diff --git a/camera/CameraParameters.cpp b/camera/CameraParameters.cpp
index c51f265..af091f4 100644
--- a/camera/CameraParameters.cpp
+++ b/camera/CameraParameters.cpp
@@ -470,7 +470,7 @@
     const size_t SIZE = 256;
     char buffer[SIZE];
     String8 result;
-    snprintf(buffer, 255, "CameraParameters::dump: mMap.size = %d\n", mMap.size());
+    snprintf(buffer, 255, "CameraParameters::dump: mMap.size = %zu\n", mMap.size());
     result.append(buffer);
     for (size_t i = 0; i < mMap.size(); i++) {
         String8 k, v;
diff --git a/cmds/screenrecord/Overlay.cpp b/cmds/screenrecord/Overlay.cpp
index 96e25b8..2e98874 100644
--- a/cmds/screenrecord/Overlay.cpp
+++ b/cmds/screenrecord/Overlay.cpp
@@ -47,7 +47,7 @@
         "ro.revision",
         "dalvik.vm.heapgrowthlimit",
         "dalvik.vm.heapsize",
-        "persist.sys.dalvik.vm.lib",
+        "persist.sys.dalvik.vm.lib.1",
         //"ro.product.cpu.abi",
         //"ro.bootloader",
         //"this-never-appears!",
diff --git a/cmds/screenrecord/screenrecord.cpp b/cmds/screenrecord/screenrecord.cpp
index 61f83e3..b6f150c 100644
--- a/cmds/screenrecord/screenrecord.cpp
+++ b/cmds/screenrecord/screenrecord.cpp
@@ -45,6 +45,7 @@
 #include <signal.h>
 #include <getopt.h>
 #include <sys/wait.h>
+#include <termios.h>
 #include <assert.h>
 
 #include "screenrecord.h"
@@ -61,6 +62,7 @@
 // Command-line parameters.
 static bool gVerbose = false;           // chatty on stdout
 static bool gRotate = false;            // rotate 90 degrees
+static bool gRawOutput = false;         // generate raw H.264 byte stream output
 static bool gSizeSpecified = false;     // was size explicitly requested?
 static bool gWantInfoScreen = false;    // do we want initial info screen?
 static bool gWantFrameTime = false;     // do we want times on each frame?
@@ -298,10 +300,12 @@
  * input frames are coming from the virtual display as fast as SurfaceFlinger
  * wants to send them.
  *
+ * Exactly one of muxer or rawFp must be non-null.
+ *
  * The muxer must *not* have been started before calling.
  */
 static status_t runEncoder(const sp<MediaCodec>& encoder,
-        const sp<MediaMuxer>& muxer, const sp<IBinder>& mainDpy,
+        const sp<MediaMuxer>& muxer, FILE* rawFp, const sp<IBinder>& mainDpy,
         const sp<IBinder>& virtualDpy, uint8_t orientation) {
     static int kTimeout = 250000;   // be responsive on signal
     status_t err;
@@ -311,6 +315,8 @@
     int64_t endWhenNsec = startWhenNsec + seconds_to_nanoseconds(gTimeLimitSec);
     DisplayInfo mainDpyInfo;
 
+    assert((rawFp == NULL && muxer != NULL) || (rawFp != NULL && muxer == NULL));
+
     Vector<sp<ABuffer> > buffers;
     err = encoder->getOutputBuffers(&buffers);
     if (err != NO_ERROR) {
@@ -342,15 +348,16 @@
         case NO_ERROR:
             // got a buffer
             if ((flags & MediaCodec::BUFFER_FLAG_CODECCONFIG) != 0) {
-                // ignore this -- we passed the CSD into MediaMuxer when
-                // we got the format change notification
-                ALOGV("Got codec config buffer (%u bytes); ignoring", size);
-                size = 0;
+                ALOGV("Got codec config buffer (%u bytes)", size);
+                if (muxer != NULL) {
+                    // ignore this -- we passed the CSD into MediaMuxer when
+                    // we got the format change notification
+                    size = 0;
+                }
             }
             if (size != 0) {
                 ALOGV("Got data in buffer %d, size=%d, pts=%lld",
                         bufIndex, size, ptsUsec);
-                assert(trackIdx != -1);
 
                 { // scope
                     ATRACE_NAME("orientation");
@@ -379,14 +386,23 @@
                     ptsUsec = systemTime(SYSTEM_TIME_MONOTONIC) / 1000;
                 }
 
-                // The MediaMuxer docs are unclear, but it appears that we
-                // need to pass either the full set of BufferInfo flags, or
-                // (flags & BUFFER_FLAG_SYNCFRAME).
-                //
-                // If this blocks for too long we could drop frames.  We may
-                // want to queue these up and do them on a different thread.
-                { // scope
+                if (muxer == NULL) {
+                    fwrite(buffers[bufIndex]->data(), 1, size, rawFp);
+                    // Flush the data immediately in case we're streaming.
+                    // We don't want to do this if all we've written is
+                    // the SPS/PPS data because mplayer gets confused.
+                    if ((flags & MediaCodec::BUFFER_FLAG_CODECCONFIG) == 0) {
+                        fflush(rawFp);
+                    }
+                } else {
+                    // The MediaMuxer docs are unclear, but it appears that we
+                    // need to pass either the full set of BufferInfo flags, or
+                    // (flags & BUFFER_FLAG_SYNCFRAME).
+                    //
+                    // If this blocks for too long we could drop frames.  We may
+                    // want to queue these up and do them on a different thread.
                     ATRACE_NAME("write sample");
+                    assert(trackIdx != -1);
                     err = muxer->writeSampleData(buffers[bufIndex], trackIdx,
                             ptsUsec, flags);
                     if (err != NO_ERROR) {
@@ -418,12 +434,14 @@
                 ALOGV("Encoder format changed");
                 sp<AMessage> newFormat;
                 encoder->getOutputFormat(&newFormat);
-                trackIdx = muxer->addTrack(newFormat);
-                ALOGV("Starting muxer");
-                err = muxer->start();
-                if (err != NO_ERROR) {
-                    fprintf(stderr, "Unable to start muxer (err=%d)\n", err);
-                    return err;
+                if (muxer != NULL) {
+                    trackIdx = muxer->addTrack(newFormat);
+                    ALOGV("Starting muxer");
+                    err = muxer->start();
+                    if (err != NO_ERROR) {
+                        fprintf(stderr, "Unable to start muxer (err=%d)\n", err);
+                        return err;
+                    }
                 }
             }
             break;
@@ -457,6 +475,44 @@
 }
 
 /*
+ * Raw H.264 byte stream output requested.  Send the output to stdout
+ * if desired.  If the output is a tty, reconfigure it to avoid the
+ * CRLF line termination that we see with "adb shell" commands.
+ */
+static FILE* prepareRawOutput(const char* fileName) {
+    FILE* rawFp = NULL;
+
+    if (strcmp(fileName, "-") == 0) {
+        if (gVerbose) {
+            fprintf(stderr, "ERROR: verbose output and '-' not compatible");
+            return NULL;
+        }
+        rawFp = stdout;
+    } else {
+        rawFp = fopen(fileName, "w");
+        if (rawFp == NULL) {
+            fprintf(stderr, "fopen raw failed: %s\n", strerror(errno));
+            return NULL;
+        }
+    }
+
+    int fd = fileno(rawFp);
+    if (isatty(fd)) {
+        // best effort -- reconfigure tty for "raw"
+        ALOGD("raw video output to tty (fd=%d)", fd);
+        struct termios term;
+        if (tcgetattr(fd, &term) == 0) {
+            cfmakeraw(&term);
+            if (tcsetattr(fd, TCSANOW, &term) == 0) {
+                ALOGD("tty successfully configured for raw");
+            }
+        }
+    }
+
+    return rawFp;
+}
+
+/*
  * Main "do work" method.
  *
  * Configures codec, muxer, and virtual display, then starts moving bits
@@ -558,16 +614,26 @@
         return err;
     }
 
-    // Configure muxer.  We have to wait for the CSD blob from the encoder
-    // before we can start it.
-    sp<MediaMuxer> muxer = new MediaMuxer(fileName,
-            MediaMuxer::OUTPUT_FORMAT_MPEG_4);
-    if (gRotate) {
-        muxer->setOrientationHint(90);  // TODO: does this do anything?
+    sp<MediaMuxer> muxer = NULL;
+    FILE* rawFp = NULL;
+    if (gRawOutput) {
+        rawFp = prepareRawOutput(fileName);
+        if (rawFp == NULL) {
+            encoder->release();
+            return -1;
+        }
+    } else {
+        // Configure muxer.  We have to wait for the CSD blob from the encoder
+        // before we can start it.
+        muxer = new MediaMuxer(fileName, MediaMuxer::OUTPUT_FORMAT_MPEG_4);
+        if (gRotate) {
+            muxer->setOrientationHint(90);  // TODO: does this do anything?
+        }
     }
 
     // Main encoder loop.
-    err = runEncoder(encoder, muxer, mainDpy, dpy, mainDpyInfo.orientation);
+    err = runEncoder(encoder, muxer, rawFp, mainDpy, dpy,
+            mainDpyInfo.orientation);
     if (err != NO_ERROR) {
         fprintf(stderr, "Encoder failed (err=%d)\n", err);
         // fall through to cleanup
@@ -584,9 +650,13 @@
         overlay->stop();
     }
     encoder->stop();
-    // If we don't stop muxer explicitly, i.e. let the destructor run,
-    // it may hang (b/11050628).
-    muxer->stop();
+    if (muxer != NULL) {
+        // If we don't stop muxer explicitly, i.e. let the destructor run,
+        // it may hang (b/11050628).
+        muxer->stop();
+    } else if (rawFp != stdout) {
+        fclose(rawFp);
+    }
     encoder->release();
 
     return err;
@@ -753,6 +823,7 @@
         { "show-frame-time",    no_argument,        NULL, 'f' },
         { "bugreport",          no_argument,        NULL, 'u' },
         { "rotate",             no_argument,        NULL, 'r' },
+        { "raw",                no_argument,        NULL, 'w' },
         { NULL,                 0,                  NULL, 0 }
     };
 
@@ -818,6 +889,10 @@
             // experimental feature
             gRotate = true;
             break;
+        case 'w':
+            // experimental feature
+            gRawOutput = true;
+            break;
         default:
             if (ic != '?') {
                 fprintf(stderr, "getopt_long returned unexpected value 0x%x\n", ic);
@@ -831,17 +906,19 @@
         return 2;
     }
 
-    // MediaMuxer tries to create the file in the constructor, but we don't
-    // learn about the failure until muxer.start(), which returns a generic
-    // error code without logging anything.  We attempt to create the file
-    // now for better diagnostics.
     const char* fileName = argv[optind];
-    int fd = open(fileName, O_CREAT | O_RDWR, 0644);
-    if (fd < 0) {
-        fprintf(stderr, "Unable to open '%s': %s\n", fileName, strerror(errno));
-        return 1;
+    if (!gRawOutput) {
+        // MediaMuxer tries to create the file in the constructor, but we don't
+        // learn about the failure until muxer.start(), which returns a generic
+        // error code without logging anything.  We attempt to create the file
+        // now for better diagnostics.
+        int fd = open(fileName, O_CREAT | O_RDWR, 0644);
+        if (fd < 0) {
+            fprintf(stderr, "Unable to open '%s': %s\n", fileName, strerror(errno));
+            return 1;
+        }
+        close(fd);
     }
-    close(fd);
 
     status_t err = recordScreen(fileName);
     if (err == NO_ERROR) {
diff --git a/cmds/screenrecord/screenrecord.h b/cmds/screenrecord/screenrecord.h
index 95e8a68..9b058c2 100644
--- a/cmds/screenrecord/screenrecord.h
+++ b/cmds/screenrecord/screenrecord.h
@@ -18,6 +18,6 @@
 #define SCREENRECORD_SCREENRECORD_H
 
 #define kVersionMajor 1
-#define kVersionMinor 1
+#define kVersionMinor 2
 
 #endif /*SCREENRECORD_SCREENRECORD_H*/
diff --git a/cmds/stagefright/SimplePlayer.cpp b/cmds/stagefright/SimplePlayer.cpp
index 5d2d721..1b2f792 100644
--- a/cmds/stagefright/SimplePlayer.cpp
+++ b/cmds/stagefright/SimplePlayer.cpp
@@ -23,6 +23,7 @@
 #include <gui/Surface.h>
 #include <media/AudioTrack.h>
 #include <media/ICrypto.h>
+#include <media/IMediaHTTPService.h>
 #include <media/stagefright/foundation/ABuffer.h>
 #include <media/stagefright/foundation/ADebug.h>
 #include <media/stagefright/foundation/AMessage.h>
@@ -275,7 +276,8 @@
 
     mExtractor = new NuMediaExtractor;
 
-    status_t err = mExtractor->setDataSource(mPath.c_str());
+    status_t err = mExtractor->setDataSource(
+            NULL /* httpService */, mPath.c_str());
 
     if (err != OK) {
         mExtractor.clear();
diff --git a/cmds/stagefright/SineSource.cpp b/cmds/stagefright/SineSource.cpp
index 14b4306..587077a 100644
--- a/cmds/stagefright/SineSource.cpp
+++ b/cmds/stagefright/SineSource.cpp
@@ -24,7 +24,7 @@
     }
 }
 
-status_t SineSource::start(MetaData *params) {
+status_t SineSource::start(MetaData * /* params */) {
     CHECK(!mStarted);
 
     mGroup = new MediaBufferGroup;
@@ -58,7 +58,7 @@
 }
 
 status_t SineSource::read(
-        MediaBuffer **out, const ReadOptions *options) {
+        MediaBuffer **out, const ReadOptions * /* options */) {
     *out = NULL;
 
     MediaBuffer *buffer;
diff --git a/cmds/stagefright/codec.cpp b/cmds/stagefright/codec.cpp
index fdfefdf..fd02bcc 100644
--- a/cmds/stagefright/codec.cpp
+++ b/cmds/stagefright/codec.cpp
@@ -16,6 +16,7 @@
 
 //#define LOG_NDEBUG 0
 #define LOG_TAG "codec"
+#include <inttypes.h>
 #include <utils/Log.h>
 
 #include "SimplePlayer.h"
@@ -23,6 +24,7 @@
 #include <binder/IServiceManager.h>
 #include <binder/ProcessState.h>
 #include <media/ICrypto.h>
+#include <media/IMediaHTTPService.h>
 #include <media/IMediaPlayerService.h>
 #include <media/stagefright/foundation/ABuffer.h>
 #include <media/stagefright/foundation/ADebug.h>
@@ -75,7 +77,7 @@
     static int64_t kTimeout = 500ll;
 
     sp<NuMediaExtractor> extractor = new NuMediaExtractor;
-    if (extractor->setDataSource(path) != OK) {
+    if (extractor->setDataSource(NULL /* httpService */, path) != OK) {
         fprintf(stderr, "unable to instantiate extractor.\n");
         return 1;
     }
@@ -291,13 +293,13 @@
         CHECK_EQ((status_t)OK, state->mCodec->release());
 
         if (state->mIsAudio) {
-            printf("track %d: %lld bytes received. %.2f KB/sec\n",
+            printf("track %zu: %" PRId64 " bytes received. %.2f KB/sec\n",
                    i,
                    state->mNumBytesDecoded,
                    state->mNumBytesDecoded * 1E6 / 1024 / elapsedTimeUs);
         } else {
-            printf("track %d: %lld frames decoded, %.2f fps. %lld bytes "
-                   "received. %.2f KB/sec\n",
+            printf("track %zu: %" PRId64 " frames decoded, %.2f fps. %" PRId64
+                    " bytes received. %.2f KB/sec\n",
                    i,
                    state->mNumBuffersDecoded,
                    state->mNumBuffersDecoded * 1E6 / elapsedTimeUs,
diff --git a/cmds/stagefright/muxer.cpp b/cmds/stagefright/muxer.cpp
index cca33e0..f4a33e8 100644
--- a/cmds/stagefright/muxer.cpp
+++ b/cmds/stagefright/muxer.cpp
@@ -16,9 +16,11 @@
 
 //#define LOG_NDEBUG 0
 #define LOG_TAG "muxer"
+#include <inttypes.h>
 #include <utils/Log.h>
 
 #include <binder/ProcessState.h>
+#include <media/IMediaHTTPService.h>
 #include <media/stagefright/foundation/ABuffer.h>
 #include <media/stagefright/foundation/ADebug.h>
 #include <media/stagefright/foundation/ALooper.h>
@@ -58,7 +60,7 @@
         int trimEndTimeMs,
         int rotationDegrees) {
     sp<NuMediaExtractor> extractor = new NuMediaExtractor;
-    if (extractor->setDataSource(path) != OK) {
+    if (extractor->setDataSource(NULL /* httpService */, path) != OK) {
         fprintf(stderr, "unable to instantiate extractor. %s\n", path);
         return 1;
     }
@@ -198,7 +200,7 @@
     trackIndexMap.clear();
 
     int64_t elapsedTimeUs = ALooper::GetNowUs() - muxerStartTimeUs;
-    fprintf(stderr, "SUCCESS: muxer generate the video in %lld ms\n",
+    fprintf(stderr, "SUCCESS: muxer generate the video in %" PRId64 " ms\n",
             elapsedTimeUs / 1000);
 
     return 0;
diff --git a/cmds/stagefright/record.cpp b/cmds/stagefright/record.cpp
index b7a40c2..fdc352e 100644
--- a/cmds/stagefright/record.cpp
+++ b/cmds/stagefright/record.cpp
@@ -296,7 +296,7 @@
 }
 #else
 
-int main(int argc, char **argv) {
+int main(int /* argc */, char ** /* argv */) {
     android::ProcessState::self()->startThreadPool();
 
     OMXClient client;
diff --git a/cmds/stagefright/recordvideo.cpp b/cmds/stagefright/recordvideo.cpp
index c30c122..1d267f9 100644
--- a/cmds/stagefright/recordvideo.cpp
+++ b/cmds/stagefright/recordvideo.cpp
@@ -16,6 +16,7 @@
 
 #include "SineSource.h"
 
+#include <inttypes.h>
 #include <binder/ProcessState.h>
 #include <media/stagefright/foundation/ADebug.h>
 #include <media/stagefright/AudioPlayer.h>
@@ -312,7 +313,7 @@
         fprintf(stderr, "record failed: %d\n", err);
         return 1;
     }
-    fprintf(stderr, "encoding %d frames in %lld us\n", nFrames, (end-start)/1000);
+    fprintf(stderr, "encoding %d frames in %" PRId64 " us\n", nFrames, (end-start)/1000);
     fprintf(stderr, "encoding speed is: %.2f fps\n", (nFrames * 1E9) / (end-start));
     return 0;
 }
diff --git a/cmds/stagefright/sf2.cpp b/cmds/stagefright/sf2.cpp
index 439b6e4..3c0c7ec 100644
--- a/cmds/stagefright/sf2.cpp
+++ b/cmds/stagefright/sf2.cpp
@@ -16,12 +16,15 @@
 
 //#define LOG_NDEBUG 0
 #define LOG_TAG "sf2"
+#include <inttypes.h>
 #include <utils/Log.h>
 
 #include <signal.h>
 
 #include <binder/ProcessState.h>
 
+#include <media/IMediaHTTPService.h>
+
 #include <media/stagefright/foundation/hexdump.h>
 #include <media/stagefright/foundation/ABuffer.h>
 #include <media/stagefright/foundation/ADebug.h>
@@ -78,17 +81,18 @@
 
     virtual void printStatistics() {
         int64_t delayUs = ALooper::GetNowUs() - mStartTimeUs;
+
         if (mDecodeAudio) {
-            printf("%lld bytes received. %.2f KB/sec\n",
-                   mTotalBytesReceived,
-                   mTotalBytesReceived * 1E6 / 1024 / delayUs);
+            printf("%" PRId64 " bytes received. %.2f KB/sec\n",
+            mTotalBytesReceived,
+            mTotalBytesReceived * 1E6 / 1024 / delayUs);
         } else {
-            printf("%d frames decoded, %.2f fps. %lld bytes "
-                   "received. %.2f KB/sec\n",
-                   mNumOutputBuffersReceived,
-                   mNumOutputBuffersReceived * 1E6 / delayUs,
-                   mTotalBytesReceived,
-                   mTotalBytesReceived * 1E6 / 1024 / delayUs);
+            printf("%d frames decoded, %.2f fps. %" PRId64 " bytes "
+                    "received. %.2f KB/sec\n",
+            mNumOutputBuffersReceived,
+            mNumOutputBuffersReceived * 1E6 / delayUs,
+            mTotalBytesReceived,
+            mTotalBytesReceived * 1E6 / 1024 / delayUs);
         }
     }
 
@@ -111,7 +115,8 @@
 #endif
 
                 sp<DataSource> dataSource =
-                    DataSource::CreateFromURI(mURI.c_str());
+                    DataSource::CreateFromURI(
+                            NULL /* httpService */, mURI.c_str());
 
                 sp<MediaExtractor> extractor =
                     MediaExtractor::Create(dataSource);
diff --git a/cmds/stagefright/stagefright.cpp b/cmds/stagefright/stagefright.cpp
index 030bf1b..daaea27 100644
--- a/cmds/stagefright/stagefright.cpp
+++ b/cmds/stagefright/stagefright.cpp
@@ -22,12 +22,14 @@
 
 #include <stdlib.h>
 #include <string.h>
+#include <inttypes.h>
 
 #include "jpeg.h"
 #include "SineSource.h"
 
 #include <binder/IServiceManager.h>
 #include <binder/ProcessState.h>
+#include <media/IMediaHTTPService.h>
 #include <media/IMediaPlayerService.h>
 #include <media/stagefright/foundation/ALooper.h>
 #include "include/NuCachedSource2.h"
@@ -89,8 +91,8 @@
     int64_t minUs = decodeTimesUs->itemAt(0);
     int64_t maxUs = decodeTimesUs->itemAt(n - 1);
 
-    printf("min decode time %lld us (%.2f secs)\n", minUs, minUs / 1E6);
-    printf("max decode time %lld us (%.2f secs)\n", maxUs, maxUs / 1E6);
+    printf("min decode time %" PRId64 " us (%.2f secs)\n", minUs, minUs / 1E6);
+    printf("max decode time %" PRId64 " us (%.2f secs)\n", maxUs, maxUs / 1E6);
 
     size_t counts[100];
     for (size_t i = 0; i < 100; ++i) {
@@ -110,7 +112,7 @@
         int64_t slotUs = minUs + (i * (maxUs - minUs) / 100);
 
         double fps = 1E6 / slotUs;
-        printf("[%.2f fps]: %d\n", fps, counts[i]);
+        printf("[%.2f fps]: %zu\n", fps, counts[i]);
     }
 }
 
@@ -262,7 +264,7 @@
                     }
                 }
 
-                printf("buffer has timestamp %lld us (%.2f secs)\n",
+                printf("buffer has timestamp %" PRId64 " us (%.2f secs)\n",
                        timestampUs, timestampUs / 1E6);
 
                 buffer->release();
@@ -285,7 +287,7 @@
                 seekTimeUs = (rand() * (float)durationUs) / RAND_MAX;
                 options.setSeekTo(seekTimeUs);
 
-                printf("seeking to %lld us (%.2f secs)\n",
+                printf("seeking to %" PRId64 " us (%.2f secs)\n",
                        seekTimeUs, seekTimeUs / 1E6);
             }
         }
@@ -388,7 +390,7 @@
         // sizes may be different across decoders.
         printf("avg. %.2f KB/sec\n", totalBytes / 1024 * 1E6 / delay);
 
-        printf("decoded a total of %lld bytes\n", totalBytes);
+        printf("decoded a total of %" PRId64 " bytes\n", totalBytes);
     }
 }
 
@@ -574,7 +576,8 @@
             int64_t timeUs;
             CHECK(buffer->meta_data()->findInt64(kKeyTime, &timeUs));
 
-            printf("%lld\t%lld\t%lld\n", seekTimeUs, timeUs, seekTimeUs - timeUs);
+            printf("%" PRId64 "\t%" PRId64 "\t%" PRId64 "\n",
+                   seekTimeUs, timeUs, seekTimeUs - timeUs);
 
             buffer->release();
             buffer = NULL;
@@ -958,7 +961,8 @@
 
         const char *filename = argv[k];
 
-        sp<DataSource> dataSource = DataSource::CreateFromURI(filename);
+        sp<DataSource> dataSource =
+            DataSource::CreateFromURI(NULL /* httpService */, filename);
 
         if (strncasecmp(filename, "sine:", 5) && dataSource == NULL) {
             fprintf(stderr, "Unable to create data source.\n");
@@ -1071,7 +1075,7 @@
 
                 int64_t thumbTimeUs;
                 if (meta->findInt64(kKeyThumbnailTime, &thumbTimeUs)) {
-                    printf("thumbnailTime: %lld us (%.2f secs)\n",
+                    printf("thumbnailTime: %" PRId64 " us (%.2f secs)\n",
                            thumbTimeUs, thumbTimeUs / 1E6);
                 }
 
diff --git a/cmds/stagefright/stream.cpp b/cmds/stagefright/stream.cpp
index dba67a9..b2abc0f 100644
--- a/cmds/stagefright/stream.cpp
+++ b/cmds/stagefright/stream.cpp
@@ -21,6 +21,7 @@
 #include <binder/ProcessState.h>
 #include <cutils/properties.h> // for property_get
 
+#include <media/IMediaHTTPService.h>
 #include <media/IStreamSource.h>
 #include <media/mediaplayer.h>
 #include <media/stagefright/foundation/ADebug.h>
@@ -159,7 +160,9 @@
 MyConvertingStreamSource::MyConvertingStreamSource(const char *filename)
     : mCurrentBufferIndex(-1),
       mCurrentBufferOffset(0) {
-    sp<DataSource> dataSource = DataSource::CreateFromURI(filename);
+    sp<DataSource> dataSource =
+        DataSource::CreateFromURI(NULL /* httpService */, filename);
+
     CHECK(dataSource != NULL);
 
     sp<MediaExtractor> extractor = MediaExtractor::Create(dataSource);
diff --git a/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/Android.mk b/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/Android.mk
index e251f82..48b0afe 100644
--- a/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/Android.mk
+++ b/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/Android.mk
@@ -61,7 +61,7 @@
     $(LOCAL_PATH)/include \
     external/openssl/include
 
-LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/drm
+LOCAL_MODULE_RELATIVE_PATH := drm
 
 LOCAL_MODULE_TAGS := optional
 
diff --git a/drm/mediadrm/plugins/mock/MockDrmCryptoPlugin.cpp b/drm/mediadrm/plugins/mock/MockDrmCryptoPlugin.cpp
index f2cadf7..69fa7a0fc 100644
--- a/drm/mediadrm/plugins/mock/MockDrmCryptoPlugin.cpp
+++ b/drm/mediadrm/plugins/mock/MockDrmCryptoPlugin.cpp
@@ -695,7 +695,7 @@
     {
         String8 result;
         for (size_t i = 0; i < numSubSamples; i++) {
-            result.appendFormat("[%d] {clear:%d, encrypted:%d} ", i,
+            result.appendFormat("[%zu] {clear:%zu, encrypted:%zu} ", i,
                                 subSamples[i].mNumBytesOfClearData,
                                 subSamples[i].mNumBytesOfEncryptedData);
         }
diff --git a/include/media/AudioEffect.h b/include/media/AudioEffect.h
index 05d834d..f3024b7 100644
--- a/include/media/AudioEffect.h
+++ b/include/media/AudioEffect.h
@@ -36,7 +36,7 @@
 
 // ----------------------------------------------------------------------------
 
-class effect_param_cblk_t;
+struct effect_param_cblk_t;
 
 // ----------------------------------------------------------------------------
 
diff --git a/include/media/AudioRecord.h b/include/media/AudioRecord.h
index b2e4518..7054fd4 100644
--- a/include/media/AudioRecord.h
+++ b/include/media/AudioRecord.h
@@ -26,7 +26,7 @@
 
 // ----------------------------------------------------------------------------
 
-class audio_track_cblk_t;
+struct audio_track_cblk_t;
 class AudioRecordClientProxy;
 
 // ----------------------------------------------------------------------------
@@ -39,8 +39,12 @@
      * Keep in sync with frameworks/base/media/java/android/media/AudioRecord.java NATIVE_EVENT_*.
      */
     enum event_type {
-        EVENT_MORE_DATA = 0,        // Request to read more data from PCM buffer.
-        EVENT_OVERRUN = 1,          // PCM buffer overrun occurred.
+        EVENT_MORE_DATA = 0,        // Request to read available data from buffer.
+                                    // If this event is delivered but the callback handler
+                                    // does not want to read the available data, the handler must
+                                    // explicitly
+                                    // ignore the event by setting frameCount to zero.
+        EVENT_OVERRUN = 1,          // Buffer overrun occurred.
         EVENT_MARKER = 2,           // Record head is at the specified marker position
                                     // (See setMarkerPosition()).
         EVENT_NEW_POS = 3,          // Record head is at a new position
@@ -63,6 +67,7 @@
                                     // (currently ignored but will make the primary field in future)
 
         size_t      size;           // input/output in bytes == frameCount * frameSize
+                                    // on output is the number of bytes actually drained
                                     // FIXME this is redundant with respect to frameCount,
                                     // and TRANSFER_OBTAIN mode is broken for 8-bit data
                                     // since we don't define the frame format
@@ -76,7 +81,7 @@
 
     /* As a convenience, if a callback is supplied, a handler thread
      * is automatically created with the appropriate priority. This thread
-     * invokes the callback when a new buffer becomes ready or various conditions occur.
+     * invokes the callback when a new buffer becomes available or various conditions occur.
      * Parameters:
      *
      * event:   type of event notified (see enum AudioRecord::event_type).
@@ -99,6 +104,8 @@
      *  - NO_ERROR: successful operation
      *  - NO_INIT: audio server or audio hardware not initialized
      *  - BAD_VALUE: unsupported configuration
+     * frameCount is guaranteed to be non-zero if status is NO_ERROR,
+     * and is undefined otherwise.
      */
 
      static status_t getMinFrameCount(size_t* frameCount,
@@ -109,7 +116,7 @@
     /* How data is transferred from AudioRecord
      */
     enum transfer_type {
-        TRANSFER_DEFAULT,   // not specified explicitly; determine from other parameters
+        TRANSFER_DEFAULT,   // not specified explicitly; determine from the other parameters
         TRANSFER_CALLBACK,  // callback EVENT_MORE_DATA
         TRANSFER_OBTAIN,    // FIXME deprecated: call obtainBuffer() and releaseBuffer()
         TRANSFER_SYNC,      // synchronous read()
@@ -137,7 +144,7 @@
      *                     be larger if the requested size is not compatible with current audio HAL
      *                     latency.  Zero means to use a default value.
      * cbf:                Callback function. If not null, this function is called periodically
-     *                     to consume new PCM data and inform of marker, position updates, etc.
+     *                     to consume new data and inform of marker, position updates, etc.
      * user:               Context for use by the callback receiver.
      * notificationFrames: The callback function is called each time notificationFrames PCM
      *                     frames are ready in record track output buffer.
@@ -155,7 +162,7 @@
                                     callback_t cbf = NULL,
                                     void* user = NULL,
                                     int notificationFrames = 0,
-                                    int sessionId = 0,
+                                    int sessionId = AUDIO_SESSION_ALLOCATE,
                                     transfer_type transferType = TRANSFER_DEFAULT,
                                     audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE);
 
@@ -171,9 +178,10 @@
      * Returned status (from utils/Errors.h) can be:
      *  - NO_ERROR: successful intialization
      *  - INVALID_OPERATION: AudioRecord is already initialized or record device is already in use
-     *  - BAD_VALUE: invalid parameter (channels, format, sampleRate...)
+     *  - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...)
      *  - NO_INIT: audio server or audio hardware not initialized
      *  - PERMISSION_DENIED: recording is not allowed for the requesting process
+     * If status is not equal to NO_ERROR, don't call any other APIs on this AudioRecord.
      *
      * Parameters not listed in the AudioRecord constructors above:
      *
@@ -188,11 +196,11 @@
                             void* user = NULL,
                             int notificationFrames = 0,
                             bool threadCanCallJava = false,
-                            int sessionId = 0,
+                            int sessionId = AUDIO_SESSION_ALLOCATE,
                             transfer_type transferType = TRANSFER_DEFAULT,
                             audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE);
 
-    /* Result of constructing the AudioRecord. This must be checked
+    /* Result of constructing the AudioRecord. This must be checked for successful initialization
      * before using any AudioRecord API (except for set()), because using
      * an uninitialized AudioRecord produces undefined results.
      * See set() method above for possible return codes.
@@ -221,7 +229,7 @@
             status_t    start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE,
                               int triggerSession = 0);
 
-    /* Stop a track. If set, the callback will cease being called.  Note that obtainBuffer() still
+    /* Stop a track.  The callback will cease being called.  Note that obtainBuffer() still
      * works and will drain buffers until the pool is exhausted, and then will return WOULD_BLOCK.
      */
             void        stop();
@@ -236,7 +244,7 @@
      * a callback with event type EVENT_MARKER is called. Calling setMarkerPosition
      * with marker == 0 cancels marker notification callback.
      * To set a marker at a position which would compute as 0,
-     * a workaround is to the set the marker at a nearby position such as ~0 or 1.
+     * a workaround is to set the marker at a nearby position such as ~0 or 1.
      * If the AudioRecord has been opened with no callback function associated,
      * the operation will fail.
      *
@@ -378,8 +386,10 @@
      * returning the current value by this function call.  Such loss typically occurs when the
      * user space process is blocked longer than the capacity of audio driver buffers.
      * Units: the number of input audio frames.
+     * FIXME The side-effect of resetting the counter may be incompatible with multi-client.
+     * Consider making it more like AudioTrack::getUnderrunFrames which doesn't have side effects.
      */
-            unsigned int  getInputFramesLost() const;
+            uint32_t    getInputFramesLost() const;
 
 private:
     /* copying audio record objects is not allowed */
@@ -412,6 +422,7 @@
         bool                mPaused;    // whether thread is requested to pause at next loop entry
         bool                mPausedInt; // whether thread internally requests pause
         nsecs_t             mPausedNs;  // if mPausedInt then associated timeout, otherwise ignored
+        bool                mIgnoreNextPausedInt;   // whether to ignore next mPausedInt request
     };
 
             // body of AudioRecordThread::threadLoop()
@@ -425,6 +436,7 @@
             nsecs_t processAudioBuffer();
 
             // caller must hold lock on mLock for all _l methods
+
             status_t openRecord_l(size_t epoch);
 
             // FIXME enum is faster than strcmp() for parameter 'from'
@@ -452,7 +464,7 @@
     // These are private to processAudioBuffer(), and are not protected by a lock
     uint32_t                mRemainingFrames;       // number of frames to request in obtainBuffer()
     bool                    mRetryOnPartialBuffer;  // sleep and retry after partial obtainBuffer()
-    int                     mObservedSequence;      // last observed value of mSequence
+    uint32_t                mObservedSequence;      // last observed value of mSequence
 
     uint32_t                mMarkerPosition;    // in wrapping (overflow) frame units
     bool                    mMarkerReached;
@@ -461,9 +473,13 @@
 
     status_t                mStatus;
 
+    size_t                  mFrameCount;            // corresponds to current IAudioRecord, value is
+                                                    // reported back by AudioFlinger to the client
+    size_t                  mReqFrameCount;         // frame count to request the first or next time
+                                                    // a new IAudioRecord is needed, non-decreasing
+
     // constant after constructor or set()
     uint32_t                mSampleRate;
-    size_t                  mFrameCount;
     audio_format_t          mFormat;
     uint32_t                mChannelCount;
     size_t                  mFrameSize;         // app-level frame size == AudioFlinger frame size
@@ -474,12 +490,11 @@
     int                     mSessionId;
     transfer_type           mTransfer;
 
-    audio_io_handle_t       mInput;             // returned by AudioSystem::getInput()
-
-    // may be changed if IAudioRecord object is re-created
+    // Next 4 fields may be changed if IAudioRecord is re-created, but always != 0
     sp<IAudioRecord>        mAudioRecord;
     sp<IMemory>             mCblkMemory;
     audio_track_cblk_t*     mCblk;              // re-load after mLock.unlock()
+    audio_io_handle_t       mInput;             // returned by AudioSystem::getInput()
 
     int                     mPreviousPriority;  // before start()
     SchedPolicy             mPreviousSchedulingGroup;
diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
index ba29e26..fd86737 100644
--- a/include/media/AudioSystem.h
+++ b/include/media/AudioSystem.h
@@ -67,12 +67,12 @@
 
     // returns true in *state if tracks are active on the specified stream or have been active
     // in the past inPastMs milliseconds
-    static status_t isStreamActive(audio_stream_type_t stream, bool *state, uint32_t inPastMs = 0);
+    static status_t isStreamActive(audio_stream_type_t stream, bool *state, uint32_t inPastMs);
     // returns true in *state if tracks are active for what qualifies as remote playback
     // on the specified stream or have been active in the past inPastMs milliseconds. Remote
     // playback isn't mutually exclusive with local playback.
     static status_t isStreamActiveRemotely(audio_stream_type_t stream, bool *state,
-            uint32_t inPastMs = 0);
+            uint32_t inPastMs);
     // returns true in *state if a recorder is currently recording with the specified source
     static status_t isSourceActive(audio_source_t source, bool *state);
 
@@ -94,12 +94,14 @@
     static float linearToLog(int volume);
     static int logToLinear(float volume);
 
+    // Returned samplingRate and frameCount output values are guaranteed
+    // to be non-zero if status == NO_ERROR
     static status_t getOutputSamplingRate(uint32_t* samplingRate,
-            audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
+            audio_stream_type_t stream);
     static status_t getOutputFrameCount(size_t* frameCount,
-            audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
+            audio_stream_type_t stream);
     static status_t getOutputLatency(uint32_t* latency,
-            audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
+            audio_stream_type_t stream);
     static status_t getSamplingRate(audio_io_handle_t output,
                                           audio_stream_type_t streamType,
                                           uint32_t* samplingRate);
@@ -131,16 +133,16 @@
     // NOTE: this feature is not supported on all hardware platforms and it is
     // necessary to check returned status before using the returned values.
     static status_t getRenderPosition(audio_io_handle_t output,
-                                      size_t *halFrames,
-                                      size_t *dspFrames,
+                                      uint32_t *halFrames,
+                                      uint32_t *dspFrames,
                                       audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
 
     // return the number of input frames lost by HAL implementation, or 0 if the handle is invalid
-    static size_t getInputFramesLost(audio_io_handle_t ioHandle);
+    static uint32_t getInputFramesLost(audio_io_handle_t ioHandle);
 
     static int newAudioSessionId();
-    static void acquireAudioSessionId(int audioSession);
-    static void releaseAudioSessionId(int audioSession);
+    static void acquireAudioSessionId(int audioSession, pid_t pid);
+    static void releaseAudioSessionId(int audioSession, pid_t pid);
 
     // types of io configuration change events received with ioConfigChanged()
     enum io_config_event {
@@ -198,24 +200,32 @@
     static status_t setPhoneState(audio_mode_t state);
     static status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config);
     static audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage);
+
+    // Client must successfully hand off the handle reference to AudioFlinger via createTrack(),
+    // or release it with releaseOutput().
     static audio_io_handle_t getOutput(audio_stream_type_t stream,
                                         uint32_t samplingRate = 0,
                                         audio_format_t format = AUDIO_FORMAT_DEFAULT,
                                         audio_channel_mask_t channelMask = AUDIO_CHANNEL_OUT_STEREO,
                                         audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
                                         const audio_offload_info_t *offloadInfo = NULL);
+
     static status_t startOutput(audio_io_handle_t output,
                                 audio_stream_type_t stream,
-                                int session = 0);
+                                int session);
     static status_t stopOutput(audio_io_handle_t output,
                                audio_stream_type_t stream,
-                               int session = 0);
+                               int session);
     static void releaseOutput(audio_io_handle_t output);
+
+    // Client must successfully hand off the handle reference to AudioFlinger via openRecord(),
+    // or release it with releaseInput().
     static audio_io_handle_t getInput(audio_source_t inputSource,
-                                    uint32_t samplingRate = 0,
-                                    audio_format_t format = AUDIO_FORMAT_DEFAULT,
-                                    audio_channel_mask_t channelMask = AUDIO_CHANNEL_IN_MONO,
-                                    int sessionId = 0);
+                                    uint32_t samplingRate,
+                                    audio_format_t format,
+                                    audio_channel_mask_t channelMask,
+                                    int sessionId);
+
     static status_t startInput(audio_io_handle_t input);
     static status_t stopInput(audio_io_handle_t input);
     static void releaseInput(audio_io_handle_t input);
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index 9f631cf..d0df710 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -27,7 +27,7 @@
 
 // ----------------------------------------------------------------------------
 
-class audio_track_cblk_t;
+struct audio_track_cblk_t;
 class AudioTrackClientProxy;
 class StaticAudioTrackClientProxy;
 
@@ -123,6 +123,8 @@
      *  - NO_ERROR: successful operation
      *  - NO_INIT: audio server or audio hardware not initialized
      *  - BAD_VALUE: unsupported configuration
+     * frameCount is guaranteed to be non-zero if status is NO_ERROR,
+     * and is undefined otherwise.
      */
 
     static status_t getMinFrameCount(size_t* frameCount,
@@ -158,7 +160,7 @@
      * sampleRate:         Data source sampling rate in Hz.
      * format:             Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed
      *                     16 bits per sample).
-     * channelMask:        Channel mask.
+     * channelMask:        Channel mask, such that audio_is_output_channel(channelMask) is true.
      * frameCount:         Minimum size of track PCM buffer in frames. This defines the
      *                     application's contribution to the
      *                     latency of the track. The actual size selected by the AudioTrack could be
@@ -185,10 +187,11 @@
                                     callback_t cbf       = NULL,
                                     void* user           = NULL,
                                     int notificationFrames = 0,
-                                    int sessionId        = 0,
+                                    int sessionId        = AUDIO_SESSION_ALLOCATE,
                                     transfer_type transferType = TRANSFER_DEFAULT,
                                     const audio_offload_info_t *offloadInfo = NULL,
-                                    int uid = -1);
+                                    int uid = -1,
+                                    pid_t pid = -1);
 
     /* Creates an audio track and registers it with AudioFlinger.
      * With this constructor, the track is configured for static buffer mode.
@@ -210,10 +213,11 @@
                                     callback_t cbf      = NULL,
                                     void* user          = NULL,
                                     int notificationFrames = 0,
-                                    int sessionId       = 0,
+                                    int sessionId       = AUDIO_SESSION_ALLOCATE,
                                     transfer_type transferType = TRANSFER_DEFAULT,
                                     const audio_offload_info_t *offloadInfo = NULL,
-                                    int uid = -1);
+                                    int uid = -1,
+                                    pid_t pid = -1);
 
     /* Terminates the AudioTrack and unregisters it from AudioFlinger.
      * Also destroys all resources associated with the AudioTrack.
@@ -248,10 +252,11 @@
                             int notificationFrames = 0,
                             const sp<IMemory>& sharedBuffer = 0,
                             bool threadCanCallJava = false,
-                            int sessionId       = 0,
+                            int sessionId       = AUDIO_SESSION_ALLOCATE,
                             transfer_type transferType = TRANSFER_DEFAULT,
                             const audio_offload_info_t *offloadInfo = NULL,
-                            int uid = -1);
+                            int uid = -1,
+                            pid_t pid = -1);
 
     /* Result of constructing the AudioTrack. This must be checked for successful initialization
      * before using any AudioTrack API (except for set()), because using
@@ -336,7 +341,7 @@
      */
             status_t    setSampleRate(uint32_t sampleRate);
 
-    /* Return current source sample rate in Hz, or 0 if unknown */
+    /* Return current source sample rate in Hz */
             uint32_t    getSampleRate() const;
 
     /* Enables looping and sets the start and end points of looping.
@@ -361,7 +366,7 @@
     /* Sets marker position. When playback reaches the number of frames specified, a callback with
      * event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker
      * notification callback.  To set a marker at a position which would compute as 0,
-     * a workaround is to the set the marker at a nearby position such as ~0 or 1.
+     * a workaround is to set the marker at a nearby position such as ~0 or 1.
      * If the AudioTrack has been opened with no callback function associated, the operation will
      * fail.
      *
@@ -452,7 +457,7 @@
      * Returned value:
      *  handle on audio hardware output
      */
-            audio_io_handle_t    getOutput();
+            audio_io_handle_t    getOutput() const;
 
     /* Returns the unique session ID associated with this track.
      *
@@ -566,7 +571,7 @@
             uint32_t    getUnderrunFrames() const;
 
     /* Get the flags */
-            audio_output_flags_t getFlags() const { return mFlags; }
+            audio_output_flags_t getFlags() const { AutoMutex _l(mLock); return mFlags; }
 
     /* Set parameters - only possible when using direct output */
             status_t    setParameters(const String8& keyValuePairs);
@@ -627,51 +632,49 @@
             //      NS_NEVER    never again
             static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3;
             nsecs_t processAudioBuffer();
-            status_t processStreamEnd(int32_t waitCount);
 
+            bool     isOffloaded() const;
 
             // caller must hold lock on mLock for all _l methods
 
-            status_t createTrack_l(audio_stream_type_t streamType,
-                                 uint32_t sampleRate,
-                                 audio_format_t format,
-                                 size_t frameCount,
-                                 audio_output_flags_t flags,
-                                 const sp<IMemory>& sharedBuffer,
-                                 audio_io_handle_t output,
-                                 size_t epoch);
+            status_t createTrack_l(size_t epoch);
 
             // can only be called when mState != STATE_ACTIVE
             void flush_l();
 
             void setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount);
-            audio_io_handle_t getOutput_l();
 
             // FIXME enum is faster than strcmp() for parameter 'from'
             status_t restoreTrack_l(const char *from);
 
-            bool     isOffloaded() const
+            bool     isOffloaded_l() const
                 { return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; }
 
-    // Next 3 fields may be changed if IAudioTrack is re-created, but always != 0
+    // Next 4 fields may be changed if IAudioTrack is re-created, but always != 0
     sp<IAudioTrack>         mAudioTrack;
     sp<IMemory>             mCblkMemory;
     audio_track_cblk_t*     mCblk;                  // re-load after mLock.unlock()
+    audio_io_handle_t       mOutput;                // returned by AudioSystem::getOutput()
 
     sp<AudioTrackThread>    mAudioTrackThread;
+
     float                   mVolume[2];
     float                   mSendLevel;
     mutable uint32_t        mSampleRate;            // mutable because getSampleRate() can update it.
-    size_t                  mFrameCount;            // corresponds to current IAudioTrack
-    size_t                  mReqFrameCount;         // frame count to request the next time a new
-                                                    // IAudioTrack is needed
+    size_t                  mFrameCount;            // corresponds to current IAudioTrack, value is
+                                                    // reported back by AudioFlinger to the client
+    size_t                  mReqFrameCount;         // frame count to request the first or next time
+                                                    // a new IAudioTrack is needed, non-decreasing
 
     // constant after constructor or set()
     audio_format_t          mFormat;                // as requested by client, not forced to 16-bit
     audio_stream_type_t     mStreamType;
     uint32_t                mChannelCount;
     audio_channel_mask_t    mChannelMask;
+    sp<IMemory>             mSharedBuffer;
     transfer_type           mTransfer;
+    audio_offload_info_t    mOffloadInfoCopy;
+    const audio_offload_info_t* mOffloadInfo;
 
     // mFrameSize is equal to mFrameSizeAF for non-PCM or 16-bit PCM data.  For 8-bit PCM data, it's
     // twice as large as mFrameSize because data is expanded to 16-bit before it's stored in buffer.
@@ -712,14 +715,17 @@
     bool                    mRetryOnPartialBuffer;  // sleep and retry after partial obtainBuffer()
     uint32_t                mObservedSequence;      // last observed value of mSequence
 
-    sp<IMemory>             mSharedBuffer;
     uint32_t                mLoopPeriod;            // in frames, zero means looping is disabled
+
     uint32_t                mMarkerPosition;        // in wrapping (overflow) frame units
     bool                    mMarkerReached;
     uint32_t                mNewPosition;           // in frames
     uint32_t                mUpdatePeriod;          // in frames, zero means no EVENT_NEW_POS
 
     audio_output_flags_t    mFlags;
+        // const after set(), except for bits AUDIO_OUTPUT_FLAG_FAST and AUDIO_OUTPUT_FLAG_OFFLOAD.
+        // mLock must be held to read or write those bits reliably.
+
     int                     mSessionId;
     int                     mAuxEffectId;
 
@@ -753,8 +759,8 @@
 
     sp<DeathNotifier>       mDeathNotifier;
     uint32_t                mSequence;              // incremented for each new IAudioTrack attempt
-    audio_io_handle_t       mOutput;                // cached output io handle
     int                     mClientUid;
+    pid_t                   mClientPid;
 };
 
 class TimedAudioTrack : public AudioTrack
diff --git a/include/media/EffectsFactoryApi.h b/include/media/EffectsFactoryApi.h
index b1143b9..b1ed7b0 100644
--- a/include/media/EffectsFactoryApi.h
+++ b/include/media/EffectsFactoryApi.h
@@ -171,30 +171,6 @@
 ////////////////////////////////////////////////////////////////////////////////
 int EffectIsNullUuid(const effect_uuid_t *pEffectUuid);
 
-////////////////////////////////////////////////////////////////////////////////
-//
-//    Function:       EffectGetSubEffects
-//
-//    Description:    Returns the descriptors of the sub effects of the effect
-//                    whose uuid is pointed to by first argument.
-//
-//    Input:
-//          pEffectUuid:    pointer to the effect uuid.
-//          size:           size of the buffer pointed by pDescriptor.
-//
-//    Input/Output:
-//          pDescriptor:    address where to return the sub effect descriptors.
-//
-//    Output:
-//        returned value:    0          successful operation.
-//                          -ENODEV     factory failed to initialize
-//                          -EINVAL     invalid pEffectUuid or pDescriptor
-//                          -ENOENT     no effect with this uuid found
-//        *pDescriptor:     updated with the sub effect descriptors.
-//
-////////////////////////////////////////////////////////////////////////////////
-int EffectGetSubEffects(const effect_uuid_t *pEffectUuid, effect_descriptor_t *pDescriptors, size_t size);
-
 #if __cplusplus
 }  // extern "C"
 #endif
diff --git a/include/media/IAudioFlinger.h b/include/media/IAudioFlinger.h
index 899d79f..7f53bfc 100644
--- a/include/media/IAudioFlinger.h
+++ b/include/media/IAudioFlinger.h
@@ -64,9 +64,12 @@
                                 uint32_t sampleRate,
                                 audio_format_t format,
                                 audio_channel_mask_t channelMask,
-                                size_t frameCount,
+                                size_t *pFrameCount,
                                 track_flags_t *flags,
                                 const sp<IMemory>& sharedBuffer,
+                                // On successful return, AudioFlinger takes over the handle
+                                // reference and will release it when the track is destroyed.
+                                // However on failure, the client is responsible for release.
                                 audio_io_handle_t output,
                                 pid_t tid,  // -1 means unused, otherwise must be valid non-0
                                 int *sessionId,
@@ -78,11 +81,14 @@
                                 status_t *status) = 0;
 
     virtual sp<IAudioRecord> openRecord(
+                                // On successful return, AudioFlinger takes over the handle
+                                // reference and will release it when the track is destroyed.
+                                // However on failure, the client is responsible for release.
                                 audio_io_handle_t input,
                                 uint32_t sampleRate,
                                 audio_format_t format,
                                 audio_channel_mask_t channelMask,
-                                size_t frameCount,
+                                size_t *pFrameCount,
                                 track_flags_t *flags,
                                 pid_t tid,  // -1 means unused, otherwise must be valid non-0
                                 int *sessionId,
@@ -167,15 +173,15 @@
 
     virtual status_t setVoiceVolume(float volume) = 0;
 
-    virtual status_t getRenderPosition(size_t *halFrames, size_t *dspFrames,
+    virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
                                     audio_io_handle_t output) const = 0;
 
-    virtual size_t getInputFramesLost(audio_io_handle_t ioHandle) const = 0;
+    virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const = 0;
 
     virtual int newAudioSessionId() = 0;
 
-    virtual void acquireAudioSessionId(int audioSession) = 0;
-    virtual void releaseAudioSessionId(int audioSession) = 0;
+    virtual void acquireAudioSessionId(int audioSession, pid_t pid) = 0;
+    virtual void releaseAudioSessionId(int audioSession, pid_t pid) = 0;
 
     virtual status_t queryNumberEffects(uint32_t *numEffects) const = 0;
 
@@ -188,6 +194,7 @@
                                     effect_descriptor_t *pDesc,
                                     const sp<IEffectClient>& client,
                                     int32_t priority,
+                                    // AudioFlinger doesn't take over handle reference from client
                                     audio_io_handle_t output,
                                     int sessionId,
                                     status_t *status,
diff --git a/include/media/IMediaHTTPConnection.h b/include/media/IMediaHTTPConnection.h
new file mode 100644
index 0000000..e048b64
--- /dev/null
+++ b/include/media/IMediaHTTPConnection.h
@@ -0,0 +1,48 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef I_MEDIA_HTTP_CONNECTION_H_
+
+#define I_MEDIA_HTTP_CONNECTION_H_
+
+#include <binder/IInterface.h>
+#include <media/stagefright/foundation/ABase.h>
+#include <utils/KeyedVector.h>
+
+namespace android {
+
+struct IMediaHTTPConnection;
+
+/** MUST stay in sync with IMediaHTTPConnection.aidl */
+
+struct IMediaHTTPConnection : public IInterface {
+    DECLARE_META_INTERFACE(MediaHTTPConnection);
+
+    virtual bool connect(
+            const char *uri, const KeyedVector<String8, String8> *headers) = 0;
+
+    virtual void disconnect() = 0;
+    virtual ssize_t readAt(off64_t offset, void *data, size_t size) = 0;
+    virtual off64_t getSize() = 0;
+    virtual status_t getMIMEType(String8 *mimeType) = 0;
+
+private:
+    DISALLOW_EVIL_CONSTRUCTORS(IMediaHTTPConnection);
+};
+
+}  // namespace android
+
+#endif  // I_MEDIA_HTTP_CONNECTION_H_
diff --git a/include/media/IMediaHTTPService.h b/include/media/IMediaHTTPService.h
new file mode 100644
index 0000000..f66d6c8
--- /dev/null
+++ b/include/media/IMediaHTTPService.h
@@ -0,0 +1,41 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef I_MEDIA_HTTP_SERVICE_H_
+
+#define I_MEDIA_HTTP_SERVICE_H_
+
+#include <binder/IInterface.h>
+#include <media/stagefright/foundation/ABase.h>
+
+namespace android {
+
+struct IMediaHTTPConnection;
+
+/** MUST stay in sync with IMediaHTTPService.aidl */
+
+struct IMediaHTTPService : public IInterface {
+    DECLARE_META_INTERFACE(MediaHTTPService);
+
+    virtual sp<IMediaHTTPConnection> makeHTTPConnection() = 0;
+
+private:
+    DISALLOW_EVIL_CONSTRUCTORS(IMediaHTTPService);
+};
+
+}  // namespace android
+
+#endif  // I_MEDIA_HTTP_SERVICE_H_
diff --git a/include/media/IMediaMetadataRetriever.h b/include/media/IMediaMetadataRetriever.h
index 6dbb2d7..2529800 100644
--- a/include/media/IMediaMetadataRetriever.h
+++ b/include/media/IMediaMetadataRetriever.h
@@ -26,6 +26,8 @@
 
 namespace android {
 
+struct IMediaHTTPService;
+
 class IMediaMetadataRetriever: public IInterface
 {
 public:
@@ -33,6 +35,7 @@
     virtual void            disconnect() = 0;
 
     virtual status_t        setDataSource(
+            const sp<IMediaHTTPService> &httpService,
             const char *srcUrl,
             const KeyedVector<String8, String8> *headers = NULL) = 0;
 
diff --git a/include/media/IMediaPlayer.h b/include/media/IMediaPlayer.h
index 0cbd269..db62cd5 100644
--- a/include/media/IMediaPlayer.h
+++ b/include/media/IMediaPlayer.h
@@ -33,6 +33,7 @@
 class Surface;
 class IStreamSource;
 class IGraphicBufferProducer;
+struct IMediaHTTPService;
 
 class IMediaPlayer: public IInterface
 {
@@ -41,8 +42,11 @@
 
     virtual void            disconnect() = 0;
 
-    virtual status_t        setDataSource(const char *url,
-                                    const KeyedVector<String8, String8>* headers) = 0;
+    virtual status_t        setDataSource(
+            const sp<IMediaHTTPService> &httpService,
+            const char *url,
+            const KeyedVector<String8, String8>* headers) = 0;
+
     virtual status_t        setDataSource(int fd, int64_t offset, int64_t length) = 0;
     virtual status_t        setDataSource(const sp<IStreamSource>& source) = 0;
     virtual status_t        setVideoSurfaceTexture(
diff --git a/include/media/IMediaPlayerService.h b/include/media/IMediaPlayerService.h
index 2998b37..5b45376 100644
--- a/include/media/IMediaPlayerService.h
+++ b/include/media/IMediaPlayerService.h
@@ -34,6 +34,7 @@
 struct ICrypto;
 struct IDrm;
 struct IHDCP;
+struct IMediaHTTPService;
 class IMediaRecorder;
 class IOMX;
 class IRemoteDisplay;
@@ -49,9 +50,14 @@
     virtual sp<IMediaMetadataRetriever> createMetadataRetriever() = 0;
     virtual sp<IMediaPlayer> create(const sp<IMediaPlayerClient>& client, int audioSessionId = 0) = 0;
 
-    virtual status_t         decode(const char* url, uint32_t *pSampleRate, int* pNumChannels,
-                                    audio_format_t* pFormat,
-                                    const sp<IMemoryHeap>& heap, size_t *pSize) = 0;
+    virtual status_t         decode(
+            const sp<IMediaHTTPService> &httpService,
+            const char* url,
+            uint32_t *pSampleRate,
+            int* pNumChannels,
+            audio_format_t* pFormat,
+            const sp<IMemoryHeap>& heap, size_t *pSize) = 0;
+
     virtual status_t         decode(int fd, int64_t offset, int64_t length, uint32_t *pSampleRate,
                                     int* pNumChannels, audio_format_t* pFormat,
                                     const sp<IMemoryHeap>& heap, size_t *pSize) = 0;
@@ -93,9 +99,6 @@
 
     virtual void addBatteryData(uint32_t params) = 0;
     virtual status_t pullBatteryData(Parcel* reply) = 0;
-
-    virtual status_t updateProxyConfig(
-            const char *host, int32_t port, const char *exclusionList) = 0;
 };
 
 // ----------------------------------------------------------------------------
diff --git a/include/media/IOMX.h b/include/media/IOMX.h
index 9c8451c..3db2c38 100644
--- a/include/media/IOMX.h
+++ b/include/media/IOMX.h
@@ -142,6 +142,8 @@
     enum InternalOptionType {
         INTERNAL_OPTION_SUSPEND,  // data is a bool
         INTERNAL_OPTION_REPEAT_PREVIOUS_FRAME_DELAY,  // data is an int64_t
+        INTERNAL_OPTION_MAX_TIMESTAMP_GAP, // data is int64_t
+        INTERNAL_OPTION_START_TIME, // data is an int64_t
     };
     virtual status_t setInternalOption(
             node_id node,
diff --git a/include/media/MediaMetadataRetrieverInterface.h b/include/media/MediaMetadataRetrieverInterface.h
index ecc3b65..bb6b97b 100644
--- a/include/media/MediaMetadataRetrieverInterface.h
+++ b/include/media/MediaMetadataRetrieverInterface.h
@@ -24,6 +24,8 @@
 
 namespace android {
 
+struct IMediaHTTPService;
+
 // Abstract base class
 class MediaMetadataRetrieverBase : public RefBase
 {
@@ -32,6 +34,7 @@
     virtual             ~MediaMetadataRetrieverBase() {}
 
     virtual status_t    setDataSource(
+            const sp<IMediaHTTPService> &httpService,
             const char *url,
             const KeyedVector<String8, String8> *headers = NULL) = 0;
 
diff --git a/include/media/MediaPlayerInterface.h b/include/media/MediaPlayerInterface.h
index 26d8729..87717da 100644
--- a/include/media/MediaPlayerInterface.h
+++ b/include/media/MediaPlayerInterface.h
@@ -137,6 +137,7 @@
     }
 
     virtual status_t    setDataSource(
+            const sp<IMediaHTTPService> &httpService,
             const char *url,
             const KeyedVector<String8, String8> *headers = NULL) = 0;
 
@@ -213,11 +214,6 @@
         return INVALID_OPERATION;
     }
 
-    virtual status_t updateProxyConfig(
-            const char *host, int32_t port, const char *exclusionList) {
-        return INVALID_OPERATION;
-    }
-
 private:
     friend class MediaPlayerService;
 
diff --git a/include/media/mediametadataretriever.h b/include/media/mediametadataretriever.h
index 0df77c1..b35cf32 100644
--- a/include/media/mediametadataretriever.h
+++ b/include/media/mediametadataretriever.h
@@ -25,6 +25,7 @@
 
 namespace android {
 
+struct IMediaHTTPService;
 class IMediaPlayerService;
 class IMediaMetadataRetriever;
 
@@ -68,6 +69,7 @@
     void disconnect();
 
     status_t setDataSource(
+            const sp<IMediaHTTPService> &httpService,
             const char *dataSourceUrl,
             const KeyedVector<String8, String8> *headers = NULL);
 
diff --git a/include/media/mediaplayer.h b/include/media/mediaplayer.h
index 4c05fc3..f8e4e3b 100644
--- a/include/media/mediaplayer.h
+++ b/include/media/mediaplayer.h
@@ -189,6 +189,8 @@
     virtual void notify(int msg, int ext1, int ext2, const Parcel *obj) = 0;
 };
 
+struct IMediaHTTPService;
+
 class MediaPlayer : public BnMediaPlayerClient,
                     public virtual IMediaDeathNotifier
 {
@@ -199,6 +201,7 @@
             void            disconnect();
 
             status_t        setDataSource(
+                    const sp<IMediaHTTPService> &httpService,
                     const char *url,
                     const KeyedVector<String8, String8> *headers);
 
@@ -224,9 +227,14 @@
             bool            isLooping();
             status_t        setVolume(float leftVolume, float rightVolume);
             void            notify(int msg, int ext1, int ext2, const Parcel *obj = NULL);
-    static  status_t        decode(const char* url, uint32_t *pSampleRate, int* pNumChannels,
-                                   audio_format_t* pFormat,
-                                   const sp<IMemoryHeap>& heap, size_t *pSize);
+    static  status_t        decode(
+            const sp<IMediaHTTPService> &httpService,
+            const char* url,
+            uint32_t *pSampleRate,
+            int* pNumChannels,
+            audio_format_t* pFormat,
+            const sp<IMemoryHeap>& heap,
+            size_t *pSize);
     static  status_t        decode(int fd, int64_t offset, int64_t length, uint32_t *pSampleRate,
                                    int* pNumChannels, audio_format_t* pFormat,
                                    const sp<IMemoryHeap>& heap, size_t *pSize);
@@ -242,9 +250,6 @@
             status_t        setRetransmitEndpoint(const char* addrString, uint16_t port);
             status_t        setNextMediaPlayer(const sp<MediaPlayer>& player);
 
-            status_t updateProxyConfig(
-                    const char *host, int32_t port, const char *exclusionList);
-
 private:
             void            clear_l();
             status_t        seekTo_l(int msec);
diff --git a/include/media/mediarecorder.h b/include/media/mediarecorder.h
index 88a42a0..142cb90 100644
--- a/include/media/mediarecorder.h
+++ b/include/media/mediarecorder.h
@@ -39,7 +39,7 @@
 enum video_source {
     VIDEO_SOURCE_DEFAULT = 0,
     VIDEO_SOURCE_CAMERA = 1,
-    VIDEO_SOURCE_GRALLOC_BUFFER = 2,
+    VIDEO_SOURCE_SURFACE = 2,
 
     VIDEO_SOURCE_LIST_END  // must be last - used to validate audio source type
 };
diff --git a/include/media/nbaio/AudioBufferProviderSource.h b/include/media/nbaio/AudioBufferProviderSource.h
index 2c4aaff..b16e20a 100644
--- a/include/media/nbaio/AudioBufferProviderSource.h
+++ b/include/media/nbaio/AudioBufferProviderSource.h
@@ -27,7 +27,7 @@
 class AudioBufferProviderSource : public NBAIO_Source {
 
 public:
-    AudioBufferProviderSource(AudioBufferProvider *provider, NBAIO_Format format);
+    AudioBufferProviderSource(AudioBufferProvider *provider, const NBAIO_Format& format);
     virtual ~AudioBufferProviderSource();
 
     // NBAIO_Port interface
diff --git a/include/media/nbaio/MonoPipe.h b/include/media/nbaio/MonoPipe.h
index d3802fe..b09b35f 100644
--- a/include/media/nbaio/MonoPipe.h
+++ b/include/media/nbaio/MonoPipe.h
@@ -41,7 +41,7 @@
     // Note: whatever shares this object with another thread needs to do so in an SMP-safe way (like
     // creating it the object before creating the other thread, or storing the object with a
     // release_store). Otherwise the other thread could see a partially-constructed object.
-    MonoPipe(size_t reqFrames, NBAIO_Format format, bool writeCanBlock = false);
+    MonoPipe(size_t reqFrames, const NBAIO_Format& format, bool writeCanBlock = false);
     virtual ~MonoPipe();
 
     // NBAIO_Port interface
diff --git a/include/media/nbaio/NBAIO.h b/include/media/nbaio/NBAIO.h
index 1da0c73..56896b9 100644
--- a/include/media/nbaio/NBAIO.h
+++ b/include/media/nbaio/NBAIO.h
@@ -52,31 +52,41 @@
 // the combinations that are actually needed within AudioFlinger.  If the list of combinations grows
 // too large, then this decision should be re-visited.
 // Sample rate and channel count are explicit, PCM interleaved 16-bit is assumed.
-typedef unsigned NBAIO_Format;
-enum {
-    Format_Invalid
+struct NBAIO_Format {
+//private:
+    unsigned    mPacked;
 };
 
+extern const NBAIO_Format Format_Invalid;
+
 // Return the frame size of an NBAIO_Format in bytes
-size_t Format_frameSize(NBAIO_Format format);
+size_t Format_frameSize(const NBAIO_Format& format);
 
 // Return the frame size of an NBAIO_Format as a bit shift
-size_t Format_frameBitShift(NBAIO_Format format);
+// or -1 if frame size is not a power of 2
+int Format_frameBitShift(const NBAIO_Format& format);
 
 // Convert a sample rate in Hz and channel count to an NBAIO_Format
+// FIXME The sample format is hard-coded to AUDIO_FORMAT_PCM_16_BIT
 NBAIO_Format Format_from_SR_C(unsigned sampleRate, unsigned channelCount);
 
 // Return the sample rate in Hz of an NBAIO_Format
-unsigned Format_sampleRate(NBAIO_Format format);
+unsigned Format_sampleRate(const NBAIO_Format& format);
 
 // Return the channel count of an NBAIO_Format
-unsigned Format_channelCount(NBAIO_Format format);
+unsigned Format_channelCount(const NBAIO_Format& format);
 
 // Callbacks used by NBAIO_Sink::writeVia() and NBAIO_Source::readVia() below.
 typedef ssize_t (*writeVia_t)(void *user, void *buffer, size_t count);
 typedef ssize_t (*readVia_t)(void *user, const void *buffer,
                              size_t count, int64_t readPTS);
 
+// Check whether an NBAIO_Format is valid
+bool Format_isValid(const NBAIO_Format& format);
+
+// Compare two NBAIO_Format values
+bool Format_isEqual(const NBAIO_Format& format1, const NBAIO_Format& format2);
+
 // Abstract class (interface) representing a data port.
 class NBAIO_Port : public RefBase {
 
@@ -115,8 +125,8 @@
     virtual NBAIO_Format format() const { return mNegotiated ? mFormat : Format_Invalid; }
 
 protected:
-    NBAIO_Port(NBAIO_Format format) : mNegotiated(false), mFormat(format),
-                                      mBitShift(Format_frameBitShift(format)) { }
+    NBAIO_Port(const NBAIO_Format& format) : mNegotiated(false), mFormat(format),
+                                             mBitShift(Format_frameBitShift(format)) { }
     virtual ~NBAIO_Port() { }
 
     // Implementations are free to ignore these if they don't need them
@@ -220,7 +230,7 @@
     virtual status_t getTimestamp(AudioTimestamp& timestamp) { return INVALID_OPERATION; }
 
 protected:
-    NBAIO_Sink(NBAIO_Format format = Format_Invalid) : NBAIO_Port(format), mFramesWritten(0) { }
+    NBAIO_Sink(const NBAIO_Format& format = Format_Invalid) : NBAIO_Port(format), mFramesWritten(0) { }
     virtual ~NBAIO_Sink() { }
 
     // Implementations are free to ignore these if they don't need them
@@ -311,7 +321,7 @@
     virtual void    onTimestamp(const AudioTimestamp& timestamp) { }
 
 protected:
-    NBAIO_Source(NBAIO_Format format = Format_Invalid) : NBAIO_Port(format), mFramesRead(0) { }
+    NBAIO_Source(const NBAIO_Format& format = Format_Invalid) : NBAIO_Port(format), mFramesRead(0) { }
     virtual ~NBAIO_Source() { }
 
     // Implementations are free to ignore these if they don't need them
diff --git a/include/media/nbaio/Pipe.h b/include/media/nbaio/Pipe.h
index 79a4eee..c784129 100644
--- a/include/media/nbaio/Pipe.h
+++ b/include/media/nbaio/Pipe.h
@@ -30,7 +30,7 @@
 
 public:
     // maxFrames will be rounded up to a power of 2, and all slots are available. Must be >= 2.
-    Pipe(size_t maxFrames, NBAIO_Format format);
+    Pipe(size_t maxFrames, const NBAIO_Format& format);
     virtual ~Pipe();
 
     // NBAIO_Port interface
diff --git a/include/media/stagefright/ACodec.h b/include/media/stagefright/ACodec.h
index f1636e6..e284109 100644
--- a/include/media/stagefright/ACodec.h
+++ b/include/media/stagefright/ACodec.h
@@ -67,6 +67,8 @@
 
     void signalRequestIDRFrame();
 
+    bool isConfiguredForAdaptivePlayback() { return mIsConfiguredForAdaptivePlayback; }
+
     struct PortDescription : public RefBase {
         size_t countBuffers();
         IOMX::buffer_id bufferIDAt(size_t index) const;
@@ -187,6 +189,7 @@
     bool mIsEncoder;
     bool mUseMetadataOnEncoderOutput;
     bool mShutdownInProgress;
+    bool mIsConfiguredForAdaptivePlayback;
 
     // If "mKeepComponentAllocated" we only transition back to Loaded state
     // and do not release the component instance.
@@ -202,6 +205,9 @@
     int32_t mMetaDataBuffersToSubmit;
 
     int64_t mRepeatFrameDelayUs;
+    int64_t mMaxPtsGapUs;
+
+    bool mCreateInputBuffersSuspended;
 
     status_t setCyclicIntraMacroblockRefresh(const sp<AMessage> &msg, int32_t mode);
     status_t allocateBuffersOnPort(OMX_U32 portIndex);
diff --git a/include/media/stagefright/DataSource.h b/include/media/stagefright/DataSource.h
index 157b1aa..f8787dd 100644
--- a/include/media/stagefright/DataSource.h
+++ b/include/media/stagefright/DataSource.h
@@ -31,6 +31,7 @@
 namespace android {
 
 struct AMessage;
+struct IMediaHTTPService;
 class String8;
 
 class DataSource : public RefBase {
@@ -43,6 +44,7 @@
     };
 
     static sp<DataSource> CreateFromURI(
+            const sp<IMediaHTTPService> &httpService,
             const char *uri,
             const KeyedVector<String8, String8> *headers = NULL);
 
diff --git a/include/media/stagefright/DataURISource.h b/include/media/stagefright/DataURISource.h
new file mode 100644
index 0000000..693562e
--- /dev/null
+++ b/include/media/stagefright/DataURISource.h
@@ -0,0 +1,49 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef DATA_URI_SOURCE_H_
+
+#define DATA_URI_SOURCE_H_
+
+#include <media/stagefright/DataSource.h>
+#include <media/stagefright/foundation/ABase.h>
+
+namespace android {
+
+struct ABuffer;
+
+struct DataURISource : public DataSource {
+    static sp<DataURISource> Create(const char *uri);
+
+    virtual status_t initCheck() const;
+    virtual ssize_t readAt(off64_t offset, void *data, size_t size);
+    virtual status_t getSize(off64_t *size);
+
+protected:
+    virtual ~DataURISource();
+
+private:
+    sp<ABuffer> mBuffer;
+
+    DataURISource(const sp<ABuffer> &buffer);
+
+    DISALLOW_EVIL_CONSTRUCTORS(DataURISource);
+};
+
+}  // namespace android
+
+#endif  // DATA_URI_SOURCE_H_
+
diff --git a/include/media/stagefright/MediaCodecSource.h b/include/media/stagefright/MediaCodecSource.h
new file mode 100644
index 0000000..4b18a0b
--- /dev/null
+++ b/include/media/stagefright/MediaCodecSource.h
@@ -0,0 +1,134 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef MediaCodecSource_H_
+#define MediaCodecSource_H_
+
+#include <media/stagefright/foundation/ABase.h>
+#include <media/stagefright/foundation/AHandlerReflector.h>
+#include <media/stagefright/MediaSource.h>
+
+namespace android {
+
+class ALooper;
+class AMessage;
+class IGraphicBufferProducer;
+class MediaCodec;
+class MetaData;
+
+struct MediaCodecSource : public MediaSource,
+                          public MediaBufferObserver {
+    enum FlagBits {
+        FLAG_USE_SURFACE_INPUT      = 1,
+        FLAG_USE_METADATA_INPUT     = 2,
+    };
+
+    static sp<MediaCodecSource> Create(
+            const sp<ALooper> &looper,
+            const sp<AMessage> &format,
+            const sp<MediaSource> &source,
+            uint32_t flags = 0);
+
+    bool isVideo() const { return mIsVideo; }
+    sp<IGraphicBufferProducer> getGraphicBufferProducer();
+
+    // MediaSource
+    virtual status_t start(MetaData *params = NULL);
+    virtual status_t stop();
+    virtual status_t pause();
+    virtual sp<MetaData> getFormat() { return mMeta; }
+    virtual status_t read(
+            MediaBuffer **buffer,
+            const ReadOptions *options = NULL);
+
+    // MediaBufferObserver
+    virtual void signalBufferReturned(MediaBuffer *buffer);
+
+    // for AHandlerReflector
+    void onMessageReceived(const sp<AMessage> &msg);
+
+protected:
+    virtual ~MediaCodecSource();
+
+private:
+    struct Puller;
+
+    enum {
+        kWhatPullerNotify,
+        kWhatEncoderActivity,
+        kWhatStart,
+        kWhatStop,
+        kWhatPause,
+    };
+
+    MediaCodecSource(
+            const sp<ALooper> &looper,
+            const sp<AMessage> &outputFormat,
+            const sp<MediaSource> &source,
+            uint32_t flags = 0);
+
+    status_t onStart(MetaData *params);
+    status_t init();
+    status_t initEncoder();
+    void releaseEncoder();
+    status_t feedEncoderInputBuffers();
+    void scheduleDoMoreWork();
+    status_t doMoreWork();
+    void suspend();
+    void resume(int64_t skipFramesBeforeUs = -1ll);
+    void signalEOS(status_t err = ERROR_END_OF_STREAM);
+    bool reachedEOS();
+    status_t postSynchronouslyAndReturnError(const sp<AMessage> &msg);
+
+    sp<ALooper> mLooper;
+    sp<ALooper> mCodecLooper;
+    sp<AHandlerReflector<MediaCodecSource> > mReflector;
+    sp<AMessage> mOutputFormat;
+    sp<MetaData> mMeta;
+    sp<Puller> mPuller;
+    sp<MediaCodec> mEncoder;
+    uint32_t mFlags;
+    List<uint32_t> mStopReplyIDQueue;
+    bool mIsVideo;
+    bool mStarted;
+    bool mStopping;
+    bool mDoMoreWorkPending;
+    bool mPullerReachedEOS;
+    sp<AMessage> mEncoderActivityNotify;
+    sp<IGraphicBufferProducer> mGraphicBufferProducer;
+    Vector<sp<ABuffer> > mEncoderInputBuffers;
+    Vector<sp<ABuffer> > mEncoderOutputBuffers;
+    List<MediaBuffer *> mInputBufferQueue;
+    List<size_t> mAvailEncoderInputIndices;
+    List<int64_t> mDecodingTimeQueue; // decoding time (us) for video
+
+    // audio drift time
+    int64_t mFirstSampleTimeUs;
+    List<int64_t> mDriftTimeQueue;
+
+    // following variables are protected by mOutputBufferLock
+    Mutex mOutputBufferLock;
+    Condition mOutputBufferCond;
+    List<MediaBuffer*> mOutputBufferQueue;
+    bool mEncodedReachedEOS;
+    status_t mErrorCode;
+
+    DISALLOW_EVIL_CONSTRUCTORS(MediaCodecSource);
+};
+
+} // namespace android
+
+#endif /* MediaCodecSource_H_ */
diff --git a/include/media/stagefright/MediaHTTP.h b/include/media/stagefright/MediaHTTP.h
new file mode 100644
index 0000000..006d8d8
--- /dev/null
+++ b/include/media/stagefright/MediaHTTP.h
@@ -0,0 +1,77 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef MEDIA_HTTP_H_
+
+#define MEDIA_HTTP_H_
+
+#include <media/stagefright/foundation/AString.h>
+
+#include "include/HTTPBase.h"
+
+namespace android {
+
+struct IMediaHTTPConnection;
+
+struct MediaHTTP : public HTTPBase {
+    MediaHTTP(const sp<IMediaHTTPConnection> &conn);
+
+    virtual status_t connect(
+            const char *uri,
+            const KeyedVector<String8, String8> *headers,
+            off64_t offset);
+
+    virtual void disconnect();
+
+    virtual status_t initCheck() const;
+
+    virtual ssize_t readAt(off64_t offset, void *data, size_t size);
+
+    virtual status_t getSize(off64_t *size);
+
+    virtual uint32_t flags();
+
+    virtual status_t reconnectAtOffset(off64_t offset);
+
+protected:
+    virtual ~MediaHTTP();
+
+    virtual sp<DecryptHandle> DrmInitialization(const char* mime);
+    virtual void getDrmInfo(sp<DecryptHandle> &handle, DrmManagerClient **client);
+    virtual String8 getUri();
+    virtual String8 getMIMEType() const;
+
+private:
+    status_t mInitCheck;
+    sp<IMediaHTTPConnection> mHTTPConnection;
+
+    KeyedVector<String8, String8> mLastHeaders;
+    AString mLastURI;
+
+    bool mCachedSizeValid;
+    off64_t mCachedSize;
+
+    sp<DecryptHandle> mDecryptHandle;
+    DrmManagerClient *mDrmManagerClient;
+
+    void clearDRMState_l();
+
+    DISALLOW_EVIL_CONSTRUCTORS(MediaHTTP);
+};
+
+}  // namespace android
+
+#endif  // MEDIA_HTTP_H_
diff --git a/include/media/stagefright/MediaSource.h b/include/media/stagefright/MediaSource.h
index 3818e63..204d1c6 100644
--- a/include/media/stagefright/MediaSource.h
+++ b/include/media/stagefright/MediaSource.h
@@ -105,7 +105,7 @@
     // This will be called after a successful start() and before the
     // first read() call.
     // Callee assumes ownership of the buffers if no error is returned.
-    virtual status_t setBuffers(const Vector<MediaBuffer *> &buffers) {
+    virtual status_t setBuffers(const Vector<MediaBuffer *> & /* buffers */) {
         return ERROR_UNSUPPORTED;
     }
 
diff --git a/include/media/stagefright/MetaData.h b/include/media/stagefright/MetaData.h
index 3a87474..db8216b 100644
--- a/include/media/stagefright/MetaData.h
+++ b/include/media/stagefright/MetaData.h
@@ -215,6 +215,8 @@
     bool findData(uint32_t key, uint32_t *type,
                   const void **data, size_t *size) const;
 
+    bool hasData(uint32_t key) const;
+
     void dumpToLog() const;
 
 protected:
diff --git a/include/media/stagefright/NuMediaExtractor.h b/include/media/stagefright/NuMediaExtractor.h
index 5ae6f6b..402e7f8 100644
--- a/include/media/stagefright/NuMediaExtractor.h
+++ b/include/media/stagefright/NuMediaExtractor.h
@@ -31,6 +31,7 @@
 struct ABuffer;
 struct AMessage;
 struct DataSource;
+struct IMediaHTTPService;
 struct MediaBuffer;
 struct MediaExtractor;
 struct MediaSource;
@@ -45,6 +46,7 @@
     NuMediaExtractor();
 
     status_t setDataSource(
+            const sp<IMediaHTTPService> &httpService,
             const char *path,
             const KeyedVector<String8, String8> *headers = NULL);
 
diff --git a/include/media/stagefright/timedtext/TimedTextDriver.h b/include/media/stagefright/timedtext/TimedTextDriver.h
index f23c337..37ef674 100644
--- a/include/media/stagefright/timedtext/TimedTextDriver.h
+++ b/include/media/stagefright/timedtext/TimedTextDriver.h
@@ -25,6 +25,7 @@
 namespace android {
 
 class ALooper;
+struct IMediaHTTPService;
 class MediaPlayerBase;
 class MediaSource;
 class Parcel;
@@ -34,7 +35,9 @@
 
 class TimedTextDriver {
 public:
-    TimedTextDriver(const wp<MediaPlayerBase> &listener);
+    TimedTextDriver(
+            const wp<MediaPlayerBase> &listener,
+            const sp<IMediaHTTPService> &httpService);
 
     ~TimedTextDriver();
 
@@ -77,6 +80,7 @@
     sp<ALooper> mLooper;
     sp<TimedTextPlayer> mPlayer;
     wp<MediaPlayerBase> mListener;
+    sp<IMediaHTTPService> mHTTPService;
 
     // Variables to be guarded by mLock.
     State mState;
diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index 85862a8..3901e79 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -65,7 +65,9 @@
 struct AudioTrackSharedStatic {
     StaticAudioTrackSingleStateQueue::Shared
                     mSingleStateQueue;
-    size_t          mBufferPosition;    // updated asynchronously by server,
+    // This field should be a size_t, but since it is located in shared memory we
+    // force to 32-bit.  The client and server may have different typedefs for size_t.
+    uint32_t        mBufferPosition;    // updated asynchronously by server,
                                         // "for entertainment purposes only"
 };
 
@@ -96,11 +98,7 @@
                                         // The value should be used "for entertainment purposes only",
                                         // which means don't make important decisions based on it.
 
-                size_t      frameCount_;    // used during creation to pass actual track buffer size
-                                            // from AudioFlinger to client, and not referenced again
-                                            // FIXME remove here and replace by createTrack() in/out
-                                            // parameter
-                                            // renamed to "_" to detect incorrect use
+                uint32_t    mPad1;      // unused
 
     volatile    int32_t     mFutex;     // event flag: down (P) by client,
                                         // up (V) by server or binderDied() or interrupt()
@@ -108,7 +106,9 @@
 
 private:
 
-                size_t      mMinimum;       // server wakes up client if available >= mMinimum
+                // This field should be a size_t, but since it is located in shared memory we
+                // force to 32-bit.  The client and server may have different typedefs for size_t.
+                uint32_t    mMinimum;       // server wakes up client if available >= mMinimum
 
                 // Channel volumes are fixed point U4.12, so 0x1000 means 1.0.
                 // Left channel is in [0:15], right channel is in [16:31].
@@ -245,7 +245,11 @@
     }
 
     void        setMinimum(size_t minimum) {
-        mCblk->mMinimum = minimum;
+        // This can only happen on a 64-bit client
+        if (minimum > UINT32_MAX) {
+            minimum = UINT32_MAX;
+        }
+        mCblk->mMinimum = (uint32_t) minimum;
     }
 
     // Return the number of frames that would need to be obtained and released
diff --git a/include/private/media/StaticAudioTrackState.h b/include/private/media/StaticAudioTrackState.h
index 46a5946..d483061 100644
--- a/include/private/media/StaticAudioTrackState.h
+++ b/include/private/media/StaticAudioTrackState.h
@@ -25,9 +25,13 @@
 // state is wrapped by a SingleStateQueue.
 struct StaticAudioTrackState {
     // do not define constructors, destructors, or virtual methods
-    size_t  mLoopStart;
-    size_t  mLoopEnd;
-    int     mLoopCount;
+
+    // These fields should both be size_t, but since they are located in shared memory we
+    // force to 32-bit.  The client and server may have different typedefs for size_t.
+    uint32_t    mLoopStart;
+    uint32_t    mLoopEnd;
+
+    int         mLoopCount;
 };
 
 }   // namespace android
diff --git a/libvideoeditor/lvpp/PreviewPlayer.cpp b/libvideoeditor/lvpp/PreviewPlayer.cpp
index 2bd9f84..b36fe0a 100755
--- a/libvideoeditor/lvpp/PreviewPlayer.cpp
+++ b/libvideoeditor/lvpp/PreviewPlayer.cpp
@@ -21,6 +21,7 @@
 
 #include <binder/IPCThreadState.h>
 #include <binder/IServiceManager.h>
+#include <media/IMediaHTTPService.h>
 #include <media/IMediaPlayerService.h>
 #include <media/stagefright/DataSource.h>
 #include <media/stagefright/MediaBuffer.h>
@@ -1160,7 +1161,8 @@
     sp<DataSource> dataSource;
     sp<MediaExtractor> extractor;
 
-    dataSource = DataSource::CreateFromURI(mUri.string(), NULL);
+    dataSource = DataSource::CreateFromURI(
+            NULL /* httpService */, mUri.string(), NULL);
 
     if (dataSource == NULL) {
         return UNKNOWN_ERROR;
diff --git a/libvideoeditor/lvpp/VideoEditorAudioPlayer.cpp b/libvideoeditor/lvpp/VideoEditorAudioPlayer.cpp
index 176f8e9..e503936 100755
--- a/libvideoeditor/lvpp/VideoEditorAudioPlayer.cpp
+++ b/libvideoeditor/lvpp/VideoEditorAudioPlayer.cpp
@@ -671,8 +671,9 @@
 
 
                         M4OSA_Void* ptr;
-                        ptr = (M4OSA_Void*)((unsigned int)mInputBuffer->data() +
-                        mInputBuffer->range_offset());
+                        ptr = reinterpret_cast<M4OSA_Void*>(
+                                reinterpret_cast<uintptr_t>(mInputBuffer->data()) +
+                                mInputBuffer->range_offset());
 
                         M4OSA_UInt32 len = mInputBuffer->range_length();
                         M4OSA_Context fp = M4OSA_NULL;
diff --git a/libvideoeditor/lvpp/VideoEditorPlayer.cpp b/libvideoeditor/lvpp/VideoEditorPlayer.cpp
index 8d656c4..f9c3879 100755
--- a/libvideoeditor/lvpp/VideoEditorPlayer.cpp
+++ b/libvideoeditor/lvpp/VideoEditorPlayer.cpp
@@ -57,6 +57,7 @@
 
 
 status_t VideoEditorPlayer::setDataSource(
+        const sp<IMediaHTTPService> &httpService,
         const char *url, const KeyedVector<String8, String8> *headers) {
     ALOGI("setDataSource('%s')", url);
     if (headers != NULL) {
diff --git a/libvideoeditor/lvpp/VideoEditorPlayer.h b/libvideoeditor/lvpp/VideoEditorPlayer.h
index b8c1254..781e4bc 100755
--- a/libvideoeditor/lvpp/VideoEditorPlayer.h
+++ b/libvideoeditor/lvpp/VideoEditorPlayer.h
@@ -98,6 +98,7 @@
     virtual status_t initCheck();
 
     virtual status_t setDataSource(
+            const sp<IMediaHTTPService> &httpService,
             const char *url, const KeyedVector<String8, String8> *headers);
 
     virtual status_t setDataSource(int fd, int64_t offset, int64_t length);
diff --git a/libvideoeditor/lvpp/VideoEditorPreviewController.cpp b/libvideoeditor/lvpp/VideoEditorPreviewController.cpp
index 149c4ea..953f35a 100755
--- a/libvideoeditor/lvpp/VideoEditorPreviewController.cpp
+++ b/libvideoeditor/lvpp/VideoEditorPreviewController.cpp
@@ -19,6 +19,7 @@
 #include <utils/Log.h>
 
 #include <gui/Surface.h>
+#include <media/IMediaHTTPService.h>
 
 #include "VideoEditorAudioPlayer.h"
 #include "PreviewRenderer.h"
@@ -967,7 +968,8 @@
     ALOGV("preparePlayer: instance %d file %d", playerInstance, index);
 
     const char* fileName = (const char*) pController->mClipList[index]->pFile;
-    pController->mVePlayer[playerInstance]->setDataSource(fileName, NULL);
+    pController->mVePlayer[playerInstance]->setDataSource(
+            NULL /* httpService */, fileName, NULL);
 
     ALOGV("preparePlayer: setDataSource instance %s",
      (const char *)pController->mClipList[index]->pFile);
@@ -1248,7 +1250,7 @@
         case MEDIA_SET_VIDEO_SIZE:
             ALOGV("MEDIA_SET_VIDEO_SIZE; New video size %d x %d", ext1, ext2);
             break;
-        case 0xAAAAAAAA:
+        case static_cast<int>(0xAAAAAAAA):
             ALOGV("VIDEO PLAYBACK ALMOST over, prepare next player");
             // Select next player and prepare it
             // If there is a clip after this one
@@ -1268,7 +1270,7 @@
                 }
             }
             break;
-        case 0xBBBBBBBB:
+        case static_cast<int>(0xBBBBBBBB):
         {
             ALOGV("VIDEO PLAYBACK, Update Overlay");
             int overlayIndex = ext2;
diff --git a/libvideoeditor/lvpp/VideoEditorSRC.cpp b/libvideoeditor/lvpp/VideoEditorSRC.cpp
index 36d0812..6beabfa 100755
--- a/libvideoeditor/lvpp/VideoEditorSRC.cpp
+++ b/libvideoeditor/lvpp/VideoEditorSRC.cpp
@@ -284,7 +284,7 @@
 
 
 void VideoEditorSRC::releaseBuffer(AudioBufferProvider::Buffer *pBuffer) {
-    ALOGV("releaseBuffer: %p", pBuffers);
+    ALOGV("releaseBuffer: %p", pBuffer);
     free(pBuffer->raw);
     pBuffer->raw = NULL;
     pBuffer->frameCount = 0;
diff --git a/libvideoeditor/osal/inc/M4OSA_Error.h b/libvideoeditor/osal/inc/M4OSA_Error.h
index 4d59529..75c3177 100755
--- a/libvideoeditor/osal/inc/M4OSA_Error.h
+++ b/libvideoeditor/osal/inc/M4OSA_Error.h
@@ -57,7 +57,7 @@
   * @arg coreID: (IN) [M4OSA_UInt32] CoreID to put in the error code
   * @arg errorID: (IN) [M4OSA_UInt32] ErrorID to put in the error code*/
 #define M4OSA_ERR_CREATE(severity, coreID, errorID)\
-   (M4OSA_Int32)((((M4OSA_UInt32)severity)<<30)+((((M4OSA_UInt32)coreID)&0x003FFF)<<16)+(((M4OSA_UInt32)errorID)&0x00FFFF))
+   (M4OSA_UInt32)((((M4OSA_UInt32)severity)<<30)+((((M4OSA_UInt32)coreID)&0x003FFF)<<16)+(((M4OSA_UInt32)errorID)&0x00FFFF))
 
 /** This macro extracts the 3 fields from the error:
   * @arg error: (IN) [M4OSA_ERR] Error code
diff --git a/libvideoeditor/vss/src/M4DECODER_Null.c b/libvideoeditor/vss/src/M4DECODER_Null.c
index a0dad30..ce260e5 100755
--- a/libvideoeditor/vss/src/M4DECODER_Null.c
+++ b/libvideoeditor/vss/src/M4DECODER_Null.c
@@ -210,7 +210,7 @@
             break;
 
         case M4DECODER_kOptionID_EnableYuvWithEffect:
-            pStreamContext->bYuvWithEffectSet = (M4OSA_Bool)pValue;
+            pStreamContext->bYuvWithEffectSet = (M4OSA_Bool)(intptr_t)pValue;
             break;
 
         case M4DECODER_kOptionID_YuvWithEffectNonContiguous:
diff --git a/libvideoeditor/vss/stagefrightshells/src/VideoEditor3gpReader.cpp b/libvideoeditor/vss/stagefrightshells/src/VideoEditor3gpReader.cpp
index 3c8915a..99cf9ec 100755
--- a/libvideoeditor/vss/stagefrightshells/src/VideoEditor3gpReader.cpp
+++ b/libvideoeditor/vss/stagefrightshells/src/VideoEditor3gpReader.cpp
@@ -776,16 +776,16 @@
         case M4READER_kOptionID_SetOsaFileReaderFctsPtr:
         break;
 
-        case M4READER_3GP_kOptionID_AudioOnly:
+        case static_cast<M4OSA_OptionID>(M4READER_3GP_kOptionID_AudioOnly):
         break;
 
-        case M4READER_3GP_kOptionID_VideoOnly:
+        case static_cast<M4OSA_OptionID>(M4READER_3GP_kOptionID_VideoOnly):
         break;
 
-        case M4READER_3GP_kOptionID_FastOpenMode:
+        case static_cast<M4OSA_OptionID>(M4READER_3GP_kOptionID_FastOpenMode):
         break;
 
-        case M4READER_kOptionID_MaxMetadataSize:
+        case static_cast<M4OSA_OptionID>(M4READER_kOptionID_MaxMetadataSize):
         break;
 
         default:
diff --git a/libvideoeditor/vss/stagefrightshells/src/VideoEditorAudioDecoder.cpp b/libvideoeditor/vss/stagefrightshells/src/VideoEditorAudioDecoder.cpp
index 9b35d07..e4c7ea1 100755
--- a/libvideoeditor/vss/stagefrightshells/src/VideoEditorAudioDecoder.cpp
+++ b/libvideoeditor/vss/stagefrightshells/src/VideoEditorAudioDecoder.cpp
@@ -809,7 +809,7 @@
     pDecoderContext = (VideoEditorAudioDecoder_Context*)pContext;
 
     switch( optionID ) {
-        case M4AD_kOptionID_UserParam:
+        case static_cast<M4OSA_UInt32>(M4AD_kOptionID_UserParam):
             ALOGV("VideoEditorAudioDecodersetOption UserParam is not supported");
             err = M4ERR_NOT_IMPLEMENTED;
             break;
diff --git a/libvideoeditor/vss/stagefrightshells/src/VideoEditorUtils.cpp b/libvideoeditor/vss/stagefrightshells/src/VideoEditorUtils.cpp
index 5a7237d..d264a2e 100755
--- a/libvideoeditor/vss/stagefrightshells/src/VideoEditorUtils.cpp
+++ b/libvideoeditor/vss/stagefrightshells/src/VideoEditorUtils.cpp
@@ -84,17 +84,17 @@
         LOG1("displayMetaData kKeyBitRate %d", int32Data);
     }
     if (meta->findData(kKeyESDS, &type, &data, &size)) {
-        LOG1("displayMetaData kKeyESDS type=%d size=%d", type, size);
+        LOG1("displayMetaData kKeyESDS type=%d size=%zu", type, size);
     }
     if (meta->findData(kKeyAVCC, &type, &data, &size)) {
-        LOG1("displayMetaData kKeyAVCC data=0x%X type=%d size=%d",
+        LOG1("displayMetaData kKeyAVCC data=0x%X type=%d size=%zu",
             *((unsigned int*)data), type, size);
     }
     if (meta->findData(kKeyVorbisInfo, &type, &data, &size)) {
-        LOG1("displayMetaData kKeyVorbisInfo type=%d size=%d", type, size);
+        LOG1("displayMetaData kKeyVorbisInfo type=%d size=%zu", type, size);
     }
     if (meta->findData(kKeyVorbisBooks, &type, &data, &size)) {
-        LOG1("displayMetaData kKeyVorbisBooks type=%d size=%d", type, size);
+        LOG1("displayMetaData kKeyVorbisBooks type=%d size=%zu", type, size);
     }
     if (meta->findInt32(kKeyWantsNALFragments, &int32Data)) {
         LOG1("displayMetaData kKeyWantsNALFragments %d", int32Data);
@@ -115,7 +115,7 @@
         LOG1("displayMetaData kKeyColorFormat %d", int32Data);
     }
     if (meta->findPointer(kKeyPlatformPrivate, &ptr)) {
-        LOG1("displayMetaData kKeyPlatformPrivate pointer=0x%x", (int32_t) ptr);
+        LOG1("displayMetaData kKeyPlatformPrivate pointer=%p", ptr);
     }
     if (meta->findCString(kKeyDecoderComponent, &charData)) {
         LOG1("displayMetaData kKeyDecoderComponent %s", charData);
@@ -151,7 +151,7 @@
         LOG1("displayMetaData kKeyYear %s", charData);
     }
     if (meta->findData(kKeyAlbumArt, &type, &data, &size)) {
-        LOG1("displayMetaData kKeyAlbumArt type=%d size=%d", type, size);
+        LOG1("displayMetaData kKeyAlbumArt type=%d size=%zu", type, size);
     }
     if (meta->findCString(kKeyAlbumArtMIME, &charData)) {
         LOG1("displayMetaData kKeyAlbumArtMIME %s", charData);
@@ -277,7 +277,7 @@
     }
 
     if (size < 4) {
-        ALOGE("Codec specific data length too short: %d", size);
+        ALOGE("Codec specific data length too short: %zu", size);
         return ERROR_MALFORMED;
     }
 
@@ -286,7 +286,7 @@
         // 2 bytes for each of the parameter set length field
         // plus the 7 bytes for the header
         if (size < 4 + 7) {
-            ALOGE("Codec specific data length too short: %d", size);
+            ALOGE("Codec specific data length too short: %zu", size);
             return ERROR_MALFORMED;
         }
 
@@ -355,7 +355,7 @@
         }
 
         if (nSeqParamSets > 0x1F) {
-            ALOGE("Too many seq parameter sets (%d) found", nSeqParamSets);
+            ALOGE("Too many seq parameter sets (%zu) found", nSeqParamSets);
             return ERROR_MALFORMED;
         }
     }
@@ -368,7 +368,7 @@
             return ERROR_MALFORMED;
         }
         if (nPicParamSets > 0xFF) {
-            ALOGE("Too many pic parameter sets (%d) found", nPicParamSets);
+            ALOGE("Too many pic parameter sets (%zu) found", nPicParamSets);
             return ERROR_MALFORMED;
         }
     }
diff --git a/media/libeffects/downmix/Android.mk b/media/libeffects/downmix/Android.mk
index 5d0a87c..2bb6dbe 100644
--- a/media/libeffects/downmix/Android.mk
+++ b/media/libeffects/downmix/Android.mk
@@ -13,7 +13,7 @@
 
 LOCAL_MODULE_TAGS := optional
 
-LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/soundfx
+LOCAL_MODULE_RELATIVE_PATH := soundfx
 
 ifeq ($(TARGET_OS)-$(TARGET_SIMULATOR),linux-true)
 LOCAL_LDLIBS += -ldl
diff --git a/media/libeffects/downmix/EffectDownmix.c b/media/libeffects/downmix/EffectDownmix.c
index f779876..4ee05f2 100644
--- a/media/libeffects/downmix/EffectDownmix.c
+++ b/media/libeffects/downmix/EffectDownmix.c
@@ -707,7 +707,7 @@
 
       case DOWNMIX_PARAM_TYPE:
         if (size != sizeof(downmix_type_t)) {
-            ALOGE("Downmix_setParameter(DOWNMIX_PARAM_TYPE) invalid size %d, should be %d",
+            ALOGE("Downmix_setParameter(DOWNMIX_PARAM_TYPE) invalid size %zu, should be %zu",
                     size, sizeof(downmix_type_t));
             return -EINVAL;
         }
@@ -760,7 +760,7 @@
 
     case DOWNMIX_PARAM_TYPE:
       if (*pSize < sizeof(int16_t)) {
-          ALOGE("Downmix_getParameter invalid parameter size %d for DOWNMIX_PARAM_TYPE", *pSize);
+          ALOGE("Downmix_getParameter invalid parameter size %zu for DOWNMIX_PARAM_TYPE", *pSize);
           return -EINVAL;
       }
       pValue16 = (int16_t *)pValue;
diff --git a/media/libeffects/factory/Android.mk b/media/libeffects/factory/Android.mk
index 60a6ce5..a932af7 100644
--- a/media/libeffects/factory/Android.mk
+++ b/media/libeffects/factory/Android.mk
@@ -9,7 +9,6 @@
 LOCAL_SHARED_LIBRARIES := \
 	libcutils liblog
 
-LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)
 LOCAL_MODULE:= libeffects
 
 LOCAL_SHARED_LIBRARIES += libdl
diff --git a/media/libeffects/factory/EffectsFactory.c b/media/libeffects/factory/EffectsFactory.c
index f8d6041..6d30d64 100644
--- a/media/libeffects/factory/EffectsFactory.c
+++ b/media/libeffects/factory/EffectsFactory.c
@@ -368,27 +368,21 @@
     }
     if (e1 == NULL) {
         ret = -ENOENT;
-        pthread_mutex_unlock(&gLibLock);
         goto exit;
     }
 
     // release effect in library
     if (fx->lib == NULL) {
         ALOGW("EffectRelease() fx %p library already unloaded", handle);
-        pthread_mutex_unlock(&gLibLock);
     } else {
         pthread_mutex_lock(&fx->lib->lock);
-        // Releasing the gLibLock here as the list access is over as the
-        // effect is removed from the list.
-        // If the gLibLock is not released, we will have a deadlock situation
-        // since we call the sub effect release inside the EffectRelease of Proxy
-        pthread_mutex_unlock(&gLibLock);
         fx->lib->desc->release_effect(fx->subItfe);
         pthread_mutex_unlock(&fx->lib->lock);
     }
     free(fx);
 
 exit:
+    pthread_mutex_unlock(&gLibLock);
     return ret;
 }
 
@@ -404,8 +398,8 @@
 // is pointed by the first argument. It searches the gSubEffectList for the
 // matching uuid and then copies the corresponding sub effect descriptors
 // to the inout param
-int EffectGetSubEffects(const effect_uuid_t *uuid,
-                        effect_descriptor_t *pDescriptors, size_t size)
+int EffectGetSubEffects(const effect_uuid_t *uuid, sub_effect_entry_t **pSube,
+                        size_t size)
 {
    ALOGV("EffectGetSubEffects() UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X"
           "%02X\n",uuid->timeLow, uuid->timeMid, uuid->timeHiAndVersion,
@@ -413,8 +407,7 @@
           uuid->node[3],uuid->node[4],uuid->node[5]);
 
    // Check if the size of the desc buffer is large enough for 2 subeffects
-   if ((uuid == NULL) || (pDescriptors == NULL) ||
-       (size < 2*sizeof(effect_descriptor_t))) {
+   if ((uuid == NULL) || (pSube == NULL) || (size < 2)) {
        ALOGW("NULL pointer or insufficient memory. Cannot query subeffects");
        return -EINVAL;
    }
@@ -432,11 +425,10 @@
            list_elem_t *subefx = e->sub_elem;
            while (subefx != NULL) {
                subeffect = (sub_effect_entry_t*)subefx->object;
-               d = (effect_descriptor_t*)(subeffect->object);
-               pDescriptors[count++] = *d;
+               pSube[count++] = subeffect;
                subefx = subefx->next;
            }
-           ALOGV("EffectGetSubEffects end - copied the sub effect descriptors");
+           ALOGV("EffectGetSubEffects end - copied the sub effect structures");
            return count;
        }
        e = e->next;
diff --git a/media/libeffects/factory/EffectsFactory.h b/media/libeffects/factory/EffectsFactory.h
index 147ff18..560b485 100644
--- a/media/libeffects/factory/EffectsFactory.h
+++ b/media/libeffects/factory/EffectsFactory.h
@@ -20,7 +20,7 @@
 #include <cutils/log.h>
 #include <pthread.h>
 #include <dirent.h>
-#include <media/EffectsFactoryApi.h>
+#include <hardware/audio_effect.h>
 
 #if __cplusplus
 extern "C" {
@@ -66,6 +66,32 @@
     void *object;
 } sub_effect_entry_t;
 
+
+////////////////////////////////////////////////////////////////////////////////
+//
+//    Function:       EffectGetSubEffects
+//
+//    Description:    Returns the descriptors of the sub effects of the effect
+//                    whose uuid is pointed to by first argument.
+//
+//    Input:
+//          pEffectUuid:    pointer to the effect uuid.
+//          size:           max number of sub_effect_entry_t * in pSube.
+//
+//    Input/Output:
+//          pSube:          address where to return the sub effect structures.
+//    Output:
+//        returned value:    0          successful operation.
+//                          -ENODEV     factory failed to initialize
+//                          -EINVAL     invalid pEffectUuid or pDescriptor
+//                          -ENOENT     no effect with this uuid found
+//        *pDescriptor:     updated with the sub effect descriptors.
+//
+////////////////////////////////////////////////////////////////////////////////
+int EffectGetSubEffects(const effect_uuid_t *pEffectUuid,
+                        sub_effect_entry_t **pSube,
+                        size_t size);
+
 #if __cplusplus
 }  // extern "C"
 #endif
diff --git a/media/libeffects/loudness/Android.mk b/media/libeffects/loudness/Android.mk
index dcb7b27..edf964e 100644
--- a/media/libeffects/loudness/Android.mk
+++ b/media/libeffects/loudness/Android.mk
@@ -14,7 +14,7 @@
 	liblog \
 	libstlport
 
-LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/soundfx
+LOCAL_MODULE_RELATIVE_PATH := soundfx
 LOCAL_MODULE:= libldnhncr
 
 LOCAL_C_INCLUDES := \
diff --git a/media/libeffects/lvm/lib/Eq/src/LVEQNB_Init.c b/media/libeffects/lvm/lib/Eq/src/LVEQNB_Init.c
index c4767a8..e01c1c5 100644
--- a/media/libeffects/lvm/lib/Eq/src/LVEQNB_Init.c
+++ b/media/libeffects/lvm/lib/Eq/src/LVEQNB_Init.c
@@ -25,6 +25,7 @@
 #include "LVEQNB.h"
 #include "LVEQNB_Private.h"
 #include "InstAlloc.h"
+#include <string.h> /* For memset */
 
 /****************************************************************************************/
 /*                                                                                      */
diff --git a/media/libeffects/lvm/wrapper/Android.mk b/media/libeffects/lvm/wrapper/Android.mk
index f1af389..68ba34c 100644
--- a/media/libeffects/lvm/wrapper/Android.mk
+++ b/media/libeffects/lvm/wrapper/Android.mk
@@ -13,7 +13,7 @@
 
 LOCAL_MODULE:= libbundlewrapper
 
-LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/soundfx
+LOCAL_MODULE_RELATIVE_PATH := soundfx
 
 LOCAL_STATIC_LIBRARIES += libmusicbundle
 
@@ -42,7 +42,7 @@
 
 LOCAL_MODULE:= libreverbwrapper
 
-LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/soundfx
+LOCAL_MODULE_RELATIVE_PATH := soundfx
 
 LOCAL_STATIC_LIBRARIES += libreverb
 
diff --git a/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp b/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp
index 28d239a..58d7767 100644
--- a/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp
+++ b/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp
@@ -138,22 +138,22 @@
 int  BassBoost_setParameter    (EffectContext *pContext, void *pParam, void *pValue);
 int  BassBoost_getParameter    (EffectContext *pContext,
                                void           *pParam,
-                               size_t         *pValueSize,
+                               uint32_t       *pValueSize,
                                void           *pValue);
 int  Virtualizer_setParameter  (EffectContext *pContext, void *pParam, void *pValue);
 int  Virtualizer_getParameter  (EffectContext *pContext,
                                void           *pParam,
-                               size_t         *pValueSize,
+                               uint32_t       *pValueSize,
                                void           *pValue);
 int  Equalizer_setParameter    (EffectContext *pContext, void *pParam, void *pValue);
 int  Equalizer_getParameter    (EffectContext *pContext,
                                 void          *pParam,
-                                size_t        *pValueSize,
+                                uint32_t      *pValueSize,
                                 void          *pValue);
 int  Volume_setParameter       (EffectContext *pContext, void *pParam, void *pValue);
 int  Volume_getParameter       (EffectContext *pContext,
                                 void          *pParam,
-                                size_t        *pValueSize,
+                                uint32_t      *pValueSize,
                                 void          *pValue);
 int Effect_setEnabled(EffectContext *pContext, bool enabled);
 
@@ -1758,7 +1758,7 @@
 
 int BassBoost_getParameter(EffectContext     *pContext,
                            void              *pParam,
-                           size_t            *pValueSize,
+                           uint32_t          *pValueSize,
                            void              *pValue){
     int status = 0;
     int32_t *pParamTemp = (int32_t *)pParam;
@@ -1876,7 +1876,7 @@
 
 int Virtualizer_getParameter(EffectContext        *pContext,
                              void                 *pParam,
-                             size_t               *pValueSize,
+                             uint32_t             *pValueSize,
                              void                 *pValue){
     int status = 0;
     int32_t *pParamTemp = (int32_t *)pParam;
@@ -1994,7 +1994,7 @@
 //----------------------------------------------------------------------------
 int Equalizer_getParameter(EffectContext     *pContext,
                            void              *pParam,
-                           size_t            *pValueSize,
+                           uint32_t          *pValueSize,
                            void              *pValue){
     int status = 0;
     int bMute = 0;
@@ -2252,7 +2252,7 @@
 
 int Volume_getParameter(EffectContext     *pContext,
                         void              *pParam,
-                        size_t            *pValueSize,
+                        uint32_t          *pValueSize,
                         void              *pValue){
     int status = 0;
     int bMute = 0;
@@ -2830,7 +2830,7 @@
 
                 p->status = android::BassBoost_getParameter(pContext,
                                                             p->data,
-                                                            (size_t  *)&p->vsize,
+                                                            &p->vsize,
                                                             p->data + voffset);
 
                 *replySize = sizeof(effect_param_t) + voffset + p->vsize;
@@ -2860,8 +2860,8 @@
                 int voffset = ((p->psize - 1) / sizeof(int32_t) + 1) * sizeof(int32_t);
 
                 p->status = android::Virtualizer_getParameter(pContext,
-                                                             (void *)p->data,
-                                                             (size_t  *)&p->vsize,
+                                                              (void *)p->data,
+                                                              &p->vsize,
                                                               p->data + voffset);
 
                 *replySize = sizeof(effect_param_t) + voffset + p->vsize;
@@ -2925,7 +2925,7 @@
 
                 p->status = android::Volume_getParameter(pContext,
                                                          (void *)p->data,
-                                                         (size_t  *)&p->vsize,
+                                                         &p->vsize,
                                                          p->data + voffset);
 
                 *replySize = sizeof(effect_param_t) + voffset + p->vsize;
diff --git a/media/libeffects/preprocessing/Android.mk b/media/libeffects/preprocessing/Android.mk
index c344352..9e8cb83 100644
--- a/media/libeffects/preprocessing/Android.mk
+++ b/media/libeffects/preprocessing/Android.mk
@@ -5,7 +5,7 @@
 
 LOCAL_MODULE:= libaudiopreprocessing
 LOCAL_MODULE_TAGS := optional
-LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/soundfx
+LOCAL_MODULE_RELATIVE_PATH := soundfx
 
 LOCAL_SRC_FILES:= \
     PreProcessing.cpp
diff --git a/media/libeffects/preprocessing/PreProcessing.cpp b/media/libeffects/preprocessing/PreProcessing.cpp
index d72eaf5..c56ff72 100644
--- a/media/libeffects/preprocessing/PreProcessing.cpp
+++ b/media/libeffects/preprocessing/PreProcessing.cpp
@@ -1233,8 +1233,8 @@
             if (session->framesIn < session->frameCount) {
                 return 0;
             }
-            size_t frIn = session->framesIn;
-            size_t frOut = session->apmFrameCount;
+            spx_uint32_t frIn = session->framesIn;
+            spx_uint32_t frOut = session->apmFrameCount;
             if (session->inChannelCount == 1) {
                 speex_resampler_process_int(session->inResampler,
                                             0,
@@ -1290,8 +1290,8 @@
         }
 
         if (session->outResampler != NULL) {
-            size_t frIn = session->apmFrameCount;
-            size_t frOut = session->frameCount;
+            spx_uint32_t frIn = session->apmFrameCount;
+            spx_uint32_t frOut = session->frameCount;
             if (session->inChannelCount == 1) {
                 speex_resampler_process_int(session->outResampler,
                                     0,
@@ -1754,8 +1754,8 @@
             if (session->framesRev < session->frameCount) {
                 return 0;
             }
-            size_t frIn = session->framesRev;
-            size_t frOut = session->apmFrameCount;
+            spx_uint32_t frIn = session->framesRev;
+            spx_uint32_t frOut = session->apmFrameCount;
             if (session->inChannelCount == 1) {
                 speex_resampler_process_int(session->revResampler,
                                             0,
diff --git a/media/libeffects/proxy/Android.mk b/media/libeffects/proxy/Android.mk
index 01b3be1..b438796 100644
--- a/media/libeffects/proxy/Android.mk
+++ b/media/libeffects/proxy/Android.mk
@@ -15,7 +15,7 @@
 LOCAL_PATH:= $(call my-dir)
 include $(CLEAR_VARS)
 LOCAL_MODULE:= libeffectproxy
-LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/soundfx
+LOCAL_MODULE_RELATIVE_PATH := soundfx
 LOCAL_MODULE_TAGS := optional
 
 
@@ -28,7 +28,8 @@
 
 LOCAL_C_INCLUDES := \
         system/media/audio_effects/include \
-        bionic/libc/include
+        bionic/libc/include \
+        frameworks/av/media/libeffects/factory
 
 include $(BUILD_SHARED_LIBRARY)
 
diff --git a/media/libeffects/proxy/EffectProxy.cpp b/media/libeffects/proxy/EffectProxy.cpp
index e6faaaf..62d3fd3 100644
--- a/media/libeffects/proxy/EffectProxy.cpp
+++ b/media/libeffects/proxy/EffectProxy.cpp
@@ -56,6 +56,8 @@
                            effect_handle_t  *pHandle) {
 
     effect_descriptor_t* desc;
+    audio_effect_library_t** aeli;
+    sub_effect_entry_t** sube;
     EffectContext* pContext;
     if (pHandle == NULL || uuid == NULL) {
         ALOGE("EffectProxyCreate() called with NULL pointer");
@@ -74,31 +76,52 @@
 
     // Get the HW and SW sub effect descriptors from the effects factory
     desc = new effect_descriptor_t[SUB_FX_COUNT];
+    aeli = new audio_effect_library_t*[SUB_FX_COUNT];
+    sube = new sub_effect_entry_t*[SUB_FX_COUNT];
+    pContext->sube = new sub_effect_entry_t*[SUB_FX_COUNT];
     pContext->desc = new effect_descriptor_t[SUB_FX_COUNT];
-    int retValue = EffectGetSubEffects(uuid, desc,
-                                sizeof(effect_descriptor_t) * SUB_FX_COUNT);
+    pContext->aeli = new audio_effect_library_t*[SUB_FX_COUNT];
+    int retValue = EffectGetSubEffects(uuid, sube, SUB_FX_COUNT);
     // EffectGetSubEffects returns the number of sub-effects copied.
     if (retValue != SUB_FX_COUNT) {
        ALOGE("EffectCreate() could not get the sub effects");
-       delete desc;
-       delete pContext->desc;
+       delete[] sube;
+       delete[] desc;
+       delete[] aeli;
+       delete[] pContext->sube;
+       delete[] pContext->desc;
+       delete[] pContext->aeli;
        return -EINVAL;
     }
     // Check which is the HW descriptor and copy the descriptors
     // to the Context desc array
     // Also check if there is only one HW and one SW descriptor.
     // HW descriptor alone has the HW_TUNNEL flag.
+    desc[0] = *(effect_descriptor_t*)(sube[0])->object;
+    desc[1] = *(effect_descriptor_t*)(sube[1])->object;
+    aeli[0] = sube[0]->lib->desc;
+    aeli[1] = sube[1]->lib->desc;
     if ((desc[0].flags & EFFECT_FLAG_HW_ACC_TUNNEL) &&
        !(desc[1].flags & EFFECT_FLAG_HW_ACC_TUNNEL)) {
+        pContext->sube[SUB_FX_OFFLOAD] = sube[0];
         pContext->desc[SUB_FX_OFFLOAD] = desc[0];
+        pContext->aeli[SUB_FX_OFFLOAD] = aeli[0];
+        pContext->sube[SUB_FX_HOST] = sube[1];
         pContext->desc[SUB_FX_HOST] = desc[1];
+        pContext->aeli[SUB_FX_HOST] = aeli[1];
     }
     else if ((desc[1].flags & EFFECT_FLAG_HW_ACC_TUNNEL) &&
              !(desc[0].flags & EFFECT_FLAG_HW_ACC_TUNNEL)) {
+        pContext->sube[SUB_FX_HOST] = sube[0];
         pContext->desc[SUB_FX_HOST] = desc[0];
+        pContext->aeli[SUB_FX_HOST] = aeli[0];
+        pContext->sube[SUB_FX_OFFLOAD] = sube[1];
         pContext->desc[SUB_FX_OFFLOAD] = desc[1];
+        pContext->aeli[SUB_FX_OFFLOAD] = aeli[1];
     }
-    delete desc;
+    delete[] desc;
+    delete[] aeli;
+    delete[] sube;
 #if (LOG_NDEBUG == 0)
     effect_uuid_t uuid_print = pContext->desc[SUB_FX_HOST].uuid;
     ALOGV("EffectCreate() UUID of HOST: %08X-%04X-%04X-%04X-%02X%02X%02X%02X"
@@ -128,13 +151,15 @@
         return -EINVAL;
     }
     ALOGV("EffectRelease");
-    delete pContext->desc;
+    delete[] pContext->desc;
     free(pContext->replyData);
 
     if (pContext->eHandle[SUB_FX_HOST])
-       EffectRelease(pContext->eHandle[SUB_FX_HOST]);
+       pContext->aeli[SUB_FX_HOST]->release_effect(pContext->eHandle[SUB_FX_HOST]);
     if (pContext->eHandle[SUB_FX_OFFLOAD])
-       EffectRelease(pContext->eHandle[SUB_FX_OFFLOAD]);
+       pContext->aeli[SUB_FX_OFFLOAD]->release_effect(pContext->eHandle[SUB_FX_OFFLOAD]);
+    delete[] pContext->aeli;
+    delete[] pContext->sube;
     delete pContext;
     pContext = NULL;
     return 0;
@@ -187,7 +212,8 @@
     }
     if (pContext->eHandle[SUB_FX_HOST] == NULL) {
         ALOGV("Effect_command() Calling HOST EffectCreate");
-        status = EffectCreate(&pContext->desc[SUB_FX_HOST].uuid,
+        status = pContext->aeli[SUB_FX_HOST]->create_effect(
+                              &pContext->desc[SUB_FX_HOST].uuid,
                               pContext->sessionId, pContext->ioId,
                               &(pContext->eHandle[SUB_FX_HOST]));
         if (status != NO_ERROR || (pContext->eHandle[SUB_FX_HOST] == NULL)) {
@@ -197,11 +223,13 @@
     }
     if (pContext->eHandle[SUB_FX_OFFLOAD] == NULL) {
         ALOGV("Effect_command() Calling OFFLOAD EffectCreate");
-        status = EffectCreate(&pContext->desc[SUB_FX_OFFLOAD].uuid,
+        status = pContext->aeli[SUB_FX_OFFLOAD]->create_effect(
+                              &pContext->desc[SUB_FX_OFFLOAD].uuid,
                               pContext->sessionId, pContext->ioId,
                               &(pContext->eHandle[SUB_FX_OFFLOAD]));
         if (status != NO_ERROR || (pContext->eHandle[SUB_FX_OFFLOAD] == NULL)) {
             ALOGV("Effect_command() Error creating HW effect");
+            pContext->eHandle[SUB_FX_OFFLOAD] = NULL;
             // Do not return error here as SW effect is created
             // Return error if the CMD_OFFLOAD sends the index as OFFLOAD
         }
@@ -233,11 +261,17 @@
         // Update the DSP wrapper with the new ioHandle.
         // Pass the OFFLOAD command to the wrapper.
         // The DSP wrapper needs to handle this CMD
-        if (pContext->eHandle[SUB_FX_OFFLOAD])
-            status = (*pContext->eHandle[SUB_FX_OFFLOAD])->command(
-                             pContext->eHandle[SUB_FX_OFFLOAD], cmdCode, cmdSize,
-                             pCmdData, replySize, pReplyData);
-        return status;
+        if (pContext->eHandle[SUB_FX_OFFLOAD]) {
+            ALOGV("Effect_command: Calling OFFLOAD command");
+            return (*pContext->eHandle[SUB_FX_OFFLOAD])->command(
+                           pContext->eHandle[SUB_FX_OFFLOAD], cmdCode, cmdSize,
+                           pCmdData, replySize, pReplyData);
+        }
+        *(int*)pReplyData = NO_ERROR;
+        ALOGV("Effect_command OFFLOAD return 0, replyData %d",
+                                                *(int*)pReplyData);
+
+        return NO_ERROR;
     }
 
     int index = pContext->index;
diff --git a/media/libeffects/proxy/EffectProxy.h b/media/libeffects/proxy/EffectProxy.h
index acbe17e..046b93e 100644
--- a/media/libeffects/proxy/EffectProxy.h
+++ b/media/libeffects/proxy/EffectProxy.h
@@ -16,6 +16,8 @@
 
 #include <hardware/audio.h>
 #include <hardware/audio_effect.h>
+#include "EffectsFactory.h"
+
 namespace android {
 enum {
     SUB_FX_HOST,       // Index of HOST in the descriptor and handle arrays
@@ -62,7 +64,9 @@
 
 struct EffectContext {
   const struct effect_interface_s  *common_itfe; // Holds the itfe of the Proxy
+  sub_effect_entry_t** sube;                     // Points to the sub effects
   effect_descriptor_t*  desc;                    // Points to the sub effect descriptors
+  audio_effect_library_t**  aeli;                // Points to the sub effect aeli
   effect_handle_t       eHandle[SUB_FX_COUNT];   // The effect handles of the sub effects
   int                   index;       // The index that is currently active - HOST or OFFLOAD
   int32_t               sessionId;   // The sessiond in which the effect is created.
diff --git a/media/libeffects/testlibs/Android.mk_ b/media/libeffects/testlibs/Android.mk_
index 2954908..672ebba 100644
--- a/media/libeffects/testlibs/Android.mk_
+++ b/media/libeffects/testlibs/Android.mk_
@@ -11,7 +11,7 @@
 LOCAL_SHARED_LIBRARIES := \
 	libcutils
 
-LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/soundfx
+LOCAL_MODULE_RELATIVE_PATH := soundfx
 LOCAL_MODULE:= libreverbtest
 
 ifeq ($(TARGET_OS)-$(TARGET_SIMULATOR),linux-true)
@@ -47,7 +47,7 @@
 LOCAL_SHARED_LIBRARIES := \
 	libcutils
 
-LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/soundfx
+LOCAL_MODULE_RELATIVE_PATH := soundfx
 LOCAL_MODULE:= libequalizertest
 
 ifeq ($(TARGET_OS)-$(TARGET_SIMULATOR),linux-true)
diff --git a/media/libeffects/visualizer/Android.mk b/media/libeffects/visualizer/Android.mk
index e196eb2..dd2d306 100644
--- a/media/libeffects/visualizer/Android.mk
+++ b/media/libeffects/visualizer/Android.mk
@@ -13,7 +13,7 @@
 	liblog \
 	libdl
 
-LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/soundfx
+LOCAL_MODULE_RELATIVE_PATH := soundfx
 LOCAL_MODULE:= libvisualizer
 
 LOCAL_C_INCLUDES := \
diff --git a/media/libmedia/Android.mk b/media/libmedia/Android.mk
index 8aa54dc..fc4b2a5 100644
--- a/media/libmedia/Android.mk
+++ b/media/libmedia/Android.mk
@@ -25,6 +25,8 @@
     AudioRecord.cpp \
     AudioSystem.cpp \
     mediaplayer.cpp \
+    IMediaHTTPConnection.cpp \
+    IMediaHTTPService.cpp \
     IMediaLogService.cpp \
     IMediaPlayerService.cpp \
     IMediaPlayerClient.cpp \
diff --git a/media/libmedia/AudioEffect.cpp b/media/libmedia/AudioEffect.cpp
index 8dfffb3..35f6557 100644
--- a/media/libmedia/AudioEffect.cpp
+++ b/media/libmedia/AudioEffect.cpp
@@ -380,9 +380,9 @@
 }
 
 void AudioEffect::commandExecuted(uint32_t cmdCode,
-                                  uint32_t cmdSize,
+                                  uint32_t cmdSize __unused,
                                   void *cmdData,
-                                  uint32_t replySize,
+                                  uint32_t replySize __unused,
                                   void *replyData)
 {
     if (cmdData == NULL || replyData == NULL) {
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 6ec8ed2..6ca499b 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -52,7 +52,7 @@
     }
 
     if (size == 0) {
-        ALOGE("Unsupported configuration: sampleRate %u, format %d, channelMask %#x",
+        ALOGE("Unsupported configuration: sampleRate %u, format %#x, channelMask %#x",
             sampleRate, format, channelMask);
         return BAD_VALUE;
     }
@@ -71,7 +71,7 @@
 // ---------------------------------------------------------------------------
 
 AudioRecord::AudioRecord()
-    : mStatus(NO_INIT), mSessionId(0),
+    : mStatus(NO_INIT), mSessionId(AUDIO_SESSION_ALLOCATE),
       mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(SP_DEFAULT)
 {
 }
@@ -87,8 +87,8 @@
         int notificationFrames,
         int sessionId,
         transfer_type transferType,
-        audio_input_flags_t flags)
-    : mStatus(NO_INIT), mSessionId(0),
+        audio_input_flags_t flags __unused)
+    : mStatus(NO_INIT), mSessionId(AUDIO_SESSION_ALLOCATE),
       mPreviousPriority(ANDROID_PRIORITY_NORMAL),
       mPreviousSchedulingGroup(SP_DEFAULT),
       mProxy(NULL)
@@ -115,7 +115,7 @@
             mAudioRecord.clear();
         }
         IPCThreadState::self()->flushCommands();
-        AudioSystem::releaseAudioSessionId(mSessionId);
+        AudioSystem::releaseAudioSessionId(mSessionId, -1);
     }
 }
 
@@ -191,12 +191,12 @@
 
     // validate parameters
     if (!audio_is_valid_format(format)) {
-        ALOGE("Invalid format %d", format);
+        ALOGE("Invalid format %#x", format);
         return BAD_VALUE;
     }
     // Temporary restriction: AudioFlinger currently supports 16-bit PCM only
     if (format != AUDIO_FORMAT_PCM_16_BIT) {
-        ALOGE("Format %d is not supported", format);
+        ALOGE("Format %#x is not supported", format);
         return BAD_VALUE;
     }
     mFormat = format;
@@ -228,12 +228,13 @@
         ALOGE("frameCount %u < minFrameCount %u", frameCount, minFrameCount);
         return BAD_VALUE;
     }
-    mFrameCount = frameCount;
+    // mFrameCount is initialized in openRecord_l
+    mReqFrameCount = frameCount;
 
     mNotificationFramesReq = notificationFrames;
     mNotificationFramesAct = 0;
 
-    if (sessionId == 0 ) {
+    if (sessionId == AUDIO_SESSION_ALLOCATE) {
         mSessionId = AudioSystem::newAudioSessionId();
     } else {
         mSessionId = sessionId;
@@ -244,7 +245,7 @@
 
     // create the IAudioRecord
     status = openRecord_l(0 /*epoch*/);
-    if (status) {
+    if (status != NO_ERROR) {
         return status;
     }
 
@@ -255,9 +256,6 @@
 
     mStatus = NO_ERROR;
 
-    // Update buffer size in case it has been limited by AudioFlinger during track creation
-    mFrameCount = mCblk->frameCount_;
-
     mActive = false;
     mCbf = cbf;
     mRefreshRemaining = true;
@@ -268,7 +266,7 @@
     mMarkerReached = false;
     mNewPosition = 0;
     mUpdatePeriod = 0;
-    AudioSystem::acquireAudioSessionId(mSessionId);
+    AudioSystem::acquireAudioSessionId(mSessionId, -1);
     mSequence = 1;
     mObservedSequence = mSequence;
     mInOverrun = false;
@@ -289,6 +287,9 @@
 
     // reset current position as seen by client to 0
     mProxy->setEpoch(mProxy->getEpoch() - mProxy->getPosition());
+    // force refresh of remaining frames by processAudioBuffer() as last
+    // read before stop could be partial.
+    mRefreshRemaining = true;
 
     mNewPosition = mProxy->getPosition() + mUpdatePeriod;
     int32_t flags = android_atomic_acquire_load(&mCblk->mFlags);
@@ -352,6 +353,7 @@
 
 status_t AudioRecord::setMarkerPosition(uint32_t marker)
 {
+    // The only purpose of setting marker position is to get a callback
     if (mCbf == NULL) {
         return INVALID_OPERATION;
     }
@@ -377,6 +379,7 @@
 
 status_t AudioRecord::setPositionUpdatePeriod(uint32_t updatePeriod)
 {
+    // The only purpose of setting position update period is to get a callback
     if (mCbf == NULL) {
         return INVALID_OPERATION;
     }
@@ -412,7 +415,7 @@
     return NO_ERROR;
 }
 
-unsigned int AudioRecord::getInputFramesLost() const
+uint32_t AudioRecord::getInputFramesLost() const
 {
     // no need to check mActive, because if inactive this will return 0, which is what we want
     return AudioSystem::getInputFramesLost(getInput());
@@ -447,11 +450,12 @@
     }
 
     mNotificationFramesAct = mNotificationFramesReq;
+    size_t frameCount = mReqFrameCount;
 
     if (!(mFlags & AUDIO_INPUT_FLAG_FAST)) {
         // Make sure that application is notified with sufficient margin before overrun
-        if (mNotificationFramesAct == 0 || mNotificationFramesAct > mFrameCount/2) {
-            mNotificationFramesAct = mFrameCount/2;
+        if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/2) {
+            mNotificationFramesAct = frameCount/2;
         }
     }
 
@@ -461,24 +465,31 @@
         ALOGE("Could not get audio input for record source %d", mInputSource);
         return BAD_VALUE;
     }
+    {
+    // Now that we have a reference to an I/O handle and have not yet handed it off to AudioFlinger,
+    // we must release it ourselves if anything goes wrong.
 
+    size_t temp = frameCount;   // temp may be replaced by a revised value of frameCount,
+                                // but we will still need the original value also
     int originalSessionId = mSessionId;
     sp<IAudioRecord> record = audioFlinger->openRecord(input,
                                                        mSampleRate, mFormat,
                                                        mChannelMask,
-                                                       mFrameCount,
+                                                       &temp,
                                                        &trackFlags,
                                                        tid,
                                                        &mSessionId,
                                                        &status);
-    ALOGE_IF(originalSessionId != 0 && mSessionId != originalSessionId,
+    ALOGE_IF(originalSessionId != AUDIO_SESSION_ALLOCATE && mSessionId != originalSessionId,
             "session ID changed from %d to %d", originalSessionId, mSessionId);
 
     if (record == 0 || status != NO_ERROR) {
         ALOGE("AudioFlinger could not create record track, status: %d", status);
-        AudioSystem::releaseInput(input);
-        return status;
+        goto release;
     }
+    // AudioFlinger now owns the reference to the I/O handle,
+    // so we are no longer responsible for releasing it.
+
     sp<IMemory> iMem = record->getCblk();
     if (iMem == 0) {
         ALOGE("Could not get control block");
@@ -493,11 +504,24 @@
         mAudioRecord->asBinder()->unlinkToDeath(mDeathNotifier, this);
         mDeathNotifier.clear();
     }
+
+    // We retain a copy of the I/O handle, but don't own the reference
     mInput = input;
     mAudioRecord = record;
     mCblkMemory = iMem;
     audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer);
     mCblk = cblk;
+    // note that temp is the (possibly revised) value of mFrameCount
+    if (temp < frameCount || (frameCount == 0 && temp == 0)) {
+        ALOGW("Requested frameCount %u but received frameCount %u", frameCount, temp);
+    }
+    frameCount = temp;
+    // If IAudioRecord is re-created, don't let the requested frameCount
+    // decrease.  This can confuse clients that cache frameCount().
+    if (frameCount > mReqFrameCount) {
+        mReqFrameCount = frameCount;
+    }
+
     // FIXME missing fast track frameCount logic
     mAwaitBoost = false;
     if (mFlags & AUDIO_INPUT_FLAG_FAST) {
@@ -521,6 +545,8 @@
     // starting address of buffers in shared memory
     void *buffers = (char*)cblk + sizeof(audio_track_cblk_t);
 
+    mFrameCount = frameCount;
+
     // update proxy
     mProxy = new AudioRecordClientProxy(cblk, buffers, mFrameCount, mFrameSize);
     mProxy->setEpoch(epoch);
@@ -530,6 +556,14 @@
     mAudioRecord->asBinder()->linkToDeath(mDeathNotifier, this);
 
     return NO_ERROR;
+    }
+
+release:
+    AudioSystem::releaseInput(input);
+    if (status == NO_ERROR) {
+        status = NO_INIT;
+    }
+    return status;
 }
 
 status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
@@ -545,13 +579,13 @@
     }
 
     const struct timespec *requested;
+    struct timespec timeout;
     if (waitCount == -1) {
         requested = &ClientProxy::kForever;
     } else if (waitCount == 0) {
         requested = &ClientProxy::kNonBlocking;
     } else if (waitCount > 0) {
         long long ms = WAIT_PERIOD_MS * (long long) waitCount;
-        struct timespec timeout;
         timeout.tv_sec = ms / 1000;
         timeout.tv_nsec = (int) (ms % 1000) * 1000000;
         requested = &timeout;
@@ -591,6 +625,9 @@
                 if (newSequence == oldSequence) {
                     status = restoreRecord_l("obtainBuffer");
                     if (status != NO_ERROR) {
+                        buffer.mFrameCount = 0;
+                        buffer.mRaw = NULL;
+                        buffer.mNonContig = 0;
                         break;
                     }
                 }
@@ -767,10 +804,10 @@
         mRetryOnPartialBuffer = false;
     }
     size_t misalignment = mProxy->getMisalignment();
-    int32_t sequence = mSequence;
+    uint32_t sequence = mSequence;
 
     // These fields don't need to be cached, because they are assigned only by set():
-    //      mTransfer, mCbf, mUserData, mSampleRate
+    //      mTransfer, mCbf, mUserData, mSampleRate, mFrameSize
 
     mLock.unlock();
 
@@ -844,8 +881,8 @@
                 "obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount);
         requested = &ClientProxy::kNonBlocking;
         size_t avail = audioBuffer.frameCount + nonContig;
-        ALOGV("obtainBuffer(%u) returned %u = %u + %u",
-                mRemainingFrames, avail, audioBuffer.frameCount, nonContig);
+        ALOGV("obtainBuffer(%u) returned %u = %u + %u err %d",
+                mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err);
         if (err != NO_ERROR) {
             if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR) {
                 break;
@@ -966,7 +1003,8 @@
 // =========================================================================
 
 AudioRecord::AudioRecordThread::AudioRecordThread(AudioRecord& receiver, bool bCanCallJava)
-    : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL)
+    : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL),
+      mIgnoreNextPausedInt(false)
 {
 }
 
@@ -983,6 +1021,10 @@
             // caller will check for exitPending()
             return true;
         }
+        if (mIgnoreNextPausedInt) {
+            mIgnoreNextPausedInt = false;
+            mPausedInt = false;
+        }
         if (mPausedInt) {
             if (mPausedNs > 0) {
                 (void) mMyCond.waitRelative(mMyLock, mPausedNs);
@@ -1017,12 +1059,7 @@
 {
     // must be in this order to avoid a race condition
     Thread::requestExit();
-    AutoMutex _l(mMyLock);
-    if (mPaused || mPausedInt) {
-        mPaused = false;
-        mPausedInt = false;
-        mMyCond.signal();
-    }
+    resume();
 }
 
 void AudioRecord::AudioRecordThread::pause()
@@ -1034,6 +1071,7 @@
 void AudioRecord::AudioRecordThread::resume()
 {
     AutoMutex _l(mMyLock);
+    mIgnoreNextPausedInt = true;
     if (mPaused || mPausedInt) {
         mPaused = false;
         mPausedInt = false;
diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp
index 9a76111..140fb66 100644
--- a/media/libmedia/AudioSystem.cpp
+++ b/media/libmedia/AudioSystem.cpp
@@ -40,10 +40,10 @@
 DefaultKeyedVector<audio_io_handle_t, AudioSystem::OutputDescriptor *> AudioSystem::gOutputs(0);
 
 // Cached values for recording queries, all protected by gLock
-uint32_t AudioSystem::gPrevInSamplingRate = 16000;
-audio_format_t AudioSystem::gPrevInFormat = AUDIO_FORMAT_PCM_16_BIT;
-audio_channel_mask_t AudioSystem::gPrevInChannelMask = AUDIO_CHANNEL_IN_MONO;
-size_t AudioSystem::gInBuffSize = 0;
+uint32_t AudioSystem::gPrevInSamplingRate;
+audio_format_t AudioSystem::gPrevInFormat;
+audio_channel_mask_t AudioSystem::gPrevInChannelMask;
+size_t AudioSystem::gInBuffSize = 0;    // zero indicates cache is invalid
 
 
 // establish binder interface to AudioFlinger service
@@ -84,13 +84,15 @@
     return DEAD_OBJECT;
 }
 
-status_t AudioSystem::muteMicrophone(bool state) {
+status_t AudioSystem::muteMicrophone(bool state)
+{
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
     if (af == 0) return PERMISSION_DENIED;
     return af->setMicMute(state);
 }
 
-status_t AudioSystem::isMicrophoneMuted(bool* state) {
+status_t AudioSystem::isMicrophoneMuted(bool* state)
+{
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
     if (af == 0) return PERMISSION_DENIED;
     *state = af->getMicMute();
@@ -175,13 +177,15 @@
     return af->setMode(mode);
 }
 
-status_t AudioSystem::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) {
+status_t AudioSystem::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs)
+{
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
     if (af == 0) return PERMISSION_DENIED;
     return af->setParameters(ioHandle, keyValuePairs);
 }
 
-String8 AudioSystem::getParameters(audio_io_handle_t ioHandle, const String8& keys) {
+String8 AudioSystem::getParameters(audio_io_handle_t ioHandle, const String8& keys)
+{
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
     String8 result = String8("");
     if (af == 0) return result;
@@ -259,6 +263,11 @@
         *samplingRate = outputDesc->samplingRate;
         gLock.unlock();
     }
+    if (*samplingRate == 0) {
+        ALOGE("AudioSystem::getSamplingRate failed for output %d stream type %d",
+                output, streamType);
+        return BAD_VALUE;
+    }
 
     ALOGV("getSamplingRate() streamType %d, output %d, sampling rate %u", streamType, output,
             *samplingRate);
@@ -299,6 +308,11 @@
         *frameCount = outputDesc->frameCount;
         gLock.unlock();
     }
+    if (*frameCount == 0) {
+        ALOGE("AudioSystem::getFrameCount failed for output %d stream type %d",
+                output, streamType);
+        return BAD_VALUE;
+    }
 
     ALOGV("getFrameCount() streamType %d, output %d, frameCount %d", streamType, output,
             *frameCount);
@@ -323,7 +337,7 @@
 }
 
 status_t AudioSystem::getLatency(audio_io_handle_t output,
-                                 audio_stream_type_t streamType,
+                                 audio_stream_type_t streamType __unused,
                                  uint32_t* latency)
 {
     OutputDescriptor *outputDesc;
@@ -359,6 +373,12 @@
             return PERMISSION_DENIED;
         }
         inBuffSize = af->getInputBufferSize(sampleRate, format, channelMask);
+        if (inBuffSize == 0) {
+            ALOGE("AudioSystem::getInputBufferSize failed sampleRate %d format %#x channelMask %x",
+                    sampleRate, format, channelMask);
+            return BAD_VALUE;
+        }
+        // A benign race is possible here: we could overwrite a fresher cache entry
         gLock.lock();
         // save the request params
         gPrevInSamplingRate = sampleRate;
@@ -380,8 +400,8 @@
     return af->setVoiceVolume(value);
 }
 
-status_t AudioSystem::getRenderPosition(audio_io_handle_t output, size_t *halFrames,
-                                        size_t *dspFrames, audio_stream_type_t stream)
+status_t AudioSystem::getRenderPosition(audio_io_handle_t output, uint32_t *halFrames,
+                                        uint32_t *dspFrames, audio_stream_type_t stream)
 {
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
     if (af == 0) return PERMISSION_DENIED;
@@ -397,9 +417,10 @@
     return af->getRenderPosition(halFrames, dspFrames, output);
 }
 
-size_t AudioSystem::getInputFramesLost(audio_io_handle_t ioHandle) {
+uint32_t AudioSystem::getInputFramesLost(audio_io_handle_t ioHandle)
+{
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
-    unsigned int result = 0;
+    uint32_t result = 0;
     if (af == 0) return result;
     if (ioHandle == 0) return result;
 
@@ -407,29 +428,33 @@
     return result;
 }
 
-int AudioSystem::newAudioSessionId() {
+int AudioSystem::newAudioSessionId()
+{
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
     if (af == 0) return 0;
     return af->newAudioSessionId();
 }
 
-void AudioSystem::acquireAudioSessionId(int audioSession) {
+void AudioSystem::acquireAudioSessionId(int audioSession, pid_t pid)
+{
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
     if (af != 0) {
-        af->acquireAudioSessionId(audioSession);
+        af->acquireAudioSessionId(audioSession, pid);
     }
 }
 
-void AudioSystem::releaseAudioSessionId(int audioSession) {
+void AudioSystem::releaseAudioSessionId(int audioSession, pid_t pid)
+{
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
     if (af != 0) {
-        af->releaseAudioSessionId(audioSession);
+        af->releaseAudioSessionId(audioSession, pid);
     }
 }
 
 // ---------------------------------------------------------------------------
 
-void AudioSystem::AudioFlingerClient::binderDied(const wp<IBinder>& who __unused) {
+void AudioSystem::AudioFlingerClient::binderDied(const wp<IBinder>& who __unused)
+{
     Mutex::Autolock _l(AudioSystem::gLock);
 
     AudioSystem::gAudioFlinger.clear();
@@ -465,7 +490,7 @@
 
         OutputDescriptor *outputDesc =  new OutputDescriptor(*desc);
         gOutputs.add(ioHandle, outputDesc);
-        ALOGV("ioConfigChanged() new output samplingRate %u, format %d channel mask %#x frameCount %u "
+        ALOGV("ioConfigChanged() new output samplingRate %u, format %#x channel mask %#x frameCount %u "
                 "latency %d",
                 outputDesc->samplingRate, outputDesc->format, outputDesc->channelMask,
                 outputDesc->frameCount, outputDesc->latency);
@@ -489,7 +514,7 @@
         if (param2 == NULL) break;
         desc = (const OutputDescriptor *)param2;
 
-        ALOGV("ioConfigChanged() new config for output %d samplingRate %u, format %d channel mask %#x "
+        ALOGV("ioConfigChanged() new config for output %d samplingRate %u, format %#x channel mask %#x "
                 "frameCount %d latency %d",
                 ioHandle, desc->samplingRate, desc->format,
                 desc->channelMask, desc->frameCount, desc->latency);
@@ -506,12 +531,14 @@
     }
 }
 
-void AudioSystem::setErrorCallback(audio_error_callback cb) {
+void AudioSystem::setErrorCallback(audio_error_callback cb)
+{
     Mutex::Autolock _l(gLock);
     gAudioErrorCallback = cb;
 }
 
-bool AudioSystem::routedToA2dpOutput(audio_stream_type_t streamType) {
+bool AudioSystem::routedToA2dpOutput(audio_stream_type_t streamType)
+{
     switch (streamType) {
     case AUDIO_STREAM_MUSIC:
     case AUDIO_STREAM_VOICE_CALL:
@@ -719,7 +746,8 @@
 audio_io_handle_t AudioSystem::getOutputForEffect(const effect_descriptor_t *desc)
 {
     const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
-    if (aps == 0) return PERMISSION_DENIED;
+    // FIXME change return type to status_t, and return PERMISSION_DENIED here
+    if (aps == 0) return 0;
     return aps->getOutputForEffect(desc);
 }
 
@@ -814,7 +842,8 @@
 
 // ---------------------------------------------------------------------------
 
-void AudioSystem::AudioPolicyServiceClient::binderDied(const wp<IBinder>& who __unused) {
+void AudioSystem::AudioPolicyServiceClient::binderDied(const wp<IBinder>& who __unused)
+{
     Mutex::Autolock _l(AudioSystem::gLock);
     AudioSystem::gAudioPolicyService.clear();
 
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index ddd4612..5c62260 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -44,9 +44,6 @@
         return BAD_VALUE;
     }
 
-    // default to 0 in case of error
-    *frameCount = 0;
-
     // FIXME merge with similar code in createTrack_l(), except we're missing
     //       some information here that is available in createTrack_l():
     //          audio_io_handle_t output
@@ -54,16 +51,26 @@
     //          audio_channel_mask_t channelMask
     //          audio_output_flags_t flags
     uint32_t afSampleRate;
-    if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
-        return NO_INIT;
+    status_t status;
+    status = AudioSystem::getOutputSamplingRate(&afSampleRate, streamType);
+    if (status != NO_ERROR) {
+        ALOGE("Unable to query output sample rate for stream type %d; status %d",
+                streamType, status);
+        return status;
     }
     size_t afFrameCount;
-    if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) {
-        return NO_INIT;
+    status = AudioSystem::getOutputFrameCount(&afFrameCount, streamType);
+    if (status != NO_ERROR) {
+        ALOGE("Unable to query output frame count for stream type %d; status %d",
+                streamType, status);
+        return status;
     }
     uint32_t afLatency;
-    if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) {
-        return NO_INIT;
+    status = AudioSystem::getOutputLatency(&afLatency, streamType);
+    if (status != NO_ERROR) {
+        ALOGE("Unable to query output latency for stream type %d; status %d",
+                streamType, status);
+        return status;
     }
 
     // Ensure that buffer depth covers at least audio hardware latency
@@ -74,6 +81,13 @@
 
     *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount :
             afFrameCount * minBufCount * sampleRate / afSampleRate;
+    // The formula above should always produce a non-zero value, but return an error
+    // in the unlikely event that it does not, as that's part of the API contract.
+    if (*frameCount == 0) {
+        ALOGE("AudioTrack::getMinFrameCount failed for streamType %d, sampleRate %d",
+                streamType, sampleRate);
+        return BAD_VALUE;
+    }
     ALOGV("getMinFrameCount=%d: afFrameCount=%d, minBufCount=%d, afSampleRate=%d, afLatency=%d",
             *frameCount, afFrameCount, minBufCount, afSampleRate, afLatency);
     return NO_ERROR;
@@ -102,7 +116,8 @@
         int sessionId,
         transfer_type transferType,
         const audio_offload_info_t *offloadInfo,
-        int uid)
+        int uid,
+        pid_t pid)
     : mStatus(NO_INIT),
       mIsTimed(false),
       mPreviousPriority(ANDROID_PRIORITY_NORMAL),
@@ -111,7 +126,7 @@
     mStatus = set(streamType, sampleRate, format, channelMask,
             frameCount, flags, cbf, user, notificationFrames,
             0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType,
-            offloadInfo, uid);
+            offloadInfo, uid, pid);
 }
 
 AudioTrack::AudioTrack(
@@ -127,7 +142,8 @@
         int sessionId,
         transfer_type transferType,
         const audio_offload_info_t *offloadInfo,
-        int uid)
+        int uid,
+        pid_t pid)
     : mStatus(NO_INIT),
       mIsTimed(false),
       mPreviousPriority(ANDROID_PRIORITY_NORMAL),
@@ -135,7 +151,8 @@
 {
     mStatus = set(streamType, sampleRate, format, channelMask,
             0 /*frameCount*/, flags, cbf, user, notificationFrames,
-            sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo, uid);
+            sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo,
+            uid, pid);
 }
 
 AudioTrack::~AudioTrack()
@@ -154,7 +171,9 @@
         mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this);
         mAudioTrack.clear();
         IPCThreadState::self()->flushCommands();
-        AudioSystem::releaseAudioSessionId(mSessionId);
+        ALOGV("~AudioTrack, releasing session id from %d on behalf of %d",
+                IPCThreadState::self()->getCallingPid(), mClientPid);
+        AudioSystem::releaseAudioSessionId(mSessionId, mClientPid);
     }
 }
 
@@ -173,7 +192,8 @@
         int sessionId,
         transfer_type transferType,
         const audio_offload_info_t *offloadInfo,
-        int uid)
+        int uid,
+        pid_t pid)
 {
     switch (transferType) {
     case TRANSFER_DEFAULT:
@@ -208,6 +228,7 @@
         ALOGE("Invalid transfer type %d", transferType);
         return BAD_VALUE;
     }
+    mSharedBuffer = sharedBuffer;
     mTransfer = transferType;
 
     // FIXME "int" here is legacy and will be replaced by size_t later
@@ -230,19 +251,24 @@
         return INVALID_OPERATION;
     }
 
-    mOutput = 0;
-
     // handle default values first.
     if (streamType == AUDIO_STREAM_DEFAULT) {
         streamType = AUDIO_STREAM_MUSIC;
     }
+    if (uint32_t(streamType) >= AUDIO_STREAM_CNT) {
+        ALOGE("Invalid stream type %d", streamType);
+        return BAD_VALUE;
+    }
+    mStreamType = streamType;
 
+    status_t status;
     if (sampleRate == 0) {
-        uint32_t afSampleRate;
-        if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
-            return NO_INIT;
+        status = AudioSystem::getOutputSamplingRate(&sampleRate, streamType);
+        if (status != NO_ERROR) {
+            ALOGE("Could not get output sample rate for stream type %d; status %d",
+                    streamType, status);
+            return status;
         }
-        sampleRate = afSampleRate;
     }
     mSampleRate = sampleRate;
 
@@ -253,9 +279,10 @@
 
     // validate parameters
     if (!audio_is_valid_format(format)) {
-        ALOGE("Invalid format %d", format);
+        ALOGE("Invalid format %#x", format);
         return BAD_VALUE;
     }
+    mFormat = format;
 
     if (!audio_is_output_channel(channelMask)) {
         ALOGE("Invalid channel mask %#x", channelMask);
@@ -296,30 +323,36 @@
         mFrameSizeAF = sizeof(uint8_t);
     }
 
-    audio_io_handle_t output = AudioSystem::getOutput(
-                                    streamType,
-                                    sampleRate, format, channelMask,
-                                    flags,
-                                    offloadInfo);
-
-    if (output == 0) {
-        ALOGE("Could not get audio output for stream type %d", streamType);
-        return BAD_VALUE;
+    // Make copy of input parameter offloadInfo so that in the future:
+    //  (a) createTrack_l doesn't need it as an input parameter
+    //  (b) we can support re-creation of offloaded tracks
+    if (offloadInfo != NULL) {
+        mOffloadInfoCopy = *offloadInfo;
+        mOffloadInfo = &mOffloadInfoCopy;
+    } else {
+        mOffloadInfo = NULL;
     }
 
     mVolume[LEFT] = 1.0f;
     mVolume[RIGHT] = 1.0f;
     mSendLevel = 0.0f;
-    mFrameCount = frameCount;
+    // mFrameCount is initialized in createTrack_l
     mReqFrameCount = frameCount;
     mNotificationFramesReq = notificationFrames;
     mNotificationFramesAct = 0;
     mSessionId = sessionId;
-    if (uid == -1 || (IPCThreadState::self()->getCallingPid() != getpid())) {
+    int callingpid = IPCThreadState::self()->getCallingPid();
+    int mypid = getpid();
+    if (uid == -1 || (callingpid != mypid)) {
         mClientUid = IPCThreadState::self()->getCallingUid();
     } else {
         mClientUid = uid;
     }
+    if (pid == -1 || (callingpid != mypid)) {
+        mClientPid = callingpid;
+    } else {
+        mClientPid = pid;
+    }
     mAuxEffectId = 0;
     mFlags = flags;
     mCbf = cbf;
@@ -330,14 +363,7 @@
     }
 
     // create the IAudioTrack
-    status_t status = createTrack_l(streamType,
-                                  sampleRate,
-                                  format,
-                                  frameCount,
-                                  flags,
-                                  sharedBuffer,
-                                  output,
-                                  0 /*epoch*/);
+    status = createTrack_l(0 /*epoch*/);
 
     if (status != NO_ERROR) {
         if (mAudioTrackThread != 0) {
@@ -345,17 +371,20 @@
             mAudioTrackThread->requestExitAndWait();
             mAudioTrackThread.clear();
         }
+        // Use of direct and offloaded output streams is ref counted by audio policy manager.
+#if 0   // FIXME This should no longer be needed
         //Use of direct and offloaded output streams is ref counted by audio policy manager.
         // As getOutput was called above and resulted in an output stream to be opened,
         // we need to release it.
-        AudioSystem::releaseOutput(output);
+        if (mOutput != 0) {
+            AudioSystem::releaseOutput(mOutput);
+            mOutput = 0;
+        }
+#endif
         return status;
     }
 
     mStatus = NO_ERROR;
-    mStreamType = streamType;
-    mFormat = format;
-    mSharedBuffer = sharedBuffer;
     mState = STATE_STOPPED;
     mUserData = user;
     mLoopPeriod = 0;
@@ -363,11 +392,10 @@
     mMarkerReached = false;
     mNewPosition = 0;
     mUpdatePeriod = 0;
-    AudioSystem::acquireAudioSessionId(mSessionId);
+    AudioSystem::acquireAudioSessionId(mSessionId, mClientPid);
     mSequence = 1;
     mObservedSequence = mSequence;
     mInUnderrun = false;
-    mOutput = output;
 
     return NO_ERROR;
 }
@@ -447,7 +475,7 @@
         return;
     }
 
-    if (isOffloaded()) {
+    if (isOffloaded_l()) {
         mState = STATE_STOPPING;
     } else {
         mState = STATE_STOPPED;
@@ -469,7 +497,7 @@
 
     sp<AudioTrackThread> t = mAudioTrackThread;
     if (t != 0) {
-        if (!isOffloaded()) {
+        if (!isOffloaded_l()) {
             t->pause();
         }
     } else {
@@ -507,7 +535,7 @@
     mRefreshRemaining = true;
 
     mState = STATE_FLUSHED;
-    if (isOffloaded()) {
+    if (isOffloaded_l()) {
         mProxy->interrupt();
     }
     mProxy->flush();
@@ -540,7 +568,7 @@
 
     mProxy->setVolumeLR((uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000));
 
-    if (isOffloaded()) {
+    if (isOffloaded_l()) {
         mAudioTrack->signal();
     }
     return NO_ERROR;
@@ -604,7 +632,7 @@
     // sample rate can be updated during playback by the offloaded decoder so we need to
     // query the HAL and update if needed.
 // FIXME use Proxy return channel to update the rate from server and avoid polling here
-    if (isOffloaded()) {
+    if (isOffloaded_l()) {
         if (mOutput != 0) {
             uint32_t sampleRate = 0;
             status_t status = AudioSystem::getSamplingRate(mOutput, mStreamType, &sampleRate);
@@ -688,6 +716,7 @@
     AutoMutex lock(mLock);
     mNewPosition = mProxy->getPosition() + updatePeriod;
     mUpdatePeriod = updatePeriod;
+
     return NO_ERROR;
 }
 
@@ -741,7 +770,7 @@
     }
 
     AutoMutex lock(mLock);
-    if (isOffloaded()) {
+    if (isOffloaded_l()) {
         uint32_t dspFrames = 0;
 
         if (mOutput != 0) {
@@ -757,7 +786,7 @@
     return NO_ERROR;
 }
 
-status_t AudioTrack::getBufferPosition(size_t *position)
+status_t AudioTrack::getBufferPosition(uint32_t *position)
 {
     if (mSharedBuffer == 0 || mIsTimed) {
         return INVALID_OPERATION;
@@ -790,23 +819,12 @@
     return NO_ERROR;
 }
 
-audio_io_handle_t AudioTrack::getOutput()
+audio_io_handle_t AudioTrack::getOutput() const
 {
     AutoMutex lock(mLock);
     return mOutput;
 }
 
-// must be called with mLock held
-audio_io_handle_t AudioTrack::getOutput_l()
-{
-    if (mOutput) {
-        return mOutput;
-    } else {
-        return AudioSystem::getOutput(mStreamType,
-                                      mSampleRate, mFormat, mChannelMask, mFlags);
-    }
-}
-
 status_t AudioTrack::attachAuxEffect(int effectId)
 {
     AutoMutex lock(mLock);
@@ -820,15 +838,7 @@
 // -------------------------------------------------------------------------
 
 // must be called with mLock held
-status_t AudioTrack::createTrack_l(
-        audio_stream_type_t streamType,
-        uint32_t sampleRate,
-        audio_format_t format,
-        size_t frameCount,
-        audio_output_flags_t flags,
-        const sp<IMemory>& sharedBuffer,
-        audio_io_handle_t output,
-        size_t epoch)
+status_t AudioTrack::createTrack_l(size_t epoch)
 {
     status_t status;
     const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
@@ -837,41 +847,54 @@
         return NO_INIT;
     }
 
+    audio_io_handle_t output = AudioSystem::getOutput(mStreamType, mSampleRate, mFormat,
+            mChannelMask, mFlags, mOffloadInfo);
+    if (output == 0) {
+        ALOGE("Could not get audio output for stream type %d, sample rate %u, format %#x, "
+              "channel mask %#x, flags %#x",
+              mStreamType, mSampleRate, mFormat, mChannelMask, mFlags);
+        return BAD_VALUE;
+    }
+    {
+    // Now that we have a reference to an I/O handle and have not yet handed it off to AudioFlinger,
+    // we must release it ourselves if anything goes wrong.
+
     // Not all of these values are needed under all conditions, but it is easier to get them all
 
     uint32_t afLatency;
-    status = AudioSystem::getLatency(output, streamType, &afLatency);
+    status = AudioSystem::getLatency(output, mStreamType, &afLatency);
     if (status != NO_ERROR) {
         ALOGE("getLatency(%d) failed status %d", output, status);
-        return NO_INIT;
+        goto release;
     }
 
     size_t afFrameCount;
-    status = AudioSystem::getFrameCount(output, streamType, &afFrameCount);
+    status = AudioSystem::getFrameCount(output, mStreamType, &afFrameCount);
     if (status != NO_ERROR) {
-        ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, streamType, status);
-        return NO_INIT;
+        ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, mStreamType, status);
+        goto release;
     }
 
     uint32_t afSampleRate;
-    status = AudioSystem::getSamplingRate(output, streamType, &afSampleRate);
+    status = AudioSystem::getSamplingRate(output, mStreamType, &afSampleRate);
     if (status != NO_ERROR) {
-        ALOGE("getSamplingRate(output=%d, streamType=%d) status %d", output, streamType, status);
-        return NO_INIT;
+        ALOGE("getSamplingRate(output=%d, streamType=%d) status %d", output, mStreamType, status);
+        goto release;
     }
 
     // Client decides whether the track is TIMED (see below), but can only express a preference
     // for FAST.  Server will perform additional tests.
-    if ((flags & AUDIO_OUTPUT_FLAG_FAST) && !(
+    if ((mFlags & AUDIO_OUTPUT_FLAG_FAST) && !((
             // either of these use cases:
             // use case 1: shared buffer
-            (sharedBuffer != 0) ||
+            (mSharedBuffer != 0) ||
             // use case 2: callback handler
-            (mCbf != NULL))) {
+            (mCbf != NULL)) &&
+            // matching sample rate
+            (mSampleRate == afSampleRate))) {
         ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client");
         // once denied, do not request again if IAudioTrack is re-created
-        flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST);
-        mFlags = flags;
+        mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST);
     }
     ALOGV("createTrack_l() output %d afLatency %d", output, afLatency);
 
@@ -882,43 +905,45 @@
     //  n = 3   normal track, with sample rate conversion
     //          (pessimistic; some non-1:1 conversion ratios don't actually need triple-buffering)
     //  n > 3   very high latency or very small notification interval; nBuffering is ignored
-    const uint32_t nBuffering = (sampleRate == afSampleRate) ? 2 : 3;
+    const uint32_t nBuffering = (mSampleRate == afSampleRate) ? 2 : 3;
 
     mNotificationFramesAct = mNotificationFramesReq;
 
-    if (!audio_is_linear_pcm(format)) {
+    size_t frameCount = mReqFrameCount;
+    if (!audio_is_linear_pcm(mFormat)) {
 
-        if (sharedBuffer != 0) {
+        if (mSharedBuffer != 0) {
             // Same comment as below about ignoring frameCount parameter for set()
-            frameCount = sharedBuffer->size();
+            frameCount = mSharedBuffer->size();
         } else if (frameCount == 0) {
             frameCount = afFrameCount;
         }
         if (mNotificationFramesAct != frameCount) {
             mNotificationFramesAct = frameCount;
         }
-    } else if (sharedBuffer != 0) {
+    } else if (mSharedBuffer != 0) {
 
         // Ensure that buffer alignment matches channel count
         // 8-bit data in shared memory is not currently supported by AudioFlinger
-        size_t alignment = /* format == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2;
+        size_t alignment = /* mFormat == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2;
         if (mChannelCount > 1) {
             // More than 2 channels does not require stronger alignment than stereo
             alignment <<= 1;
         }
-        if (((size_t)sharedBuffer->pointer() & (alignment - 1)) != 0) {
+        if (((uintptr_t)mSharedBuffer->pointer() & (alignment - 1)) != 0) {
             ALOGE("Invalid buffer alignment: address %p, channel count %u",
-                    sharedBuffer->pointer(), mChannelCount);
-            return BAD_VALUE;
+                    mSharedBuffer->pointer(), mChannelCount);
+            status = BAD_VALUE;
+            goto release;
         }
 
         // When initializing a shared buffer AudioTrack via constructors,
         // there's no frameCount parameter.
         // But when initializing a shared buffer AudioTrack via set(),
         // there _is_ a frameCount parameter.  We silently ignore it.
-        frameCount = sharedBuffer->size()/mChannelCount/sizeof(int16_t);
+        frameCount = mSharedBuffer->size()/mChannelCount/sizeof(int16_t);
 
-    } else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) {
+    } else if (!(mFlags & AUDIO_OUTPUT_FLAG_FAST)) {
 
         // FIXME move these calculations and associated checks to server
 
@@ -930,10 +955,10 @@
             minBufCount = nBuffering;
         }
 
-        size_t minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate;
+        size_t minFrameCount = (afFrameCount*mSampleRate*minBufCount)/afSampleRate;
         ALOGV("minFrameCount: %u, afFrameCount=%d, minBufCount=%d, sampleRate=%u, afSampleRate=%u"
                 ", afLatency=%d",
-                minFrameCount, afFrameCount, minBufCount, sampleRate, afSampleRate, afLatency);
+                minFrameCount, afFrameCount, minBufCount, mSampleRate, afSampleRate, afLatency);
 
         if (frameCount == 0) {
             frameCount = minFrameCount;
@@ -958,26 +983,28 @@
     }
 
     pid_t tid = -1;
-    if (flags & AUDIO_OUTPUT_FLAG_FAST) {
+    if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
         trackFlags |= IAudioFlinger::TRACK_FAST;
         if (mAudioTrackThread != 0) {
             tid = mAudioTrackThread->getTid();
         }
     }
 
-    if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
+    if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
         trackFlags |= IAudioFlinger::TRACK_OFFLOAD;
     }
 
-    sp<IAudioTrack> track = audioFlinger->createTrack(streamType,
-                                                      sampleRate,
+    size_t temp = frameCount;   // temp may be replaced by a revised value of frameCount,
+                                // but we will still need the original value also
+    sp<IAudioTrack> track = audioFlinger->createTrack(mStreamType,
+                                                      mSampleRate,
                                                       // AudioFlinger only sees 16-bit PCM
-                                                      format == AUDIO_FORMAT_PCM_8_BIT ?
-                                                              AUDIO_FORMAT_PCM_16_BIT : format,
+                                                      mFormat == AUDIO_FORMAT_PCM_8_BIT ?
+                                                              AUDIO_FORMAT_PCM_16_BIT : mFormat,
                                                       mChannelMask,
-                                                      frameCount,
+                                                      &temp,
                                                       &trackFlags,
-                                                      sharedBuffer,
+                                                      mSharedBuffer,
                                                       output,
                                                       tid,
                                                       &mSessionId,
@@ -987,13 +1014,21 @@
 
     if (track == 0) {
         ALOGE("AudioFlinger could not create track, status: %d", status);
-        return status;
+        goto release;
     }
+    // AudioFlinger now owns the reference to the I/O handle,
+    // so we are no longer responsible for releasing it.
+
     sp<IMemory> iMem = track->getCblk();
     if (iMem == 0) {
         ALOGE("Could not get control block");
         return NO_INIT;
     }
+    void *iMemPointer = iMem->pointer();
+    if (iMemPointer == NULL) {
+        ALOGE("Could not get control block pointer");
+        return NO_INIT;
+    }
     // invariant that mAudioTrack != 0 is true only after set() returns successfully
     if (mAudioTrack != 0) {
         mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this);
@@ -1001,9 +1036,9 @@
     }
     mAudioTrack = track;
     mCblkMemory = iMem;
-    audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMem->pointer());
+    audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer);
     mCblk = cblk;
-    size_t temp = cblk->frameCount_;
+    // note that temp is the (possibly revised) value of frameCount
     if (temp < frameCount || (frameCount == 0 && temp == 0)) {
         // In current design, AudioTrack client checks and ensures frame count validity before
         // passing it to AudioFlinger so AudioFlinger should not return a different value except
@@ -1012,11 +1047,11 @@
     }
     frameCount = temp;
     mAwaitBoost = false;
-    if (flags & AUDIO_OUTPUT_FLAG_FAST) {
+    if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
         if (trackFlags & IAudioFlinger::TRACK_FAST) {
             ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", frameCount);
             mAwaitBoost = true;
-            if (sharedBuffer == 0) {
+            if (mSharedBuffer == 0) {
                 // Theoretically double-buffering is not required for fast tracks,
                 // due to tighter scheduling.  But in practice, to accommodate kernels with
                 // scheduling jitter, and apps with computation jitter, we use double-buffering.
@@ -1027,26 +1062,27 @@
         } else {
             ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", frameCount);
             // once denied, do not request again if IAudioTrack is re-created
-            flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST);
-            mFlags = flags;
-            if (sharedBuffer == 0) {
+            mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST);
+            if (mSharedBuffer == 0) {
                 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) {
                     mNotificationFramesAct = frameCount/nBuffering;
                 }
             }
         }
     }
-    if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
+    if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
         if (trackFlags & IAudioFlinger::TRACK_OFFLOAD) {
             ALOGV("AUDIO_OUTPUT_FLAG_OFFLOAD successful");
         } else {
             ALOGW("AUDIO_OUTPUT_FLAG_OFFLOAD denied by server");
-            flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
-            mFlags = flags;
-            return NO_INIT;
+            mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
+            // FIXME This is a warning, not an error, so don't return error status
+            //return NO_INIT;
         }
     }
 
+    // We retain a copy of the I/O handle, but don't own the reference
+    mOutput = output;
     mRefreshRemaining = true;
 
     // Starting address of buffers in shared memory.  If there is a shared buffer, buffers
@@ -1054,15 +1090,15 @@
     // immediately after the control block.  This address is for the mapping within client
     // address space.  AudioFlinger::TrackBase::mBuffer is for the server address space.
     void* buffers;
-    if (sharedBuffer == 0) {
+    if (mSharedBuffer == 0) {
         buffers = (char*)cblk + sizeof(audio_track_cblk_t);
     } else {
-        buffers = sharedBuffer->pointer();
+        buffers = mSharedBuffer->pointer();
     }
 
     mAudioTrack->attachAuxEffect(mAuxEffectId);
     // FIXME don't believe this lie
-    mLatency = afLatency + (1000*frameCount) / sampleRate;
+    mLatency = afLatency + (1000*frameCount) / mSampleRate;
     mFrameCount = frameCount;
     // If IAudioTrack is re-created, don't let the requested frameCount
     // decrease.  This can confuse clients that cache frameCount().
@@ -1071,7 +1107,7 @@
     }
 
     // update proxy
-    if (sharedBuffer == 0) {
+    if (mSharedBuffer == 0) {
         mStaticProxy.clear();
         mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF);
     } else {
@@ -1089,6 +1125,14 @@
     mAudioTrack->asBinder()->linkToDeath(mDeathNotifier, this);
 
     return NO_ERROR;
+    }
+
+release:
+    AudioSystem::releaseOutput(output);
+    if (status == NO_ERROR) {
+        status = NO_INIT;
+    }
+    return status;
 }
 
 status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
@@ -1104,13 +1148,13 @@
     }
 
     const struct timespec *requested;
+    struct timespec timeout;
     if (waitCount == -1) {
         requested = &ClientProxy::kForever;
     } else if (waitCount == 0) {
         requested = &ClientProxy::kNonBlocking;
     } else if (waitCount > 0) {
         long long ms = WAIT_PERIOD_MS * (long long) waitCount;
-        struct timespec timeout;
         timeout.tv_sec = ms / 1000;
         timeout.tv_nsec = (int) (ms % 1000) * 1000000;
         requested = &timeout;
@@ -1235,7 +1279,7 @@
     if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) {
         // Sanity-check: user is most-likely passing an error code, and it would
         // make the return value ambiguous (actualSize vs error).
-        ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)", buffer, userSize, userSize);
+        ALOGE("AudioTrack::write(buffer=%p, size=%zu (%zd)", buffer, userSize, userSize);
         return BAD_VALUE;
     }
 
@@ -1379,7 +1423,7 @@
         // for offloaded tracks restoreTrack_l() will just update the sequence and clear
         // AudioSystem cache. We should not exit here but after calling the callback so
         // that the upper layers can recreate the track
-        if (!isOffloaded() || (mSequence == mObservedSequence)) {
+        if (!isOffloaded_l() || (mSequence == mObservedSequence)) {
             status_t status = restoreTrack_l("processAudioBuffer");
             mLock.unlock();
             // Run again immediately, but with a new IAudioTrack
@@ -1600,7 +1644,6 @@
         size_t reqSize = audioBuffer.size;
         mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer);
         size_t writtenSize = audioBuffer.size;
-        size_t writtenFrames = writtenSize / mFrameSize;
 
         // Sanity check on returned size
         if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) {
@@ -1666,22 +1709,19 @@
 status_t AudioTrack::restoreTrack_l(const char *from)
 {
     ALOGW("dead IAudioTrack, %s, creating a new one from %s()",
-          isOffloaded() ? "Offloaded" : "PCM", from);
+          isOffloaded_l() ? "Offloaded" : "PCM", from);
     ++mSequence;
     status_t result;
 
     // refresh the audio configuration cache in this process to make sure we get new
-    // output parameters in getOutput_l() and createTrack_l()
+    // output parameters in createTrack_l()
     AudioSystem::clearAudioConfigCache();
 
-    if (isOffloaded()) {
+    if (isOffloaded_l()) {
+        // FIXME re-creation of offloaded tracks is not yet implemented
         return DEAD_OBJECT;
     }
 
-    // force new output query from audio policy manager;
-    mOutput = 0;
-    audio_io_handle_t output = getOutput_l();
-
     // if the new IAudioTrack is created, createTrack_l() will modify the
     // following member variables: mAudioTrack, mCblkMemory and mCblk.
     // It will also delete the strong references on previous IAudioTrack and IMemory
@@ -1689,14 +1729,7 @@
     // take the frames that will be lost by track recreation into account in saved position
     size_t position = mProxy->getPosition() + mProxy->getFramesFilled();
     size_t bufferPosition = mStaticProxy != NULL ? mStaticProxy->getBufferPosition() : 0;
-    result = createTrack_l(mStreamType,
-                           mSampleRate,
-                           mFormat,
-                           mReqFrameCount,  // so that frame count never goes down
-                           mFlags,
-                           mSharedBuffer,
-                           output,
-                           position /*epoch*/);
+    result = createTrack_l(position /*epoch*/);
 
     if (result == NO_ERROR) {
         // continue playback from last known position, but
@@ -1724,10 +1757,16 @@
         }
     }
     if (result != NO_ERROR) {
+        // Use of direct and offloaded output streams is ref counted by audio policy manager.
+#if 0   // FIXME This should no longer be needed
         //Use of direct and offloaded output streams is ref counted by audio policy manager.
         // As getOutput was called above and resulted in an output stream to be opened,
         // we need to release it.
-        AudioSystem::releaseOutput(output);
+        if (mOutput != 0) {
+            AudioSystem::releaseOutput(mOutput);
+            mOutput = 0;
+        }
+#endif
         ALOGW("restoreTrack_l() failed status %d", result);
         mState = STATE_STOPPED;
     }
@@ -1760,13 +1799,20 @@
 
 String8 AudioTrack::getParameters(const String8& keys)
 {
-    if (mOutput) {
-        return AudioSystem::getParameters(mOutput, keys);
+    audio_io_handle_t output = getOutput();
+    if (output != 0) {
+        return AudioSystem::getParameters(output, keys);
     } else {
         return String8::empty();
     }
 }
 
+bool AudioTrack::isOffloaded() const
+{
+    AutoMutex lock(mLock);
+    return isOffloaded_l();
+}
+
 status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const
 {
 
@@ -1778,7 +1824,7 @@
     snprintf(buffer, 255, "  stream type(%d), left - right volume(%f, %f)\n", mStreamType,
             mVolume[0], mVolume[1]);
     result.append(buffer);
-    snprintf(buffer, 255, "  format(%d), channel count(%d), frame count(%d)\n", mFormat,
+    snprintf(buffer, 255, "  format(%d), channel count(%d), frame count(%zu)\n", mFormat,
             mChannelCount, mFrameCount);
     result.append(buffer);
     snprintf(buffer, 255, "  sample rate(%u), status(%d)\n", mSampleRate, mStatus);
diff --git a/media/libmedia/AudioTrackShared.cpp b/media/libmedia/AudioTrackShared.cpp
index 7a1e207..fdd1a12 100644
--- a/media/libmedia/AudioTrackShared.cpp
+++ b/media/libmedia/AudioTrackShared.cpp
@@ -26,7 +26,7 @@
 namespace android {
 
 audio_track_cblk_t::audio_track_cblk_t()
-    : mServer(0), frameCount_(0), mFutex(0), mMinimum(0),
+    : mServer(0), mFutex(0), mMinimum(0),
     mVolumeLR(0x10001000), mSampleRate(0), mSendLevel(0), mFlags(0)
 {
     memset(&u, 0, sizeof(u));
@@ -475,9 +475,14 @@
 
 void StaticAudioTrackClientProxy::setLoop(size_t loopStart, size_t loopEnd, int loopCount)
 {
+    // This can only happen on a 64-bit client
+    if (loopStart > UINT32_MAX || loopEnd > UINT32_MAX) {
+        // FIXME Should return an error status
+        return;
+    }
     StaticAudioTrackState newState;
-    newState.mLoopStart = loopStart;
-    newState.mLoopEnd = loopEnd;
+    newState.mLoopStart = (uint32_t) loopStart;
+    newState.mLoopEnd = (uint32_t) loopEnd;
     newState.mLoopCount = loopCount;
     mBufferPosition = loopStart;
     (void) mMutator.push(newState);
@@ -487,7 +492,7 @@
 {
     size_t bufferPosition;
     if (mMutator.ack()) {
-        bufferPosition = mCblk->u.mStatic.mBufferPosition;
+        bufferPosition = (size_t) mCblk->u.mStatic.mBufferPosition;
         if (bufferPosition > mFrameCount) {
             bufferPosition = mFrameCount;
         }
@@ -622,7 +627,7 @@
     if (half == 0) {
         half = 1;
     }
-    size_t minimum = cblk->mMinimum;
+    size_t minimum = (size_t) cblk->mMinimum;
     if (minimum == 0) {
         minimum = mIsOut ? half : 1;
     } else if (minimum > half) {
@@ -760,7 +765,8 @@
             mIsShutdown = true;
             return (ssize_t) NO_INIT;
         }
-        mCblk->u.mStatic.mBufferPosition = position;
+        // This may overflow, but client is not supposed to rely on it
+        mCblk->u.mStatic.mBufferPosition = (uint32_t) position;
     }
     return (ssize_t) position;
 }
@@ -836,7 +842,8 @@
     mPosition = newPosition;
 
     cblk->mServer += stepCount;
-    cblk->u.mStatic.mBufferPosition = newPosition;
+    // This may overflow, but client is not supposed to rely on it
+    cblk->u.mStatic.mBufferPosition = (uint32_t) newPosition;
     if (setFlags != 0) {
         (void) android_atomic_or(setFlags, &cblk->mFlags);
         // this would be a good place to wake a futex
diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp
index 9df10f0..7b15e68 100644
--- a/media/libmedia/IAudioFlinger.cpp
+++ b/media/libmedia/IAudioFlinger.cpp
@@ -89,7 +89,7 @@
                                 uint32_t sampleRate,
                                 audio_format_t format,
                                 audio_channel_mask_t channelMask,
-                                size_t frameCount,
+                                size_t *pFrameCount,
                                 track_flags_t *flags,
                                 const sp<IMemory>& sharedBuffer,
                                 audio_io_handle_t output,
@@ -106,6 +106,7 @@
         data.writeInt32(sampleRate);
         data.writeInt32(format);
         data.writeInt32(channelMask);
+        size_t frameCount = pFrameCount != NULL ? *pFrameCount : 0;
         data.writeInt32(frameCount);
         track_flags_t lFlags = flags != NULL ? *flags : (track_flags_t) TRACK_DEFAULT;
         data.writeInt32(lFlags);
@@ -117,7 +118,7 @@
         }
         data.writeInt32((int32_t) output);
         data.writeInt32((int32_t) tid);
-        int lSessionId = 0;
+        int lSessionId = AUDIO_SESSION_ALLOCATE;
         if (sessionId != NULL) {
             lSessionId = *sessionId;
         }
@@ -127,6 +128,10 @@
         if (lStatus != NO_ERROR) {
             ALOGE("createTrack error: %s", strerror(-lStatus));
         } else {
+            frameCount = reply.readInt32();
+            if (pFrameCount != NULL) {
+                *pFrameCount = frameCount;
+            }
             lFlags = reply.readInt32();
             if (flags != NULL) {
                 *flags = lFlags;
@@ -138,6 +143,17 @@
             name = reply.readString8();
             lStatus = reply.readInt32();
             track = interface_cast<IAudioTrack>(reply.readStrongBinder());
+            if (lStatus == NO_ERROR) {
+                if (track == 0) {
+                    ALOGE("createTrack should have returned an IAudioTrack");
+                    lStatus = UNKNOWN_ERROR;
+                }
+            } else {
+                if (track != 0) {
+                    ALOGE("createTrack returned an IAudioTrack but with status %d", lStatus);
+                    track.clear();
+                }
+            }
         }
         if (status != NULL) {
             *status = lStatus;
@@ -150,7 +166,7 @@
                                 uint32_t sampleRate,
                                 audio_format_t format,
                                 audio_channel_mask_t channelMask,
-                                size_t frameCount,
+                                size_t *pFrameCount,
                                 track_flags_t *flags,
                                 pid_t tid,
                                 int *sessionId,
@@ -163,11 +179,12 @@
         data.writeInt32(sampleRate);
         data.writeInt32(format);
         data.writeInt32(channelMask);
+        size_t frameCount = pFrameCount != NULL ? *pFrameCount : 0;
         data.writeInt32(frameCount);
         track_flags_t lFlags = flags != NULL ? *flags : (track_flags_t) TRACK_DEFAULT;
         data.writeInt32(lFlags);
         data.writeInt32((int32_t) tid);
-        int lSessionId = 0;
+        int lSessionId = AUDIO_SESSION_ALLOCATE;
         if (sessionId != NULL) {
             lSessionId = *sessionId;
         }
@@ -176,6 +193,10 @@
         if (lStatus != NO_ERROR) {
             ALOGE("openRecord error: %s", strerror(-lStatus));
         } else {
+            frameCount = reply.readInt32();
+            if (pFrameCount != NULL) {
+                *pFrameCount = frameCount;
+            }
             lFlags = reply.readInt32();
             if (flags != NULL) {
                 *flags = lFlags;
@@ -543,7 +564,7 @@
         return reply.readInt32();
     }
 
-    virtual status_t getRenderPosition(size_t *halFrames, size_t *dspFrames,
+    virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
             audio_io_handle_t output) const
     {
         Parcel data, reply;
@@ -564,13 +585,16 @@
         return status;
     }
 
-    virtual size_t getInputFramesLost(audio_io_handle_t ioHandle) const
+    virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
         data.writeInt32((int32_t) ioHandle);
-        remote()->transact(GET_INPUT_FRAMES_LOST, data, &reply);
-        return reply.readInt32();
+        status_t status = remote()->transact(GET_INPUT_FRAMES_LOST, data, &reply);
+        if (status != NO_ERROR) {
+            return 0;
+        }
+        return (uint32_t) reply.readInt32();
     }
 
     virtual int newAudioSessionId()
@@ -585,19 +609,21 @@
         return id;
     }
 
-    virtual void acquireAudioSessionId(int audioSession)
+    virtual void acquireAudioSessionId(int audioSession, int pid)
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
         data.writeInt32(audioSession);
+        data.writeInt32(pid);
         remote()->transact(ACQUIRE_AUDIO_SESSION_ID, data, &reply);
     }
 
-    virtual void releaseAudioSessionId(int audioSession)
+    virtual void releaseAudioSessionId(int audioSession, int pid)
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
         data.writeInt32(audioSession);
+        data.writeInt32(pid);
         remote()->transact(RELEASE_AUDIO_SESSION_ID, data, &reply);
     }
 
@@ -793,9 +819,11 @@
             } else {
                 track = createTrack(
                         (audio_stream_type_t) streamType, sampleRate, format,
-                        channelMask, frameCount, &flags, buffer, output, tid,
+                        channelMask, &frameCount, &flags, buffer, output, tid,
                         &sessionId, name, clientUid, &status);
+                LOG_ALWAYS_FATAL_IF((track != 0) != (status == NO_ERROR));
             }
+            reply->writeInt32(frameCount);
             reply->writeInt32(flags);
             reply->writeInt32(sessionId);
             reply->writeString8(name);
@@ -815,8 +843,9 @@
             int sessionId = data.readInt32();
             status_t status;
             sp<IAudioRecord> record = openRecord(input,
-                    sampleRate, format, channelMask, frameCount, &flags, tid, &sessionId, &status);
+                    sampleRate, format, channelMask, &frameCount, &flags, tid, &sessionId, &status);
             LOG_ALWAYS_FATAL_IF((record != 0) != (status == NO_ERROR));
+            reply->writeInt32(frameCount);
             reply->writeInt32(flags);
             reply->writeInt32(sessionId);
             reply->writeInt32(status);
@@ -1031,8 +1060,8 @@
         case GET_RENDER_POSITION: {
             CHECK_INTERFACE(IAudioFlinger, data, reply);
             audio_io_handle_t output = (audio_io_handle_t) data.readInt32();
-            size_t halFrames;
-            size_t dspFrames;
+            uint32_t halFrames;
+            uint32_t dspFrames;
             status_t status = getRenderPosition(&halFrames, &dspFrames, output);
             reply->writeInt32(status);
             if (status == NO_ERROR) {
@@ -1044,7 +1073,7 @@
         case GET_INPUT_FRAMES_LOST: {
             CHECK_INTERFACE(IAudioFlinger, data, reply);
             audio_io_handle_t ioHandle = (audio_io_handle_t) data.readInt32();
-            reply->writeInt32(getInputFramesLost(ioHandle));
+            reply->writeInt32((int32_t) getInputFramesLost(ioHandle));
             return NO_ERROR;
         } break;
         case NEW_AUDIO_SESSION_ID: {
@@ -1055,13 +1084,15 @@
         case ACQUIRE_AUDIO_SESSION_ID: {
             CHECK_INTERFACE(IAudioFlinger, data, reply);
             int audioSession = data.readInt32();
-            acquireAudioSessionId(audioSession);
+            int pid = data.readInt32();
+            acquireAudioSessionId(audioSession, pid);
             return NO_ERROR;
         } break;
         case RELEASE_AUDIO_SESSION_ID: {
             CHECK_INTERFACE(IAudioFlinger, data, reply);
             int audioSession = data.readInt32();
-            releaseAudioSessionId(audioSession);
+            int pid = data.readInt32();
+            releaseAudioSessionId(audioSession, pid);
             return NO_ERROR;
         } break;
         case QUERY_NUM_EFFECTS: {
diff --git a/media/libmedia/IMediaHTTPConnection.cpp b/media/libmedia/IMediaHTTPConnection.cpp
new file mode 100644
index 0000000..622d9cf
--- /dev/null
+++ b/media/libmedia/IMediaHTTPConnection.cpp
@@ -0,0 +1,158 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "IMediaHTTPConnection"
+#include <utils/Log.h>
+
+#include <media/IMediaHTTPConnection.h>
+
+#include <binder/IMemory.h>
+#include <binder/Parcel.h>
+#include <utils/String8.h>
+#include <media/stagefright/foundation/ADebug.h>
+
+namespace android {
+
+enum {
+    CONNECT = IBinder::FIRST_CALL_TRANSACTION,
+    DISCONNECT,
+    READ_AT,
+    GET_SIZE,
+    GET_MIME_TYPE,
+};
+
+struct BpMediaHTTPConnection : public BpInterface<IMediaHTTPConnection> {
+    BpMediaHTTPConnection(const sp<IBinder> &impl)
+        : BpInterface<IMediaHTTPConnection>(impl) {
+    }
+
+    virtual bool connect(
+            const char *uri, const KeyedVector<String8, String8> *headers) {
+        Parcel data, reply;
+        data.writeInterfaceToken(
+                IMediaHTTPConnection::getInterfaceDescriptor());
+
+        String16 tmp(uri);
+        data.writeString16(tmp);
+
+        tmp = String16("");
+        if (headers != NULL) {
+            for (size_t i = 0; i < headers->size(); ++i) {
+                String16 key(headers->keyAt(i).string());
+                String16 val(headers->valueAt(i).string());
+
+                tmp.append(key);
+                tmp.append(String16(": "));
+                tmp.append(val);
+                tmp.append(String16("\r\n"));
+            }
+        }
+        data.writeString16(tmp);
+
+        remote()->transact(CONNECT, data, &reply);
+
+        int32_t exceptionCode = reply.readExceptionCode();
+
+        if (exceptionCode) {
+            return UNKNOWN_ERROR;
+        }
+
+        sp<IBinder> binder = reply.readStrongBinder();
+        mMemory = interface_cast<IMemory>(binder);
+
+        return mMemory != NULL;
+    }
+
+    virtual void disconnect() {
+        Parcel data, reply;
+        data.writeInterfaceToken(
+                IMediaHTTPConnection::getInterfaceDescriptor());
+
+        remote()->transact(DISCONNECT, data, &reply);
+    }
+
+    virtual ssize_t readAt(off64_t offset, void *buffer, size_t size) {
+        Parcel data, reply;
+        data.writeInterfaceToken(
+                IMediaHTTPConnection::getInterfaceDescriptor());
+
+        data.writeInt64(offset);
+        data.writeInt32(size);
+
+        status_t err = remote()->transact(READ_AT, data, &reply);
+        CHECK_EQ(err, (status_t)OK);
+
+        int32_t exceptionCode = reply.readExceptionCode();
+
+        if (exceptionCode) {
+            return UNKNOWN_ERROR;
+        }
+
+        int32_t len = reply.readInt32();
+
+        if (len > 0) {
+            memcpy(buffer, mMemory->pointer(), len);
+        }
+
+        return len;
+    }
+
+    virtual off64_t getSize() {
+        Parcel data, reply;
+        data.writeInterfaceToken(
+                IMediaHTTPConnection::getInterfaceDescriptor());
+
+        remote()->transact(GET_SIZE, data, &reply);
+
+        int32_t exceptionCode = reply.readExceptionCode();
+
+        if (exceptionCode) {
+            return UNKNOWN_ERROR;
+        }
+
+        return reply.readInt64();
+    }
+
+    virtual status_t getMIMEType(String8 *mimeType) {
+        *mimeType = String8("");
+
+        Parcel data, reply;
+        data.writeInterfaceToken(
+                IMediaHTTPConnection::getInterfaceDescriptor());
+
+        remote()->transact(GET_MIME_TYPE, data, &reply);
+
+        int32_t exceptionCode = reply.readExceptionCode();
+
+        if (exceptionCode) {
+            return UNKNOWN_ERROR;
+        }
+
+        *mimeType = String8(reply.readString16());
+
+        return OK;
+    }
+
+private:
+    sp<IMemory> mMemory;
+};
+
+IMPLEMENT_META_INTERFACE(
+        MediaHTTPConnection, "android.media.IMediaHTTPConnection");
+
+}  // namespace android
+
diff --git a/media/libmedia/IMediaHTTPService.cpp b/media/libmedia/IMediaHTTPService.cpp
new file mode 100644
index 0000000..1260582
--- /dev/null
+++ b/media/libmedia/IMediaHTTPService.cpp
@@ -0,0 +1,58 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "IMediaHTTPService"
+#include <utils/Log.h>
+
+#include <media/IMediaHTTPService.h>
+
+#include <binder/Parcel.h>
+#include <media/IMediaHTTPConnection.h>
+
+namespace android {
+
+enum {
+    MAKE_HTTP = IBinder::FIRST_CALL_TRANSACTION,
+};
+
+struct BpMediaHTTPService : public BpInterface<IMediaHTTPService> {
+    BpMediaHTTPService(const sp<IBinder> &impl)
+        : BpInterface<IMediaHTTPService>(impl) {
+    }
+
+    virtual sp<IMediaHTTPConnection> makeHTTPConnection() {
+        Parcel data, reply;
+        data.writeInterfaceToken(
+                IMediaHTTPService::getInterfaceDescriptor());
+
+        remote()->transact(MAKE_HTTP, data, &reply);
+
+        status_t err = reply.readInt32();
+
+        if (err != OK) {
+            return NULL;
+        }
+
+        return interface_cast<IMediaHTTPConnection>(reply.readStrongBinder());
+    }
+};
+
+IMPLEMENT_META_INTERFACE(
+        MediaHTTPService, "android.media.IMediaHTTPService");
+
+}  // namespace android
+
diff --git a/media/libmedia/IMediaMetadataRetriever.cpp b/media/libmedia/IMediaMetadataRetriever.cpp
index bb066a0..c7d9d51 100644
--- a/media/libmedia/IMediaMetadataRetriever.cpp
+++ b/media/libmedia/IMediaMetadataRetriever.cpp
@@ -18,6 +18,7 @@
 #include <stdint.h>
 #include <sys/types.h>
 #include <binder/Parcel.h>
+#include <media/IMediaHTTPService.h>
 #include <media/IMediaMetadataRetriever.h>
 #include <utils/String8.h>
 #include <utils/KeyedVector.h>
@@ -84,10 +85,16 @@
     }
 
     status_t setDataSource(
-            const char *srcUrl, const KeyedVector<String8, String8> *headers)
+            const sp<IMediaHTTPService> &httpService,
+            const char *srcUrl,
+            const KeyedVector<String8, String8> *headers)
     {
         Parcel data, reply;
         data.writeInterfaceToken(IMediaMetadataRetriever::getInterfaceDescriptor());
+        data.writeInt32(httpService != NULL);
+        if (httpService != NULL) {
+            data.writeStrongBinder(httpService->asBinder());
+        }
         data.writeCString(srcUrl);
 
         if (headers == NULL) {
@@ -195,6 +202,13 @@
         } break;
         case SET_DATA_SOURCE_URL: {
             CHECK_INTERFACE(IMediaMetadataRetriever, data, reply);
+
+            sp<IMediaHTTPService> httpService;
+            if (data.readInt32()) {
+                httpService =
+                    interface_cast<IMediaHTTPService>(data.readStrongBinder());
+            }
+
             const char* srcUrl = data.readCString();
 
             KeyedVector<String8, String8> headers;
@@ -206,7 +220,8 @@
             }
 
             reply->writeInt32(
-                    setDataSource(srcUrl, numHeaders > 0 ? &headers : NULL));
+                    setDataSource(
+                        httpService, srcUrl, numHeaders > 0 ? &headers : NULL));
 
             return NO_ERROR;
         } break;
diff --git a/media/libmedia/IMediaPlayer.cpp b/media/libmedia/IMediaPlayer.cpp
index e79bcd2..d778d05 100644
--- a/media/libmedia/IMediaPlayer.cpp
+++ b/media/libmedia/IMediaPlayer.cpp
@@ -21,6 +21,7 @@
 
 #include <binder/Parcel.h>
 
+#include <media/IMediaHTTPService.h>
 #include <media/IMediaPlayer.h>
 #include <media/IStreamSource.h>
 
@@ -75,11 +76,17 @@
         remote()->transact(DISCONNECT, data, &reply);
     }
 
-    status_t setDataSource(const char* url,
+    status_t setDataSource(
+            const sp<IMediaHTTPService> &httpService,
+            const char* url,
             const KeyedVector<String8, String8>* headers)
     {
         Parcel data, reply;
         data.writeInterfaceToken(IMediaPlayer::getInterfaceDescriptor());
+        data.writeInt32(httpService != NULL);
+        if (httpService != NULL) {
+            data.writeStrongBinder(httpService->asBinder());
+        }
         data.writeCString(url);
         if (headers == NULL) {
             data.writeInt32(0);
@@ -355,6 +362,13 @@
         } break;
         case SET_DATA_SOURCE_URL: {
             CHECK_INTERFACE(IMediaPlayer, data, reply);
+
+            sp<IMediaHTTPService> httpService;
+            if (data.readInt32()) {
+                httpService =
+                    interface_cast<IMediaHTTPService>(data.readStrongBinder());
+            }
+
             const char* url = data.readCString();
             KeyedVector<String8, String8> headers;
             int32_t numHeaders = data.readInt32();
@@ -363,7 +377,8 @@
                 String8 value = data.readString8();
                 headers.add(key, value);
             }
-            reply->writeInt32(setDataSource(url, numHeaders > 0 ? &headers : NULL));
+            reply->writeInt32(setDataSource(
+                        httpService, url, numHeaders > 0 ? &headers : NULL));
             return NO_ERROR;
         } break;
         case SET_DATA_SOURCE_FD: {
diff --git a/media/libmedia/IMediaPlayerService.cpp b/media/libmedia/IMediaPlayerService.cpp
index 3c22b4c..d116b14 100644
--- a/media/libmedia/IMediaPlayerService.cpp
+++ b/media/libmedia/IMediaPlayerService.cpp
@@ -23,6 +23,7 @@
 #include <media/ICrypto.h>
 #include <media/IDrm.h>
 #include <media/IHDCP.h>
+#include <media/IMediaHTTPService.h>
 #include <media/IMediaPlayerService.h>
 #include <media/IMediaRecorder.h>
 #include <media/IOMX.h>
@@ -48,7 +49,6 @@
     ADD_BATTERY_DATA,
     PULL_BATTERY_DATA,
     LISTEN_FOR_REMOTE_DISPLAY,
-    UPDATE_PROXY_CONFIG,
 };
 
 class BpMediaPlayerService: public BpInterface<IMediaPlayerService>
@@ -86,12 +86,21 @@
         return interface_cast<IMediaRecorder>(reply.readStrongBinder());
     }
 
-    virtual status_t decode(const char* url, uint32_t *pSampleRate, int* pNumChannels,
-                               audio_format_t* pFormat,
-                               const sp<IMemoryHeap>& heap, size_t *pSize)
+    virtual status_t decode(
+            const sp<IMediaHTTPService> &httpService,
+            const char* url,
+            uint32_t *pSampleRate,
+            int* pNumChannels,
+            audio_format_t* pFormat,
+            const sp<IMemoryHeap>& heap,
+            size_t *pSize)
     {
         Parcel data, reply;
         data.writeInterfaceToken(IMediaPlayerService::getInterfaceDescriptor());
+        data.writeInt32(httpService != NULL);
+        if (httpService != NULL) {
+            data.writeStrongBinder(httpService->asBinder());
+        }
         data.writeCString(url);
         data.writeStrongBinder(heap->asBinder());
         status_t status = remote()->transact(DECODE_URL, data, &reply);
@@ -182,25 +191,6 @@
         remote()->transact(LISTEN_FOR_REMOTE_DISPLAY, data, &reply);
         return interface_cast<IRemoteDisplay>(reply.readStrongBinder());
     }
-
-    virtual status_t updateProxyConfig(
-            const char *host, int32_t port, const char *exclusionList) {
-        Parcel data, reply;
-
-        data.writeInterfaceToken(IMediaPlayerService::getInterfaceDescriptor());
-        if (host == NULL) {
-            data.writeInt32(0);
-        } else {
-            data.writeInt32(1);
-            data.writeCString(host);
-            data.writeInt32(port);
-            data.writeCString(exclusionList);
-        }
-
-        remote()->transact(UPDATE_PROXY_CONFIG, data, &reply);
-
-        return reply.readInt32();
-    }
 };
 
 IMPLEMENT_META_INTERFACE(MediaPlayerService, "android.media.IMediaPlayerService");
@@ -222,13 +212,25 @@
         } break;
         case DECODE_URL: {
             CHECK_INTERFACE(IMediaPlayerService, data, reply);
+            sp<IMediaHTTPService> httpService;
+            if (data.readInt32()) {
+                httpService =
+                    interface_cast<IMediaHTTPService>(data.readStrongBinder());
+            }
             const char* url = data.readCString();
             sp<IMemoryHeap> heap = interface_cast<IMemoryHeap>(data.readStrongBinder());
             uint32_t sampleRate;
             int numChannels;
             audio_format_t format;
             size_t size;
-            status_t status = decode(url, &sampleRate, &numChannels, &format, heap, &size);
+            status_t status =
+                decode(httpService,
+                       url,
+                       &sampleRate,
+                       &numChannels,
+                       &format,
+                       heap,
+                       &size);
             reply->writeInt32(status);
             if (status == NO_ERROR) {
                 reply->writeInt32(sampleRate);
@@ -316,24 +318,6 @@
             reply->writeStrongBinder(display->asBinder());
             return NO_ERROR;
         } break;
-        case UPDATE_PROXY_CONFIG:
-        {
-            CHECK_INTERFACE(IMediaPlayerService, data, reply);
-
-            const char *host = NULL;
-            int32_t port = 0;
-            const char *exclusionList = NULL;
-
-            if (data.readInt32()) {
-                host = data.readCString();
-                port = data.readInt32();
-                exclusionList = data.readCString();
-            }
-
-            reply->writeInt32(updateProxyConfig(host, port, exclusionList));
-
-            return OK;
-        }
         default:
             return BBinder::onTransact(code, data, reply, flags);
     }
diff --git a/media/libmedia/MediaProfiles.cpp b/media/libmedia/MediaProfiles.cpp
index 8319cd7..1074da9 100644
--- a/media/libmedia/MediaProfiles.cpp
+++ b/media/libmedia/MediaProfiles.cpp
@@ -81,8 +81,14 @@
     {"timelapseqvga", CAMCORDER_QUALITY_TIME_LAPSE_QVGA},
 };
 
+#if LOG_NDEBUG
+#define UNUSED __unused
+#else
+#define UNUSED
+#endif
+
 /*static*/ void
-MediaProfiles::logVideoCodec(const MediaProfiles::VideoCodec& codec)
+MediaProfiles::logVideoCodec(const MediaProfiles::VideoCodec& codec UNUSED)
 {
     ALOGV("video codec:");
     ALOGV("codec = %d", codec.mCodec);
@@ -93,7 +99,7 @@
 }
 
 /*static*/ void
-MediaProfiles::logAudioCodec(const MediaProfiles::AudioCodec& codec)
+MediaProfiles::logAudioCodec(const MediaProfiles::AudioCodec& codec UNUSED)
 {
     ALOGV("audio codec:");
     ALOGV("codec = %d", codec.mCodec);
@@ -103,7 +109,7 @@
 }
 
 /*static*/ void
-MediaProfiles::logVideoEncoderCap(const MediaProfiles::VideoEncoderCap& cap)
+MediaProfiles::logVideoEncoderCap(const MediaProfiles::VideoEncoderCap& cap UNUSED)
 {
     ALOGV("video encoder cap:");
     ALOGV("codec = %d", cap.mCodec);
@@ -114,7 +120,7 @@
 }
 
 /*static*/ void
-MediaProfiles::logAudioEncoderCap(const MediaProfiles::AudioEncoderCap& cap)
+MediaProfiles::logAudioEncoderCap(const MediaProfiles::AudioEncoderCap& cap UNUSED)
 {
     ALOGV("audio encoder cap:");
     ALOGV("codec = %d", cap.mCodec);
@@ -124,21 +130,21 @@
 }
 
 /*static*/ void
-MediaProfiles::logVideoDecoderCap(const MediaProfiles::VideoDecoderCap& cap)
+MediaProfiles::logVideoDecoderCap(const MediaProfiles::VideoDecoderCap& cap UNUSED)
 {
     ALOGV("video decoder cap:");
     ALOGV("codec = %d", cap.mCodec);
 }
 
 /*static*/ void
-MediaProfiles::logAudioDecoderCap(const MediaProfiles::AudioDecoderCap& cap)
+MediaProfiles::logAudioDecoderCap(const MediaProfiles::AudioDecoderCap& cap UNUSED)
 {
     ALOGV("audio codec cap:");
     ALOGV("codec = %d", cap.mCodec);
 }
 
 /*static*/ void
-MediaProfiles::logVideoEditorCap(const MediaProfiles::VideoEditorCap& cap)
+MediaProfiles::logVideoEditorCap(const MediaProfiles::VideoEditorCap& cap UNUSED)
 {
     ALOGV("videoeditor cap:");
     ALOGV("mMaxInputFrameWidth = %d", cap.mMaxInputFrameWidth);
diff --git a/media/libmedia/MemoryLeakTrackUtil.cpp b/media/libmedia/MemoryLeakTrackUtil.cpp
index 6a108ae..f004ca4 100644
--- a/media/libmedia/MemoryLeakTrackUtil.cpp
+++ b/media/libmedia/MemoryLeakTrackUtil.cpp
@@ -49,7 +49,7 @@
     }
 
     void append(const char *s) {
-        strcat(mPtr, s);
+        strncat(mPtr, s, MAX_SIZE - size() - 1);
     }
 
     const char *string() const {
@@ -60,6 +60,10 @@
         return strlen(mPtr);
     }
 
+    void clear() {
+        *mPtr = '\0';
+    }
+
 private:
     char *mPtr;
 
@@ -139,6 +143,9 @@
             }
         } while (moved);
 
+        write(fd, result.string(), result.size());
+        result.clear();
+
         for (size_t i = 0; i < count; i++) {
             AllocEntry *e = &entries[i];
 
@@ -152,13 +159,14 @@
                 result.append(buffer);
             }
             result.append("\n");
+
+            write(fd, result.string(), result.size());
+            result.clear();
         }
 
         delete[] entries;
         free_malloc_leak_info(info);
     }
-
-    write(fd, result.string(), result.size());
 }
 
 #else
diff --git a/media/libmedia/SoundPool.cpp b/media/libmedia/SoundPool.cpp
index 98acd1f..4885b4f 100644
--- a/media/libmedia/SoundPool.cpp
+++ b/media/libmedia/SoundPool.cpp
@@ -21,6 +21,7 @@
 #define USE_SHARED_MEM_BUFFER
 
 #include <media/AudioTrack.h>
+#include <media/IMediaHTTPService.h>
 #include <media/mediaplayer.h>
 #include <media/SoundPool.h>
 #include "SoundPoolThread.h"
@@ -496,7 +497,14 @@
 
     ALOGV("Start decode");
     if (mUrl) {
-        status = MediaPlayer::decode(mUrl, &sampleRate, &numChannels, &format, mHeap, &mSize);
+        status = MediaPlayer::decode(
+                NULL /* httpService */,
+                mUrl,
+                &sampleRate,
+                &numChannels,
+                &format,
+                mHeap,
+                &mSize);
     } else {
         status = MediaPlayer::decode(mFd, mOffset, mLength, &sampleRate, &numChannels, &format,
                                      mHeap, &mSize);
@@ -729,7 +737,8 @@
                     count = b->size;
                 }
                 memcpy(q, p, count);
-//              ALOGV("fill: q=%p, p=%p, mPos=%u, b->size=%u, count=%d", q, p, mPos, b->size, count);
+//              ALOGV("fill: q=%p, p=%p, mPos=%u, b->size=%u, count=%d", q, p, mPos, b->size,
+//                      count);
             } else if (mPos < mAudioBufferSize) {
                 count = mAudioBufferSize - mPos;
                 if (count > b->size) {
diff --git a/media/libmedia/mediametadataretriever.cpp b/media/libmedia/mediametadataretriever.cpp
index bad2494..1d6bb6f 100644
--- a/media/libmedia/mediametadataretriever.cpp
+++ b/media/libmedia/mediametadataretriever.cpp
@@ -21,6 +21,7 @@
 #include <binder/IServiceManager.h>
 #include <binder/IPCThreadState.h>
 #include <media/mediametadataretriever.h>
+#include <media/IMediaHTTPService.h>
 #include <media/IMediaPlayerService.h>
 #include <utils/Log.h>
 #include <dlfcn.h>
@@ -93,7 +94,9 @@
 }
 
 status_t MediaMetadataRetriever::setDataSource(
-        const char *srcUrl, const KeyedVector<String8, String8> *headers)
+        const sp<IMediaHTTPService> &httpService,
+        const char *srcUrl,
+        const KeyedVector<String8, String8> *headers)
 {
     ALOGV("setDataSource");
     Mutex::Autolock _l(mLock);
@@ -106,7 +109,7 @@
         return UNKNOWN_ERROR;
     }
     ALOGV("data source (%s)", srcUrl);
-    return mRetriever->setDataSource(srcUrl, headers);
+    return mRetriever->setDataSource(httpService, srcUrl, headers);
 }
 
 status_t MediaMetadataRetriever::setDataSource(int fd, int64_t offset, int64_t length)
diff --git a/media/libmedia/mediaplayer.cpp b/media/libmedia/mediaplayer.cpp
index 7a6f31d..d94c7c5 100644
--- a/media/libmedia/mediaplayer.cpp
+++ b/media/libmedia/mediaplayer.cpp
@@ -58,7 +58,7 @@
     mVideoWidth = mVideoHeight = 0;
     mLockThreadId = 0;
     mAudioSessionId = AudioSystem::newAudioSessionId();
-    AudioSystem::acquireAudioSessionId(mAudioSessionId);
+    AudioSystem::acquireAudioSessionId(mAudioSessionId, -1);
     mSendLevel = 0;
     mRetransmitEndpointValid = false;
 }
@@ -66,7 +66,7 @@
 MediaPlayer::~MediaPlayer()
 {
     ALOGV("destructor");
-    AudioSystem::releaseAudioSessionId(mAudioSessionId);
+    AudioSystem::releaseAudioSessionId(mAudioSessionId, -1);
     disconnect();
     IPCThreadState::self()->flushCommands();
 }
@@ -136,6 +136,7 @@
 }
 
 status_t MediaPlayer::setDataSource(
+        const sp<IMediaHTTPService> &httpService,
         const char *url, const KeyedVector<String8, String8> *headers)
 {
     ALOGV("setDataSource(%s)", url);
@@ -145,7 +146,7 @@
         if (service != 0) {
             sp<IMediaPlayer> player(service->create(this, mAudioSessionId));
             if ((NO_ERROR != doSetRetransmitEndpoint(player)) ||
-                (NO_ERROR != player->setDataSource(url, headers))) {
+                (NO_ERROR != player->setDataSource(httpService, url, headers))) {
                 player.clear();
             }
             err = attachNewPlayer(player);
@@ -575,8 +576,8 @@
         return BAD_VALUE;
     }
     if (sessionId != mAudioSessionId) {
-        AudioSystem::acquireAudioSessionId(sessionId);
-        AudioSystem::releaseAudioSessionId(mAudioSessionId);
+        AudioSystem::acquireAudioSessionId(sessionId, -1);
+        AudioSystem::releaseAudioSessionId(mAudioSessionId, -1);
         mAudioSessionId = sessionId;
     }
     return NO_ERROR;
@@ -776,15 +777,20 @@
     }
 }
 
-/*static*/ status_t MediaPlayer::decode(const char* url, uint32_t *pSampleRate,
-                                           int* pNumChannels, audio_format_t* pFormat,
-                                           const sp<IMemoryHeap>& heap, size_t *pSize)
+/*static*/ status_t MediaPlayer::decode(
+        const sp<IMediaHTTPService> &httpService,
+        const char* url,
+        uint32_t *pSampleRate,
+        int* pNumChannels,
+        audio_format_t* pFormat,
+        const sp<IMemoryHeap>& heap,
+        size_t *pSize)
 {
     ALOGV("decode(%s)", url);
     status_t status;
     const sp<IMediaPlayerService>& service = getMediaPlayerService();
     if (service != 0) {
-        status = service->decode(url, pSampleRate, pNumChannels, pFormat, heap, pSize);
+        status = service->decode(httpService, url, pSampleRate, pNumChannels, pFormat, heap, pSize);
     } else {
         ALOGE("Unable to locate media service");
         status = DEAD_OBJECT;
@@ -832,15 +838,4 @@
     return mPlayer->setNextPlayer(next == NULL ? NULL : next->mPlayer);
 }
 
-status_t MediaPlayer::updateProxyConfig(
-        const char *host, int32_t port, const char *exclusionList) {
-    const sp<IMediaPlayerService>& service = getMediaPlayerService();
-
-    if (service != NULL) {
-        return service->updateProxyConfig(host, port, exclusionList);
-    }
-
-    return INVALID_OPERATION;
-}
-
 }; // namespace android
diff --git a/media/libmediaplayerservice/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp
index 9ac9105..142788d 100644
--- a/media/libmediaplayerservice/MediaPlayerService.cpp
+++ b/media/libmediaplayerservice/MediaPlayerService.cpp
@@ -44,6 +44,7 @@
 #include <utils/SystemClock.h>
 #include <utils/Vector.h>
 
+#include <media/IMediaHTTPService.h>
 #include <media/IRemoteDisplay.h>
 #include <media/IRemoteDisplayClient.h>
 #include <media/MediaPlayerInterface.h>
@@ -306,11 +307,6 @@
     return new RemoteDisplay(client, iface.string());
 }
 
-status_t MediaPlayerService::updateProxyConfig(
-        const char *host, int32_t port, const char *exclusionList) {
-    return HTTPBase::UpdateProxyConfig(host, port, exclusionList);
-}
-
 status_t MediaPlayerService::AudioCache::dump(int fd, const Vector<String16>& args) const
 {
     const size_t SIZE = 256;
@@ -319,7 +315,7 @@
 
     result.append(" AudioCache\n");
     if (mHeap != 0) {
-        snprintf(buffer, 255, "  heap base(%p), size(%d), flags(%d)\n",
+        snprintf(buffer, 255, "  heap base(%p), size(%zu), flags(%d)\n",
                 mHeap->getBase(), mHeap->getSize(), mHeap->getFlags());
         result.append(buffer);
     }
@@ -590,7 +586,8 @@
     }
 
     if (!p->hardwareOutput()) {
-        mAudioOutput = new AudioOutput(mAudioSessionId, IPCThreadState::self()->getCallingUid());
+        mAudioOutput = new AudioOutput(mAudioSessionId, IPCThreadState::self()->getCallingUid(),
+                mPid);
         static_cast<MediaPlayerInterface*>(p.get())->setAudioSink(mAudioOutput);
     }
 
@@ -622,7 +619,9 @@
 }
 
 status_t MediaPlayerService::Client::setDataSource(
-        const char *url, const KeyedVector<String8, String8> *headers)
+        const sp<IMediaHTTPService> &httpService,
+        const char *url,
+        const KeyedVector<String8, String8> *headers)
 {
     ALOGV("setDataSource(%s)", url);
     if (url == NULL)
@@ -657,7 +656,7 @@
             return NO_INIT;
         }
 
-        setDataSource_post(p, p->setDataSource(url, headers));
+        setDataSource_post(p, p->setDataSource(httpService, url, headers));
         return mStatus;
     }
 }
@@ -1176,9 +1175,14 @@
 }
 #endif
 
-status_t MediaPlayerService::decode(const char* url, uint32_t *pSampleRate, int* pNumChannels,
-                                       audio_format_t* pFormat,
-                                       const sp<IMemoryHeap>& heap, size_t *pSize)
+status_t MediaPlayerService::decode(
+        const sp<IMediaHTTPService> &httpService,
+        const char* url,
+        uint32_t *pSampleRate,
+        int* pNumChannels,
+        audio_format_t* pFormat,
+        const sp<IMemoryHeap>& heap,
+        size_t *pSize)
 {
     ALOGV("decode(%s)", url);
     sp<MediaPlayerBase> player;
@@ -1206,7 +1210,7 @@
     static_cast<MediaPlayerInterface*>(player.get())->setAudioSink(cache);
 
     // set data source
-    if (player->setDataSource(url) != NO_ERROR) goto Exit;
+    if (player->setDataSource(httpService, url) != NO_ERROR) goto Exit;
 
     ALOGV("prepare");
     player->prepareAsync();
@@ -1296,13 +1300,14 @@
 
 #undef LOG_TAG
 #define LOG_TAG "AudioSink"
-MediaPlayerService::AudioOutput::AudioOutput(int sessionId, int uid)
+MediaPlayerService::AudioOutput::AudioOutput(int sessionId, int uid, int pid)
     : mCallback(NULL),
       mCallbackCookie(NULL),
       mCallbackData(NULL),
       mBytesWritten(0),
       mSessionId(sessionId),
       mUid(uid),
+      mPid(pid),
       mFlags(AUDIO_OUTPUT_FLAG_NONE) {
     ALOGV("AudioOutput(%d)", sessionId);
     mStreamType = AUDIO_STREAM_MUSIC;
@@ -1551,7 +1556,8 @@
                     mSessionId,
                     AudioTrack::TRANSFER_CALLBACK,
                     offloadInfo,
-                    mUid);
+                    mUid,
+                    mPid);
         } else {
             t = new AudioTrack(
                     mStreamType,
@@ -1566,7 +1572,8 @@
                     mSessionId,
                     AudioTrack::TRANSFER_DEFAULT,
                     NULL, // offload info
-                    mUid);
+                    mUid,
+                    mPid);
         }
 
         if ((t == 0) || (t->initCheck() != NO_ERROR)) {
diff --git a/media/libmediaplayerservice/MediaPlayerService.h b/media/libmediaplayerservice/MediaPlayerService.h
index 9c084e1..448f27a 100644
--- a/media/libmediaplayerservice/MediaPlayerService.h
+++ b/media/libmediaplayerservice/MediaPlayerService.h
@@ -72,7 +72,7 @@
         class CallbackData;
 
      public:
-                                AudioOutput(int sessionId, int uid);
+                                AudioOutput(int sessionId, int uid, int pid);
         virtual                 ~AudioOutput();
 
         virtual bool            ready() const { return mTrack != 0; }
@@ -140,6 +140,7 @@
         float                   mMsecsPerFrame;
         int                     mSessionId;
         int                     mUid;
+        int                     mPid;
         float                   mSendLevel;
         int                     mAuxEffectId;
         static bool             mIsOnEmulator;
@@ -211,12 +212,12 @@
         virtual void            flush() {}
         virtual void            pause() {}
         virtual void            close() {}
-                void            setAudioStreamType(audio_stream_type_t streamType) {}
+                void            setAudioStreamType(audio_stream_type_t streamType __unused) {}
                 // stream type is not used for AudioCache
         virtual audio_stream_type_t getAudioStreamType() const { return AUDIO_STREAM_DEFAULT; }
 
-                void            setVolume(float left, float right) {}
-        virtual status_t        setPlaybackRatePermille(int32_t ratePermille) { return INVALID_OPERATION; }
+                void            setVolume(float left __unused, float right __unused) {}
+        virtual status_t        setPlaybackRatePermille(int32_t ratePermille __unused) { return INVALID_OPERATION; }
                 uint32_t        sampleRate() const { return mSampleRate; }
                 audio_format_t  format() const { return mFormat; }
                 size_t          size() const { return mSize; }
@@ -256,9 +257,15 @@
 
     virtual sp<IMediaPlayer>    create(const sp<IMediaPlayerClient>& client, int audioSessionId);
 
-    virtual status_t            decode(const char* url, uint32_t *pSampleRate, int* pNumChannels,
-                                       audio_format_t* pFormat,
-                                       const sp<IMemoryHeap>& heap, size_t *pSize);
+    virtual status_t            decode(
+            const sp<IMediaHTTPService> &httpService,
+            const char* url,
+            uint32_t *pSampleRate,
+            int* pNumChannels,
+            audio_format_t* pFormat,
+            const sp<IMemoryHeap>& heap,
+            size_t *pSize);
+
     virtual status_t            decode(int fd, int64_t offset, int64_t length,
                                        uint32_t *pSampleRate, int* pNumChannels,
                                        audio_format_t* pFormat,
@@ -272,9 +279,6 @@
             const String8& iface);
     virtual status_t            dump(int fd, const Vector<String16>& args);
 
-    virtual status_t        updateProxyConfig(
-            const char *host, int32_t port, const char *exclusionList);
-
             void                removeClient(wp<Client> client);
 
     // For battery usage tracking purpose
@@ -356,6 +360,7 @@
         sp<MediaPlayerBase>     createPlayer(player_type playerType);
 
         virtual status_t        setDataSource(
+                        const sp<IMediaHTTPService> &httpService,
                         const char *url,
                         const KeyedVector<String8, String8> *headers);
 
diff --git a/media/libmediaplayerservice/MetadataRetrieverClient.cpp b/media/libmediaplayerservice/MetadataRetrieverClient.cpp
index 348957f..c61cf89 100644
--- a/media/libmediaplayerservice/MetadataRetrieverClient.cpp
+++ b/media/libmediaplayerservice/MetadataRetrieverClient.cpp
@@ -31,6 +31,7 @@
 #include <binder/MemoryHeapBase.h>
 #include <binder/IPCThreadState.h>
 #include <binder/IServiceManager.h>
+#include <media/IMediaHTTPService.h>
 #include <media/MediaMetadataRetrieverInterface.h>
 #include <media/MediaPlayerInterface.h>
 #include <private/media/VideoFrame.h>
@@ -106,7 +107,9 @@
 }
 
 status_t MetadataRetrieverClient::setDataSource(
-        const char *url, const KeyedVector<String8, String8> *headers)
+        const sp<IMediaHTTPService> &httpService,
+        const char *url,
+        const KeyedVector<String8, String8> *headers)
 {
     ALOGV("setDataSource(%s)", url);
     Mutex::Autolock lock(mLock);
@@ -127,7 +130,7 @@
     ALOGV("player type = %d", playerType);
     sp<MediaMetadataRetrieverBase> p = createRetriever(playerType);
     if (p == NULL) return NO_INIT;
-    status_t ret = p->setDataSource(url, headers);
+    status_t ret = p->setDataSource(httpService, url, headers);
     if (ret == NO_ERROR) mRetriever = p;
     return ret;
 }
diff --git a/media/libmediaplayerservice/MetadataRetrieverClient.h b/media/libmediaplayerservice/MetadataRetrieverClient.h
index f08f933..9d3fbe9 100644
--- a/media/libmediaplayerservice/MetadataRetrieverClient.h
+++ b/media/libmediaplayerservice/MetadataRetrieverClient.h
@@ -30,6 +30,7 @@
 
 namespace android {
 
+struct IMediaHTTPService;
 class IMediaPlayerService;
 class MemoryDealer;
 
@@ -43,7 +44,9 @@
     virtual void                    disconnect();
 
     virtual status_t                setDataSource(
-            const char *url, const KeyedVector<String8, String8> *headers);
+            const sp<IMediaHTTPService> &httpService,
+            const char *url,
+            const KeyedVector<String8, String8> *headers);
 
     virtual status_t                setDataSource(int fd, int64_t offset, int64_t length);
     virtual sp<IMemory>             getFrameAtTime(int64_t timeUs, int option);
diff --git a/media/libmediaplayerservice/MidiFile.cpp b/media/libmediaplayerservice/MidiFile.cpp
index 0a6aa90..deeddd1 100644
--- a/media/libmediaplayerservice/MidiFile.cpp
+++ b/media/libmediaplayerservice/MidiFile.cpp
@@ -114,7 +114,9 @@
 }
 
 status_t MidiFile::setDataSource(
-        const char* path, const KeyedVector<String8, String8> *) {
+        const sp<IMediaHTTPService> &httpService,
+        const char* path,
+        const KeyedVector<String8, String8> *) {
     ALOGV("MidiFile::setDataSource url=%s", path);
     Mutex::Autolock lock(mMutex);
 
diff --git a/media/libmediaplayerservice/MidiFile.h b/media/libmediaplayerservice/MidiFile.h
index 24d59b4..12802ba 100644
--- a/media/libmediaplayerservice/MidiFile.h
+++ b/media/libmediaplayerservice/MidiFile.h
@@ -32,7 +32,9 @@
     virtual status_t    initCheck();
 
     virtual status_t    setDataSource(
-            const char* path, const KeyedVector<String8, String8> *headers);
+            const sp<IMediaHTTPService> &httpService,
+            const char* path,
+            const KeyedVector<String8, String8> *headers);
 
     virtual status_t    setDataSource(int fd, int64_t offset, int64_t length);
     virtual status_t    setVideoSurfaceTexture(
diff --git a/media/libmediaplayerservice/MidiMetadataRetriever.cpp b/media/libmediaplayerservice/MidiMetadataRetriever.cpp
index 465209f..f3cf6ef 100644
--- a/media/libmediaplayerservice/MidiMetadataRetriever.cpp
+++ b/media/libmediaplayerservice/MidiMetadataRetriever.cpp
@@ -22,6 +22,8 @@
 #include "MidiMetadataRetriever.h"
 #include <media/mediametadataretriever.h>
 
+#include <media/IMediaHTTPService.h>
+
 namespace android {
 
 static status_t ERROR_NOT_OPEN = -1;
@@ -36,7 +38,9 @@
 }
 
 status_t MidiMetadataRetriever::setDataSource(
-        const char *url, const KeyedVector<String8, String8> *headers)
+        const sp<IMediaHTTPService> &httpService,
+        const char *url,
+        const KeyedVector<String8, String8> *headers)
 {
     ALOGV("setDataSource: %s", url? url: "NULL pointer");
     Mutex::Autolock lock(mLock);
@@ -44,7 +48,7 @@
     if (mMidiPlayer == 0) {
         mMidiPlayer = new MidiFile();
     }
-    return mMidiPlayer->setDataSource(url, headers);
+    return mMidiPlayer->setDataSource(httpService, url, headers);
 }
 
 status_t MidiMetadataRetriever::setDataSource(int fd, int64_t offset, int64_t length)
diff --git a/media/libmediaplayerservice/MidiMetadataRetriever.h b/media/libmediaplayerservice/MidiMetadataRetriever.h
index 4cee42d..b8214ee 100644
--- a/media/libmediaplayerservice/MidiMetadataRetriever.h
+++ b/media/libmediaplayerservice/MidiMetadataRetriever.h
@@ -32,7 +32,9 @@
                                    ~MidiMetadataRetriever() {}
 
     virtual status_t                setDataSource(
-            const char *url, const KeyedVector<String8, String8> *headers);
+            const sp<IMediaHTTPService> &httpService,
+            const char *url,
+            const KeyedVector<String8, String8> *headers);
 
     virtual status_t                setDataSource(int fd, int64_t offset, int64_t length);
     virtual const char*             extractMetadata(int keyCode);
diff --git a/media/libmediaplayerservice/StagefrightPlayer.cpp b/media/libmediaplayerservice/StagefrightPlayer.cpp
index de61d9b..b37aee3 100644
--- a/media/libmediaplayerservice/StagefrightPlayer.cpp
+++ b/media/libmediaplayerservice/StagefrightPlayer.cpp
@@ -54,8 +54,10 @@
 }
 
 status_t StagefrightPlayer::setDataSource(
-        const char *url, const KeyedVector<String8, String8> *headers) {
-    return mPlayer->setDataSource(url, headers);
+        const sp<IMediaHTTPService> &httpService,
+        const char *url,
+        const KeyedVector<String8, String8> *headers) {
+    return mPlayer->setDataSource(httpService, url, headers);
 }
 
 // Warning: The filedescriptor passed into this method will only be valid until
@@ -187,7 +189,7 @@
 }
 
 status_t StagefrightPlayer::getMetadata(
-        const media::Metadata::Filter& ids, Parcel *records) {
+        const media::Metadata::Filter& /* ids */, Parcel *records) {
     using media::Metadata;
 
     uint32_t flags = mPlayer->flags();
diff --git a/media/libmediaplayerservice/StagefrightPlayer.h b/media/libmediaplayerservice/StagefrightPlayer.h
index 600945e..e6c30ff 100644
--- a/media/libmediaplayerservice/StagefrightPlayer.h
+++ b/media/libmediaplayerservice/StagefrightPlayer.h
@@ -34,7 +34,9 @@
     virtual status_t setUID(uid_t uid);
 
     virtual status_t setDataSource(
-            const char *url, const KeyedVector<String8, String8> *headers);
+            const sp<IMediaHTTPService> &httpService,
+            const char *url,
+            const KeyedVector<String8, String8> *headers);
 
     virtual status_t setDataSource(int fd, int64_t offset, int64_t length);
 
diff --git a/media/libmediaplayerservice/StagefrightRecorder.cpp b/media/libmediaplayerservice/StagefrightRecorder.cpp
index 78dad19..d377acd 100644
--- a/media/libmediaplayerservice/StagefrightRecorder.cpp
+++ b/media/libmediaplayerservice/StagefrightRecorder.cpp
@@ -16,6 +16,7 @@
 
 //#define LOG_NDEBUG 0
 #define LOG_TAG "StagefrightRecorder"
+#include <inttypes.h>
 #include <utils/Log.h>
 
 #include "StagefrightRecorder.h"
@@ -24,8 +25,10 @@
 #include <binder/IServiceManager.h>
 
 #include <media/IMediaPlayerService.h>
-#include <media/openmax/OMX_Audio.h>
+#include <media/stagefright/foundation/ABuffer.h>
 #include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/foundation/ALooper.h>
 #include <media/stagefright/AudioSource.h>
 #include <media/stagefright/AMRWriter.h>
 #include <media/stagefright/AACWriter.h>
@@ -35,13 +38,12 @@
 #include <media/stagefright/MPEG4Writer.h>
 #include <media/stagefright/MediaDefs.h>
 #include <media/stagefright/MetaData.h>
+#include <media/stagefright/MediaCodecSource.h>
 #include <media/stagefright/OMXClient.h>
 #include <media/stagefright/OMXCodec.h>
-#include <media/stagefright/SurfaceMediaSource.h>
 #include <media/MediaProfiles.h>
 #include <camera/ICamera.h>
 #include <camera/CameraParameters.h>
-#include <gui/Surface.h>
 
 #include <utils/Errors.h>
 #include <sys/types.h>
@@ -71,8 +73,7 @@
       mAudioSource(AUDIO_SOURCE_CNT),
       mVideoSource(VIDEO_SOURCE_LIST_END),
       mCaptureTimeLapse(false),
-      mStarted(false),
-      mSurfaceMediaSource(NULL) {
+      mStarted(false) {
 
     ALOGV("Constructor");
     reset();
@@ -81,10 +82,19 @@
 StagefrightRecorder::~StagefrightRecorder() {
     ALOGV("Destructor");
     stop();
+
+    if (mLooper != NULL) {
+        mLooper->stop();
+    }
 }
 
 status_t StagefrightRecorder::init() {
     ALOGV("init");
+
+    mLooper = new ALooper;
+    mLooper->setName("recorder_looper");
+    mLooper->start();
+
     return OK;
 }
 
@@ -93,7 +103,7 @@
 // while encoding GL Frames
 sp<IGraphicBufferProducer> StagefrightRecorder::querySurfaceMediaSource() const {
     ALOGV("Get SurfaceMediaSource");
-    return mSurfaceMediaSource->getBufferQueue();
+    return mGraphicBufferProducer;
 }
 
 status_t StagefrightRecorder::setAudioSource(audio_source_t as) {
@@ -233,7 +243,7 @@
     return OK;
 }
 
-status_t StagefrightRecorder::setOutputFile(const char *path) {
+status_t StagefrightRecorder::setOutputFile(const char * /* path */) {
     ALOGE("setOutputFile(const char*) must not be called");
     // We don't actually support this at all, as the media_server process
     // no longer has permissions to create files.
@@ -739,15 +749,61 @@
 }
 
 status_t StagefrightRecorder::prepare() {
-    return OK;
-}
-
-status_t StagefrightRecorder::start() {
-    CHECK_GE(mOutputFd, 0);
+    ALOGV("prepare");
+    if (mOutputFd < 0) {
+        ALOGE("Output file descriptor is invalid");
+        return INVALID_OPERATION;
+    }
 
     // Get UID here for permission checking
     mClientUid = IPCThreadState::self()->getCallingUid();
-    if (mWriter != NULL) {
+
+    status_t status = OK;
+
+    switch (mOutputFormat) {
+        case OUTPUT_FORMAT_DEFAULT:
+        case OUTPUT_FORMAT_THREE_GPP:
+        case OUTPUT_FORMAT_MPEG_4:
+            status = setupMPEG4Recording();
+            break;
+
+        case OUTPUT_FORMAT_AMR_NB:
+        case OUTPUT_FORMAT_AMR_WB:
+            status = setupAMRRecording();
+            break;
+
+        case OUTPUT_FORMAT_AAC_ADIF:
+        case OUTPUT_FORMAT_AAC_ADTS:
+            status = setupAACRecording();
+            break;
+
+        case OUTPUT_FORMAT_RTP_AVP:
+            status = setupRTPRecording();
+            break;
+
+        case OUTPUT_FORMAT_MPEG2TS:
+            status = setupMPEG2TSRecording();
+            break;
+
+        default:
+            ALOGE("Unsupported output file format: %d", mOutputFormat);
+            status = UNKNOWN_ERROR;
+            break;
+    }
+
+    return status;
+}
+
+status_t StagefrightRecorder::start() {
+    ALOGV("start");
+    if (mOutputFd < 0) {
+        ALOGE("Output file descriptor is invalid");
+        return INVALID_OPERATION;
+    }
+
+    // Get UID here for permission checking
+    mClientUid = IPCThreadState::self()->getCallingUid();
+    if (mWriter == NULL) {
         ALOGE("File writer is not avaialble");
         return UNKNOWN_ERROR;
     }
@@ -758,31 +814,35 @@
         case OUTPUT_FORMAT_DEFAULT:
         case OUTPUT_FORMAT_THREE_GPP:
         case OUTPUT_FORMAT_MPEG_4:
-            status = startMPEG4Recording();
+        {
+            sp<MetaData> meta = new MetaData;
+            setupMPEG4MetaData(&meta);
+            status = mWriter->start(meta.get());
             break;
+        }
 
         case OUTPUT_FORMAT_AMR_NB:
         case OUTPUT_FORMAT_AMR_WB:
-            status = startAMRRecording();
-            break;
-
         case OUTPUT_FORMAT_AAC_ADIF:
         case OUTPUT_FORMAT_AAC_ADTS:
-            status = startAACRecording();
-            break;
-
         case OUTPUT_FORMAT_RTP_AVP:
-            status = startRTPRecording();
-            break;
-
         case OUTPUT_FORMAT_MPEG2TS:
-            status = startMPEG2TSRecording();
+        {
+            status = mWriter->start();
             break;
+        }
 
         default:
+        {
             ALOGE("Unsupported output file format: %d", mOutputFormat);
             status = UNKNOWN_ERROR;
             break;
+        }
+    }
+
+    if (status != OK) {
+        mWriter.clear();
+        mWriter = NULL;
     }
 
     if ((status == OK) && (!mStarted)) {
@@ -816,58 +876,54 @@
         return NULL;
     }
 
-    sp<MetaData> encMeta = new MetaData;
+    sp<AMessage> format = new AMessage;
     const char *mime;
     switch (mAudioEncoder) {
         case AUDIO_ENCODER_AMR_NB:
         case AUDIO_ENCODER_DEFAULT:
-            mime = MEDIA_MIMETYPE_AUDIO_AMR_NB;
+            format->setString("mime", MEDIA_MIMETYPE_AUDIO_AMR_NB);
             break;
         case AUDIO_ENCODER_AMR_WB:
-            mime = MEDIA_MIMETYPE_AUDIO_AMR_WB;
+            format->setString("mime", MEDIA_MIMETYPE_AUDIO_AMR_WB);
             break;
         case AUDIO_ENCODER_AAC:
-            mime = MEDIA_MIMETYPE_AUDIO_AAC;
-            encMeta->setInt32(kKeyAACProfile, OMX_AUDIO_AACObjectLC);
+            format->setString("mime", MEDIA_MIMETYPE_AUDIO_AAC);
+            format->setInt32("aac-profile", OMX_AUDIO_AACObjectLC);
             break;
         case AUDIO_ENCODER_HE_AAC:
-            mime = MEDIA_MIMETYPE_AUDIO_AAC;
-            encMeta->setInt32(kKeyAACProfile, OMX_AUDIO_AACObjectHE);
+            format->setString("mime", MEDIA_MIMETYPE_AUDIO_AAC);
+            format->setInt32("aac-profile", OMX_AUDIO_AACObjectHE);
             break;
         case AUDIO_ENCODER_AAC_ELD:
-            mime = MEDIA_MIMETYPE_AUDIO_AAC;
-            encMeta->setInt32(kKeyAACProfile, OMX_AUDIO_AACObjectELD);
+            format->setString("mime", MEDIA_MIMETYPE_AUDIO_AAC);
+            format->setInt32("aac-profile", OMX_AUDIO_AACObjectELD);
             break;
 
         default:
             ALOGE("Unknown audio encoder: %d", mAudioEncoder);
             return NULL;
     }
-    encMeta->setCString(kKeyMIMEType, mime);
 
     int32_t maxInputSize;
     CHECK(audioSource->getFormat()->findInt32(
                 kKeyMaxInputSize, &maxInputSize));
 
-    encMeta->setInt32(kKeyMaxInputSize, maxInputSize);
-    encMeta->setInt32(kKeyChannelCount, mAudioChannels);
-    encMeta->setInt32(kKeySampleRate, mSampleRate);
-    encMeta->setInt32(kKeyBitRate, mAudioBitRate);
+    format->setInt32("max-input-size", maxInputSize);
+    format->setInt32("channel-count", mAudioChannels);
+    format->setInt32("sample-rate", mSampleRate);
+    format->setInt32("bitrate", mAudioBitRate);
     if (mAudioTimeScale > 0) {
-        encMeta->setInt32(kKeyTimeScale, mAudioTimeScale);
+        format->setInt32("time-scale", mAudioTimeScale);
     }
 
-    OMXClient client;
-    CHECK_EQ(client.connect(), (status_t)OK);
     sp<MediaSource> audioEncoder =
-        OMXCodec::Create(client.interface(), encMeta,
-                         true /* createEncoder */, audioSource);
+            MediaCodecSource::Create(mLooper, format, audioSource);
     mAudioSourceNode = audioSource;
 
     return audioEncoder;
 }
 
-status_t StagefrightRecorder::startAACRecording() {
+status_t StagefrightRecorder::setupAACRecording() {
     // FIXME:
     // Add support for OUTPUT_FORMAT_AAC_ADIF
     CHECK_EQ(mOutputFormat, OUTPUT_FORMAT_AAC_ADTS);
@@ -878,16 +934,10 @@
     CHECK(mAudioSource != AUDIO_SOURCE_CNT);
 
     mWriter = new AACWriter(mOutputFd);
-    status_t status = startRawAudioRecording();
-    if (status != OK) {
-        mWriter.clear();
-        mWriter = NULL;
-    }
-
-    return status;
+    return setupRawAudioRecording();
 }
 
-status_t StagefrightRecorder::startAMRRecording() {
+status_t StagefrightRecorder::setupAMRRecording() {
     CHECK(mOutputFormat == OUTPUT_FORMAT_AMR_NB ||
           mOutputFormat == OUTPUT_FORMAT_AMR_WB);
 
@@ -907,15 +957,10 @@
     }
 
     mWriter = new AMRWriter(mOutputFd);
-    status_t status = startRawAudioRecording();
-    if (status != OK) {
-        mWriter.clear();
-        mWriter = NULL;
-    }
-    return status;
+    return setupRawAudioRecording();
 }
 
-status_t StagefrightRecorder::startRawAudioRecording() {
+status_t StagefrightRecorder::setupRawAudioRecording() {
     if (mAudioSource >= AUDIO_SOURCE_CNT) {
         ALOGE("Invalid audio source: %d", mAudioSource);
         return BAD_VALUE;
@@ -941,12 +986,11 @@
         mWriter->setMaxFileSize(mMaxFileSizeBytes);
     }
     mWriter->setListener(mListener);
-    mWriter->start();
 
     return OK;
 }
 
-status_t StagefrightRecorder::startRTPRecording() {
+status_t StagefrightRecorder::setupRTPRecording() {
     CHECK_EQ(mOutputFormat, OUTPUT_FORMAT_RTP_AVP);
 
     if ((mAudioSource != AUDIO_SOURCE_CNT
@@ -983,10 +1027,10 @@
     mWriter->addSource(source);
     mWriter->setListener(mListener);
 
-    return mWriter->start();
+    return OK;
 }
 
-status_t StagefrightRecorder::startMPEG2TSRecording() {
+status_t StagefrightRecorder::setupMPEG2TSRecording() {
     CHECK_EQ(mOutputFormat, OUTPUT_FORMAT_MPEG2TS);
 
     sp<MediaWriter> writer = new MPEG2TSWriter(mOutputFd);
@@ -1036,7 +1080,7 @@
 
     mWriter = writer;
 
-    return mWriter->start();
+    return OK;
 }
 
 void StagefrightRecorder::clipVideoFrameRate() {
@@ -1277,49 +1321,14 @@
             return err;
         }
         *mediaSource = cameraSource;
-    } else if (mVideoSource == VIDEO_SOURCE_GRALLOC_BUFFER) {
-        // If using GRAlloc buffers, setup surfacemediasource.
-        // Later a handle to that will be passed
-        // to the client side when queried
-        status_t err = setupSurfaceMediaSource();
-        if (err != OK) {
-            return err;
-        }
-        *mediaSource = mSurfaceMediaSource;
+    } else if (mVideoSource == VIDEO_SOURCE_SURFACE) {
+        *mediaSource = NULL;
     } else {
         return INVALID_OPERATION;
     }
     return OK;
 }
 
-// setupSurfaceMediaSource creates a source with the given
-// width and height and framerate.
-// TODO: This could go in a static function inside SurfaceMediaSource
-// similar to that in CameraSource
-status_t StagefrightRecorder::setupSurfaceMediaSource() {
-    status_t err = OK;
-    mSurfaceMediaSource = new SurfaceMediaSource(mVideoWidth, mVideoHeight);
-    if (mSurfaceMediaSource == NULL) {
-        return NO_INIT;
-    }
-
-    if (mFrameRate == -1) {
-        int32_t frameRate = 0;
-        CHECK (mSurfaceMediaSource->getFormat()->findInt32(
-                                        kKeyFrameRate, &frameRate));
-        ALOGI("Frame rate is not explicitly set. Use the current frame "
-             "rate (%d fps)", frameRate);
-        mFrameRate = frameRate;
-    } else {
-        err = mSurfaceMediaSource->setFrameRate(mFrameRate);
-    }
-    CHECK(mFrameRate != -1);
-
-    mIsMetaDataStoredInVideoBuffers =
-        mSurfaceMediaSource->isMetaDataStoredInVideoBuffers();
-    return err;
-}
-
 status_t StagefrightRecorder::setupCameraSource(
         sp<CameraSource> *cameraSource) {
     status_t err = OK;
@@ -1386,21 +1395,19 @@
         sp<MediaSource> *source) {
     source->clear();
 
-    sp<MetaData> enc_meta = new MetaData;
-    enc_meta->setInt32(kKeyBitRate, mVideoBitRate);
-    enc_meta->setInt32(kKeyFrameRate, mFrameRate);
+    sp<AMessage> format = new AMessage();
 
     switch (mVideoEncoder) {
         case VIDEO_ENCODER_H263:
-            enc_meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_VIDEO_H263);
+            format->setString("mime", MEDIA_MIMETYPE_VIDEO_H263);
             break;
 
         case VIDEO_ENCODER_MPEG_4_SP:
-            enc_meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_VIDEO_MPEG4);
+            format->setString("mime", MEDIA_MIMETYPE_VIDEO_MPEG4);
             break;
 
         case VIDEO_ENCODER_H264:
-            enc_meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_VIDEO_AVC);
+            format->setString("mime", MEDIA_MIMETYPE_VIDEO_AVC);
             break;
 
         default:
@@ -1408,59 +1415,69 @@
             break;
     }
 
-    sp<MetaData> meta = cameraSource->getFormat();
+    if (cameraSource != NULL) {
+        sp<MetaData> meta = cameraSource->getFormat();
 
-    int32_t width, height, stride, sliceHeight, colorFormat;
-    CHECK(meta->findInt32(kKeyWidth, &width));
-    CHECK(meta->findInt32(kKeyHeight, &height));
-    CHECK(meta->findInt32(kKeyStride, &stride));
-    CHECK(meta->findInt32(kKeySliceHeight, &sliceHeight));
-    CHECK(meta->findInt32(kKeyColorFormat, &colorFormat));
+        int32_t width, height, stride, sliceHeight, colorFormat;
+        CHECK(meta->findInt32(kKeyWidth, &width));
+        CHECK(meta->findInt32(kKeyHeight, &height));
+        CHECK(meta->findInt32(kKeyStride, &stride));
+        CHECK(meta->findInt32(kKeySliceHeight, &sliceHeight));
+        CHECK(meta->findInt32(kKeyColorFormat, &colorFormat));
 
-    enc_meta->setInt32(kKeyWidth, width);
-    enc_meta->setInt32(kKeyHeight, height);
-    enc_meta->setInt32(kKeyIFramesInterval, mIFramesIntervalSec);
-    enc_meta->setInt32(kKeyStride, stride);
-    enc_meta->setInt32(kKeySliceHeight, sliceHeight);
-    enc_meta->setInt32(kKeyColorFormat, colorFormat);
+        format->setInt32("width", width);
+        format->setInt32("height", height);
+        format->setInt32("stride", stride);
+        format->setInt32("slice-height", sliceHeight);
+        format->setInt32("color-format", colorFormat);
+    } else {
+        format->setInt32("width", mVideoWidth);
+        format->setInt32("height", mVideoHeight);
+        format->setInt32("stride", mVideoWidth);
+        format->setInt32("slice-height", mVideoWidth);
+        format->setInt32("color-format", OMX_COLOR_FormatAndroidOpaque);
+    }
+
+    format->setInt32("bitrate", mVideoBitRate);
+    format->setInt32("frame-rate", mFrameRate);
+    format->setInt32("i-frame-interval", mIFramesIntervalSec);
+
     if (mVideoTimeScale > 0) {
-        enc_meta->setInt32(kKeyTimeScale, mVideoTimeScale);
+        format->setInt32("time-scale", mVideoTimeScale);
     }
     if (mVideoEncoderProfile != -1) {
-        enc_meta->setInt32(kKeyVideoProfile, mVideoEncoderProfile);
+        format->setInt32("profile", mVideoEncoderProfile);
     }
     if (mVideoEncoderLevel != -1) {
-        enc_meta->setInt32(kKeyVideoLevel, mVideoEncoderLevel);
+        format->setInt32("level", mVideoEncoderLevel);
     }
 
-    OMXClient client;
-    CHECK_EQ(client.connect(), (status_t)OK);
-
-    uint32_t encoder_flags = 0;
+    uint32_t flags = 0;
     if (mIsMetaDataStoredInVideoBuffers) {
-        encoder_flags |= OMXCodec::kStoreMetaDataInVideoBuffers;
+        flags |= MediaCodecSource::FLAG_USE_METADATA_INPUT;
     }
 
-    // Do not wait for all the input buffers to become available.
-    // This give timelapse video recording faster response in
-    // receiving output from video encoder component.
-    if (mCaptureTimeLapse) {
-        encoder_flags |= OMXCodec::kOnlySubmitOneInputBufferAtOneTime;
+    if (cameraSource == NULL) {
+        flags |= MediaCodecSource::FLAG_USE_SURFACE_INPUT;
     }
 
-    sp<MediaSource> encoder = OMXCodec::Create(
-            client.interface(), enc_meta,
-            true /* createEncoder */, cameraSource,
-            NULL, encoder_flags);
+    sp<MediaCodecSource> encoder =
+            MediaCodecSource::Create(mLooper, format, cameraSource, flags);
     if (encoder == NULL) {
         ALOGW("Failed to create the encoder");
         // When the encoder fails to be created, we need
         // release the camera source due to the camera's lock
         // and unlock mechanism.
-        cameraSource->stop();
+        if (cameraSource != NULL) {
+            cameraSource->stop();
+        }
         return UNKNOWN_ERROR;
     }
 
+    if (cameraSource == NULL) {
+        mGraphicBufferProducer = encoder->getGraphicBufferProducer();
+    }
+
     *source = encoder;
 
     return OK;
@@ -1494,9 +1511,10 @@
     return OK;
 }
 
-status_t StagefrightRecorder::setupMPEG4Recording(int32_t *totalBitRate) {
+status_t StagefrightRecorder::setupMPEG4Recording() {
     mWriter.clear();
-    *totalBitRate = 0;
+    mTotalBitRate = 0;
+
     status_t err = OK;
     sp<MediaWriter> writer = new MPEG4Writer(mOutputFd);
 
@@ -1515,7 +1533,7 @@
         }
 
         writer->addSource(encoder);
-        *totalBitRate += mVideoBitRate;
+        mTotalBitRate += mVideoBitRate;
     }
 
     // Audio source is added at the end if it exists.
@@ -1524,7 +1542,7 @@
     if (!mCaptureTimeLapse && (mAudioSource != AUDIO_SOURCE_CNT)) {
         err = setupAudioEncoder(writer);
         if (err != OK) return err;
-        *totalBitRate += mAudioBitRate;
+        mTotalBitRate += mAudioBitRate;
     }
 
     if (mInterleaveDurationUs > 0) {
@@ -1542,7 +1560,13 @@
         writer->setMaxFileSize(mMaxFileSizeBytes);
     }
 
-    mStartTimeOffsetMs = mEncoderProfiles->getStartTimeOffsetMs(mCameraId);
+    if (mVideoSource == VIDEO_SOURCE_DEFAULT
+            || mVideoSource == VIDEO_SOURCE_CAMERA) {
+        mStartTimeOffsetMs = mEncoderProfiles->getStartTimeOffsetMs(mCameraId);
+    } else if (mVideoSource == VIDEO_SOURCE_SURFACE) {
+        // surface source doesn't need large initial delay
+        mStartTimeOffsetMs = 200;
+    }
     if (mStartTimeOffsetMs > 0) {
         reinterpret_cast<MPEG4Writer *>(writer.get())->
             setStartTimeOffsetMs(mStartTimeOffsetMs);
@@ -1553,11 +1577,11 @@
     return OK;
 }
 
-void StagefrightRecorder::setupMPEG4MetaData(int64_t startTimeUs, int32_t totalBitRate,
-        sp<MetaData> *meta) {
+void StagefrightRecorder::setupMPEG4MetaData(sp<MetaData> *meta) {
+    int64_t startTimeUs = systemTime() / 1000;
     (*meta)->setInt64(kKeyTime, startTimeUs);
     (*meta)->setInt32(kKeyFileType, mOutputFormat);
-    (*meta)->setInt32(kKeyBitRate, totalBitRate);
+    (*meta)->setInt32(kKeyBitRate, mTotalBitRate);
     (*meta)->setInt32(kKey64BitFileOffset, mUse64BitFileOffset);
     if (mMovieTimeScale > 0) {
         (*meta)->setInt32(kKeyTimeScale, mMovieTimeScale);
@@ -1570,25 +1594,6 @@
     }
 }
 
-status_t StagefrightRecorder::startMPEG4Recording() {
-    int32_t totalBitRate;
-    status_t err = setupMPEG4Recording(&totalBitRate);
-    if (err != OK) {
-        return err;
-    }
-
-    int64_t startTimeUs = systemTime() / 1000;
-    sp<MetaData> meta = new MetaData;
-    setupMPEG4MetaData(startTimeUs, totalBitRate, &meta);
-
-    err = mWriter->start(meta.get());
-    if (err != OK) {
-        return err;
-    }
-
-    return OK;
-}
-
 status_t StagefrightRecorder::pause() {
     ALOGV("pause");
     if (mWriter == NULL) {
@@ -1628,6 +1633,8 @@
         mWriter.clear();
     }
 
+    mGraphicBufferProducer.clear();
+
     if (mOutputFd >= 0) {
         ::close(mOutputFd);
         mOutputFd = -1;
@@ -1647,7 +1654,6 @@
         addBatteryData(params);
     }
 
-
     return err;
 }
 
@@ -1699,6 +1705,7 @@
     mRotationDegrees = 0;
     mLatitudex10000 = -3600000;
     mLongitudex10000 = -3600000;
+    mTotalBitRate = 0;
 
     mOutputFd = -1;
 
@@ -1739,15 +1746,15 @@
     result.append(buffer);
     snprintf(buffer, SIZE, "     File format: %d\n", mOutputFormat);
     result.append(buffer);
-    snprintf(buffer, SIZE, "     Max file size (bytes): %lld\n", mMaxFileSizeBytes);
+    snprintf(buffer, SIZE, "     Max file size (bytes): %" PRId64 "\n", mMaxFileSizeBytes);
     result.append(buffer);
-    snprintf(buffer, SIZE, "     Max file duration (us): %lld\n", mMaxFileDurationUs);
+    snprintf(buffer, SIZE, "     Max file duration (us): %" PRId64 "\n", mMaxFileDurationUs);
     result.append(buffer);
     snprintf(buffer, SIZE, "     File offset length (bits): %d\n", mUse64BitFileOffset? 64: 32);
     result.append(buffer);
     snprintf(buffer, SIZE, "     Interleave duration (us): %d\n", mInterleaveDurationUs);
     result.append(buffer);
-    snprintf(buffer, SIZE, "     Progress notification: %lld us\n", mTrackEveryTimeDurationUs);
+    snprintf(buffer, SIZE, "     Progress notification: %" PRId64 " us\n", mTrackEveryTimeDurationUs);
     result.append(buffer);
     snprintf(buffer, SIZE, "   Audio\n");
     result.append(buffer);
diff --git a/media/libmediaplayerservice/StagefrightRecorder.h b/media/libmediaplayerservice/StagefrightRecorder.h
index bc43488..7d6abd3 100644
--- a/media/libmediaplayerservice/StagefrightRecorder.h
+++ b/media/libmediaplayerservice/StagefrightRecorder.h
@@ -37,6 +37,7 @@
 class MediaProfiles;
 class IGraphicBufferProducer;
 class SurfaceMediaSource;
+class ALooper;
 
 struct StagefrightRecorder : public MediaRecorderBase {
     StagefrightRecorder();
@@ -106,6 +107,7 @@
     int32_t mLatitudex10000;
     int32_t mLongitudex10000;
     int32_t mStartTimeOffsetMs;
+    int32_t mTotalBitRate;
 
     bool mCaptureTimeLapse;
     int64_t mTimeBetweenTimeLapseFrameCaptureUs;
@@ -122,17 +124,16 @@
     // An <IGraphicBufferProducer> pointer
     // will be sent to the client side using which the
     // frame buffers will be queued and dequeued
-    sp<SurfaceMediaSource> mSurfaceMediaSource;
+    sp<IGraphicBufferProducer> mGraphicBufferProducer;
+    sp<ALooper> mLooper;
 
-    status_t setupMPEG4Recording(int32_t *totalBitRate);
-    void setupMPEG4MetaData(int64_t startTimeUs, int32_t totalBitRate,
-        sp<MetaData> *meta);
-    status_t startMPEG4Recording();
-    status_t startAMRRecording();
-    status_t startAACRecording();
-    status_t startRawAudioRecording();
-    status_t startRTPRecording();
-    status_t startMPEG2TSRecording();
+    status_t setupMPEG4Recording();
+    void setupMPEG4MetaData(sp<MetaData> *meta);
+    status_t setupAMRRecording();
+    status_t setupAACRecording();
+    status_t setupRawAudioRecording();
+    status_t setupRTPRecording();
+    status_t setupMPEG2TSRecording();
     sp<MediaSource> createAudioSource();
     status_t checkVideoEncoderCapabilities(
             bool *supportsCameraSourceMetaDataMode);
@@ -142,9 +143,6 @@
     // depending on the videosource type
     status_t setupMediaSource(sp<MediaSource> *mediaSource);
     status_t setupCameraSource(sp<CameraSource> *cameraSource);
-    // setup the surfacemediasource for the encoder
-    status_t setupSurfaceMediaSource();
-
     status_t setupAudioEncoder(const sp<MediaWriter>& writer);
     status_t setupVideoEncoder(sp<MediaSource> cameraSource, sp<MediaSource> *source);
 
diff --git a/media/libmediaplayerservice/TestPlayerStub.cpp b/media/libmediaplayerservice/TestPlayerStub.cpp
index 5d9728a..5795773 100644
--- a/media/libmediaplayerservice/TestPlayerStub.cpp
+++ b/media/libmediaplayerservice/TestPlayerStub.cpp
@@ -113,7 +113,9 @@
 // Create the test player.
 // Call setDataSource on the test player with the url in param.
 status_t TestPlayerStub::setDataSource(
-        const char *url, const KeyedVector<String8, String8> *headers) {
+        const sp<IMediaHTTPService> &httpService,
+        const char *url,
+        const KeyedVector<String8, String8> *headers) {
     if (!isTestUrl(url) || NULL != mHandle) {
         return INVALID_OPERATION;
     }
@@ -162,7 +164,7 @@
     }
 
     mPlayer = (*mNewPlayer)();
-    return mPlayer->setDataSource(mContentUrl, headers);
+    return mPlayer->setDataSource(httpService, mContentUrl, headers);
 }
 
 // Internal cleanup.
diff --git a/media/libmediaplayerservice/TestPlayerStub.h b/media/libmediaplayerservice/TestPlayerStub.h
index a3802eb..55bf2c8 100644
--- a/media/libmediaplayerservice/TestPlayerStub.h
+++ b/media/libmediaplayerservice/TestPlayerStub.h
@@ -66,7 +66,9 @@
 
     // @param url Should be a test url. See class comment.
     virtual status_t setDataSource(
-            const char* url, const KeyedVector<String8, String8> *headers);
+            const sp<IMediaHTTPService> &httpService,
+            const char* url,
+            const KeyedVector<String8, String8> *headers);
 
     // Test player for a file descriptor source is not supported.
     virtual status_t setDataSource(int, int64_t, int64_t)  {
diff --git a/media/libmediaplayerservice/nuplayer/Android.mk b/media/libmediaplayerservice/nuplayer/Android.mk
index f946c1c..f97ba57 100644
--- a/media/libmediaplayerservice/nuplayer/Android.mk
+++ b/media/libmediaplayerservice/nuplayer/Android.mk
@@ -11,7 +11,6 @@
         NuPlayerStreamListener.cpp      \
         RTSPSource.cpp                  \
         StreamingSource.cpp             \
-        mp4/MP4Source.cpp               \
 
 LOCAL_C_INCLUDES := \
 	$(TOP)/frameworks/av/media/libstagefright/httplive            \
diff --git a/media/libmediaplayerservice/nuplayer/GenericSource.cpp b/media/libmediaplayerservice/nuplayer/GenericSource.cpp
index b04e7a6..06aac33 100644
--- a/media/libmediaplayerservice/nuplayer/GenericSource.cpp
+++ b/media/libmediaplayerservice/nuplayer/GenericSource.cpp
@@ -33,17 +33,16 @@
 
 NuPlayer::GenericSource::GenericSource(
         const sp<AMessage> &notify,
+        const sp<IMediaHTTPService> &httpService,
         const char *url,
-        const KeyedVector<String8, String8> *headers,
-        bool uidValid,
-        uid_t uid)
+        const KeyedVector<String8, String8> *headers)
     : Source(notify),
       mDurationUs(0ll),
       mAudioIsVorbis(false) {
     DataSource::RegisterDefaultSniffers();
 
     sp<DataSource> dataSource =
-        DataSource::CreateFromURI(url, headers);
+        DataSource::CreateFromURI(httpService, url, headers);
     CHECK(dataSource != NULL);
 
     initFromDataSource(dataSource);
diff --git a/media/libmediaplayerservice/nuplayer/GenericSource.h b/media/libmediaplayerservice/nuplayer/GenericSource.h
index 2da680c..20d597e 100644
--- a/media/libmediaplayerservice/nuplayer/GenericSource.h
+++ b/media/libmediaplayerservice/nuplayer/GenericSource.h
@@ -33,10 +33,9 @@
 struct NuPlayer::GenericSource : public NuPlayer::Source {
     GenericSource(
             const sp<AMessage> &notify,
+            const sp<IMediaHTTPService> &httpService,
             const char *url,
-            const KeyedVector<String8, String8> *headers,
-            bool uidValid = false,
-            uid_t uid = 0);
+            const KeyedVector<String8, String8> *headers);
 
     GenericSource(
             const sp<AMessage> &notify,
diff --git a/media/libmediaplayerservice/nuplayer/HTTPLiveSource.cpp b/media/libmediaplayerservice/nuplayer/HTTPLiveSource.cpp
index f1782cc..cbedf5c 100644
--- a/media/libmediaplayerservice/nuplayer/HTTPLiveSource.cpp
+++ b/media/libmediaplayerservice/nuplayer/HTTPLiveSource.cpp
@@ -24,6 +24,7 @@
 #include "LiveDataSource.h"
 #include "LiveSession.h"
 
+#include <media/IMediaHTTPService.h>
 #include <media/stagefright/foundation/ABuffer.h>
 #include <media/stagefright/foundation/ADebug.h>
 #include <media/stagefright/foundation/AMessage.h>
@@ -34,13 +35,12 @@
 
 NuPlayer::HTTPLiveSource::HTTPLiveSource(
         const sp<AMessage> &notify,
+        const sp<IMediaHTTPService> &httpService,
         const char *url,
-        const KeyedVector<String8, String8> *headers,
-        bool uidValid, uid_t uid)
+        const KeyedVector<String8, String8> *headers)
     : Source(notify),
+      mHTTPService(httpService),
       mURL(url),
-      mUIDValid(uidValid),
-      mUID(uid),
       mFlags(0),
       mFinalResult(OK),
       mOffset(0),
@@ -79,8 +79,7 @@
     mLiveSession = new LiveSession(
             notify,
             (mFlags & kFlagIncognito) ? LiveSession::kFlagIncognito : 0,
-            mUIDValid,
-            mUID);
+            mHTTPService);
 
     mLiveLooper->registerHandler(mLiveSession);
 
@@ -140,7 +139,7 @@
     // LiveSession::selectTrack returns BAD_VALUE when selecting the currently
     // selected track, or unselecting a non-selected track. In this case it's an
     // no-op so we return OK.
-    return (err == OK || err == BAD_VALUE) ? OK : err;
+    return (err == OK || err == BAD_VALUE) ? (status_t)OK : err;
 }
 
 status_t NuPlayer::HTTPLiveSource::seekTo(int64_t seekTimeUs) {
diff --git a/media/libmediaplayerservice/nuplayer/HTTPLiveSource.h b/media/libmediaplayerservice/nuplayer/HTTPLiveSource.h
index bcc3f8b..4d7251f 100644
--- a/media/libmediaplayerservice/nuplayer/HTTPLiveSource.h
+++ b/media/libmediaplayerservice/nuplayer/HTTPLiveSource.h
@@ -28,10 +28,9 @@
 struct NuPlayer::HTTPLiveSource : public NuPlayer::Source {
     HTTPLiveSource(
             const sp<AMessage> &notify,
+            const sp<IMediaHTTPService> &httpService,
             const char *url,
-            const KeyedVector<String8, String8> *headers,
-            bool uidValid = false,
-            uid_t uid = 0);
+            const KeyedVector<String8, String8> *headers);
 
     virtual void prepareAsync();
     virtual void start();
@@ -61,10 +60,9 @@
         kWhatFetchSubtitleData,
     };
 
+    sp<IMediaHTTPService> mHTTPService;
     AString mURL;
     KeyedVector<String8, String8> mExtraHeaders;
-    bool mUIDValid;
-    uid_t mUID;
     uint32_t mFlags;
     status_t mFinalResult;
     off64_t mOffset;
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
index 3669a5b..d47ac98 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
@@ -28,13 +28,11 @@
 #include "RTSPSource.h"
 #include "StreamingSource.h"
 #include "GenericSource.h"
-#include "mp4/MP4Source.h"
 
 #include "ATSParser.h"
 
 #include "SoftwareRenderer.h"
 
-#include <cutils/properties.h> // for property_get
 #include <media/stagefright/foundation/hexdump.h>
 #include <media/stagefright/foundation/ABuffer.h>
 #include <media/stagefright/foundation/ADebug.h>
@@ -183,14 +181,7 @@
 
     sp<AMessage> notify = new AMessage(kWhatSourceNotify, id());
 
-    char prop[PROPERTY_VALUE_MAX];
-    if (property_get("media.stagefright.use-mp4source", prop, NULL)
-            && (!strcmp(prop, "1") || !strcasecmp(prop, "true"))) {
-        msg->setObject("source", new MP4Source(notify, source));
-    } else {
-        msg->setObject("source", new StreamingSource(notify, source));
-    }
-
+    msg->setObject("source", new StreamingSource(notify, source));
     msg->post();
 }
 
@@ -212,7 +203,9 @@
 }
 
 void NuPlayer::setDataSourceAsync(
-        const char *url, const KeyedVector<String8, String8> *headers) {
+        const sp<IMediaHTTPService> &httpService,
+        const char *url,
+        const KeyedVector<String8, String8> *headers) {
     sp<AMessage> msg = new AMessage(kWhatSetDataSource, id());
     size_t len = strlen(url);
 
@@ -220,16 +213,18 @@
 
     sp<Source> source;
     if (IsHTTPLiveURL(url)) {
-        source = new HTTPLiveSource(notify, url, headers, mUIDValid, mUID);
+        source = new HTTPLiveSource(notify, httpService, url, headers);
     } else if (!strncasecmp(url, "rtsp://", 7)) {
-        source = new RTSPSource(notify, url, headers, mUIDValid, mUID);
+        source = new RTSPSource(
+                notify, httpService, url, headers, mUIDValid, mUID);
     } else if ((!strncasecmp(url, "http://", 7)
                 || !strncasecmp(url, "https://", 8))
                     && ((len >= 4 && !strcasecmp(".sdp", &url[len - 4]))
                     || strstr(url, ".sdp?"))) {
-        source = new RTSPSource(notify, url, headers, mUIDValid, mUID, true);
+        source = new RTSPSource(
+                notify, httpService, url, headers, mUIDValid, mUID, true);
     } else {
-        source = new GenericSource(notify, url, headers, mUIDValid, mUID);
+        source = new GenericSource(notify, httpService, url, headers);
     }
 
     msg->setObject("source", source);
@@ -1524,7 +1519,7 @@
     notify->post();
 }
 
-void NuPlayer::Source::onMessageReceived(const sp<AMessage> &msg) {
+void NuPlayer::Source::onMessageReceived(const sp<AMessage> & /* msg */) {
     TRESPASS();
 }
 
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.h b/media/libmediaplayerservice/nuplayer/NuPlayer.h
index 590e1f2..9dfe4a0 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayer.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayer.h
@@ -38,7 +38,9 @@
     void setDataSourceAsync(const sp<IStreamSource> &source);
 
     void setDataSourceAsync(
-            const char *url, const KeyedVector<String8, String8> *headers);
+            const sp<IMediaHTTPService> &httpService,
+            const char *url,
+            const KeyedVector<String8, String8> *headers);
 
     void setDataSourceAsync(int fd, int64_t offset, int64_t length);
 
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
index 47834fd..e4850f0 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
@@ -16,6 +16,7 @@
 
 //#define LOG_NDEBUG 0
 #define LOG_TAG "NuPlayerDriver"
+#include <inttypes.h>
 #include <utils/Log.h>
 
 #include "NuPlayerDriver.h"
@@ -70,7 +71,9 @@
 }
 
 status_t NuPlayerDriver::setDataSource(
-        const char *url, const KeyedVector<String8, String8> *headers) {
+        const sp<IMediaHTTPService> &httpService,
+        const char *url,
+        const KeyedVector<String8, String8> *headers) {
     Mutex::Autolock autoLock(mLock);
 
     if (mState != STATE_IDLE) {
@@ -79,7 +82,7 @@
 
     mState = STATE_SET_DATASOURCE_PENDING;
 
-    mPlayer->setDataSourceAsync(url, headers);
+    mPlayer->setDataSourceAsync(httpService, url, headers);
 
     while (mState == STATE_SET_DATASOURCE_PENDING) {
         mCondition.wait(mLock);
@@ -364,7 +367,7 @@
     return OK;
 }
 
-status_t NuPlayerDriver::setLooping(int loop) {
+status_t NuPlayerDriver::setLooping(int /* loop */) {
     return INVALID_OPERATION;
 }
 
@@ -420,16 +423,17 @@
     mPlayer->setAudioSink(audioSink);
 }
 
-status_t NuPlayerDriver::setParameter(int key, const Parcel &request) {
+status_t NuPlayerDriver::setParameter(
+        int /* key */, const Parcel & /* request */) {
     return INVALID_OPERATION;
 }
 
-status_t NuPlayerDriver::getParameter(int key, Parcel *reply) {
+status_t NuPlayerDriver::getParameter(int /* key */, Parcel * /* reply */) {
     return INVALID_OPERATION;
 }
 
 status_t NuPlayerDriver::getMetadata(
-        const media::Metadata::Filter& ids, Parcel *records) {
+        const media::Metadata::Filter& /* ids */, Parcel *records) {
     Mutex::Autolock autoLock(mLock);
 
     using media::Metadata;
@@ -493,13 +497,14 @@
     mNumFramesDropped = numFramesDropped;
 }
 
-status_t NuPlayerDriver::dump(int fd, const Vector<String16> &args) const {
+status_t NuPlayerDriver::dump(
+        int fd, const Vector<String16> & /* args */) const {
     Mutex::Autolock autoLock(mLock);
 
     FILE *out = fdopen(dup(fd), "w");
 
     fprintf(out, " NuPlayer\n");
-    fprintf(out, "  numFramesTotal(%lld), numFramesDropped(%lld), "
+    fprintf(out, "  numFramesTotal(%" PRId64 "), numFramesDropped(%" PRId64 "), "
                  "percentageDropped(%.2f)\n",
                  mNumFramesTotal,
                  mNumFramesDropped,
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.h b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.h
index 99f72a6..0148fb1 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.h
@@ -31,7 +31,9 @@
     virtual status_t setUID(uid_t uid);
 
     virtual status_t setDataSource(
-            const char *url, const KeyedVector<String8, String8> *headers);
+            const sp<IMediaHTTPService> &httpService,
+            const char *url,
+            const KeyedVector<String8, String8> *headers);
 
     virtual status_t setDataSource(int fd, int64_t offset, int64_t length);
 
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
index 92b9a92..bf5271e 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
@@ -422,7 +422,9 @@
         ALOGV("video late by %lld us (%.2f secs)",
              mVideoLateByUs, mVideoLateByUs / 1E6);
     } else {
-        ALOGV("rendering video at media time %.2f secs", mediaTimeUs / 1E6);
+        ALOGV("rendering video at media time %.2f secs",
+                (mFlags & FLAG_REAL_TIME ? realTimeUs :
+                (realTimeUs + mAnchorTimeMediaUs - mAnchorTimeRealUs)) / 1E6);
         if (mSoftRenderer != NULL) {
             mSoftRenderer->render(entry->mBuffer->data(), entry->mBuffer->size(), NULL);
         }
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerSource.h b/media/libmediaplayerservice/nuplayer/NuPlayerSource.h
index e50533a..11279fc 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerSource.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerSource.h
@@ -68,19 +68,19 @@
     virtual status_t dequeueAccessUnit(
             bool audio, sp<ABuffer> *accessUnit) = 0;
 
-    virtual status_t getDuration(int64_t *durationUs) {
+    virtual status_t getDuration(int64_t * /* durationUs */) {
         return INVALID_OPERATION;
     }
 
-    virtual status_t getTrackInfo(Parcel* reply) const {
+    virtual status_t getTrackInfo(Parcel* /* reply */) const {
         return INVALID_OPERATION;
     }
 
-    virtual status_t selectTrack(size_t trackIndex, bool select) {
+    virtual status_t selectTrack(size_t /* trackIndex */, bool /* select */) {
         return INVALID_OPERATION;
     }
 
-    virtual status_t seekTo(int64_t seekTimeUs) {
+    virtual status_t seekTo(int64_t /* seekTimeUs */) {
         return INVALID_OPERATION;
     }
 
@@ -93,7 +93,7 @@
 
     virtual void onMessageReceived(const sp<AMessage> &msg);
 
-    virtual sp<MetaData> getFormatMeta(bool audio) { return NULL; }
+    virtual sp<MetaData> getFormatMeta(bool /* audio */) { return NULL; }
 
     sp<AMessage> dupNotify() const { return mNotify->dup(); }
 
diff --git a/media/libmediaplayerservice/nuplayer/RTSPSource.cpp b/media/libmediaplayerservice/nuplayer/RTSPSource.cpp
index 18cf6d1..94800ba 100644
--- a/media/libmediaplayerservice/nuplayer/RTSPSource.cpp
+++ b/media/libmediaplayerservice/nuplayer/RTSPSource.cpp
@@ -24,6 +24,7 @@
 #include "MyHandler.h"
 #include "SDPLoader.h"
 
+#include <media/IMediaHTTPService.h>
 #include <media/stagefright/MediaDefs.h>
 #include <media/stagefright/MetaData.h>
 
@@ -33,12 +34,14 @@
 
 NuPlayer::RTSPSource::RTSPSource(
         const sp<AMessage> &notify,
+        const sp<IMediaHTTPService> &httpService,
         const char *url,
         const KeyedVector<String8, String8> *headers,
         bool uidValid,
         uid_t uid,
         bool isSDP)
     : Source(notify),
+      mHTTPService(httpService),
       mURL(url),
       mUIDValid(uidValid),
       mUID(uid),
@@ -92,7 +95,7 @@
     if (mIsSDP) {
         mSDPLoader = new SDPLoader(notify,
                 (mFlags & kFlagIncognito) ? SDPLoader::kFlagIncognito : 0,
-                mUIDValid, mUID);
+                mHTTPService);
 
         mSDPLoader->load(
                 mURL.c_str(), mExtraHeaders.isEmpty() ? NULL : &mExtraHeaders);
diff --git a/media/libmediaplayerservice/nuplayer/RTSPSource.h b/media/libmediaplayerservice/nuplayer/RTSPSource.h
index 8cf34a0..3718bf9 100644
--- a/media/libmediaplayerservice/nuplayer/RTSPSource.h
+++ b/media/libmediaplayerservice/nuplayer/RTSPSource.h
@@ -34,6 +34,7 @@
 struct NuPlayer::RTSPSource : public NuPlayer::Source {
     RTSPSource(
             const sp<AMessage> &notify,
+            const sp<IMediaHTTPService> &httpService,
             const char *url,
             const KeyedVector<String8, String8> *headers,
             bool uidValid = false,
@@ -88,6 +89,7 @@
         bool mNPTMappingValid;
     };
 
+    sp<IMediaHTTPService> mHTTPService;
     AString mURL;
     KeyedVector<String8, String8> mExtraHeaders;
     bool mUIDValid;
diff --git a/media/libmediaplayerservice/nuplayer/mp4/MP4Source.cpp b/media/libmediaplayerservice/nuplayer/mp4/MP4Source.cpp
deleted file mode 100644
index d31d947..0000000
--- a/media/libmediaplayerservice/nuplayer/mp4/MP4Source.cpp
+++ /dev/null
@@ -1,144 +0,0 @@
-/*
- * Copyright (C) 2012 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#include "MP4Source.h"
-
-#include "FragmentedMP4Parser.h"
-#include "../NuPlayerStreamListener.h"
-
-#include <media/IStreamSource.h>
-#include <media/stagefright/foundation/AMessage.h>
-#include <media/stagefright/MediaErrors.h>
-#include <media/stagefright/MetaData.h>
-
-namespace android {
-
-struct StreamSource : public FragmentedMP4Parser::Source {
-    StreamSource(const sp<IStreamSource> &source)
-        : mListener(new NuPlayer::NuPlayerStreamListener(source, 0)),
-          mPosition(0) {
-        mListener->start();
-    }
-
-    virtual ssize_t readAt(off64_t offset, void *data, size_t size) {
-        if (offset < mPosition) {
-            return -EPIPE;
-        }
-
-        while (offset > mPosition) {
-            char buffer[1024];
-            off64_t skipBytes = offset - mPosition;
-            if (skipBytes > sizeof(buffer)) {
-                skipBytes = sizeof(buffer);
-            }
-
-            sp<AMessage> extra;
-            ssize_t n;
-            for (;;) {
-                n = mListener->read(buffer, skipBytes, &extra);
-
-                if (n == -EWOULDBLOCK) {
-                    usleep(10000);
-                    continue;
-                }
-
-                break;
-            }
-
-            ALOGV("skipped %ld bytes at offset %lld", n, mPosition);
-
-            if (n < 0) {
-                return n;
-            }
-
-            mPosition += n;
-        }
-
-        sp<AMessage> extra;
-        size_t total = 0;
-        while (total < size) {
-            ssize_t n = mListener->read(
-                    (uint8_t *)data + total, size - total, &extra);
-
-            if (n == -EWOULDBLOCK) {
-                usleep(10000);
-                continue;
-            } else if (n == 0) {
-                break;
-            } else if (n < 0) {
-                mPosition += total;
-                return n;
-            }
-
-            total += n;
-        }
-
-        ALOGV("read %ld bytes at offset %lld", n, mPosition);
-
-        mPosition += total;
-
-        return total;
-    }
-
-    bool isSeekable() {
-        return false;
-    }
-
-private:
-    sp<NuPlayer::NuPlayerStreamListener> mListener;
-    off64_t mPosition;
-
-    DISALLOW_EVIL_CONSTRUCTORS(StreamSource);
-};
-
-MP4Source::MP4Source(
-        const sp<AMessage> &notify, const sp<IStreamSource> &source)
-    : Source(notify),
-      mSource(source),
-      mLooper(new ALooper),
-      mParser(new FragmentedMP4Parser),
-      mEOS(false) {
-    mLooper->registerHandler(mParser);
-}
-
-MP4Source::~MP4Source() {
-}
-
-void MP4Source::prepareAsync() {
-    notifyVideoSizeChanged(0, 0);
-    notifyFlagsChanged(0);
-    notifyPrepared();
-}
-
-void MP4Source::start() {
-    mLooper->start(false /* runOnCallingThread */);
-    mParser->start(new StreamSource(mSource));
-}
-
-status_t MP4Source::feedMoreTSData() {
-    return mEOS ? ERROR_END_OF_STREAM : (status_t)OK;
-}
-
-sp<AMessage> MP4Source::getFormat(bool audio) {
-    return mParser->getFormat(audio);
-}
-
-status_t MP4Source::dequeueAccessUnit(
-        bool audio, sp<ABuffer> *accessUnit) {
-    return mParser->dequeueAccessUnit(audio, accessUnit);
-}
-
-}  // namespace android
diff --git a/media/libmediaplayerservice/nuplayer/mp4/MP4Source.h b/media/libmediaplayerservice/nuplayer/mp4/MP4Source.h
deleted file mode 100644
index a6ef622..0000000
--- a/media/libmediaplayerservice/nuplayer/mp4/MP4Source.h
+++ /dev/null
@@ -1,53 +0,0 @@
-/*
- * Copyright (C) 2012 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef MP4_SOURCE_H
-#define MP4_SOURCE_H
-
-#include "NuPlayerSource.h"
-
-namespace android {
-
-struct FragmentedMP4Parser;
-
-struct MP4Source : public NuPlayer::Source {
-    MP4Source(const sp<AMessage> &notify, const sp<IStreamSource> &source);
-
-    virtual void prepareAsync();
-    virtual void start();
-
-    virtual status_t feedMoreTSData();
-
-    virtual sp<AMessage> getFormat(bool audio);
-
-    virtual status_t dequeueAccessUnit(
-            bool audio, sp<ABuffer> *accessUnit);
-
-protected:
-    virtual ~MP4Source();
-
-private:
-    sp<IStreamSource> mSource;
-    sp<ALooper> mLooper;
-    sp<FragmentedMP4Parser> mParser;
-    bool mEOS;
-
-    DISALLOW_EVIL_CONSTRUCTORS(MP4Source);
-};
-
-}  // namespace android
-
-#endif // MP4_SOURCE_H
diff --git a/media/libnbaio/AudioBufferProviderSource.cpp b/media/libnbaio/AudioBufferProviderSource.cpp
index 74a6fdb..4a69104 100644
--- a/media/libnbaio/AudioBufferProviderSource.cpp
+++ b/media/libnbaio/AudioBufferProviderSource.cpp
@@ -24,11 +24,11 @@
 namespace android {
 
 AudioBufferProviderSource::AudioBufferProviderSource(AudioBufferProvider *provider,
-                                                     NBAIO_Format format) :
+                                                     const NBAIO_Format& format) :
     NBAIO_Source(format), mProvider(provider), mConsumed(0)
 {
     ALOG_ASSERT(provider != NULL);
-    ALOG_ASSERT(format != Format_Invalid);
+    ALOG_ASSERT(Format_isValid(format));
 }
 
 AudioBufferProviderSource::~AudioBufferProviderSource()
diff --git a/media/libnbaio/AudioStreamInSource.cpp b/media/libnbaio/AudioStreamInSource.cpp
index 05273f6..ae8fac8 100644
--- a/media/libnbaio/AudioStreamInSource.cpp
+++ b/media/libnbaio/AudioStreamInSource.cpp
@@ -40,7 +40,7 @@
 ssize_t AudioStreamInSource::negotiate(const NBAIO_Format offers[], size_t numOffers,
                                       NBAIO_Format counterOffers[], size_t& numCounterOffers)
 {
-    if (mFormat == Format_Invalid) {
+    if (!Format_isValid(mFormat)) {
         mStreamBufferSizeBytes = mStream->common.get_buffer_size(&mStream->common);
         audio_format_t streamFormat = mStream->common.get_format(&mStream->common);
         if (streamFormat == AUDIO_FORMAT_PCM_16_BIT) {
@@ -67,7 +67,7 @@
 
 ssize_t AudioStreamInSource::read(void *buffer, size_t count)
 {
-    if (CC_UNLIKELY(mFormat == Format_Invalid)) {
+    if (CC_UNLIKELY(!Format_isValid(mFormat))) {
         return NEGOTIATE;
     }
     ssize_t bytesRead = mStream->read(mStream, buffer, count << mBitShift);
diff --git a/media/libnbaio/AudioStreamOutSink.cpp b/media/libnbaio/AudioStreamOutSink.cpp
index e4341d7..aa9810e 100644
--- a/media/libnbaio/AudioStreamOutSink.cpp
+++ b/media/libnbaio/AudioStreamOutSink.cpp
@@ -37,7 +37,7 @@
 ssize_t AudioStreamOutSink::negotiate(const NBAIO_Format offers[], size_t numOffers,
                                       NBAIO_Format counterOffers[], size_t& numCounterOffers)
 {
-    if (mFormat == Format_Invalid) {
+    if (!Format_isValid(mFormat)) {
         mStreamBufferSizeBytes = mStream->common.get_buffer_size(&mStream->common);
         audio_format_t streamFormat = mStream->common.get_format(&mStream->common);
         if (streamFormat == AUDIO_FORMAT_PCM_16_BIT) {
@@ -56,7 +56,7 @@
     if (!mNegotiated) {
         return NEGOTIATE;
     }
-    ALOG_ASSERT(mFormat != Format_Invalid);
+    ALOG_ASSERT(Format_isValid(mFormat));
     ssize_t ret = mStream->write(mStream, buffer, count << mBitShift);
     if (ret > 0) {
         ret >>= mBitShift;
diff --git a/media/libnbaio/MonoPipe.cpp b/media/libnbaio/MonoPipe.cpp
index de0ad28..b23967b 100644
--- a/media/libnbaio/MonoPipe.cpp
+++ b/media/libnbaio/MonoPipe.cpp
@@ -30,7 +30,7 @@
 
 namespace android {
 
-MonoPipe::MonoPipe(size_t reqFrames, NBAIO_Format format, bool writeCanBlock) :
+MonoPipe::MonoPipe(size_t reqFrames, const NBAIO_Format& format, bool writeCanBlock) :
         NBAIO_Sink(format),
         mUpdateSeq(0),
         mReqFrames(reqFrames),
@@ -183,7 +183,7 @@
             }
         }
         if (ns > 0) {
-            const struct timespec req = {0, ns};
+            const struct timespec req = {0, static_cast<long>(ns)};
             nanosleep(&req, NULL);
         }
         // record the time that this write() completed
diff --git a/media/libnbaio/NBAIO.cpp b/media/libnbaio/NBAIO.cpp
index e0d2c21..51514de 100644
--- a/media/libnbaio/NBAIO.cpp
+++ b/media/libnbaio/NBAIO.cpp
@@ -22,17 +22,22 @@
 
 namespace android {
 
-size_t Format_frameSize(NBAIO_Format format)
+size_t Format_frameSize(const NBAIO_Format& format)
 {
+    // FIXME The sample format is hard-coded to AUDIO_FORMAT_PCM_16_BIT
     return Format_channelCount(format) * sizeof(short);
 }
 
-size_t Format_frameBitShift(NBAIO_Format format)
+int Format_frameBitShift(const NBAIO_Format& format)
 {
+    // FIXME The sample format is hard-coded to AUDIO_FORMAT_PCM_16_BIT
     // sizeof(short) == 2, so frame size == 1 << channels
     return Format_channelCount(format);
+    // FIXME must return -1 for non-power of 2
 }
 
+const NBAIO_Format Format_Invalid = { 0 };
+
 enum {
     Format_SR_8000,
     Format_SR_11025,
@@ -51,12 +56,12 @@
     Format_C_Mask = 0x18
 };
 
-unsigned Format_sampleRate(NBAIO_Format format)
+unsigned Format_sampleRate(const NBAIO_Format& format)
 {
-    if (format == Format_Invalid) {
+    if (!Format_isValid(format)) {
         return 0;
     }
-    switch (format & Format_SR_Mask) {
+    switch (format.mPacked & Format_SR_Mask) {
     case Format_SR_8000:
         return 8000;
     case Format_SR_11025:
@@ -78,12 +83,12 @@
     }
 }
 
-unsigned Format_channelCount(NBAIO_Format format)
+unsigned Format_channelCount(const NBAIO_Format& format)
 {
-    if (format == Format_Invalid) {
+    if (!Format_isValid(format)) {
         return 0;
     }
-    switch (format & Format_C_Mask) {
+    switch (format.mPacked & Format_C_Mask) {
     case Format_C_1:
         return 1;
     case Format_C_2:
@@ -95,7 +100,7 @@
 
 NBAIO_Format Format_from_SR_C(unsigned sampleRate, unsigned channelCount)
 {
-    NBAIO_Format format;
+    unsigned format;
     switch (sampleRate) {
     case 8000:
         format = Format_SR_8000;
@@ -134,7 +139,9 @@
     default:
         return Format_Invalid;
     }
-    return format;
+    NBAIO_Format ret;
+    ret.mPacked = format;
+    return ret;
 }
 
 // This is a default implementation; it is expected that subclasses will optimize this.
@@ -216,9 +223,9 @@
 {
     ALOGV("negotiate offers=%p numOffers=%u countersOffers=%p numCounterOffers=%u",
             offers, numOffers, counterOffers, numCounterOffers);
-    if (mFormat != Format_Invalid) {
+    if (Format_isValid(mFormat)) {
         for (size_t i = 0; i < numOffers; ++i) {
-            if (offers[i] == mFormat) {
+            if (Format_isEqual(offers[i], mFormat)) {
                 mNegotiated = true;
                 return i;
             }
@@ -233,4 +240,14 @@
     return (ssize_t) NEGOTIATE;
 }
 
+bool Format_isValid(const NBAIO_Format& format)
+{
+    return format.mPacked != Format_Invalid.mPacked;
+}
+
+bool Format_isEqual(const NBAIO_Format& format1, const NBAIO_Format& format2)
+{
+    return format1.mPacked == format2.mPacked;
+}
+
 }   // namespace android
diff --git a/media/libnbaio/NBLog.cpp b/media/libnbaio/NBLog.cpp
index ba8d0b4..190824d 100644
--- a/media/libnbaio/NBLog.cpp
+++ b/media/libnbaio/NBLog.cpp
@@ -340,7 +340,7 @@
     if (i > 0) {
         lost += i;
         if (fd >= 0) {
-            fdprintf(fd, "%*swarning: lost %u bytes worth of events\n", indent, "", lost);
+            fdprintf(fd, "%*swarning: lost %zu bytes worth of events\n", indent, "", lost);
         } else {
             ALOGI("%*swarning: lost %u bytes worth of events\n", indent, "", lost);
         }
diff --git a/media/libnbaio/Pipe.cpp b/media/libnbaio/Pipe.cpp
index 1c21f9c..115f311 100644
--- a/media/libnbaio/Pipe.cpp
+++ b/media/libnbaio/Pipe.cpp
@@ -25,7 +25,7 @@
 
 namespace android {
 
-Pipe::Pipe(size_t maxFrames, NBAIO_Format format) :
+Pipe::Pipe(size_t maxFrames, const NBAIO_Format& format) :
         NBAIO_Sink(format),
         mMaxFrames(roundup(maxFrames)),
         mBuffer(malloc(mMaxFrames * Format_frameSize(format))),
diff --git a/media/libnbaio/PipeReader.cpp b/media/libnbaio/PipeReader.cpp
index d786b84..24da1bd 100644
--- a/media/libnbaio/PipeReader.cpp
+++ b/media/libnbaio/PipeReader.cpp
@@ -59,7 +59,7 @@
     return avail;
 }
 
-ssize_t PipeReader::read(void *buffer, size_t count, int64_t readPTS)
+ssize_t PipeReader::read(void *buffer, size_t count, int64_t readPTS __unused)
 {
     ssize_t avail = availableToRead();
     if (CC_UNLIKELY(avail <= 0)) {
diff --git a/media/libstagefright/AACExtractor.cpp b/media/libstagefright/AACExtractor.cpp
index 4d1072f..196f6ee 100644
--- a/media/libstagefright/AACExtractor.cpp
+++ b/media/libstagefright/AACExtractor.cpp
@@ -219,7 +219,7 @@
     return new AACSource(mDataSource, mMeta, mOffsetVector, mFrameDurationUs);
 }
 
-sp<MetaData> AACExtractor::getTrackMetaData(size_t index, uint32_t flags) {
+sp<MetaData> AACExtractor::getTrackMetaData(size_t index, uint32_t /* flags */) {
     if (mInitCheck != OK || index != 0) {
         return NULL;
     }
@@ -252,7 +252,7 @@
     }
 }
 
-status_t AACSource::start(MetaData *params) {
+status_t AACSource::start(MetaData * /* params */) {
     CHECK(!mStarted);
 
     if (mOffsetVector.empty()) {
diff --git a/media/libstagefright/AACWriter.cpp b/media/libstagefright/AACWriter.cpp
index a6f7cfb..deee8e7 100644
--- a/media/libstagefright/AACWriter.cpp
+++ b/media/libstagefright/AACWriter.cpp
@@ -111,7 +111,7 @@
     return OK;
 }
 
-status_t AACWriter::start(MetaData *params) {
+status_t AACWriter::start(MetaData * /* params */) {
     if (mInitCheck != OK) {
         return mInitCheck;
     }
@@ -171,7 +171,7 @@
     void *dummy;
     pthread_join(mThread, &dummy);
 
-    status_t err = (status_t) dummy;
+    status_t err = static_cast<status_t>(reinterpret_cast<uintptr_t>(dummy));
     {
         status_t status = mSource->stop();
         if (err == OK &&
@@ -200,7 +200,7 @@
 
 // static
 void *AACWriter::ThreadWrapper(void *me) {
-    return (void *) static_cast<AACWriter *>(me)->threadFunc();
+    return (void *)(uintptr_t)static_cast<AACWriter *>(me)->threadFunc();
 }
 
 /*
diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index eb274a8..ac78d6c 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -365,6 +365,7 @@
       mIsEncoder(false),
       mUseMetadataOnEncoderOutput(false),
       mShutdownInProgress(false),
+      mIsConfiguredForAdaptivePlayback(false),
       mEncoderDelay(0),
       mEncoderPadding(0),
       mChannelMaskPresent(false),
@@ -372,7 +373,9 @@
       mDequeueCounter(0),
       mStoreMetaDataInOutputBuffers(false),
       mMetaDataBuffersToSubmit(0),
-      mRepeatFrameDelayUs(-1ll) {
+      mRepeatFrameDelayUs(-1ll),
+      mMaxPtsGapUs(-1l),
+      mCreateInputBuffersSuspended(false) {
     mUninitializedState = new UninitializedState(this);
     mLoadedState = new LoadedState(this);
     mLoadedToIdleState = new LoadedToIdleState(this);
@@ -1114,6 +1117,16 @@
                     &mRepeatFrameDelayUs)) {
             mRepeatFrameDelayUs = -1ll;
         }
+
+        if (!msg->findInt64("max-pts-gap-to-encoder", &mMaxPtsGapUs)) {
+            mMaxPtsGapUs = -1l;
+        }
+
+        if (!msg->findInt32(
+                    "create-input-buffers-suspended",
+                    (int32_t*)&mCreateInputBuffersSuspended)) {
+            mCreateInputBuffersSuspended = false;
+        }
     }
 
     // Always try to enable dynamic output buffers on native surface
@@ -1121,6 +1134,7 @@
     int32_t haveNativeWindow = msg->findObject("native-window", &obj) &&
             obj != NULL;
     mStoreMetaDataInOutputBuffers = false;
+    mIsConfiguredForAdaptivePlayback = false;
     if (!encoder && video && haveNativeWindow) {
         err = mOMX->storeMetaDataInBuffers(mNode, kPortIndexOutput, OMX_TRUE);
         if (err != OK) {
@@ -1157,7 +1171,7 @@
             if (canDoAdaptivePlayback &&
                 msg->findInt32("max-width", &maxWidth) &&
                 msg->findInt32("max-height", &maxHeight)) {
-                ALOGV("[%s] prepareForAdaptivePlayback(%ldx%ld)",
+                ALOGV("[%s] prepareForAdaptivePlayback(%dx%d)",
                       mComponentName.c_str(), maxWidth, maxHeight);
 
                 err = mOMX->prepareForAdaptivePlayback(
@@ -1165,12 +1179,14 @@
                 ALOGW_IF(err != OK,
                         "[%s] prepareForAdaptivePlayback failed w/ err %d",
                         mComponentName.c_str(), err);
+                mIsConfiguredForAdaptivePlayback = (err == OK);
             }
             // allow failure
             err = OK;
         } else {
             ALOGV("[%s] storeMetaDataInBuffers succeeded", mComponentName.c_str());
             mStoreMetaDataInOutputBuffers = true;
+            mIsConfiguredForAdaptivePlayback = true;
         }
 
         int32_t push;
@@ -2951,7 +2967,8 @@
       mCodec(codec) {
 }
 
-ACodec::BaseState::PortMode ACodec::BaseState::getPortMode(OMX_U32 portIndex) {
+ACodec::BaseState::PortMode ACodec::BaseState::getPortMode(
+        OMX_U32 /* portIndex */) {
     return KEEP_BUFFERS;
 }
 
@@ -3318,11 +3335,11 @@
                 mCodec->mInputEOSResult = err;
             }
             break;
-
-            default:
-                CHECK_EQ((int)mode, (int)FREE_BUFFERS);
-                break;
         }
+
+        default:
+            CHECK_EQ((int)mode, (int)FREE_BUFFERS);
+            break;
     }
 }
 
@@ -3360,8 +3377,8 @@
         size_t rangeOffset, size_t rangeLength,
         OMX_U32 flags,
         int64_t timeUs,
-        void *platformPrivate,
-        void *dataPtr) {
+        void * /* platformPrivate */,
+        void * /* dataPtr */) {
     ALOGV("[%s] onOMXFillBufferDone %p time %lld us, flags = 0x%08lx",
          mCodec->mComponentName.c_str(), bufferID, timeUs, flags);
 
@@ -3498,7 +3515,7 @@
     int32_t render;
     if (mCodec->mNativeWindow != NULL
             && msg->findInt32("render", &render) && render != 0
-            && (info->mData == NULL || info->mData->size() != 0)) {
+            && info->mData != NULL && info->mData->size() != 0) {
         // The client wants this buffer to be rendered.
 
         status_t err;
@@ -3773,6 +3790,7 @@
     mCodec->mDequeueCounter = 0;
     mCodec->mMetaDataBuffersToSubmit = 0;
     mCodec->mRepeatFrameDelayUs = -1ll;
+    mCodec->mIsConfiguredForAdaptivePlayback = false;
 
     if (mCodec->mShutdownInProgress) {
         bool keepComponentAllocated = mCodec->mKeepComponentAllocated;
@@ -3893,7 +3911,7 @@
 }
 
 void ACodec::LoadedState::onCreateInputSurface(
-        const sp<AMessage> &msg) {
+        const sp<AMessage> & /* msg */) {
     ALOGV("onCreateInputSurface");
 
     sp<AMessage> notify = mCodec->mNotify->dup();
@@ -3921,6 +3939,37 @@
         }
     }
 
+    if (err == OK && mCodec->mMaxPtsGapUs > 0l) {
+        err = mCodec->mOMX->setInternalOption(
+                mCodec->mNode,
+                kPortIndexInput,
+                IOMX::INTERNAL_OPTION_MAX_TIMESTAMP_GAP,
+                &mCodec->mMaxPtsGapUs,
+                sizeof(mCodec->mMaxPtsGapUs));
+
+        if (err != OK) {
+            ALOGE("[%s] Unable to configure max timestamp gap (err %d)",
+                    mCodec->mComponentName.c_str(),
+                    err);
+          }
+      }
+
+    if (err == OK && mCodec->mCreateInputBuffersSuspended) {
+        bool suspend = true;
+        err = mCodec->mOMX->setInternalOption(
+                mCodec->mNode,
+                kPortIndexInput,
+                IOMX::INTERNAL_OPTION_SUSPEND,
+                &suspend,
+                sizeof(suspend));
+
+        if (err != OK) {
+            ALOGE("[%s] Unable to configure option to suspend (err %d)",
+                  mCodec->mComponentName.c_str(),
+                  err);
+        }
+    }
+
     if (err == OK) {
         notify->setObject("input-surface",
                 new BufferProducerWrapper(bufferProducer));
@@ -3978,6 +4027,7 @@
 
 bool ACodec::LoadedToIdleState::onMessageReceived(const sp<AMessage> &msg) {
     switch (msg->what()) {
+        case kWhatSetParameters:
         case kWhatShutdown:
         {
             mCodec->deferMessage(msg);
@@ -4044,6 +4094,7 @@
 
 bool ACodec::IdleToExecutingState::onMessageReceived(const sp<AMessage> &msg) {
     switch (msg->what()) {
+        case kWhatSetParameters:
         case kWhatShutdown:
         {
             mCodec->deferMessage(msg);
@@ -4104,7 +4155,7 @@
 }
 
 ACodec::BaseState::PortMode ACodec::ExecutingState::getPortMode(
-        OMX_U32 portIndex) {
+        OMX_U32 /* portIndex */) {
     return RESUBMIT_BUFFERS;
 }
 
@@ -4313,6 +4364,22 @@
         }
     }
 
+    int64_t skipFramesBeforeUs;
+    if (params->findInt64("skip-frames-before", &skipFramesBeforeUs)) {
+        status_t err =
+            mOMX->setInternalOption(
+                     mNode,
+                     kPortIndexInput,
+                     IOMX::INTERNAL_OPTION_START_TIME,
+                     &skipFramesBeforeUs,
+                     sizeof(skipFramesBeforeUs));
+
+        if (err != OK) {
+            ALOGE("Failed to set parameter 'skip-frames-before' (err %d)", err);
+            return err;
+        }
+    }
+
     int32_t dropInputFrames;
     if (params->findInt32("drop-input-frames", &dropInputFrames)) {
         bool suspend = dropInputFrames != 0;
diff --git a/media/libstagefright/AMRExtractor.cpp b/media/libstagefright/AMRExtractor.cpp
index 03dcbf9..3f592ed 100644
--- a/media/libstagefright/AMRExtractor.cpp
+++ b/media/libstagefright/AMRExtractor.cpp
@@ -189,7 +189,7 @@
             mOffsetTable, mOffsetTableLength);
 }
 
-sp<MetaData> AMRExtractor::getTrackMetaData(size_t index, uint32_t flags) {
+sp<MetaData> AMRExtractor::getTrackMetaData(size_t index, uint32_t /* flags */) {
     if (mInitCheck != OK || index != 0) {
         return NULL;
     }
@@ -221,7 +221,7 @@
     }
 }
 
-status_t AMRSource::start(MetaData *params) {
+status_t AMRSource::start(MetaData * /* params */) {
     CHECK(!mStarted);
 
     mOffset = mIsWide ? 9 : 6;
@@ -258,7 +258,7 @@
         int64_t seekFrame = seekTimeUs / 20000ll;  // 20ms per frame.
         mCurrentTimeUs = seekFrame * 20000ll;
 
-        int index = seekFrame / 50;
+        size_t index = seekFrame < 0 ? 0 : seekFrame / 50;
         if (index >= mOffsetTableLength) {
             index = mOffsetTableLength - 1;
         }
diff --git a/media/libstagefright/AMRWriter.cpp b/media/libstagefright/AMRWriter.cpp
index 8d5eec8..653ca36 100644
--- a/media/libstagefright/AMRWriter.cpp
+++ b/media/libstagefright/AMRWriter.cpp
@@ -105,7 +105,7 @@
     return OK;
 }
 
-status_t AMRWriter::start(MetaData *params) {
+status_t AMRWriter::start(MetaData * /* params */) {
     if (mInitCheck != OK) {
         return mInitCheck;
     }
@@ -162,7 +162,7 @@
     void *dummy;
     pthread_join(mThread, &dummy);
 
-    status_t err = (status_t) dummy;
+    status_t err = static_cast<status_t>(reinterpret_cast<uintptr_t>(dummy));
     {
         status_t status = mSource->stop();
         if (err == OK &&
@@ -191,7 +191,7 @@
 
 // static
 void *AMRWriter::ThreadWrapper(void *me) {
-    return (void *) static_cast<AMRWriter *>(me)->threadFunc();
+    return (void *)(uintptr_t) static_cast<AMRWriter *>(me)->threadFunc();
 }
 
 status_t AMRWriter::threadFunc() {
diff --git a/media/libstagefright/Android.mk b/media/libstagefright/Android.mk
index 6a2a696..0636dcc 100644
--- a/media/libstagefright/Android.mk
+++ b/media/libstagefright/Android.mk
@@ -15,6 +15,7 @@
         CameraSource.cpp                  \
         CameraSourceTimeLapse.cpp         \
         DataSource.cpp                    \
+        DataURISource.cpp                 \
         DRMExtractor.cpp                  \
         ESDS.cpp                          \
         FileSource.cpp                    \
@@ -30,8 +31,10 @@
         MediaBufferGroup.cpp              \
         MediaCodec.cpp                    \
         MediaCodecList.cpp                \
+        MediaCodecSource.cpp              \
         MediaDefs.cpp                     \
         MediaExtractor.cpp                \
+        http/MediaHTTP.cpp                \
         MediaMuxer.cpp                    \
         MediaSource.cpp                   \
         MetaData.cpp                      \
@@ -55,8 +58,6 @@
         WVMExtractor.cpp                  \
         XINGSeeker.cpp                    \
         avc_utils.cpp                     \
-        mp4/FragmentedMP4Parser.cpp       \
-        mp4/TrackFragment.cpp             \
 
 LOCAL_C_INCLUDES:= \
         $(TOP)/frameworks/av/include/media/stagefright/timedtext \
@@ -103,13 +104,6 @@
         libFLAC \
         libmedia_helper
 
-LOCAL_SRC_FILES += \
-        chromium_http_stub.cpp
-LOCAL_CPPFLAGS += -DCHROMIUM_AVAILABLE=1
-
-LOCAL_SHARED_LIBRARIES += libstlport
-include external/stlport/libstlport.mk
-
 LOCAL_SHARED_LIBRARIES += \
         libstagefright_enc_common \
         libstagefright_avc_common \
diff --git a/media/libstagefright/AudioPlayer.cpp b/media/libstagefright/AudioPlayer.cpp
index 05ee34e..8623100 100644
--- a/media/libstagefright/AudioPlayer.cpp
+++ b/media/libstagefright/AudioPlayer.cpp
@@ -410,7 +410,7 @@
 
 // static
 size_t AudioPlayer::AudioSinkCallback(
-        MediaPlayerBase::AudioSink *audioSink,
+        MediaPlayerBase::AudioSink * /* audioSink */,
         void *buffer, size_t size, void *cookie,
         MediaPlayerBase::AudioSink::cb_event_t event) {
     AudioPlayer *me = (AudioPlayer *)cookie;
diff --git a/media/libstagefright/AudioSource.cpp b/media/libstagefright/AudioSource.cpp
index d7223d9..df7da0a 100644
--- a/media/libstagefright/AudioSource.cpp
+++ b/media/libstagefright/AudioSource.cpp
@@ -68,7 +68,7 @@
         int frameCount = kMaxBufferSize / sizeof(int16_t) / channelCount;
 
         // make sure that the AudioRecord total buffer size is large enough
-        int bufCount = 2;
+        size_t bufCount = 2;
         while ((bufCount * frameCount) < minFrameCount) {
             bufCount++;
         }
@@ -208,7 +208,7 @@
 }
 
 status_t AudioSource::read(
-        MediaBuffer **out, const ReadOptions *options) {
+        MediaBuffer **out, const ReadOptions * /* options */) {
     Mutex::Autolock autoLock(mLock);
     *out = NULL;
 
@@ -278,7 +278,7 @@
 
     // Drop retrieved and previously lost audio data.
     if (mNumFramesReceived == 0 && timeUs < mStartTimeUs) {
-        mRecord->getInputFramesLost();
+        (void) mRecord->getInputFramesLost();
         ALOGV("Drop audio data at %lld/%lld us", timeUs, mStartTimeUs);
         return OK;
     }
@@ -308,7 +308,7 @@
     if (numLostBytes > 0) {
         // Loss of audio frames should happen rarely; thus the LOGW should
         // not cause a logging spam
-        ALOGW("Lost audio record data: %d bytes", numLostBytes);
+        ALOGW("Lost audio record data: %zu bytes", numLostBytes);
     }
 
     while (numLostBytes > 0) {
diff --git a/media/libstagefright/AwesomePlayer.cpp b/media/libstagefright/AwesomePlayer.cpp
index 130207d..e83ec62 100644
--- a/media/libstagefright/AwesomePlayer.cpp
+++ b/media/libstagefright/AwesomePlayer.cpp
@@ -19,6 +19,7 @@
 //#define LOG_NDEBUG 0
 #define LOG_TAG "AwesomePlayer"
 #define ATRACE_TAG ATRACE_TAG_VIDEO
+#include <inttypes.h>
 #include <utils/Log.h>
 #include <utils/Trace.h>
 
@@ -34,6 +35,8 @@
 
 #include <binder/IPCThreadState.h>
 #include <binder/IServiceManager.h>
+#include <media/IMediaHTTPConnection.h>
+#include <media/IMediaHTTPService.h>
 #include <media/IMediaPlayerService.h>
 #include <media/stagefright/foundation/hexdump.h>
 #include <media/stagefright/foundation/ADebug.h>
@@ -44,6 +47,7 @@
 #include <media/stagefright/MediaBuffer.h>
 #include <media/stagefright/MediaDefs.h>
 #include <media/stagefright/MediaExtractor.h>
+#include <media/stagefright/MediaHTTP.h>
 #include <media/stagefright/MediaSource.h>
 #include <media/stagefright/MetaData.h>
 #include <media/stagefright/OMXCodec.h>
@@ -82,7 +86,7 @@
 protected:
     virtual ~AwesomeEvent() {}
 
-    virtual void fire(TimedEventQueue *queue, int64_t /* now_us */) {
+    virtual void fire(TimedEventQueue * /* queue */, int64_t /* now_us */) {
         (mPlayer->*mMethod)();
     }
 
@@ -276,15 +280,20 @@
 }
 
 status_t AwesomePlayer::setDataSource(
-        const char *uri, const KeyedVector<String8, String8> *headers) {
+        const sp<IMediaHTTPService> &httpService,
+        const char *uri,
+        const KeyedVector<String8, String8> *headers) {
     Mutex::Autolock autoLock(mLock);
-    return setDataSource_l(uri, headers);
+    return setDataSource_l(httpService, uri, headers);
 }
 
 status_t AwesomePlayer::setDataSource_l(
-        const char *uri, const KeyedVector<String8, String8> *headers) {
+        const sp<IMediaHTTPService> &httpService,
+        const char *uri,
+        const KeyedVector<String8, String8> *headers) {
     reset_l();
 
+    mHTTPService = httpService;
     mUri = uri;
 
     if (headers) {
@@ -581,6 +590,7 @@
     mSeekNotificationSent = true;
     mSeekTimeUs = 0;
 
+    mHTTPService.clear();
     mUri.setTo("");
     mUriHeaders.clear();
 
@@ -725,7 +735,7 @@
 
                 if ((mFlags & PLAYING) && !eos
                         && (cachedDataRemaining < kLowWaterMarkBytes)) {
-                    ALOGI("cache is running low (< %d) , pausing.",
+                    ALOGI("cache is running low (< %zu) , pausing.",
                          kLowWaterMarkBytes);
                     modifyFlags(CACHE_UNDERRUN, SET);
                     pause_l();
@@ -734,12 +744,12 @@
                     notifyListener_l(MEDIA_INFO, MEDIA_INFO_BUFFERING_START);
                 } else if (eos || cachedDataRemaining > kHighWaterMarkBytes) {
                     if (mFlags & CACHE_UNDERRUN) {
-                        ALOGI("cache has filled up (> %d), resuming.",
+                        ALOGI("cache has filled up (> %zu), resuming.",
                              kHighWaterMarkBytes);
                         modifyFlags(CACHE_UNDERRUN, CLEAR);
                         play_l();
                     } else if (mFlags & PREPARING) {
-                        ALOGV("cache has filled up (> %d), prepare is done",
+                        ALOGV("cache has filled up (> %zu), prepare is done",
                              kHighWaterMarkBytes);
                         finishAsyncPrepare_l();
                     }
@@ -1482,7 +1492,7 @@
     CHECK(source != NULL);
 
     if (mTextDriver == NULL) {
-        mTextDriver = new TimedTextDriver(mListener);
+        mTextDriver = new TimedTextDriver(mListener, mHTTPService);
     }
 
     mTextDriver->addInBandTextSource(trackIndex, source);
@@ -2096,7 +2106,10 @@
             mSeekNotificationSent = true;
         }
 
-        mSeeking = NO_SEEK;
+        if (mVideoSource == NULL) {
+            // For video the mSeeking flag is always reset in finishSeekIfNecessary
+            mSeeking = NO_SEEK;
+        }
 
         notifyIfMediaStarted_l();
     }
@@ -2189,15 +2202,14 @@
     if (!strncasecmp("http://", mUri.string(), 7)
             || !strncasecmp("https://", mUri.string(), 8)
             || isWidevineStreaming) {
-        mConnectingDataSource = HTTPBase::Create(
-                (mFlags & INCOGNITO)
-                    ? HTTPBase::kFlagIncognito
-                    : 0);
-
-        if (mUIDValid) {
-            mConnectingDataSource->setUID(mUID);
+        if (mHTTPService == NULL) {
+            ALOGE("Attempt to play media from http URI without HTTP service.");
+            return UNKNOWN_ERROR;
         }
 
+        sp<IMediaHTTPConnection> conn = mHTTPService->makeHTTPConnection();
+        mConnectingDataSource = new MediaHTTP(conn);
+
         String8 cacheConfig;
         bool disconnectAtHighwatermark;
         NuCachedSource2::RemoveCacheSpecificHeaders(
@@ -2292,8 +2304,8 @@
                         sniffedMIME = tmp.string();
 
                         if (meta == NULL
-                                || !meta->findInt64(
-                                    "meta-data-size", &metaDataSize)) {
+                                || !meta->findInt64("meta-data-size",
+                                     reinterpret_cast<int64_t*>(&metaDataSize))) {
                             metaDataSize = kHighWaterMarkBytes;
                         }
 
@@ -2313,7 +2325,8 @@
             }
         }
     } else {
-        dataSource = DataSource::CreateFromURI(mUri.string(), &mUriHeaders);
+        dataSource = DataSource::CreateFromURI(
+                mHTTPService, mUri.string(), &mUriHeaders);
     }
 
     if (dataSource == NULL) {
@@ -2580,12 +2593,12 @@
 status_t AwesomePlayer::selectAudioTrack_l(
         const sp<MediaSource>& source, size_t trackIndex) {
 
-    ALOGI("selectAudioTrack_l: trackIndex=%d, mFlags=0x%x", trackIndex, mFlags);
+    ALOGI("selectAudioTrack_l: trackIndex=%zu, mFlags=0x%x", trackIndex, mFlags);
 
     {
         Mutex::Autolock autoLock(mStatsLock);
         if ((ssize_t)trackIndex == mActiveAudioTrackIndex) {
-            ALOGI("Track %d is active. Does nothing.", trackIndex);
+            ALOGI("Track %zu is active. Does nothing.", trackIndex);
             return OK;
         }
         //mStats.mFlags = mFlags;
@@ -2658,7 +2671,7 @@
         trackCount += mTextDriver->countExternalTracks();
     }
     if (trackIndex >= trackCount) {
-        ALOGE("Track index (%d) is out of range [0, %d)", trackIndex, trackCount);
+        ALOGE("Track index (%zu) is out of range [0, %zu)", trackIndex, trackCount);
         return ERROR_OUT_OF_RANGE;
     }
 
@@ -2670,14 +2683,14 @@
         isAudioTrack = !strncasecmp(mime, "audio/", 6);
 
         if (!isAudioTrack && strcasecmp(mime, MEDIA_MIMETYPE_TEXT_3GPP) != 0) {
-            ALOGE("Track %d is not either audio or timed text", trackIndex);
+            ALOGE("Track %zu is not either audio or timed text", trackIndex);
             return ERROR_UNSUPPORTED;
         }
     }
 
     if (isAudioTrack) {
         if (!select) {
-            ALOGE("Deselect an audio track (%d) is not supported", trackIndex);
+            ALOGE("Deselect an audio track (%zu) is not supported", trackIndex);
             return ERROR_UNSUPPORTED;
         }
         return selectAudioTrack_l(mExtractor->getTrack(trackIndex), trackIndex);
@@ -2755,7 +2768,7 @@
         {
             Mutex::Autolock autoLock(mLock);
             if (mTextDriver == NULL) {
-                mTextDriver = new TimedTextDriver(mListener);
+                mTextDriver = new TimedTextDriver(mListener, mHTTPService);
             }
             // String values written in Parcel are UTF-16 values.
             String8 uri(request.readString16());
@@ -2767,7 +2780,7 @@
         {
             Mutex::Autolock autoLock(mLock);
             if (mTextDriver == NULL) {
-                mTextDriver = new TimedTextDriver(mListener);
+                mTextDriver = new TimedTextDriver(mListener, mHTTPService);
             }
             int fd         = request.readFileDescriptor();
             off64_t offset = request.readInt64();
@@ -2800,7 +2813,8 @@
     return mCachedSource != NULL || mWVMExtractor != NULL;
 }
 
-status_t AwesomePlayer::dump(int fd, const Vector<String16> &args) const {
+status_t AwesomePlayer::dump(
+        int fd, const Vector<String16> & /* args */) const {
     Mutex::Autolock autoLock(mStatsLock);
 
     FILE *out = fdopen(dup(fd), "w");
@@ -2815,7 +2829,7 @@
     fprintf(out, ", flags(0x%08x)", mStats.mFlags);
 
     if (mStats.mBitrate >= 0) {
-        fprintf(out, ", bitrate(%lld bps)", mStats.mBitrate);
+        fprintf(out, ", bitrate(%" PRId64 " bps)", mStats.mBitrate);
     }
 
     fprintf(out, "\n");
@@ -2823,7 +2837,7 @@
     for (size_t i = 0; i < mStats.mTracks.size(); ++i) {
         const TrackStat &stat = mStats.mTracks.itemAt(i);
 
-        fprintf(out, "  Track %d\n", i + 1);
+        fprintf(out, "  Track %zu\n", i + 1);
         fprintf(out, "   MIME(%s)", stat.mMIME.string());
 
         if (!stat.mDecoderName.isEmpty()) {
@@ -2835,8 +2849,8 @@
         if ((ssize_t)i == mStats.mVideoTrackIndex) {
             fprintf(out,
                     "   videoDimensions(%d x %d), "
-                    "numVideoFramesDecoded(%lld), "
-                    "numVideoFramesDropped(%lld)\n",
+                    "numVideoFramesDecoded(%" PRId64 "), "
+                    "numVideoFramesDropped(%" PRId64 ")\n",
                     mStats.mVideoWidth,
                     mStats.mVideoHeight,
                     mStats.mNumVideoFramesDecoded,
@@ -2896,6 +2910,8 @@
     // get current position so we can start recreated stream from here
     getPosition(&mAudioTearDownPosition);
 
+    sp<IMediaHTTPService> savedHTTPService = mHTTPService;
+
     // Reset and recreate
     reset_l();
 
@@ -2905,7 +2921,7 @@
         mFileSource = fileSource;
         err = setDataSource_l(fileSource);
     } else {
-        err = setDataSource_l(uri, &uriHeaders);
+        err = setDataSource_l(savedHTTPService, uri, &uriHeaders);
     }
 
     mFlags |= PREPARING;
diff --git a/media/libstagefright/CameraSource.cpp b/media/libstagefright/CameraSource.cpp
index 3017fe7..f3ff792 100644
--- a/media/libstagefright/CameraSource.cpp
+++ b/media/libstagefright/CameraSource.cpp
@@ -31,6 +31,12 @@
 #include <utils/String8.h>
 #include <cutils/properties.h>
 
+#if LOG_NDEBUG
+#define UNUSED_UNLESS_VERBOSE(x) (void)(x)
+#else
+#define UNUSED_UNLESS_VERBOSE(x)
+#endif
+
 namespace android {
 
 static const int64_t CAMERA_SOURCE_TIMEOUT_NS = 3000000000LL;
@@ -63,11 +69,14 @@
 }
 
 void CameraSourceListener::notify(int32_t msgType, int32_t ext1, int32_t ext2) {
+    UNUSED_UNLESS_VERBOSE(msgType);
+    UNUSED_UNLESS_VERBOSE(ext1);
+    UNUSED_UNLESS_VERBOSE(ext2);
     ALOGV("notify(%d, %d, %d)", msgType, ext1, ext2);
 }
 
 void CameraSourceListener::postData(int32_t msgType, const sp<IMemory> &dataPtr,
-                                    camera_frame_metadata_t *metadata) {
+                                    camera_frame_metadata_t * /* metadata */) {
     ALOGV("postData(%d, ptr:%p, size:%d)",
          msgType, dataPtr->pointer(), dataPtr->size());
 
diff --git a/media/libstagefright/CameraSourceTimeLapse.cpp b/media/libstagefright/CameraSourceTimeLapse.cpp
index 86844b8..60cdf66 100644
--- a/media/libstagefright/CameraSourceTimeLapse.cpp
+++ b/media/libstagefright/CameraSourceTimeLapse.cpp
@@ -232,7 +232,7 @@
     return newMemory;
 }
 
-bool CameraSourceTimeLapse::skipCurrentFrame(int64_t timestampUs) {
+bool CameraSourceTimeLapse::skipCurrentFrame(int64_t /* timestampUs */) {
     ALOGV("skipCurrentFrame");
     if (mSkipCurrentFrame) {
         mSkipCurrentFrame = false;
diff --git a/media/libstagefright/DataSource.cpp b/media/libstagefright/DataSource.cpp
index 97987e2..6e0f37a 100644
--- a/media/libstagefright/DataSource.cpp
+++ b/media/libstagefright/DataSource.cpp
@@ -16,10 +16,6 @@
 
 #include "include/AMRExtractor.h"
 
-#if CHROMIUM_AVAILABLE
-#include "include/chromium_http_stub.h"
-#endif
-
 #include "include/AACExtractor.h"
 #include "include/DRMExtractor.h"
 #include "include/FLACExtractor.h"
@@ -35,10 +31,14 @@
 
 #include "matroska/MatroskaExtractor.h"
 
+#include <media/IMediaHTTPConnection.h>
+#include <media/IMediaHTTPService.h>
 #include <media/stagefright/foundation/AMessage.h>
 #include <media/stagefright/DataSource.h>
+#include <media/stagefright/DataURISource.h>
 #include <media/stagefright/FileSource.h>
 #include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/MediaHTTP.h>
 #include <utils/String8.h>
 
 #include <cutils/properties.h>
@@ -180,7 +180,9 @@
 
 // static
 sp<DataSource> DataSource::CreateFromURI(
-        const char *uri, const KeyedVector<String8, String8> *headers) {
+        const sp<IMediaHTTPService> &httpService,
+        const char *uri,
+        const KeyedVector<String8, String8> *headers) {
     bool isWidevine = !strncasecmp("widevine://", uri, 11);
 
     sp<DataSource> source;
@@ -189,7 +191,7 @@
     } else if (!strncasecmp("http://", uri, 7)
             || !strncasecmp("https://", uri, 8)
             || isWidevine) {
-        sp<HTTPBase> httpSource = HTTPBase::Create();
+        sp<HTTPBase> httpSource = new MediaHTTP(httpService->makeHTTPConnection());
 
         String8 tmp;
         if (isWidevine) {
@@ -220,11 +222,8 @@
             // in the widevine:// case.
             source = httpSource;
         }
-
-# if CHROMIUM_AVAILABLE
     } else if (!strncasecmp("data:", uri, 5)) {
-        source = createDataUriSource(uri);
-#endif
+        source = DataURISource::Create(uri);
     } else {
         // Assume it's a filename.
         source = new FileSource(uri);
diff --git a/media/libstagefright/DataURISource.cpp b/media/libstagefright/DataURISource.cpp
new file mode 100644
index 0000000..377bc85
--- /dev/null
+++ b/media/libstagefright/DataURISource.cpp
@@ -0,0 +1,109 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <media/stagefright/DataURISource.h>
+
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/AString.h>
+#include <media/stagefright/foundation/base64.h>
+
+namespace android {
+
+// static
+sp<DataURISource> DataURISource::Create(const char *uri) {
+    if (strncasecmp("data:", uri, 5)) {
+        return NULL;
+    }
+
+    char *commaPos = strrchr(uri, ',');
+
+    if (commaPos == NULL) {
+        return NULL;
+    }
+
+    sp<ABuffer> buffer;
+
+    AString tmp(&uri[5], commaPos - &uri[5]);
+
+    if (tmp.endsWith(";base64")) {
+        AString encoded(commaPos + 1);
+
+        // Strip CR and LF...
+        for (size_t i = encoded.size(); i-- > 0;) {
+            if (encoded.c_str()[i] == '\r' || encoded.c_str()[i] == '\n') {
+                encoded.erase(i, 1);
+            }
+        }
+
+        buffer = decodeBase64(encoded);
+
+        if (buffer == NULL) {
+            ALOGE("Malformed base64 encoded content found.");
+            return NULL;
+        }
+    } else {
+#if 0
+        size_t dataLen = strlen(uri) - tmp.size() - 6;
+        buffer = new ABuffer(dataLen);
+        memcpy(buffer->data(), commaPos + 1, dataLen);
+
+        // unescape
+#else
+        // MediaPlayer doesn't care for this right now as we don't
+        // play any text-based media.
+        return NULL;
+#endif
+    }
+
+    // We don't really care about charset or mime type.
+
+    return new DataURISource(buffer);
+}
+
+DataURISource::DataURISource(const sp<ABuffer> &buffer)
+    : mBuffer(buffer) {
+}
+
+DataURISource::~DataURISource() {
+}
+
+status_t DataURISource::initCheck() const {
+    return OK;
+}
+
+ssize_t DataURISource::readAt(off64_t offset, void *data, size_t size) {
+    if (offset >= mBuffer->size()) {
+        return 0;
+    }
+
+    size_t copy = mBuffer->size() - offset;
+    if (copy > size) {
+        copy = size;
+    }
+
+    memcpy(data, mBuffer->data() + offset, copy);
+
+    return copy;
+}
+
+status_t DataURISource::getSize(off64_t *size) {
+    *size = mBuffer->size();
+
+    return OK;
+}
+
+}  // namespace android
+
diff --git a/media/libstagefright/FLACExtractor.cpp b/media/libstagefright/FLACExtractor.cpp
index 098fcf9..fa7251c 100644
--- a/media/libstagefright/FLACExtractor.cpp
+++ b/media/libstagefright/FLACExtractor.cpp
@@ -208,55 +208,55 @@
 // with the same parameter list, but discard redundant information.
 
 FLAC__StreamDecoderReadStatus FLACParser::read_callback(
-        const FLAC__StreamDecoder *decoder, FLAC__byte buffer[],
+        const FLAC__StreamDecoder * /* decoder */, FLAC__byte buffer[],
         size_t *bytes, void *client_data)
 {
     return ((FLACParser *) client_data)->readCallback(buffer, bytes);
 }
 
 FLAC__StreamDecoderSeekStatus FLACParser::seek_callback(
-        const FLAC__StreamDecoder *decoder,
+        const FLAC__StreamDecoder * /* decoder */,
         FLAC__uint64 absolute_byte_offset, void *client_data)
 {
     return ((FLACParser *) client_data)->seekCallback(absolute_byte_offset);
 }
 
 FLAC__StreamDecoderTellStatus FLACParser::tell_callback(
-        const FLAC__StreamDecoder *decoder,
+        const FLAC__StreamDecoder * /* decoder */,
         FLAC__uint64 *absolute_byte_offset, void *client_data)
 {
     return ((FLACParser *) client_data)->tellCallback(absolute_byte_offset);
 }
 
 FLAC__StreamDecoderLengthStatus FLACParser::length_callback(
-        const FLAC__StreamDecoder *decoder,
+        const FLAC__StreamDecoder * /* decoder */,
         FLAC__uint64 *stream_length, void *client_data)
 {
     return ((FLACParser *) client_data)->lengthCallback(stream_length);
 }
 
 FLAC__bool FLACParser::eof_callback(
-        const FLAC__StreamDecoder *decoder, void *client_data)
+        const FLAC__StreamDecoder * /* decoder */, void *client_data)
 {
     return ((FLACParser *) client_data)->eofCallback();
 }
 
 FLAC__StreamDecoderWriteStatus FLACParser::write_callback(
-        const FLAC__StreamDecoder *decoder, const FLAC__Frame *frame,
+        const FLAC__StreamDecoder * /* decoder */, const FLAC__Frame *frame,
         const FLAC__int32 * const buffer[], void *client_data)
 {
     return ((FLACParser *) client_data)->writeCallback(frame, buffer);
 }
 
 void FLACParser::metadata_callback(
-        const FLAC__StreamDecoder *decoder,
+        const FLAC__StreamDecoder * /* decoder */,
         const FLAC__StreamMetadata *metadata, void *client_data)
 {
     ((FLACParser *) client_data)->metadataCallback(metadata);
 }
 
 void FLACParser::error_callback(
-        const FLAC__StreamDecoder *decoder,
+        const FLAC__StreamDecoder * /* decoder */,
         FLAC__StreamDecoderErrorStatus status, void *client_data)
 {
     ((FLACParser *) client_data)->errorCallback(status);
@@ -380,15 +380,21 @@
 // Copy samples from FLAC native 32-bit non-interleaved to 16-bit interleaved.
 // These are candidates for optimization if needed.
 
-static void copyMono8(short *dst, const int *const *src, unsigned nSamples, unsigned nChannels)
-{
+static void copyMono8(
+        short *dst,
+        const int *const *src,
+        unsigned nSamples,
+        unsigned /* nChannels */) {
     for (unsigned i = 0; i < nSamples; ++i) {
         *dst++ = src[0][i] << 8;
     }
 }
 
-static void copyStereo8(short *dst, const int *const *src, unsigned nSamples, unsigned nChannels)
-{
+static void copyStereo8(
+        short *dst,
+        const int *const *src,
+        unsigned nSamples,
+        unsigned /* nChannels */) {
     for (unsigned i = 0; i < nSamples; ++i) {
         *dst++ = src[0][i] << 8;
         *dst++ = src[1][i] << 8;
@@ -404,15 +410,21 @@
     }
 }
 
-static void copyMono16(short *dst, const int *const *src, unsigned nSamples, unsigned nChannels)
-{
+static void copyMono16(
+        short *dst,
+        const int *const *src,
+        unsigned nSamples,
+        unsigned /* nChannels */) {
     for (unsigned i = 0; i < nSamples; ++i) {
         *dst++ = src[0][i];
     }
 }
 
-static void copyStereo16(short *dst, const int *const *src, unsigned nSamples, unsigned nChannels)
-{
+static void copyStereo16(
+        short *dst,
+        const int *const *src,
+        unsigned nSamples,
+        unsigned /* nChannels */) {
     for (unsigned i = 0; i < nSamples; ++i) {
         *dst++ = src[0][i];
         *dst++ = src[1][i];
@@ -430,15 +442,21 @@
 
 // 24-bit versions should do dithering or noise-shaping, here or in AudioFlinger
 
-static void copyMono24(short *dst, const int *const *src, unsigned nSamples, unsigned nChannels)
-{
+static void copyMono24(
+        short *dst,
+        const int *const *src,
+        unsigned nSamples,
+        unsigned /* nChannels */) {
     for (unsigned i = 0; i < nSamples; ++i) {
         *dst++ = src[0][i] >> 8;
     }
 }
 
-static void copyStereo24(short *dst, const int *const *src, unsigned nSamples, unsigned nChannels)
-{
+static void copyStereo24(
+        short *dst,
+        const int *const *src,
+        unsigned nSamples,
+        unsigned /* nChannels */) {
     for (unsigned i = 0; i < nSamples; ++i) {
         *dst++ = src[0][i] >> 8;
         *dst++ = src[1][i] >> 8;
@@ -454,8 +472,11 @@
     }
 }
 
-static void copyTrespass(short *dst, const int *const *src, unsigned nSamples, unsigned nChannels)
-{
+static void copyTrespass(
+        short * /* dst */,
+        const int *const * /* src */,
+        unsigned /* nSamples */,
+        unsigned /* nChannels */) {
     TRESPASS();
 }
 
@@ -700,7 +721,7 @@
     }
 }
 
-status_t FLACSource::start(MetaData *params)
+status_t FLACSource::start(MetaData * /* params */)
 {
     ALOGV("FLACSource::start");
 
@@ -792,8 +813,7 @@
 }
 
 sp<MetaData> FLACExtractor::getTrackMetaData(
-        size_t index, uint32_t flags)
-{
+        size_t index, uint32_t /* flags */) {
     if (mInitCheck != OK || index > 0) {
         return NULL;
     }
diff --git a/media/libstagefright/HTTPBase.cpp b/media/libstagefright/HTTPBase.cpp
index 5fa4b6f..ca68c3d 100644
--- a/media/libstagefright/HTTPBase.cpp
+++ b/media/libstagefright/HTTPBase.cpp
@@ -20,10 +20,6 @@
 
 #include "include/HTTPBase.h"
 
-#if CHROMIUM_AVAILABLE
-#include "include/chromium_http_stub.h"
-#endif
-
 #include <media/stagefright/foundation/ADebug.h>
 #include <media/stagefright/foundation/ALooper.h>
 
@@ -40,34 +36,7 @@
       mTotalTransferBytes(0),
       mPrevBandwidthMeasureTimeUs(0),
       mPrevEstimatedBandWidthKbps(0),
-      mBandWidthCollectFreqMs(5000),
-      mUIDValid(false),
-      mUID(0) {
-}
-
-// static
-sp<HTTPBase> HTTPBase::Create(uint32_t flags) {
-#if CHROMIUM_AVAILABLE
-        HTTPBase *dataSource = createChromiumHTTPDataSource(flags);
-        if (dataSource) {
-           return dataSource;
-        }
-#endif
-    {
-        TRESPASS();
-
-        return NULL;
-    }
-}
-
-// static
-status_t HTTPBase::UpdateProxyConfig(
-        const char *host, int32_t port, const char *exclusionList) {
-#if CHROMIUM_AVAILABLE
-    return UpdateChromiumHTTPDataSourceProxyConfig(host, port, exclusionList);
-#else
-    return INVALID_OPERATION;
-#endif
+      mBandWidthCollectFreqMs(5000) {
 }
 
 void HTTPBase::addBandwidthMeasurement(
@@ -135,21 +104,6 @@
     return OK;
 }
 
-void HTTPBase::setUID(uid_t uid) {
-    mUIDValid = true;
-    mUID = uid;
-}
-
-bool HTTPBase::getUID(uid_t *uid) const {
-    if (!mUIDValid) {
-        return false;
-    }
-
-    *uid = mUID;
-
-    return true;
-}
-
 // static
 void HTTPBase::RegisterSocketUserTag(int sockfd, uid_t uid, uint32_t kTag) {
     int res = qtaguid_tagSocket(sockfd, kTag, uid);
diff --git a/media/libstagefright/MP3Extractor.cpp b/media/libstagefright/MP3Extractor.cpp
index 380dab4..4a63152 100644
--- a/media/libstagefright/MP3Extractor.cpp
+++ b/media/libstagefright/MP3Extractor.cpp
@@ -398,7 +398,8 @@
             mSeeker);
 }
 
-sp<MetaData> MP3Extractor::getTrackMetaData(size_t index, uint32_t flags) {
+sp<MetaData> MP3Extractor::getTrackMetaData(
+        size_t index, uint32_t /* flags */) {
     if (mInitCheck != OK || index != 0) {
         return NULL;
     }
diff --git a/media/libstagefright/MPEG2TSWriter.cpp b/media/libstagefright/MPEG2TSWriter.cpp
index c9ed5bb..78c12e1 100644
--- a/media/libstagefright/MPEG2TSWriter.cpp
+++ b/media/libstagefright/MPEG2TSWriter.cpp
@@ -555,7 +555,7 @@
     return OK;
 }
 
-status_t MPEG2TSWriter::start(MetaData *param) {
+status_t MPEG2TSWriter::start(MetaData * /* param */) {
     CHECK(!mStarted);
 
     mStarted = true;
@@ -596,7 +596,8 @@
     return !mStarted || (mNumSourcesDone == mSources.size() ? true : false);
 }
 
-status_t MPEG2TSWriter::dump(int fd, const Vector<String16> &args) {
+status_t MPEG2TSWriter::dump(
+        int /* fd */, const Vector<String16> & /* args */) {
     return OK;
 }
 
diff --git a/media/libstagefright/MPEG4Extractor.cpp b/media/libstagefright/MPEG4Extractor.cpp
index dc73980..4756b3e 100644
--- a/media/libstagefright/MPEG4Extractor.cpp
+++ b/media/libstagefright/MPEG4Extractor.cpp
@@ -39,6 +39,7 @@
 #include <utils/String8.h>
 
 #include <byteswap.h>
+#include "include/ID3.h"
 
 namespace android {
 
@@ -259,7 +260,7 @@
     const uint8_t *data = (const uint8_t *)_data;
     size_t offset = 0;
     while (offset < size) {
-        printf("0x%04x  ", offset);
+        printf("0x%04zx  ", offset);
 
         size_t n = size - offset;
         if (n > 16) {
@@ -570,7 +571,8 @@
     return size;
 }
 
-status_t MPEG4Extractor::parseDrmSINF(off64_t *offset, off64_t data_offset) {
+status_t MPEG4Extractor::parseDrmSINF(
+        off64_t * /* offset */, off64_t data_offset) {
     uint8_t updateIdTag;
     if (mDataSource->readAt(data_offset, &updateIdTag, 1) < 1) {
         return ERROR_IO;
@@ -1787,6 +1789,18 @@
             break;
         }
 
+        case FOURCC('I', 'D', '3', '2'):
+        {
+            if (chunk_data_size < 6) {
+                return ERROR_MALFORMED;
+            }
+
+            parseID3v2MetaData(data_offset + 6);
+
+            *offset += chunk_size;
+            break;
+        }
+
         case FOURCC('-', '-', '-', '-'):
         {
             mLastCommentMean.clear();
@@ -2167,7 +2181,7 @@
             break;
     }
 
-    if (size >= 8 && metadataKey) {
+    if (size >= 8 && metadataKey && !mFileMetaData->hasData(metadataKey)) {
         if (metadataKey == kKeyAlbumArt) {
             mFileMetaData->setData(
                     kKeyAlbumArt, MetaData::TYPE_NONE,
@@ -2316,6 +2330,62 @@
     return OK;
 }
 
+void MPEG4Extractor::parseID3v2MetaData(off64_t offset) {
+    ID3 id3(mDataSource, true /* ignorev1 */, offset);
+
+    if (id3.isValid()) {
+        struct Map {
+            int key;
+            const char *tag1;
+            const char *tag2;
+        };
+        static const Map kMap[] = {
+            { kKeyAlbum, "TALB", "TAL" },
+            { kKeyArtist, "TPE1", "TP1" },
+            { kKeyAlbumArtist, "TPE2", "TP2" },
+            { kKeyComposer, "TCOM", "TCM" },
+            { kKeyGenre, "TCON", "TCO" },
+            { kKeyTitle, "TIT2", "TT2" },
+            { kKeyYear, "TYE", "TYER" },
+            { kKeyAuthor, "TXT", "TEXT" },
+            { kKeyCDTrackNumber, "TRK", "TRCK" },
+            { kKeyDiscNumber, "TPA", "TPOS" },
+            { kKeyCompilation, "TCP", "TCMP" },
+        };
+        static const size_t kNumMapEntries = sizeof(kMap) / sizeof(kMap[0]);
+
+        for (size_t i = 0; i < kNumMapEntries; ++i) {
+            if (!mFileMetaData->hasData(kMap[i].key)) {
+                ID3::Iterator *it = new ID3::Iterator(id3, kMap[i].tag1);
+                if (it->done()) {
+                    delete it;
+                    it = new ID3::Iterator(id3, kMap[i].tag2);
+                }
+
+                if (it->done()) {
+                    delete it;
+                    continue;
+                }
+
+                String8 s;
+                it->getString(&s);
+                delete it;
+
+                mFileMetaData->setCString(kMap[i].key, s);
+            }
+        }
+
+        size_t dataSize;
+        String8 mime;
+        const void *data = id3.getAlbumArt(&dataSize, &mime);
+
+        if (data) {
+            mFileMetaData->setData(kKeyAlbumArt, MetaData::TYPE_NONE, data, dataSize);
+            mFileMetaData->setCString(kKeyAlbumArtMIME, mime.string());
+        }
+    }
+}
+
 sp<MediaSource> MPEG4Extractor::getTrack(size_t index) {
     status_t err;
     if ((err = readMetaData()) != OK) {
@@ -2372,6 +2442,58 @@
     return OK;
 }
 
+typedef enum {
+    //AOT_NONE             = -1,
+    //AOT_NULL_OBJECT      = 0,
+    //AOT_AAC_MAIN         = 1, /**< Main profile                              */
+    AOT_AAC_LC           = 2,   /**< Low Complexity object                     */
+    //AOT_AAC_SSR          = 3,
+    //AOT_AAC_LTP          = 4,
+    AOT_SBR              = 5,
+    //AOT_AAC_SCAL         = 6,
+    //AOT_TWIN_VQ          = 7,
+    //AOT_CELP             = 8,
+    //AOT_HVXC             = 9,
+    //AOT_RSVD_10          = 10, /**< (reserved)                                */
+    //AOT_RSVD_11          = 11, /**< (reserved)                                */
+    //AOT_TTSI             = 12, /**< TTSI Object                               */
+    //AOT_MAIN_SYNTH       = 13, /**< Main Synthetic object                     */
+    //AOT_WAV_TAB_SYNTH    = 14, /**< Wavetable Synthesis object                */
+    //AOT_GEN_MIDI         = 15, /**< General MIDI object                       */
+    //AOT_ALG_SYNTH_AUD_FX = 16, /**< Algorithmic Synthesis and Audio FX object */
+    AOT_ER_AAC_LC        = 17,   /**< Error Resilient(ER) AAC Low Complexity    */
+    //AOT_RSVD_18          = 18, /**< (reserved)                                */
+    //AOT_ER_AAC_LTP       = 19, /**< Error Resilient(ER) AAC LTP object        */
+    AOT_ER_AAC_SCAL      = 20,   /**< Error Resilient(ER) AAC Scalable object   */
+    //AOT_ER_TWIN_VQ       = 21, /**< Error Resilient(ER) TwinVQ object         */
+    AOT_ER_BSAC          = 22,   /**< Error Resilient(ER) BSAC object           */
+    AOT_ER_AAC_LD        = 23,   /**< Error Resilient(ER) AAC LowDelay object   */
+    //AOT_ER_CELP          = 24, /**< Error Resilient(ER) CELP object           */
+    //AOT_ER_HVXC          = 25, /**< Error Resilient(ER) HVXC object           */
+    //AOT_ER_HILN          = 26, /**< Error Resilient(ER) HILN object           */
+    //AOT_ER_PARA          = 27, /**< Error Resilient(ER) Parametric object     */
+    //AOT_RSVD_28          = 28, /**< might become SSC                          */
+    AOT_PS               = 29,   /**< PS, Parametric Stereo (includes SBR)      */
+    //AOT_MPEGS            = 30, /**< MPEG Surround                             */
+
+    AOT_ESCAPE           = 31,   /**< Signal AOT uses more than 5 bits          */
+
+    //AOT_MP3ONMP4_L1      = 32, /**< MPEG-Layer1 in mp4                        */
+    //AOT_MP3ONMP4_L2      = 33, /**< MPEG-Layer2 in mp4                        */
+    //AOT_MP3ONMP4_L3      = 34, /**< MPEG-Layer3 in mp4                        */
+    //AOT_RSVD_35          = 35, /**< might become DST                          */
+    //AOT_RSVD_36          = 36, /**< might become ALS                          */
+    //AOT_AAC_SLS          = 37, /**< AAC + SLS                                 */
+    //AOT_SLS              = 38, /**< SLS                                       */
+    //AOT_ER_AAC_ELD       = 39, /**< AAC Enhanced Low Delay                    */
+
+    //AOT_USAC             = 42, /**< USAC                                      */
+    //AOT_SAOC             = 43, /**< SAOC                                      */
+    //AOT_LD_MPEGS         = 44, /**< Low Delay MPEG Surround                   */
+
+    //AOT_RSVD50           = 50,  /**< Interim AOT for Rsvd50                   */
+} AUDIO_OBJECT_TYPE;
+
 status_t MPEG4Extractor::updateAudioTrackInfoFromESDS_MPEG4Audio(
         const void *esds_data, size_t esds_size) {
     ESDS esds(esds_data, esds_size);
@@ -2454,7 +2576,7 @@
         sampleRate = kSamplingRate[freqIndex];
     }
 
-    if (objectType == 5 || objectType == 29) { // SBR specific config per 14496-3 table 1.13
+    if (objectType == AOT_SBR || objectType == AOT_PS) {//SBR specific config per 14496-3 table 1.13
         uint32_t extFreqIndex = br.getBits(4);
         int32_t extSampleRate;
         if (extFreqIndex == 15) {
@@ -2472,6 +2594,111 @@
         //      mLastTrack->meta->setInt32(kKeyExtSampleRate, extSampleRate);
     }
 
+    switch (numChannels) {
+        // values defined in 14496-3_2009 amendment-4 Table 1.19 - Channel Configuration
+        case 0:
+        case 1:// FC
+        case 2:// FL FR
+        case 3:// FC, FL FR
+        case 4:// FC, FL FR, RC
+        case 5:// FC, FL FR, SL SR
+        case 6:// FC, FL FR, SL SR, LFE
+            //numChannels already contains the right value
+            break;
+        case 11:// FC, FL FR, SL SR, RC, LFE
+            numChannels = 7;
+            break;
+        case 7: // FC, FCL FCR, FL FR, SL SR, LFE
+        case 12:// FC, FL  FR,  SL SR, RL RR, LFE
+        case 14:// FC, FL  FR,  SL SR, LFE, FHL FHR
+            numChannels = 8;
+            break;
+        default:
+            return ERROR_UNSUPPORTED;
+    }
+
+    {
+        if (objectType == AOT_SBR || objectType == AOT_PS) {
+            const int32_t extensionSamplingFrequency = br.getBits(4);
+            objectType = br.getBits(5);
+
+            if (objectType == AOT_ESCAPE) {
+                objectType = 32 + br.getBits(6);
+            }
+        }
+        if (objectType == AOT_AAC_LC || objectType == AOT_ER_AAC_LC ||
+                objectType == AOT_ER_AAC_LD || objectType == AOT_ER_AAC_SCAL ||
+                objectType == AOT_ER_BSAC) {
+            const int32_t frameLengthFlag = br.getBits(1);
+
+            const int32_t dependsOnCoreCoder = br.getBits(1);
+
+            if (dependsOnCoreCoder ) {
+                const int32_t coreCoderDelay = br.getBits(14);
+            }
+
+            const int32_t extensionFlag = br.getBits(1);
+
+            if (numChannels == 0 ) {
+                int32_t channelsEffectiveNum = 0;
+                int32_t channelsNum = 0;
+                const int32_t ElementInstanceTag = br.getBits(4);
+                const int32_t Profile = br.getBits(2);
+                const int32_t SamplingFrequencyIndex = br.getBits(4);
+                const int32_t NumFrontChannelElements = br.getBits(4);
+                const int32_t NumSideChannelElements = br.getBits(4);
+                const int32_t NumBackChannelElements = br.getBits(4);
+                const int32_t NumLfeChannelElements = br.getBits(2);
+                const int32_t NumAssocDataElements = br.getBits(3);
+                const int32_t NumValidCcElements = br.getBits(4);
+
+                const int32_t MonoMixdownPresent = br.getBits(1);
+                if (MonoMixdownPresent != 0) {
+                    const int32_t MonoMixdownElementNumber = br.getBits(4);
+                }
+
+                const int32_t StereoMixdownPresent = br.getBits(1);
+                if (StereoMixdownPresent != 0) {
+                    const int32_t StereoMixdownElementNumber = br.getBits(4);
+                }
+
+                const int32_t MatrixMixdownIndexPresent = br.getBits(1);
+                if (MatrixMixdownIndexPresent != 0) {
+                    const int32_t MatrixMixdownIndex = br.getBits(2);
+                    const int32_t PseudoSurroundEnable = br.getBits(1);
+                }
+
+                int i;
+                for (i=0; i < NumFrontChannelElements; i++) {
+                    const int32_t FrontElementIsCpe = br.getBits(1);
+                    const int32_t FrontElementTagSelect = br.getBits(4);
+                    channelsNum += FrontElementIsCpe ? 2 : 1;
+                }
+
+                for (i=0; i < NumSideChannelElements; i++) {
+                    const int32_t SideElementIsCpe = br.getBits(1);
+                    const int32_t SideElementTagSelect = br.getBits(4);
+                    channelsNum += SideElementIsCpe ? 2 : 1;
+                }
+
+                for (i=0; i < NumBackChannelElements; i++) {
+                    const int32_t BackElementIsCpe = br.getBits(1);
+                    const int32_t BackElementTagSelect = br.getBits(4);
+                    channelsNum += BackElementIsCpe ? 2 : 1;
+                }
+                channelsEffectiveNum = channelsNum;
+
+                for (i=0; i < NumLfeChannelElements; i++) {
+                    const int32_t LfeElementTagSelect = br.getBits(4);
+                    channelsNum += 1;
+                }
+                ALOGV("mpeg4 audio channelsNum = %d", channelsNum);
+                ALOGV("mpeg4 audio channelsEffectiveNum = %d", channelsEffectiveNum);
+                numChannels = channelsNum;
+            }
+        }
+    }
+
     if (numChannels == 0) {
         return ERROR_UNSUPPORTED;
     }
@@ -2733,7 +2960,8 @@
     return OK;
 }
 
-status_t MPEG4Source::parseSampleAuxiliaryInformationSizes(off64_t offset, off64_t size) {
+status_t MPEG4Source::parseSampleAuxiliaryInformationSizes(
+        off64_t offset, off64_t /* size */) {
     ALOGV("parseSampleAuxiliaryInformationSizes");
     // 14496-12 8.7.12
     uint8_t version;
@@ -2795,7 +3023,8 @@
     return OK;
 }
 
-status_t MPEG4Source::parseSampleAuxiliaryInformationOffsets(off64_t offset, off64_t size) {
+status_t MPEG4Source::parseSampleAuxiliaryInformationOffsets(
+        off64_t offset, off64_t /* size */) {
     ALOGV("parseSampleAuxiliaryInformationOffsets");
     // 14496-12 8.7.13
     uint8_t version;
@@ -3506,7 +3735,7 @@
     }
 
     off64_t offset = 0;
-    size_t size;
+    size_t size = 0;
     uint32_t cts = 0;
     bool isSyncSample = false;
     bool newBuffer = false;
diff --git a/media/libstagefright/MPEG4Writer.cpp b/media/libstagefright/MPEG4Writer.cpp
index a0f17b5..c839560 100644
--- a/media/libstagefright/MPEG4Writer.cpp
+++ b/media/libstagefright/MPEG4Writer.cpp
@@ -16,6 +16,7 @@
 
 //#define LOG_NDEBUG 0
 #define LOG_TAG "MPEG4Writer"
+#include <inttypes.h>
 #include <utils/Log.h>
 
 #include <arpa/inet.h>
@@ -43,7 +44,9 @@
 namespace android {
 
 static const int64_t kMinStreamableFileSizeInBytes = 5 * 1024 * 1024;
-static const int64_t kMax32BitFileSize = 0x007fffffffLL;
+static const int64_t kMax32BitFileSize = 0x00ffffffffLL; // 2^32-1 : max FAT32
+                                                         // filesystem file size
+                                                         // used by most SD cards
 static const uint8_t kNalUnitTypeSeqParamSet = 0x07;
 static const uint8_t kNalUnitTypePicParamSet = 0x08;
 static const int64_t kInitialDelayTimeUs     = 700000LL;
@@ -406,7 +409,7 @@
 }
 
 status_t MPEG4Writer::Track::dump(
-        int fd, const Vector<String16>& args) const {
+        int fd, const Vector<String16>& /* args */) const {
     const size_t SIZE = 256;
     char buffer[SIZE];
     String8 result;
@@ -417,7 +420,7 @@
     result.append(buffer);
     snprintf(buffer, SIZE, "       frames encoded : %d\n", mStszTableEntries->count());
     result.append(buffer);
-    snprintf(buffer, SIZE, "       duration encoded : %lld us\n", mTrackDurationUs);
+    snprintf(buffer, SIZE, "       duration encoded : %" PRId64 " us\n", mTrackDurationUs);
     result.append(buffer);
     ::write(fd, result.string(), result.size());
     return OK;
@@ -860,11 +863,11 @@
     // Fix up the size of the 'mdat' chunk.
     if (mUse32BitOffset) {
         lseek64(mFd, mMdatOffset, SEEK_SET);
-        int32_t size = htonl(static_cast<int32_t>(mOffset - mMdatOffset));
+        uint32_t size = htonl(static_cast<uint32_t>(mOffset - mMdatOffset));
         ::write(mFd, &size, 4);
     } else {
         lseek64(mFd, mMdatOffset + 8, SEEK_SET);
-        int64_t size = mOffset - mMdatOffset;
+        uint64_t size = mOffset - mMdatOffset;
         size = hton64(size);
         ::write(mFd, &size, 8);
     }
@@ -972,13 +975,16 @@
     if (param && param->findInt32(kKeyFileType, &fileType) &&
         fileType != OUTPUT_FORMAT_MPEG_4) {
         writeFourcc("3gp4");
-    } else {
+        writeInt32(0);
         writeFourcc("isom");
+        writeFourcc("3gp4");
+    } else {
+        writeFourcc("mp42");
+        writeInt32(0);
+        writeFourcc("isom");
+        writeFourcc("mp42");
     }
 
-    writeInt32(0);
-    writeFourcc("isom");
-    writeFourcc("3gp4");
     endBox();
 }
 
@@ -1404,7 +1410,7 @@
         size_t sampleCount, int32_t duration) {
 
     if (duration == 0) {
-        ALOGW("0-duration samples found: %d", sampleCount);
+        ALOGW("0-duration samples found: %zu", sampleCount);
     }
     mSttsTableEntries->add(htonl(sampleCount));
     mSttsTableEntries->add(htonl(duration));
@@ -1584,7 +1590,7 @@
     sendSessionSummary();
 
     mChunkInfos.clear();
-    ALOGD("%d chunks are written in the last batch", outstandingChunks);
+    ALOGD("%zu chunks are written in the last batch", outstandingChunks);
 }
 
 bool MPEG4Writer::findChunkToWrite(Chunk *chunk) {
@@ -1760,7 +1766,7 @@
 }
 
 status_t MPEG4Writer::Track::stop() {
-    ALOGD("Stopping %s track", mIsAudio? "Audio": "Video");
+    ALOGD("%s track stopping", mIsAudio? "Audio": "Video");
     if (!mStarted) {
         ALOGE("Stop() called but track is not started");
         return ERROR_END_OF_STREAM;
@@ -1771,18 +1777,13 @@
     }
     mDone = true;
 
+    ALOGD("%s track source stopping", mIsAudio? "Audio": "Video");
+    mSource->stop();
+    ALOGD("%s track source stopped", mIsAudio? "Audio": "Video");
+
     void *dummy;
     pthread_join(mThread, &dummy);
-
-    status_t err = (status_t) dummy;
-
-    ALOGD("Stopping %s track source", mIsAudio? "Audio": "Video");
-    {
-        status_t status = mSource->stop();
-        if (err == OK && status != OK && status != ERROR_END_OF_STREAM) {
-            err = status;
-        }
-    }
+    status_t err = static_cast<status_t>(reinterpret_cast<uintptr_t>(dummy));
 
     ALOGD("%s track stopped", mIsAudio? "Audio": "Video");
     return err;
@@ -1797,7 +1798,7 @@
     Track *track = static_cast<Track *>(me);
 
     status_t err = track->threadEntry();
-    return (void *) err;
+    return (void *)(uintptr_t)err;
 }
 
 static void getNalUnitType(uint8_t byte, uint8_t* type) {
@@ -1869,7 +1870,7 @@
     // 2 bytes for each of the parameter set length field
     // plus the 7 bytes for the header
     if (size < 4 + 7) {
-        ALOGE("Codec specific data length too short: %d", size);
+        ALOGE("Codec specific data length too short: %zu", size);
         return ERROR_MALFORMED;
     }
 
@@ -1938,7 +1939,7 @@
         }
 
         if (nSeqParamSets > 0x1F) {
-            ALOGE("Too many seq parameter sets (%d) found", nSeqParamSets);
+            ALOGE("Too many seq parameter sets (%zu) found", nSeqParamSets);
             return ERROR_MALFORMED;
         }
     }
@@ -1951,7 +1952,7 @@
             return ERROR_MALFORMED;
         }
         if (nPicParamSets > 0xFF) {
-            ALOGE("Too many pic parameter sets (%d) found", nPicParamSets);
+            ALOGE("Too many pic parameter sets (%zd) found", nPicParamSets);
             return ERROR_MALFORMED;
         }
     }
@@ -1981,7 +1982,7 @@
     }
 
     if (size < 4) {
-        ALOGE("Codec specific data length too short: %d", size);
+        ALOGE("Codec specific data length too short: %zu", size);
         return ERROR_MALFORMED;
     }
 
@@ -2279,6 +2280,22 @@
             return UNKNOWN_ERROR;
         }
 
+        // if the duration is different for this sample, see if it is close enough to the previous
+        // duration that we can fudge it and use the same value, to avoid filling the stts table
+        // with lots of near-identical entries.
+        // "close enough" here means that the current duration needs to be adjusted by less
+        // than 0.1 milliseconds
+        if (lastDurationTicks && (currDurationTicks != lastDurationTicks)) {
+            int64_t deltaUs = ((lastDurationTicks - currDurationTicks) * 1000000LL
+                    + (mTimeScale / 2)) / mTimeScale;
+            if (deltaUs > -100 && deltaUs < 100) {
+                // use previous ticks, and adjust timestamp as if it was actually that number
+                // of ticks
+                currDurationTicks = lastDurationTicks;
+                timestampUs += deltaUs;
+            }
+        }
+
         mStszTableEntries->add(htonl(sampleSize));
         if (mStszTableEntries->count() > 2) {
 
diff --git a/media/libstagefright/MediaAdapter.cpp b/media/libstagefright/MediaAdapter.cpp
index 2484212..d680e0c 100644
--- a/media/libstagefright/MediaAdapter.cpp
+++ b/media/libstagefright/MediaAdapter.cpp
@@ -36,7 +36,7 @@
     CHECK(mCurrentMediaBuffer == NULL);
 }
 
-status_t MediaAdapter::start(MetaData *params) {
+status_t MediaAdapter::start(MetaData * /* params */) {
     Mutex::Autolock autoLock(mAdapterLock);
     if (!mStarted) {
         mStarted = true;
@@ -75,7 +75,7 @@
 }
 
 status_t MediaAdapter::read(
-            MediaBuffer **buffer, const ReadOptions *options) {
+            MediaBuffer **buffer, const ReadOptions * /* options */) {
     Mutex::Autolock autoLock(mAdapterLock);
     if (!mStarted) {
         ALOGV("Read before even started!");
diff --git a/media/libstagefright/MediaCodec.cpp b/media/libstagefright/MediaCodec.cpp
index 8af1aaf..fe21296 100644
--- a/media/libstagefright/MediaCodec.cpp
+++ b/media/libstagefright/MediaCodec.cpp
@@ -115,7 +115,7 @@
         if (codecIdx >= 0) {
             Vector<AString> types;
             if (mcl->getSupportedTypes(codecIdx, &types) == OK) {
-                for (int i = 0; i < types.size(); i++) {
+                for (size_t i = 0; i < types.size(); i++) {
                     if (types[i].startsWith("video/")) {
                         needDedicatedLooper = true;
                         break;
@@ -1680,7 +1680,7 @@
         return -EACCES;
     }
 
-    if (render && (info->mData == NULL || info->mData->size() != 0)) {
+    if (render && info->mData != NULL && info->mData->size() != 0) {
         info->mNotify->setInt32("render", true);
 
         if (mSoftRenderer != NULL) {
diff --git a/media/libstagefright/MediaCodecSource.cpp b/media/libstagefright/MediaCodecSource.cpp
new file mode 100644
index 0000000..924173c
--- /dev/null
+++ b/media/libstagefright/MediaCodecSource.cpp
@@ -0,0 +1,881 @@
+/*
+ * Copyright 2014, The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *     http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "MediaCodecSource"
+#define DEBUG_DRIFT_TIME 0
+#include <gui/IGraphicBufferProducer.h>
+#include <gui/Surface.h>
+#include <media/ICrypto.h>
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/ALooper.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/MediaBuffer.h>
+#include <media/stagefright/MediaCodec.h>
+#include <media/stagefright/MetaData.h>
+#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/MediaSource.h>
+#include <media/stagefright/MediaCodecSource.h>
+#include <media/stagefright/Utils.h>
+
+namespace android {
+
+static void ReleaseMediaBufferReference(const sp<ABuffer> &accessUnit) {
+    void *mbuf;
+    if (accessUnit->meta()->findPointer("mediaBuffer", &mbuf)
+            && mbuf != NULL) {
+        ALOGV("releasing mbuf %p", mbuf);
+
+        accessUnit->meta()->setPointer("mediaBuffer", NULL);
+
+        static_cast<MediaBuffer *>(mbuf)->release();
+        mbuf = NULL;
+    }
+}
+
+struct MediaCodecSource::Puller : public AHandler {
+    Puller(const sp<MediaSource> &source);
+
+    status_t start(const sp<MetaData> &meta, const sp<AMessage> &notify);
+    void stopAsync();
+
+    void pause();
+    void resume();
+
+protected:
+    virtual void onMessageReceived(const sp<AMessage> &msg);
+    virtual ~Puller();
+
+private:
+    enum {
+        kWhatStart = 'msta',
+        kWhatStop,
+        kWhatPull,
+        kWhatPause,
+        kWhatResume,
+    };
+
+    sp<MediaSource> mSource;
+    sp<AMessage> mNotify;
+    sp<ALooper> mLooper;
+    int32_t mPullGeneration;
+    bool mIsAudio;
+    bool mPaused;
+    bool mReachedEOS;
+
+    status_t postSynchronouslyAndReturnError(const sp<AMessage> &msg);
+    void schedulePull();
+    void handleEOS();
+
+    DISALLOW_EVIL_CONSTRUCTORS(Puller);
+};
+
+MediaCodecSource::Puller::Puller(const sp<MediaSource> &source)
+    : mSource(source),
+      mLooper(new ALooper()),
+      mPullGeneration(0),
+      mIsAudio(false),
+      mPaused(false),
+      mReachedEOS(false) {
+    sp<MetaData> meta = source->getFormat();
+    const char *mime;
+    CHECK(meta->findCString(kKeyMIMEType, &mime));
+
+    mIsAudio = !strncasecmp(mime, "audio/", 6);
+
+    mLooper->setName("pull_looper");
+}
+
+MediaCodecSource::Puller::~Puller() {
+    mLooper->unregisterHandler(id());
+    mLooper->stop();
+}
+
+status_t MediaCodecSource::Puller::postSynchronouslyAndReturnError(
+        const sp<AMessage> &msg) {
+    sp<AMessage> response;
+    status_t err = msg->postAndAwaitResponse(&response);
+
+    if (err != OK) {
+        return err;
+    }
+
+    if (!response->findInt32("err", &err)) {
+        err = OK;
+    }
+
+    return err;
+}
+
+status_t MediaCodecSource::Puller::start(const sp<MetaData> &meta,
+        const sp<AMessage> &notify) {
+    ALOGV("puller (%s) start", mIsAudio ? "audio" : "video");
+    mLooper->start(
+            false /* runOnCallingThread */,
+            false /* canCallJava */,
+            PRIORITY_AUDIO);
+    mLooper->registerHandler(this);
+    mNotify = notify;
+
+    sp<AMessage> msg = new AMessage(kWhatStart, id());
+    msg->setObject("meta", meta);
+    return postSynchronouslyAndReturnError(msg);
+}
+
+void MediaCodecSource::Puller::stopAsync() {
+    ALOGV("puller (%s) stopAsync", mIsAudio ? "audio" : "video");
+    (new AMessage(kWhatStop, id()))->post();
+}
+
+void MediaCodecSource::Puller::pause() {
+    (new AMessage(kWhatPause, id()))->post();
+}
+
+void MediaCodecSource::Puller::resume() {
+    (new AMessage(kWhatResume, id()))->post();
+}
+
+void MediaCodecSource::Puller::schedulePull() {
+    sp<AMessage> msg = new AMessage(kWhatPull, id());
+    msg->setInt32("generation", mPullGeneration);
+    msg->post();
+}
+
+void MediaCodecSource::Puller::handleEOS() {
+    if (!mReachedEOS) {
+        ALOGV("puller (%s) posting EOS", mIsAudio ? "audio" : "video");
+        mReachedEOS = true;
+        sp<AMessage> notify = mNotify->dup();
+        notify->setPointer("accessUnit", NULL);
+        notify->post();
+    }
+}
+
+void MediaCodecSource::Puller::onMessageReceived(const sp<AMessage> &msg) {
+    switch (msg->what()) {
+        case kWhatStart:
+        {
+            sp<RefBase> obj;
+            CHECK(msg->findObject("meta", &obj));
+
+            mReachedEOS = false;
+
+            status_t err = mSource->start(static_cast<MetaData *>(obj.get()));
+
+            if (err == OK) {
+                schedulePull();
+            }
+
+            sp<AMessage> response = new AMessage;
+            response->setInt32("err", err);
+
+            uint32_t replyID;
+            CHECK(msg->senderAwaitsResponse(&replyID));
+            response->postReply(replyID);
+            break;
+        }
+
+        case kWhatStop:
+        {
+            ALOGV("source (%s) stopping", mIsAudio ? "audio" : "video");
+            mSource->stop();
+            ALOGV("source (%s) stopped", mIsAudio ? "audio" : "video");
+            ++mPullGeneration;
+
+            handleEOS();
+            break;
+        }
+
+        case kWhatPull:
+        {
+            int32_t generation;
+            CHECK(msg->findInt32("generation", &generation));
+
+            if (generation != mPullGeneration) {
+                break;
+            }
+
+            MediaBuffer *mbuf;
+            status_t err = mSource->read(&mbuf);
+
+            if (mPaused) {
+                if (err == OK) {
+                    mbuf->release();
+                    mbuf = NULL;
+                }
+
+                msg->post();
+                break;
+            }
+
+            if (err != OK) {
+                if (err == ERROR_END_OF_STREAM) {
+                    ALOGV("stream ended, mbuf %p", mbuf);
+                } else {
+                    ALOGE("error %d reading stream.", err);
+                }
+                handleEOS();
+            } else {
+                sp<AMessage> notify = mNotify->dup();
+
+                notify->setPointer("accessUnit", mbuf);
+                notify->post();
+
+                msg->post();
+            }
+            break;
+        }
+
+        case kWhatPause:
+        {
+            mPaused = true;
+            break;
+        }
+
+        case kWhatResume:
+        {
+            mPaused = false;
+            break;
+        }
+
+        default:
+            TRESPASS();
+    }
+}
+
+// static
+sp<MediaCodecSource> MediaCodecSource::Create(
+        const sp<ALooper> &looper,
+        const sp<AMessage> &format,
+        const sp<MediaSource> &source,
+        uint32_t flags) {
+    sp<MediaCodecSource> mediaSource =
+            new MediaCodecSource(looper, format, source, flags);
+
+    if (mediaSource->init() == OK) {
+        return mediaSource;
+    }
+    return NULL;
+}
+
+status_t MediaCodecSource::start(MetaData* params) {
+    sp<AMessage> msg = new AMessage(kWhatStart, mReflector->id());
+    msg->setObject("meta", params);
+    return postSynchronouslyAndReturnError(msg);
+}
+
+status_t MediaCodecSource::stop() {
+    sp<AMessage> msg = new AMessage(kWhatStop, mReflector->id());
+    return postSynchronouslyAndReturnError(msg);
+}
+
+status_t MediaCodecSource::pause() {
+    (new AMessage(kWhatPause, mReflector->id()))->post();
+    return OK;
+}
+
+sp<IGraphicBufferProducer> MediaCodecSource::getGraphicBufferProducer() {
+    CHECK(mFlags & FLAG_USE_SURFACE_INPUT);
+    return mGraphicBufferProducer;
+}
+
+status_t MediaCodecSource::read(
+        MediaBuffer** buffer, const ReadOptions* /* options */) {
+    Mutex::Autolock autolock(mOutputBufferLock);
+
+    *buffer = NULL;
+    while (mOutputBufferQueue.size() == 0 && !mEncodedReachedEOS) {
+        mOutputBufferCond.wait(mOutputBufferLock);
+    }
+    if (!mEncodedReachedEOS) {
+        *buffer = *mOutputBufferQueue.begin();
+        mOutputBufferQueue.erase(mOutputBufferQueue.begin());
+        return OK;
+    }
+    return mErrorCode;
+}
+
+void MediaCodecSource::signalBufferReturned(MediaBuffer *buffer) {
+    buffer->setObserver(0);
+    buffer->release();
+}
+
+MediaCodecSource::MediaCodecSource(
+        const sp<ALooper> &looper,
+        const sp<AMessage> &outputFormat,
+        const sp<MediaSource> &source,
+        uint32_t flags)
+    : mLooper(looper),
+      mOutputFormat(outputFormat),
+      mMeta(new MetaData),
+      mFlags(flags),
+      mIsVideo(false),
+      mStarted(false),
+      mStopping(false),
+      mDoMoreWorkPending(false),
+      mPullerReachedEOS(false),
+      mFirstSampleTimeUs(-1ll),
+      mEncodedReachedEOS(false),
+      mErrorCode(OK) {
+    CHECK(mLooper != NULL);
+
+    AString mime;
+    CHECK(mOutputFormat->findString("mime", &mime));
+
+    if (!strncasecmp("video/", mime.c_str(), 6)) {
+        mIsVideo = true;
+    }
+
+    if (!(mFlags & FLAG_USE_SURFACE_INPUT)) {
+        mPuller = new Puller(source);
+    }
+}
+
+MediaCodecSource::~MediaCodecSource() {
+    releaseEncoder();
+
+    mCodecLooper->stop();
+    mLooper->unregisterHandler(mReflector->id());
+}
+
+status_t MediaCodecSource::init() {
+    status_t err = initEncoder();
+
+    if (err != OK) {
+        releaseEncoder();
+    }
+
+    return err;
+}
+
+status_t MediaCodecSource::initEncoder() {
+    mReflector = new AHandlerReflector<MediaCodecSource>(this);
+    mLooper->registerHandler(mReflector);
+
+    mCodecLooper = new ALooper;
+    mCodecLooper->setName("codec_looper");
+    mCodecLooper->start();
+
+    if (mFlags & FLAG_USE_METADATA_INPUT) {
+        mOutputFormat->setInt32("store-metadata-in-buffers", 1);
+    }
+
+    if (mFlags & FLAG_USE_SURFACE_INPUT) {
+        mOutputFormat->setInt32("create-input-buffers-suspended", 1);
+    }
+
+    AString outputMIME;
+    CHECK(mOutputFormat->findString("mime", &outputMIME));
+
+    mEncoder = MediaCodec::CreateByType(
+            mCodecLooper, outputMIME.c_str(), true /* encoder */);
+
+    if (mEncoder == NULL) {
+        return NO_INIT;
+    }
+
+    ALOGV("output format is '%s'", mOutputFormat->debugString(0).c_str());
+
+    status_t err = mEncoder->configure(
+                mOutputFormat,
+                NULL /* nativeWindow */,
+                NULL /* crypto */,
+                MediaCodec::CONFIGURE_FLAG_ENCODE);
+
+    if (err != OK) {
+        return err;
+    }
+
+    mEncoder->getOutputFormat(&mOutputFormat);
+    convertMessageToMetaData(mOutputFormat, mMeta);
+
+    if (mFlags & FLAG_USE_SURFACE_INPUT) {
+        CHECK(mIsVideo);
+
+        err = mEncoder->createInputSurface(&mGraphicBufferProducer);
+
+        if (err != OK) {
+            return err;
+        }
+    }
+
+    err = mEncoder->start();
+
+    if (err != OK) {
+        return err;
+    }
+
+    err = mEncoder->getInputBuffers(&mEncoderInputBuffers);
+
+    if (err != OK) {
+        return err;
+    }
+
+    err = mEncoder->getOutputBuffers(&mEncoderOutputBuffers);
+
+    if (err != OK) {
+        return err;
+    }
+
+    mEncodedReachedEOS = false;
+    mErrorCode = OK;
+
+    return OK;
+}
+
+void MediaCodecSource::releaseEncoder() {
+    if (mEncoder == NULL) {
+        return;
+    }
+
+    mEncoder->release();
+    mEncoder.clear();
+
+    while (!mInputBufferQueue.empty()) {
+        MediaBuffer *mbuf = *mInputBufferQueue.begin();
+        mInputBufferQueue.erase(mInputBufferQueue.begin());
+        if (mbuf != NULL) {
+            mbuf->release();
+        }
+    }
+
+    for (size_t i = 0; i < mEncoderInputBuffers.size(); ++i) {
+        sp<ABuffer> accessUnit = mEncoderInputBuffers.itemAt(i);
+        ReleaseMediaBufferReference(accessUnit);
+    }
+
+    mEncoderInputBuffers.clear();
+    mEncoderOutputBuffers.clear();
+}
+
+bool MediaCodecSource::reachedEOS() {
+    return mEncodedReachedEOS && ((mPuller == NULL) || mPullerReachedEOS);
+}
+
+status_t MediaCodecSource::postSynchronouslyAndReturnError(
+        const sp<AMessage> &msg) {
+    sp<AMessage> response;
+    status_t err = msg->postAndAwaitResponse(&response);
+
+    if (err != OK) {
+        return err;
+    }
+
+    if (!response->findInt32("err", &err)) {
+        err = OK;
+    }
+
+    return err;
+}
+
+void MediaCodecSource::signalEOS(status_t err) {
+    if (!mEncodedReachedEOS) {
+        ALOGI("encoder (%s) reached EOS", mIsVideo ? "video" : "audio");
+        {
+            Mutex::Autolock autoLock(mOutputBufferLock);
+            // release all unread media buffers
+            for (List<MediaBuffer*>::iterator it = mOutputBufferQueue.begin();
+                    it != mOutputBufferQueue.end(); it++) {
+                (*it)->release();
+            }
+            mOutputBufferQueue.clear();
+            mEncodedReachedEOS = true;
+            mErrorCode = err;
+            mOutputBufferCond.signal();
+        }
+
+        releaseEncoder();
+    }
+    if (mStopping && reachedEOS()) {
+        ALOGI("MediaCodecSource (%s) fully stopped",
+                mIsVideo ? "video" : "audio");
+        // posting reply to everyone that's waiting
+        List<uint32_t>::iterator it;
+        for (it = mStopReplyIDQueue.begin();
+                it != mStopReplyIDQueue.end(); it++) {
+            (new AMessage)->postReply(*it);
+        }
+        mStopReplyIDQueue.clear();
+        mStopping = false;
+    }
+}
+
+void MediaCodecSource::suspend() {
+    CHECK(mFlags & FLAG_USE_SURFACE_INPUT);
+    if (mEncoder != NULL) {
+        sp<AMessage> params = new AMessage;
+        params->setInt32("drop-input-frames", true);
+        mEncoder->setParameters(params);
+    }
+}
+
+void MediaCodecSource::resume(int64_t skipFramesBeforeUs) {
+    CHECK(mFlags & FLAG_USE_SURFACE_INPUT);
+    if (mEncoder != NULL) {
+        sp<AMessage> params = new AMessage;
+        params->setInt32("drop-input-frames", false);
+        if (skipFramesBeforeUs > 0) {
+            params->setInt64("skip-frames-before", skipFramesBeforeUs);
+        }
+        mEncoder->setParameters(params);
+    }
+}
+
+void MediaCodecSource::scheduleDoMoreWork() {
+    if (mDoMoreWorkPending) {
+        return;
+    }
+
+    mDoMoreWorkPending = true;
+
+    if (mEncoderActivityNotify == NULL) {
+        mEncoderActivityNotify = new AMessage(
+                kWhatEncoderActivity, mReflector->id());
+    }
+    mEncoder->requestActivityNotification(mEncoderActivityNotify);
+}
+
+status_t MediaCodecSource::feedEncoderInputBuffers() {
+    while (!mInputBufferQueue.empty()
+            && !mAvailEncoderInputIndices.empty()) {
+        MediaBuffer* mbuf = *mInputBufferQueue.begin();
+        mInputBufferQueue.erase(mInputBufferQueue.begin());
+
+        size_t bufferIndex = *mAvailEncoderInputIndices.begin();
+        mAvailEncoderInputIndices.erase(mAvailEncoderInputIndices.begin());
+
+        int64_t timeUs = 0ll;
+        uint32_t flags = 0;
+        size_t size = 0;
+
+        if (mbuf != NULL) {
+            CHECK(mbuf->meta_data()->findInt64(kKeyTime, &timeUs));
+
+            // push decoding time for video, or drift time for audio
+            if (mIsVideo) {
+                mDecodingTimeQueue.push_back(timeUs);
+            } else {
+#if DEBUG_DRIFT_TIME
+                if (mFirstSampleTimeUs < 0ll) {
+                    mFirstSampleTimeUs = timeUs;
+                }
+
+                int64_t driftTimeUs = 0;
+                if (mbuf->meta_data()->findInt64(kKeyDriftTime, &driftTimeUs)
+                        && driftTimeUs) {
+                    driftTimeUs = timeUs - mFirstSampleTimeUs - driftTimeUs;
+                }
+                mDriftTimeQueue.push_back(driftTimeUs);
+#endif // DEBUG_DRIFT_TIME
+            }
+
+            size = mbuf->size();
+
+            memcpy(mEncoderInputBuffers.itemAt(bufferIndex)->data(),
+                   mbuf->data(), size);
+
+            if (mIsVideo) {
+                // video encoder will release MediaBuffer when done
+                // with underlying data.
+                mEncoderInputBuffers.itemAt(bufferIndex)->meta()
+                        ->setPointer("mediaBuffer", mbuf);
+            } else {
+                mbuf->release();
+            }
+        } else {
+            flags = MediaCodec::BUFFER_FLAG_EOS;
+        }
+
+        status_t err = mEncoder->queueInputBuffer(
+                bufferIndex, 0, size, timeUs, flags);
+
+        if (err != OK) {
+            return err;
+        }
+    }
+
+    return OK;
+}
+
+status_t MediaCodecSource::doMoreWork() {
+    status_t err;
+
+    if (!(mFlags & FLAG_USE_SURFACE_INPUT)) {
+        for (;;) {
+            size_t bufferIndex;
+            err = mEncoder->dequeueInputBuffer(&bufferIndex);
+
+            if (err != OK) {
+                break;
+            }
+
+            mAvailEncoderInputIndices.push_back(bufferIndex);
+        }
+
+        feedEncoderInputBuffers();
+    }
+
+    for (;;) {
+        size_t bufferIndex;
+        size_t offset;
+        size_t size;
+        int64_t timeUs;
+        uint32_t flags;
+        native_handle_t* handle = NULL;
+        err = mEncoder->dequeueOutputBuffer(
+                &bufferIndex, &offset, &size, &timeUs, &flags);
+
+        if (err != OK) {
+            if (err == INFO_FORMAT_CHANGED) {
+                continue;
+            } else if (err == INFO_OUTPUT_BUFFERS_CHANGED) {
+                mEncoder->getOutputBuffers(&mEncoderOutputBuffers);
+                continue;
+            }
+
+            if (err == -EAGAIN) {
+                err = OK;
+            }
+            break;
+        }
+        if (!(flags & MediaCodec::BUFFER_FLAG_EOS)) {
+            sp<ABuffer> outbuf = mEncoderOutputBuffers.itemAt(bufferIndex);
+
+            MediaBuffer *mbuf = new MediaBuffer(outbuf->size());
+            memcpy(mbuf->data(), outbuf->data(), outbuf->size());
+
+            if (!(flags & MediaCodec::BUFFER_FLAG_CODECCONFIG)) {
+                if (mIsVideo) {
+                    int64_t decodingTimeUs;
+                    if (mFlags & FLAG_USE_SURFACE_INPUT) {
+                        // GraphicBufferSource is supposed to discard samples
+                        // queued before start, and offset timeUs by start time
+                        CHECK_GE(timeUs, 0ll);
+                        // TODO:
+                        // Decoding time for surface source is unavailable,
+                        // use presentation time for now. May need to move
+                        // this logic into MediaCodec.
+                        decodingTimeUs = timeUs;
+                    } else {
+                        CHECK(!mDecodingTimeQueue.empty());
+                        decodingTimeUs = *(mDecodingTimeQueue.begin());
+                        mDecodingTimeQueue.erase(mDecodingTimeQueue.begin());
+                    }
+                    mbuf->meta_data()->setInt64(kKeyDecodingTime, decodingTimeUs);
+
+                    ALOGV("[video] time %lld us (%.2f secs), dts/pts diff %lld",
+                            timeUs, timeUs / 1E6, decodingTimeUs - timeUs);
+                } else {
+                    int64_t driftTimeUs = 0;
+#if DEBUG_DRIFT_TIME
+                    CHECK(!mDriftTimeQueue.empty());
+                    driftTimeUs = *(mDriftTimeQueue.begin());
+                    mDriftTimeQueue.erase(mDriftTimeQueue.begin());
+                    mbuf->meta_data()->setInt64(kKeyDriftTime, driftTimeUs);
+#endif // DEBUG_DRIFT_TIME
+                    ALOGV("[audio] time %lld us (%.2f secs), drift %lld",
+                            timeUs, timeUs / 1E6, driftTimeUs);
+                }
+                mbuf->meta_data()->setInt64(kKeyTime, timeUs);
+            } else {
+                mbuf->meta_data()->setInt32(kKeyIsCodecConfig, true);
+            }
+            if (flags & MediaCodec::BUFFER_FLAG_SYNCFRAME) {
+                mbuf->meta_data()->setInt32(kKeyIsSyncFrame, true);
+            }
+            mbuf->setObserver(this);
+            mbuf->add_ref();
+
+            {
+                Mutex::Autolock autoLock(mOutputBufferLock);
+                mOutputBufferQueue.push_back(mbuf);
+                mOutputBufferCond.signal();
+            }
+        }
+
+        mEncoder->releaseOutputBuffer(bufferIndex);
+
+        if (flags & MediaCodec::BUFFER_FLAG_EOS) {
+            err = ERROR_END_OF_STREAM;
+            break;
+        }
+    }
+
+    return err;
+}
+
+status_t MediaCodecSource::onStart(MetaData *params) {
+    if (mStopping) {
+        ALOGE("Failed to start while we're stopping");
+        return INVALID_OPERATION;
+    }
+
+    if (mStarted) {
+        ALOGI("MediaCodecSource (%s) resuming", mIsVideo ? "video" : "audio");
+        if (mFlags & FLAG_USE_SURFACE_INPUT) {
+            resume();
+        } else {
+            CHECK(mPuller != NULL);
+            mPuller->resume();
+        }
+        return OK;
+    }
+
+    ALOGI("MediaCodecSource (%s) starting", mIsVideo ? "video" : "audio");
+
+    status_t err = OK;
+
+    if (mFlags & FLAG_USE_SURFACE_INPUT) {
+        int64_t startTimeUs;
+        if (!params || !params->findInt64(kKeyTime, &startTimeUs)) {
+            startTimeUs = -1ll;
+        }
+        resume(startTimeUs);
+        scheduleDoMoreWork();
+    } else {
+        CHECK(mPuller != NULL);
+        sp<AMessage> notify = new AMessage(
+                kWhatPullerNotify, mReflector->id());
+        err = mPuller->start(params, notify);
+        if (err != OK) {
+            mPullerReachedEOS = true;
+            return err;
+        }
+    }
+
+    ALOGI("MediaCodecSource (%s) started", mIsVideo ? "video" : "audio");
+
+    mStarted = true;
+    return OK;
+}
+
+void MediaCodecSource::onMessageReceived(const sp<AMessage> &msg) {
+    switch (msg->what()) {
+    case kWhatPullerNotify:
+    {
+        MediaBuffer *mbuf;
+        CHECK(msg->findPointer("accessUnit", (void**)&mbuf));
+
+        if (mbuf == NULL) {
+            ALOGI("puller (%s) reached EOS",
+                    mIsVideo ? "video" : "audio");
+            mPullerReachedEOS = true;
+        }
+
+        if (mEncoder == NULL) {
+            ALOGV("got msg '%s' after encoder shutdown.",
+                  msg->debugString().c_str());
+
+            if (mbuf != NULL) {
+                mbuf->release();
+            } else {
+                signalEOS();
+            }
+            break;
+        }
+
+        mInputBufferQueue.push_back(mbuf);
+
+        feedEncoderInputBuffers();
+        scheduleDoMoreWork();
+
+        break;
+    }
+    case kWhatEncoderActivity:
+    {
+        mDoMoreWorkPending = false;
+
+        if (mEncoder == NULL) {
+            break;
+        }
+
+        status_t err = doMoreWork();
+
+        if (err == OK) {
+            scheduleDoMoreWork();
+        } else {
+            // reached EOS, or error
+            signalEOS(err);
+        }
+
+        break;
+    }
+    case kWhatStart:
+    {
+        uint32_t replyID;
+        CHECK(msg->senderAwaitsResponse(&replyID));
+
+        sp<RefBase> obj;
+        CHECK(msg->findObject("meta", &obj));
+        MetaData *params = static_cast<MetaData *>(obj.get());
+
+        sp<AMessage> response = new AMessage;
+        response->setInt32("err", onStart(params));
+        response->postReply(replyID);
+        break;
+    }
+    case kWhatStop:
+    {
+        ALOGI("MediaCodecSource (%s) stopping", mIsVideo ? "video" : "audio");
+
+        uint32_t replyID;
+        CHECK(msg->senderAwaitsResponse(&replyID));
+
+        if (reachedEOS()) {
+            // if we already reached EOS, reply and return now
+            ALOGI("MediaCodecSource (%s) already stopped",
+                    mIsVideo ? "video" : "audio");
+            (new AMessage)->postReply(replyID);
+            break;
+        }
+
+        mStopReplyIDQueue.push_back(replyID);
+        if (mStopping) {
+            // nothing to do if we're already stopping, reply will be posted
+            // to all when we're stopped.
+            break;
+        }
+
+        mStopping = true;
+
+        // if using surface, signal source EOS and wait for EOS to come back.
+        // otherwise, release encoder and post EOS if haven't done already
+        if (mFlags & FLAG_USE_SURFACE_INPUT) {
+            mEncoder->signalEndOfInputStream();
+        } else {
+            CHECK(mPuller != NULL);
+            mPuller->stopAsync();
+            signalEOS();
+        }
+        break;
+    }
+    case kWhatPause:
+    {
+        if (mFlags && FLAG_USE_SURFACE_INPUT) {
+            suspend();
+        } else {
+            CHECK(mPuller != NULL);
+            mPuller->pause();
+        }
+        break;
+    }
+    default:
+        TRESPASS();
+    }
+}
+
+} // namespace android
diff --git a/media/libstagefright/MetaData.cpp b/media/libstagefright/MetaData.cpp
index 7b60afc..74234a6 100644
--- a/media/libstagefright/MetaData.cpp
+++ b/media/libstagefright/MetaData.cpp
@@ -16,6 +16,7 @@
 
 //#define LOG_NDEBUG 0
 #define LOG_TAG "MetaData"
+#include <inttypes.h>
 #include <utils/Log.h>
 
 #include <stdlib.h>
@@ -221,6 +222,16 @@
     return true;
 }
 
+bool MetaData::hasData(uint32_t key) const {
+    ssize_t i = mItems.indexOfKey(key);
+
+    if (i < 0) {
+        return false;
+    }
+
+    return true;
+}
+
 MetaData::typed_data::typed_data()
     : mType(0),
       mSize(0) {
@@ -297,7 +308,7 @@
     const void *data = storage();
     switch(mType) {
         case TYPE_NONE:
-            out = String8::format("no type, size %d)", mSize);
+            out = String8::format("no type, size %zu)", mSize);
             break;
         case TYPE_C_STRING:
             out = String8::format("(char*) %s", (const char *)data);
@@ -306,7 +317,7 @@
             out = String8::format("(int32_t) %d", *(int32_t *)data);
             break;
         case TYPE_INT64:
-            out = String8::format("(int64_t) %lld", *(int64_t *)data);
+            out = String8::format("(int64_t) %" PRId64, *(int64_t *)data);
             break;
         case TYPE_FLOAT:
             out = String8::format("(float) %f", *(float *)data);
@@ -323,7 +334,7 @@
         }
 
         default:
-            out = String8::format("(unknown type %d, size %d)", mType, mSize);
+            out = String8::format("(unknown type %d, size %zu)", mType, mSize);
             if (mSize <= 48) { // if it's less than three lines of hex data, dump it
                 AString foo;
                 hexdump(data, mSize, 0, &foo);
diff --git a/media/libstagefright/NuCachedSource2.cpp b/media/libstagefright/NuCachedSource2.cpp
index 05e599b..72ea32d 100644
--- a/media/libstagefright/NuCachedSource2.cpp
+++ b/media/libstagefright/NuCachedSource2.cpp
@@ -213,7 +213,14 @@
 
     mLooper->setName("NuCachedSource2");
     mLooper->registerHandler(mReflector);
-    mLooper->start();
+
+    // Since it may not be obvious why our looper thread needs to be
+    // able to call into java since it doesn't appear to do so at all...
+    // IMediaHTTPConnection may be (and most likely is) implemented in JAVA
+    // and a local JAVA IBinder will call directly into JNI methods.
+    // So whenever we call DataSource::readAt it may end up in a call to
+    // IMediaHTTPConnection::readAt and therefore call back into JAVA.
+    mLooper->start(false /* runOnCallingThread */, true /* canCallJava */);
 
     Mutex::Autolock autoLock(mLock);
     (new AMessage(kWhatFetchMore, mReflector->id()))->post();
@@ -326,7 +333,7 @@
             mNumRetriesLeft = 0;
         }
 
-        ALOGE("source returned error %ld, %d retries left", n, mNumRetriesLeft);
+        ALOGE("source returned error %d, %d retries left", n, mNumRetriesLeft);
         mCache->releasePage(page);
     } else if (n == 0) {
         ALOGI("ERROR_END_OF_STREAM");
@@ -641,7 +648,7 @@
     ssize_t lowwaterMarkKb, highwaterMarkKb;
     int keepAliveSecs;
 
-    if (sscanf(s, "%ld/%ld/%d",
+    if (sscanf(s, "%d/%d/%d",
                &lowwaterMarkKb, &highwaterMarkKb, &keepAliveSecs) != 3) {
         ALOGE("Failed to parse cache parameters from '%s'.", s);
         return;
diff --git a/media/libstagefright/NuMediaExtractor.cpp b/media/libstagefright/NuMediaExtractor.cpp
index 7bc7da2..64f56e9 100644
--- a/media/libstagefright/NuMediaExtractor.cpp
+++ b/media/libstagefright/NuMediaExtractor.cpp
@@ -58,7 +58,9 @@
 }
 
 status_t NuMediaExtractor::setDataSource(
-        const char *path, const KeyedVector<String8, String8> *headers) {
+        const sp<IMediaHTTPService> &httpService,
+        const char *path,
+        const KeyedVector<String8, String8> *headers) {
     Mutex::Autolock autoLock(mLock);
 
     if (mImpl != NULL) {
@@ -66,7 +68,7 @@
     }
 
     sp<DataSource> dataSource =
-        DataSource::CreateFromURI(path, headers);
+        DataSource::CreateFromURI(httpService, path, headers);
 
     if (dataSource == NULL) {
         return -ENOENT;
diff --git a/media/libstagefright/OggExtractor.cpp b/media/libstagefright/OggExtractor.cpp
index 5e79e78..f3eeb03 100644
--- a/media/libstagefright/OggExtractor.cpp
+++ b/media/libstagefright/OggExtractor.cpp
@@ -151,7 +151,7 @@
     return mExtractor->mImpl->getFormat();
 }
 
-status_t OggSource::start(MetaData *params) {
+status_t OggSource::start(MetaData * /* params */) {
     if (mStarted) {
         return INVALID_OPERATION;
     }
@@ -381,7 +381,7 @@
     ssize_t n;
     if ((n = mSource->readAt(offset, header, sizeof(header)))
             < (ssize_t)sizeof(header)) {
-        ALOGV("failed to read %d bytes at offset 0x%016llx, got %ld bytes",
+        ALOGV("failed to read %zu bytes at offset 0x%016llx, got %d bytes",
              sizeof(header), offset, n);
 
         if (n < 0) {
@@ -505,7 +505,7 @@
                     packetSize);
 
             if (n < (ssize_t)packetSize) {
-                ALOGV("failed to read %d bytes at 0x%016llx, got %ld bytes",
+                ALOGV("failed to read %zu bytes at 0x%016llx, got %d bytes",
                      packetSize, dataOffset, n);
                 return ERROR_IO;
             }
@@ -546,7 +546,7 @@
                 buffer = NULL;
             }
 
-            ALOGV("readPage returned %ld", n);
+            ALOGV("readPage returned %d", n);
 
             return n < 0 ? n : (status_t)ERROR_END_OF_STREAM;
         }
@@ -998,7 +998,7 @@
 }
 
 sp<MetaData> OggExtractor::getTrackMetaData(
-        size_t index, uint32_t flags) {
+        size_t index, uint32_t /* flags */) {
     if (index >= 1) {
         return NULL;
     }
diff --git a/media/libstagefright/StagefrightMediaScanner.cpp b/media/libstagefright/StagefrightMediaScanner.cpp
index af8186c..fe20835 100644
--- a/media/libstagefright/StagefrightMediaScanner.cpp
+++ b/media/libstagefright/StagefrightMediaScanner.cpp
@@ -24,6 +24,7 @@
 
 #include <media/stagefright/StagefrightMediaScanner.h>
 
+#include <media/IMediaHTTPService.h>
 #include <media/mediametadataretriever.h>
 #include <private/media/VideoFrame.h>
 
@@ -117,7 +118,7 @@
 }
 
 MediaScanResult StagefrightMediaScanner::processFileInternal(
-        const char *path, const char *mimeType,
+        const char *path, const char * /* mimeType */,
         MediaScannerClient &client) {
     const char *extension = strrchr(path, '.');
 
@@ -147,7 +148,7 @@
     status_t status;
     if (fd < 0) {
         // couldn't open it locally, maybe the media server can?
-        status = mRetriever->setDataSource(path);
+        status = mRetriever->setDataSource(NULL /* httpService */, path);
     } else {
         status = mRetriever->setDataSource(fd, 0, 0x7ffffffffffffffL);
         close(fd);
diff --git a/media/libstagefright/StagefrightMetadataRetriever.cpp b/media/libstagefright/StagefrightMetadataRetriever.cpp
index 19af4fb..9475d05 100644
--- a/media/libstagefright/StagefrightMetadataRetriever.cpp
+++ b/media/libstagefright/StagefrightMetadataRetriever.cpp
@@ -16,10 +16,12 @@
 
 //#define LOG_NDEBUG 0
 #define LOG_TAG "StagefrightMetadataRetriever"
+#include <inttypes.h>
 #include <utils/Log.h>
 
 #include "include/StagefrightMetadataRetriever.h"
 
+#include <media/IMediaHTTPService.h>
 #include <media/stagefright/foundation/ADebug.h>
 #include <media/stagefright/ColorConverter.h>
 #include <media/stagefright/DataSource.h>
@@ -50,7 +52,9 @@
 }
 
 status_t StagefrightMetadataRetriever::setDataSource(
-        const char *uri, const KeyedVector<String8, String8> *headers) {
+        const sp<IMediaHTTPService> &httpService,
+        const char *uri,
+        const KeyedVector<String8, String8> *headers) {
     ALOGV("setDataSource(%s)", uri);
 
     mParsedMetaData = false;
@@ -58,7 +62,7 @@
     delete mAlbumArt;
     mAlbumArt = NULL;
 
-    mSource = DataSource::CreateFromURI(uri, headers);
+    mSource = DataSource::CreateFromURI(httpService, uri, headers);
 
     if (mSource == NULL) {
         ALOGE("Unable to create data source for '%s'.", uri);
@@ -488,7 +492,7 @@
     size_t numTracks = mExtractor->countTracks();
 
     char tmp[32];
-    sprintf(tmp, "%d", numTracks);
+    sprintf(tmp, "%zu", numTracks);
 
     mMetaData.add(METADATA_KEY_NUM_TRACKS, String8(tmp));
 
@@ -545,7 +549,7 @@
     }
 
     // The duration value is a string representing the duration in ms.
-    sprintf(tmp, "%lld", (maxDurationUs + 500) / 1000);
+    sprintf(tmp, "%" PRId64, (maxDurationUs + 500) / 1000);
     mMetaData.add(METADATA_KEY_DURATION, String8(tmp));
 
     if (hasAudio) {
@@ -573,7 +577,7 @@
         if (mSource->getSize(&sourceSize) == OK) {
             int64_t avgBitRate = (int64_t)(sourceSize * 8E6 / maxDurationUs);
 
-            sprintf(tmp, "%lld", avgBitRate);
+            sprintf(tmp, "%" PRId64, avgBitRate);
             mMetaData.add(METADATA_KEY_BITRATE, String8(tmp));
         }
     }
diff --git a/media/libstagefright/SurfaceMediaSource.cpp b/media/libstagefright/SurfaceMediaSource.cpp
index 6b934d4..10c00f4 100644
--- a/media/libstagefright/SurfaceMediaSource.cpp
+++ b/media/libstagefright/SurfaceMediaSource.cpp
@@ -99,8 +99,11 @@
     dump(result, "", buffer, 1024);
 }
 
-void SurfaceMediaSource::dump(String8& result, const char* prefix,
-        char* buffer, size_t SIZE) const
+void SurfaceMediaSource::dump(
+        String8& result,
+        const char* /* prefix */,
+        char* buffer,
+        size_t /* SIZE */) const
 {
     Mutex::Autolock lock(mMutex);
 
@@ -202,6 +205,9 @@
         return OK;
     }
 
+    mStarted = false;
+    mFrameAvailableCondition.signal();
+
     while (mNumPendingBuffers > 0) {
         ALOGI("Still waiting for %d buffers to be returned.",
                 mNumPendingBuffers);
@@ -215,8 +221,6 @@
         mMediaBuffersAvailableCondition.wait(mMutex);
     }
 
-    mStarted = false;
-    mFrameAvailableCondition.signal();
     mMediaBuffersAvailableCondition.signal();
 
     return mBufferQueue->consumerDisconnect();
@@ -269,9 +273,8 @@
             bufferHandle, (*buffer)->range_length(), (*buffer)->range_offset());
 }
 
-status_t SurfaceMediaSource::read( MediaBuffer **buffer,
-                                    const ReadOptions *options)
-{
+status_t SurfaceMediaSource::read(
+        MediaBuffer **buffer, const ReadOptions * /* options */) {
     ALOGV("read");
     Mutex::Autolock lock(mMutex);
 
diff --git a/media/libstagefright/TimedEventQueue.cpp b/media/libstagefright/TimedEventQueue.cpp
index dedd186..3d2eb1f 100644
--- a/media/libstagefright/TimedEventQueue.cpp
+++ b/media/libstagefright/TimedEventQueue.cpp
@@ -318,7 +318,7 @@
 
 void TimedEventQueue::acquireWakeLock_l()
 {
-    if (mWakeLockCount++ == 0) {
+    if (mWakeLockCount == 0) {
         CHECK(mWakeLockToken == 0);
         if (mPowerManager == 0) {
             // use checkService() to avoid blocking if power service is not up yet
@@ -341,21 +341,23 @@
             IPCThreadState::self()->restoreCallingIdentity(token);
             if (status == NO_ERROR) {
                 mWakeLockToken = binder;
+                mWakeLockCount++;
             }
         }
+    } else {
+        mWakeLockCount++;
     }
 }
 
 void TimedEventQueue::releaseWakeLock_l(bool force)
 {
+    if (mWakeLockCount == 0) {
+        return;
+    }
     if (force) {
-        if (mWakeLockCount == 0) {
-            return;
-        }
         // Force wakelock release below by setting reference count to 1.
         mWakeLockCount = 1;
     }
-    CHECK(mWakeLockCount != 0);
     if (--mWakeLockCount == 0) {
         CHECK(mWakeLockToken != 0);
         if (mPowerManager != 0) {
@@ -374,8 +376,8 @@
     mPowerManager.clear();
 }
 
-void TimedEventQueue::PMDeathRecipient::binderDied(const wp<IBinder>& who)
-{
+void TimedEventQueue::PMDeathRecipient::binderDied(
+        const wp<IBinder>& /* who */) {
     mQueue->clearPowerManager();
 }
 
diff --git a/media/libstagefright/VBRISeeker.cpp b/media/libstagefright/VBRISeeker.cpp
index a245f2c..af858b9 100644
--- a/media/libstagefright/VBRISeeker.cpp
+++ b/media/libstagefright/VBRISeeker.cpp
@@ -119,7 +119,7 @@
 
         seeker->mSegments.push(numBytes);
 
-        ALOGV("entry #%d: %d offset 0x%08lx", i, numBytes, offset);
+        ALOGV("entry #%d: %u offset 0x%016llx", i, numBytes, offset);
         offset += numBytes;
     }
 
@@ -160,7 +160,7 @@
         *pos += mSegments.itemAt(segmentIndex++);
     }
 
-    ALOGV("getOffsetForTime %lld us => 0x%08lx", *timeUs, *pos);
+    ALOGV("getOffsetForTime %lld us => 0x%016llx", *timeUs, *pos);
 
     *timeUs = nowUs;
 
diff --git a/media/libstagefright/WAVExtractor.cpp b/media/libstagefright/WAVExtractor.cpp
index 22af6fb..fe9058b 100644
--- a/media/libstagefright/WAVExtractor.cpp
+++ b/media/libstagefright/WAVExtractor.cpp
@@ -127,7 +127,7 @@
 }
 
 sp<MetaData> WAVExtractor::getTrackMetaData(
-        size_t index, uint32_t flags) {
+        size_t index, uint32_t /* flags */) {
     if (mInitCheck != OK || index > 0) {
         return NULL;
     }
@@ -358,7 +358,7 @@
     }
 }
 
-status_t WAVSource::start(MetaData *params) {
+status_t WAVSource::start(MetaData * /* params */) {
     ALOGV("WAVSource::start");
 
     CHECK(!mStarted);
diff --git a/media/libstagefright/avc_utils.cpp b/media/libstagefright/avc_utils.cpp
index b822868..c6ac0da 100644
--- a/media/libstagefright/avc_utils.cpp
+++ b/media/libstagefright/avc_utils.cpp
@@ -251,9 +251,7 @@
     return OK;
 }
 
-static sp<ABuffer> FindNAL(
-        const uint8_t *data, size_t size, unsigned nalType,
-        size_t *stopOffset) {
+static sp<ABuffer> FindNAL(const uint8_t *data, size_t size, unsigned nalType) {
     const uint8_t *nalStart;
     size_t nalSize;
     while (getNextNALUnit(&data, &size, &nalStart, &nalSize, true) == OK) {
@@ -293,7 +291,7 @@
     const uint8_t *data = accessUnit->data();
     size_t size = accessUnit->size();
 
-    sp<ABuffer> seqParamSet = FindNAL(data, size, 7, NULL);
+    sp<ABuffer> seqParamSet = FindNAL(data, size, 7);
     if (seqParamSet == NULL) {
         return NULL;
     }
@@ -303,8 +301,7 @@
     FindAVCDimensions(
             seqParamSet, &width, &height, &sarWidth, &sarHeight);
 
-    size_t stopOffset;
-    sp<ABuffer> picParamSet = FindNAL(data, size, 8, &stopOffset);
+    sp<ABuffer> picParamSet = FindNAL(data, size, 8);
     CHECK(picParamSet != NULL);
 
     size_t csdSize =
diff --git a/media/libstagefright/chromium_http/Android.mk b/media/libstagefright/chromium_http/Android.mk
deleted file mode 100644
index f26f386..0000000
--- a/media/libstagefright/chromium_http/Android.mk
+++ /dev/null
@@ -1,37 +0,0 @@
-LOCAL_PATH:= $(call my-dir)
-
-ifneq ($(TARGET_BUILD_PDK), true)
-include $(CLEAR_VARS)
-
-LOCAL_SRC_FILES:=       \
-        DataUriSource.cpp \
-        ChromiumHTTPDataSource.cpp \
-        support.cpp \
-        chromium_http_stub.cpp
-
-LOCAL_C_INCLUDES:= \
-        $(TOP)/frameworks/av/media/libstagefright \
-        $(TOP)/frameworks/native/include/media/openmax \
-        external/chromium \
-        external/chromium/android
-
-LOCAL_CFLAGS += -Wno-multichar
-
-LOCAL_SHARED_LIBRARIES += \
-        libstlport \
-        libchromium_net \
-        libutils \
-        libcutils \
-        liblog \
-        libstagefright_foundation \
-        libstagefright \
-        libdrmframework
-
-include external/stlport/libstlport.mk
-
-LOCAL_MODULE:= libstagefright_chromium_http
-
-LOCAL_MODULE_TAGS := optional
-
-include $(BUILD_SHARED_LIBRARY)
-endif
diff --git a/media/libstagefright/chromium_http/ChromiumHTTPDataSource.cpp b/media/libstagefright/chromium_http/ChromiumHTTPDataSource.cpp
deleted file mode 100644
index a862d8b..0000000
--- a/media/libstagefright/chromium_http/ChromiumHTTPDataSource.cpp
+++ /dev/null
@@ -1,350 +0,0 @@
-/*
- * Copyright (C) 2011 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "ChromiumHTTPDataSource"
-#include <media/stagefright/foundation/ADebug.h>
-
-#include "include/ChromiumHTTPDataSource.h"
-
-#include <media/stagefright/foundation/ALooper.h>
-#include <media/stagefright/MediaErrors.h>
-
-#include "support.h"
-
-#include <cutils/properties.h> // for property_get
-
-namespace android {
-
-ChromiumHTTPDataSource::ChromiumHTTPDataSource(uint32_t flags)
-    : mFlags(flags),
-      mState(DISCONNECTED),
-      mDelegate(new SfDelegate),
-      mCurrentOffset(0),
-      mIOResult(OK),
-      mContentSize(-1),
-      mDecryptHandle(NULL),
-      mDrmManagerClient(NULL) {
-    mDelegate->setOwner(this);
-}
-
-ChromiumHTTPDataSource::~ChromiumHTTPDataSource() {
-    disconnect();
-
-    delete mDelegate;
-    mDelegate = NULL;
-
-    clearDRMState_l();
-
-    if (mDrmManagerClient != NULL) {
-        delete mDrmManagerClient;
-        mDrmManagerClient = NULL;
-    }
-}
-
-status_t ChromiumHTTPDataSource::connect(
-        const char *uri,
-        const KeyedVector<String8, String8> *headers,
-        off64_t offset) {
-    Mutex::Autolock autoLock(mLock);
-
-    uid_t uid;
-    if (getUID(&uid)) {
-        mDelegate->setUID(uid);
-    }
-
-#if defined(LOG_NDEBUG) && !LOG_NDEBUG
-    LOG_PRI(ANDROID_LOG_VERBOSE, LOG_TAG, "connect on behalf of uid %d", uid);
-#endif
-
-    return connect_l(uri, headers, offset);
-}
-
-status_t ChromiumHTTPDataSource::connect_l(
-        const char *uri,
-        const KeyedVector<String8, String8> *headers,
-        off64_t offset) {
-    if (mState != DISCONNECTED) {
-        disconnect_l();
-    }
-
-#if defined(LOG_NDEBUG) && !LOG_NDEBUG
-    LOG_PRI(ANDROID_LOG_VERBOSE, LOG_TAG,
-                "connect to <URL suppressed> @%lld", offset);
-#endif
-
-    mURI = uri;
-    mContentType = String8("application/octet-stream");
-
-    if (headers != NULL) {
-        mHeaders = *headers;
-    } else {
-        mHeaders.clear();
-    }
-
-    mState = CONNECTING;
-    mContentSize = -1;
-    mCurrentOffset = offset;
-
-    mDelegate->initiateConnection(mURI.c_str(), &mHeaders, offset);
-
-    while (mState == CONNECTING || mState == DISCONNECTING) {
-        mCondition.wait(mLock);
-    }
-
-    return mState == CONNECTED ? OK : mIOResult;
-}
-
-void ChromiumHTTPDataSource::onConnectionEstablished(
-        int64_t contentSize, const char *contentType) {
-    Mutex::Autolock autoLock(mLock);
-
-    if (mState != CONNECTING) {
-        // We may have initiated disconnection.
-        CHECK_EQ(mState, DISCONNECTING);
-        return;
-    }
-
-    mState = CONNECTED;
-    mContentSize = (contentSize < 0) ? -1 : contentSize + mCurrentOffset;
-    mContentType = String8(contentType);
-    mCondition.broadcast();
-}
-
-void ChromiumHTTPDataSource::onConnectionFailed(status_t err) {
-    Mutex::Autolock autoLock(mLock);
-    mState = DISCONNECTED;
-    mCondition.broadcast();
-
-    // mURI.clear();
-
-    mIOResult = err;
-}
-
-void ChromiumHTTPDataSource::disconnect() {
-    Mutex::Autolock autoLock(mLock);
-    disconnect_l();
-}
-
-void ChromiumHTTPDataSource::disconnect_l() {
-    if (mState == DISCONNECTED) {
-        return;
-    }
-
-    mState = DISCONNECTING;
-    mIOResult = -EINTR;
-
-    mDelegate->initiateDisconnect();
-
-    while (mState == DISCONNECTING) {
-        mCondition.wait(mLock);
-    }
-
-    CHECK_EQ((int)mState, (int)DISCONNECTED);
-}
-
-status_t ChromiumHTTPDataSource::initCheck() const {
-    Mutex::Autolock autoLock(mLock);
-
-    return mState == CONNECTED ? OK : NO_INIT;
-}
-
-ssize_t ChromiumHTTPDataSource::readAt(off64_t offset, void *data, size_t size) {
-    Mutex::Autolock autoLock(mLock);
-
-    if (mState != CONNECTED) {
-        return INVALID_OPERATION;
-    }
-
-#if 0
-    char value[PROPERTY_VALUE_MAX];
-    if (property_get("media.stagefright.disable-net", value, 0)
-            && (!strcasecmp(value, "true") || !strcmp(value, "1"))) {
-        LOG_PRI(ANDROID_LOG_INFO, LOG_TAG, "Simulating that the network is down.");
-        disconnect_l();
-        return ERROR_IO;
-    }
-#endif
-
-    if (offset != mCurrentOffset) {
-        AString tmp = mURI;
-        KeyedVector<String8, String8> tmpHeaders = mHeaders;
-
-        disconnect_l();
-
-        status_t err = connect_l(tmp.c_str(), &tmpHeaders, offset);
-
-        if (err != OK) {
-            return err;
-        }
-    }
-
-    mState = READING;
-
-    int64_t startTimeUs = ALooper::GetNowUs();
-
-    mDelegate->initiateRead(data, size);
-
-    while (mState == READING) {
-        mCondition.wait(mLock);
-    }
-
-    if (mIOResult < OK) {
-        return mIOResult;
-    }
-
-    if (mState == CONNECTED) {
-        int64_t delayUs = ALooper::GetNowUs() - startTimeUs;
-
-        // The read operation was successful, mIOResult contains
-        // the number of bytes read.
-        addBandwidthMeasurement(mIOResult, delayUs);
-
-        mCurrentOffset += mIOResult;
-        return mIOResult;
-    }
-
-    return ERROR_IO;
-}
-
-void ChromiumHTTPDataSource::onReadCompleted(ssize_t size) {
-    Mutex::Autolock autoLock(mLock);
-
-    mIOResult = size;
-
-    if (mState == READING) {
-        mState = CONNECTED;
-        mCondition.broadcast();
-    }
-}
-
-status_t ChromiumHTTPDataSource::getSize(off64_t *size) {
-    Mutex::Autolock autoLock(mLock);
-
-    if (mContentSize < 0) {
-        return ERROR_UNSUPPORTED;
-    }
-
-    *size = mContentSize;
-
-    return OK;
-}
-
-uint32_t ChromiumHTTPDataSource::flags() {
-    return kWantsPrefetching | kIsHTTPBasedSource;
-}
-
-// static
-void ChromiumHTTPDataSource::InitiateRead(
-        ChromiumHTTPDataSource *me, void *data, size_t size) {
-    me->initiateRead(data, size);
-}
-
-void ChromiumHTTPDataSource::initiateRead(void *data, size_t size) {
-    mDelegate->initiateRead(data, size);
-}
-
-void ChromiumHTTPDataSource::onDisconnectComplete() {
-    Mutex::Autolock autoLock(mLock);
-    CHECK_EQ((int)mState, (int)DISCONNECTING);
-
-    mState = DISCONNECTED;
-    // mURI.clear();
-    mIOResult = -ENOTCONN;
-
-    mCondition.broadcast();
-}
-
-sp<DecryptHandle> ChromiumHTTPDataSource::DrmInitialization(const char* mime) {
-    Mutex::Autolock autoLock(mLock);
-
-    if (mDrmManagerClient == NULL) {
-        mDrmManagerClient = new DrmManagerClient();
-    }
-
-    if (mDrmManagerClient == NULL) {
-        return NULL;
-    }
-
-    if (mDecryptHandle == NULL) {
-        /* Note if redirect occurs, mUri is the redirect uri instead of the
-         * original one
-         */
-        mDecryptHandle = mDrmManagerClient->openDecryptSession(
-                String8(mURI.c_str()), mime);
-    }
-
-    if (mDecryptHandle == NULL) {
-        delete mDrmManagerClient;
-        mDrmManagerClient = NULL;
-    }
-
-    return mDecryptHandle;
-}
-
-void ChromiumHTTPDataSource::getDrmInfo(
-        sp<DecryptHandle> &handle, DrmManagerClient **client) {
-    Mutex::Autolock autoLock(mLock);
-
-    handle = mDecryptHandle;
-    *client = mDrmManagerClient;
-}
-
-String8 ChromiumHTTPDataSource::getUri() {
-    Mutex::Autolock autoLock(mLock);
-
-    return String8(mURI.c_str());
-}
-
-String8 ChromiumHTTPDataSource::getMIMEType() const {
-    Mutex::Autolock autoLock(mLock);
-
-    return mContentType;
-}
-
-void ChromiumHTTPDataSource::clearDRMState_l() {
-    if (mDecryptHandle != NULL) {
-        // To release mDecryptHandle
-        CHECK(mDrmManagerClient);
-        mDrmManagerClient->closeDecryptSession(mDecryptHandle);
-        mDecryptHandle = NULL;
-    }
-}
-
-status_t ChromiumHTTPDataSource::reconnectAtOffset(off64_t offset) {
-    Mutex::Autolock autoLock(mLock);
-
-    if (mURI.empty()) {
-        return INVALID_OPERATION;
-    }
-
-    LOG_PRI(ANDROID_LOG_INFO, LOG_TAG, "Reconnecting...");
-    status_t err = connect_l(mURI.c_str(), &mHeaders, offset);
-    if (err != OK) {
-        LOG_PRI(ANDROID_LOG_INFO, LOG_TAG, "Reconnect failed w/ err 0x%08x", err);
-    }
-
-    return err;
-}
-
-// static
-status_t ChromiumHTTPDataSource::UpdateProxyConfig(
-        const char *host, int32_t port, const char *exclusionList) {
-    return SfDelegate::UpdateProxyConfig(host, port, exclusionList);
-}
-
-}  // namespace android
-
diff --git a/media/libstagefright/chromium_http/DataUriSource.cpp b/media/libstagefright/chromium_http/DataUriSource.cpp
deleted file mode 100644
index ecf3fa1..0000000
--- a/media/libstagefright/chromium_http/DataUriSource.cpp
+++ /dev/null
@@ -1,68 +0,0 @@
-/*
- * Copyright (C) 2011 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#include <include/DataUriSource.h>
-
-#include <net/base/data_url.h>
-#include <googleurl/src/gurl.h>
-
-
-namespace android {
-
-DataUriSource::DataUriSource(const char *uri) :
-    mDataUri(uri),
-    mInited(NO_INIT) {
-
-    // Copy1: const char *uri -> String8 mDataUri.
-    std::string mimeTypeStr, unusedCharsetStr, dataStr;
-    // Copy2: String8 mDataUri -> std::string
-    const bool ret = net::DataURL::Parse(
-            GURL(std::string(mDataUri.string())),
-            &mimeTypeStr, &unusedCharsetStr, &dataStr);
-    // Copy3: std::string dataStr -> AString mData
-    mData.setTo(dataStr.data(), dataStr.length());
-    mInited = ret ? OK : UNKNOWN_ERROR;
-
-    // The chromium data url implementation defaults to using "text/plain"
-    // if no mime type is specified. We prefer to leave this unspecified
-    // instead, since the mime type is sniffed in most cases.
-    if (mimeTypeStr != "text/plain") {
-        mMimeType = mimeTypeStr.c_str();
-    }
-}
-
-ssize_t DataUriSource::readAt(off64_t offset, void *out, size_t size) {
-    if (mInited != OK) {
-        return mInited;
-    }
-
-    const off64_t length = mData.size();
-    if (offset >= length) {
-        return UNKNOWN_ERROR;
-    }
-
-    const char *dataBuf = mData.c_str();
-    const size_t bytesToCopy =
-            offset + size >= length ? (length - offset) : size;
-
-    if (bytesToCopy > 0) {
-        memcpy(out, dataBuf + offset, bytesToCopy);
-    }
-
-    return bytesToCopy;
-}
-
-}  // namespace android
diff --git a/media/libstagefright/chromium_http/chromium_http_stub.cpp b/media/libstagefright/chromium_http/chromium_http_stub.cpp
deleted file mode 100644
index 289f6de..0000000
--- a/media/libstagefright/chromium_http/chromium_http_stub.cpp
+++ /dev/null
@@ -1,38 +0,0 @@
-/*
- * Copyright (C) 2012 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#include <dlfcn.h>
-
-#include <include/chromium_http_stub.h>
-#include <include/ChromiumHTTPDataSource.h>
-#include <include/DataUriSource.h>
-
-namespace android {
-
-HTTPBase *createChromiumHTTPDataSource(uint32_t flags) {
-    return new ChromiumHTTPDataSource(flags);
-}
-
-status_t UpdateChromiumHTTPDataSourceProxyConfig(
-        const char *host, int32_t port, const char *exclusionList) {
-    return ChromiumHTTPDataSource::UpdateProxyConfig(host, port, exclusionList);
-}
-
-DataSource *createDataUriSource(const char *uri) {
-    return new DataUriSource(uri);
-}
-
-}
diff --git a/media/libstagefright/chromium_http/support.cpp b/media/libstagefright/chromium_http/support.cpp
deleted file mode 100644
index 0a8e3e3..0000000
--- a/media/libstagefright/chromium_http/support.cpp
+++ /dev/null
@@ -1,558 +0,0 @@
-/*
- * Copyright (C) 2011 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "ChromiumHTTPDataSourceSupport"
-#include <utils/Log.h>
-
-#include <media/stagefright/foundation/AString.h>
-
-#include "support.h"
-
-#include "android/net/android_network_library_impl.h"
-#include "base/logging.h"
-#include "base/threading/thread.h"
-#include "net/base/cert_verifier.h"
-#include "net/base/cookie_monster.h"
-#include "net/base/host_resolver.h"
-#include "net/base/ssl_config_service.h"
-#include "net/http/http_auth_handler_factory.h"
-#include "net/http/http_cache.h"
-#include "net/proxy/proxy_config_service_android.h"
-
-#include "include/ChromiumHTTPDataSource.h"
-
-#include <cutils/log.h>
-#include <media/stagefright/MediaErrors.h>
-#include <media/stagefright/Utils.h>
-#include <string>
-
-namespace android {
-
-static Mutex gNetworkThreadLock;
-static base::Thread *gNetworkThread = NULL;
-static scoped_refptr<SfRequestContext> gReqContext;
-static scoped_ptr<net::NetworkChangeNotifier> gNetworkChangeNotifier;
-
-bool logMessageHandler(
-        int severity,
-        const char* file,
-        int line,
-        size_t message_start,
-        const std::string& str) {
-    int androidSeverity = ANDROID_LOG_VERBOSE;
-    switch(severity) {
-    case logging::LOG_FATAL:
-        androidSeverity = ANDROID_LOG_FATAL;
-        break;
-    case logging::LOG_ERROR_REPORT:
-    case logging::LOG_ERROR:
-        androidSeverity = ANDROID_LOG_ERROR;
-        break;
-    case logging::LOG_WARNING:
-        androidSeverity = ANDROID_LOG_WARN;
-        break;
-    default:
-        androidSeverity = ANDROID_LOG_VERBOSE;
-        break;
-    }
-    android_printLog(androidSeverity, "chromium-libstagefright",
-                    "%s:%d: %s", file, line, str.c_str());
-    return false;
-}
-
-struct AutoPrioritySaver {
-    AutoPrioritySaver()
-        : mTID(androidGetTid()),
-          mPrevPriority(androidGetThreadPriority(mTID)) {
-        androidSetThreadPriority(mTID, ANDROID_PRIORITY_NORMAL);
-    }
-
-    ~AutoPrioritySaver() {
-        androidSetThreadPriority(mTID, mPrevPriority);
-    }
-
-private:
-    pid_t mTID;
-    int mPrevPriority;
-
-    DISALLOW_EVIL_CONSTRUCTORS(AutoPrioritySaver);
-};
-
-static void InitializeNetworkThreadIfNecessary() {
-    Mutex::Autolock autoLock(gNetworkThreadLock);
-
-    if (gNetworkThread == NULL) {
-        // Make sure any threads spawned by the chromium framework are
-        // running at normal priority instead of inheriting this thread's.
-        AutoPrioritySaver saver;
-
-        gNetworkThread = new base::Thread("network");
-        base::Thread::Options options;
-        options.message_loop_type = MessageLoop::TYPE_IO;
-        CHECK(gNetworkThread->StartWithOptions(options));
-
-        gReqContext = new SfRequestContext;
-
-        gNetworkChangeNotifier.reset(net::NetworkChangeNotifier::Create());
-
-        net::AndroidNetworkLibrary::RegisterSharedInstance(
-                new SfNetworkLibrary);
-        logging::SetLogMessageHandler(logMessageHandler);
-    }
-}
-
-static void MY_LOGI(const char *s) {
-    LOG_PRI(ANDROID_LOG_INFO, LOG_TAG, "%s", s);
-}
-
-static void MY_LOGV(const char *s) {
-#if !defined(LOG_NDEBUG) || LOG_NDEBUG == 0
-    LOG_PRI(ANDROID_LOG_VERBOSE, LOG_TAG, "%s", s);
-#endif
-}
-
-SfNetLog::SfNetLog()
-    : mNextID(1) {
-}
-
-void SfNetLog::AddEntry(
-        EventType type,
-        const base::TimeTicks &time,
-        const Source &source,
-        EventPhase phase,
-        EventParameters *params) {
-#if 0
-    MY_LOGI(StringPrintf(
-                "AddEntry time=%s type=%s source=%s phase=%s\n",
-                TickCountToString(time).c_str(),
-                EventTypeToString(type),
-                SourceTypeToString(source.type),
-                EventPhaseToString(phase)).c_str());
-#endif
-}
-
-uint32 SfNetLog::NextID() {
-    return mNextID++;
-}
-
-net::NetLog::LogLevel SfNetLog::GetLogLevel() const {
-    return LOG_BASIC;
-}
-
-////////////////////////////////////////////////////////////////////////////////
-
-SfRequestContext::SfRequestContext() {
-    mUserAgent = MakeUserAgent().c_str();
-
-    set_net_log(new SfNetLog());
-
-    set_host_resolver(
-        net::CreateSystemHostResolver(
-                net::HostResolver::kDefaultParallelism,
-                NULL /* resolver_proc */,
-                net_log()));
-
-    set_ssl_config_service(
-        net::SSLConfigService::CreateSystemSSLConfigService());
-
-    mProxyConfigService = new net::ProxyConfigServiceAndroid;
-
-    set_proxy_service(net::ProxyService::CreateWithoutProxyResolver(
-        mProxyConfigService, net_log()));
-
-    set_http_transaction_factory(new net::HttpCache(
-            host_resolver(),
-            new net::CertVerifier(),
-            dnsrr_resolver(),
-            dns_cert_checker(),
-            proxy_service(),
-            ssl_config_service(),
-            net::HttpAuthHandlerFactory::CreateDefault(host_resolver()),
-            network_delegate(),
-            net_log(),
-            NULL));  // backend_factory
-
-    set_cookie_store(new net::CookieMonster(NULL, NULL));
-}
-
-const std::string &SfRequestContext::GetUserAgent(const GURL &url) const {
-    return mUserAgent;
-}
-
-status_t SfRequestContext::updateProxyConfig(
-        const char *host, int32_t port, const char *exclusionList) {
-    Mutex::Autolock autoLock(mProxyConfigLock);
-
-    if (host == NULL || *host == '\0') {
-        MY_LOGV("updateProxyConfig NULL");
-
-        std::string proxy;
-        std::string exList;
-        mProxyConfigService->UpdateProxySettings(proxy, exList);
-    } else {
-#if !defined(LOG_NDEBUG) || LOG_NDEBUG == 0
-        LOG_PRI(ANDROID_LOG_VERBOSE, LOG_TAG,
-                "updateProxyConfig %s:%d, exclude '%s'",
-                host, port, exclusionList);
-#endif
-
-        std::string proxy = StringPrintf("%s:%d", host, port).c_str();
-        std::string exList = exclusionList;
-        mProxyConfigService->UpdateProxySettings(proxy, exList);
-    }
-
-    return OK;
-}
-
-////////////////////////////////////////////////////////////////////////////////
-
-SfNetworkLibrary::SfNetworkLibrary() {}
-
-SfNetworkLibrary::VerifyResult SfNetworkLibrary::VerifyX509CertChain(
-        const std::vector<std::string>& cert_chain,
-        const std::string& hostname,
-        const std::string& auth_type) {
-    return VERIFY_OK;
-}
-
-////////////////////////////////////////////////////////////////////////////////
-
-SfDelegate::SfDelegate()
-    : mOwner(NULL),
-      mURLRequest(NULL),
-      mReadBuffer(new net::IOBufferWithSize(8192)),
-      mNumBytesRead(0),
-      mNumBytesTotal(0),
-      mDataDestination(NULL),
-      mAtEOS(false) {
-    InitializeNetworkThreadIfNecessary();
-}
-
-SfDelegate::~SfDelegate() {
-    CHECK(mURLRequest == NULL);
-}
-
-// static
-status_t SfDelegate::UpdateProxyConfig(
-        const char *host, int32_t port, const char *exclusionList) {
-    InitializeNetworkThreadIfNecessary();
-
-    return gReqContext->updateProxyConfig(host, port, exclusionList);
-}
-
-void SfDelegate::setOwner(ChromiumHTTPDataSource *owner) {
-    mOwner = owner;
-}
-
-void SfDelegate::setUID(uid_t uid) {
-    gReqContext->setUID(uid);
-}
-
-bool SfDelegate::getUID(uid_t *uid) const {
-    return gReqContext->getUID(uid);
-}
-
-void SfDelegate::OnReceivedRedirect(
-            net::URLRequest *request, const GURL &new_url, bool *defer_redirect) {
-    MY_LOGV("OnReceivedRedirect");
-}
-
-void SfDelegate::OnAuthRequired(
-            net::URLRequest *request, net::AuthChallengeInfo *auth_info) {
-    MY_LOGV("OnAuthRequired");
-
-    inherited::OnAuthRequired(request, auth_info);
-}
-
-void SfDelegate::OnCertificateRequested(
-            net::URLRequest *request, net::SSLCertRequestInfo *cert_request_info) {
-    MY_LOGV("OnCertificateRequested");
-
-    inherited::OnCertificateRequested(request, cert_request_info);
-}
-
-void SfDelegate::OnSSLCertificateError(
-            net::URLRequest *request, int cert_error, net::X509Certificate *cert) {
-    fprintf(stderr, "OnSSLCertificateError cert_error=%d\n", cert_error);
-
-    inherited::OnSSLCertificateError(request, cert_error, cert);
-}
-
-void SfDelegate::OnGetCookies(net::URLRequest *request, bool blocked_by_policy) {
-    MY_LOGV("OnGetCookies");
-}
-
-void SfDelegate::OnSetCookie(
-        net::URLRequest *request,
-        const std::string &cookie_line,
-        const net::CookieOptions &options,
-        bool blocked_by_policy) {
-    MY_LOGV("OnSetCookie");
-}
-
-void SfDelegate::OnResponseStarted(net::URLRequest *request) {
-    if (request->status().status() != net::URLRequestStatus::SUCCESS) {
-        MY_LOGI(StringPrintf(
-                    "Request failed with status %d and os_error %d",
-                    request->status().status(),
-                    request->status().os_error()).c_str());
-
-        delete mURLRequest;
-        mURLRequest = NULL;
-
-        mOwner->onConnectionFailed(ERROR_IO);
-        return;
-    } else if (mRangeRequested && request->GetResponseCode() != 206) {
-        MY_LOGI(StringPrintf(
-                    "We requested a content range, but server didn't "
-                    "support that. (responded with %d)",
-                    request->GetResponseCode()).c_str());
-
-        delete mURLRequest;
-        mURLRequest = NULL;
-
-        mOwner->onConnectionFailed(-EPIPE);
-        return;
-    } else if ((request->GetResponseCode() / 100) != 2) {
-        MY_LOGI(StringPrintf(
-                    "Server responded with http status %d",
-                    request->GetResponseCode()).c_str());
-
-        delete mURLRequest;
-        mURLRequest = NULL;
-
-        mOwner->onConnectionFailed(ERROR_IO);
-        return;
-    }
-
-    MY_LOGV("OnResponseStarted");
-
-    std::string headers;
-    request->GetAllResponseHeaders(&headers);
-
-    MY_LOGV(StringPrintf("response headers: %s", headers.c_str()).c_str());
-
-    std::string contentType;
-    request->GetResponseHeaderByName("Content-Type", &contentType);
-
-    mOwner->onConnectionEstablished(
-            request->GetExpectedContentSize(), contentType.c_str());
-}
-
-void SfDelegate::OnReadCompleted(net::URLRequest *request, int bytes_read) {
-    if (bytes_read == -1) {
-        MY_LOGI(StringPrintf(
-                    "OnReadCompleted, read failed, status %d",
-                    request->status().status()).c_str());
-
-        mOwner->onReadCompleted(ERROR_IO);
-        return;
-    }
-
-    MY_LOGV(StringPrintf("OnReadCompleted, read %d bytes", bytes_read).c_str());
-
-    if (bytes_read < 0) {
-        MY_LOGI(StringPrintf(
-                    "Read failed w/ status %d\n",
-                    request->status().status()).c_str());
-
-        mOwner->onReadCompleted(ERROR_IO);
-        return;
-    } else if (bytes_read == 0) {
-        mAtEOS = true;
-        mOwner->onReadCompleted(mNumBytesRead);
-        return;
-    }
-
-    CHECK_GT(bytes_read, 0);
-    CHECK_LE(mNumBytesRead + bytes_read, mNumBytesTotal);
-
-    memcpy((uint8_t *)mDataDestination + mNumBytesRead,
-           mReadBuffer->data(),
-           bytes_read);
-
-    mNumBytesRead += bytes_read;
-
-    readMore(request);
-}
-
-void SfDelegate::readMore(net::URLRequest *request) {
-    while (mNumBytesRead < mNumBytesTotal) {
-        size_t copy = mNumBytesTotal - mNumBytesRead;
-        if (copy > mReadBuffer->size()) {
-            copy = mReadBuffer->size();
-        }
-
-        int n;
-        if (request->Read(mReadBuffer, copy, &n)) {
-            MY_LOGV(StringPrintf("Read %d bytes directly.", n).c_str());
-
-            CHECK_LE((size_t)n, copy);
-
-            memcpy((uint8_t *)mDataDestination + mNumBytesRead,
-                   mReadBuffer->data(),
-                   n);
-
-            mNumBytesRead += n;
-
-            if (n == 0) {
-                mAtEOS = true;
-                break;
-            }
-        } else {
-            MY_LOGV("readMore pending read");
-
-            if (request->status().status() != net::URLRequestStatus::IO_PENDING) {
-                MY_LOGI(StringPrintf(
-                            "Direct read failed w/ status %d\n",
-                            request->status().status()).c_str());
-
-                mOwner->onReadCompleted(ERROR_IO);
-                return;
-            }
-
-            return;
-        }
-    }
-
-    mOwner->onReadCompleted(mNumBytesRead);
-}
-
-void SfDelegate::initiateConnection(
-        const char *uri,
-        const KeyedVector<String8, String8> *headers,
-        off64_t offset) {
-    GURL url(uri);
-
-    MessageLoop *loop = gNetworkThread->message_loop();
-    loop->PostTask(
-            FROM_HERE,
-            NewRunnableFunction(
-                &SfDelegate::OnInitiateConnectionWrapper,
-                this,
-                url,
-                headers,
-                offset));
-
-}
-
-// static
-void SfDelegate::OnInitiateConnectionWrapper(
-        SfDelegate *me, GURL url,
-        const KeyedVector<String8, String8> *headers,
-        off64_t offset) {
-    me->onInitiateConnection(url, headers, offset);
-}
-
-void SfDelegate::onInitiateConnection(
-        const GURL &url,
-        const KeyedVector<String8, String8> *extra,
-        off64_t offset) {
-    CHECK(mURLRequest == NULL);
-
-    mURLRequest = new net::URLRequest(url, this);
-    mAtEOS = false;
-
-    mRangeRequested = false;
-
-    if (offset != 0 || extra != NULL) {
-        net::HttpRequestHeaders headers =
-            mURLRequest->extra_request_headers();
-
-        if (offset != 0) {
-            headers.AddHeaderFromString(
-                    StringPrintf("Range: bytes=%lld-", offset).c_str());
-
-            mRangeRequested = true;
-        }
-
-        if (extra != NULL) {
-            for (size_t i = 0; i < extra->size(); ++i) {
-                AString s;
-                s.append(extra->keyAt(i).string());
-                s.append(": ");
-                s.append(extra->valueAt(i).string());
-
-                headers.AddHeaderFromString(s.c_str());
-            }
-        }
-
-        mURLRequest->SetExtraRequestHeaders(headers);
-    }
-
-    mURLRequest->set_context(gReqContext);
-
-    mURLRequest->Start();
-}
-
-void SfDelegate::initiateDisconnect() {
-    MessageLoop *loop = gNetworkThread->message_loop();
-    loop->PostTask(
-            FROM_HERE,
-            NewRunnableFunction(
-                &SfDelegate::OnInitiateDisconnectWrapper, this));
-}
-
-// static
-void SfDelegate::OnInitiateDisconnectWrapper(SfDelegate *me) {
-    me->onInitiateDisconnect();
-}
-
-void SfDelegate::onInitiateDisconnect() {
-    if (mURLRequest == NULL) {
-        return;
-    }
-
-    mURLRequest->Cancel();
-
-    delete mURLRequest;
-    mURLRequest = NULL;
-
-    mOwner->onDisconnectComplete();
-}
-
-void SfDelegate::initiateRead(void *data, size_t size) {
-    MessageLoop *loop = gNetworkThread->message_loop();
-    loop->PostTask(
-            FROM_HERE,
-            NewRunnableFunction(
-                &SfDelegate::OnInitiateReadWrapper, this, data, size));
-}
-
-// static
-void SfDelegate::OnInitiateReadWrapper(
-        SfDelegate *me, void *data, size_t size) {
-    me->onInitiateRead(data, size);
-}
-
-void SfDelegate::onInitiateRead(void *data, size_t size) {
-    CHECK(mURLRequest != NULL);
-
-    mNumBytesRead = 0;
-    mNumBytesTotal = size;
-    mDataDestination = data;
-
-    if (mAtEOS) {
-        mOwner->onReadCompleted(0);
-        return;
-    }
-
-    readMore(mURLRequest);
-}
-
-}  // namespace android
-
diff --git a/media/libstagefright/chromium_http/support.h b/media/libstagefright/chromium_http/support.h
deleted file mode 100644
index 975a1d3..0000000
--- a/media/libstagefright/chromium_http/support.h
+++ /dev/null
@@ -1,178 +0,0 @@
-/*
- * Copyright (C) 2011 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef SUPPORT_H_
-
-#define SUPPORT_H_
-
-#include <assert.h>
-
-#include "net/base/net_log.h"
-#include "net/url_request/url_request.h"
-#include "net/url_request/url_request_context.h"
-#include "net/base/android_network_library.h"
-#include "net/base/io_buffer.h"
-
-#include <utils/KeyedVector.h>
-#include <utils/Mutex.h>
-#include <utils/String8.h>
-
-namespace net {
-    struct ProxyConfigServiceAndroid;
-};
-
-namespace android {
-
-struct SfNetLog : public net::NetLog {
-    SfNetLog();
-
-    virtual void AddEntry(
-            EventType type,
-            const base::TimeTicks &time,
-            const Source &source,
-            EventPhase phase,
-            EventParameters *params);
-
-    virtual uint32 NextID();
-    virtual LogLevel GetLogLevel() const;
-
-private:
-    uint32 mNextID;
-
-    DISALLOW_EVIL_CONSTRUCTORS(SfNetLog);
-};
-
-struct SfRequestContext : public net::URLRequestContext {
-    SfRequestContext();
-
-    virtual const std::string &GetUserAgent(const GURL &url) const;
-
-    status_t updateProxyConfig(
-            const char *host, int32_t port, const char *exclusionList);
-
-private:
-    Mutex mProxyConfigLock;
-
-    std::string mUserAgent;
-    net::ProxyConfigServiceAndroid *mProxyConfigService;
-
-    DISALLOW_EVIL_CONSTRUCTORS(SfRequestContext);
-};
-
-// This is required for https support, we don't really verify certificates,
-// we accept anything...
-struct SfNetworkLibrary : public net::AndroidNetworkLibrary {
-    SfNetworkLibrary();
-
-    virtual VerifyResult VerifyX509CertChain(
-            const std::vector<std::string>& cert_chain,
-            const std::string& hostname,
-            const std::string& auth_type);
-
-private:
-    DISALLOW_EVIL_CONSTRUCTORS(SfNetworkLibrary);
-};
-
-struct ChromiumHTTPDataSource;
-
-struct SfDelegate : public net::URLRequest::Delegate {
-    SfDelegate();
-    virtual ~SfDelegate();
-
-    void initiateConnection(
-            const char *uri,
-            const KeyedVector<String8, String8> *headers,
-            off64_t offset);
-
-    void initiateDisconnect();
-    void initiateRead(void *data, size_t size);
-
-    void setOwner(ChromiumHTTPDataSource *mOwner);
-
-    // Gets the UID of the calling process
-    bool getUID(uid_t *uid) const;
-
-    void setUID(uid_t uid);
-
-    virtual void OnReceivedRedirect(
-            net::URLRequest *request, const GURL &new_url, bool *defer_redirect);
-
-    virtual void OnAuthRequired(
-            net::URLRequest *request, net::AuthChallengeInfo *auth_info);
-
-    virtual void OnCertificateRequested(
-            net::URLRequest *request, net::SSLCertRequestInfo *cert_request_info);
-
-    virtual void OnSSLCertificateError(
-            net::URLRequest *request, int cert_error, net::X509Certificate *cert);
-
-    virtual void OnGetCookies(net::URLRequest *request, bool blocked_by_policy);
-
-    virtual void OnSetCookie(
-            net::URLRequest *request,
-            const std::string &cookie_line,
-            const net::CookieOptions &options,
-            bool blocked_by_policy);
-
-    virtual void OnResponseStarted(net::URLRequest *request);
-
-    virtual void OnReadCompleted(net::URLRequest *request, int bytes_read);
-
-    static status_t UpdateProxyConfig(
-            const char *host, int32_t port, const char *exclusionList);
-
-private:
-    typedef Delegate inherited;
-
-    ChromiumHTTPDataSource *mOwner;
-
-    net::URLRequest *mURLRequest;
-    scoped_refptr<net::IOBufferWithSize> mReadBuffer;
-
-    size_t mNumBytesRead;
-    size_t mNumBytesTotal;
-    void *mDataDestination;
-
-    bool mRangeRequested;
-    bool mAtEOS;
-
-    void readMore(net::URLRequest *request);
-
-    static void OnInitiateConnectionWrapper(
-            SfDelegate *me,
-            GURL url,
-            const KeyedVector<String8, String8> *headers,
-            off64_t offset);
-
-    static void OnInitiateDisconnectWrapper(SfDelegate *me);
-
-    static void OnInitiateReadWrapper(
-            SfDelegate *me, void *data, size_t size);
-
-    void onInitiateConnection(
-            const GURL &url,
-            const KeyedVector<String8, String8> *headers,
-            off64_t offset);
-
-    void onInitiateDisconnect();
-    void onInitiateRead(void *data, size_t size);
-
-    DISALLOW_EVIL_CONSTRUCTORS(SfDelegate);
-};
-
-}  // namespace android
-
-#endif  // SUPPORT_H_
diff --git a/media/libstagefright/chromium_http_stub.cpp b/media/libstagefright/chromium_http_stub.cpp
deleted file mode 100644
index ed8a878..0000000
--- a/media/libstagefright/chromium_http_stub.cpp
+++ /dev/null
@@ -1,102 +0,0 @@
-/*
- * Copyright (C) 2012 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#include <dlfcn.h>
-
-#include <media/stagefright/DataSource.h>
-
-#include "include/chromium_http_stub.h"
-#include "include/HTTPBase.h"
-
-namespace android {
-
-static bool gFirst = true;
-static void *gHandle;
-static Mutex gLibMutex;
-
-HTTPBase *(*gLib_createChromiumHTTPDataSource)(uint32_t flags);
-DataSource *(*gLib_createDataUriSource)(const char *uri);
-
-status_t (*gLib_UpdateChromiumHTTPDataSourceProxyConfig)(
-        const char *host, int32_t port, const char *exclusionList);
-
-static bool load_libstagefright_chromium_http() {
-    Mutex::Autolock autoLock(gLibMutex);
-    void *sym;
-
-    if (!gFirst) {
-        return (gHandle != NULL);
-    }
-
-    gFirst = false;
-
-    gHandle = dlopen("libstagefright_chromium_http.so", RTLD_NOW);
-    if (gHandle == NULL) {
-        return false;
-    }
-
-    sym = dlsym(gHandle, "createChromiumHTTPDataSource");
-    if (sym == NULL) {
-        gHandle = NULL;
-        return false;
-    }
-    gLib_createChromiumHTTPDataSource = (HTTPBase *(*)(uint32_t))sym;
-
-    sym = dlsym(gHandle, "createDataUriSource");
-    if (sym == NULL) {
-        gHandle = NULL;
-        return false;
-    }
-    gLib_createDataUriSource = (DataSource *(*)(const char *))sym;
-
-    sym = dlsym(gHandle, "UpdateChromiumHTTPDataSourceProxyConfig");
-    if (sym == NULL) {
-        gHandle = NULL;
-        return false;
-    }
-    gLib_UpdateChromiumHTTPDataSourceProxyConfig =
-        (status_t (*)(const char *, int32_t, const char *))sym;
-
-    return true;
-}
-
-HTTPBase *createChromiumHTTPDataSource(uint32_t flags) {
-    if (!load_libstagefright_chromium_http()) {
-        return NULL;
-    }
-
-    return gLib_createChromiumHTTPDataSource(flags);
-}
-
-status_t UpdateChromiumHTTPDataSourceProxyConfig(
-        const char *host, int32_t port, const char *exclusionList) {
-    if (!load_libstagefright_chromium_http()) {
-        return INVALID_OPERATION;
-    }
-
-    return gLib_UpdateChromiumHTTPDataSourceProxyConfig(
-            host, port, exclusionList);
-}
-
-DataSource *createDataUriSource(const char *uri) {
-    if (!load_libstagefright_chromium_http()) {
-        return NULL;
-    }
-
-    return gLib_createDataUriSource(uri);
-}
-
-}
diff --git a/media/libstagefright/codecs/aacdec/Android.mk b/media/libstagefright/codecs/aacdec/Android.mk
index ffa64f9..49ff238 100644
--- a/media/libstagefright/codecs/aacdec/Android.mk
+++ b/media/libstagefright/codecs/aacdec/Android.mk
@@ -17,6 +17,8 @@
 
 LOCAL_CFLAGS :=
 
+LOCAL_CFLAGS += -Werror
+
 LOCAL_STATIC_LIBRARIES := libFraunhoferAAC
 
 LOCAL_SHARED_LIBRARIES := \
diff --git a/media/libstagefright/codecs/aacdec/SoftAAC2.cpp b/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
index 2f5eff4..4ac8999 100644
--- a/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
+++ b/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
@@ -377,7 +377,7 @@
                 mNumSamplesOutput = 0;
             }
 
-            if (mIsADTS) {
+            if (mIsADTS && inHeader->nFilledLen) {
                 size_t adtsHeaderSize = 0;
                 // skip 30 bits, aac_frame_length follows.
                 // ssssssss ssssiiip ppffffPc ccohCCll llllllll lll?????
diff --git a/media/libstagefright/codecs/aacenc/Android.mk b/media/libstagefright/codecs/aacenc/Android.mk
index 057c69b..58ec3ba 100644
--- a/media/libstagefright/codecs/aacenc/Android.mk
+++ b/media/libstagefright/codecs/aacenc/Android.mk
@@ -82,6 +82,8 @@
 LOCAL_C_INCLUDES += $(LOCAL_PATH)/src/asm/ARMV7
 endif
 
+LOCAL_CFLAGS += -Werror
+
 include $(BUILD_STATIC_LIBRARY)
 
 ################################################################################
@@ -106,6 +108,8 @@
 
   LOCAL_CFLAGS :=
 
+  LOCAL_CFLAGS += -Werror
+
   LOCAL_STATIC_LIBRARIES := libFraunhoferAAC
 
   LOCAL_SHARED_LIBRARIES := \
@@ -128,6 +132,8 @@
 
   LOCAL_CFLAGS := -DOSCL_IMPORT_REF=
 
+  LOCAL_CFLAGS += -Werror
+
   LOCAL_STATIC_LIBRARIES := \
           libstagefright_aacenc
 
diff --git a/media/libstagefright/codecs/aacenc/SoftAACEncoder2.cpp b/media/libstagefright/codecs/aacenc/SoftAACEncoder2.cpp
index ff2b503..9a91579 100644
--- a/media/libstagefright/codecs/aacenc/SoftAACEncoder2.cpp
+++ b/media/libstagefright/codecs/aacenc/SoftAACEncoder2.cpp
@@ -338,7 +338,7 @@
     return OK;
 }
 
-void SoftAACEncoder2::onQueueFilled(OMX_U32 portIndex) {
+void SoftAACEncoder2::onQueueFilled(OMX_U32 /* portIndex */) {
     if (mSignalledError) {
         return;
     }
diff --git a/media/libstagefright/codecs/aacenc/basic_op/oper_32b.c b/media/libstagefright/codecs/aacenc/basic_op/oper_32b.c
index cc01927..1d029fc 100644
--- a/media/libstagefright/codecs/aacenc/basic_op/oper_32b.c
+++ b/media/libstagefright/codecs/aacenc/basic_op/oper_32b.c
@@ -24,6 +24,8 @@
 #include "basic_op.h"
 #include "oper_32b.h"
 
+#define UNUSED(x) (void)(x)
+
 /*****************************************************************************
  *                                                                           *
  *  Function L_Extract()                                                     *
@@ -243,6 +245,8 @@
 Word32 rsqrt(Word32 value,     /*!< Operand to square root (0.0 ... 1) */
              Word32 accuracy)  /*!< Number of valid bits that will be calculated */
 {
+    UNUSED(accuracy);
+
     Word32 root = 0;
 	Word32 scale;
 
diff --git a/media/libstagefright/codecs/aacenc/src/aacenc.c b/media/libstagefright/codecs/aacenc/src/aacenc.c
index d1c8621..40db92c 100644
--- a/media/libstagefright/codecs/aacenc/src/aacenc.c
+++ b/media/libstagefright/codecs/aacenc/src/aacenc.c
@@ -27,6 +27,8 @@
 #include "cmnMemory.h"
 #include "memalign.h"
 
+#define UNUSED(x) (void)(x)
+
 /**
 * Init the audio codec module and return codec handle
 * \param phCodec [OUT] Return the video codec handle
@@ -46,6 +48,8 @@
 	VO_MEM_OPERATOR *pMemOP;
 	int interMem;
 
+        UNUSED(vType);
+
 	interMem = 0;
 	error = 0;
 
@@ -471,6 +475,10 @@
 */
 VO_U32 VO_API voAACEncGetParam(VO_HANDLE hCodec, VO_S32 uParamID, VO_PTR pData)
 {
+        UNUSED(hCodec);
+        UNUSED(uParamID);
+        UNUSED(pData);
+
 	return VO_ERR_NONE;
 }
 
diff --git a/media/libstagefright/codecs/aacenc/src/adj_thr.c b/media/libstagefright/codecs/aacenc/src/adj_thr.c
index ccfe883..471631c 100644
--- a/media/libstagefright/codecs/aacenc/src/adj_thr.c
+++ b/media/libstagefright/codecs/aacenc/src/adj_thr.c
@@ -72,7 +72,7 @@
                           const Word16 nChannels)
 {
   Word16 ch, sfb, sfbGrp;
-  Word32 *pthrExp, *psfbThre;
+  Word32 *pthrExp = NULL, *psfbThre;
   for (ch=0; ch<nChannels; ch++) {
     PSY_OUT_CHANNEL *psyOutChan = &psyOutChannel[ch];
 	 for(sfbGrp = 0; sfbGrp < psyOutChan->sfbCnt; sfbGrp+= psyOutChan->sfbPerGroup)
diff --git a/media/libstagefright/codecs/aacenc/src/bitenc.c b/media/libstagefright/codecs/aacenc/src/bitenc.c
index fcc12dd..d1fd647 100644
--- a/media/libstagefright/codecs/aacenc/src/bitenc.c
+++ b/media/libstagefright/codecs/aacenc/src/bitenc.c
@@ -26,6 +26,7 @@
 #include "qc_data.h"
 #include "interface.h"
 
+#define UNUSED(x) (void)(x)
 
 static const  Word16 globalGainOffset = 100;
 static const  Word16 icsReservedBit   = 0;
@@ -585,6 +586,8 @@
   Word16 elementUsedBits;
   Word16 frameBits=0;
 
+  UNUSED(ancBytes);
+
   /*   struct bitbuffer bsWriteCopy; */
   bitMarkUp = GetBitsAvail(hBitStream);
   if(qcOut->qcElement.adtsUsed)  /*  write adts header*/
diff --git a/media/libstagefright/codecs/aacenc/src/dyn_bits.c b/media/libstagefright/codecs/aacenc/src/dyn_bits.c
index 7769188..4d763d0 100644
--- a/media/libstagefright/codecs/aacenc/src/dyn_bits.c
+++ b/media/libstagefright/codecs/aacenc/src/dyn_bits.c
@@ -25,7 +25,6 @@
 #include "bit_cnt.h"
 #include "psy_const.h"
 
-
 /*****************************************************************************
 *
 * function name: buildBitLookUp
@@ -226,7 +225,7 @@
   }
 
   while (TRUE) {
-    Word16 maxMergeGain, maxNdx, maxNdxNext, maxNdxLast;
+    Word16 maxMergeGain, maxNdx = 0, maxNdxNext, maxNdxLast;
 
     maxMergeGain = findMaxMerge(mergeGainLookUp, sectionInfo, maxSfb, &maxNdx);
 
diff --git a/media/libstagefright/codecs/aacenc/src/psy_main.c b/media/libstagefright/codecs/aacenc/src/psy_main.c
index 4e9218c..6f0679c 100644
--- a/media/libstagefright/codecs/aacenc/src/psy_main.c
+++ b/media/libstagefright/codecs/aacenc/src/psy_main.c
@@ -38,6 +38,8 @@
 #include "tns_func.h"
 #include "memalign.h"
 
+#define UNUSED(x) (void)(x)
+
 /*                                    long       start       short       stop */
 static Word16 blockType2windowShape[] = {KBD_WINDOW,SINE_WINDOW,SINE_WINDOW,KBD_WINDOW};
 
@@ -170,7 +172,9 @@
 *****************************************************************************/
 Word16 PsyOutDelete(PSY_OUT *hPsyOut, VO_MEM_OPERATOR *pMemOP)
 {
-  hPsyOut=NULL;
+  UNUSED(hPsyOut);
+  UNUSED(pMemOP);
+
   return 0;
 }
 
diff --git a/media/libstagefright/codecs/aacenc/src/qc_main.c b/media/libstagefright/codecs/aacenc/src/qc_main.c
index 48ff300..e5d78aa 100644
--- a/media/libstagefright/codecs/aacenc/src/qc_main.c
+++ b/media/libstagefright/codecs/aacenc/src/qc_main.c
@@ -33,6 +33,7 @@
 #include "channel_map.h"
 #include "memalign.h"
 
+#define UNUSED(x) (void)(x)
 
 typedef enum{
   FRAME_LEN_BYTES_MODULO =  1,
@@ -204,11 +205,8 @@
 **********************************************************************************/
 void QCDelete(QC_STATE *hQC, VO_MEM_OPERATOR *pMemOP)
 {
-
-  /*
-     nothing to do
-  */
-  hQC=NULL;
+  UNUSED(hQC);
+  UNUSED(pMemOP);
 }
 
 /*********************************************************************************
diff --git a/media/libstagefright/codecs/aacenc/src/tns.c b/media/libstagefright/codecs/aacenc/src/tns.c
index 455a864..5172612 100644
--- a/media/libstagefright/codecs/aacenc/src/tns.c
+++ b/media/libstagefright/codecs/aacenc/src/tns.c
@@ -30,6 +30,8 @@
 #include "psy_configuration.h"
 #include "tns_func.h"
 
+#define UNUSED(x) (void)(x)
+
 #define TNS_MODIFY_BEGIN         2600  /* Hz */
 #define RATIO_PATCH_LOWER_BORDER 380   /* Hz */
 #define TNS_GAIN_THRESH			 141   /* 1.41*100 */
@@ -643,6 +645,8 @@
   Word32 i;
   Word32 tnsOrderPlus1 = tnsOrder + 1;
 
+  UNUSED(window);
+
   assert(tnsOrder <= TNS_MAX_ORDER);      /* remove asserts later? (btg) */
 
   for(i=0;i<tnsOrder;i++) {
diff --git a/media/libstagefright/codecs/amrnb/common/Android.mk b/media/libstagefright/codecs/amrnb/common/Android.mk
index 30ce29c..a2b3c8f 100644
--- a/media/libstagefright/codecs/amrnb/common/Android.mk
+++ b/media/libstagefright/codecs/amrnb/common/Android.mk
@@ -69,6 +69,8 @@
 LOCAL_CFLAGS := \
         -DOSCL_UNUSED_ARG= -DOSCL_IMPORT_REF= -DOSCL_EXPORT_REF=
 
+LOCAL_CFLAGS += -Werror
+
 LOCAL_MODULE := libstagefright_amrnb_common
 
 include $(BUILD_SHARED_LIBRARY)
diff --git a/media/libstagefright/codecs/amrnb/dec/Android.mk b/media/libstagefright/codecs/amrnb/dec/Android.mk
index 8d6c6f8..b067456 100644
--- a/media/libstagefright/codecs/amrnb/dec/Android.mk
+++ b/media/libstagefright/codecs/amrnb/dec/Android.mk
@@ -47,6 +47,8 @@
 LOCAL_CFLAGS := \
         -DOSCL_UNUSED_ARG= -DOSCL_IMPORT_REF=
 
+LOCAL_CFLAGS += -Werror
+
 LOCAL_MODULE := libstagefright_amrnbdec
 
 include $(BUILD_STATIC_LIBRARY)
@@ -68,6 +70,8 @@
 
 LOCAL_CFLAGS := -DOSCL_IMPORT_REF=
 
+LOCAL_CFLAGS += -Werror
+
 LOCAL_STATIC_LIBRARIES := \
         libstagefright_amrnbdec libstagefright_amrwbdec
 
diff --git a/media/libstagefright/codecs/amrnb/dec/SoftAMR.cpp b/media/libstagefright/codecs/amrnb/dec/SoftAMR.cpp
index 3320688..d1b0f76 100644
--- a/media/libstagefright/codecs/amrnb/dec/SoftAMR.cpp
+++ b/media/libstagefright/codecs/amrnb/dec/SoftAMR.cpp
@@ -274,7 +274,7 @@
     return frameSize;
 }
 
-void SoftAMR::onQueueFilled(OMX_U32 portIndex) {
+void SoftAMR::onQueueFilled(OMX_U32 /* portIndex */) {
     List<BufferInfo *> &inQueue = getPortQueue(0);
     List<BufferInfo *> &outQueue = getPortQueue(1);
 
@@ -428,7 +428,7 @@
     }
 }
 
-void SoftAMR::onPortFlushCompleted(OMX_U32 portIndex) {
+void SoftAMR::onPortFlushCompleted(OMX_U32 /* portIndex */) {
 }
 
 void SoftAMR::onPortEnableCompleted(OMX_U32 portIndex, bool enabled) {
diff --git a/media/libstagefright/codecs/amrnb/enc/Android.mk b/media/libstagefright/codecs/amrnb/enc/Android.mk
index f4e467a..afc0b89 100644
--- a/media/libstagefright/codecs/amrnb/enc/Android.mk
+++ b/media/libstagefright/codecs/amrnb/enc/Android.mk
@@ -69,6 +69,8 @@
 LOCAL_CFLAGS := \
         -DOSCL_UNUSED_ARG=
 
+LOCAL_CFLAGS += -Werror
+
 LOCAL_MODULE := libstagefright_amrnbenc
 
 include $(BUILD_STATIC_LIBRARY)
@@ -88,6 +90,8 @@
         $(LOCAL_PATH)/../common/include \
         $(LOCAL_PATH)/../common
 
+LOCAL_CFLAGS += -Werror
+
 LOCAL_STATIC_LIBRARIES := \
         libstagefright_amrnbenc
 
diff --git a/media/libstagefright/codecs/amrnb/enc/SoftAMRNBEncoder.cpp b/media/libstagefright/codecs/amrnb/enc/SoftAMRNBEncoder.cpp
index 50b739c..9489457 100644
--- a/media/libstagefright/codecs/amrnb/enc/SoftAMRNBEncoder.cpp
+++ b/media/libstagefright/codecs/amrnb/enc/SoftAMRNBEncoder.cpp
@@ -270,7 +270,7 @@
     }
 }
 
-void SoftAMRNBEncoder::onQueueFilled(OMX_U32 portIndex) {
+void SoftAMRNBEncoder::onQueueFilled(OMX_U32 /* portIndex */) {
     if (mSignalledError) {
         return;
     }
diff --git a/media/libstagefright/codecs/amrwb/Android.mk b/media/libstagefright/codecs/amrwb/Android.mk
index 677107f..efdf988 100644
--- a/media/libstagefright/codecs/amrwb/Android.mk
+++ b/media/libstagefright/codecs/amrwb/Android.mk
@@ -50,6 +50,8 @@
 LOCAL_CFLAGS := \
         -DOSCL_UNUSED_ARG= -DOSCL_IMPORT_REF=
 
+LOCAL_CFLAGS += -Werror
+
 LOCAL_MODULE := libstagefright_amrwbdec
 
 include $(BUILD_STATIC_LIBRARY)
diff --git a/media/libstagefright/codecs/amrwbenc/Android.mk b/media/libstagefright/codecs/amrwbenc/Android.mk
index c5b8e0c..64fe8d1 100644
--- a/media/libstagefright/codecs/amrwbenc/Android.mk
+++ b/media/libstagefright/codecs/amrwbenc/Android.mk
@@ -112,6 +112,8 @@
 LOCAL_C_INCLUDES += $(LOCAL_PATH)/src/asm/ARMV7
 endif
 
+LOCAL_CFLAGS += -Werror
+
 include $(BUILD_STATIC_LIBRARY)
 
 ################################################################################
@@ -126,6 +128,8 @@
 	frameworks/av/media/libstagefright/codecs/common/include \
 	frameworks/native/include/media/openmax
 
+LOCAL_CFLAGS += -Werror
+
 LOCAL_STATIC_LIBRARIES := \
         libstagefright_amrwbenc
 
diff --git a/media/libstagefright/codecs/amrwbenc/SoftAMRWBEncoder.cpp b/media/libstagefright/codecs/amrwbenc/SoftAMRWBEncoder.cpp
index 9ccb49c..91a512d 100644
--- a/media/libstagefright/codecs/amrwbenc/SoftAMRWBEncoder.cpp
+++ b/media/libstagefright/codecs/amrwbenc/SoftAMRWBEncoder.cpp
@@ -317,7 +317,7 @@
     }
 }
 
-void SoftAMRWBEncoder::onQueueFilled(OMX_U32 portIndex) {
+void SoftAMRWBEncoder::onQueueFilled(OMX_U32 /* portIndex */) {
     if (mSignalledError) {
         return;
     }
diff --git a/media/libstagefright/codecs/amrwbenc/src/autocorr.c b/media/libstagefright/codecs/amrwbenc/src/autocorr.c
index 8c477ca..0b2ea89 100644
--- a/media/libstagefright/codecs/amrwbenc/src/autocorr.c
+++ b/media/libstagefright/codecs/amrwbenc/src/autocorr.c
@@ -28,6 +28,8 @@
 #include "acelp.h"
 #include "ham_wind.tab"
 
+#define UNUSED(x) (void)(x)
+
 void Autocorr(
 		Word16 x[],                           /* (i)    : Input signal                      */
 		Word16 m,                             /* (i)    : LPC order                         */
@@ -40,6 +42,8 @@
 	Word32 L_sum, L_sum1, L_tmp, F_LEN;
 	Word16 *p1,*p2,*p3;
 	const Word16 *p4;
+        UNUSED(m);
+
 	/* Windowing of signal */
 	p1 = x;
 	p4 = vo_window;
diff --git a/media/libstagefright/codecs/amrwbenc/src/convolve.c b/media/libstagefright/codecs/amrwbenc/src/convolve.c
index acba532..4c1f7d4 100644
--- a/media/libstagefright/codecs/amrwbenc/src/convolve.c
+++ b/media/libstagefright/codecs/amrwbenc/src/convolve.c
@@ -25,6 +25,8 @@
 #include "typedef.h"
 #include "basic_op.h"
 
+#define UNUSED(x) (void)(x)
+
 void Convolve (
 		Word16 x[],        /* (i)     : input vector                           */
 		Word16 h[],        /* (i)     : impulse response                       */
@@ -35,6 +37,8 @@
 	Word32  i, n;
 	Word16 *tmpH,*tmpX;
 	Word32 s;
+        UNUSED(L);
+
 	for (n = 0; n < 64;)
 	{
 		tmpH = h+n;
diff --git a/media/libstagefright/codecs/amrwbenc/src/pitch_f4.c b/media/libstagefright/codecs/amrwbenc/src/pitch_f4.c
index 0d66c31..b66b55e 100644
--- a/media/libstagefright/codecs/amrwbenc/src/pitch_f4.c
+++ b/media/libstagefright/codecs/amrwbenc/src/pitch_f4.c
@@ -31,6 +31,8 @@
 #define UP_SAMP      4
 #define L_INTERPOL1  4
 
+#define UNUSED(x) (void)(x)
+
 /* Local functions */
 
 #ifdef ASM_OPT
@@ -171,6 +173,7 @@
 	Word32 corr, exp_corr, norm, exp, scale;
 	Word16 exp_norm, excf[L_SUBFR], tmp;
 	Word32 L_tmp, L_tmp1, L_tmp2;
+        UNUSED(L_subfr);
 
 	/* compute the filtered excitation for the first delay t_min */
 	k = -t_min;
diff --git a/media/libstagefright/codecs/amrwbenc/src/syn_filt.c b/media/libstagefright/codecs/amrwbenc/src/syn_filt.c
index 1bda05a..961aadc 100644
--- a/media/libstagefright/codecs/amrwbenc/src/syn_filt.c
+++ b/media/libstagefright/codecs/amrwbenc/src/syn_filt.c
@@ -26,6 +26,8 @@
 #include "math_op.h"
 #include "cnst.h"
 
+#define UNUSED(x) (void)(x)
+
 void Syn_filt(
 		Word16 a[],                           /* (i) Q12 : a[m+1] prediction coefficients           */
 		Word16 x[],                           /* (i)     : input signal                             */
@@ -95,6 +97,8 @@
 	Word32 i,a0;
 	Word32 L_tmp, L_tmp1;
 	Word16 *p1, *p2, *p3;
+        UNUSED(m);
+
 	a0 = a[0] >> (4 + Qnew);          /* input / 16 and >>Qnew */
 	/* Do the filtering. */
 	for (i = 0; i < lg; i++)
diff --git a/media/libstagefright/codecs/amrwbenc/src/voAMRWBEnc.c b/media/libstagefright/codecs/amrwbenc/src/voAMRWBEnc.c
index ea9da52..df7b9b3 100644
--- a/media/libstagefright/codecs/amrwbenc/src/voAMRWBEnc.c
+++ b/media/libstagefright/codecs/amrwbenc/src/voAMRWBEnc.c
@@ -39,6 +39,8 @@
 #include "mem_align.h"
 #include "cmnMemory.h"
 
+#define UNUSED(x) (void)(x)
+
 #ifdef __cplusplus
 extern "C" {
 #endif
@@ -1602,6 +1604,8 @@
 	VO_MEM_OPERATOR voMemoprator;
 #endif
 	VO_MEM_OPERATOR *pMemOP;
+        UNUSED(vType);
+
 	int interMem = 0;
 
 	if(pUserData == NULL || pUserData->memflag != VO_IMF_USERMEMOPERATOR || pUserData->memData == NULL )
diff --git a/media/libstagefright/codecs/avc/common/Android.mk b/media/libstagefright/codecs/avc/common/Android.mk
index 22dee15..844ef0a 100644
--- a/media/libstagefright/codecs/avc/common/Android.mk
+++ b/media/libstagefright/codecs/avc/common/Android.mk
@@ -16,4 +16,6 @@
 	$(LOCAL_PATH)/src \
  	$(LOCAL_PATH)/include
 
+LOCAL_CFLAGS += -Werror
+
 include $(BUILD_SHARED_LIBRARY)
diff --git a/media/libstagefright/codecs/avc/enc/Android.mk b/media/libstagefright/codecs/avc/enc/Android.mk
index 7d17c2a..537ba42 100644
--- a/media/libstagefright/codecs/avc/enc/Android.mk
+++ b/media/libstagefright/codecs/avc/enc/Android.mk
@@ -30,6 +30,8 @@
 LOCAL_CFLAGS := \
     -DOSCL_IMPORT_REF= -DOSCL_UNUSED_ARG= -DOSCL_EXPORT_REF=
 
+LOCAL_CFLAGS += -Werror
+
 include $(BUILD_STATIC_LIBRARY)
 
 ################################################################################
@@ -69,4 +71,6 @@
 LOCAL_MODULE := libstagefright_soft_h264enc
 LOCAL_MODULE_TAGS := optional
 
+LOCAL_CFLAGS += -Werror
+
 include $(BUILD_SHARED_LIBRARY)
diff --git a/media/libstagefright/codecs/avc/enc/SoftAVCEncoder.cpp b/media/libstagefright/codecs/avc/enc/SoftAVCEncoder.cpp
index 4a21a3e..a15b040 100644
--- a/media/libstagefright/codecs/avc/enc/SoftAVCEncoder.cpp
+++ b/media/libstagefright/codecs/avc/enc/SoftAVCEncoder.cpp
@@ -34,6 +34,12 @@
 
 #include "SoftAVCEncoder.h"
 
+#if LOG_NDEBUG
+#define UNUSED_UNLESS_VERBOSE(x) (void)(x)
+#else
+#define UNUSED_UNLESS_VERBOSE(x)
+#endif
+
 namespace android {
 
 template<class T>
@@ -136,14 +142,14 @@
 }
 
 static void* MallocWrapper(
-        void *userData, int32_t size, int32_t attrs) {
+        void * /* userData */, int32_t size, int32_t /* attrs */) {
     void *ptr = malloc(size);
     if (ptr)
         memset(ptr, 0, size);
     return ptr;
 }
 
-static void FreeWrapper(void *userData, void* ptr) {
+static void FreeWrapper(void * /* userData */, void* ptr) {
     free(ptr);
 }
 
@@ -217,7 +223,7 @@
     mHandle->CBAVC_Free = FreeWrapper;
 
     CHECK(mEncParams != NULL);
-    memset(mEncParams, 0, sizeof(mEncParams));
+    memset(mEncParams, 0, sizeof(*mEncParams));
     mEncParams->rate_control = AVC_ON;
     mEncParams->initQP = 0;
     mEncParams->init_CBP_removal_delay = 1600;
@@ -593,6 +599,17 @@
                 mVideoHeight = def->format.video.nFrameHeight;
                 mVideoFrameRate = def->format.video.xFramerate >> 16;
                 mVideoColorFormat = def->format.video.eColorFormat;
+
+                OMX_PARAM_PORTDEFINITIONTYPE *portDef =
+                    &editPortInfo(0)->mDef;
+                portDef->format.video.nFrameWidth = mVideoWidth;
+                portDef->format.video.nFrameHeight = mVideoHeight;
+                portDef->format.video.xFramerate = def->format.video.xFramerate;
+                portDef->format.video.eColorFormat =
+                    (OMX_COLOR_FORMATTYPE) mVideoColorFormat;
+                portDef = &editPortInfo(1)->mDef;
+                portDef->format.video.nFrameWidth = mVideoWidth;
+                portDef->format.video.nFrameHeight = mVideoHeight;
             } else {
                 mVideoBitRate = def->format.video.nBitrate;
             }
@@ -711,7 +728,7 @@
     }
 }
 
-void SoftAVCEncoder::onQueueFilled(OMX_U32 portIndex) {
+void SoftAVCEncoder::onQueueFilled(OMX_U32 /* portIndex */) {
     if (mSignalledError || mSawInputEOS) {
         return;
     }
@@ -784,7 +801,7 @@
             }
         }
 
-        buffer_handle_t srcBuffer; // for MetaDataMode only
+        buffer_handle_t srcBuffer = NULL; // for MetaDataMode only
 
         // Get next input video frame
         if (mReadyForNextFrame) {
@@ -953,6 +970,7 @@
 }
 
 void SoftAVCEncoder::signalBufferReturned(MediaBuffer *buffer) {
+    UNUSED_UNLESS_VERBOSE(buffer);
     ALOGV("signalBufferReturned: %p", buffer);
 }
 
diff --git a/media/libstagefright/codecs/avc/enc/src/bitstream_io.cpp b/media/libstagefright/codecs/avc/enc/src/bitstream_io.cpp
index 0e3037f..d71c327 100644
--- a/media/libstagefright/codecs/avc/enc/src/bitstream_io.cpp
+++ b/media/libstagefright/codecs/avc/enc/src/bitstream_io.cpp
@@ -103,6 +103,15 @@
     {
         num_bits -= 8;
         byte = (current_word >> num_bits) & 0xFF;
+        if (stream->count_zeros == 2)
+        {   /* for num_bits = 32, this can add 2 more bytes extra for EPBS */
+            if (byte <= 3)
+            {
+                *write_pnt++ = 0x3;
+                stream->write_pos++;
+                stream->count_zeros = 0;
+            }
+        }
         if (byte != 0)
         {
             *write_pnt++ = byte;
@@ -114,12 +123,6 @@
             stream->count_zeros++;
             *write_pnt++ = byte;
             stream->write_pos++;
-            if (stream->count_zeros == 2)
-            {   /* for num_bits = 32, this can add 2 more bytes extra for EPBS */
-                *write_pnt++ = 0x3;
-                stream->write_pos++;
-                stream->count_zeros = 0;
-            }
         }
     }
 
diff --git a/media/libstagefright/codecs/common/Android.mk b/media/libstagefright/codecs/common/Android.mk
index a33cb92..b0010ff 100644
--- a/media/libstagefright/codecs/common/Android.mk
+++ b/media/libstagefright/codecs/common/Android.mk
@@ -14,6 +14,8 @@
 LOCAL_C_INCLUDES := \
 	$(LOCAL_PATH)/include
 
+LOCAL_CFLAGS += -Werror
+
 include $(BUILD_SHARED_LIBRARY)
 
 
diff --git a/media/libstagefright/codecs/common/cmnMemory.c b/media/libstagefright/codecs/common/cmnMemory.c
index aa52bd9..5bb6cc4 100644
--- a/media/libstagefright/codecs/common/cmnMemory.c
+++ b/media/libstagefright/codecs/common/cmnMemory.c
@@ -26,8 +26,12 @@
 
 //VO_MEM_OPERATOR		g_memOP;
 
+#define UNUSED(x) (void)(x)
+
 VO_U32 cmnMemAlloc (VO_S32 uID,  VO_MEM_INFO * pMemInfo)
 {
+        UNUSED(uID);
+
 	if (!pMemInfo)
 		return VO_ERR_INVALID_ARG;
 
@@ -37,34 +41,48 @@
 
 VO_U32 cmnMemFree (VO_S32 uID, VO_PTR pMem)
 {
+        UNUSED(uID);
+
 	free (pMem);
 	return 0;
 }
 
 VO_U32	cmnMemSet (VO_S32 uID, VO_PTR pBuff, VO_U8 uValue, VO_U32 uSize)
 {
+        UNUSED(uID);
+
 	memset (pBuff, uValue, uSize);
 	return 0;
 }
 
 VO_U32	cmnMemCopy (VO_S32 uID, VO_PTR pDest, VO_PTR pSource, VO_U32 uSize)
 {
+        UNUSED(uID);
+
 	memcpy (pDest, pSource, uSize);
 	return 0;
 }
 
 VO_U32	cmnMemCheck (VO_S32 uID, VO_PTR pBuffer, VO_U32 uSize)
 {
+        UNUSED(uID);
+        UNUSED(pBuffer);
+        UNUSED(uSize);
+
 	return 0;
 }
 
 VO_S32 cmnMemCompare (VO_S32 uID, VO_PTR pBuffer1, VO_PTR pBuffer2, VO_U32 uSize)
 {
+        UNUSED(uID);
+
 	return memcmp(pBuffer1, pBuffer2, uSize);
 }
 
 VO_U32	cmnMemMove (VO_S32 uID, VO_PTR pDest, VO_PTR pSource, VO_U32 uSize)
 {
+        UNUSED(uID);
+
 	memmove (pDest, pSource, uSize);
 	return 0;
 }
diff --git a/media/libstagefright/codecs/flac/enc/Android.mk b/media/libstagefright/codecs/flac/enc/Android.mk
index f01d605..59a11de 100644
--- a/media/libstagefright/codecs/flac/enc/Android.mk
+++ b/media/libstagefright/codecs/flac/enc/Android.mk
@@ -9,6 +9,8 @@
         frameworks/native/include/media/openmax \
         external/flac/include
 
+LOCAL_CFLAGS += -Werror
+
 LOCAL_SHARED_LIBRARIES := \
         libstagefright libstagefright_omx libstagefright_foundation libutils liblog
 
diff --git a/media/libstagefright/codecs/flac/enc/SoftFlacEncoder.cpp b/media/libstagefright/codecs/flac/enc/SoftFlacEncoder.cpp
index e64fe72..40661e7 100644
--- a/media/libstagefright/codecs/flac/enc/SoftFlacEncoder.cpp
+++ b/media/libstagefright/codecs/flac/enc/SoftFlacEncoder.cpp
@@ -27,6 +27,12 @@
 #define FLAC_COMPRESSION_LEVEL_DEFAULT 5
 #define FLAC_COMPRESSION_LEVEL_MAX     8
 
+#if LOG_NDEBUG
+#define UNUSED_UNLESS_VERBOSE(x) (void)(x)
+#else
+#define UNUSED_UNLESS_VERBOSE(x)
+#endif
+
 namespace android {
 
 template<class T>
@@ -257,7 +263,7 @@
 }
 
 void SoftFlacEncoder::onQueueFilled(OMX_U32 portIndex) {
-
+    UNUSED_UNLESS_VERBOSE(portIndex);
     ALOGV("SoftFlacEncoder::onQueueFilled(portIndex=%ld)", portIndex);
 
     if (mSignalledError) {
@@ -343,10 +349,11 @@
     }
 }
 
-
 FLAC__StreamEncoderWriteStatus SoftFlacEncoder::onEncodedFlacAvailable(
             const FLAC__byte buffer[],
-            size_t bytes, unsigned samples, unsigned current_frame) {
+            size_t bytes, unsigned samples,
+            unsigned current_frame) {
+    UNUSED_UNLESS_VERBOSE(current_frame);
     ALOGV("SoftFlacEncoder::onEncodedFlacAvailable(bytes=%d, samples=%d, curr_frame=%d)",
             bytes, samples, current_frame);
 
@@ -444,8 +451,12 @@
 
 // static
 FLAC__StreamEncoderWriteStatus SoftFlacEncoder::flacEncoderWriteCallback(
-            const FLAC__StreamEncoder *encoder, const FLAC__byte buffer[],
-            size_t bytes, unsigned samples, unsigned current_frame, void *client_data) {
+            const FLAC__StreamEncoder * /* encoder */,
+            const FLAC__byte buffer[],
+            size_t bytes,
+            unsigned samples,
+            unsigned current_frame,
+            void *client_data) {
     return ((SoftFlacEncoder*) client_data)->onEncodedFlacAvailable(
             buffer, bytes, samples, current_frame);
 }
diff --git a/media/libstagefright/codecs/g711/dec/Android.mk b/media/libstagefright/codecs/g711/dec/Android.mk
index 4c80da6..a0112e1 100644
--- a/media/libstagefright/codecs/g711/dec/Android.mk
+++ b/media/libstagefright/codecs/g711/dec/Android.mk
@@ -14,4 +14,6 @@
 LOCAL_MODULE := libstagefright_soft_g711dec
 LOCAL_MODULE_TAGS := optional
 
+LOCAL_CFLAGS += -Werror
+
 include $(BUILD_SHARED_LIBRARY)
diff --git a/media/libstagefright/codecs/g711/dec/SoftG711.cpp b/media/libstagefright/codecs/g711/dec/SoftG711.cpp
index bcdd3c7..160ada0 100644
--- a/media/libstagefright/codecs/g711/dec/SoftG711.cpp
+++ b/media/libstagefright/codecs/g711/dec/SoftG711.cpp
@@ -182,7 +182,7 @@
     }
 }
 
-void SoftG711::onQueueFilled(OMX_U32 portIndex) {
+void SoftG711::onQueueFilled(OMX_U32 /* portIndex */) {
     if (mSignalledError) {
         return;
     }
diff --git a/media/libstagefright/codecs/gsm/dec/Android.mk b/media/libstagefright/codecs/gsm/dec/Android.mk
index 71613d2..30868d5 100644
--- a/media/libstagefright/codecs/gsm/dec/Android.mk
+++ b/media/libstagefright/codecs/gsm/dec/Android.mk
@@ -9,6 +9,8 @@
         frameworks/native/include/media/openmax \
         external/libgsm/inc
 
+LOCAL_CFLAGS += -Werror
+
 LOCAL_SHARED_LIBRARIES := \
         libstagefright libstagefright_omx libstagefright_foundation libutils liblog
 
diff --git a/media/libstagefright/codecs/gsm/dec/SoftGSM.cpp b/media/libstagefright/codecs/gsm/dec/SoftGSM.cpp
index 00e0c85..18f7d29 100644
--- a/media/libstagefright/codecs/gsm/dec/SoftGSM.cpp
+++ b/media/libstagefright/codecs/gsm/dec/SoftGSM.cpp
@@ -172,7 +172,7 @@
     }
 }
 
-void SoftGSM::onQueueFilled(OMX_U32 portIndex) {
+void SoftGSM::onQueueFilled(OMX_U32 /* portIndex */) {
     if (mSignalledError) {
         return;
     }
diff --git a/media/libstagefright/codecs/m4v_h263/dec/Android.mk b/media/libstagefright/codecs/m4v_h263/dec/Android.mk
index a3d5779..1d232c6 100644
--- a/media/libstagefright/codecs/m4v_h263/dec/Android.mk
+++ b/media/libstagefright/codecs/m4v_h263/dec/Android.mk
@@ -46,6 +46,8 @@
 
 LOCAL_CFLAGS := -DOSCL_EXPORT_REF= -DOSCL_IMPORT_REF=
 
+LOCAL_CFLAGS += -Werror
+
 include $(BUILD_STATIC_LIBRARY)
 
 ################################################################################
@@ -72,4 +74,6 @@
 LOCAL_MODULE := libstagefright_soft_mpeg4dec
 LOCAL_MODULE_TAGS := optional
 
+LOCAL_CFLAGS += -Werror
+
 include $(BUILD_SHARED_LIBRARY)
diff --git a/media/libstagefright/codecs/m4v_h263/dec/SoftMPEG4.cpp b/media/libstagefright/codecs/m4v_h263/dec/SoftMPEG4.cpp
index fb2a430..0d1ab71 100644
--- a/media/libstagefright/codecs/m4v_h263/dec/SoftMPEG4.cpp
+++ b/media/libstagefright/codecs/m4v_h263/dec/SoftMPEG4.cpp
@@ -91,7 +91,7 @@
     return OK;
 }
 
-void SoftMPEG4::onQueueFilled(OMX_U32 portIndex) {
+void SoftMPEG4::onQueueFilled(OMX_U32 /* portIndex */) {
     if (mSignalledError || mOutputPortSettingsChange != NONE) {
         return;
     }
diff --git a/media/libstagefright/codecs/m4v_h263/dec/src/get_pred_adv_b_add.cpp b/media/libstagefright/codecs/m4v_h263/dec/src/get_pred_adv_b_add.cpp
index e23f23d..fe9e7dc 100644
--- a/media/libstagefright/codecs/m4v_h263/dec/src/get_pred_adv_b_add.cpp
+++ b/media/libstagefright/codecs/m4v_h263/dec/src/get_pred_adv_b_add.cpp
@@ -96,7 +96,7 @@
     offset = width - B_SIZE; /* offset for prev */
     offset2 = (pred_width_rnd >> 1) - 4; /* offset for pred_block */
 
-    tmp = (uint32)prev & 0x3;
+    tmp = (uintptr_t)prev & 0x3;
     pred_block -= offset2; /* preset */
 
     if (tmp == 0)  /* word-aligned */
@@ -203,7 +203,7 @@
     /* Branch based on pixel location (half-pel or full-pel) for x and y */
     pred_block -= offset2; /* preset */
 
-    tmp = (uint32)prev & 3;
+    tmp = (uintptr_t)prev & 3;
     mask = 254;
     mask |= (mask << 8);
     mask |= (mask << 16); /* 0xFEFEFEFE */
@@ -532,7 +532,7 @@
     /* Branch based on pixel location (half-pel or full-pel) for x and y */
     pred_block -= offset2; /* preset */
 
-    tmp = (uint32)prev & 3;
+    tmp = (uintptr_t)prev & 3;
     mask = 254;
     mask |= (mask << 8);
     mask |= (mask << 16); /* 0xFEFEFEFE */
@@ -884,7 +884,7 @@
     mask |= (mask << 8);
     mask |= (mask << 16); /* 0x3f3f3f3f */
 
-    tmp = (uint32)prev & 3;
+    tmp = (uintptr_t)prev & 3;
 
     pred_block -= 4; /* preset */
 
diff --git a/media/libstagefright/codecs/m4v_h263/enc/Android.mk b/media/libstagefright/codecs/m4v_h263/enc/Android.mk
index 83a2dd2..c9006d9 100644
--- a/media/libstagefright/codecs/m4v_h263/enc/Android.mk
+++ b/media/libstagefright/codecs/m4v_h263/enc/Android.mk
@@ -33,6 +33,8 @@
     $(TOP)/frameworks/av/media/libstagefright/include \
     $(TOP)/frameworks/native/include/media/openmax
 
+LOCAL_CFLAGS += -Werror
+
 include $(BUILD_STATIC_LIBRARY)
 
 ################################################################################
@@ -72,4 +74,6 @@
 LOCAL_MODULE := libstagefright_soft_mpeg4enc
 LOCAL_MODULE_TAGS := optional
 
+LOCAL_CFLAGS += -Werror
+
 include $(BUILD_SHARED_LIBRARY)
diff --git a/media/libstagefright/codecs/m4v_h263/enc/SoftMPEG4Encoder.cpp b/media/libstagefright/codecs/m4v_h263/enc/SoftMPEG4Encoder.cpp
index e02af90..2c73e57 100644
--- a/media/libstagefright/codecs/m4v_h263/enc/SoftMPEG4Encoder.cpp
+++ b/media/libstagefright/codecs/m4v_h263/enc/SoftMPEG4Encoder.cpp
@@ -620,7 +620,7 @@
     }
 }
 
-void SoftMPEG4Encoder::onQueueFilled(OMX_U32 portIndex) {
+void SoftMPEG4Encoder::onQueueFilled(OMX_U32 /* portIndex */) {
     if (mSignalledError || mSawInputEOS) {
         return;
     }
@@ -677,7 +677,7 @@
             mSawInputEOS = true;
         }
 
-        buffer_handle_t srcBuffer; // for MetaDataMode only
+        buffer_handle_t srcBuffer = NULL; // for MetaDataMode only
         if (inHeader->nFilledLen > 0) {
             uint8_t *inputData = NULL;
             if (mStoreMetaDataInBuffers) {
diff --git a/media/libstagefright/codecs/m4v_h263/enc/src/dct.cpp b/media/libstagefright/codecs/m4v_h263/enc/src/dct.cpp
index fa50eeb..fa4ae23 100644
--- a/media/libstagefright/codecs/m4v_h263/enc/src/dct.cpp
+++ b/media/libstagefright/codecs/m4v_h263/enc/src/dct.cpp
@@ -250,7 +250,7 @@
             out[40] = k4 ;   /* row 5 */
             out++;
         }
-        while ((UInt)out < (UInt)dst) ;
+        while ((uintptr_t)out < (uintptr_t)dst) ;
 
         return ;
     }
@@ -455,7 +455,7 @@
             out[8] = k5 ;       /* row 1 */
             out++;
         }
-        while ((UInt)out < (UInt)dst) ;
+        while ((uintptr_t)out < (uintptr_t)dst) ;
 
         return ;
     }
@@ -635,7 +635,7 @@
             out[8] = k5 ;       /* row 1 */
             out++;
         }
-        while ((UInt)out < (UInt)dst) ;
+        while ((uintptr_t)out < (uintptr_t)dst) ;
 
         return ;
     }
@@ -846,7 +846,7 @@
             out[40] = k4 ;   /* row 5 */
             out++;
         }
-        while ((UInt)out < (UInt)dst) ;
+        while ((uintptr_t)out < (uintptr_t)dst) ;
 
         return ;
     }
@@ -1033,7 +1033,7 @@
             out[8] = k5 ;       /* row 1 */
             out++;
         }
-        while ((UInt)out < (UInt)dst) ;
+        while ((uintptr_t)out < (uintptr_t)dst) ;
 
         return ;
     }
@@ -1195,7 +1195,7 @@
             out[8] = k5 ;       /* row 1 */
             out++;
         }
-        while ((UInt)out < (UInt)dst) ;
+        while ((uintptr_t)out < (uintptr_t)dst) ;
 
         return ;
     }
diff --git a/media/libstagefright/codecs/m4v_h263/enc/src/fastcodemb.cpp b/media/libstagefright/codecs/m4v_h263/enc/src/fastcodemb.cpp
index 6fd41c3..0ad39a6 100644
--- a/media/libstagefright/codecs/m4v_h263/enc/src/fastcodemb.cpp
+++ b/media/libstagefright/codecs/m4v_h263/enc/src/fastcodemb.cpp
@@ -572,7 +572,7 @@
         cur2    = cur2 & (mask << 8);   /* mask first and third bytes */
         sum2    = sum2 + ((UInt)cur2 >> 8);
     }
-    while ((UInt)curInt < (UInt)end);
+    while ((uintptr_t)curInt < (uintptr_t)end);
 
     cur1 = sum4 - (sum2 << 8);  /* get even-sum */
     cur1 = cur1 + sum2;         /* add 16 bit even-sum and odd-sum*/
@@ -611,7 +611,7 @@
         load2 = load2 & (mask << 8); /* even bytes */
         sum2 += ((UInt)load2 >> 8); /* sum even bytes, 16 bit */
     }
-    while ((UInt)curInt < (UInt)end);
+    while ((uintptr_t)curInt < (uintptr_t)end);
     load1 = sum4 - (sum2 << 8);     /* get even-sum */
     load1 = load1 + sum2;           /* add 16 bit even-sum and odd-sum*/
     load1 = load1 + (load1 << 16);  /* add upper and lower 16 bit sum */
diff --git a/media/libstagefright/codecs/m4v_h263/enc/src/motion_comp.cpp b/media/libstagefright/codecs/m4v_h263/enc/src/motion_comp.cpp
index b81d278..06e8926 100644
--- a/media/libstagefright/codecs/m4v_h263/enc/src/motion_comp.cpp
+++ b/media/libstagefright/codecs/m4v_h263/enc/src/motion_comp.cpp
@@ -1959,7 +1959,7 @@
         dst += offset;
         src += offset;
     }
-    while ((UInt)src < (UInt)end);
+    while ((uintptr_t)src < (uintptr_t)end);
 
     return ;
 }
diff --git a/media/libstagefright/codecs/m4v_h263/enc/src/sad_inline.h b/media/libstagefright/codecs/m4v_h263/enc/src/sad_inline.h
index ba77dfd..b865f23 100644
--- a/media/libstagefright/codecs/m4v_h263/enc/src/sad_inline.h
+++ b/media/libstagefright/codecs/m4v_h263/enc/src/sad_inline.h
@@ -85,7 +85,7 @@
 
         x9 = 0x80808080; /* const. */
 
-        x8 = (uint32)ref & 0x3;
+        x8 = (uintptr_t)ref & 0x3;
         if (x8 == 3)
             goto SadMBOffset3;
         if (x8 == 2)
diff --git a/media/libstagefright/codecs/mp3dec/Android.mk b/media/libstagefright/codecs/mp3dec/Android.mk
index 135c715..8284490 100644
--- a/media/libstagefright/codecs/mp3dec/Android.mk
+++ b/media/libstagefright/codecs/mp3dec/Android.mk
@@ -50,6 +50,8 @@
 LOCAL_CFLAGS := \
         -DOSCL_UNUSED_ARG=
 
+LOCAL_CFLAGS += -Werror
+
 LOCAL_MODULE := libstagefright_mp3dec
 
 LOCAL_ARM_MODE := arm
@@ -69,6 +71,8 @@
         $(LOCAL_PATH)/src \
         $(LOCAL_PATH)/include
 
+LOCAL_CFLAGS += -Werror
+
 LOCAL_SHARED_LIBRARIES := \
         libstagefright libstagefright_omx libstagefright_foundation libutils liblog
 
diff --git a/media/libstagefright/codecs/mp3dec/SoftMP3.cpp b/media/libstagefright/codecs/mp3dec/SoftMP3.cpp
index 877e3cb..a09ab7c 100644
--- a/media/libstagefright/codecs/mp3dec/SoftMP3.cpp
+++ b/media/libstagefright/codecs/mp3dec/SoftMP3.cpp
@@ -188,7 +188,7 @@
     }
 }
 
-void SoftMP3::onQueueFilled(OMX_U32 portIndex) {
+void SoftMP3::onQueueFilled(OMX_U32 /* portIndex */) {
     if (mSignalledError || mOutputPortSettingsChange != NONE) {
         return;
     }
@@ -335,6 +335,9 @@
         // depend on fragments from the last one decoded.
         pvmp3_InitDecoder(mConfig, mDecoderBuf);
         mIsFirst = true;
+        mSignalledError = false;
+        mSawInputEos = false;
+        mSignalledOutputEos = false;
     }
 }
 
diff --git a/media/libstagefright/codecs/mp3dec/src/pvmp3_mpeg2_get_scale_data.cpp b/media/libstagefright/codecs/mp3dec/src/pvmp3_mpeg2_get_scale_data.cpp
index ee42dc5..499672b 100644
--- a/media/libstagefright/codecs/mp3dec/src/pvmp3_mpeg2_get_scale_data.cpp
+++ b/media/libstagefright/codecs/mp3dec/src/pvmp3_mpeg2_get_scale_data.cpp
@@ -139,7 +139,7 @@
     int16 blocknumber = 0;
 
     granuleInfo *gr_info = &(si->ch[ch].gran[gr]);
-    uint32 scalefac_comp, int_scalefac_comp, new_slen[4];
+    uint32 scalefac_comp, int_scalefac_comp, new_slen[4] = { 0,0,0,0 };
 
     scalefac_comp =  gr_info->scalefac_compress;
 
diff --git a/media/libstagefright/codecs/on2/dec/Android.mk b/media/libstagefright/codecs/on2/dec/Android.mk
index 7f2c46d..93ff64c 100644
--- a/media/libstagefright/codecs/on2/dec/Android.mk
+++ b/media/libstagefright/codecs/on2/dec/Android.mk
@@ -20,4 +20,6 @@
 LOCAL_MODULE := libstagefright_soft_vpxdec
 LOCAL_MODULE_TAGS := optional
 
+LOCAL_CFLAGS += -Werror
+
 include $(BUILD_SHARED_LIBRARY)
diff --git a/media/libstagefright/codecs/on2/dec/SoftVPX.cpp b/media/libstagefright/codecs/on2/dec/SoftVPX.cpp
index 476e986..423a057 100644
--- a/media/libstagefright/codecs/on2/dec/SoftVPX.cpp
+++ b/media/libstagefright/codecs/on2/dec/SoftVPX.cpp
@@ -85,7 +85,7 @@
     return OK;
 }
 
-void SoftVPX::onQueueFilled(OMX_U32 portIndex) {
+void SoftVPX::onQueueFilled(OMX_U32 /* portIndex */) {
     if (mOutputPortSettingsChange != NONE) {
         return;
     }
diff --git a/media/libstagefright/codecs/on2/enc/SoftVPXEncoder.cpp b/media/libstagefright/codecs/on2/enc/SoftVPXEncoder.cpp
index 8375cac..5efe022 100644
--- a/media/libstagefright/codecs/on2/enc/SoftVPXEncoder.cpp
+++ b/media/libstagefright/codecs/on2/enc/SoftVPXEncoder.cpp
@@ -677,6 +677,9 @@
         def->format.video.nFrameHeight = mHeight;
         def->format.video.xFramerate = port->format.video.xFramerate;
         def->format.video.eColorFormat = mColorFormat;
+        def = &editPortInfo(kOutputPortIndex)->mDef;
+        def->format.video.nFrameWidth = mWidth;
+        def->format.video.nFrameHeight = mHeight;
 
         return OMX_ErrorNone;
     } else if (port->nPortIndex == kOutputPortIndex) {
diff --git a/media/libstagefright/codecs/on2/h264dec/SoftAVC.cpp b/media/libstagefright/codecs/on2/h264dec/SoftAVC.cpp
index 7ddb13c..a7bde97 100644
--- a/media/libstagefright/codecs/on2/h264dec/SoftAVC.cpp
+++ b/media/libstagefright/codecs/on2/h264dec/SoftAVC.cpp
@@ -98,7 +98,7 @@
     return UNKNOWN_ERROR;
 }
 
-void SoftAVC::onQueueFilled(OMX_U32 portIndex) {
+void SoftAVC::onQueueFilled(OMX_U32 /* portIndex */) {
     if (mSignalledError || mOutputPortSettingsChange != NONE) {
         return;
     }
diff --git a/media/libstagefright/codecs/on2/h264dec/source/H264SwDecApi.c b/media/libstagefright/codecs/on2/h264dec/source/H264SwDecApi.c
index 2bb4c4d..524a3f0 100644
--- a/media/libstagefright/codecs/on2/h264dec/source/H264SwDecApi.c
+++ b/media/libstagefright/codecs/on2/h264dec/source/H264SwDecApi.c
@@ -42,6 +42,8 @@
 #include "h264bsd_decoder.h"
 #include "h264bsd_util.h"
 
+#define UNUSED(x) (void)(x)
+
 /*------------------------------------------------------------------------------
        Version Information
 ------------------------------------------------------------------------------*/
@@ -73,6 +75,7 @@
 #endif
 
 void H264SwDecTrace(char *string) {
+    UNUSED(string);
 }
 
 void* H264SwDecMalloc(u32 size) {
diff --git a/media/libstagefright/codecs/on2/h264dec/source/h264bsd_conceal.c b/media/libstagefright/codecs/on2/h264dec/source/h264bsd_conceal.c
index 493fb9e..7a262ed 100755
--- a/media/libstagefright/codecs/on2/h264dec/source/h264bsd_conceal.c
+++ b/media/libstagefright/codecs/on2/h264dec/source/h264bsd_conceal.c
@@ -267,7 +267,7 @@
     i32 firstPhase[16];
     i32 *pTmp;
     /* neighbours above, below, left and right */
-    i32 a[4], b[4], l[4], r[4];
+    i32 a[4] = { 0,0,0,0 }, b[4], l[4] = { 0,0,0,0 }, r[4];
     u32 A, B, L, R;
 #ifdef H264DEC_OMXDL
     u8 fillBuff[32*21 + 15 + 32];
diff --git a/media/libstagefright/codecs/on2/h264dec/source/h264bsd_reconstruct.c b/media/libstagefright/codecs/on2/h264dec/source/h264bsd_reconstruct.c
index c948776..b409a06 100755
--- a/media/libstagefright/codecs/on2/h264dec/source/h264bsd_reconstruct.c
+++ b/media/libstagefright/codecs/on2/h264dec/source/h264bsd_reconstruct.c
@@ -42,6 +42,8 @@
 #include "armVC.h"
 #endif /* H264DEC_OMXDL */
 
+#define UNUSED(x) (void)(x)
+
 /*------------------------------------------------------------------------------
     2. External compiler flags
 --------------------------------------------------------------------------------
@@ -2136,7 +2138,8 @@
   i32 center,
   i32 right)
 {
-
+    UNUSED(left);
+    UNUSED(right);
     ASSERT(ref);
     ASSERT(fill);
 
diff --git a/media/libstagefright/codecs/on2/h264dec/source/h264bsd_slice_header.c b/media/libstagefright/codecs/on2/h264dec/source/h264bsd_slice_header.c
index a7c6f64..23401c6 100755
--- a/media/libstagefright/codecs/on2/h264dec/source/h264bsd_slice_header.c
+++ b/media/libstagefright/codecs/on2/h264dec/source/h264bsd_slice_header.c
@@ -47,6 +47,8 @@
 #include "h264bsd_nal_unit.h"
 #include "h264bsd_dpb.h"
 
+#define UNUSED(x) (void)(x)
+
 /*------------------------------------------------------------------------------
     2. External compiler flags
 --------------------------------------------------------------------------------
@@ -1407,6 +1409,7 @@
     u32 tmp, value, i;
     i32 ivalue;
     strmData_t tmpStrmData[1];
+    UNUSED(nalUnitType);
 
 /* Code */
 
diff --git a/media/libstagefright/codecs/on2/h264dec/source/h264bsd_util.c b/media/libstagefright/codecs/on2/h264dec/source/h264bsd_util.c
index cc838fd..fb97a28 100755
--- a/media/libstagefright/codecs/on2/h264dec/source/h264bsd_util.c
+++ b/media/libstagefright/codecs/on2/h264dec/source/h264bsd_util.c
@@ -186,7 +186,7 @@
         return(HANTRO_FALSE);
 
     if ( (bits > 8) ||
-         ((h264bsdShowBits32(pStrmData)>>(32-bits)) != (1 << (bits-1))) )
+         ((h264bsdShowBits32(pStrmData)>>(32-bits)) != (1ul << (bits-1))) )
         return(HANTRO_TRUE);
     else
         return(HANTRO_FALSE);
diff --git a/media/libstagefright/codecs/on2/h264dec/source/h264bsd_util.h b/media/libstagefright/codecs/on2/h264dec/source/h264bsd_util.h
index cb3adda..216ad04 100755
--- a/media/libstagefright/codecs/on2/h264dec/source/h264bsd_util.h
+++ b/media/libstagefright/codecs/on2/h264dec/source/h264bsd_util.h
@@ -42,6 +42,7 @@
 #include <stdio.h>
 #endif
 
+#include <stdint.h>
 #include "basetype.h"
 #include "h264bsd_stream.h"
 #include "h264bsd_image.h"
@@ -150,7 +151,7 @@
 }
 
 #define ALIGN(ptr, bytePos) \
-        (ptr + ( ((bytePos - (int)ptr) & (bytePos - 1)) / sizeof(*ptr) ))
+        (ptr + ( ((bytePos - (uintptr_t)ptr) & (bytePos - 1)) / sizeof(*ptr) ))
 
 extern const u32 h264bsdQpC[52];
 
diff --git a/media/libstagefright/codecs/raw/Android.mk b/media/libstagefright/codecs/raw/Android.mk
index fe90a03..87080e7 100644
--- a/media/libstagefright/codecs/raw/Android.mk
+++ b/media/libstagefright/codecs/raw/Android.mk
@@ -8,6 +8,8 @@
         frameworks/av/media/libstagefright/include \
         frameworks/native/include/media/openmax
 
+LOCAL_CFLAGS += -Werror
+
 LOCAL_SHARED_LIBRARIES := \
         libstagefright_omx libstagefright_foundation libutils liblog
 
diff --git a/media/libstagefright/codecs/raw/SoftRaw.cpp b/media/libstagefright/codecs/raw/SoftRaw.cpp
index 19d6f13..9d514a6 100644
--- a/media/libstagefright/codecs/raw/SoftRaw.cpp
+++ b/media/libstagefright/codecs/raw/SoftRaw.cpp
@@ -163,7 +163,7 @@
     }
 }
 
-void SoftRaw::onQueueFilled(OMX_U32 portIndex) {
+void SoftRaw::onQueueFilled(OMX_U32 /* portIndex */) {
     if (mSignalledError) {
         return;
     }
diff --git a/media/libstagefright/codecs/vorbis/dec/Android.mk b/media/libstagefright/codecs/vorbis/dec/Android.mk
index 2232353..217a6d2 100644
--- a/media/libstagefright/codecs/vorbis/dec/Android.mk
+++ b/media/libstagefright/codecs/vorbis/dec/Android.mk
@@ -16,4 +16,6 @@
 LOCAL_MODULE := libstagefright_soft_vorbisdec
 LOCAL_MODULE_TAGS := optional
 
+LOCAL_CFLAGS += -Werror
+
 include $(BUILD_SHARED_LIBRARY)
diff --git a/media/libstagefright/foundation/ANetworkSession.cpp b/media/libstagefright/foundation/ANetworkSession.cpp
index e629588..08c4a87 100644
--- a/media/libstagefright/foundation/ANetworkSession.cpp
+++ b/media/libstagefright/foundation/ANetworkSession.cpp
@@ -521,7 +521,7 @@
     return err;
 }
 
-void ANetworkSession::Session::dumpFragmentStats(const Fragment &frag) {
+void ANetworkSession::Session::dumpFragmentStats(const Fragment & /* frag */) {
 #if 0
     int64_t nowUs = ALooper::GetNowUs();
     int64_t delayMs = (nowUs - frag.mTimeUs) / 1000ll;
diff --git a/media/libstagefright/foundation/Android.mk b/media/libstagefright/foundation/Android.mk
index ad2dab5..90a6a23 100644
--- a/media/libstagefright/foundation/Android.mk
+++ b/media/libstagefright/foundation/Android.mk
@@ -24,7 +24,7 @@
         libutils          \
         liblog
 
-LOCAL_CFLAGS += -Wno-multichar
+LOCAL_CFLAGS += -Wno-multichar -Werror
 
 LOCAL_MODULE:= libstagefright_foundation
 
diff --git a/media/libstagefright/foundation/base64.cpp b/media/libstagefright/foundation/base64.cpp
index d5fb4e0..dcf5bef 100644
--- a/media/libstagefright/foundation/base64.cpp
+++ b/media/libstagefright/foundation/base64.cpp
@@ -33,6 +33,10 @@
 
         if (n >= 2 && s.c_str()[n - 2] == '=') {
             padding = 2;
+
+            if (n >= 3 && s.c_str()[n - 3] == '=') {
+                padding = 3;
+            }
         }
     }
 
@@ -71,7 +75,7 @@
         if (((i + 1) % 4) == 0) {
             out[j++] = (accum >> 16);
 
-            if (j < outLen) { out[j++] = (accum >> 8) & 0xff; } 
+            if (j < outLen) { out[j++] = (accum >> 8) & 0xff; }
             if (j < outLen) { out[j++] = accum & 0xff; }
 
             accum = 0;
diff --git a/media/libstagefright/http/Android.mk b/media/libstagefright/http/Android.mk
new file mode 100644
index 0000000..7f3307d
--- /dev/null
+++ b/media/libstagefright/http/Android.mk
@@ -0,0 +1,28 @@
+LOCAL_PATH:= $(call my-dir)
+
+ifneq ($(TARGET_BUILD_PDK), true)
+
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES:=       \
+        HTTPHelper.cpp          \
+
+LOCAL_C_INCLUDES:= \
+	$(TOP)/frameworks/av/media/libstagefright \
+	$(TOP)/frameworks/native/include/media/openmax \
+	$(TOP)/frameworks/base/core/jni \
+
+LOCAL_SHARED_LIBRARIES := \
+	libstagefright liblog libutils libbinder libstagefright_foundation \
+        libandroid_runtime \
+        libmedia
+
+LOCAL_MODULE:= libstagefright_http_support
+
+LOCAL_CFLAGS += -Wno-multichar
+
+LOCAL_CFLAGS += -Werror
+
+include $(BUILD_SHARED_LIBRARY)
+
+endif
diff --git a/media/libstagefright/http/HTTPHelper.cpp b/media/libstagefright/http/HTTPHelper.cpp
new file mode 100644
index 0000000..77845e2
--- /dev/null
+++ b/media/libstagefright/http/HTTPHelper.cpp
@@ -0,0 +1,70 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "HTTPHelper"
+#include <utils/Log.h>
+
+#include "HTTPHelper.h"
+
+#include "android_runtime/AndroidRuntime.h"
+#include "android_util_Binder.h"
+#include <media/IMediaHTTPService.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <nativehelper/ScopedLocalRef.h>
+#include "jni.h"
+
+namespace android {
+
+sp<IMediaHTTPService> CreateHTTPServiceInCurrentJavaContext() {
+    if (AndroidRuntime::getJavaVM() == NULL) {
+        ALOGE("CreateHTTPServiceInCurrentJavaContext called outside "
+              "JAVA environment.");
+        return NULL;
+    }
+
+    JNIEnv *env = AndroidRuntime::getJNIEnv();
+
+    ScopedLocalRef<jclass> clazz(
+            env, env->FindClass("android/media/MediaHTTPService"));
+    CHECK(clazz.get() != NULL);
+
+    jmethodID constructID = env->GetMethodID(clazz.get(), "<init>", "()V");
+    CHECK(constructID != NULL);
+
+    ScopedLocalRef<jobject> httpServiceObj(
+            env, env->NewObject(clazz.get(), constructID));
+
+    sp<IMediaHTTPService> httpService;
+    if (httpServiceObj.get() != NULL) {
+        jmethodID asBinderID =
+            env->GetMethodID(clazz.get(), "asBinder", "()Landroid/os/IBinder;");
+        CHECK(asBinderID != NULL);
+
+        ScopedLocalRef<jobject> httpServiceBinderObj(
+                env, env->CallObjectMethod(httpServiceObj.get(), asBinderID));
+        CHECK(httpServiceBinderObj.get() != NULL);
+
+        sp<IBinder> binder =
+            ibinderForJavaObject(env, httpServiceBinderObj.get());
+
+        httpService = interface_cast<IMediaHTTPService>(binder);
+    }
+
+    return httpService;
+}
+
+}  // namespace android
diff --git a/media/libstagefright/http/HTTPHelper.h b/media/libstagefright/http/HTTPHelper.h
new file mode 100644
index 0000000..8aef115
--- /dev/null
+++ b/media/libstagefright/http/HTTPHelper.h
@@ -0,0 +1,31 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef HTTP_HELPER_H_
+
+#define HTTP_HELPER_H_
+
+#include <utils/RefBase.h>
+
+namespace android {
+
+struct IMediaHTTPService;
+
+sp<IMediaHTTPService> CreateHTTPServiceInCurrentJavaContext();
+
+}  // namespace android
+
+#endif  // HTTP_HELPER_H_
diff --git a/media/libstagefright/http/MediaHTTP.cpp b/media/libstagefright/http/MediaHTTP.cpp
new file mode 100644
index 0000000..157d967
--- /dev/null
+++ b/media/libstagefright/http/MediaHTTP.cpp
@@ -0,0 +1,201 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "MediaHTTP"
+#include <utils/Log.h>
+
+#include <media/stagefright/MediaHTTP.h>
+
+#include <binder/IServiceManager.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/ALooper.h>
+#include <media/stagefright/Utils.h>
+
+#include <media/IMediaHTTPConnection.h>
+
+namespace android {
+
+MediaHTTP::MediaHTTP(const sp<IMediaHTTPConnection> &conn)
+    : mInitCheck(NO_INIT),
+      mHTTPConnection(conn),
+      mCachedSizeValid(false),
+      mCachedSize(0ll),
+      mDrmManagerClient(NULL) {
+    mInitCheck = OK;
+}
+
+MediaHTTP::~MediaHTTP() {
+    clearDRMState_l();
+}
+
+status_t MediaHTTP::connect(
+        const char *uri,
+        const KeyedVector<String8, String8> *headers,
+        off64_t /* offset */) {
+    if (mInitCheck != OK) {
+        return mInitCheck;
+    }
+
+    KeyedVector<String8, String8> extHeaders;
+    if (headers != NULL) {
+        extHeaders = *headers;
+    }
+    extHeaders.add(String8("User-Agent"), String8(MakeUserAgent().c_str()));
+
+    bool success = mHTTPConnection->connect(uri, &extHeaders);
+
+    mLastHeaders = extHeaders;
+    mLastURI = uri;
+
+    mCachedSizeValid = false;
+
+    return success ? OK : UNKNOWN_ERROR;
+}
+
+void MediaHTTP::disconnect() {
+    if (mInitCheck != OK) {
+        return;
+    }
+
+    mHTTPConnection->disconnect();
+}
+
+status_t MediaHTTP::initCheck() const {
+    return mInitCheck;
+}
+
+ssize_t MediaHTTP::readAt(off64_t offset, void *data, size_t size) {
+    if (mInitCheck != OK) {
+        return mInitCheck;
+    }
+
+    int64_t startTimeUs = ALooper::GetNowUs();
+
+    size_t numBytesRead = 0;
+    while (numBytesRead < size) {
+        size_t copy = size - numBytesRead;
+
+        if (copy > 64 * 1024) {
+            // limit the buffer sizes transferred across binder boundaries
+            // to avoid spurious transaction failures.
+            copy = 64 * 1024;
+        }
+
+        ssize_t n = mHTTPConnection->readAt(
+                offset + numBytesRead, (uint8_t *)data + numBytesRead, copy);
+
+        if (n < 0) {
+            return n;
+        } else if (n == 0) {
+            break;
+        }
+
+        numBytesRead += n;
+    }
+
+    int64_t delayUs = ALooper::GetNowUs() - startTimeUs;
+
+    addBandwidthMeasurement(numBytesRead, delayUs);
+
+    return numBytesRead;
+}
+
+status_t MediaHTTP::getSize(off64_t *size) {
+    if (mInitCheck != OK) {
+        return mInitCheck;
+    }
+
+    // Caching the returned size so that it stays valid even after a
+    // disconnect. NuCachedSource2 relies on this.
+
+    if (!mCachedSizeValid) {
+        mCachedSize = mHTTPConnection->getSize();
+        mCachedSizeValid = true;
+    }
+
+    *size = mCachedSize;
+
+    return *size < 0 ? *size : OK;
+}
+
+uint32_t MediaHTTP::flags() {
+    return kWantsPrefetching | kIsHTTPBasedSource;
+}
+
+status_t MediaHTTP::reconnectAtOffset(off64_t offset) {
+    return connect(mLastURI.c_str(), &mLastHeaders, offset);
+}
+
+// DRM...
+
+sp<DecryptHandle> MediaHTTP::DrmInitialization(const char* mime) {
+    if (mDrmManagerClient == NULL) {
+        mDrmManagerClient = new DrmManagerClient();
+    }
+
+    if (mDrmManagerClient == NULL) {
+        return NULL;
+    }
+
+    if (mDecryptHandle == NULL) {
+        mDecryptHandle = mDrmManagerClient->openDecryptSession(
+                String8(mLastURI.c_str()), mime);
+    }
+
+    if (mDecryptHandle == NULL) {
+        delete mDrmManagerClient;
+        mDrmManagerClient = NULL;
+    }
+
+    return mDecryptHandle;
+}
+
+void MediaHTTP::getDrmInfo(
+        sp<DecryptHandle> &handle, DrmManagerClient **client) {
+    handle = mDecryptHandle;
+    *client = mDrmManagerClient;
+}
+
+String8 MediaHTTP::getUri() {
+    return String8(mLastURI.c_str());
+}
+
+String8 MediaHTTP::getMIMEType() const {
+    if (mInitCheck != OK) {
+        return String8("application/octet-stream");
+    }
+
+    String8 mimeType;
+    status_t err = mHTTPConnection->getMIMEType(&mimeType);
+
+    if (err != OK) {
+        return String8("application/octet-stream");
+    }
+
+    return mimeType;
+}
+
+void MediaHTTP::clearDRMState_l() {
+    if (mDecryptHandle != NULL) {
+        // To release mDecryptHandle
+        CHECK(mDrmManagerClient);
+        mDrmManagerClient->closeDecryptSession(mDecryptHandle);
+        mDecryptHandle = NULL;
+    }
+}
+
+}  // namespace android
diff --git a/media/libstagefright/httplive/Android.mk b/media/libstagefright/httplive/Android.mk
index f3529f9..e8d558c 100644
--- a/media/libstagefright/httplive/Android.mk
+++ b/media/libstagefright/httplive/Android.mk
@@ -13,6 +13,8 @@
 	$(TOP)/frameworks/native/include/media/openmax \
 	$(TOP)/external/openssl/include
 
+LOCAL_CFLAGS += -Werror
+
 LOCAL_SHARED_LIBRARIES := \
         libbinder \
         libcrypto \
diff --git a/media/libstagefright/httplive/LiveSession.cpp b/media/libstagefright/httplive/LiveSession.cpp
index 233db44..149a817 100644
--- a/media/libstagefright/httplive/LiveSession.cpp
+++ b/media/libstagefright/httplive/LiveSession.cpp
@@ -27,6 +27,8 @@
 #include "mpeg2ts/AnotherPacketSource.h"
 
 #include <cutils/properties.h>
+#include <media/IMediaHTTPConnection.h>
+#include <media/IMediaHTTPService.h>
 #include <media/stagefright/foundation/hexdump.h>
 #include <media/stagefright/foundation/ABuffer.h>
 #include <media/stagefright/foundation/ADebug.h>
@@ -34,6 +36,7 @@
 #include <media/stagefright/DataSource.h>
 #include <media/stagefright/FileSource.h>
 #include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/MediaHTTP.h>
 #include <media/stagefright/MetaData.h>
 #include <media/stagefright/Utils.h>
 
@@ -44,17 +47,13 @@
 namespace android {
 
 LiveSession::LiveSession(
-        const sp<AMessage> &notify, uint32_t flags, bool uidValid, uid_t uid)
+        const sp<AMessage> &notify, uint32_t flags,
+        const sp<IMediaHTTPService> &httpService)
     : mNotify(notify),
       mFlags(flags),
-      mUIDValid(uidValid),
-      mUID(uid),
+      mHTTPService(httpService),
       mInPreparationPhase(true),
-      mHTTPDataSource(
-              HTTPBase::Create(
-                  (mFlags & kFlagIncognito)
-                    ? HTTPBase::kFlagIncognito
-                    : 0)),
+      mHTTPDataSource(new MediaHTTP(mHTTPService->makeHTTPConnection())),
       mPrevBandwidthIndex(-1),
       mStreamMask(0),
       mCheckBandwidthGeneration(0),
@@ -62,10 +61,6 @@
       mRealTimeBaseUs(0ll),
       mReconfigurationInProgress(false),
       mDisconnectReplyID(0) {
-    if (mUIDValid) {
-        mHTTPDataSource->setUID(mUID);
-    }
-
     mPacketSources.add(
             STREAMTYPE_AUDIO, new AnotherPacketSource(NULL /* meta */));
 
diff --git a/media/libstagefright/httplive/LiveSession.h b/media/libstagefright/httplive/LiveSession.h
index 99b480a8..ee10e70 100644
--- a/media/libstagefright/httplive/LiveSession.h
+++ b/media/libstagefright/httplive/LiveSession.h
@@ -28,6 +28,7 @@
 struct AnotherPacketSource;
 struct DataSource;
 struct HTTPBase;
+struct IMediaHTTPService;
 struct LiveDataSource;
 struct M3UParser;
 struct PlaylistFetcher;
@@ -40,7 +41,8 @@
     };
     LiveSession(
             const sp<AMessage> &notify,
-            uint32_t flags = 0, bool uidValid = false, uid_t uid = 0);
+            uint32_t flags,
+            const sp<IMediaHTTPService> &httpService);
 
     enum StreamType {
         STREAMTYPE_AUDIO        = 1,
@@ -107,8 +109,7 @@
 
     sp<AMessage> mNotify;
     uint32_t mFlags;
-    bool mUIDValid;
-    uid_t mUID;
+    sp<IMediaHTTPService> mHTTPService;
 
     bool mInPreparationPhase;
 
diff --git a/media/libstagefright/httplive/M3UParser.cpp b/media/libstagefright/httplive/M3UParser.cpp
index 243888c..39d80fc 100644
--- a/media/libstagefright/httplive/M3UParser.cpp
+++ b/media/libstagefright/httplive/M3UParser.cpp
@@ -125,7 +125,7 @@
                 mSelectedIndex = strtoul(value, &end, 10);
                 CHECK(end > value && *end == '\0');
 
-                if (mSelectedIndex >= mMediaItems.size()) {
+                if (mSelectedIndex >= (ssize_t)mMediaItems.size()) {
                     mSelectedIndex = mMediaItems.size() - 1;
                 }
             } else {
@@ -165,14 +165,14 @@
             ALOGE("track %d does not exist", index);
             return INVALID_OPERATION;
         }
-        if (mSelectedIndex == index) {
+        if (mSelectedIndex == (ssize_t)index) {
             ALOGE("track %d already selected", index);
             return BAD_VALUE;
         }
         ALOGV("selected track %d", index);
         mSelectedIndex = index;
     } else {
-        if (mSelectedIndex != index) {
+        if (mSelectedIndex != (ssize_t)index) {
             ALOGE("track %d is not selected", index);
             return BAD_VALUE;
         }
@@ -416,22 +416,32 @@
     } else {
         // URL is a relative path
 
-        size_t n = strlen(baseURL);
-        if (baseURL[n - 1] == '/') {
-            out->setTo(baseURL);
-            out->append(url);
+        // Check for a possible query string
+        const char *qsPos = strchr(baseURL, '?');
+        size_t end;
+        if (qsPos != NULL) {
+            end = qsPos - baseURL;
         } else {
-            const char *slashPos = strrchr(baseURL, '/');
-
-            if (slashPos > &baseURL[6]) {
-                out->setTo(baseURL, slashPos - baseURL);
-            } else {
-                out->setTo(baseURL);
-            }
-
-            out->append("/");
-            out->append(url);
+            end = strlen(baseURL);
         }
+        // Check for the last slash before a potential query string
+        for (ssize_t pos = end - 1; pos >= 0; pos--) {
+            if (baseURL[pos] == '/') {
+                end = pos;
+                break;
+            }
+        }
+
+        // Check whether the found slash actually is part of the path
+        // and not part of the "http://".
+        if (end > 6) {
+            out->setTo(baseURL, end);
+        } else {
+            out->setTo(baseURL);
+        }
+
+        out->append("/");
+        out->append(url);
     }
 
     ALOGV("base:'%s', url:'%s' => '%s'", baseURL, url, out->c_str());
@@ -608,7 +618,7 @@
     if (meta->get() == NULL) {
         *meta = new AMessage;
     }
-    (*meta)->setInt64(key, (int64_t)x * 1E6);
+    (*meta)->setInt64(key, (int64_t)(x * 1E6));
 
     return OK;
 }
diff --git a/media/libstagefright/httplive/PlaylistFetcher.cpp b/media/libstagefright/httplive/PlaylistFetcher.cpp
index f095987..4d29366 100644
--- a/media/libstagefright/httplive/PlaylistFetcher.cpp
+++ b/media/libstagefright/httplive/PlaylistFetcher.cpp
@@ -602,7 +602,7 @@
 
         if (mPlaylist->isComplete() || mPlaylist->isEvent()) {
             mSeqNumber = getSeqNumberForTime(mStartTimeUs);
-            ALOGV("Initial sequence number for time %lld is %ld from (%ld .. %ld)",
+            ALOGV("Initial sequence number for time %lld is %d from (%d .. %d)",
                     mStartTimeUs, mSeqNumber, firstSeqNumberInPlaylist,
                     lastSeqNumberInPlaylist);
         } else {
@@ -611,7 +611,7 @@
             if (mSeqNumber < firstSeqNumberInPlaylist) {
                 mSeqNumber = firstSeqNumberInPlaylist;
             }
-            ALOGV("Initial sequence number for live event %ld from (%ld .. %ld)",
+            ALOGV("Initial sequence number for live event %d from (%d .. %d)",
                     mSeqNumber, firstSeqNumberInPlaylist,
                     lastSeqNumberInPlaylist);
         }
@@ -635,7 +635,8 @@
                 if (delayUs > kMaxMonitorDelayUs) {
                     delayUs = kMaxMonitorDelayUs;
                 }
-                ALOGV("sequence number high: %ld from (%ld .. %ld), monitor in %lld (retry=%d)",
+                ALOGV("sequence number high: %d from (%d .. %d), "
+                      "monitor in %lld (retry=%d)",
                         mSeqNumber, firstSeqNumberInPlaylist,
                         lastSeqNumberInPlaylist, delayUs, mNumRetries);
                 postMonitorQueue(delayUs);
diff --git a/media/libstagefright/id3/Android.mk b/media/libstagefright/id3/Android.mk
index bf6f7bb..2194c38 100644
--- a/media/libstagefright/id3/Android.mk
+++ b/media/libstagefright/id3/Android.mk
@@ -4,6 +4,8 @@
 LOCAL_SRC_FILES := \
 	ID3.cpp
 
+LOCAL_CFLAGS += -Werror
+
 LOCAL_MODULE := libstagefright_id3
 
 include $(BUILD_STATIC_LIBRARY)
@@ -15,6 +17,8 @@
 LOCAL_SRC_FILES := \
 	testid3.cpp
 
+LOCAL_CFLAGS += -Werror
+
 LOCAL_SHARED_LIBRARIES := \
 	libstagefright libutils liblog libbinder libstagefright_foundation
 
diff --git a/media/libstagefright/id3/ID3.cpp b/media/libstagefright/id3/ID3.cpp
index a486522..f0f203c 100644
--- a/media/libstagefright/id3/ID3.cpp
+++ b/media/libstagefright/id3/ID3.cpp
@@ -56,14 +56,14 @@
     DISALLOW_EVIL_CONSTRUCTORS(MemorySource);
 };
 
-ID3::ID3(const sp<DataSource> &source, bool ignoreV1)
+ID3::ID3(const sp<DataSource> &source, bool ignoreV1, off64_t offset)
     : mIsValid(false),
       mData(NULL),
       mSize(0),
       mFirstFrameOffset(0),
       mVersion(ID3_UNKNOWN),
       mRawSize(0) {
-    mIsValid = parseV2(source);
+    mIsValid = parseV2(source, offset);
 
     if (!mIsValid && !ignoreV1) {
         mIsValid = parseV1(source);
@@ -79,7 +79,7 @@
       mRawSize(0) {
     sp<MemorySource> source = new MemorySource(data, size);
 
-    mIsValid = parseV2(source);
+    mIsValid = parseV2(source, 0);
 
     if (!mIsValid && !ignoreV1) {
         mIsValid = parseV1(source);
@@ -115,7 +115,7 @@
     return true;
 }
 
-bool ID3::parseV2(const sp<DataSource> &source) {
+bool ID3::parseV2(const sp<DataSource> &source, off64_t offset) {
 struct id3_header {
     char id[3];
     uint8_t version_major;
@@ -126,7 +126,7 @@
 
     id3_header header;
     if (source->readAt(
-                0, &header, sizeof(header)) != (ssize_t)sizeof(header)) {
+                offset, &header, sizeof(header)) != (ssize_t)sizeof(header)) {
         return false;
     }
 
@@ -185,7 +185,7 @@
     mSize = size;
     mRawSize = mSize + sizeof(header);
 
-    if (source->readAt(sizeof(header), mData, mSize) != (ssize_t)mSize) {
+    if (source->readAt(offset + sizeof(header), mData, mSize) != (ssize_t)mSize) {
         free(mData);
         mData = NULL;
 
diff --git a/media/libstagefright/id3/testid3.cpp b/media/libstagefright/id3/testid3.cpp
index bc4572c..b2f4188 100644
--- a/media/libstagefright/id3/testid3.cpp
+++ b/media/libstagefright/id3/testid3.cpp
@@ -33,7 +33,7 @@
     const uint8_t *data = (const uint8_t *)_data;
     size_t offset = 0;
     while (offset < size) {
-        printf("0x%04x  ", offset);
+        printf("0x%04zx  ", offset);
 
         size_t n = size - offset;
         if (n > 16) {
@@ -101,7 +101,7 @@
         const void *data = tag.getAlbumArt(&dataSize, &mime);
 
         if (data) {
-            printf("found album art: size=%d mime='%s'\n", dataSize,
+            printf("found album art: size=%zu mime='%s'\n", dataSize,
                    mime.string());
 
             hexdump(data, dataSize > 128 ? 128 : dataSize);
diff --git a/media/libstagefright/include/AwesomePlayer.h b/media/libstagefright/include/AwesomePlayer.h
index 271df8e..a81bbba 100644
--- a/media/libstagefright/include/AwesomePlayer.h
+++ b/media/libstagefright/include/AwesomePlayer.h
@@ -63,6 +63,7 @@
     void setUID(uid_t uid);
 
     status_t setDataSource(
+            const sp<IMediaHTTPService> &httpService,
             const char *uri,
             const KeyedVector<String8, String8> *headers = NULL);
 
@@ -159,6 +160,7 @@
     SystemTimeSource mSystemTimeSource;
     TimeSource *mTimeSource;
 
+    sp<IMediaHTTPService> mHTTPService;
     String8 mUri;
     KeyedVector<String8, String8> mUriHeaders;
 
@@ -247,6 +249,7 @@
     sp<MediaExtractor> mExtractor;
 
     status_t setDataSource_l(
+            const sp<IMediaHTTPService> &httpService,
             const char *uri,
             const KeyedVector<String8, String8> *headers = NULL);
 
diff --git a/media/libstagefright/include/ChromiumHTTPDataSource.h b/media/libstagefright/include/ChromiumHTTPDataSource.h
deleted file mode 100644
index 785f939..0000000
--- a/media/libstagefright/include/ChromiumHTTPDataSource.h
+++ /dev/null
@@ -1,124 +0,0 @@
-/*
- * Copyright (C) 2011 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef CHROME_HTTP_DATA_SOURCE_H_
-
-#define CHROME_HTTP_DATA_SOURCE_H_
-
-#include <media/stagefright/foundation/AString.h>
-#include <utils/threads.h>
-
-#include "HTTPBase.h"
-
-namespace android {
-
-struct SfDelegate;
-
-struct ChromiumHTTPDataSource : public HTTPBase {
-    ChromiumHTTPDataSource(uint32_t flags = 0);
-
-    virtual status_t connect(
-            const char *uri,
-            const KeyedVector<String8, String8> *headers = NULL,
-            off64_t offset = 0);
-
-    virtual void disconnect();
-
-    virtual status_t initCheck() const;
-
-    virtual ssize_t readAt(off64_t offset, void *data, size_t size);
-    virtual status_t getSize(off64_t *size);
-    virtual uint32_t flags();
-
-    virtual sp<DecryptHandle> DrmInitialization(const char *mime);
-
-    virtual void getDrmInfo(sp<DecryptHandle> &handle, DrmManagerClient **client);
-
-    virtual String8 getUri();
-
-    virtual String8 getMIMEType() const;
-
-    virtual status_t reconnectAtOffset(off64_t offset);
-
-    static status_t UpdateProxyConfig(
-            const char *host, int32_t port, const char *exclusionList);
-
-protected:
-    virtual ~ChromiumHTTPDataSource();
-
-private:
-    friend struct SfDelegate;
-
-    enum State {
-        DISCONNECTED,
-        CONNECTING,
-        CONNECTED,
-        READING,
-        DISCONNECTING
-    };
-
-    const uint32_t mFlags;
-
-    mutable Mutex mLock;
-    Condition mCondition;
-
-    State mState;
-
-    SfDelegate *mDelegate;
-
-    AString mURI;
-    KeyedVector<String8, String8> mHeaders;
-
-    off64_t mCurrentOffset;
-
-    // Any connection error or the result of a read operation
-    // (for the lattter this is the number of bytes read, if successful).
-    ssize_t mIOResult;
-
-    int64_t mContentSize;
-
-    String8 mContentType;
-
-    sp<DecryptHandle> mDecryptHandle;
-    DrmManagerClient *mDrmManagerClient;
-
-    void disconnect_l();
-
-    status_t connect_l(
-            const char *uri,
-            const KeyedVector<String8, String8> *headers,
-            off64_t offset);
-
-    static void InitiateRead(
-            ChromiumHTTPDataSource *me, void *data, size_t size);
-
-    void initiateRead(void *data, size_t size);
-
-    void onConnectionEstablished(
-            int64_t contentSize, const char *contentType);
-
-    void onConnectionFailed(status_t err);
-    void onReadCompleted(ssize_t size);
-    void onDisconnectComplete();
-
-    void clearDRMState_l();
-
-    DISALLOW_EVIL_CONSTRUCTORS(ChromiumHTTPDataSource);
-};
-
-}  // namespace android
-
-#endif  // CHROME_HTTP_DATA_SOURCE_H_
diff --git a/media/libstagefright/include/FragmentedMP4Parser.h b/media/libstagefright/include/FragmentedMP4Parser.h
deleted file mode 100644
index dbe02b8..0000000
--- a/media/libstagefright/include/FragmentedMP4Parser.h
+++ /dev/null
@@ -1,274 +0,0 @@
-/*
- * Copyright (C) 2012 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef PARSER_H_
-
-#define PARSER_H_
-
-#include <media/stagefright/foundation/AHandler.h>
-#include <media/stagefright/DataSource.h>
-#include <utils/Vector.h>
-
-namespace android {
-
-struct ABuffer;
-
-struct FragmentedMP4Parser : public AHandler {
-    struct Source : public RefBase {
-        Source() {}
-
-        virtual ssize_t readAt(off64_t offset, void *data, size_t size) = 0;
-        virtual bool isSeekable() = 0;
-
-        protected:
-        virtual ~Source() {}
-
-        private:
-        DISALLOW_EVIL_CONSTRUCTORS(Source);
-    };
-
-    FragmentedMP4Parser();
-
-    void start(const char *filename);
-    void start(const sp<Source> &source);
-    void start(sp<DataSource> &source);
-
-    sp<AMessage> getFormat(bool audio, bool synchronous = false);
-    status_t dequeueAccessUnit(bool audio, sp<ABuffer> *accessUnit, bool synchronous = false);
-    status_t seekTo(bool audio, int64_t timeUs);
-    bool isSeekable() const;
-
-    virtual void onMessageReceived(const sp<AMessage> &msg);
-
-protected:
-    virtual ~FragmentedMP4Parser();
-
-private:
-    enum {
-        kWhatStart,
-        kWhatProceed,
-        kWhatReadMore,
-        kWhatGetFormat,
-        kWhatDequeueAccessUnit,
-        kWhatSeekTo,
-    };
-
-    struct TrackFragment;
-    struct DynamicTrackFragment;
-    struct StaticTrackFragment;
-
-    struct DispatchEntry {
-        uint32_t mType;
-        uint32_t mParentType;
-        status_t (FragmentedMP4Parser::*mHandler)(uint32_t, size_t, uint64_t);
-    };
-
-    struct Container {
-        uint64_t mOffset;
-        uint64_t mBytesRemaining;
-        uint32_t mType;
-        bool mExtendsToEOF;
-    };
-
-    struct SampleDescription {
-        uint32_t mType;
-        uint16_t mDataRefIndex;
-
-        sp<AMessage> mFormat;
-    };
-
-    struct SampleInfo {
-        off64_t mOffset;
-        size_t mSize;
-        uint32_t mPresentationTime;
-        size_t mSampleDescIndex;
-        uint32_t mFlags;
-    };
-
-    struct MediaDataInfo {
-        sp<ABuffer> mBuffer;
-        off64_t mOffset;
-    };
-
-    struct SidxEntry {
-        size_t mSize;
-        uint32_t mDurationUs;
-    };
-
-    struct TrackInfo {
-        enum Flags {
-            kTrackEnabled     = 0x01,
-            kTrackInMovie     = 0x02,
-            kTrackInPreview   = 0x04,
-        };
-
-        uint32_t mTrackID;
-        uint32_t mFlags;
-        uint32_t mDuration;  // This is the duration in terms of movie timescale!
-        uint64_t mSidxDuration; // usec, from sidx box, which can use a different timescale
-
-        uint32_t mMediaTimeScale;
-
-        uint32_t mMediaHandlerType;
-        Vector<SampleDescription> mSampleDescs;
-
-        // from track extends:
-        uint32_t mDefaultSampleDescriptionIndex;
-        uint32_t mDefaultSampleDuration;
-        uint32_t mDefaultSampleSize;
-        uint32_t mDefaultSampleFlags;
-
-        uint32_t mDecodingTime;
-
-        Vector<SidxEntry> mSidx;
-        sp<StaticTrackFragment> mStaticFragment;
-        List<sp<TrackFragment> > mFragments;
-    };
-
-    struct TrackFragmentHeaderInfo {
-        enum Flags {
-            kBaseDataOffsetPresent         = 0x01,
-            kSampleDescriptionIndexPresent = 0x02,
-            kDefaultSampleDurationPresent  = 0x08,
-            kDefaultSampleSizePresent      = 0x10,
-            kDefaultSampleFlagsPresent     = 0x20,
-            kDurationIsEmpty               = 0x10000,
-        };
-
-        uint32_t mTrackID;
-        uint32_t mFlags;
-        uint64_t mBaseDataOffset;
-        uint32_t mSampleDescriptionIndex;
-        uint32_t mDefaultSampleDuration;
-        uint32_t mDefaultSampleSize;
-        uint32_t mDefaultSampleFlags;
-
-        uint64_t mDataOffset;
-    };
-
-    static const DispatchEntry kDispatchTable[];
-
-    sp<Source> mSource;
-    off_t mBufferPos;
-    bool mSuspended;
-    bool mDoneWithMoov;
-    off_t mFirstMoofOffset; // used as the starting point for offsets calculated from the sidx box
-    sp<ABuffer> mBuffer;
-    Vector<Container> mStack;
-    KeyedVector<uint32_t, TrackInfo> mTracks;  // TrackInfo by trackID
-    Vector<MediaDataInfo> mMediaData;
-
-    uint32_t mCurrentTrackID;
-
-    status_t mFinalResult;
-
-    TrackFragmentHeaderInfo mTrackFragmentHeaderInfo;
-
-    status_t onProceed();
-    status_t onDequeueAccessUnit(size_t trackIndex, sp<ABuffer> *accessUnit);
-    status_t onSeekTo(bool wantAudio, int64_t position);
-
-    void enter(off64_t offset, uint32_t type, uint64_t size);
-
-    uint16_t readU16(size_t offset);
-    uint32_t readU32(size_t offset);
-    uint64_t readU64(size_t offset);
-    void skip(off_t distance);
-    status_t need(size_t size);
-    bool fitsContainer(uint64_t size) const;
-
-    status_t parseTrackHeader(
-            uint32_t type, size_t offset, uint64_t size);
-
-    status_t parseMediaHeader(
-            uint32_t type, size_t offset, uint64_t size);
-
-    status_t parseMediaHandler(
-            uint32_t type, size_t offset, uint64_t size);
-
-    status_t parseTrackExtends(
-            uint32_t type, size_t offset, uint64_t size);
-
-    status_t parseTrackFragmentHeader(
-            uint32_t type, size_t offset, uint64_t size);
-
-    status_t parseTrackFragmentRun(
-            uint32_t type, size_t offset, uint64_t size);
-
-    status_t parseVisualSampleEntry(
-            uint32_t type, size_t offset, uint64_t size);
-
-    status_t parseAudioSampleEntry(
-            uint32_t type, size_t offset, uint64_t size);
-
-    status_t parseSampleSizes(
-            uint32_t type, size_t offset, uint64_t size);
-
-    status_t parseCompactSampleSizes(
-            uint32_t type, size_t offset, uint64_t size);
-
-    status_t parseSampleToChunk(
-            uint32_t type, size_t offset, uint64_t size);
-
-    status_t parseChunkOffsets(
-            uint32_t type, size_t offset, uint64_t size);
-
-    status_t parseChunkOffsets64(
-            uint32_t type, size_t offset, uint64_t size);
-
-    status_t parseAVCCodecSpecificData(
-            uint32_t type, size_t offset, uint64_t size);
-
-    status_t parseESDSCodecSpecificData(
-            uint32_t type, size_t offset, uint64_t size);
-
-    status_t parseMediaData(
-            uint32_t type, size_t offset, uint64_t size);
-
-    status_t parseSegmentIndex(
-            uint32_t type, size_t offset, uint64_t size);
-
-    TrackInfo *editTrack(uint32_t trackID, bool createIfNecessary = false);
-
-    ssize_t findTrack(bool wantAudio) const;
-
-    status_t makeAccessUnit(
-            TrackInfo *info,
-            const SampleInfo &sample,
-            const MediaDataInfo &mdatInfo,
-            sp<ABuffer> *accessUnit);
-
-    status_t getSample(
-            TrackInfo *info,
-            sp<TrackFragment> *fragment,
-            SampleInfo *sampleInfo);
-
-    static int CompareSampleLocation(
-        const SampleInfo &sample, const MediaDataInfo &mdatInfo);
-
-    void resumeIfNecessary();
-
-    void copyBuffer(
-            sp<ABuffer> *dst,
-            size_t offset, uint64_t size) const;
-
-    DISALLOW_EVIL_CONSTRUCTORS(FragmentedMP4Parser);
-};
-
-}  // namespace android
-
-#endif  // PARSER_H_
-
diff --git a/media/libstagefright/include/HTTPBase.h b/media/libstagefright/include/HTTPBase.h
index d4b7f9f..1c3cd5e 100644
--- a/media/libstagefright/include/HTTPBase.h
+++ b/media/libstagefright/include/HTTPBase.h
@@ -48,14 +48,6 @@
 
     virtual status_t setBandwidthStatCollectFreq(int32_t freqMs);
 
-    static status_t UpdateProxyConfig(
-            const char *host, int32_t port, const char *exclusionList);
-
-    void setUID(uid_t uid);
-    bool getUID(uid_t *uid) const;
-
-    static sp<HTTPBase> Create(uint32_t flags = 0);
-
     static void RegisterSocketUserTag(int sockfd, uid_t uid, uint32_t kTag);
     static void UnRegisterSocketUserTag(int sockfd);
 
@@ -87,9 +79,6 @@
     int32_t mPrevEstimatedBandWidthKbps;
     int32_t mBandWidthCollectFreqMs;
 
-    bool mUIDValid;
-    uid_t mUID;
-
     DISALLOW_EVIL_CONSTRUCTORS(HTTPBase);
 };
 
diff --git a/media/libstagefright/include/ID3.h b/media/libstagefright/include/ID3.h
index cca83ab..e83f3ef 100644
--- a/media/libstagefright/include/ID3.h
+++ b/media/libstagefright/include/ID3.h
@@ -35,7 +35,7 @@
         ID3_V2_4,
     };
 
-    ID3(const sp<DataSource> &source, bool ignoreV1 = false);
+    ID3(const sp<DataSource> &source, bool ignoreV1 = false, off64_t offset = 0);
     ID3(const uint8_t *data, size_t size, bool ignoreV1 = false);
     ~ID3();
 
@@ -86,7 +86,7 @@
     size_t mRawSize;
 
     bool parseV1(const sp<DataSource> &source);
-    bool parseV2(const sp<DataSource> &source);
+    bool parseV2(const sp<DataSource> &source, off64_t offset);
     void removeUnsynchronization();
     bool removeUnsynchronizationV2_4(bool iTunesHack);
 
diff --git a/media/libstagefright/include/MPEG4Extractor.h b/media/libstagefright/include/MPEG4Extractor.h
index bd5e4b9..7b4bc6d 100644
--- a/media/libstagefright/include/MPEG4Extractor.h
+++ b/media/libstagefright/include/MPEG4Extractor.h
@@ -97,6 +97,7 @@
     status_t parseChunk(off64_t *offset, int depth);
     status_t parseITunesMetaData(off64_t offset, size_t size);
     status_t parse3GPPMetaData(off64_t offset, size_t size, int depth);
+    void parseID3v2MetaData(off64_t offset);
 
     status_t updateAudioTrackInfoFromESDS_MPEG4Audio(
             const void *esds_data, size_t esds_size);
diff --git a/media/libstagefright/include/SDPLoader.h b/media/libstagefright/include/SDPLoader.h
index ca59dc0..2c4f543 100644
--- a/media/libstagefright/include/SDPLoader.h
+++ b/media/libstagefright/include/SDPLoader.h
@@ -25,6 +25,7 @@
 namespace android {
 
 struct HTTPBase;
+struct IMediaHTTPService;
 
 struct SDPLoader : public AHandler {
     enum Flags {
@@ -34,7 +35,10 @@
     enum {
         kWhatSDPLoaded = 'sdpl'
     };
-    SDPLoader(const sp<AMessage> &notify, uint32_t flags = 0, bool uidValid = false, uid_t uid = 0);
+    SDPLoader(
+            const sp<AMessage> &notify,
+            uint32_t flags,
+            const sp<IMediaHTTPService> &httpService);
 
     void load(const char* url, const KeyedVector<String8, String8> *headers);
 
@@ -55,8 +59,6 @@
     sp<AMessage> mNotify;
     const char* mUrl;
     uint32_t mFlags;
-    bool mUIDValid;
-    uid_t mUID;
     sp<ALooper> mNetLooper;
     bool mCancelled;
 
diff --git a/media/libstagefright/include/StagefrightMetadataRetriever.h b/media/libstagefright/include/StagefrightMetadataRetriever.h
index b02ed0e..6632c27 100644
--- a/media/libstagefright/include/StagefrightMetadataRetriever.h
+++ b/media/libstagefright/include/StagefrightMetadataRetriever.h
@@ -33,6 +33,7 @@
     virtual ~StagefrightMetadataRetriever();
 
     virtual status_t setDataSource(
+            const sp<IMediaHTTPService> &httpService,
             const char *url,
             const KeyedVector<String8, String8> *headers);
 
diff --git a/media/libstagefright/include/TimedEventQueue.h b/media/libstagefright/include/TimedEventQueue.h
index 3e84256..2963150 100644
--- a/media/libstagefright/include/TimedEventQueue.h
+++ b/media/libstagefright/include/TimedEventQueue.h
@@ -122,7 +122,7 @@
     };
 
     struct StopEvent : public TimedEventQueue::Event {
-        virtual void fire(TimedEventQueue *queue, int64_t now_us) {
+        virtual void fire(TimedEventQueue *queue, int64_t /* now_us */) {
             queue->mStopped = true;
         }
     };
diff --git a/media/libstagefright/include/chromium_http_stub.h b/media/libstagefright/include/chromium_http_stub.h
deleted file mode 100644
index e0651a4..0000000
--- a/media/libstagefright/include/chromium_http_stub.h
+++ /dev/null
@@ -1,34 +0,0 @@
-/*
- * Copyright (C) 2012 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef CHROMIUM_HTTP_STUB_H_
-#define CHROMIUM_HTTP_STUB_H_
-
-#include <include/HTTPBase.h>
-#include <media/stagefright/DataSource.h>
-
-namespace android {
-extern "C" {
-HTTPBase *createChromiumHTTPDataSource(uint32_t flags);
-
-status_t UpdateChromiumHTTPDataSourceProxyConfig(
-        const char *host, int32_t port, const char *exclusionList);
-
-DataSource *createDataUriSource(const char *uri);
-}
-}
-
-#endif
diff --git a/media/libstagefright/matroska/Android.mk b/media/libstagefright/matroska/Android.mk
index 2d8c1e1..446ff8c 100644
--- a/media/libstagefright/matroska/Android.mk
+++ b/media/libstagefright/matroska/Android.mk
@@ -8,7 +8,7 @@
         $(TOP)/external/libvpx/libwebm \
         $(TOP)/frameworks/native/include/media/openmax \
 
-LOCAL_CFLAGS += -Wno-multichar
+LOCAL_CFLAGS += -Wno-multichar -Werror
 
 LOCAL_MODULE:= libstagefright_matroska
 
diff --git a/media/libstagefright/matroska/MatroskaExtractor.cpp b/media/libstagefright/matroska/MatroskaExtractor.cpp
index d260d0f..6f69d0b 100644
--- a/media/libstagefright/matroska/MatroskaExtractor.cpp
+++ b/media/libstagefright/matroska/MatroskaExtractor.cpp
@@ -193,7 +193,7 @@
     clearPendingFrames();
 }
 
-status_t MatroskaSource::start(MetaData *params) {
+status_t MatroskaSource::start(MetaData * /* params */) {
     mBlockIter.reset();
 
     return OK;
@@ -410,7 +410,7 @@
             // Accept the first key frame
             *actualFrameTimeUs = (block()->GetTime(mCluster) + 500LL) / 1000LL;
             ALOGV("Requested seek point: %lld actual: %lld",
-                  seekTimeUs, actualFrameTimeUs);
+                  seekTimeUs, *actualFrameTimeUs);
             break;
         }
     }
@@ -716,41 +716,61 @@
     return mIsLiveStreaming;
 }
 
+static int bytesForSize(size_t size) {
+    // use at most 28 bits (4 times 7)
+    CHECK(size <= 0xfffffff);
+
+    if (size > 0x1fffff) {
+        return 4;
+    } else if (size > 0x3fff) {
+        return 3;
+    } else if (size > 0x7f) {
+        return 2;
+    }
+    return 1;
+}
+
+static void storeSize(uint8_t *data, size_t &idx, size_t size) {
+    int numBytes = bytesForSize(size);
+    idx += numBytes;
+
+    data += idx;
+    size_t next = 0;
+    while (numBytes--) {
+        *--data = (size & 0x7f) | next;
+        size >>= 7;
+        next = 0x80;
+    }
+}
+
 static void addESDSFromCodecPrivate(
         const sp<MetaData> &meta,
         bool isAudio, const void *priv, size_t privSize) {
-    static const uint8_t kStaticESDS[] = {
-        0x03, 22,
-        0x00, 0x00,     // ES_ID
-        0x00,           // streamDependenceFlag, URL_Flag, OCRstreamFlag
 
-        0x04, 17,
-        0x40,           // ObjectTypeIndication
-        0x00, 0x00, 0x00, 0x00,
-        0x00, 0x00, 0x00, 0x00,
-        0x00, 0x00, 0x00, 0x00,
-
-        0x05,
-        // CodecSpecificInfo (with size prefix) follows
-    };
-
-    // Make sure all sizes can be coded in a single byte.
-    CHECK(privSize + 22 - 2 < 128);
-    size_t esdsSize = sizeof(kStaticESDS) + privSize + 1;
+    int privSizeBytesRequired = bytesForSize(privSize);
+    int esdsSize2 = 14 + privSizeBytesRequired + privSize;
+    int esdsSize2BytesRequired = bytesForSize(esdsSize2);
+    int esdsSize1 = 4 + esdsSize2BytesRequired + esdsSize2;
+    int esdsSize1BytesRequired = bytesForSize(esdsSize1);
+    size_t esdsSize = 1 + esdsSize1BytesRequired + esdsSize1;
     uint8_t *esds = new uint8_t[esdsSize];
-    memcpy(esds, kStaticESDS, sizeof(kStaticESDS));
-    uint8_t *ptr = esds + sizeof(kStaticESDS);
-    *ptr++ = privSize;
-    memcpy(ptr, priv, privSize);
 
-    // Increment by codecPrivateSize less 2 bytes that are accounted for
-    // already in lengths of 22/17
-    esds[1] += privSize - 2;
-    esds[6] += privSize - 2;
-
-    // Set ObjectTypeIndication.
-    esds[7] = isAudio ? 0x40   // Audio ISO/IEC 14496-3
-                      : 0x20;  // Visual ISO/IEC 14496-2
+    size_t idx = 0;
+    esds[idx++] = 0x03;
+    storeSize(esds, idx, esdsSize1);
+    esds[idx++] = 0x00; // ES_ID
+    esds[idx++] = 0x00; // ES_ID
+    esds[idx++] = 0x00; // streamDependenceFlag, URL_Flag, OCRstreamFlag
+    esds[idx++] = 0x04;
+    storeSize(esds, idx, esdsSize2);
+    esds[idx++] = isAudio ? 0x40   // Audio ISO/IEC 14496-3
+                          : 0x20;  // Visual ISO/IEC 14496-2
+    for (int i = 0; i < 12; i++) {
+        esds[idx++] = 0x00;
+    }
+    esds[idx++] = 0x05;
+    storeSize(esds, idx, privSize);
+    memcpy(esds + idx, priv, privSize);
 
     meta->setData(kKeyESDS, 0, esds, esdsSize);
 
diff --git a/media/libstagefright/mp4/FragmentedMP4Parser.cpp b/media/libstagefright/mp4/FragmentedMP4Parser.cpp
deleted file mode 100644
index 0102656..0000000
--- a/media/libstagefright/mp4/FragmentedMP4Parser.cpp
+++ /dev/null
@@ -1,1993 +0,0 @@
-/*
- * Copyright (C) 2012 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "FragmentedMP4Parser"
-#include <utils/Log.h>
-
-#include "include/avc_utils.h"
-#include "include/ESDS.h"
-#include "include/FragmentedMP4Parser.h"
-#include "TrackFragment.h"
-
-
-#include <media/stagefright/foundation/ABuffer.h>
-#include <media/stagefright/foundation/ADebug.h>
-#include <media/stagefright/foundation/AMessage.h>
-#include <media/stagefright/foundation/hexdump.h>
-#include <media/stagefright/MediaDefs.h>
-#include <media/stagefright/MediaErrors.h>
-#include <media/stagefright/Utils.h>
-
-
-namespace android {
-
-static const char *Fourcc2String(uint32_t fourcc) {
-    static char buffer[5];
-    buffer[4] = '\0';
-    buffer[0] = fourcc >> 24;
-    buffer[1] = (fourcc >> 16) & 0xff;
-    buffer[2] = (fourcc >> 8) & 0xff;
-    buffer[3] = fourcc & 0xff;
-
-    return buffer;
-}
-
-static const char *IndentString(size_t n) {
-    static const char kSpace[] = "                              ";
-    return kSpace + sizeof(kSpace) - 2 * n - 1;
-}
-
-// static
-const FragmentedMP4Parser::DispatchEntry FragmentedMP4Parser::kDispatchTable[] = {
-    { FOURCC('m', 'o', 'o', 'v'), 0, NULL },
-    { FOURCC('t', 'r', 'a', 'k'), FOURCC('m', 'o', 'o', 'v'), NULL },
-    { FOURCC('u', 'd', 't', 'a'), FOURCC('t', 'r', 'a', 'k'), NULL },
-    { FOURCC('u', 'd', 't', 'a'), FOURCC('m', 'o', 'o', 'v'), NULL },
-    { FOURCC('m', 'e', 't', 'a'), FOURCC('u', 'd', 't', 'a'), NULL },
-    { FOURCC('i', 'l', 's', 't'), FOURCC('m', 'e', 't', 'a'), NULL },
-
-    { FOURCC('t', 'k', 'h', 'd'), FOURCC('t', 'r', 'a', 'k'),
-        &FragmentedMP4Parser::parseTrackHeader
-    },
-
-    { FOURCC('m', 'v', 'e', 'x'), FOURCC('m', 'o', 'o', 'v'), NULL },
-
-    { FOURCC('t', 'r', 'e', 'x'), FOURCC('m', 'v', 'e', 'x'),
-        &FragmentedMP4Parser::parseTrackExtends
-    },
-
-    { FOURCC('e', 'd', 't', 's'), FOURCC('t', 'r', 'a', 'k'), NULL },
-    { FOURCC('m', 'd', 'i', 'a'), FOURCC('t', 'r', 'a', 'k'), NULL },
-
-    { FOURCC('m', 'd', 'h', 'd'), FOURCC('m', 'd', 'i', 'a'),
-        &FragmentedMP4Parser::parseMediaHeader
-    },
-
-    { FOURCC('h', 'd', 'l', 'r'), FOURCC('m', 'd', 'i', 'a'),
-        &FragmentedMP4Parser::parseMediaHandler
-    },
-
-    { FOURCC('m', 'i', 'n', 'f'), FOURCC('m', 'd', 'i', 'a'), NULL },
-    { FOURCC('d', 'i', 'n', 'f'), FOURCC('m', 'i', 'n', 'f'), NULL },
-    { FOURCC('s', 't', 'b', 'l'), FOURCC('m', 'i', 'n', 'f'), NULL },
-    { FOURCC('s', 't', 's', 'd'), FOURCC('s', 't', 'b', 'l'), NULL },
-
-    { FOURCC('s', 't', 's', 'z'), FOURCC('s', 't', 'b', 'l'),
-        &FragmentedMP4Parser::parseSampleSizes },
-
-    { FOURCC('s', 't', 'z', '2'), FOURCC('s', 't', 'b', 'l'),
-        &FragmentedMP4Parser::parseCompactSampleSizes },
-
-    { FOURCC('s', 't', 's', 'c'), FOURCC('s', 't', 'b', 'l'),
-        &FragmentedMP4Parser::parseSampleToChunk },
-
-    { FOURCC('s', 't', 'c', 'o'), FOURCC('s', 't', 'b', 'l'),
-        &FragmentedMP4Parser::parseChunkOffsets },
-
-    { FOURCC('c', 'o', '6', '4'), FOURCC('s', 't', 'b', 'l'),
-        &FragmentedMP4Parser::parseChunkOffsets64 },
-
-    { FOURCC('a', 'v', 'c', 'C'), FOURCC('a', 'v', 'c', '1'),
-        &FragmentedMP4Parser::parseAVCCodecSpecificData },
-
-    { FOURCC('e', 's', 'd', 's'), FOURCC('m', 'p', '4', 'a'),
-        &FragmentedMP4Parser::parseESDSCodecSpecificData },
-
-    { FOURCC('e', 's', 'd', 's'), FOURCC('m', 'p', '4', 'v'),
-        &FragmentedMP4Parser::parseESDSCodecSpecificData },
-
-    { FOURCC('m', 'd', 'a', 't'), 0, &FragmentedMP4Parser::parseMediaData },
-
-    { FOURCC('m', 'o', 'o', 'f'), 0, NULL },
-    { FOURCC('t', 'r', 'a', 'f'), FOURCC('m', 'o', 'o', 'f'), NULL },
-
-    { FOURCC('t', 'f', 'h', 'd'), FOURCC('t', 'r', 'a', 'f'),
-        &FragmentedMP4Parser::parseTrackFragmentHeader
-    },
-    { FOURCC('t', 'r', 'u', 'n'), FOURCC('t', 'r', 'a', 'f'),
-        &FragmentedMP4Parser::parseTrackFragmentRun
-    },
-
-    { FOURCC('m', 'f', 'r', 'a'), 0, NULL },
-
-    { FOURCC('s', 'i', 'd', 'x'), 0, &FragmentedMP4Parser::parseSegmentIndex },
-};
-
-struct FileSource : public FragmentedMP4Parser::Source {
-    FileSource(const char *filename)
-        : mFile(fopen(filename, "rb")) {
-            CHECK(mFile != NULL);
-        }
-
-    virtual ~FileSource() {
-        fclose(mFile);
-    }
-
-    virtual ssize_t readAt(off64_t offset, void *data, size_t size) {
-        fseek(mFile, offset, SEEK_SET);
-        return fread(data, 1, size, mFile);
-    }
-
-    virtual bool isSeekable() {
-        return true;
-    }
-
-    private:
-    FILE *mFile;
-
-    DISALLOW_EVIL_CONSTRUCTORS(FileSource);
-};
-
-struct ReadTracker : public RefBase {
-    ReadTracker(off64_t size) {
-        allocSize = 1 + size / 8192; // 1 bit per kilobyte
-        bitmap = (char*) calloc(1, allocSize);
-    }
-    virtual ~ReadTracker() {
-        dumpToLog();
-        free(bitmap);
-    }
-    void mark(off64_t offset, size_t size) {
-        int firstbit = offset / 1024;
-        int lastbit = (offset + size - 1) / 1024;
-        for (int i = firstbit; i <= lastbit; i++) {
-            bitmap[i/8] |= (0x80 >> (i & 7));
-        }
-    }
-
- private:
-    void dumpToLog() {
-        // 96 chars per line, each char represents one kilobyte, 1 kb per bit
-        int numlines = allocSize / 12;
-        char buf[97];
-        char *cur = bitmap;
-        for (int i = 0; i < numlines; i++ && cur) {
-            for (int j = 0; j < 12; j++) {
-                for (int k = 0; k < 8; k++) {
-                    buf[(j * 8) + k] = (*cur & (0x80 >> k)) ? 'X' : '.';
-                }
-                cur++;
-            }
-            buf[96] = '\0';
-            ALOGI("%5dk: %s", i * 96, buf);
-        }
-    }
-
-    size_t allocSize;
-    char *bitmap;
-};
-
-struct DataSourceSource : public FragmentedMP4Parser::Source {
-    DataSourceSource(sp<DataSource> &source)
-        : mDataSource(source) {
-            CHECK(mDataSource != NULL);
-#if 0
-            off64_t size;
-            if (source->getSize(&size) == OK) {
-                mReadTracker = new ReadTracker(size);
-            } else {
-                ALOGE("couldn't get data source size");
-            }
-#endif
-        }
-
-    virtual ssize_t readAt(off64_t offset, void *data, size_t size) {
-        if (mReadTracker != NULL) {
-            mReadTracker->mark(offset, size);
-        }
-        return mDataSource->readAt(offset, data, size);
-    }
-
-    virtual bool isSeekable() {
-        return true;
-    }
-
-    private:
-    sp<DataSource> mDataSource;
-    sp<ReadTracker> mReadTracker;
-
-    DISALLOW_EVIL_CONSTRUCTORS(DataSourceSource);
-};
-
-FragmentedMP4Parser::FragmentedMP4Parser()
-    : mBufferPos(0),
-      mSuspended(false),
-      mDoneWithMoov(false),
-      mFirstMoofOffset(0),
-      mFinalResult(OK) {
-}
-
-FragmentedMP4Parser::~FragmentedMP4Parser() {
-}
-
-void FragmentedMP4Parser::start(const char *filename) {
-    sp<AMessage> msg = new AMessage(kWhatStart, id());
-    msg->setObject("source", new FileSource(filename));
-    msg->post();
-    ALOGV("Parser::start(%s)", filename);
-}
-
-void FragmentedMP4Parser::start(const sp<Source> &source) {
-    sp<AMessage> msg = new AMessage(kWhatStart, id());
-    msg->setObject("source", source);
-    msg->post();
-    ALOGV("Parser::start(Source)");
-}
-
-void FragmentedMP4Parser::start(sp<DataSource> &source) {
-    sp<AMessage> msg = new AMessage(kWhatStart, id());
-    msg->setObject("source", new DataSourceSource(source));
-    msg->post();
-    ALOGV("Parser::start(DataSource)");
-}
-
-sp<AMessage> FragmentedMP4Parser::getFormat(bool audio, bool synchronous) {
-
-    while (true) {
-        bool moovDone = mDoneWithMoov;
-        sp<AMessage> msg = new AMessage(kWhatGetFormat, id());
-        msg->setInt32("audio", audio);
-
-        sp<AMessage> response;
-        status_t err = msg->postAndAwaitResponse(&response);
-
-        if (err != OK) {
-            ALOGV("getFormat post failed: %d", err);
-            return NULL;
-        }
-
-        if (response->findInt32("err", &err) && err != OK) {
-            if (synchronous && err == -EWOULDBLOCK && !moovDone) {
-                resumeIfNecessary();
-                ALOGV("@getFormat parser not ready yet, retrying");
-                usleep(10000);
-                continue;
-            }
-            ALOGV("getFormat failed: %d", err);
-            return NULL;
-        }
-
-        sp<AMessage> format;
-        CHECK(response->findMessage("format", &format));
-
-        ALOGV("returning format %s", format->debugString().c_str());
-        return format;
-    }
-}
-
-status_t FragmentedMP4Parser::seekTo(bool wantAudio, int64_t timeUs) {
-    sp<AMessage> msg = new AMessage(kWhatSeekTo, id());
-    msg->setInt32("audio", wantAudio);
-    msg->setInt64("position", timeUs);
-
-    sp<AMessage> response;
-    status_t err = msg->postAndAwaitResponse(&response);
-    return err;
-}
-
-bool FragmentedMP4Parser::isSeekable() const {
-    while (mFirstMoofOffset == 0 && mFinalResult == OK) {
-        usleep(10000);
-    }
-    bool seekable = mSource->isSeekable();
-    for (size_t i = 0; seekable && i < mTracks.size(); i++) {
-        const TrackInfo *info = &mTracks.valueAt(i);
-        seekable &= !info->mSidx.empty();
-    }
-    return seekable;
-}
-
-status_t FragmentedMP4Parser::onSeekTo(bool wantAudio, int64_t position) {
-    status_t err = -EINVAL;
-    ssize_t trackIndex = findTrack(wantAudio);
-    if (trackIndex < 0) {
-        err = trackIndex;
-    } else {
-        TrackInfo *info = &mTracks.editValueAt(trackIndex);
-
-        int numSidxEntries = info->mSidx.size();
-        int64_t totalTime = 0;
-        off_t totalOffset = mFirstMoofOffset;
-        for (int i = 0; i < numSidxEntries; i++) {
-            const SidxEntry *se = &info->mSidx[i];
-            if (totalTime + se->mDurationUs > position) {
-                mBuffer->setRange(0,0);
-                mBufferPos = totalOffset;
-                if (mFinalResult == ERROR_END_OF_STREAM) {
-                    mFinalResult = OK;
-                    mSuspended = true; // force resume
-                    resumeIfNecessary();
-                }
-                info->mFragments.clear();
-                info->mDecodingTime = totalTime * info->mMediaTimeScale / 1000000ll;
-                return OK;
-            }
-            totalTime += se->mDurationUs;
-            totalOffset += se->mSize;
-        }
-    }
-    ALOGV("seekTo out of range");
-    return err;
-}
-
-status_t FragmentedMP4Parser::dequeueAccessUnit(bool audio, sp<ABuffer> *accessUnit,
-                                                bool synchronous) {
-
-    while (true) {
-        sp<AMessage> msg = new AMessage(kWhatDequeueAccessUnit, id());
-        msg->setInt32("audio", audio);
-
-        sp<AMessage> response;
-        status_t err = msg->postAndAwaitResponse(&response);
-
-        if (err != OK) {
-            ALOGV("dequeue fail 1: %d", err);
-            return err;
-        }
-
-        if (response->findInt32("err", &err) && err != OK) {
-            if (synchronous && err == -EWOULDBLOCK) {
-                resumeIfNecessary();
-                ALOGV("Parser not ready yet, retrying");
-                usleep(10000);
-                continue;
-            }
-            ALOGV("dequeue fail 2: %d, %d", err, synchronous);
-            return err;
-        }
-
-        CHECK(response->findBuffer("accessUnit", accessUnit));
-
-        return OK;
-    }
-}
-
-ssize_t FragmentedMP4Parser::findTrack(bool wantAudio) const {
-    for (size_t i = 0; i < mTracks.size(); ++i) {
-        const TrackInfo *info = &mTracks.valueAt(i);
-
-        bool isAudio =
-            info->mMediaHandlerType == FOURCC('s', 'o', 'u', 'n');
-
-        bool isVideo =
-            info->mMediaHandlerType == FOURCC('v', 'i', 'd', 'e');
-
-        if ((wantAudio && isAudio) || (!wantAudio && !isAudio)) {
-            if (info->mSampleDescs.empty()) {
-                break;
-            }
-
-            return i;
-        }
-    }
-
-    return -EWOULDBLOCK;
-}
-
-void FragmentedMP4Parser::onMessageReceived(const sp<AMessage> &msg) {
-    switch (msg->what()) {
-        case kWhatStart:
-        {
-            sp<RefBase> obj;
-            CHECK(msg->findObject("source", &obj));
-
-            mSource = static_cast<Source *>(obj.get());
-
-            mBuffer = new ABuffer(512 * 1024);
-            mBuffer->setRange(0, 0);
-
-            enter(0ll, 0, 0);
-
-            (new AMessage(kWhatProceed, id()))->post();
-            break;
-        }
-
-        case kWhatProceed:
-        {
-            CHECK(!mSuspended);
-
-            status_t err = onProceed();
-
-            if (err == OK) {
-                if (!mSuspended) {
-                    msg->post();
-                }
-            } else if (err != -EAGAIN) {
-                ALOGE("onProceed returned error %d", err);
-            }
-
-            break;
-        }
-
-        case kWhatReadMore:
-        {
-            size_t needed;
-            CHECK(msg->findSize("needed", &needed));
-
-            memmove(mBuffer->base(), mBuffer->data(), mBuffer->size());
-            mBufferPos += mBuffer->offset();
-            mBuffer->setRange(0, mBuffer->size());
-
-            size_t maxBytesToRead = mBuffer->capacity() - mBuffer->size();
-
-            if (maxBytesToRead < needed) {
-                ALOGV("resizing buffer.");
-
-                sp<ABuffer> newBuffer =
-                    new ABuffer((mBuffer->size() + needed + 1023) & ~1023);
-                memcpy(newBuffer->data(), mBuffer->data(), mBuffer->size());
-                newBuffer->setRange(0, mBuffer->size());
-
-                mBuffer = newBuffer;
-                maxBytesToRead = mBuffer->capacity() - mBuffer->size();
-            }
-
-            CHECK_GE(maxBytesToRead, needed);
-
-            ssize_t n = mSource->readAt(
-                    mBufferPos + mBuffer->size(),
-                    mBuffer->data() + mBuffer->size(), needed);
-
-            if (n < (ssize_t)needed) {
-                ALOGV("Reached EOF when reading %d @ %d + %d", needed, mBufferPos, mBuffer->size());
-                if (n < 0) {
-                    mFinalResult = n;
-                } else if (n == 0) {
-                    mFinalResult = ERROR_END_OF_STREAM;
-                } else {
-                    mFinalResult = ERROR_IO;
-                }
-            } else {
-                mBuffer->setRange(0, mBuffer->size() + n);
-                (new AMessage(kWhatProceed, id()))->post();
-            }
-
-            break;
-        }
-
-        case kWhatGetFormat:
-        {
-            int32_t wantAudio;
-            CHECK(msg->findInt32("audio", &wantAudio));
-
-            status_t err = -EWOULDBLOCK;
-            sp<AMessage> response = new AMessage;
-
-            ssize_t trackIndex = findTrack(wantAudio);
-
-            if (trackIndex < 0) {
-                err = trackIndex;
-            } else {
-                TrackInfo *info = &mTracks.editValueAt(trackIndex);
-
-                sp<AMessage> format = info->mSampleDescs.itemAt(0).mFormat;
-                if (info->mSidxDuration) {
-                    format->setInt64("durationUs", info->mSidxDuration);
-                } else {
-                    // this is probably going to be zero. Oh well...
-                    format->setInt64("durationUs",
-                                     1000000ll * info->mDuration / info->mMediaTimeScale);
-                }
-                response->setMessage(
-                        "format", format);
-
-                err = OK;
-            }
-
-            response->setInt32("err", err);
-
-            uint32_t replyID;
-            CHECK(msg->senderAwaitsResponse(&replyID));
-
-            response->postReply(replyID);
-            break;
-        }
-
-        case kWhatDequeueAccessUnit:
-        {
-            int32_t wantAudio;
-            CHECK(msg->findInt32("audio", &wantAudio));
-
-            status_t err = -EWOULDBLOCK;
-            sp<AMessage> response = new AMessage;
-
-            ssize_t trackIndex = findTrack(wantAudio);
-
-            if (trackIndex < 0) {
-                err = trackIndex;
-            } else {
-                sp<ABuffer> accessUnit;
-                err = onDequeueAccessUnit(trackIndex, &accessUnit);
-
-                if (err == OK) {
-                    response->setBuffer("accessUnit", accessUnit);
-                }
-            }
-
-            response->setInt32("err", err);
-
-            uint32_t replyID;
-            CHECK(msg->senderAwaitsResponse(&replyID));
-
-            response->postReply(replyID);
-            break;
-        }
-
-        case kWhatSeekTo:
-        {
-            ALOGV("kWhatSeekTo");
-            int32_t wantAudio;
-            CHECK(msg->findInt32("audio", &wantAudio));
-            int64_t position;
-            CHECK(msg->findInt64("position", &position));
-
-            status_t err = -EWOULDBLOCK;
-            sp<AMessage> response = new AMessage;
-
-            ssize_t trackIndex = findTrack(wantAudio);
-
-            if (trackIndex < 0) {
-                err = trackIndex;
-            } else {
-                err = onSeekTo(wantAudio, position);
-            }
-            response->setInt32("err", err);
-            uint32_t replyID;
-            CHECK(msg->senderAwaitsResponse(&replyID));
-            response->postReply(replyID);
-            break;
-        }
-        default:
-            TRESPASS();
-    }
-}
-
-status_t FragmentedMP4Parser::onProceed() {
-    status_t err;
-
-    if ((err = need(8)) != OK) {
-        return err;
-    }
-
-    uint64_t size = readU32(0);
-    uint32_t type = readU32(4);
-
-    size_t offset = 8;
-
-    if (size == 1) {
-        if ((err = need(16)) != OK) {
-            return err;
-        }
-
-        size = readU64(offset);
-        offset += 8;
-    }
-
-    uint8_t userType[16];
-
-    if (type == FOURCC('u', 'u', 'i', 'd')) {
-        if ((err = need(offset + 16)) != OK) {
-            return err;
-        }
-
-        memcpy(userType, mBuffer->data() + offset, 16);
-        offset += 16;
-    }
-
-    CHECK(!mStack.isEmpty());
-    uint32_t ptype = mStack.itemAt(mStack.size() - 1).mType;
-
-    static const size_t kNumDispatchers =
-        sizeof(kDispatchTable) / sizeof(kDispatchTable[0]);
-
-    size_t i;
-    for (i = 0; i < kNumDispatchers; ++i) {
-        if (kDispatchTable[i].mType == type
-                && kDispatchTable[i].mParentType == ptype) {
-            break;
-        }
-    }
-
-    // SampleEntry boxes are container boxes that start with a variable
-    // amount of data depending on the media handler type.
-    // We don't look inside 'hint' type SampleEntry boxes.
-
-    bool isSampleEntryBox =
-        (ptype == FOURCC('s', 't', 's', 'd'))
-        && editTrack(mCurrentTrackID)->mMediaHandlerType
-        != FOURCC('h', 'i', 'n', 't');
-
-    if ((i < kNumDispatchers && kDispatchTable[i].mHandler == 0)
-            || isSampleEntryBox || ptype == FOURCC('i', 'l', 's', 't')) {
-        // This is a container box.
-        if (type == FOURCC('m', 'o', 'o', 'f')) {
-            if (mFirstMoofOffset == 0) {
-                ALOGV("first moof @ %08x", mBufferPos + offset);
-                mFirstMoofOffset = mBufferPos + offset - 8; // point at the size
-            }
-        }
-        if (type == FOURCC('m', 'e', 't', 'a')) {
-            if ((err = need(offset + 4)) < OK) {
-                return err;
-            }
-
-            if (readU32(offset) != 0) {
-                return -EINVAL;
-            }
-
-            offset += 4;
-        } else if (type == FOURCC('s', 't', 's', 'd')) {
-            if ((err = need(offset + 8)) < OK) {
-                return err;
-            }
-
-            if (readU32(offset) != 0) {
-                return -EINVAL;
-            }
-
-            if (readU32(offset + 4) == 0) {
-                // We need at least some entries.
-                return -EINVAL;
-            }
-
-            offset += 8;
-        } else if (isSampleEntryBox) {
-            size_t headerSize;
-
-            switch (editTrack(mCurrentTrackID)->mMediaHandlerType) {
-                case FOURCC('v', 'i', 'd', 'e'):
-                {
-                    // 8 bytes SampleEntry + 70 bytes VisualSampleEntry
-                    headerSize = 78;
-                    break;
-                }
-
-                case FOURCC('s', 'o', 'u', 'n'):
-                {
-                    // 8 bytes SampleEntry + 20 bytes AudioSampleEntry
-                    headerSize = 28;
-                    break;
-                }
-
-                case FOURCC('m', 'e', 't', 'a'):
-                {
-                    headerSize = 8;  // 8 bytes SampleEntry
-                    break;
-                }
-
-                default:
-                    TRESPASS();
-            }
-
-            if (offset + headerSize > size) {
-                return -EINVAL;
-            }
-
-            if ((err = need(offset + headerSize)) != OK) {
-                return err;
-            }
-
-            switch (editTrack(mCurrentTrackID)->mMediaHandlerType) {
-                case FOURCC('v', 'i', 'd', 'e'):
-                {
-                    err = parseVisualSampleEntry(
-                            type, offset, offset + headerSize);
-                    break;
-                }
-
-                case FOURCC('s', 'o', 'u', 'n'):
-                {
-                    err = parseAudioSampleEntry(
-                            type, offset, offset + headerSize);
-                    break;
-                }
-
-                case FOURCC('m', 'e', 't', 'a'):
-                {
-                    err = OK;
-                    break;
-                }
-
-                default:
-                    TRESPASS();
-            }
-
-            if (err != OK) {
-                return err;
-            }
-
-            offset += headerSize;
-        }
-
-        skip(offset);
-
-        ALOGV("%sentering box of type '%s'",
-                IndentString(mStack.size()), Fourcc2String(type));
-
-        enter(mBufferPos - offset, type, size - offset);
-    } else {
-        if (!fitsContainer(size)) {
-            return -EINVAL;
-        }
-
-        if (i < kNumDispatchers && kDispatchTable[i].mHandler != 0) {
-            // We have a handler for this box type.
-
-            if ((err = need(size)) != OK) {
-                return err;
-            }
-
-            ALOGV("%sparsing box of type '%s'",
-                    IndentString(mStack.size()), Fourcc2String(type));
-
-            if ((err = (this->*kDispatchTable[i].mHandler)(
-                            type, offset, size)) != OK) {
-                return err;
-            }
-        } else {
-            // Unknown box type
-
-            ALOGV("%sskipping box of type '%s', size %llu",
-                    IndentString(mStack.size()),
-                    Fourcc2String(type), size);
-
-        }
-
-        skip(size);
-    }
-
-    return OK;
-}
-
-// static
-int FragmentedMP4Parser::CompareSampleLocation(
-        const SampleInfo &sample, const MediaDataInfo &mdatInfo) {
-    if (sample.mOffset + sample.mSize < mdatInfo.mOffset) {
-        return -1;
-    }
-
-    if (sample.mOffset >= mdatInfo.mOffset + mdatInfo.mBuffer->size()) {
-        return 1;
-    }
-
-    // Otherwise make sure the sample is completely contained within this
-    // media data block.
-
-    CHECK_GE(sample.mOffset, mdatInfo.mOffset);
-
-    CHECK_LE(sample.mOffset + sample.mSize,
-             mdatInfo.mOffset + mdatInfo.mBuffer->size());
-
-    return 0;
-}
-
-void FragmentedMP4Parser::resumeIfNecessary() {
-    if (!mSuspended) {
-        return;
-    }
-
-    ALOGV("resuming.");
-
-    mSuspended = false;
-    (new AMessage(kWhatProceed, id()))->post();
-}
-
-status_t FragmentedMP4Parser::getSample(
-        TrackInfo *info, sp<TrackFragment> *fragment, SampleInfo *sampleInfo) {
-    for (;;) {
-        if (info->mFragments.empty()) {
-            if (mFinalResult != OK) {
-                return mFinalResult;
-            }
-
-            resumeIfNecessary();
-            return -EWOULDBLOCK;
-        }
-
-        *fragment = *info->mFragments.begin();
-
-        status_t err = (*fragment)->getSample(sampleInfo);
-
-        if (err == OK) {
-            return OK;
-        } else if (err != ERROR_END_OF_STREAM) {
-            return err;
-        }
-
-        // Really, end of this fragment...
-
-        info->mFragments.erase(info->mFragments.begin());
-    }
-}
-
-status_t FragmentedMP4Parser::onDequeueAccessUnit(
-        size_t trackIndex, sp<ABuffer> *accessUnit) {
-    TrackInfo *info = &mTracks.editValueAt(trackIndex);
-
-    sp<TrackFragment> fragment;
-    SampleInfo sampleInfo;
-    status_t err = getSample(info, &fragment, &sampleInfo);
-
-    if (err == -EWOULDBLOCK) {
-        resumeIfNecessary();
-        return err;
-    } else if (err != OK) {
-        return err;
-    }
-
-    err = -EWOULDBLOCK;
-
-    bool checkDroppable = false;
-
-    for (size_t i = 0; i < mMediaData.size(); ++i) {
-        const MediaDataInfo &mdatInfo = mMediaData.itemAt(i);
-
-        int cmp = CompareSampleLocation(sampleInfo, mdatInfo);
-
-        if (cmp < 0 && !mSource->isSeekable()) {
-            return -EPIPE;
-        } else if (cmp == 0) {
-            if (i > 0) {
-                checkDroppable = true;
-            }
-
-            err = makeAccessUnit(info, sampleInfo, mdatInfo, accessUnit);
-            break;
-        }
-    }
-
-    if (err != OK) {
-        return err;
-    }
-
-    fragment->advance();
-
-    if (!mMediaData.empty() && checkDroppable) {
-        size_t numDroppable = 0;
-        bool done = false;
-
-        // XXX FIXME: if one of the tracks is not advanced (e.g. if you play an audio+video
-        // file with sf2), then mMediaData will not be pruned and keeps growing
-        for (size_t i = 0; !done && i < mMediaData.size(); ++i) {
-            const MediaDataInfo &mdatInfo = mMediaData.itemAt(i);
-
-            for (size_t j = 0; j < mTracks.size(); ++j) {
-                TrackInfo *info = &mTracks.editValueAt(j);
-
-                sp<TrackFragment> fragment;
-                SampleInfo sampleInfo;
-                err = getSample(info, &fragment, &sampleInfo);
-
-                if (err != OK) {
-                    done = true;
-                    break;
-                }
-
-                int cmp = CompareSampleLocation(sampleInfo, mdatInfo);
-
-                if (cmp <= 0) {
-                    done = true;
-                    break;
-                }
-            }
-
-            if (!done) {
-                ++numDroppable;
-            }
-        }
-
-        if (numDroppable > 0) {
-            mMediaData.removeItemsAt(0, numDroppable);
-
-            if (mMediaData.size() < 5) {
-                resumeIfNecessary();
-            }
-        }
-    }
-
-    return err;
-}
-
-static size_t parseNALSize(size_t nalLengthSize, const uint8_t *data) {
-    switch (nalLengthSize) {
-        case 1:
-            return *data;
-        case 2:
-            return U16_AT(data);
-        case 3:
-            return ((size_t)data[0] << 16) | U16_AT(&data[1]);
-        case 4:
-            return U32_AT(data);
-    }
-
-    // This cannot happen, mNALLengthSize springs to life by adding 1 to
-    // a 2-bit integer.
-    TRESPASS();
-
-    return 0;
-}
-
-status_t FragmentedMP4Parser::makeAccessUnit(
-        TrackInfo *info,
-        const SampleInfo &sample,
-        const MediaDataInfo &mdatInfo,
-        sp<ABuffer> *accessUnit) {
-    if (sample.mSampleDescIndex < 1
-            || sample.mSampleDescIndex > info->mSampleDescs.size()) {
-        return ERROR_MALFORMED;
-    }
-
-    int64_t presentationTimeUs =
-        1000000ll * sample.mPresentationTime / info->mMediaTimeScale;
-
-    const SampleDescription &sampleDesc =
-        info->mSampleDescs.itemAt(sample.mSampleDescIndex - 1);
-
-    size_t nalLengthSize;
-    if (!sampleDesc.mFormat->findSize("nal-length-size", &nalLengthSize)) {
-        *accessUnit = new ABuffer(sample.mSize);
-
-        memcpy((*accessUnit)->data(),
-               mdatInfo.mBuffer->data() + (sample.mOffset - mdatInfo.mOffset),
-               sample.mSize);
-
-        (*accessUnit)->meta()->setInt64("timeUs", presentationTimeUs);
-        if (IsIDR(*accessUnit)) {
-            (*accessUnit)->meta()->setInt32("is-sync-frame", 1);
-        }
-
-        return OK;
-    }
-
-    const uint8_t *srcPtr =
-        mdatInfo.mBuffer->data() + (sample.mOffset - mdatInfo.mOffset);
-
-    for (int i = 0; i < 2 ; ++i) {
-        size_t srcOffset = 0;
-        size_t dstOffset = 0;
-
-        while (srcOffset < sample.mSize) {
-            if (srcOffset + nalLengthSize > sample.mSize) {
-                return ERROR_MALFORMED;
-            }
-
-            size_t nalSize = parseNALSize(nalLengthSize, &srcPtr[srcOffset]);
-            srcOffset += nalLengthSize;
-
-            if (srcOffset + nalSize > sample.mSize) {
-                return ERROR_MALFORMED;
-            }
-
-            if (i == 1) {
-                memcpy((*accessUnit)->data() + dstOffset,
-                       "\x00\x00\x00\x01",
-                       4);
-
-                memcpy((*accessUnit)->data() + dstOffset + 4,
-                       srcPtr + srcOffset,
-                       nalSize);
-            }
-
-            srcOffset += nalSize;
-            dstOffset += nalSize + 4;
-        }
-
-        if (i == 0) {
-            (*accessUnit) = new ABuffer(dstOffset);
-            (*accessUnit)->meta()->setInt64(
-                    "timeUs", presentationTimeUs);
-        }
-    }
-    if (IsIDR(*accessUnit)) {
-        (*accessUnit)->meta()->setInt32("is-sync-frame", 1);
-    }
-
-    return OK;
-}
-
-status_t FragmentedMP4Parser::need(size_t size) {
-    if (!fitsContainer(size)) {
-        return -EINVAL;
-    }
-
-    if (size <= mBuffer->size()) {
-        return OK;
-    }
-
-    sp<AMessage> msg = new AMessage(kWhatReadMore, id());
-    msg->setSize("needed", size - mBuffer->size());
-    msg->post();
-
-    // ALOGV("need(%d) returning -EAGAIN, only have %d", size, mBuffer->size());
-
-    return -EAGAIN;
-}
-
-void FragmentedMP4Parser::enter(off64_t offset, uint32_t type, uint64_t size) {
-    Container container;
-    container.mOffset = offset;
-    container.mType = type;
-    container.mExtendsToEOF = (size == 0);
-    container.mBytesRemaining = size;
-
-    mStack.push(container);
-}
-
-bool FragmentedMP4Parser::fitsContainer(uint64_t size) const {
-    CHECK(!mStack.isEmpty());
-    const Container &container = mStack.itemAt(mStack.size() - 1);
-
-    return container.mExtendsToEOF || size <= container.mBytesRemaining;
-}
-
-uint16_t FragmentedMP4Parser::readU16(size_t offset) {
-    CHECK_LE(offset + 2, mBuffer->size());
-
-    const uint8_t *ptr = mBuffer->data() + offset;
-    return (ptr[0] << 8) | ptr[1];
-}
-
-uint32_t FragmentedMP4Parser::readU32(size_t offset) {
-    CHECK_LE(offset + 4, mBuffer->size());
-
-    const uint8_t *ptr = mBuffer->data() + offset;
-    return (ptr[0] << 24) | (ptr[1] << 16) | (ptr[2] << 8) | ptr[3];
-}
-
-uint64_t FragmentedMP4Parser::readU64(size_t offset) {
-    return (((uint64_t)readU32(offset)) << 32) | readU32(offset + 4);
-}
-
-void FragmentedMP4Parser::skip(off_t distance) {
-    CHECK(!mStack.isEmpty());
-    for (size_t i = mStack.size(); i-- > 0;) {
-        Container *container = &mStack.editItemAt(i);
-        if (!container->mExtendsToEOF) {
-            CHECK_LE(distance, (off_t)container->mBytesRemaining);
-
-            container->mBytesRemaining -= distance;
-
-            if (container->mBytesRemaining == 0) {
-                ALOGV("%sleaving box of type '%s'",
-                        IndentString(mStack.size() - 1),
-                        Fourcc2String(container->mType));
-
-#if 0
-                if (container->mType == FOURCC('s', 't', 's', 'd')) {
-                    TrackInfo *trackInfo = editTrack(mCurrentTrackID);
-                    for (size_t i = 0;
-                            i < trackInfo->mSampleDescs.size(); ++i) {
-                        ALOGI("format #%d: %s",
-                              i,
-                              trackInfo->mSampleDescs.itemAt(i)
-                                .mFormat->debugString().c_str());
-                    }
-                }
-#endif
-
-                if (container->mType == FOURCC('s', 't', 'b', 'l')) {
-                    TrackInfo *trackInfo = editTrack(mCurrentTrackID);
-
-                    trackInfo->mStaticFragment->signalCompletion();
-
-                    CHECK(trackInfo->mFragments.empty());
-                    trackInfo->mFragments.push_back(trackInfo->mStaticFragment);
-                    trackInfo->mStaticFragment.clear();
-                } else if (container->mType == FOURCC('t', 'r', 'a', 'f')) {
-                    TrackInfo *trackInfo =
-                        editTrack(mTrackFragmentHeaderInfo.mTrackID);
-
-                    const sp<TrackFragment> &fragment =
-                        *--trackInfo->mFragments.end();
-
-                    static_cast<DynamicTrackFragment *>(
-                            fragment.get())->signalCompletion();
-                } else if (container->mType == FOURCC('m', 'o', 'o', 'v')) {
-                    mDoneWithMoov = true;
-                }
-
-                container = NULL;
-                mStack.removeItemsAt(i);
-            }
-        }
-    }
-
-    if (distance < (off_t)mBuffer->size()) {
-        mBuffer->setRange(mBuffer->offset() + distance, mBuffer->size() - distance);
-        mBufferPos += distance;
-        return;
-    }
-
-    mBuffer->setRange(0, 0);
-    mBufferPos += distance;
-}
-
-status_t FragmentedMP4Parser::parseTrackHeader(
-        uint32_t type, size_t offset, uint64_t size) {
-    if (offset + 4 > size) {
-        return -EINVAL;
-    }
-
-    uint32_t flags = readU32(offset);
-
-    uint32_t version = flags >> 24;
-    flags &= 0xffffff;
-
-    uint32_t trackID;
-    uint64_t duration;
-
-    if (version == 1) {
-        if (offset + 36 > size) {
-            return -EINVAL;
-        }
-
-        trackID = readU32(offset + 20);
-        duration = readU64(offset + 28);
-
-        offset += 36;
-    } else if (version == 0) {
-        if (offset + 24 > size) {
-            return -EINVAL;
-        }
-
-        trackID = readU32(offset + 12);
-        duration = readU32(offset + 20);
-
-        offset += 24;
-    } else {
-        return -EINVAL;
-    }
-
-    TrackInfo *info = editTrack(trackID, true /* createIfNecessary */);
-    info->mFlags = flags;
-    info->mDuration = duration;
-    if (info->mDuration == 0xffffffff) {
-        // ffmpeg sets this to -1, which is incorrect.
-        info->mDuration = 0;
-    }
-
-    info->mStaticFragment = new StaticTrackFragment;
-
-    mCurrentTrackID = trackID;
-
-    return OK;
-}
-
-status_t FragmentedMP4Parser::parseMediaHeader(
-        uint32_t type, size_t offset, uint64_t size) {
-    if (offset + 4 > size) {
-        return -EINVAL;
-    }
-
-    uint32_t versionAndFlags = readU32(offset);
-
-    if (versionAndFlags & 0xffffff) {
-        return ERROR_MALFORMED;
-    }
-
-    uint32_t version = versionAndFlags >> 24;
-
-    TrackInfo *info = editTrack(mCurrentTrackID);
-
-    if (version == 1) {
-        if (offset + 4 + 32 > size) {
-            return -EINVAL;
-        }
-        info->mMediaTimeScale = U32_AT(mBuffer->data() + offset + 20);
-    } else if (version == 0) {
-        if (offset + 4 + 20 > size) {
-            return -EINVAL;
-        }
-        info->mMediaTimeScale = U32_AT(mBuffer->data() + offset + 12);
-    } else {
-        return ERROR_MALFORMED;
-    }
-
-    return OK;
-}
-
-status_t FragmentedMP4Parser::parseMediaHandler(
-        uint32_t type, size_t offset, uint64_t size) {
-    if (offset + 12 > size) {
-        return -EINVAL;
-    }
-
-    if (readU32(offset) != 0) {
-        return -EINVAL;
-    }
-
-    uint32_t handlerType = readU32(offset + 8);
-
-    switch (handlerType) {
-        case FOURCC('v', 'i', 'd', 'e'):
-        case FOURCC('s', 'o', 'u', 'n'):
-        case FOURCC('h', 'i', 'n', 't'):
-        case FOURCC('m', 'e', 't', 'a'):
-            break;
-
-        default:
-            return -EINVAL;
-    }
-
-    editTrack(mCurrentTrackID)->mMediaHandlerType = handlerType;
-
-    return OK;
-}
-
-status_t FragmentedMP4Parser::parseVisualSampleEntry(
-        uint32_t type, size_t offset, uint64_t size) {
-    if (offset + 78 > size) {
-        return -EINVAL;
-    }
-
-    TrackInfo *trackInfo = editTrack(mCurrentTrackID);
-
-    trackInfo->mSampleDescs.push();
-    SampleDescription *sampleDesc =
-        &trackInfo->mSampleDescs.editItemAt(
-                trackInfo->mSampleDescs.size() - 1);
-
-    sampleDesc->mType = type;
-    sampleDesc->mDataRefIndex = readU16(offset + 6);
-
-    sp<AMessage> format = new AMessage;
-
-    switch (type) {
-        case FOURCC('a', 'v', 'c', '1'):
-            format->setString("mime", MEDIA_MIMETYPE_VIDEO_AVC);
-            break;
-        case FOURCC('m', 'p', '4', 'v'):
-            format->setString("mime", MEDIA_MIMETYPE_VIDEO_MPEG4);
-            break;
-        case FOURCC('s', '2', '6', '3'):
-        case FOURCC('h', '2', '6', '3'):
-        case FOURCC('H', '2', '6', '3'):
-            format->setString("mime", MEDIA_MIMETYPE_VIDEO_H263);
-            break;
-        default:
-            format->setString("mime", "application/octet-stream");
-            break;
-    }
-
-    format->setInt32("width", readU16(offset + 8 + 16));
-    format->setInt32("height", readU16(offset + 8 + 18));
-
-    sampleDesc->mFormat = format;
-
-    return OK;
-}
-
-status_t FragmentedMP4Parser::parseAudioSampleEntry(
-        uint32_t type, size_t offset, uint64_t size) {
-    if (offset + 28 > size) {
-        return -EINVAL;
-    }
-
-    TrackInfo *trackInfo = editTrack(mCurrentTrackID);
-
-    trackInfo->mSampleDescs.push();
-    SampleDescription *sampleDesc =
-        &trackInfo->mSampleDescs.editItemAt(
-                trackInfo->mSampleDescs.size() - 1);
-
-    sampleDesc->mType = type;
-    sampleDesc->mDataRefIndex = readU16(offset + 6);
-
-    sp<AMessage> format = new AMessage;
-
-    format->setInt32("channel-count", readU16(offset + 8 + 8));
-    format->setInt32("sample-size", readU16(offset + 8 + 10));
-    format->setInt32("sample-rate", readU32(offset + 8 + 16) / 65536.0f);
-
-    switch (type) {
-        case FOURCC('m', 'p', '4', 'a'):
-            format->setString("mime", MEDIA_MIMETYPE_AUDIO_AAC);
-            break;
-
-        case FOURCC('s', 'a', 'm', 'r'):
-            format->setString("mime", MEDIA_MIMETYPE_AUDIO_AMR_NB);
-            format->setInt32("channel-count", 1);
-            format->setInt32("sample-rate", 8000);
-            break;
-
-        case FOURCC('s', 'a', 'w', 'b'):
-            format->setString("mime", MEDIA_MIMETYPE_AUDIO_AMR_WB);
-            format->setInt32("channel-count", 1);
-            format->setInt32("sample-rate", 16000);
-            break;
-        default:
-            format->setString("mime", "application/octet-stream");
-            break;
-    }
-
-    sampleDesc->mFormat = format;
-
-    return OK;
-}
-
-static void addCodecSpecificData(
-        const sp<AMessage> &format, int32_t index,
-        const void *data, size_t size,
-        bool insertStartCode = false) {
-    sp<ABuffer> csd = new ABuffer(insertStartCode ? size + 4 : size);
-
-    memcpy(csd->data() + (insertStartCode ? 4 : 0), data, size);
-
-    if (insertStartCode) {
-        memcpy(csd->data(), "\x00\x00\x00\x01", 4);
-    }
-
-    csd->meta()->setInt32("csd", true);
-    csd->meta()->setInt64("timeUs", 0ll);
-
-    format->setBuffer(StringPrintf("csd-%d", index).c_str(), csd);
-}
-
-status_t FragmentedMP4Parser::parseSampleSizes(
-        uint32_t type, size_t offset, uint64_t size) {
-    return editTrack(mCurrentTrackID)->mStaticFragment->parseSampleSizes(
-            this, type, offset, size);
-}
-
-status_t FragmentedMP4Parser::parseCompactSampleSizes(
-        uint32_t type, size_t offset, uint64_t size) {
-    return editTrack(mCurrentTrackID)->mStaticFragment->parseCompactSampleSizes(
-            this, type, offset, size);
-}
-
-status_t FragmentedMP4Parser::parseSampleToChunk(
-        uint32_t type, size_t offset, uint64_t size) {
-    return editTrack(mCurrentTrackID)->mStaticFragment->parseSampleToChunk(
-            this, type, offset, size);
-}
-
-status_t FragmentedMP4Parser::parseChunkOffsets(
-        uint32_t type, size_t offset, uint64_t size) {
-    return editTrack(mCurrentTrackID)->mStaticFragment->parseChunkOffsets(
-            this, type, offset, size);
-}
-
-status_t FragmentedMP4Parser::parseChunkOffsets64(
-        uint32_t type, size_t offset, uint64_t size) {
-    return editTrack(mCurrentTrackID)->mStaticFragment->parseChunkOffsets64(
-            this, type, offset, size);
-}
-
-status_t FragmentedMP4Parser::parseAVCCodecSpecificData(
-        uint32_t type, size_t offset, uint64_t size) {
-    TrackInfo *trackInfo = editTrack(mCurrentTrackID);
-
-    SampleDescription *sampleDesc =
-        &trackInfo->mSampleDescs.editItemAt(
-                trackInfo->mSampleDescs.size() - 1);
-
-    if (sampleDesc->mType != FOURCC('a', 'v', 'c', '1')) {
-        return -EINVAL;
-    }
-
-    const uint8_t *ptr = mBuffer->data() + offset;
-
-    size -= offset;
-    offset = 0;
-
-    if (size < 7 || ptr[0] != 0x01) {
-        return ERROR_MALFORMED;
-    }
-
-    sampleDesc->mFormat->setSize("nal-length-size", 1 + (ptr[4] & 3));
-
-    size_t numSPS = ptr[5] & 31;
-
-    ptr += 6;
-    size -= 6;
-
-    for (size_t i = 0; i < numSPS; ++i) {
-        if (size < 2) {
-            return ERROR_MALFORMED;
-        }
-
-        size_t length = U16_AT(ptr);
-
-        ptr += 2;
-        size -= 2;
-
-        if (size < length) {
-            return ERROR_MALFORMED;
-        }
-
-        addCodecSpecificData(
-                sampleDesc->mFormat, i, ptr, length,
-                true /* insertStartCode */);
-
-        ptr += length;
-        size -= length;
-    }
-
-    if (size < 1) {
-        return ERROR_MALFORMED;
-    }
-
-    size_t numPPS = *ptr;
-    ++ptr;
-    --size;
-
-    for (size_t i = 0; i < numPPS; ++i) {
-        if (size < 2) {
-            return ERROR_MALFORMED;
-        }
-
-        size_t length = U16_AT(ptr);
-
-        ptr += 2;
-        size -= 2;
-
-        if (size < length) {
-            return ERROR_MALFORMED;
-        }
-
-        addCodecSpecificData(
-                sampleDesc->mFormat, numSPS + i, ptr, length,
-                true /* insertStartCode */);
-
-        ptr += length;
-        size -= length;
-    }
-
-    return OK;
-}
-
-status_t FragmentedMP4Parser::parseESDSCodecSpecificData(
-        uint32_t type, size_t offset, uint64_t size) {
-    TrackInfo *trackInfo = editTrack(mCurrentTrackID);
-
-    SampleDescription *sampleDesc =
-        &trackInfo->mSampleDescs.editItemAt(
-                trackInfo->mSampleDescs.size() - 1);
-
-    if (sampleDesc->mType != FOURCC('m', 'p', '4', 'a')
-            && sampleDesc->mType != FOURCC('m', 'p', '4', 'v')) {
-        return -EINVAL;
-    }
-
-    const uint8_t *ptr = mBuffer->data() + offset;
-
-    size -= offset;
-    offset = 0;
-
-    if (size < 4) {
-        return -EINVAL;
-    }
-
-    if (U32_AT(ptr) != 0) {
-        return -EINVAL;
-    }
-
-    ptr += 4;
-    size -=4;
-
-    ESDS esds(ptr, size);
-
-    uint8_t objectTypeIndication;
-    if (esds.getObjectTypeIndication(&objectTypeIndication) != OK) {
-        return ERROR_MALFORMED;
-    }
-
-    const uint8_t *csd;
-    size_t csd_size;
-    if (esds.getCodecSpecificInfo(
-                (const void **)&csd, &csd_size) != OK) {
-        return ERROR_MALFORMED;
-    }
-
-    addCodecSpecificData(sampleDesc->mFormat, 0, csd, csd_size);
-
-    if (sampleDesc->mType != FOURCC('m', 'p', '4', 'a')) {
-        return OK;
-    }
-
-    if (csd_size == 0) {
-        // There's no further information, i.e. no codec specific data
-        // Let's assume that the information provided in the mpeg4 headers
-        // is accurate and hope for the best.
-
-        return OK;
-    }
-
-    if (csd_size < 2) {
-        return ERROR_MALFORMED;
-    }
-
-    uint32_t objectType = csd[0] >> 3;
-
-    if (objectType == 31) {
-        return ERROR_UNSUPPORTED;
-    }
-
-    uint32_t freqIndex = (csd[0] & 7) << 1 | (csd[1] >> 7);
-    int32_t sampleRate = 0;
-    int32_t numChannels = 0;
-    if (freqIndex == 15) {
-        if (csd_size < 5) {
-            return ERROR_MALFORMED;
-        }
-
-        sampleRate = (csd[1] & 0x7f) << 17
-                        | csd[2] << 9
-                        | csd[3] << 1
-                        | (csd[4] >> 7);
-
-        numChannels = (csd[4] >> 3) & 15;
-    } else {
-        static uint32_t kSamplingRate[] = {
-            96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
-            16000, 12000, 11025, 8000, 7350
-        };
-
-        if (freqIndex == 13 || freqIndex == 14) {
-            return ERROR_MALFORMED;
-        }
-
-        sampleRate = kSamplingRate[freqIndex];
-        numChannels = (csd[1] >> 3) & 15;
-    }
-
-    if (numChannels == 0) {
-        return ERROR_UNSUPPORTED;
-    }
-
-    sampleDesc->mFormat->setInt32("sample-rate", sampleRate);
-    sampleDesc->mFormat->setInt32("channel-count", numChannels);
-
-    return OK;
-}
-
-status_t FragmentedMP4Parser::parseMediaData(
-        uint32_t type, size_t offset, uint64_t size) {
-    ALOGV("skipping 'mdat' chunk at offsets 0x%08lx-0x%08llx.",
-          mBufferPos + offset, mBufferPos + size);
-
-    sp<ABuffer> buffer = new ABuffer(size - offset);
-    memcpy(buffer->data(), mBuffer->data() + offset, size - offset);
-
-    mMediaData.push();
-    MediaDataInfo *info = &mMediaData.editItemAt(mMediaData.size() - 1);
-    info->mBuffer = buffer;
-    info->mOffset = mBufferPos + offset;
-
-    if (mMediaData.size() > 10) {
-        ALOGV("suspending for now.");
-        mSuspended = true;
-    }
-
-    return OK;
-}
-
-status_t FragmentedMP4Parser::parseSegmentIndex(
-        uint32_t type, size_t offset, uint64_t size) {
-    ALOGV("sidx box type %d, offset %d, size %d", type, int(offset), int(size));
-//    AString sidxstr;
-//    hexdump(mBuffer->data() + offset, size, 0 /* indent */, &sidxstr);
-//    ALOGV("raw sidx:");
-//    ALOGV("%s", sidxstr.c_str());
-    if (offset + 12 > size) {
-        return -EINVAL;
-    }
-
-    uint32_t flags = readU32(offset);
-
-    uint32_t version = flags >> 24;
-    flags &= 0xffffff;
-
-    ALOGV("sidx version %d", version);
-
-    uint32_t referenceId = readU32(offset + 4);
-    uint32_t timeScale = readU32(offset + 8);
-    ALOGV("sidx refid/timescale: %d/%d", referenceId, timeScale);
-
-    uint64_t earliestPresentationTime;
-    uint64_t firstOffset;
-
-    offset += 12;
-
-    if (version == 0) {
-        if (offset + 8 > size) {
-            return -EINVAL;
-        }
-        earliestPresentationTime = readU32(offset);
-        firstOffset = readU32(offset + 4);
-        offset += 8;
-    } else {
-        if (offset + 16 > size) {
-            return -EINVAL;
-        }
-        earliestPresentationTime = readU64(offset);
-        firstOffset = readU64(offset + 8);
-        offset += 16;
-    }
-    ALOGV("sidx pres/off: %Ld/%Ld", earliestPresentationTime, firstOffset);
-
-    if (offset + 4 > size) {
-        return -EINVAL;
-    }
-    if (readU16(offset) != 0) { // reserved
-        return -EINVAL;
-    }
-    int32_t referenceCount = readU16(offset + 2);
-    offset += 4;
-    ALOGV("refcount: %d", referenceCount);
-
-    if (offset + referenceCount * 12 > size) {
-        return -EINVAL;
-    }
-
-    TrackInfo *info = editTrack(mCurrentTrackID);
-    uint64_t total_duration = 0;
-    for (int i = 0; i < referenceCount; i++) {
-        uint32_t d1 = readU32(offset);
-        uint32_t d2 = readU32(offset + 4);
-        uint32_t d3 = readU32(offset + 8);
-
-        if (d1 & 0x80000000) {
-            ALOGW("sub-sidx boxes not supported yet");
-        }
-        bool sap = d3 & 0x80000000;
-        bool saptype = d3 >> 28;
-        if (!sap || saptype > 2) {
-            ALOGW("not a stream access point, or unsupported type");
-        }
-        total_duration += d2;
-        offset += 12;
-        ALOGV(" item %d, %08x %08x %08x", i, d1, d2, d3);
-        SidxEntry se;
-        se.mSize = d1 & 0x7fffffff;
-        se.mDurationUs = 1000000LL * d2 / timeScale;
-        info->mSidx.add(se);
-    }
-
-    info->mSidxDuration = total_duration * 1000000 / timeScale;
-    ALOGV("duration: %lld", info->mSidxDuration);
-    return OK;
-}
-
-status_t FragmentedMP4Parser::parseTrackExtends(
-        uint32_t type, size_t offset, uint64_t size) {
-    if (offset + 24 > size) {
-        return -EINVAL;
-    }
-
-    if (readU32(offset) != 0) {
-        return -EINVAL;
-    }
-
-    uint32_t trackID = readU32(offset + 4);
-
-    TrackInfo *info = editTrack(trackID, true /* createIfNecessary */);
-    info->mDefaultSampleDescriptionIndex = readU32(offset + 8);
-    info->mDefaultSampleDuration = readU32(offset + 12);
-    info->mDefaultSampleSize = readU32(offset + 16);
-    info->mDefaultSampleFlags = readU32(offset + 20);
-
-    return OK;
-}
-
-FragmentedMP4Parser::TrackInfo *FragmentedMP4Parser::editTrack(
-        uint32_t trackID, bool createIfNecessary) {
-    ssize_t i = mTracks.indexOfKey(trackID);
-
-    if (i >= 0) {
-        return &mTracks.editValueAt(i);
-    }
-
-    if (!createIfNecessary) {
-        return NULL;
-    }
-
-    TrackInfo info;
-    info.mTrackID = trackID;
-    info.mFlags = 0;
-    info.mDuration = 0xffffffff;
-    info.mSidxDuration = 0;
-    info.mMediaTimeScale = 0;
-    info.mMediaHandlerType = 0;
-    info.mDefaultSampleDescriptionIndex = 0;
-    info.mDefaultSampleDuration = 0;
-    info.mDefaultSampleSize = 0;
-    info.mDefaultSampleFlags = 0;
-
-    info.mDecodingTime = 0;
-
-    mTracks.add(trackID, info);
-    return &mTracks.editValueAt(mTracks.indexOfKey(trackID));
-}
-
-status_t FragmentedMP4Parser::parseTrackFragmentHeader(
-        uint32_t type, size_t offset, uint64_t size) {
-    if (offset + 8 > size) {
-        return -EINVAL;
-    }
-
-    uint32_t flags = readU32(offset);
-
-    if (flags & 0xff000000) {
-        return -EINVAL;
-    }
-
-    mTrackFragmentHeaderInfo.mFlags = flags;
-
-    mTrackFragmentHeaderInfo.mTrackID = readU32(offset + 4);
-    offset += 8;
-
-    if (flags & TrackFragmentHeaderInfo::kBaseDataOffsetPresent) {
-        if (offset + 8 > size) {
-            return -EINVAL;
-        }
-
-        mTrackFragmentHeaderInfo.mBaseDataOffset = readU64(offset);
-        offset += 8;
-    }
-
-    if (flags & TrackFragmentHeaderInfo::kSampleDescriptionIndexPresent) {
-        if (offset + 4 > size) {
-            return -EINVAL;
-        }
-
-        mTrackFragmentHeaderInfo.mSampleDescriptionIndex = readU32(offset);
-        offset += 4;
-    }
-
-    if (flags & TrackFragmentHeaderInfo::kDefaultSampleDurationPresent) {
-        if (offset + 4 > size) {
-            return -EINVAL;
-        }
-
-        mTrackFragmentHeaderInfo.mDefaultSampleDuration = readU32(offset);
-        offset += 4;
-    }
-
-    if (flags & TrackFragmentHeaderInfo::kDefaultSampleSizePresent) {
-        if (offset + 4 > size) {
-            return -EINVAL;
-        }
-
-        mTrackFragmentHeaderInfo.mDefaultSampleSize = readU32(offset);
-        offset += 4;
-    }
-
-    if (flags & TrackFragmentHeaderInfo::kDefaultSampleFlagsPresent) {
-        if (offset + 4 > size) {
-            return -EINVAL;
-        }
-
-        mTrackFragmentHeaderInfo.mDefaultSampleFlags = readU32(offset);
-        offset += 4;
-    }
-
-    if (!(flags & TrackFragmentHeaderInfo::kBaseDataOffsetPresent)) {
-        // This should point to the position of the first byte of the
-        // enclosing 'moof' container for the first track and
-        // the end of the data of the preceding fragment for subsequent
-        // tracks.
-
-        CHECK_GE(mStack.size(), 2u);
-
-        mTrackFragmentHeaderInfo.mBaseDataOffset =
-            mStack.itemAt(mStack.size() - 2).mOffset;
-
-        // XXX TODO: This does not do the right thing for the 2nd and
-        // subsequent tracks yet.
-    }
-
-    mTrackFragmentHeaderInfo.mDataOffset =
-        mTrackFragmentHeaderInfo.mBaseDataOffset;
-
-    TrackInfo *trackInfo = editTrack(mTrackFragmentHeaderInfo.mTrackID);
-
-    if (trackInfo->mFragments.empty()
-            || (*trackInfo->mFragments.begin())->complete()) {
-        trackInfo->mFragments.push_back(new DynamicTrackFragment);
-    }
-
-    return OK;
-}
-
-status_t FragmentedMP4Parser::parseTrackFragmentRun(
-        uint32_t type, size_t offset, uint64_t size) {
-    if (offset + 8 > size) {
-        return -EINVAL;
-    }
-
-    enum {
-        kDataOffsetPresent                  = 0x01,
-        kFirstSampleFlagsPresent            = 0x04,
-        kSampleDurationPresent              = 0x100,
-        kSampleSizePresent                  = 0x200,
-        kSampleFlagsPresent                 = 0x400,
-        kSampleCompositionTimeOffsetPresent = 0x800,
-    };
-
-    uint32_t flags = readU32(offset);
-
-    if (flags & 0xff000000) {
-        return -EINVAL;
-    }
-
-    if ((flags & kFirstSampleFlagsPresent) && (flags & kSampleFlagsPresent)) {
-        // These two shall not be used together.
-        return -EINVAL;
-    }
-
-    uint32_t sampleCount = readU32(offset + 4);
-    offset += 8;
-
-    uint64_t dataOffset = mTrackFragmentHeaderInfo.mDataOffset;
-
-    uint32_t firstSampleFlags = 0;
-
-    if (flags & kDataOffsetPresent) {
-        if (offset + 4 > size) {
-            return -EINVAL;
-        }
-
-        int32_t dataOffsetDelta = (int32_t)readU32(offset);
-
-        dataOffset = mTrackFragmentHeaderInfo.mBaseDataOffset + dataOffsetDelta;
-
-        offset += 4;
-    }
-
-    if (flags & kFirstSampleFlagsPresent) {
-        if (offset + 4 > size) {
-            return -EINVAL;
-        }
-
-        firstSampleFlags = readU32(offset);
-        offset += 4;
-    }
-
-    TrackInfo *info = editTrack(mTrackFragmentHeaderInfo.mTrackID);
-
-    if (info == NULL) {
-        return -EINVAL;
-    }
-
-    uint32_t sampleDuration = 0, sampleSize = 0, sampleFlags = 0,
-             sampleCtsOffset = 0;
-
-    size_t bytesPerSample = 0;
-    if (flags & kSampleDurationPresent) {
-        bytesPerSample += 4;
-    } else if (mTrackFragmentHeaderInfo.mFlags
-            & TrackFragmentHeaderInfo::kDefaultSampleDurationPresent) {
-        sampleDuration = mTrackFragmentHeaderInfo.mDefaultSampleDuration;
-    } else {
-        sampleDuration = info->mDefaultSampleDuration;
-    }
-
-    if (flags & kSampleSizePresent) {
-        bytesPerSample += 4;
-    } else if (mTrackFragmentHeaderInfo.mFlags
-            & TrackFragmentHeaderInfo::kDefaultSampleSizePresent) {
-        sampleSize = mTrackFragmentHeaderInfo.mDefaultSampleSize;
-    } else {
-        sampleSize = info->mDefaultSampleSize;
-    }
-
-    if (flags & kSampleFlagsPresent) {
-        bytesPerSample += 4;
-    } else if (mTrackFragmentHeaderInfo.mFlags
-            & TrackFragmentHeaderInfo::kDefaultSampleFlagsPresent) {
-        sampleFlags = mTrackFragmentHeaderInfo.mDefaultSampleFlags;
-    } else {
-        sampleFlags = info->mDefaultSampleFlags;
-    }
-
-    if (flags & kSampleCompositionTimeOffsetPresent) {
-        bytesPerSample += 4;
-    } else {
-        sampleCtsOffset = 0;
-    }
-
-    if (offset + sampleCount * bytesPerSample > size) {
-        return -EINVAL;
-    }
-
-    uint32_t sampleDescIndex =
-        (mTrackFragmentHeaderInfo.mFlags
-            & TrackFragmentHeaderInfo::kSampleDescriptionIndexPresent)
-            ? mTrackFragmentHeaderInfo.mSampleDescriptionIndex
-            : info->mDefaultSampleDescriptionIndex;
-
-    for (uint32_t i = 0; i < sampleCount; ++i) {
-        if (flags & kSampleDurationPresent) {
-            sampleDuration = readU32(offset);
-            offset += 4;
-        }
-
-        if (flags & kSampleSizePresent) {
-            sampleSize = readU32(offset);
-            offset += 4;
-        }
-
-        if (flags & kSampleFlagsPresent) {
-            sampleFlags = readU32(offset);
-            offset += 4;
-        }
-
-        if (flags & kSampleCompositionTimeOffsetPresent) {
-            sampleCtsOffset = readU32(offset);
-            offset += 4;
-        }
-
-        ALOGV("adding sample at offset 0x%08llx, size %u, duration %u, "
-              "sampleDescIndex=%u, flags 0x%08x",
-                dataOffset, sampleSize, sampleDuration,
-                sampleDescIndex,
-                (flags & kFirstSampleFlagsPresent) && i == 0
-                    ? firstSampleFlags : sampleFlags);
-
-        const sp<TrackFragment> &fragment = *--info->mFragments.end();
-
-        uint32_t decodingTime = info->mDecodingTime;
-        info->mDecodingTime += sampleDuration;
-        uint32_t presentationTime = decodingTime + sampleCtsOffset;
-
-        static_cast<DynamicTrackFragment *>(
-                fragment.get())->addSample(
-                    dataOffset,
-                    sampleSize,
-                    presentationTime,
-                    sampleDescIndex,
-                    ((flags & kFirstSampleFlagsPresent) && i == 0)
-                        ? firstSampleFlags : sampleFlags);
-
-        dataOffset += sampleSize;
-    }
-
-    mTrackFragmentHeaderInfo.mDataOffset = dataOffset;
-
-    return OK;
-}
-
-void FragmentedMP4Parser::copyBuffer(
-        sp<ABuffer> *dst, size_t offset, uint64_t size) const {
-    sp<ABuffer> buf = new ABuffer(size);
-    memcpy(buf->data(), mBuffer->data() + offset, size);
-
-    *dst = buf;
-}
-
-}  // namespace android
diff --git a/media/libstagefright/mp4/TrackFragment.cpp b/media/libstagefright/mp4/TrackFragment.cpp
deleted file mode 100644
index 3699038..0000000
--- a/media/libstagefright/mp4/TrackFragment.cpp
+++ /dev/null
@@ -1,364 +0,0 @@
-/*
- * Copyright (C) 2012 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "TrackFragment"
-#include <utils/Log.h>
-
-#include "TrackFragment.h"
-
-#include <media/stagefright/MediaErrors.h>
-#include <media/stagefright/Utils.h>
-#include <media/stagefright/foundation/ABuffer.h>
-#include <media/stagefright/foundation/ADebug.h>
-#include <media/stagefright/foundation/hexdump.h>
-
-namespace android {
-
-FragmentedMP4Parser::DynamicTrackFragment::DynamicTrackFragment()
-    : mComplete(false),
-      mSampleIndex(0) {
-}
-
-FragmentedMP4Parser::DynamicTrackFragment::~DynamicTrackFragment() {
-}
-
-status_t FragmentedMP4Parser::DynamicTrackFragment::getSample(SampleInfo *info) {
-    if (mSampleIndex >= mSamples.size()) {
-        return mComplete ? ERROR_END_OF_STREAM : -EWOULDBLOCK;
-    }
-
-    *info = mSamples.itemAt(mSampleIndex);
-
-    return OK;
-}
-
-void FragmentedMP4Parser::DynamicTrackFragment::advance() {
-    ++mSampleIndex;
-}
-
-void FragmentedMP4Parser::DynamicTrackFragment::addSample(
-        off64_t dataOffset, size_t sampleSize,
-        uint32_t presentationTime,
-        size_t sampleDescIndex,
-        uint32_t flags) {
-    mSamples.push();
-    SampleInfo *sampleInfo = &mSamples.editItemAt(mSamples.size() - 1);
-
-    sampleInfo->mOffset = dataOffset;
-    sampleInfo->mSize = sampleSize;
-    sampleInfo->mPresentationTime = presentationTime;
-    sampleInfo->mSampleDescIndex = sampleDescIndex;
-    sampleInfo->mFlags = flags;
-}
-
-status_t FragmentedMP4Parser::DynamicTrackFragment::signalCompletion() {
-    mComplete = true;
-
-    return OK;
-}
-
-bool FragmentedMP4Parser::DynamicTrackFragment::complete() const {
-    return mComplete;
-}
-
-////////////////////////////////////////////////////////////////////////////////
-
-FragmentedMP4Parser::StaticTrackFragment::StaticTrackFragment()
-    : mSampleIndex(0),
-      mSampleCount(0),
-      mChunkIndex(0),
-      mSampleToChunkIndex(-1),
-      mSampleToChunkRemaining(0),
-      mPrevChunkIndex(0xffffffff),
-      mNextSampleOffset(0) {
-}
-
-FragmentedMP4Parser::StaticTrackFragment::~StaticTrackFragment() {
-}
-
-status_t FragmentedMP4Parser::StaticTrackFragment::getSample(SampleInfo *info) {
-    if (mSampleIndex >= mSampleCount) {
-        return ERROR_END_OF_STREAM;
-    }
-
-    *info = mSampleInfo;
-
-    ALOGV("returning sample %d at [0x%08llx, 0x%08llx)",
-          mSampleIndex,
-          info->mOffset, info->mOffset + info->mSize);
-
-    return OK;
-}
-
-void FragmentedMP4Parser::StaticTrackFragment::updateSampleInfo() {
-    if (mSampleIndex >= mSampleCount) {
-        return;
-    }
-
-    if (mSampleSizes != NULL) {
-        uint32_t defaultSampleSize = U32_AT(mSampleSizes->data() + 4);
-        if (defaultSampleSize > 0) {
-            mSampleInfo.mSize = defaultSampleSize;
-        } else {
-            mSampleInfo.mSize= U32_AT(mSampleSizes->data() + 12 + 4 * mSampleIndex);
-        }
-    } else {
-        CHECK(mCompactSampleSizes != NULL);
-
-        uint32_t fieldSize = U32_AT(mCompactSampleSizes->data() + 4);
-
-        switch (fieldSize) {
-            case 4:
-            {
-                unsigned byte = mCompactSampleSizes->data()[12 + mSampleIndex / 2];
-                mSampleInfo.mSize = (mSampleIndex & 1) ? byte & 0x0f : byte >> 4;
-                break;
-            }
-
-            case 8:
-            {
-                mSampleInfo.mSize = mCompactSampleSizes->data()[12 + mSampleIndex];
-                break;
-            }
-
-            default:
-            {
-                CHECK_EQ(fieldSize, 16);
-                mSampleInfo.mSize =
-                    U16_AT(mCompactSampleSizes->data() + 12 + mSampleIndex * 2);
-                break;
-            }
-        }
-    }
-
-    CHECK_GT(mSampleToChunkRemaining, 0);
-
-    // The sample desc index is 1-based... XXX
-    mSampleInfo.mSampleDescIndex =
-        U32_AT(mSampleToChunk->data() + 8 + 12 * mSampleToChunkIndex + 8);
-
-    if (mChunkIndex != mPrevChunkIndex) {
-        mPrevChunkIndex = mChunkIndex;
-
-        if (mChunkOffsets != NULL) {
-            uint32_t entryCount = U32_AT(mChunkOffsets->data() + 4);
-
-            if (mChunkIndex >= entryCount) {
-                mSampleIndex = mSampleCount;
-                return;
-            }
-
-            mNextSampleOffset =
-                U32_AT(mChunkOffsets->data() + 8 + 4 * mChunkIndex);
-        } else {
-            CHECK(mChunkOffsets64 != NULL);
-
-            uint32_t entryCount = U32_AT(mChunkOffsets64->data() + 4);
-
-            if (mChunkIndex >= entryCount) {
-                mSampleIndex = mSampleCount;
-                return;
-            }
-
-            mNextSampleOffset =
-                U64_AT(mChunkOffsets64->data() + 8 + 8 * mChunkIndex);
-        }
-    }
-
-    mSampleInfo.mOffset = mNextSampleOffset;
-
-    mSampleInfo.mPresentationTime = 0;
-    mSampleInfo.mFlags = 0;
-}
-
-void FragmentedMP4Parser::StaticTrackFragment::advance() {
-    mNextSampleOffset += mSampleInfo.mSize;
-
-    ++mSampleIndex;
-    if (--mSampleToChunkRemaining == 0) {
-        ++mChunkIndex;
-
-        uint32_t entryCount = U32_AT(mSampleToChunk->data() + 4);
-
-        // If this is the last entry in the sample to chunk table, we will
-        // stay on this entry.
-        if ((uint32_t)(mSampleToChunkIndex + 1) < entryCount) {
-            uint32_t nextChunkIndex =
-                U32_AT(mSampleToChunk->data() + 8 + 12 * (mSampleToChunkIndex + 1));
-
-            CHECK_GE(nextChunkIndex, 1u);
-            --nextChunkIndex;
-
-            if (mChunkIndex >= nextChunkIndex) {
-                CHECK_EQ(mChunkIndex, nextChunkIndex);
-                ++mSampleToChunkIndex;
-            }
-        }
-
-        mSampleToChunkRemaining =
-            U32_AT(mSampleToChunk->data() + 8 + 12 * mSampleToChunkIndex + 4);
-    }
-
-    updateSampleInfo();
-}
-
-static void setU32At(uint8_t *ptr, uint32_t x) {
-    ptr[0] = x >> 24;
-    ptr[1] = (x >> 16) & 0xff;
-    ptr[2] = (x >> 8) & 0xff;
-    ptr[3] = x & 0xff;
-}
-
-status_t FragmentedMP4Parser::StaticTrackFragment::signalCompletion() {
-    mSampleToChunkIndex = 0;
-
-    mSampleToChunkRemaining =
-        (mSampleToChunk == NULL)
-            ? 0
-            : U32_AT(mSampleToChunk->data() + 8 + 12 * mSampleToChunkIndex + 4);
-
-    updateSampleInfo();
-
-    return OK;
-}
-
-bool FragmentedMP4Parser::StaticTrackFragment::complete() const {
-    return true;
-}
-
-status_t FragmentedMP4Parser::StaticTrackFragment::parseSampleSizes(
-        FragmentedMP4Parser *parser, uint32_t type, size_t offset, uint64_t size) {
-    if (offset + 12 > size) {
-        return ERROR_MALFORMED;
-    }
-
-    if (parser->readU32(offset) != 0) {
-        return ERROR_MALFORMED;
-    }
-
-    uint32_t sampleSize = parser->readU32(offset + 4);
-    uint32_t sampleCount = parser->readU32(offset + 8);
-
-    if (sampleSize == 0 && offset + 12 + sampleCount * 4 != size) {
-        return ERROR_MALFORMED;
-    }
-
-    parser->copyBuffer(&mSampleSizes, offset, size);
-
-    mSampleCount = sampleCount;
-
-    return OK;
-}
-
-status_t FragmentedMP4Parser::StaticTrackFragment::parseCompactSampleSizes(
-        FragmentedMP4Parser *parser, uint32_t type, size_t offset, uint64_t size) {
-    if (offset + 12 > size) {
-        return ERROR_MALFORMED;
-    }
-
-    if (parser->readU32(offset) != 0) {
-        return ERROR_MALFORMED;
-    }
-
-    uint32_t fieldSize = parser->readU32(offset + 4);
-
-    if (fieldSize != 4 && fieldSize != 8 && fieldSize != 16) {
-        return ERROR_MALFORMED;
-    }
-
-    uint32_t sampleCount = parser->readU32(offset + 8);
-
-    if (offset + 12 + (sampleCount * fieldSize + 4) / 8 != size) {
-        return ERROR_MALFORMED;
-    }
-
-    parser->copyBuffer(&mCompactSampleSizes, offset, size);
-
-    mSampleCount = sampleCount;
-
-    return OK;
-}
-
-status_t FragmentedMP4Parser::StaticTrackFragment::parseSampleToChunk(
-        FragmentedMP4Parser *parser, uint32_t type, size_t offset, uint64_t size) {
-    if (offset + 8 > size) {
-        return ERROR_MALFORMED;
-    }
-
-    if (parser->readU32(offset) != 0) {
-        return ERROR_MALFORMED;
-    }
-
-    uint32_t entryCount = parser->readU32(offset + 4);
-
-    if (entryCount == 0) {
-        return OK;
-    }
-
-    if (offset + 8 + entryCount * 12 != size) {
-        return ERROR_MALFORMED;
-    }
-
-    parser->copyBuffer(&mSampleToChunk, offset, size);
-
-    return OK;
-}
-
-status_t FragmentedMP4Parser::StaticTrackFragment::parseChunkOffsets(
-        FragmentedMP4Parser *parser, uint32_t type, size_t offset, uint64_t size) {
-    if (offset + 8 > size) {
-        return ERROR_MALFORMED;
-    }
-
-    if (parser->readU32(offset) != 0) {
-        return ERROR_MALFORMED;
-    }
-
-    uint32_t entryCount = parser->readU32(offset + 4);
-
-    if (offset + 8 + entryCount * 4 != size) {
-        return ERROR_MALFORMED;
-    }
-
-    parser->copyBuffer(&mChunkOffsets, offset, size);
-
-    return OK;
-}
-
-status_t FragmentedMP4Parser::StaticTrackFragment::parseChunkOffsets64(
-        FragmentedMP4Parser *parser, uint32_t type, size_t offset, uint64_t size) {
-    if (offset + 8 > size) {
-        return ERROR_MALFORMED;
-    }
-
-    if (parser->readU32(offset) != 0) {
-        return ERROR_MALFORMED;
-    }
-
-    uint32_t entryCount = parser->readU32(offset + 4);
-
-    if (offset + 8 + entryCount * 8 != size) {
-        return ERROR_MALFORMED;
-    }
-
-    parser->copyBuffer(&mChunkOffsets64, offset, size);
-
-    return OK;
-}
-
-}  // namespace android
-
diff --git a/media/libstagefright/mp4/TrackFragment.h b/media/libstagefright/mp4/TrackFragment.h
deleted file mode 100644
index e1ad46e..0000000
--- a/media/libstagefright/mp4/TrackFragment.h
+++ /dev/null
@@ -1,122 +0,0 @@
-/*
- * Copyright (C) 2012 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef TRACK_FRAGMENT_H_
-
-#define TRACK_FRAGMENT_H_
-
-#include "include/FragmentedMP4Parser.h"
-
-namespace android {
-
-struct FragmentedMP4Parser::TrackFragment : public RefBase {
-    TrackFragment() {}
-
-    virtual status_t getSample(SampleInfo *info) = 0;
-    virtual void advance() = 0;
-
-    virtual status_t signalCompletion() = 0;
-    virtual bool complete() const = 0;
-
-protected:
-    virtual ~TrackFragment() {}
-
-private:
-    DISALLOW_EVIL_CONSTRUCTORS(TrackFragment);
-};
-
-struct FragmentedMP4Parser::DynamicTrackFragment : public FragmentedMP4Parser::TrackFragment {
-    DynamicTrackFragment();
-
-    virtual status_t getSample(SampleInfo *info);
-    virtual void advance();
-
-    void addSample(
-            off64_t dataOffset, size_t sampleSize,
-            uint32_t presentationTime,
-            size_t sampleDescIndex,
-            uint32_t flags);
-
-    // No more samples will be added to this fragment.
-    virtual status_t signalCompletion();
-
-    virtual bool complete() const;
-
-protected:
-    virtual ~DynamicTrackFragment();
-
-private:
-    bool mComplete;
-    size_t mSampleIndex;
-    Vector<SampleInfo> mSamples;
-
-    DISALLOW_EVIL_CONSTRUCTORS(DynamicTrackFragment);
-};
-
-struct FragmentedMP4Parser::StaticTrackFragment : public FragmentedMP4Parser::TrackFragment {
-    StaticTrackFragment();
-
-    virtual status_t getSample(SampleInfo *info);
-    virtual void advance();
-
-    virtual status_t signalCompletion();
-    virtual bool complete() const;
-
-    status_t parseSampleSizes(
-            FragmentedMP4Parser *parser, uint32_t type, size_t offset, uint64_t size);
-
-    status_t parseCompactSampleSizes(
-            FragmentedMP4Parser *parser, uint32_t type, size_t offset, uint64_t size);
-
-    status_t parseSampleToChunk(
-            FragmentedMP4Parser *parser, uint32_t type, size_t offset, uint64_t size);
-
-    status_t parseChunkOffsets(
-            FragmentedMP4Parser *parser, uint32_t type, size_t offset, uint64_t size);
-
-    status_t parseChunkOffsets64(
-            FragmentedMP4Parser *parser, uint32_t type, size_t offset, uint64_t size);
-
-protected:
-    virtual ~StaticTrackFragment();
-
-private:
-    size_t mSampleIndex;
-    size_t mSampleCount;
-    uint32_t mChunkIndex;
-
-    SampleInfo mSampleInfo;
-
-    sp<ABuffer> mSampleSizes;
-    sp<ABuffer> mCompactSampleSizes;
-
-    sp<ABuffer> mSampleToChunk;
-    ssize_t mSampleToChunkIndex;
-    size_t mSampleToChunkRemaining;
-
-    sp<ABuffer> mChunkOffsets;
-    sp<ABuffer> mChunkOffsets64;
-    uint32_t mPrevChunkIndex;
-    uint64_t mNextSampleOffset;
-
-    void updateSampleInfo();
-
-    DISALLOW_EVIL_CONSTRUCTORS(StaticTrackFragment);
-};
-
-}  // namespace android
-
-#endif  // TRACK_FRAGMENT_H_
diff --git a/media/libstagefright/mpeg2ts/ATSParser.cpp b/media/libstagefright/mpeg2ts/ATSParser.cpp
index cb57a2f..d039f7d 100644
--- a/media/libstagefright/mpeg2ts/ATSParser.cpp
+++ b/media/libstagefright/mpeg2ts/ATSParser.cpp
@@ -862,7 +862,7 @@
 }
 
 void ATSParser::Stream::onPayloadData(
-        unsigned PTS_DTS_flags, uint64_t PTS, uint64_t DTS,
+        unsigned PTS_DTS_flags, uint64_t PTS, uint64_t /* DTS */,
         const uint8_t *data, size_t size) {
 #if 0
     ALOGI("payload streamType 0x%02x, PTS = 0x%016llx, dPTS = %lld",
@@ -1267,7 +1267,7 @@
 }
 
 void ATSParser::updatePCR(
-        unsigned PID, uint64_t PCR, size_t byteOffsetFromStart) {
+        unsigned /* PID */, uint64_t PCR, size_t byteOffsetFromStart) {
     ALOGV("PCR 0x%016llx @ %d", PCR, byteOffsetFromStart);
 
     if (mNumPCRs == 2) {
diff --git a/media/libstagefright/mpeg2ts/Android.mk b/media/libstagefright/mpeg2ts/Android.mk
index c1a7a9d..c17a0b7 100644
--- a/media/libstagefright/mpeg2ts/Android.mk
+++ b/media/libstagefright/mpeg2ts/Android.mk
@@ -13,6 +13,8 @@
 	$(TOP)/frameworks/av/media/libstagefright \
 	$(TOP)/frameworks/native/include/media/openmax
 
+LOCAL_CFLAGS += -Werror
+
 LOCAL_MODULE:= libstagefright_mpeg2ts
 
 ifeq ($(TARGET_ARCH),arm)
diff --git a/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp b/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp
index 2b0bf30..6dfaa94 100644
--- a/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp
+++ b/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp
@@ -62,7 +62,7 @@
 AnotherPacketSource::~AnotherPacketSource() {
 }
 
-status_t AnotherPacketSource::start(MetaData *params) {
+status_t AnotherPacketSource::start(MetaData * /* params */) {
     return OK;
 }
 
diff --git a/media/libstagefright/mpeg2ts/ESQueue.cpp b/media/libstagefright/mpeg2ts/ESQueue.cpp
index ea79885..c0c9717 100644
--- a/media/libstagefright/mpeg2ts/ESQueue.cpp
+++ b/media/libstagefright/mpeg2ts/ESQueue.cpp
@@ -60,7 +60,7 @@
 // update metadata only applicable, and return the payload size
 static unsigned parseAC3SyncFrame(
         const uint8_t *ptr, size_t size, sp<MetaData> *metaData) {
-    static const unsigned channelCountTable[] = {2, 1, 2, 3, 4, 4, 5, 6};
+    static const unsigned channelCountTable[] = {2, 1, 2, 3, 3, 4, 4, 5};
     static const unsigned samplingRateTable[] = {48000, 44100, 32000};
     static const unsigned rates[] = {32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256,
             320, 384, 448, 512, 576, 640};
@@ -794,7 +794,9 @@
                 dstOffset += pos.nalSize + 4;
             }
 
+#if !LOG_NDEBUG
             ALOGV("accessUnit contains nal types %s", out.c_str());
+#endif
 
             const NALPosition &pos = nals.itemAt(nals.size() - 1);
             size_t nextScan = pos.nalOffset + pos.nalSize;
diff --git a/media/libstagefright/mpeg2ts/MPEG2PSExtractor.cpp b/media/libstagefright/mpeg2ts/MPEG2PSExtractor.cpp
index dd714c99..bc2a16d 100644
--- a/media/libstagefright/mpeg2ts/MPEG2PSExtractor.cpp
+++ b/media/libstagefright/mpeg2ts/MPEG2PSExtractor.cpp
@@ -130,7 +130,8 @@
     return new WrappedTrack(this, mTracks.valueAt(index));
 }
 
-sp<MetaData> MPEG2PSExtractor::getTrackMetaData(size_t index, uint32_t flags) {
+sp<MetaData> MPEG2PSExtractor::getTrackMetaData(
+        size_t index, uint32_t /* flags */) {
     if (index >= mTracks.size()) {
         return NULL;
     }
@@ -625,7 +626,7 @@
 
 status_t MPEG2PSExtractor::Track::appendPESData(
         unsigned PTS_DTS_flags,
-        uint64_t PTS, uint64_t DTS,
+        uint64_t PTS, uint64_t /* DTS */,
         const uint8_t *data, size_t size) {
     if (mQueue == NULL) {
         return OK;
diff --git a/media/libstagefright/mpeg2ts/MPEG2TSExtractor.cpp b/media/libstagefright/mpeg2ts/MPEG2TSExtractor.cpp
index d449c34..35ca118 100644
--- a/media/libstagefright/mpeg2ts/MPEG2TSExtractor.cpp
+++ b/media/libstagefright/mpeg2ts/MPEG2TSExtractor.cpp
@@ -141,7 +141,7 @@
 }
 
 sp<MetaData> MPEG2TSExtractor::getTrackMetaData(
-        size_t index, uint32_t flags) {
+        size_t index, uint32_t /* flags */) {
     return index < mSourceImpls.size()
         ? mSourceImpls.editItemAt(index)->getFormat() : NULL;
 }
diff --git a/media/libstagefright/omx/GraphicBufferSource.cpp b/media/libstagefright/omx/GraphicBufferSource.cpp
index b8970ad..7672204 100644
--- a/media/libstagefright/omx/GraphicBufferSource.cpp
+++ b/media/libstagefright/omx/GraphicBufferSource.cpp
@@ -41,8 +41,13 @@
     mNumFramesAvailable(0),
     mEndOfStream(false),
     mEndOfStreamSent(false),
+    mMaxTimestampGapUs(-1ll),
+    mPrevOriginalTimeUs(-1ll),
+    mPrevModifiedTimeUs(-1ll),
+    mSkipFramesBeforeNs(-1ll),
     mRepeatAfterUs(-1ll),
     mRepeatLastFrameGeneration(0),
+    mRepeatLastFrameTimestamp(-1ll),
     mLatestSubmittedBufferId(-1),
     mLatestSubmittedBufferFrameNum(0),
     mLatestSubmittedBufferUseCount(0),
@@ -299,6 +304,32 @@
     return;
 }
 
+void GraphicBufferSource::codecBufferFilled(OMX_BUFFERHEADERTYPE* header) {
+    Mutex::Autolock autoLock(mMutex);
+
+    if (mMaxTimestampGapUs > 0ll
+            && !(header->nFlags & OMX_BUFFERFLAG_CODECCONFIG)) {
+        ssize_t index = mOriginalTimeUs.indexOfKey(header->nTimeStamp);
+        if (index >= 0) {
+            ALOGV("OUT timestamp: %lld -> %lld",
+                    header->nTimeStamp, mOriginalTimeUs[index]);
+            header->nTimeStamp = mOriginalTimeUs[index];
+            mOriginalTimeUs.removeItemsAt(index);
+        } else {
+            // giving up the effort as encoder doesn't appear to preserve pts
+            ALOGW("giving up limiting timestamp gap (pts = %lld)",
+                    header->nTimeStamp);
+            mMaxTimestampGapUs = -1ll;
+        }
+        if (mOriginalTimeUs.size() > BufferQueue::NUM_BUFFER_SLOTS) {
+            // something terribly wrong must have happened, giving up...
+            ALOGE("mOriginalTimeUs has too many entries (%d)",
+                    mOriginalTimeUs.size());
+            mMaxTimestampGapUs = -1ll;
+        }
+    }
+}
+
 void GraphicBufferSource::suspend(bool suspend) {
     Mutex::Autolock autoLock(mMutex);
 
@@ -384,7 +415,18 @@
         mBufferSlot[item.mBuf] = item.mGraphicBuffer;
     }
 
-    err = submitBuffer_l(item, cbi);
+    err = UNKNOWN_ERROR;
+
+    // only submit sample if start time is unspecified, or sample
+    // is queued after the specified start time
+    if (mSkipFramesBeforeNs < 0ll || item.mTimestamp >= mSkipFramesBeforeNs) {
+        // if start time is set, offset time stamp by start time
+        if (mSkipFramesBeforeNs > 0) {
+            item.mTimestamp -= mSkipFramesBeforeNs;
+        }
+        err = submitBuffer_l(item, cbi);
+    }
+
     if (err != OK) {
         ALOGV("submitBuffer_l failed, releasing bq buf %d", item.mBuf);
         mBufferQueue->releaseBuffer(item.mBuf, item.mFrameNumber,
@@ -431,6 +473,7 @@
     BufferQueue::BufferItem item;
     item.mBuf = mLatestSubmittedBufferId;
     item.mFrameNumber = mLatestSubmittedBufferFrameNum;
+    item.mTimestamp = mRepeatLastFrameTimestamp;
 
     status_t err = submitBuffer_l(item, cbi);
 
@@ -440,6 +483,20 @@
 
     ++mLatestSubmittedBufferUseCount;
 
+    /* repeat last frame up to kRepeatLastFrameCount times.
+     * in case of static scene, a single repeat might not get rid of encoder
+     * ghosting completely, refresh a couple more times to get better quality
+     */
+    if (--mRepeatLastFrameCount > 0) {
+        mRepeatLastFrameTimestamp = item.mTimestamp + mRepeatAfterUs * 1000;
+
+        if (mReflector != NULL) {
+            sp<AMessage> msg = new AMessage(kWhatRepeatLastFrame, mReflector->id());
+            msg->setInt32("generation", ++mRepeatLastFrameGeneration);
+            msg->post(mRepeatAfterUs);
+        }
+    }
+
     return true;
 }
 
@@ -460,8 +517,11 @@
 
     mLatestSubmittedBufferId = item.mBuf;
     mLatestSubmittedBufferFrameNum = item.mFrameNumber;
+    mRepeatLastFrameTimestamp = item.mTimestamp + mRepeatAfterUs * 1000;
+
     mLatestSubmittedBufferUseCount = 1;
     mRepeatBufferDeferred = false;
+    mRepeatLastFrameCount = kRepeatLastFrameCount;
 
     if (mReflector != NULL) {
         sp<AMessage> msg = new AMessage(kWhatRepeatLastFrame, mReflector->id());
@@ -497,6 +557,39 @@
     return OK;
 }
 
+int64_t GraphicBufferSource::getTimestamp(const BufferQueue::BufferItem &item) {
+    int64_t timeUs = item.mTimestamp / 1000;
+
+    if (mMaxTimestampGapUs > 0ll) {
+        /* Cap timestamp gap between adjacent frames to specified max
+         *
+         * In the scenario of cast mirroring, encoding could be suspended for
+         * prolonged periods. Limiting the pts gap to workaround the problem
+         * where encoder's rate control logic produces huge frames after a
+         * long period of suspension.
+         */
+
+        int64_t originalTimeUs = timeUs;
+        if (mPrevOriginalTimeUs >= 0ll) {
+            if (originalTimeUs < mPrevOriginalTimeUs) {
+                // Drop the frame if it's going backward in time. Bad timestamp
+                // could disrupt encoder's rate control completely.
+                ALOGV("Dropping frame that's going backward in time");
+                return -1;
+            }
+            int64_t timestampGapUs = originalTimeUs - mPrevOriginalTimeUs;
+            timeUs = (timestampGapUs < mMaxTimestampGapUs ?
+                    timestampGapUs : mMaxTimestampGapUs) + mPrevModifiedTimeUs;
+        }
+        mPrevOriginalTimeUs = originalTimeUs;
+        mPrevModifiedTimeUs = timeUs;
+        mOriginalTimeUs.add(timeUs, originalTimeUs);
+        ALOGV("IN  timestamp: %lld -> %lld", originalTimeUs, timeUs);
+    }
+
+    return timeUs;
+}
+
 status_t GraphicBufferSource::submitBuffer_l(
         const BufferQueue::BufferItem &item, int cbi) {
     ALOGV("submitBuffer_l cbi=%d", cbi);
@@ -513,9 +606,15 @@
     memcpy(data, &type, 4);
     memcpy(data + 4, &handle, sizeof(buffer_handle_t));
 
+    int64_t timeUs = getTimestamp(item);
+    if (timeUs < 0ll) {
+        ALOGE("Dropping frame with bad timestamp");
+        return UNKNOWN_ERROR;
+    }
+
     status_t err = mNodeInstance->emptyDirectBuffer(header, 0,
             4 + sizeof(buffer_handle_t), OMX_BUFFERFLAG_ENDOFFRAME,
-            item.mTimestamp / 1000);
+            timeUs);
     if (err != OK) {
         ALOGW("WARNING: emptyDirectBuffer failed: 0x%x", err);
         codecBuffer.mGraphicBuffer = NULL;
@@ -609,6 +708,12 @@
         BufferQueue::BufferItem item;
         status_t err = mBufferQueue->acquireBuffer(&item, 0);
         if (err == OK) {
+            // If this is the first time we're seeing this buffer, add it to our
+            // slot table.
+            if (item.mGraphicBuffer != NULL) {
+                ALOGV("fillCodecBuffer_l: setting mBufferSlot %d", item.mBuf);
+                mBufferSlot[item.mBuf] = item.mGraphicBuffer;
+            }
             mBufferQueue->releaseBuffer(item.mBuf, item.mFrameNumber,
                     EGL_NO_DISPLAY, EGL_NO_SYNC_KHR, item.mFence);
         }
@@ -658,6 +763,25 @@
     return OK;
 }
 
+status_t GraphicBufferSource::setMaxTimestampGapUs(int64_t maxGapUs) {
+    Mutex::Autolock autoLock(mMutex);
+
+    if (mExecuting || maxGapUs <= 0ll) {
+        return INVALID_OPERATION;
+    }
+
+    mMaxTimestampGapUs = maxGapUs;
+
+    return OK;
+}
+
+void GraphicBufferSource::setSkipFramesBeforeUs(int64_t skipFramesBeforeUs) {
+    Mutex::Autolock autoLock(mMutex);
+
+    mSkipFramesBeforeNs =
+            (skipFramesBeforeUs > 0) ? (skipFramesBeforeUs * 1000) : -1ll;
+}
+
 void GraphicBufferSource::onMessageReceived(const sp<AMessage> &msg) {
     switch (msg->what()) {
         case kWhatRepeatLastFrame:
diff --git a/media/libstagefright/omx/GraphicBufferSource.h b/media/libstagefright/omx/GraphicBufferSource.h
index 9e5eee6..153f2a0 100644
--- a/media/libstagefright/omx/GraphicBufferSource.h
+++ b/media/libstagefright/omx/GraphicBufferSource.h
@@ -87,6 +87,10 @@
     // fill it with a new frame of data; otherwise, just mark it as available.
     void codecBufferEmptied(OMX_BUFFERHEADERTYPE* header);
 
+    // Called when omx_message::FILL_BUFFER_DONE is received. (Currently the
+    // buffer source will fix timestamp in the header if needed.)
+    void codecBufferFilled(OMX_BUFFERHEADERTYPE* header);
+
     // This is called after the last input frame has been submitted.  We
     // need to submit an empty buffer with the EOS flag set.  If we don't
     // have a codec buffer ready, we just set the mEndOfStream flag.
@@ -105,6 +109,19 @@
     // state and once this behaviour is specified it cannot be reset.
     status_t setRepeatPreviousFrameDelayUs(int64_t repeatAfterUs);
 
+    // When set, the timestamp fed to the encoder will be modified such that
+    // the gap between two adjacent frames is capped at maxGapUs. Timestamp
+    // will be restored to the original when the encoded frame is returned to
+    // the client.
+    // This is to solve a problem in certain real-time streaming case, where
+    // encoder's rate control logic produces huge frames after a long period
+    // of suspension on input.
+    status_t setMaxTimestampGapUs(int64_t maxGapUs);
+
+    // Sets the start time us (in system time), samples before which should
+    // be dropped and not submitted to encoder
+    void setSkipFramesBeforeUs(int64_t startTimeUs);
+
 protected:
     // BufferQueue::ConsumerListener interface, called when a new frame of
     // data is available.  If we're executing and a codec buffer is
@@ -165,6 +182,7 @@
 
     void setLatestSubmittedBuffer_l(const BufferQueue::BufferItem &item);
     bool repeatLatestSubmittedBuffer_l();
+    int64_t getTimestamp(const BufferQueue::BufferItem &item);
 
     // Lock, covers all member variables.
     mutable Mutex mMutex;
@@ -206,13 +224,23 @@
     enum {
         kWhatRepeatLastFrame,
     };
+    enum {
+        kRepeatLastFrameCount = 10,
+    };
 
-    int64_t mRepeatAfterUs;
+    KeyedVector<int64_t, int64_t> mOriginalTimeUs;
+    int64_t mMaxTimestampGapUs;
+    int64_t mPrevOriginalTimeUs;
+    int64_t mPrevModifiedTimeUs;
+    int64_t mSkipFramesBeforeNs;
 
     sp<ALooper> mLooper;
     sp<AHandlerReflector<GraphicBufferSource> > mReflector;
 
+    int64_t mRepeatAfterUs;
     int32_t mRepeatLastFrameGeneration;
+    int64_t mRepeatLastFrameTimestamp;
+    int32_t mRepeatLastFrameCount;
 
     int mLatestSubmittedBufferId;
     uint64_t mLatestSubmittedBufferFrameNum;
diff --git a/media/libstagefright/omx/OMX.cpp b/media/libstagefright/omx/OMX.cpp
index 84a0e10..a608479 100644
--- a/media/libstagefright/omx/OMX.cpp
+++ b/media/libstagefright/omx/OMX.cpp
@@ -185,7 +185,7 @@
     instance->onObserverDied(mMaster);
 }
 
-bool OMX::livesLocally(node_id node, pid_t pid) {
+bool OMX::livesLocally(node_id /* node */, pid_t pid) {
     return pid == getpid();
 }
 
@@ -424,7 +424,7 @@
         OMX_IN OMX_EVENTTYPE eEvent,
         OMX_IN OMX_U32 nData1,
         OMX_IN OMX_U32 nData2,
-        OMX_IN OMX_PTR pEventData) {
+        OMX_IN OMX_PTR /* pEventData */) {
     ALOGV("OnEvent(%d, %ld, %ld)", eEvent, nData1, nData2);
 
     // Forward to OMXNodeInstance.
@@ -479,7 +479,7 @@
 OMX::node_id OMX::makeNodeID(OMXNodeInstance *instance) {
     // mLock is already held.
 
-    node_id node = (node_id)++mNodeCounter;
+    node_id node = (node_id)(uintptr_t)++mNodeCounter;
     mNodeIDToInstance.add(node, instance);
 
     return node;
diff --git a/media/libstagefright/omx/OMXNodeInstance.cpp b/media/libstagefright/omx/OMXNodeInstance.cpp
index 5f104fc..aa96389 100644
--- a/media/libstagefright/omx/OMXNodeInstance.cpp
+++ b/media/libstagefright/omx/OMXNodeInstance.cpp
@@ -266,7 +266,7 @@
 }
 
 status_t OMXNodeInstance::getParameter(
-        OMX_INDEXTYPE index, void *params, size_t size) {
+        OMX_INDEXTYPE index, void *params, size_t /* size */) {
     Mutex::Autolock autoLock(mLock);
 
     OMX_ERRORTYPE err = OMX_GetParameter(mHandle, index, params);
@@ -275,7 +275,7 @@
 }
 
 status_t OMXNodeInstance::setParameter(
-        OMX_INDEXTYPE index, const void *params, size_t size) {
+        OMX_INDEXTYPE index, const void *params, size_t /* size */) {
     Mutex::Autolock autoLock(mLock);
 
     OMX_ERRORTYPE err = OMX_SetParameter(
@@ -285,7 +285,7 @@
 }
 
 status_t OMXNodeInstance::getConfig(
-        OMX_INDEXTYPE index, void *params, size_t size) {
+        OMX_INDEXTYPE index, void *params, size_t /* size */) {
     Mutex::Autolock autoLock(mLock);
 
     OMX_ERRORTYPE err = OMX_GetConfig(mHandle, index, params);
@@ -293,7 +293,7 @@
 }
 
 status_t OMXNodeInstance::setConfig(
-        OMX_INDEXTYPE index, const void *params, size_t size) {
+        OMX_INDEXTYPE index, const void *params, size_t /* size */) {
     Mutex::Autolock autoLock(mLock);
 
     OMX_ERRORTYPE err = OMX_SetConfig(
@@ -610,7 +610,7 @@
 }
 
 status_t OMXNodeInstance::updateGraphicBufferInMeta(
-        OMX_U32 portIndex, const sp<GraphicBuffer>& graphicBuffer,
+        OMX_U32 /* portIndex */, const sp<GraphicBuffer>& graphicBuffer,
         OMX::buffer_id buffer) {
     Mutex::Autolock autoLock(mLock);
 
@@ -849,6 +849,8 @@
     switch (type) {
         case IOMX::INTERNAL_OPTION_SUSPEND:
         case IOMX::INTERNAL_OPTION_REPEAT_PREVIOUS_FRAME_DELAY:
+        case IOMX::INTERNAL_OPTION_MAX_TIMESTAMP_GAP:
+        case IOMX::INTERNAL_OPTION_START_TIME:
         {
             const sp<GraphicBufferSource> &bufferSource =
                 getGraphicBufferSource();
@@ -864,7 +866,8 @@
 
                 bool suspend = *(bool *)data;
                 bufferSource->suspend(suspend);
-            } else {
+            } else if (type ==
+                    IOMX::INTERNAL_OPTION_REPEAT_PREVIOUS_FRAME_DELAY){
                 if (size != sizeof(int64_t)) {
                     return INVALID_OPERATION;
                 }
@@ -872,6 +875,23 @@
                 int64_t delayUs = *(int64_t *)data;
 
                 return bufferSource->setRepeatPreviousFrameDelayUs(delayUs);
+            } else if (type ==
+                    IOMX::INTERNAL_OPTION_MAX_TIMESTAMP_GAP){
+                if (size != sizeof(int64_t)) {
+                    return INVALID_OPERATION;
+                }
+
+                int64_t maxGapUs = *(int64_t *)data;
+
+                return bufferSource->setMaxTimestampGapUs(maxGapUs);
+            } else { // IOMX::INTERNAL_OPTION_START_TIME
+                if (size != sizeof(int64_t)) {
+                    return INVALID_OPERATION;
+                }
+
+                int64_t skipFramesBeforeUs = *(int64_t *)data;
+
+                bufferSource->setSkipFramesBeforeUs(skipFramesBeforeUs);
             }
 
             return OK;
@@ -883,6 +903,8 @@
 }
 
 void OMXNodeInstance::onMessage(const omx_message &msg) {
+    const sp<GraphicBufferSource>& bufferSource(getGraphicBufferSource());
+
     if (msg.type == omx_message::FILL_BUFFER_DONE) {
         OMX_BUFFERHEADERTYPE *buffer =
             static_cast<OMX_BUFFERHEADERTYPE *>(
@@ -892,10 +914,18 @@
             static_cast<BufferMeta *>(buffer->pAppPrivate);
 
         buffer_meta->CopyFromOMX(buffer);
-    } else if (msg.type == omx_message::EMPTY_BUFFER_DONE) {
-        const sp<GraphicBufferSource>& bufferSource(getGraphicBufferSource());
 
         if (bufferSource != NULL) {
+            // fix up the buffer info (especially timestamp) if needed
+            bufferSource->codecBufferFilled(buffer);
+
+            omx_message newMsg = msg;
+            newMsg.u.extended_buffer_data.timestamp = buffer->nTimeStamp;
+            mObserver->onMessage(newMsg);
+            return;
+        }
+    } else if (msg.type == omx_message::EMPTY_BUFFER_DONE) {
+        if (bufferSource != NULL) {
             // This is one of the buffers used exclusively by
             // GraphicBufferSource.
             // Don't dispatch a message back to ACodec, since it doesn't
@@ -941,7 +971,7 @@
 
 // static
 OMX_ERRORTYPE OMXNodeInstance::OnEvent(
-        OMX_IN OMX_HANDLETYPE hComponent,
+        OMX_IN OMX_HANDLETYPE /* hComponent */,
         OMX_IN OMX_PTR pAppData,
         OMX_IN OMX_EVENTTYPE eEvent,
         OMX_IN OMX_U32 nData1,
@@ -957,7 +987,7 @@
 
 // static
 OMX_ERRORTYPE OMXNodeInstance::OnEmptyBufferDone(
-        OMX_IN OMX_HANDLETYPE hComponent,
+        OMX_IN OMX_HANDLETYPE /* hComponent */,
         OMX_IN OMX_PTR pAppData,
         OMX_IN OMX_BUFFERHEADERTYPE* pBuffer) {
     OMXNodeInstance *instance = static_cast<OMXNodeInstance *>(pAppData);
@@ -969,7 +999,7 @@
 
 // static
 OMX_ERRORTYPE OMXNodeInstance::OnFillBufferDone(
-        OMX_IN OMX_HANDLETYPE hComponent,
+        OMX_IN OMX_HANDLETYPE /* hComponent */,
         OMX_IN OMX_PTR pAppData,
         OMX_IN OMX_BUFFERHEADERTYPE* pBuffer) {
     OMXNodeInstance *instance = static_cast<OMXNodeInstance *>(pAppData);
diff --git a/media/libstagefright/omx/SoftOMXComponent.cpp b/media/libstagefright/omx/SoftOMXComponent.cpp
index b1c34dc..646cd32 100644
--- a/media/libstagefright/omx/SoftOMXComponent.cpp
+++ b/media/libstagefright/omx/SoftOMXComponent.cpp
@@ -257,69 +257,69 @@
 ////////////////////////////////////////////////////////////////////////////////
 
 OMX_ERRORTYPE SoftOMXComponent::sendCommand(
-        OMX_COMMANDTYPE cmd, OMX_U32 param, OMX_PTR data) {
+        OMX_COMMANDTYPE /* cmd */, OMX_U32 /* param */, OMX_PTR /* data */) {
     return OMX_ErrorUndefined;
 }
 
 OMX_ERRORTYPE SoftOMXComponent::getParameter(
-        OMX_INDEXTYPE index, OMX_PTR params) {
+        OMX_INDEXTYPE /* index */, OMX_PTR /* params */) {
     return OMX_ErrorUndefined;
 }
 
 OMX_ERRORTYPE SoftOMXComponent::setParameter(
-        OMX_INDEXTYPE index, const OMX_PTR params) {
+        OMX_INDEXTYPE /* index */, const OMX_PTR /* params */) {
     return OMX_ErrorUndefined;
 }
 
 OMX_ERRORTYPE SoftOMXComponent::getConfig(
-        OMX_INDEXTYPE index, OMX_PTR params) {
+        OMX_INDEXTYPE /* index */, OMX_PTR /* params */) {
     return OMX_ErrorUndefined;
 }
 
 OMX_ERRORTYPE SoftOMXComponent::setConfig(
-        OMX_INDEXTYPE index, const OMX_PTR params) {
+        OMX_INDEXTYPE /* index */, const OMX_PTR /* params */) {
     return OMX_ErrorUndefined;
 }
 
 OMX_ERRORTYPE SoftOMXComponent::getExtensionIndex(
-        const char *name, OMX_INDEXTYPE *index) {
+        const char * /* name */, OMX_INDEXTYPE * /* index */) {
     return OMX_ErrorUndefined;
 }
 
 OMX_ERRORTYPE SoftOMXComponent::useBuffer(
-        OMX_BUFFERHEADERTYPE **buffer,
-        OMX_U32 portIndex,
-        OMX_PTR appPrivate,
-        OMX_U32 size,
-        OMX_U8 *ptr) {
+        OMX_BUFFERHEADERTYPE ** /* buffer */,
+        OMX_U32 /* portIndex */,
+        OMX_PTR /* appPrivate */,
+        OMX_U32 /* size */,
+        OMX_U8 * /* ptr */) {
     return OMX_ErrorUndefined;
 }
 
 OMX_ERRORTYPE SoftOMXComponent::allocateBuffer(
-        OMX_BUFFERHEADERTYPE **buffer,
-        OMX_U32 portIndex,
-        OMX_PTR appPrivate,
-        OMX_U32 size) {
+        OMX_BUFFERHEADERTYPE ** /* buffer */,
+        OMX_U32 /* portIndex */,
+        OMX_PTR /* appPrivate */,
+        OMX_U32 /* size */) {
     return OMX_ErrorUndefined;
 }
 
 OMX_ERRORTYPE SoftOMXComponent::freeBuffer(
-        OMX_U32 portIndex,
-        OMX_BUFFERHEADERTYPE *buffer) {
+        OMX_U32 /* portIndex */,
+        OMX_BUFFERHEADERTYPE * /* buffer */) {
     return OMX_ErrorUndefined;
 }
 
 OMX_ERRORTYPE SoftOMXComponent::emptyThisBuffer(
-        OMX_BUFFERHEADERTYPE *buffer) {
+        OMX_BUFFERHEADERTYPE * /* buffer */) {
     return OMX_ErrorUndefined;
 }
 
 OMX_ERRORTYPE SoftOMXComponent::fillThisBuffer(
-        OMX_BUFFERHEADERTYPE *buffer) {
+        OMX_BUFFERHEADERTYPE * /* buffer */) {
     return OMX_ErrorUndefined;
 }
 
-OMX_ERRORTYPE SoftOMXComponent::getState(OMX_STATETYPE *state) {
+OMX_ERRORTYPE SoftOMXComponent::getState(OMX_STATETYPE * /* state */) {
     return OMX_ErrorUndefined;
 }
 
diff --git a/media/libstagefright/omx/SoftOMXPlugin.cpp b/media/libstagefright/omx/SoftOMXPlugin.cpp
index d6cde73..d49e50b 100644
--- a/media/libstagefright/omx/SoftOMXPlugin.cpp
+++ b/media/libstagefright/omx/SoftOMXPlugin.cpp
@@ -154,7 +154,7 @@
 
 OMX_ERRORTYPE SoftOMXPlugin::enumerateComponents(
         OMX_STRING name,
-        size_t size,
+        size_t /* size */,
         OMX_U32 index) {
     if (index >= kNumComponents) {
         return OMX_ErrorNoMore;
diff --git a/media/libstagefright/omx/tests/Android.mk b/media/libstagefright/omx/tests/Android.mk
index 1061c39..8b79af4 100644
--- a/media/libstagefright/omx/tests/Android.mk
+++ b/media/libstagefright/omx/tests/Android.mk
@@ -11,6 +11,8 @@
 	$(TOP)/frameworks/av/media/libstagefright \
 	$(TOP)/frameworks/native/include/media/openmax
 
+LOCAL_CFLAGS += -Werror
+
 LOCAL_MODULE := omx_tests
 
 LOCAL_MODULE_TAGS := tests
diff --git a/media/libstagefright/omx/tests/OMXHarness.cpp b/media/libstagefright/omx/tests/OMXHarness.cpp
index 4bee808..03725df 100644
--- a/media/libstagefright/omx/tests/OMXHarness.cpp
+++ b/media/libstagefright/omx/tests/OMXHarness.cpp
@@ -16,6 +16,7 @@
 
 //#define LOG_NDEBUG 0
 #define LOG_TAG "OMXHarness"
+#include <inttypes.h>
 #include <utils/Log.h>
 
 #include "OMXHarness.h"
@@ -25,6 +26,7 @@
 #include <binder/ProcessState.h>
 #include <binder/IServiceManager.h>
 #include <binder/MemoryDealer.h>
+#include <media/IMediaHTTPService.h>
 #include <media/IMediaPlayerService.h>
 #include <media/stagefright/foundation/ADebug.h>
 #include <media/stagefright/foundation/ALooper.h>
@@ -241,7 +243,8 @@
 };
 
 static sp<MediaExtractor> CreateExtractorFromURI(const char *uri) {
-    sp<DataSource> source = DataSource::CreateFromURI(uri);
+    sp<DataSource> source =
+        DataSource::CreateFromURI(NULL /* httpService */, uri);
 
     if (source == NULL) {
         return NULL;
@@ -711,11 +714,11 @@
             int64_t bufferTimeUs;
             CHECK(buffer->meta_data()->findInt64(kKeyTime, &bufferTimeUs));
             if (!CloseEnough(bufferTimeUs, actualSeekTimeUs)) {
-                printf("\n  * Attempted seeking to %lld us (%.2f secs)",
+                printf("\n  * Attempted seeking to %" PRId64 " us (%.2f secs)",
                        requestedSeekTimeUs, requestedSeekTimeUs / 1E6);
-                printf("\n  * Nearest keyframe is at %lld us (%.2f secs)",
+                printf("\n  * Nearest keyframe is at %" PRId64 " us (%.2f secs)",
                        actualSeekTimeUs, actualSeekTimeUs / 1E6);
-                printf("\n  * Returned buffer was at %lld us (%.2f secs)\n\n",
+                printf("\n  * Returned buffer was at %" PRId64 " us (%.2f secs)\n\n",
                        bufferTimeUs, bufferTimeUs / 1E6);
 
                 buffer->release();
diff --git a/media/libstagefright/rtsp/AMPEG2TSAssembler.cpp b/media/libstagefright/rtsp/AMPEG2TSAssembler.cpp
index 4c9bf5b..dca5c89 100644
--- a/media/libstagefright/rtsp/AMPEG2TSAssembler.cpp
+++ b/media/libstagefright/rtsp/AMPEG2TSAssembler.cpp
@@ -34,7 +34,9 @@
 namespace android {
 
 AMPEG2TSAssembler::AMPEG2TSAssembler(
-        const sp<AMessage> &notify, const char *desc, const AString &params)
+        const sp<AMessage> &notify,
+        const char * /* desc */,
+        const AString & /* params */)
     : mNotifyMsg(notify),
       mNextExpectedSeqNoValid(false),
       mNextExpectedSeqNo(0) {
diff --git a/media/libstagefright/rtsp/APacketSource.cpp b/media/libstagefright/rtsp/APacketSource.cpp
index 462c384..09f52bc 100644
--- a/media/libstagefright/rtsp/APacketSource.cpp
+++ b/media/libstagefright/rtsp/APacketSource.cpp
@@ -23,7 +23,7 @@
 #include "ARawAudioAssembler.h"
 #include "ASessionDescription.h"
 
-#include "avc_utils.h"
+#include "include/avc_utils.h"
 
 #include <ctype.h>
 
diff --git a/media/libstagefright/rtsp/ARTPWriter.cpp b/media/libstagefright/rtsp/ARTPWriter.cpp
index 0d07043..c46d16f 100644
--- a/media/libstagefright/rtsp/ARTPWriter.cpp
+++ b/media/libstagefright/rtsp/ARTPWriter.cpp
@@ -114,7 +114,7 @@
     return (mFlags & kFlagEOS) != 0;
 }
 
-status_t ARTPWriter::start(MetaData *params) {
+status_t ARTPWriter::start(MetaData * /* params */) {
     Mutex::Autolock autoLock(mLock);
     if (mFlags & kFlagStarted) {
         return INVALID_OPERATION;
diff --git a/media/libstagefright/rtsp/ARTSPConnection.cpp b/media/libstagefright/rtsp/ARTSPConnection.cpp
index efde7a9..4054da6 100644
--- a/media/libstagefright/rtsp/ARTSPConnection.cpp
+++ b/media/libstagefright/rtsp/ARTSPConnection.cpp
@@ -33,7 +33,7 @@
 #include <openssl/md5.h>
 #include <sys/socket.h>
 
-#include "HTTPBase.h"
+#include "include/HTTPBase.h"
 
 namespace android {
 
diff --git a/media/libstagefright/rtsp/ARawAudioAssembler.cpp b/media/libstagefright/rtsp/ARawAudioAssembler.cpp
index 0da5dd2..167f7a4 100644
--- a/media/libstagefright/rtsp/ARawAudioAssembler.cpp
+++ b/media/libstagefright/rtsp/ARawAudioAssembler.cpp
@@ -34,7 +34,9 @@
 namespace android {
 
 ARawAudioAssembler::ARawAudioAssembler(
-        const sp<AMessage> &notify, const char *desc, const AString &params)
+        const sp<AMessage> &notify,
+        const char * /* desc */,
+        const AString & /* params */)
     : mNotifyMsg(notify),
       mNextExpectedSeqNoValid(false),
       mNextExpectedSeqNo(0) {
diff --git a/media/libstagefright/rtsp/Android.mk b/media/libstagefright/rtsp/Android.mk
index e77c69c..39eedc0 100644
--- a/media/libstagefright/rtsp/Android.mk
+++ b/media/libstagefright/rtsp/Android.mk
@@ -20,7 +20,7 @@
         SDPLoader.cpp               \
 
 LOCAL_C_INCLUDES:= \
-	$(TOP)/frameworks/av/media/libstagefright/include \
+	$(TOP)/frameworks/av/media/libstagefright \
 	$(TOP)/frameworks/native/include/media/openmax \
 	$(TOP)/external/openssl/include
 
@@ -30,6 +30,8 @@
     LOCAL_CFLAGS += -Wno-psabi
 endif
 
+LOCAL_CFLAGS += -Werror
+
 include $(BUILD_STATIC_LIBRARY)
 
 ################################################################################
diff --git a/media/libstagefright/rtsp/MyHandler.h b/media/libstagefright/rtsp/MyHandler.h
index cd77aa0..45470a3 100644
--- a/media/libstagefright/rtsp/MyHandler.h
+++ b/media/libstagefright/rtsp/MyHandler.h
@@ -19,7 +19,11 @@
 #define MY_HANDLER_H_
 
 //#define LOG_NDEBUG 0
+
+#ifndef LOG_TAG
 #define LOG_TAG "MyHandler"
+#endif
+
 #include <utils/Log.h>
 
 #include "APacketSource.h"
@@ -42,6 +46,12 @@
 
 #include "HTTPBase.h"
 
+#if LOG_NDEBUG
+#define UNUSED_UNLESS_VERBOSE(x) (void)(x)
+#else
+#define UNUSED_UNLESS_VERBOSE(x)
+#endif
+
 // If no access units are received within 5 secs, assume that the rtp
 // stream has ended and signal end of stream.
 static int64_t kAccessUnitTimeoutUs = 10000000ll;
@@ -178,7 +188,7 @@
         mConn->connect(mOriginalSessionURL.c_str(), reply);
     }
 
-    AString getControlURL(sp<ASessionDescription> desc) {
+    AString getControlURL() {
         AString sessionLevelControlURL;
         if (mSessionDesc->findAttribute(
                 0,
@@ -545,7 +555,7 @@
                                 mBaseURL = tmp;
                             }
 
-                            mControlURL = getControlURL(mSessionDesc);
+                            mControlURL = getControlURL();
 
                             if (mSessionDesc->countTracks() < 2) {
                                 // There's no actual tracks in this session.
@@ -591,7 +601,7 @@
 
                         mSeekable = !isLiveStream(mSessionDesc);
 
-                        mControlURL = getControlURL(mSessionDesc);
+                        mControlURL = getControlURL();
 
                         if (mSessionDesc->countTracks() < 2) {
                             // There's no actual tracks in this session.
@@ -1805,6 +1815,8 @@
     bool addMediaTimestamp(
             int32_t trackIndex, const TrackInfo *track,
             const sp<ABuffer> &accessUnit) {
+        UNUSED_UNLESS_VERBOSE(trackIndex);
+
         uint32_t rtpTime;
         CHECK(accessUnit->meta()->findInt32(
                     "rtp-time", (int32_t *)&rtpTime));
diff --git a/media/libstagefright/rtsp/SDPLoader.cpp b/media/libstagefright/rtsp/SDPLoader.cpp
index ed3fa7e..ce1e89d 100644
--- a/media/libstagefright/rtsp/SDPLoader.cpp
+++ b/media/libstagefright/rtsp/SDPLoader.cpp
@@ -18,11 +18,13 @@
 #define LOG_TAG "SDPLoader"
 #include <utils/Log.h>
 
-#include "SDPLoader.h"
+#include "include/SDPLoader.h"
 
 #include "ASessionDescription.h"
-#include "HTTPBase.h"
 
+#include <media/IMediaHTTPConnection.h>
+#include <media/IMediaHTTPService.h>
+#include <media/stagefright/MediaHTTP.h>
 #include <media/stagefright/foundation/ABuffer.h>
 #include <media/stagefright/foundation/ADebug.h>
 
@@ -30,22 +32,15 @@
 
 namespace android {
 
-SDPLoader::SDPLoader(const sp<AMessage> &notify, uint32_t flags, bool uidValid, uid_t uid)
+SDPLoader::SDPLoader(
+        const sp<AMessage> &notify,
+        uint32_t flags,
+        const sp<IMediaHTTPService> &httpService)
     : mNotify(notify),
       mFlags(flags),
-      mUIDValid(uidValid),
-      mUID(uid),
       mNetLooper(new ALooper),
       mCancelled(false),
-      mHTTPDataSource(
-              HTTPBase::Create(
-                  (mFlags & kFlagIncognito)
-                    ? HTTPBase::kFlagIncognito
-                    : 0)) {
-    if (mUIDValid) {
-        mHTTPDataSource->setUID(mUID);
-    }
-
+      mHTTPDataSource(new MediaHTTP(httpService->makeHTTPConnection())) {
     mNetLooper->setName("sdp net");
     mNetLooper->start(false /* runOnCallingThread */,
                       false /* canCallJava */,
@@ -130,7 +125,7 @@
         ssize_t readSize = mHTTPDataSource->readAt(0, buffer->data(), sdpSize);
 
         if (readSize < 0) {
-            ALOGE("Failed to read SDP, error code = %ld", readSize);
+            ALOGE("Failed to read SDP, error code = %d", readSize);
             err = UNKNOWN_ERROR;
         } else {
             desc = new ASessionDescription;
diff --git a/media/libstagefright/tests/DummyRecorder.cpp b/media/libstagefright/tests/DummyRecorder.cpp
index ac37b28..8f17088 100644
--- a/media/libstagefright/tests/DummyRecorder.cpp
+++ b/media/libstagefright/tests/DummyRecorder.cpp
@@ -61,7 +61,7 @@
     mSource->stop();
     void *dummy;
     pthread_join(mThread, &dummy);
-    status_t err = (status_t) dummy;
+    status_t err = static_cast<status_t>(reinterpret_cast<uintptr_t>(dummy));
 
     ALOGV("Ending the reading thread");
     return err;
diff --git a/media/libstagefright/tests/SurfaceMediaSource_test.cpp b/media/libstagefright/tests/SurfaceMediaSource_test.cpp
index 49ffcd6..aeecdbc 100644
--- a/media/libstagefright/tests/SurfaceMediaSource_test.cpp
+++ b/media/libstagefright/tests/SurfaceMediaSource_test.cpp
@@ -527,7 +527,8 @@
 }
 
 // Dequeuing and queuing the buffer without really filling it in.
-void SurfaceMediaSourceTest::oneBufferPassNoFill(int width, int height ) {
+void SurfaceMediaSourceTest::oneBufferPassNoFill(
+        int /* width */, int /* height  */) {
     ANativeWindowBuffer* anb;
     ASSERT_EQ(NO_ERROR, native_window_dequeue_buffer_and_wait(mANW.get(), &anb));
     ASSERT_TRUE(anb != NULL);
@@ -746,9 +747,8 @@
     CHECK(fd >= 0);
 
     sp<MediaRecorder> mr = SurfaceMediaSourceGLTest::setUpMediaRecorder(fd,
-            VIDEO_SOURCE_GRALLOC_BUFFER,
-            OUTPUT_FORMAT_MPEG_4, VIDEO_ENCODER_H264, mYuvTexWidth,
-            mYuvTexHeight, 30);
+            VIDEO_SOURCE_SURFACE, OUTPUT_FORMAT_MPEG_4, VIDEO_ENCODER_H264,
+            mYuvTexWidth, mYuvTexHeight, 30);
     // get the reference to the surfacemediasource living in
     // mediaserver that is created by stagefrightrecorder
     sp<IGraphicBufferProducer> iST = mr->querySurfaceMediaSourceFromMediaServer();
@@ -880,7 +880,7 @@
     }
     CHECK(fd >= 0);
 
-    sp<MediaRecorder> mr = setUpMediaRecorder(fd, VIDEO_SOURCE_GRALLOC_BUFFER,
+    sp<MediaRecorder> mr = setUpMediaRecorder(fd, VIDEO_SOURCE_SURFACE,
             OUTPUT_FORMAT_MPEG_4, VIDEO_ENCODER_H264, mYuvTexWidth, mYuvTexHeight, 30);
 
     // get the reference to the surfacemediasource living in
@@ -923,7 +923,7 @@
     }
     CHECK(fd >= 0);
 
-    sp<MediaRecorder> mr = setUpMediaRecorder(fd, VIDEO_SOURCE_GRALLOC_BUFFER,
+    sp<MediaRecorder> mr = setUpMediaRecorder(fd, VIDEO_SOURCE_SURFACE,
             OUTPUT_FORMAT_MPEG_4, VIDEO_ENCODER_H264, mYuvTexWidth, mYuvTexHeight, 30);
 
     // get the reference to the surfacemediasource living in
diff --git a/media/libstagefright/timedtext/Android.mk b/media/libstagefright/timedtext/Android.mk
index f099bbd..6a8b9fc 100644
--- a/media/libstagefright/timedtext/Android.mk
+++ b/media/libstagefright/timedtext/Android.mk
@@ -9,7 +9,8 @@
         TimedTextSRTSource.cpp    \
         TimedTextPlayer.cpp
 
-LOCAL_CFLAGS += -Wno-multichar
+LOCAL_CFLAGS += -Wno-multichar -Werror
+
 LOCAL_C_INCLUDES:= \
         $(TOP)/frameworks/av/include/media/stagefright/timedtext \
         $(TOP)/frameworks/av/media/libstagefright
diff --git a/media/libstagefright/timedtext/TimedTextDriver.cpp b/media/libstagefright/timedtext/TimedTextDriver.cpp
index 12fd7f4..71aa21e 100644
--- a/media/libstagefright/timedtext/TimedTextDriver.cpp
+++ b/media/libstagefright/timedtext/TimedTextDriver.cpp
@@ -20,6 +20,7 @@
 
 #include <binder/IPCThreadState.h>
 
+#include <media/IMediaHTTPService.h>
 #include <media/mediaplayer.h>
 #include <media/MediaPlayerInterface.h>
 #include <media/stagefright/DataSource.h>
@@ -40,9 +41,11 @@
 namespace android {
 
 TimedTextDriver::TimedTextDriver(
-        const wp<MediaPlayerBase> &listener)
+        const wp<MediaPlayerBase> &listener,
+        const sp<IMediaHTTPService> &httpService)
     : mLooper(new ALooper),
       mListener(listener),
+      mHTTPService(httpService),
       mState(UNINITIALIZED),
       mCurrentTrackIndex(UINT_MAX) {
     mLooper->setName("TimedTextDriver");
@@ -207,7 +210,7 @@
     }
 
     sp<DataSource> dataSource =
-            DataSource::CreateFromURI(uri);
+            DataSource::CreateFromURI(mHTTPService, uri);
     return createOutOfBandTextSource(trackIndex, mimeType, dataSource);
 }
 
diff --git a/media/libstagefright/timedtext/TimedTextSource.h b/media/libstagefright/timedtext/TimedTextSource.h
index 756cc31..8c1c1cd 100644
--- a/media/libstagefright/timedtext/TimedTextSource.h
+++ b/media/libstagefright/timedtext/TimedTextSource.h
@@ -47,7 +47,7 @@
           int64_t *endTimeUs,
           Parcel *parcel,
           const MediaSource::ReadOptions *options = NULL) = 0;
-  virtual status_t extractGlobalDescriptions(Parcel *parcel) {
+  virtual status_t extractGlobalDescriptions(Parcel * /* parcel */) {
       return INVALID_OPERATION;
   }
   virtual sp<MetaData> getFormat();
diff --git a/media/libstagefright/wifi-display/rtp/RTPSender.cpp b/media/libstagefright/wifi-display/rtp/RTPSender.cpp
index 1887b8b..e88a3bd 100644
--- a/media/libstagefright/wifi-display/rtp/RTPSender.cpp
+++ b/media/libstagefright/wifi-display/rtp/RTPSender.cpp
@@ -685,9 +685,8 @@
     return OK;
 }
 
-status_t RTPSender::parseReceiverReport(const uint8_t *data, size_t size) {
-    // hexdump(data, size);
-
+status_t RTPSender::parseReceiverReport(
+        const uint8_t *data, size_t /* size */) {
     float fractionLost = data[12] / 256.0f;
 
     ALOGI("lost %.2f %% of packets during report interval.",
diff --git a/media/libstagefright/wifi-display/source/PlaybackSession.cpp b/media/libstagefright/wifi-display/source/PlaybackSession.cpp
index 286ea13..1a5acba 100644
--- a/media/libstagefright/wifi-display/source/PlaybackSession.cpp
+++ b/media/libstagefright/wifi-display/source/PlaybackSession.cpp
@@ -29,6 +29,7 @@
 #include <binder/IServiceManager.h>
 #include <cutils/properties.h>
 #include <media/IHDCP.h>
+#include <media/IMediaHTTPService.h>
 #include <media/stagefright/foundation/ABitReader.h>
 #include <media/stagefright/foundation/ABuffer.h>
 #include <media/stagefright/foundation/ADebug.h>
@@ -749,7 +750,8 @@
 
     mExtractor = new NuMediaExtractor;
 
-    status_t err = mExtractor->setDataSource(mMediaPath.c_str());
+    status_t err = mExtractor->setDataSource(
+            NULL /* httpService */, mMediaPath.c_str());
 
     if (err != OK) {
         return err;
diff --git a/media/libstagefright/wifi-display/source/RepeaterSource.cpp b/media/libstagefright/wifi-display/source/RepeaterSource.cpp
index cc8dee3..59d7e6e 100644
--- a/media/libstagefright/wifi-display/source/RepeaterSource.cpp
+++ b/media/libstagefright/wifi-display/source/RepeaterSource.cpp
@@ -79,6 +79,8 @@
 
     ALOGV("stopping");
 
+    status_t err = mSource->stop();
+
     if (mLooper != NULL) {
         mLooper->stop();
         mLooper.clear();
@@ -92,7 +94,6 @@
         mBuffer = NULL;
     }
 
-    status_t err = mSource->stop();
 
     ALOGV("stopped");
 
diff --git a/media/libstagefright/wifi-display/source/TSPacketizer.cpp b/media/libstagefright/wifi-display/source/TSPacketizer.cpp
index c674700..50d317a 100644
--- a/media/libstagefright/wifi-display/source/TSPacketizer.cpp
+++ b/media/libstagefright/wifi-display/source/TSPacketizer.cpp
@@ -216,7 +216,7 @@
     uint8_t *ptr = dup->data();
 
     *ptr++ = 0xff;
-    *ptr++ = 0xf1;  // b11110001, ID=0, layer=0, protection_absent=1
+    *ptr++ = 0xf9;  // b11111001, ID=1(MPEG-2), layer=0, protection_absent=1
 
     *ptr++ =
         profile << 6
@@ -565,7 +565,7 @@
             }
         }
 
-        // size_t numPaddingBytes = sizeAvailableForPayload - numBytesOfPayload;
+        size_t numPaddingBytes = sizeAvailableForPayload - numBytesOfPayload;
         ALOGV("packet 1 contains %zd padding bytes and %zd bytes of payload",
               numPaddingBytes, numBytesOfPayload);
 
diff --git a/media/libstagefright/wifi-display/source/WifiDisplaySource.cpp b/media/libstagefright/wifi-display/source/WifiDisplaySource.cpp
index 05e4018..da405e2 100644
--- a/media/libstagefright/wifi-display/source/WifiDisplaySource.cpp
+++ b/media/libstagefright/wifi-display/source/WifiDisplaySource.cpp
@@ -746,7 +746,7 @@
 }
 
 status_t WifiDisplaySource::onReceiveM1Response(
-        int32_t sessionID, const sp<ParsedMessage> &msg) {
+        int32_t /* sessionID */, const sp<ParsedMessage> &msg) {
     int32_t statusCode;
     if (!msg->getStatusCode(&statusCode)) {
         return ERROR_MALFORMED;
@@ -991,7 +991,7 @@
 }
 
 status_t WifiDisplaySource::onReceiveM5Response(
-        int32_t sessionID, const sp<ParsedMessage> &msg) {
+        int32_t /* sessionID */, const sp<ParsedMessage> &msg) {
     int32_t statusCode;
     if (!msg->getStatusCode(&statusCode)) {
         return ERROR_MALFORMED;
@@ -1005,7 +1005,7 @@
 }
 
 status_t WifiDisplaySource::onReceiveM16Response(
-        int32_t sessionID, const sp<ParsedMessage> &msg) {
+        int32_t sessionID, const sp<ParsedMessage> & /* msg */) {
     // If only the response was required to include a "Session:" header...
 
     CHECK_EQ(sessionID, mClientSessionID);
@@ -1680,7 +1680,7 @@
 }
 
 void WifiDisplaySource::HDCPObserver::notify(
-        int msg, int ext1, int ext2, const Parcel *obj) {
+        int msg, int ext1, int ext2, const Parcel * /* obj */) {
     sp<AMessage> notify = mNotify->dup();
     notify->setInt32("msg", msg);
     notify->setInt32("ext1", ext1);
diff --git a/media/libstagefright/yuv/Android.mk b/media/libstagefright/yuv/Android.mk
index b3f7b1b..bb86dfc 100644
--- a/media/libstagefright/yuv/Android.mk
+++ b/media/libstagefright/yuv/Android.mk
@@ -12,5 +12,7 @@
 LOCAL_MODULE:= libstagefright_yuv
 
 
+LOCAL_CFLAGS += -Werror
+
 
 include $(BUILD_SHARED_LIBRARY)
diff --git a/media/libstagefright/yuv/YUVImage.cpp b/media/libstagefright/yuv/YUVImage.cpp
index 7b9000b..bb3e2fd 100644
--- a/media/libstagefright/yuv/YUVImage.cpp
+++ b/media/libstagefright/yuv/YUVImage.cpp
@@ -226,8 +226,8 @@
             &ySrcOffsetIncrement, &uSrcOffsetIncrement, &vSrcOffsetIncrement);
 
     int32_t yDestOffsetIncrement;
-    int32_t uDestOffsetIncrement;
-    int32_t vDestOffsetIncrement;
+    int32_t uDestOffsetIncrement = 0;
+    int32_t vDestOffsetIncrement = 0;
     destImage.getOffsetIncrementsPerDataRow(
             &yDestOffsetIncrement, &uDestOffsetIncrement, &vDestOffsetIncrement);
 
@@ -309,7 +309,7 @@
 
     int32_t yDestOffsetIncrement;
     int32_t uDestOffsetIncrement;
-    int32_t vDestOffsetIncrement;
+    int32_t vDestOffsetIncrement = 0;
     destImage.getOffsetIncrementsPerDataRow(
             &yDestOffsetIncrement, &uDestOffsetIncrement, &vDestOffsetIncrement);
 
@@ -393,9 +393,9 @@
     fprintf(fp, "255\n");
     for (int32_t y = 0; y < mHeight; ++y) {
         for (int32_t x = 0; x < mWidth; ++x) {
-            uint8_t yValue;
-            uint8_t uValue;
-            uint8_t vValue;
+            uint8_t yValue = 0u;
+            uint8_t uValue = 0u;
+            uint8_t vValue = 0u;
             getPixelValue(x, y, &yValue, &uValue, & vValue);
 
             uint8_t rValue;
diff --git a/media/mediaserver/Android.mk b/media/mediaserver/Android.mk
index 1ac647a..f848054 100644
--- a/media/mediaserver/Android.mk
+++ b/media/mediaserver/Android.mk
@@ -15,6 +15,7 @@
 
 LOCAL_SHARED_LIBRARIES := \
 	libaudioflinger \
+	libcamera_metadata\
 	libcameraservice \
 	libmedialogservice \
 	libcutils \
diff --git a/media/mediaserver/main_mediaserver.cpp b/media/mediaserver/main_mediaserver.cpp
index d5207d5..a347951 100644
--- a/media/mediaserver/main_mediaserver.cpp
+++ b/media/mediaserver/main_mediaserver.cpp
@@ -37,7 +37,7 @@
 
 using namespace android;
 
-int main(int argc, char** argv)
+int main(int argc __unused, char** argv)
 {
     signal(SIGPIPE, SIG_IGN);
     char value[PROPERTY_VALUE_MAX];
diff --git a/media/mtp/MtpDataPacket.cpp b/media/mtp/MtpDataPacket.cpp
index 930f0b0..c4f87a0 100644
--- a/media/mtp/MtpDataPacket.cpp
+++ b/media/mtp/MtpDataPacket.cpp
@@ -331,7 +331,7 @@
 
 void MtpDataPacket::putString(const uint16_t* string) {
     int count = 0;
-    for (int i = 0; i < 256; i++) {
+    for (int i = 0; i <= MTP_STRING_MAX_CHARACTER_NUMBER; i++) {
         if (string[i])
             count++;
         else
diff --git a/media/mtp/MtpProperty.cpp b/media/mtp/MtpProperty.cpp
index 64dd45b..375ed9a 100644
--- a/media/mtp/MtpProperty.cpp
+++ b/media/mtp/MtpProperty.cpp
@@ -16,6 +16,7 @@
 
 #define LOG_TAG "MtpProperty"
 
+#include <inttypes.h>
 #include "MtpDataPacket.h"
 #include "MtpDebug.h"
 #include "MtpProperty.h"
@@ -385,10 +386,10 @@
             buffer.appendFormat("%d", value.u.u32);
             break;
         case MTP_TYPE_INT64:
-            buffer.appendFormat("%lld", value.u.i64);
+            buffer.appendFormat("%" PRId64, value.u.i64);
             break;
         case MTP_TYPE_UINT64:
-            buffer.appendFormat("%lld", value.u.u64);
+            buffer.appendFormat("%" PRIu64, value.u.u64);
             break;
         case MTP_TYPE_INT128:
             buffer.appendFormat("%08X%08X%08X%08X", value.u.i128[0], value.u.i128[1],
diff --git a/media/mtp/MtpStringBuffer.cpp b/media/mtp/MtpStringBuffer.cpp
index fe8cf04..f3420a4 100644
--- a/media/mtp/MtpStringBuffer.cpp
+++ b/media/mtp/MtpStringBuffer.cpp
@@ -56,42 +56,47 @@
 }
 
 void MtpStringBuffer::set(const char* src) {
-    int length = strlen(src);
-    if (length >= sizeof(mBuffer))
-        length = sizeof(mBuffer) - 1;
-    memcpy(mBuffer, src, length);
-
     // count the characters
     int count = 0;
     char ch;
-    while ((ch = *src++) != 0) {
+    char* dest = (char*)mBuffer;
+
+    while ((ch = *src++) != 0 && count < MTP_STRING_MAX_CHARACTER_NUMBER) {
         if ((ch & 0x80) == 0) {
             // single byte character
+            *dest++ = ch;
         } else if ((ch & 0xE0) == 0xC0) {
             // two byte character
-            if (! *src++) {
+            char ch1 = *src++;
+            if (! ch1) {
                 // last character was truncated, so ignore last byte
-                length--;
                 break;
             }
+
+            *dest++ = ch;
+            *dest++ = ch1;
         } else if ((ch & 0xF0) == 0xE0) {
             // 3 byte char
-            if (! *src++) {
+            char ch1 = *src++;
+            if (! ch1) {
                 // last character was truncated, so ignore last byte
-                length--;
                 break;
             }
-            if (! *src++) {
-                // last character was truncated, so ignore last two bytes
-                length -= 2;
+            char ch2 = *src++;
+            if (! ch2) {
+                // last character was truncated, so ignore last byte
                 break;
             }
+
+            *dest++ = ch;
+            *dest++ = ch1;
+            *dest++ = ch2;
         }
         count++;
     }
 
-    mByteCount = length + 1;
-    mBuffer[length] = 0;
+    *dest++ = 0;
+    mByteCount = dest - (char*)mBuffer;
     mCharCount = count;
 }
 
@@ -100,7 +105,7 @@
     uint16_t ch;
     uint8_t* dest = mBuffer;
 
-    while ((ch = *src++) != 0 && count < 255) {
+    while ((ch = *src++) != 0 && count < MTP_STRING_MAX_CHARACTER_NUMBER) {
         if (ch >= 0x0800) {
             *dest++ = (uint8_t)(0xE0 | (ch >> 12));
             *dest++ = (uint8_t)(0x80 | ((ch >> 6) & 0x3F));
diff --git a/media/mtp/MtpStringBuffer.h b/media/mtp/MtpStringBuffer.h
index cbc8307..e5150df 100644
--- a/media/mtp/MtpStringBuffer.h
+++ b/media/mtp/MtpStringBuffer.h
@@ -19,6 +19,9 @@
 
 #include <stdint.h>
 
+// Max Character number of a MTP String
+#define MTP_STRING_MAX_CHARACTER_NUMBER             255
+
 namespace android {
 
 class MtpDataPacket;
@@ -29,7 +32,7 @@
 private:
     // mBuffer contains string in UTF8 format
     // maximum 3 bytes/character, with 1 extra for zero termination
-    uint8_t         mBuffer[255 * 3 + 1];
+    uint8_t         mBuffer[MTP_STRING_MAX_CHARACTER_NUMBER * 3 + 1];
     int             mCharCount;
     int             mByteCount;
 
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 443051c..b74fa89 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -104,6 +104,27 @@
 
 // ----------------------------------------------------------------------------
 
+const char *formatToString(audio_format_t format) {
+    switch(format) {
+    case AUDIO_FORMAT_PCM_SUB_8_BIT: return "pcm8";
+    case AUDIO_FORMAT_PCM_SUB_16_BIT: return "pcm16";
+    case AUDIO_FORMAT_PCM_SUB_32_BIT: return "pcm32";
+    case AUDIO_FORMAT_PCM_SUB_8_24_BIT: return "pcm8.24";
+    case AUDIO_FORMAT_PCM_SUB_24_BIT_PACKED: return "pcm24";
+    case AUDIO_FORMAT_PCM_SUB_FLOAT: return "pcmfloat";
+    case AUDIO_FORMAT_MP3: return "mp3";
+    case AUDIO_FORMAT_AMR_NB: return "amr-nb";
+    case AUDIO_FORMAT_AMR_WB: return "amr-wb";
+    case AUDIO_FORMAT_AAC: return "aac";
+    case AUDIO_FORMAT_HE_AAC_V1: return "he-aac-v1";
+    case AUDIO_FORMAT_HE_AAC_V2: return "he-aac-v2";
+    case AUDIO_FORMAT_VORBIS: return "vorbis";
+    default:
+        break;
+    }
+    return "unknown";
+}
+
 static int load_audio_interface(const char *if_name, audio_hw_device_t **dev)
 {
     const hw_module_t *mod;
@@ -264,7 +285,7 @@
     return NULL;
 }
 
-void AudioFlinger::dumpClients(int fd, const Vector<String16>& args)
+void AudioFlinger::dumpClients(int fd, const Vector<String16>& args __unused)
 {
     const size_t SIZE = 256;
     char buffer[SIZE];
@@ -286,17 +307,17 @@
     }
 
     result.append("Global session refs:\n");
-    result.append(" session pid count\n");
+    result.append("  session   pid count\n");
     for (size_t i = 0; i < mAudioSessionRefs.size(); i++) {
         AudioSessionRef *r = mAudioSessionRefs[i];
-        snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt);
+        snprintf(buffer, SIZE, "  %7d %5d %5d\n", r->mSessionid, r->mPid, r->mCnt);
         result.append(buffer);
     }
     write(fd, result.string(), result.size());
 }
 
 
-void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args)
+void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args __unused)
 {
     const size_t SIZE = 256;
     char buffer[SIZE];
@@ -311,7 +332,7 @@
     write(fd, result.string(), result.size());
 }
 
-void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args)
+void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args __unused)
 {
     const size_t SIZE = 256;
     char buffer[SIZE];
@@ -481,7 +502,7 @@
         uint32_t sampleRate,
         audio_format_t format,
         audio_channel_mask_t channelMask,
-        size_t frameCount,
+        size_t *frameCount,
         IAudioFlinger::track_flags_t *flags,
         const sp<IMemory>& sharedBuffer,
         audio_io_handle_t output,
@@ -508,7 +529,7 @@
     // client is responsible for conversion of 8-bit PCM to 16-bit PCM,
     // and we don't yet support 8.24 or 32-bit PCM
     if (audio_is_linear_pcm(format) && format != AUDIO_FORMAT_PCM_16_BIT) {
-        ALOGE("createTrack() invalid format %d", format);
+        ALOGE("createTrack() invalid format %#x", format);
         lStatus = BAD_VALUE;
         goto Exit;
     }
@@ -534,7 +555,7 @@
         client = registerPid_l(pid);
 
         ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId);
-        if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
+        if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) {
             // check if an effect chain with the same session ID is present on another
             // output thread and move it here.
             for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
@@ -559,6 +580,7 @@
 
         track = thread->createTrack_l(client, streamType, sampleRate, format,
                 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus);
+        LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0));
         // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless
 
         // move effect chain to this output thread if an effect on same session was waiting
@@ -1058,7 +1080,7 @@
     return size;
 }
 
-unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const
+uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const
 {
     Mutex::Autolock _l(mLock);
 
@@ -1090,7 +1112,7 @@
     return ret;
 }
 
-status_t AudioFlinger::getRenderPosition(size_t *halFrames, size_t *dspFrames,
+status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
         audio_io_handle_t output) const
 {
     status_t status;
@@ -1258,7 +1280,7 @@
 {
 }
 
-void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who)
+void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused)
 {
     sp<NotificationClient> keep(this);
     mAudioFlinger->removeNotificationClient(mPid);
@@ -1276,7 +1298,7 @@
         uint32_t sampleRate,
         audio_format_t format,
         audio_channel_mask_t channelMask,
-        size_t frameCount,
+        size_t *frameCount,
         IAudioFlinger::track_flags_t *flags,
         pid_t tid,
         int *sessionId,
@@ -1298,7 +1320,7 @@
     }
 
     if (format != AUDIO_FORMAT_PCM_16_BIT) {
-        ALOGE("openRecord() invalid format %d", format);
+        ALOGE("openRecord() invalid format %#x", format);
         lStatus = BAD_VALUE;
         goto Exit;
     }
@@ -1324,7 +1346,7 @@
         client = registerPid_l(pid);
 
         // If no audio session id is provided, create one here
-        if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
+        if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) {
             lSessionId = *sessionId;
         } else {
             lSessionId = nextUniqueId();
@@ -1339,7 +1361,7 @@
                                                   frameCount, lSessionId,
                                                   IPCThreadState::self()->getCallingUid(),
                                                   flags, tid, &lStatus);
-        LOG_ALWAYS_FATAL_IF((recordTrack != 0) != (lStatus == NO_ERROR));
+        LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0));
     }
 
     if (lStatus != NO_ERROR) {
@@ -1745,7 +1767,7 @@
     audio_stream_in_t *inStream = NULL;
     status_t status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config,
                                         &inStream);
-    ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, "
+    ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %#x, Channels %x, "
             "status %d",
             inStream,
             config.sample_rate,
@@ -1782,7 +1804,7 @@
                                         popcount(inStream->common.get_channels(&inStream->common)));
         if (!mTeeSinkInputEnabled) {
             kind = TEE_SINK_NO;
-        } else if (format == Format_Invalid) {
+        } else if (!Format_isValid(format)) {
             kind = TEE_SINK_NO;
         } else if (mRecordTeeSink == 0) {
             kind = TEE_SINK_NEW;
@@ -1907,18 +1929,21 @@
     return nextUniqueId();
 }
 
-void AudioFlinger::acquireAudioSessionId(int audioSession)
+void AudioFlinger::acquireAudioSessionId(int audioSession, pid_t pid)
 {
     Mutex::Autolock _l(mLock);
     pid_t caller = IPCThreadState::self()->getCallingPid();
-    ALOGV("acquiring %d from %d", audioSession, caller);
+    ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid);
+    if (pid != -1 && (caller == getpid_cached)) {
+        caller = pid;
+    }
 
     // Ignore requests received from processes not known as notification client. The request
     // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be
     // called from a different pid leaving a stale session reference.  Also we don't know how
     // to clear this reference if the client process dies.
     if (mNotificationClients.indexOfKey(caller) < 0) {
-        ALOGV("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession);
+        ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession);
         return;
     }
 
@@ -1935,11 +1960,14 @@
     ALOGV(" added new entry for %d", audioSession);
 }
 
-void AudioFlinger::releaseAudioSessionId(int audioSession)
+void AudioFlinger::releaseAudioSessionId(int audioSession, pid_t pid)
 {
     Mutex::Autolock _l(mLock);
     pid_t caller = IPCThreadState::self()->getCallingPid();
-    ALOGV("releasing %d from %d", audioSession, caller);
+    ALOGV("releasing %d from %d for %d", audioSession, caller, pid);
+    if (pid != -1 && (caller == getpid_cached)) {
+        caller = pid;
+    }
     size_t num = mAudioSessionRefs.size();
     for (size_t i = 0; i< num; i++) {
         AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 066d5d5..4799beb 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -60,8 +60,8 @@
 
 namespace android {
 
-class audio_track_cblk_t;
-class effect_param_cblk_t;
+struct audio_track_cblk_t;
+struct effect_param_cblk_t;
 class AudioMixer;
 class AudioBuffer;
 class AudioResampler;
@@ -102,7 +102,7 @@
                                 uint32_t sampleRate,
                                 audio_format_t format,
                                 audio_channel_mask_t channelMask,
-                                size_t frameCount,
+                                size_t *pFrameCount,
                                 IAudioFlinger::track_flags_t *flags,
                                 const sp<IMemory>& sharedBuffer,
                                 audio_io_handle_t output,
@@ -117,7 +117,7 @@
                                 uint32_t sampleRate,
                                 audio_format_t format,
                                 audio_channel_mask_t channelMask,
-                                size_t frameCount,
+                                size_t *pFrameCount,
                                 IAudioFlinger::track_flags_t *flags,
                                 pid_t tid,
                                 int *sessionId,
@@ -186,16 +186,16 @@
 
     virtual status_t setVoiceVolume(float volume);
 
-    virtual status_t getRenderPosition(size_t *halFrames, size_t *dspFrames,
+    virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
                                        audio_io_handle_t output) const;
 
-    virtual     unsigned int  getInputFramesLost(audio_io_handle_t ioHandle) const;
+    virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const;
 
     virtual int newAudioSessionId();
 
-    virtual void acquireAudioSessionId(int audioSession);
+    virtual void acquireAudioSessionId(int audioSession, pid_t pid);
 
-    virtual void releaseAudioSessionId(int audioSession);
+    virtual void releaseAudioSessionId(int audioSession, pid_t pid);
 
     virtual status_t queryNumberEffects(uint32_t *numEffects) const;
 
@@ -655,6 +655,8 @@
 
 #undef INCLUDING_FROM_AUDIOFLINGER_H
 
+const char *formatToString(audio_format_t format);
+
 // ----------------------------------------------------------------------------
 
 }; // namespace android
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index 8bea752..67d83b1 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -229,7 +229,7 @@
     return status;
 }
 
-void AudioMixer::unprepareTrackForDownmix(track_t* pTrack, int trackName) {
+void AudioMixer::unprepareTrackForDownmix(track_t* pTrack, int trackName __unused) {
     ALOGV("AudioMixer::unprepareTrackForDownmix(%d)", trackName);
 
     if (pTrack->downmixerBufferProvider != NULL) {
@@ -398,15 +398,16 @@
     ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name);
     track_t& track = mState.tracks[name];
 
-    int valueInt = (int)value;
-    int32_t *valueBuf = (int32_t *)value;
+    int valueInt = static_cast<int>(reinterpret_cast<uintptr_t>(value));
+    int32_t *valueBuf = reinterpret_cast<int32_t*>(value);
 
     switch (target) {
 
     case TRACK:
         switch (param) {
         case CHANNEL_MASK: {
-            audio_channel_mask_t mask = (audio_channel_mask_t) value;
+            audio_channel_mask_t mask =
+                static_cast<audio_channel_mask_t>(reinterpret_cast<uintptr_t>(value));
             if (track.channelMask != mask) {
                 uint32_t channelCount = popcount(mask);
                 ALOG_ASSERT((channelCount <= MAX_NUM_CHANNELS_TO_DOWNMIX) && channelCount);
@@ -536,7 +537,7 @@
                 // Should have a way to distinguish tracks with static ratios vs. dynamic ratios.
                 if (!((value == 44100 && devSampleRate == 48000) ||
                       (value == 48000 && devSampleRate == 44100))) {
-                    quality = AudioResampler::LOW_QUALITY;
+                    quality = AudioResampler::DYN_LOW_QUALITY;
                 } else {
                     quality = AudioResampler::DEFAULT_QUALITY;
                 }
@@ -783,8 +784,8 @@
     }
 }
 
-void AudioMixer::track__nop(track_t* t, int32_t* out, size_t outFrameCount, int32_t* temp,
-        int32_t* aux)
+void AudioMixer::track__nop(track_t* t __unused, int32_t* out __unused,
+        size_t outFrameCount __unused, int32_t* temp __unused, int32_t* aux __unused)
 {
 }
 
@@ -860,8 +861,8 @@
     }
 }
 
-void AudioMixer::track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
-        int32_t* aux)
+void AudioMixer::track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount,
+        int32_t* temp __unused, int32_t* aux)
 {
     const int16_t *in = static_cast<const int16_t *>(t->in);
 
@@ -951,8 +952,8 @@
     t->in = in;
 }
 
-void AudioMixer::track__16BitsMono(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
-        int32_t* aux)
+void AudioMixer::track__16BitsMono(track_t* t, int32_t* out, size_t frameCount,
+        int32_t* temp __unused, int32_t* aux)
 {
     const int16_t *in = static_cast<int16_t const *>(t->in);
 
@@ -1100,11 +1101,6 @@
         t.bufferProvider->getNextBuffer(&t.buffer, pts);
         t.frameCount = t.buffer.frameCount;
         t.in = t.buffer.raw;
-        // t.in == NULL can happen if the track was flushed just after having
-        // been enabled for mixing.
-        if (t.in == NULL) {
-            enabledTracks &= ~(1<<i);
-        }
     }
 
     e0 = enabledTracks;
@@ -1140,6 +1136,13 @@
                     aux = t.auxBuffer + numFrames;
                 }
                 while (outFrames) {
+                    // t.in == NULL can happen if the track was flushed just after having
+                    // been enabled for mixing.
+                   if (t.in == NULL) {
+                        enabledTracks &= ~(1<<i);
+                        e1 &= ~(1<<i);
+                        break;
+                    }
                     size_t inFrames = (t.frameCount > outFrames)?outFrames:t.frameCount;
                     if (inFrames > 0) {
                         t.hook(&t, outTemp + (BLOCKSIZE-outFrames)*MAX_NUM_CHANNELS, inFrames,
diff --git a/services/audioflinger/AudioMixer.h b/services/audioflinger/AudioMixer.h
index d5c9da7..d286986 100644
--- a/services/audioflinger/AudioMixer.h
+++ b/services/audioflinger/AudioMixer.h
@@ -216,7 +216,7 @@
         NBLog::Writer*  mLog;
         int32_t         reserved[1];
         // FIXME allocate dynamically to save some memory when maxNumTracks < MAX_NUM_TRACKS
-        track_t         tracks[MAX_NUM_TRACKS]; __attribute__((aligned(32)));
+        track_t         tracks[MAX_NUM_TRACKS] __attribute__((aligned(32)));
     };
 
     // AudioBufferProvider that wraps a track AudioBufferProvider by a call to a downmix effect
diff --git a/services/audioflinger/AudioPolicyService.cpp b/services/audioflinger/AudioPolicyService.cpp
index c5ad2c0..9980344 100644
--- a/services/audioflinger/AudioPolicyService.cpp
+++ b/services/audioflinger/AudioPolicyService.cpp
@@ -60,7 +60,7 @@
 // ----------------------------------------------------------------------------
 
 AudioPolicyService::AudioPolicyService()
-    : BnAudioPolicyService() , mpAudioPolicyDev(NULL) , mpAudioPolicy(NULL)
+    : BnAudioPolicyService(), mpAudioPolicyDev(NULL), mpAudioPolicy(NULL)
 {
     char value[PROPERTY_VALUE_MAX];
     const struct hw_module_t *module;
@@ -475,8 +475,9 @@
 
 audio_io_handle_t AudioPolicyService::getOutputForEffect(const effect_descriptor_t *desc)
 {
+    // FIXME change return type to status_t, and return NO_INIT here
     if (mpAudioPolicy == NULL) {
-        return NO_INIT;
+        return 0;
     }
     Mutex::Autolock _l(mLock);
     return mpAudioPolicy->get_output_for_effect(mpAudioPolicy, desc);
@@ -612,7 +613,7 @@
     return NO_ERROR;
 }
 
-status_t AudioPolicyService::dump(int fd, const Vector<String16>& args)
+status_t AudioPolicyService::dump(int fd, const Vector<String16>& args __unused)
 {
     if (!dumpAllowed()) {
         dumpPermissionDenial(fd);
@@ -1441,6 +1442,14 @@
     loadEffects(root, effects);
     loadInputSources(root, effects);
 
+    // delete effects to fix memory leak.
+    // as effects is local var and valgrind would treat this as memory leak
+    // and although it only did in mediaserver init, but free it in case mediaserver reboot
+    size_t i;
+    for (i = 0; i < effects.size(); i++) {
+      delete effects[i];
+    }
+
     config_free(root);
     free(root);
     free(data);
@@ -1452,7 +1461,7 @@
 extern "C" {
 
 
-static audio_module_handle_t aps_load_hw_module(void *service,
+static audio_module_handle_t aps_load_hw_module(void *service __unused,
                                              const char *name)
 {
     sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
@@ -1465,7 +1474,7 @@
 }
 
 // deprecated: replaced by aps_open_output_on_module()
-static audio_io_handle_t aps_open_output(void *service,
+static audio_io_handle_t aps_open_output(void *service __unused,
                                          audio_devices_t *pDevices,
                                          uint32_t *pSamplingRate,
                                          audio_format_t *pFormat,
@@ -1483,7 +1492,7 @@
                           pLatencyMs, flags);
 }
 
-static audio_io_handle_t aps_open_output_on_module(void *service,
+static audio_io_handle_t aps_open_output_on_module(void *service __unused,
                                                    audio_module_handle_t module,
                                                    audio_devices_t *pDevices,
                                                    uint32_t *pSamplingRate,
@@ -1502,7 +1511,7 @@
                           pLatencyMs, flags, offloadInfo);
 }
 
-static audio_io_handle_t aps_open_dup_output(void *service,
+static audio_io_handle_t aps_open_dup_output(void *service __unused,
                                                  audio_io_handle_t output1,
                                                  audio_io_handle_t output2)
 {
@@ -1514,7 +1523,7 @@
     return af->openDuplicateOutput(output1, output2);
 }
 
-static int aps_close_output(void *service, audio_io_handle_t output)
+static int aps_close_output(void *service __unused, audio_io_handle_t output)
 {
     sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
     if (af == 0) {
@@ -1524,7 +1533,7 @@
     return af->closeOutput(output);
 }
 
-static int aps_suspend_output(void *service, audio_io_handle_t output)
+static int aps_suspend_output(void *service __unused, audio_io_handle_t output)
 {
     sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
     if (af == 0) {
@@ -1535,7 +1544,7 @@
     return af->suspendOutput(output);
 }
 
-static int aps_restore_output(void *service, audio_io_handle_t output)
+static int aps_restore_output(void *service __unused, audio_io_handle_t output)
 {
     sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
     if (af == 0) {
@@ -1547,12 +1556,12 @@
 }
 
 // deprecated: replaced by aps_open_input_on_module(), and acoustics parameter is ignored
-static audio_io_handle_t aps_open_input(void *service,
+static audio_io_handle_t aps_open_input(void *service __unused,
                                         audio_devices_t *pDevices,
                                         uint32_t *pSamplingRate,
                                         audio_format_t *pFormat,
                                         audio_channel_mask_t *pChannelMask,
-                                        audio_in_acoustics_t acoustics)
+                                        audio_in_acoustics_t acoustics __unused)
 {
     sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
     if (af == 0) {
@@ -1563,7 +1572,7 @@
     return af->openInput((audio_module_handle_t)0, pDevices, pSamplingRate, pFormat, pChannelMask);
 }
 
-static audio_io_handle_t aps_open_input_on_module(void *service,
+static audio_io_handle_t aps_open_input_on_module(void *service __unused,
                                                   audio_module_handle_t module,
                                                   audio_devices_t *pDevices,
                                                   uint32_t *pSamplingRate,
@@ -1579,7 +1588,7 @@
     return af->openInput(module, pDevices, pSamplingRate, pFormat, pChannelMask);
 }
 
-static int aps_close_input(void *service, audio_io_handle_t input)
+static int aps_close_input(void *service __unused, audio_io_handle_t input)
 {
     sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
     if (af == 0) {
@@ -1589,7 +1598,7 @@
     return af->closeInput(input);
 }
 
-static int aps_set_stream_output(void *service, audio_stream_type_t stream,
+static int aps_set_stream_output(void *service __unused, audio_stream_type_t stream,
                                      audio_io_handle_t output)
 {
     sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
@@ -1600,7 +1609,7 @@
     return af->setStreamOutput(stream, output);
 }
 
-static int aps_move_effects(void *service, int session,
+static int aps_move_effects(void *service __unused, int session,
                                 audio_io_handle_t src_output,
                                 audio_io_handle_t dst_output)
 {
@@ -1612,7 +1621,7 @@
     return af->moveEffects(session, src_output, dst_output);
 }
 
-static char * aps_get_parameters(void *service, audio_io_handle_t io_handle,
+static char * aps_get_parameters(void *service __unused, audio_io_handle_t io_handle,
                                      const char *keys)
 {
     String8 result = AudioSystem::getParameters(io_handle, String8(keys));
@@ -1663,24 +1672,24 @@
 
 namespace {
     struct audio_policy_service_ops aps_ops = {
-        open_output           : aps_open_output,
-        open_duplicate_output : aps_open_dup_output,
-        close_output          : aps_close_output,
-        suspend_output        : aps_suspend_output,
-        restore_output        : aps_restore_output,
-        open_input            : aps_open_input,
-        close_input           : aps_close_input,
-        set_stream_volume     : aps_set_stream_volume,
-        set_stream_output     : aps_set_stream_output,
-        set_parameters        : aps_set_parameters,
-        get_parameters        : aps_get_parameters,
-        start_tone            : aps_start_tone,
-        stop_tone             : aps_stop_tone,
-        set_voice_volume      : aps_set_voice_volume,
-        move_effects          : aps_move_effects,
-        load_hw_module        : aps_load_hw_module,
-        open_output_on_module : aps_open_output_on_module,
-        open_input_on_module  : aps_open_input_on_module,
+        .open_output           = aps_open_output,
+        .open_duplicate_output = aps_open_dup_output,
+        .close_output          = aps_close_output,
+        .suspend_output        = aps_suspend_output,
+        .restore_output        = aps_restore_output,
+        .open_input            = aps_open_input,
+        .close_input           = aps_close_input,
+        .set_stream_volume     = aps_set_stream_volume,
+        .set_stream_output     = aps_set_stream_output,
+        .set_parameters        = aps_set_parameters,
+        .get_parameters        = aps_get_parameters,
+        .start_tone            = aps_start_tone,
+        .stop_tone             = aps_stop_tone,
+        .set_voice_volume      = aps_set_voice_volume,
+        .move_effects          = aps_move_effects,
+        .load_hw_module        = aps_load_hw_module,
+        .open_output_on_module = aps_open_output_on_module,
+        .open_input_on_module  = aps_open_input_on_module,
     };
 }; // namespace <unnamed>
 
diff --git a/services/audioflinger/AudioResampler.cpp b/services/audioflinger/AudioResampler.cpp
index 3b5a8c1..b206116 100644
--- a/services/audioflinger/AudioResampler.cpp
+++ b/services/audioflinger/AudioResampler.cpp
@@ -78,6 +78,9 @@
     int mX0R;
 };
 
+/*static*/
+const double AudioResampler::kPhaseMultiplier = 1L << AudioResampler::kNumPhaseBits;
+
 bool AudioResampler::qualityIsSupported(src_quality quality)
 {
     switch (quality) {
@@ -158,6 +161,16 @@
         atFinalQuality = true;
     }
 
+    /* if the caller requests DEFAULT_QUALITY and af.resampler.property
+     * has not been set, the target resampler quality is set to DYN_MED_QUALITY,
+     * and allowed to "throttle" down to DYN_LOW_QUALITY if necessary
+     * due to estimated CPU load of having too many active resamplers
+     * (the code below the if).
+     */
+    if (quality == DEFAULT_QUALITY) {
+        quality = DYN_MED_QUALITY;
+    }
+
     // naive implementation of CPU load throttling doesn't account for whether resampler is active
     pthread_mutex_lock(&mutex);
     for (;;) {
@@ -172,7 +185,6 @@
         // not enough CPU available for proposed quality level, so try next lowest level
         switch (quality) {
         default:
-        case DEFAULT_QUALITY:
         case LOW_QUALITY:
             atFinalQuality = true;
             break;
@@ -202,7 +214,6 @@
 
     switch (quality) {
     default:
-    case DEFAULT_QUALITY:
     case LOW_QUALITY:
         ALOGV("Create linear Resampler");
         resampler = new AudioResamplerOrder1(bitDepth, inChannelCount, sampleRate);
@@ -541,6 +552,16 @@
             size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr,
             uint32_t &phaseFraction, uint32_t phaseIncrement)
 {
+    (void)maxOutPt; // remove unused parameter warnings
+    (void)maxInIdx;
+    (void)outputIndex;
+    (void)out;
+    (void)inputIndex;
+    (void)vl;
+    (void)vr;
+    (void)phaseFraction;
+    (void)phaseIncrement;
+    (void)in;
 #define MO_PARAM5   "36"        // offset of parameter 5 (outputIndex)
 
     asm(
@@ -553,7 +574,7 @@
         "   ldr r8, [sp, #" MO_PARAM5 " + 4]\n"     // out
         "   ldr r0, [sp, #" MO_PARAM5 " + 0]\n"     // &outputIndex
         "   ldr r0, [r0]\n"                         // outputIndex
-        "   add r8, r0, asl #2\n"                   // curOut
+        "   add r8, r8, r0, asl #2\n"               // curOut
         "   ldr r9, [sp, #" MO_PARAM5 " + 24]\n"    // phaseIncrement
         "   ldr r10, [sp, #" MO_PARAM5 " + 12]\n"   // vl
         "   ldr r11, [sp, #" MO_PARAM5 " + 16]\n"   // vr
@@ -652,6 +673,16 @@
             size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr,
             uint32_t &phaseFraction, uint32_t phaseIncrement)
 {
+    (void)maxOutPt; // remove unused parameter warnings
+    (void)maxInIdx;
+    (void)outputIndex;
+    (void)out;
+    (void)inputIndex;
+    (void)vl;
+    (void)vr;
+    (void)phaseFraction;
+    (void)phaseIncrement;
+    (void)in;
 #define ST_PARAM5    "40"     // offset of parameter 5 (outputIndex)
     asm(
         "stmfd  sp!, {r4, r5, r6, r7, r8, r9, r10, r11, r12, lr}\n"
@@ -663,7 +694,7 @@
         "   ldr r8, [sp, #" ST_PARAM5 " + 4]\n"     // out
         "   ldr r0, [sp, #" ST_PARAM5 " + 0]\n"     // &outputIndex
         "   ldr r0, [r0]\n"                         // outputIndex
-        "   add r8, r0, asl #2\n"                   // curOut
+        "   add r8, r8, r0, asl #2\n"               // curOut
         "   ldr r9, [sp, #" ST_PARAM5 " + 24]\n"    // phaseIncrement
         "   ldr r10, [sp, #" ST_PARAM5 " + 12]\n"   // vl
         "   ldr r11, [sp, #" ST_PARAM5 " + 16]\n"   // vr
diff --git a/services/audioflinger/AudioResampler.h b/services/audioflinger/AudioResampler.h
index c341325..dc33f29 100644
--- a/services/audioflinger/AudioResampler.h
+++ b/services/audioflinger/AudioResampler.h
@@ -84,7 +84,7 @@
     static const uint32_t kPhaseMask = (1LU<<kNumPhaseBits)-1;
 
     // multiplier to calculate fixed point phase increment
-    static const double kPhaseMultiplier = 1L << kNumPhaseBits;
+    static const double kPhaseMultiplier;
 
     AudioResampler(int bitDepth, int inChannelCount, int32_t sampleRate, src_quality quality);
 
diff --git a/services/audioflinger/AudioResamplerDyn.cpp b/services/audioflinger/AudioResamplerDyn.cpp
index 1e38d3e..54c2309 100644
--- a/services/audioflinger/AudioResamplerDyn.cpp
+++ b/services/audioflinger/AudioResamplerDyn.cpp
@@ -161,7 +161,8 @@
 AudioResamplerDyn::AudioResamplerDyn(int bitDepth,
         int inChannelCount, int32_t sampleRate, src_quality quality)
     : AudioResampler(bitDepth, inChannelCount, sampleRate, quality),
-    mResampleType(0), mFilterSampleRate(0), mCoefBuffer(NULL)
+    mResampleType(0), mFilterSampleRate(0), mFilterQuality(DEFAULT_QUALITY),
+    mCoefBuffer(NULL)
 {
     mVolumeSimd[0] = mVolumeSimd[1] = 0;
     mConstants.set(128, 8, mSampleRate, mSampleRate); // TODO: set better
@@ -213,19 +214,16 @@
     // test the filter and report results
     double fp = (fcr - tbw/2)/c.mL;
     double fs = (fcr + tbw/2)/c.mL;
-    double fmin, fmax;
-    testFir(buf, c.mL, c.mHalfNumCoefs, 0., fp, 100, fmin, fmax);
-    double d1 = (fmax - fmin)/2.;
-    double ap = -20.*log10(1. - d1); // passband ripple
-    printf("passband(%lf, %lf): %.8lf %.8lf %.8lf\n", 0., fp, (fmax + fmin)/2., d1, ap);
-    testFir(buf, c.mL, c.mHalfNumCoefs, fs, 0.5, 100, fmin, fmax);
-    double d2 = fmax;
-    double as = -20.*log10(d2); // stopband attenuation
-    printf("stopband(%lf, %lf): %.8lf %.8lf %.3lf\n", fs, 0.5, (fmax + fmin)/2., d2, as);
+    double passMin, passMax, passRipple;
+    double stopMax, stopRipple;
+    testFir(buf, c.mL, c.mHalfNumCoefs, fp, fs, /*passSteps*/ 1000, /*stopSteps*/ 100000,
+            passMin, passMax, passRipple, stopMax, stopRipple);
+    printf("passband(%lf, %lf): %.8lf %.8lf %.8lf\n", 0., fp, passMin, passMax, passRipple);
+    printf("stopband(%lf, %lf): %.8lf %.3lf\n", fs, 0.5, stopMax, stopRipple);
 #endif
 }
 
-// recursive gcd (TODO: verify tail recursion elimination should make this iterate)
+// recursive gcd. Using objdump, it appears the tail recursion is converted to a while loop.
 static int gcd(int n, int m) {
     if (m == 0) {
         return n;
@@ -233,7 +231,16 @@
     return gcd(m, n % m);
 }
 
-static bool isClose(int32_t newSampleRate, int32_t prevSampleRate, int32_t filterSampleRate) {
+static bool isClose(int32_t newSampleRate, int32_t prevSampleRate,
+        int32_t filterSampleRate, int32_t outSampleRate) {
+
+    // different upsampling ratios do not need a filter change.
+    if (filterSampleRate != 0
+            && filterSampleRate < outSampleRate
+            && newSampleRate < outSampleRate)
+        return true;
+
+    // check design criteria again if downsampling is detected.
     int pdiff = absdiff(newSampleRate, prevSampleRate);
     int adiff = absdiff(newSampleRate, filterSampleRate);
 
@@ -255,8 +262,10 @@
 
     // TODO: Add precalculated Equiripple filters
 
-    if (!isClose(inSampleRate, oldSampleRate, mFilterSampleRate)) {
+    if (mFilterQuality != getQuality() ||
+            !isClose(inSampleRate, oldSampleRate, mFilterSampleRate, mSampleRate)) {
         mFilterSampleRate = inSampleRate;
+        mFilterQuality = getQuality();
 
         // Begin Kaiser Filter computation
         //
@@ -270,37 +279,46 @@
         double stopBandAtten;
         double tbwCheat = 1.; // how much we "cheat" into aliasing
         int halfLength;
-        if (getQuality() == DYN_HIGH_QUALITY) {
+        if (mFilterQuality == DYN_HIGH_QUALITY) {
             // 32b coefficients, 64 length
             useS32 = true;
             stopBandAtten = 98.;
-            halfLength = 32;
-        } else if (getQuality() == DYN_LOW_QUALITY) {
+            if (inSampleRate >= mSampleRate * 4) {
+                halfLength = 48;
+            } else if (inSampleRate >= mSampleRate * 2) {
+                halfLength = 40;
+            } else {
+                halfLength = 32;
+            }
+        } else if (mFilterQuality == DYN_LOW_QUALITY) {
             // 16b coefficients, 16-32 length
             useS32 = false;
             stopBandAtten = 80.;
-            if (mSampleRate >= inSampleRate * 2) {
+            if (inSampleRate >= mSampleRate * 4) {
+                halfLength = 24;
+            } else if (inSampleRate >= mSampleRate * 2) {
                 halfLength = 16;
             } else {
                 halfLength = 8;
             }
-            if (mSampleRate >= inSampleRate) {
+            if (inSampleRate <= mSampleRate) {
                 tbwCheat = 1.05;
             } else {
                 tbwCheat = 1.03;
             }
-        } else { // medium quality
+        } else { // DYN_MED_QUALITY
             // 16b coefficients, 32-64 length
+            // note: > 64 length filters with 16b coefs can have quantization noise problems
             useS32 = false;
             stopBandAtten = 84.;
-            if (mSampleRate >= inSampleRate * 4) {
+            if (inSampleRate >= mSampleRate * 4) {
                 halfLength = 32;
-            } else if (mSampleRate >= inSampleRate * 2) {
+            } else if (inSampleRate >= mSampleRate * 2) {
                 halfLength = 24;
             } else {
                 halfLength = 16;
             }
-            if (mSampleRate >= inSampleRate) {
+            if (inSampleRate <= mSampleRate) {
                 tbwCheat = 1.03;
             } else {
                 tbwCheat = 1.01;
@@ -315,9 +333,20 @@
 
         int phases = mSampleRate / gcd(mSampleRate, inSampleRate);
 
-        while (phases<63) { // too few phases, allow room for interpolation
+        // TODO: Once dynamic sample rate change is an option, the code below
+        // should be modified to execute only when dynamic sample rate change is enabled.
+        //
+        // as above, #phases less than 63 is too few phases for accurate linear interpolation.
+        // we increase the phases to compensate, but more phases means more memory per
+        // filter and more time to compute the filter.
+        //
+        // if we know that the filter will be used for dynamic sample rate changes,
+        // that would allow us skip this part for fixed sample rate resamplers.
+        //
+        while (phases<63) {
             phases *= 2; // this code only needed to support dynamic rate changes
         }
+
         if (phases>=256) {  // too many phases, always interpolate
             phases = 127;
         }
@@ -351,11 +380,7 @@
     if (locked) {
         mPhaseFraction = mPhaseFraction >> c.mShift << c.mShift; // remove fractional phase
     }
-    if (!USE_NEON) {
-        stride = 2; // C version only
-    }
-    // TODO: Remove this for testing
-    //stride = 2;
+
     mResampleType = RESAMPLETYPE(mChannelCount, locked, stride, !!useS32);
 #ifdef DEBUG_RESAMPLER
     printf("channels:%d  %s  stride:%d  %s  coef:%d  shift:%d\n",
@@ -371,7 +396,7 @@
     // 24 cases - this perhaps can be reduced later, as testing might take too long
     switch (mResampleType) {
 
-    // stride 16 (stride 2 for machines that do not support NEON)
+    // stride 16 (falls back to stride 2 for machines that do not support NEON)
     case RESAMPLETYPE(1, true, 16, 0):
         return resample<1, true, 16>(out, outFrameCount, mConstants.mFirCoefsS16, provider);
     case RESAMPLETYPE(2, true, 16, 0):
diff --git a/services/audioflinger/AudioResamplerDyn.h b/services/audioflinger/AudioResamplerDyn.h
index 85a01ab..df1fdbe 100644
--- a/services/audioflinger/AudioResamplerDyn.h
+++ b/services/audioflinger/AudioResamplerDyn.h
@@ -113,7 +113,8 @@
     Constants mConstants;  // current set of coefficient parameters
     int32_t __attribute__ ((aligned (8))) mVolumeSimd[2];
     int32_t mResampleType; // contains the resample type.
-    int32_t mFilterSampleRate; // designed sample rate for the filter
+    int32_t mFilterSampleRate; // designed filter sample rate.
+    src_quality mFilterQuality; // designed filter quality.
     void* mCoefBuffer; // if a filter is created, this is not null
 };
 
diff --git a/services/audioflinger/AudioResamplerFirGen.h b/services/audioflinger/AudioResamplerFirGen.h
index fa6ee3e..fac3001 100644
--- a/services/audioflinger/AudioResamplerFirGen.h
+++ b/services/audioflinger/AudioResamplerFirGen.h
@@ -20,19 +20,105 @@
 namespace android {
 
 /*
- * Sinc function is the traditional variant.
+ * generates a sine wave at equal steps.
  *
- * TODO: Investigate optimizations (regular sampling grid, NEON vector accelerations)
- * TODO: Remove comparison at 0 and trap at a higher level.
+ * As most of our functions use sine or cosine at equal steps,
+ * it is very efficient to compute them that way (single multiply and subtract),
+ * rather than invoking the math library sin() or cos() each time.
+ *
+ * SineGen uses Goertzel's Algorithm (as a generator not a filter)
+ * to calculate sine(wstart + n * wstep) or cosine(wstart + n * wstep)
+ * by stepping through 0, 1, ... n.
+ *
+ * e^i(wstart+wstep) = 2cos(wstep) * e^i(wstart) - e^i(wstart-wstep)
+ *
+ * or looking at just the imaginary sine term, as the cosine follows identically:
+ *
+ * sin(wstart+wstep) = 2cos(wstep) * sin(wstart) - sin(wstart-wstep)
+ *
+ * Goertzel's algorithm is more efficient than the angle addition formula,
+ * e^i(wstart+wstep) = e^i(wstart) * e^i(wstep), which takes up to
+ * 4 multiplies and 2 adds (or 3* and 3+) and requires both sine and
+ * cosine generation due to the complex * complex multiply (full rotation).
+ *
+ * See: http://en.wikipedia.org/wiki/Goertzel_algorithm
  *
  */
 
-static inline double sinc(double x) {
-    if (fabs(x) < FLT_MIN) {
-        return 1.;
+class SineGen {
+public:
+    SineGen(double wstart, double wstep, bool cosine = false) {
+        if (cosine) {
+            mCurrent = cos(wstart);
+            mPrevious = cos(wstart - wstep);
+        } else {
+            mCurrent = sin(wstart);
+            mPrevious = sin(wstart - wstep);
+        }
+        mTwoCos = 2.*cos(wstep);
     }
-    return sin(x) / x;
-}
+    SineGen(double expNow, double expPrev, double twoCosStep) {
+        mCurrent = expNow;
+        mPrevious = expPrev;
+        mTwoCos = twoCosStep;
+    }
+    inline double value() const {
+        return mCurrent;
+    }
+    inline void advance() {
+        double tmp = mCurrent;
+        mCurrent = mCurrent*mTwoCos - mPrevious;
+        mPrevious = tmp;
+    }
+    inline double valueAdvance() {
+        double tmp = mCurrent;
+        mCurrent = mCurrent*mTwoCos - mPrevious;
+        mPrevious = tmp;
+        return tmp;
+    }
+
+private:
+    double mCurrent; // current value of sine/cosine
+    double mPrevious; // previous value of sine/cosine
+    double mTwoCos; // stepping factor
+};
+
+/*
+ * generates a series of sine generators, phase offset by fixed steps.
+ *
+ * This is used to generate polyphase sine generators, one per polyphase
+ * in the filter code below.
+ *
+ * The SineGen returned by value() starts at innerStart = outerStart + n*outerStep;
+ * increments by innerStep.
+ *
+ */
+
+class SineGenGen {
+public:
+    SineGenGen(double outerStart, double outerStep, double innerStep, bool cosine = false)
+            : mSineInnerCur(outerStart, outerStep, cosine),
+              mSineInnerPrev(outerStart-innerStep, outerStep, cosine)
+    {
+        mTwoCos = 2.*cos(innerStep);
+    }
+    inline SineGen value() {
+        return SineGen(mSineInnerCur.value(), mSineInnerPrev.value(), mTwoCos);
+    }
+    inline void advance() {
+        mSineInnerCur.advance();
+        mSineInnerPrev.advance();
+    }
+    inline SineGen valueAdvance() {
+        return SineGen(mSineInnerCur.valueAdvance(), mSineInnerPrev.valueAdvance(), mTwoCos);
+    }
+
+private:
+    SineGen mSineInnerCur; // generate the inner sine values (stepped by outerStep).
+    SineGen mSineInnerPrev; // generate the inner sine previous values
+                            // (behind by innerStep, stepped by outerStep).
+    double mTwoCos; // the inner stepping factor for the returned SineGen.
+};
 
 static inline double sqr(double x) {
     return x * x;
@@ -46,8 +132,7 @@
  * The other variant is a non-noise shaped version for
  * S32 coefficients (noise shaping doesn't gain much).
  *
- * Caution: No bounds saturation is applied, but isn't needed in
- * this case.
+ * Caution: No bounds saturation is applied, but isn't needed in this case.
  *
  * @param x is the value to round.
  *
@@ -56,12 +141,15 @@
  * FIR coefficients generated here are never that close to 1.0 to pose an overflow condition).
  *
  * @param err is the previous error (actual - rounded) for the previous rounding op.
+ * For 16b coefficients this can improve stopband dB performance by up to 2dB.
+ *
+ * Many variants exist for the noise shaping: http://en.wikipedia.org/wiki/Noise_shaping
  *
  */
 
 static inline int64_t toint(double x, int64_t maxval, double& err) {
     double val = x * maxval;
-    double ival = floor(val + 0.5 + err*0.17);
+    double ival = floor(val + 0.5 + err*0.2);
     err = val - ival;
     return static_cast<int64_t>(ival);
 }
@@ -74,7 +162,8 @@
  * Modified Bessel function of the first kind
  * http://en.wikipedia.org/wiki/Bessel_function
  *
- * The formulas are taken from Abramowitz and Stegun:
+ * The formulas are taken from Abramowitz and Stegun,
+ * _Handbook of Mathematical Functions_ (links below):
  *
  * http://people.math.sfu.ca/~cbm/aands/page_375.htm
  * http://people.math.sfu.ca/~cbm/aands/page_378.htm
@@ -92,11 +181,13 @@
  * We use a bit of template math here, constexpr would probably be
  * more appropriate for a C++11 compiler.
  *
+ * For the intermediate range 3.75 < x < 15, we use minimax polynomial fit.
+ *
  */
 
 template <int N>
 struct I0Term {
-    static const double value = I0Term<N-1>::value/ (4. * N * N);
+    static const double value = I0Term<N-1>::value / (4. * N * N);
 };
 
 template <>
@@ -114,68 +205,275 @@
     static const double value = 0.398942280401432677939946059934381868475858631164934657665925;
 };
 
+#if USE_HORNERS_METHOD
+/* Polynomial evaluation of A + Bx + Cx^2 + Dx^3 + ...
+ * using Horner's Method: http://en.wikipedia.org/wiki/Horner's_method
+ *
+ * This has fewer multiplications than Estrin's method below, but has back to back
+ * floating point dependencies.
+ *
+ * On ARM this appears to work slower, so USE_HORNERS_METHOD is not default enabled.
+ */
+
+inline double Poly2(double A, double B, double x) {
+    return A + x * B;
+}
+
+inline double Poly4(double A, double B, double C, double D, double x) {
+    return A + x * (B + x * (C + x * (D)));
+}
+
+inline double Poly7(double A, double B, double C, double D, double E, double F, double G,
+        double x) {
+    return A + x * (B + x * (C + x * (D + x * (E + x * (F + x * (G))))));
+}
+
+inline double Poly9(double A, double B, double C, double D, double E, double F, double G,
+        double H, double I, double x) {
+    return A + x * (B + x * (C + x * (D + x * (E + x * (F + x * (G + x * (H + x * (I))))))));
+}
+
+#else
+/* Polynomial evaluation of A + Bx + Cx^2 + Dx^3 + ...
+ * using Estrin's Method: http://en.wikipedia.org/wiki/Estrin's_scheme
+ *
+ * This is typically faster, perhaps gains about 5-10% overall on ARM processors
+ * over Horner's method above.
+ */
+
+inline double Poly2(double A, double B, double x) {
+    return A + B * x;
+}
+
+inline double Poly3(double A, double B, double C, double x, double x2) {
+    return Poly2(A, B, x) + C * x2;
+}
+
+inline double Poly3(double A, double B, double C, double x) {
+    return Poly2(A, B, x) + C * x * x;
+}
+
+inline double Poly4(double A, double B, double C, double D, double x, double x2) {
+    return Poly2(A, B, x) + Poly2(C, D, x) * x2; // same as poly2(poly2, poly2, x2);
+}
+
+inline double Poly4(double A, double B, double C, double D, double x) {
+    return Poly4(A, B, C, D, x, x * x);
+}
+
+inline double Poly7(double A, double B, double C, double D, double E, double F, double G,
+        double x) {
+    double x2 = x * x;
+    return Poly4(A, B, C, D, x, x2) + Poly3(E, F, G, x, x2) * (x2 * x2);
+}
+
+inline double Poly8(double A, double B, double C, double D, double E, double F, double G,
+        double H, double x, double x2, double x4) {
+    return Poly4(A, B, C, D, x, x2) + Poly4(E, F, G, H, x, x2) * x4;
+}
+
+inline double Poly9(double A, double B, double C, double D, double E, double F, double G,
+        double H, double I, double x) {
+    double x2 = x * x;
+#if 1
+    // It does not seem faster to explicitly decompose Poly8 into Poly4, but
+    // could depend on compiler floating point scheduling.
+    double x4 = x2 * x2;
+    return Poly8(A, B, C, D, E, F, G, H, x, x2, x4) + I * (x4 * x4);
+#else
+    double val = Poly4(A, B, C, D, x, x2);
+    double x4 = x2 * x2;
+    return val + Poly4(E, F, G, H, x, x2) * x4 + I * (x4 * x4);
+#endif
+}
+#endif
+
 static inline double I0(double x) {
-    if (x < 3.75) { // TODO: Estrin's method instead of Horner's method?
+    if (x < 3.75) {
         x *= x;
-        return I0Term<0>::value + x*(
-                I0Term<1>::value + x*(
-                I0Term<2>::value + x*(
-                I0Term<3>::value + x*(
-                I0Term<4>::value + x*(
-                I0Term<5>::value + x*(
-                I0Term<6>::value)))))); // e < 1.6e-7
+        return Poly7(I0Term<0>::value, I0Term<1>::value,
+                I0Term<2>::value, I0Term<3>::value,
+                I0Term<4>::value, I0Term<5>::value,
+                I0Term<6>::value, x); // e < 1.6e-7
     }
-    // a bit ugly here - perhaps we expand the top series
-    // to permit computation to x < 20 (a reasonable range)
-    double y = 1./x;
-    return exp(x) * sqrt(y) * (
-            // note: reciprocal squareroot may be easier!
-            // http://en.wikipedia.org/wiki/Fast_inverse_square_root
-            I0ATerm<0>::value + y*(
-            I0ATerm<1>::value + y*(
-            I0ATerm<2>::value + y*(
-            I0ATerm<3>::value + y*(
-            I0ATerm<4>::value + y*(
-            I0ATerm<5>::value + y*(
-            I0ATerm<6>::value + y*(
-            I0ATerm<7>::value + y*(
-            I0ATerm<8>::value))))))))); // (... e) < 1.9e-7
+    if (1) {
+        /*
+         * Series expansion coefs are easy to calculate, but are expanded around 0,
+         * so error is unequal over the interval 0 < x < 3.75, the error being
+         * significantly better near 0.
+         *
+         * A better solution is to use precise minimax polynomial fits.
+         *
+         * We use a slightly more complicated solution for 3.75 < x < 15, based on
+         * the tables in Blair and Edwards, "Stable Rational Minimax Approximations
+         * to the Modified Bessel Functions I0(x) and I1(x)", Chalk Hill Nuclear Laboratory,
+         * AECL-4928.
+         *
+         * http://www.iaea.org/inis/collection/NCLCollectionStore/_Public/06/178/6178667.pdf
+         *
+         * See Table 11 for 0 < x < 15; e < 10^(-7.13).
+         *
+         * Note: Beta cannot exceed 15 (hence Stopband cannot exceed 144dB = 24b).
+         *
+         * This speeds up overall computation by about 40% over using the else clause below,
+         * which requires sqrt and exp.
+         *
+         */
+
+        x *= x;
+        double num = Poly9(-0.13544938430e9, -0.33153754512e8,
+                -0.19406631946e7, -0.48058318783e5,
+                -0.63269783360e3, -0.49520779070e1,
+                -0.24970910370e-1, -0.74741159550e-4,
+                -0.18257612460e-6, x);
+        double y = x - 225.; // reflection around 15 (squared)
+        double den = Poly4(-0.34598737196e8, 0.23852643181e6,
+                -0.70699387620e3, 0.10000000000e1, y);
+        return num / den;
+
+#if IO_EXTENDED_BETA
+        /* Table 42 for x > 15; e < 10^(-8.11).
+         * This is used for Beta>15, but is disabled here as
+         * we never use Beta that high.
+         *
+         * NOTE: This should be enabled only for x > 15.
+         */
+
+        double y = 1./x;
+        double z = y - (1./15);
+        double num = Poly2(0.415079861746e1, -0.5149092496e1, z);
+        double den = Poly3(0.103150763823e2, -0.14181687413e2,
+                0.1000000000e1, z);
+        return exp(x) * sqrt(y) * num / den;
+#endif
+    } else {
+        /*
+         * NOT USED, but reference for large Beta.
+         *
+         * Abramowitz and Stegun asymptotic formula.
+         * works for x > 3.75.
+         */
+        double y = 1./x;
+        return exp(x) * sqrt(y) *
+                // note: reciprocal squareroot may be easier!
+                // http://en.wikipedia.org/wiki/Fast_inverse_square_root
+                Poly9(I0ATerm<0>::value, I0ATerm<1>::value,
+                        I0ATerm<2>::value, I0ATerm<3>::value,
+                        I0ATerm<4>::value, I0ATerm<5>::value,
+                        I0ATerm<6>::value, I0ATerm<7>::value,
+                        I0ATerm<8>::value, y); // (... e) < 1.9e-7
+    }
 }
 
 /*
  * calculates the transition bandwidth for a Kaiser filter
  *
- * Formula 3.2.8, Multirate Systems and Filter Banks, PP Vaidyanathan, pg. 48
+ * Formula 3.2.8, Vaidyanathan, _Multirate Systems and Filter Banks_, p. 48
+ * Formula 7.76, Oppenheim and Schafer, _Discrete-time Signal Processing, 3e_, p. 542
  *
  * @param halfNumCoef is half the number of coefficients per filter phase.
+ *
  * @param stopBandAtten is the stop band attenuation desired.
+ *
  * @return the transition bandwidth in normalized frequency (0 <= f <= 0.5)
  */
 static inline double firKaiserTbw(int halfNumCoef, double stopBandAtten) {
-    return (stopBandAtten - 7.95)/(2.*14.36*halfNumCoef);
+    return (stopBandAtten - 7.95)/((2.*14.36)*halfNumCoef);
 }
 
 /*
- * calculates the fir transfer response.
+ * calculates the fir transfer response of the overall polyphase filter at w.
  *
- * calculates the transfer coefficient H(w) for 0 <= w <= PI.
- * Be careful be careful to consider the fact that this is an interpolated filter
- * of length L, so normalizing H(w)/L is probably what you expect.
+ * Calculates the DTFT transfer coefficient H(w) for 0 <= w <= PI, utilizing the
+ * fact that h[n] is symmetric (cosines only, no complex arithmetic).
+ *
+ * We use Goertzel's algorithm to accelerate the computation to essentially
+ * a single multiply and 2 adds per filter coefficient h[].
+ *
+ * Be careful be careful to consider that h[n] is the overall polyphase filter,
+ * with L phases, so rescaling H(w)/L is probably what you expect for "unity gain",
+ * as you only use one of the polyphases at a time.
  */
 template <typename T>
 static inline double firTransfer(const T* coef, int L, int halfNumCoef, double w) {
-    double accum = static_cast<double>(coef[0])*0.5;
-    coef += halfNumCoef;    // skip first row.
+    double accum = static_cast<double>(coef[0])*0.5;  // "center coefficient" from first bank
+    coef += halfNumCoef;    // skip first filterbank (picked up by the last filterbank).
+#if SLOW_FIRTRANSFER
+    /* Original code for reference.  This is equivalent to the code below, but slower. */
     for (int i=1 ; i<=L ; ++i) {
         for (int j=0, ix=i ; j<halfNumCoef ; ++j, ix+=L) {
             accum += cos(ix*w)*static_cast<double>(*coef++);
         }
     }
+#else
+    /*
+     * Our overall filter is stored striped by polyphases, not a contiguous h[n].
+     * We could fetch coefficients in a non-contiguous fashion
+     * but that will not scale to vector processing.
+     *
+     * We apply Goertzel's algorithm directly to each polyphase filter bank instead of
+     * using cosine generation/multiplication, thereby saving one multiply per inner loop.
+     *
+     * See: http://en.wikipedia.org/wiki/Goertzel_algorithm
+     * Also: Oppenheim and Schafer, _Discrete Time Signal Processing, 3e_, p. 720.
+     *
+     * We use the basic recursion to incorporate the cosine steps into real sequence x[n]:
+     * s[n] = x[n] + (2cosw)*s[n-1] + s[n-2]
+     *
+     * y[n] = s[n] - e^(iw)s[n-1]
+     *      = sum_{k=-\infty}^{n} x[k]e^(-iw(n-k))
+     *      = e^(-iwn) sum_{k=0}^{n} x[k]e^(iwk)
+     *
+     * The summation contains the frequency steps we want multiplied by the source
+     * (similar to a DTFT).
+     *
+     * Using symmetry, and just the real part (be careful, this must happen
+     * after any internal complex multiplications), the polyphase filterbank
+     * transfer function is:
+     *
+     * Hpp[n, w, w_0] = sum_{k=0}^{n} x[k] * cos(wk + w_0)
+     *                = Re{ e^(iwn + iw_0) y[n]}
+     *                = cos(wn+w_0) * s[n] - cos(w(n+1)+w_0) * s[n-1]
+     *
+     * using the fact that s[n] of real x[n] is real.
+     *
+     */
+    double dcos = 2. * cos(L*w);
+    int start = ((halfNumCoef)*L + 1);
+    SineGen cc((start - L) * w, w, true); // cosine
+    SineGen cp(start * w, w, true); // cosine
+    for (int i=1 ; i<=L ; ++i) {
+        double sc = 0;
+        double sp = 0;
+        for (int j=0 ; j<halfNumCoef ; ++j) {
+            double tmp = sc;
+            sc  = static_cast<double>(*coef++) + dcos*sc - sp;
+            sp = tmp;
+        }
+        // If we are awfully clever, we can apply Goertzel's algorithm
+        // again on the sc and sp sequences returned here.
+        accum += cc.valueAdvance() * sc - cp.valueAdvance() * sp;
+    }
+#endif
     return accum*2.;
 }
 
 /*
- * returns the minimum and maximum |H(f)| bounds
+ * evaluates the minimum and maximum |H(f)| bound in a band region.
+ *
+ * This is usually done with equally spaced increments in the target band in question.
+ * The passband is often very small, and sampled that way. The stopband is often much
+ * larger.
+ *
+ * We use the fact that the overall polyphase filter has an additional bank at the end
+ * for interpolation; hence it is overspecified for the H(f) computation.  Thus the
+ * first polyphase is never actually checked, excepting its first term.
+ *
+ * In this code we use the firTransfer() evaluator above, which uses Goertzel's
+ * algorithm to calculate the transfer function at each point.
+ *
+ * TODO: An alternative with equal spacing is the FFT/DFT.  An alternative with unequal
+ * spacing is a chirp transform.
  *
  * @param coef is the designed polyphase filter banks
  *
@@ -231,7 +529,63 @@
 }
 
 /*
- * Calculates the polyphase filter banks based on a windowed sinc function.
+ * evaluates the |H(f)| lowpass band characteristics.
+ *
+ * This function tests the lowpass characteristics for the overall polyphase filter,
+ * and is used to verify the design.  For this case, fp should be set to the
+ * passband normalized frequency from 0 to 0.5 for the overall filter (thus it
+ * is the designed polyphase bank value / L).  Likewise for fs.
+ *
+ * @param coef is the designed polyphase filter banks
+ *
+ * @param L is the number of phases (for interpolation)
+ *
+ * @param halfNumCoef should be half the number of coefficients for a single
+ * polyphase.
+ *
+ * @param fp is the passband normalized frequency, 0 < fp < fs < 0.5.
+ *
+ * @param fs is the stopband normalized frequency, 0 < fp < fs < 0.5.
+ *
+ * @param passSteps is the number of passband sampling steps.
+ *
+ * @param stopSteps is the number of stopband sampling steps.
+ *
+ * @param passMin is the minimum value in the passband
+ *
+ * @param passMax is the maximum value in the passband (useful for scaling).  This should
+ * be less than 1., to avoid sine wave test overflow.
+ *
+ * @param passRipple is the passband ripple.  Typically this should be less than 0.1 for
+ * an audio filter.  Generally speaker/headphone device characteristics will dominate
+ * the passband term.
+ *
+ * @param stopMax is the maximum value in the stopband.
+ *
+ * @param stopRipple is the stopband ripple, also known as stopband attenuation.
+ * Typically this should be greater than ~80dB for low quality, and greater than
+ * ~100dB for full 16b quality, otherwise aliasing may become noticeable.
+ *
+ */
+template <typename T>
+static void testFir(const T* coef, int L, int halfNumCoef,
+        double fp, double fs, int passSteps, int stopSteps,
+        double &passMin, double &passMax, double &passRipple,
+        double &stopMax, double &stopRipple) {
+    double fmin, fmax;
+    testFir(coef, L, halfNumCoef, 0., fp, passSteps, fmin, fmax);
+    double d1 = (fmax - fmin)/2.;
+    passMin = fmin;
+    passMax = fmax;
+    passRipple = -20.*log10(1. - d1); // passband ripple
+    testFir(coef, L, halfNumCoef, fs, 0.5, stopSteps, fmin, fmax);
+    // fmin is really not important for the stopband.
+    stopMax = fmax;
+    stopRipple = -20.*log10(fmax); // stopband ripple/attenuation
+}
+
+/*
+ * Calculates the overall polyphase filter based on a windowed sinc function.
  *
  * The windowed sinc is an odd length symmetric filter of exactly L*halfNumCoef*2+1
  * taps for the entire kernel.  This is then decomposed into L+1 polyphase filterbanks.
@@ -239,6 +593,9 @@
  * of the first bank shifted by one sample), and is unnecessary if one does
  * not do interpolation.
  *
+ * We use the last filterbank for some transfer function calculation purposes,
+ * so it needs to be generated anyways.
+ *
  * @param coef is the caller allocated space for coefficients.  This should be
  * exactly (L+1)*halfNumCoef in size.
  *
@@ -261,7 +618,8 @@
 static inline void firKaiserGen(T* coef, int L, int halfNumCoef,
         double stopBandAtten, double fcr, double atten) {
     //
-    // Formula 3.2.5, 3.2.7, Multirate Systems and Filter Banks, PP Vaidyanathan, pg. 48
+    // Formula 3.2.5, 3.2.7, Vaidyanathan, _Multirate Systems and Filter Banks_, p. 48
+    // Formula 7.75, Oppenheim and Schafer, _Discrete-time Signal Processing, 3e_, p. 542
     //
     // See also: http://melodi.ee.washington.edu/courses/ee518/notes/lec17.pdf
     //
@@ -284,13 +642,32 @@
 
     const int N = L * halfNumCoef; // non-negative half
     const double beta = 0.1102 * (stopBandAtten - 8.7); // >= 50dB always
-    const double yscale = 2. * atten * fcr / I0(beta);
-    const double xstep = 2. * M_PI * fcr / L;
+    const double xstep = (2. * M_PI) * fcr / L;
     const double xfrac = 1. / N;
-    double err = 0; // for noise shaping on int16_t coefficients
+    const double yscale = atten * L / (I0(beta) * M_PI);
+
+    // We use sine generators, which computes sines on regular step intervals.
+    // This speeds up overall computation about 40% from computing the sine directly.
+
+    SineGenGen sgg(0., xstep, L*xstep); // generates sine generators (one per polyphase)
+
     for (int i=0 ; i<=L ; ++i) { // generate an extra set of coefs for interpolation
+
+        // computation for a single polyphase of the overall filter.
+        SineGen sg = sgg.valueAdvance(); // current sine generator for "j" inner loop.
+        double err = 0; // for noise shaping on int16_t coefficients (over each polyphase)
+
         for (int j=0, ix=i ; j<halfNumCoef ; ++j, ix+=L) {
-            double y = I0(beta * sqrt(1.0 - sqr(ix * xfrac))) * sinc(ix * xstep) * yscale;
+            double y;
+            if (CC_LIKELY(ix)) {
+                double x = static_cast<double>(ix);
+
+                // sine generator: sg.valueAdvance() returns sin(ix*xstep);
+                y = I0(beta * sqrt(1.0 - sqr(x * xfrac))) * yscale * sg.valueAdvance() / x;
+            } else {
+                y = 2. * atten * fcr; // center of filter, sinc(0) = 1.
+                sg.advance();
+            }
 
             // (caution!) float version does not need rounding
             if (is_same<T, int16_t>::value) { // int16_t needs noise shaping
diff --git a/services/audioflinger/Configuration.h b/services/audioflinger/Configuration.h
index bc2038a..0754d9d 100644
--- a/services/audioflinger/Configuration.h
+++ b/services/audioflinger/Configuration.h
@@ -32,9 +32,6 @@
 // uncomment to enable fast mixer to take performance samples for later statistical analysis
 #define FAST_MIXER_STATISTICS
 
-// uncomment to allow fast tracks at non-native sample rate
-//#define FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE
-
 // uncomment for debugging timing problems related to StateQueue::push()
 //#define STATE_QUEUE_DUMP
 
diff --git a/services/audioflinger/Effects.cpp b/services/audioflinger/Effects.cpp
index 59b4770..29b56db 100644
--- a/services/audioflinger/Effects.cpp
+++ b/services/audioflinger/Effects.cpp
@@ -805,7 +805,112 @@
     return mOffloaded;
 }
 
-void AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args)
+String8 effectFlagsToString(uint32_t flags) {
+    String8 s;
+
+    s.append("conn. mode: ");
+    switch (flags & EFFECT_FLAG_TYPE_MASK) {
+    case EFFECT_FLAG_TYPE_INSERT: s.append("insert"); break;
+    case EFFECT_FLAG_TYPE_AUXILIARY: s.append("auxiliary"); break;
+    case EFFECT_FLAG_TYPE_REPLACE: s.append("replace"); break;
+    case EFFECT_FLAG_TYPE_PRE_PROC: s.append("preproc"); break;
+    case EFFECT_FLAG_TYPE_POST_PROC: s.append("postproc"); break;
+    default: s.append("unknown/reserved"); break;
+    }
+    s.append(", ");
+
+    s.append("insert pref: ");
+    switch (flags & EFFECT_FLAG_INSERT_MASK) {
+    case EFFECT_FLAG_INSERT_ANY: s.append("any"); break;
+    case EFFECT_FLAG_INSERT_FIRST: s.append("first"); break;
+    case EFFECT_FLAG_INSERT_LAST: s.append("last"); break;
+    case EFFECT_FLAG_INSERT_EXCLUSIVE: s.append("exclusive"); break;
+    default: s.append("unknown/reserved"); break;
+    }
+    s.append(", ");
+
+    s.append("volume mgmt: ");
+    switch (flags & EFFECT_FLAG_VOLUME_MASK) {
+    case EFFECT_FLAG_VOLUME_NONE: s.append("none"); break;
+    case EFFECT_FLAG_VOLUME_CTRL: s.append("implements control"); break;
+    case EFFECT_FLAG_VOLUME_IND: s.append("requires indication"); break;
+    default: s.append("unknown/reserved"); break;
+    }
+    s.append(", ");
+
+    uint32_t devind = flags & EFFECT_FLAG_DEVICE_MASK;
+    if (devind) {
+        s.append("device indication: ");
+        switch (devind) {
+        case EFFECT_FLAG_DEVICE_IND: s.append("requires updates"); break;
+        default: s.append("unknown/reserved"); break;
+        }
+        s.append(", ");
+    }
+
+    s.append("input mode: ");
+    switch (flags & EFFECT_FLAG_INPUT_MASK) {
+    case EFFECT_FLAG_INPUT_DIRECT: s.append("direct"); break;
+    case EFFECT_FLAG_INPUT_PROVIDER: s.append("provider"); break;
+    case EFFECT_FLAG_INPUT_BOTH: s.append("direct+provider"); break;
+    default: s.append("not set"); break;
+    }
+    s.append(", ");
+
+    s.append("output mode: ");
+    switch (flags & EFFECT_FLAG_OUTPUT_MASK) {
+    case EFFECT_FLAG_OUTPUT_DIRECT: s.append("direct"); break;
+    case EFFECT_FLAG_OUTPUT_PROVIDER: s.append("provider"); break;
+    case EFFECT_FLAG_OUTPUT_BOTH: s.append("direct+provider"); break;
+    default: s.append("not set"); break;
+    }
+    s.append(", ");
+
+    uint32_t accel = flags & EFFECT_FLAG_HW_ACC_MASK;
+    if (accel) {
+        s.append("hardware acceleration: ");
+        switch (accel) {
+        case EFFECT_FLAG_HW_ACC_SIMPLE: s.append("non-tunneled"); break;
+        case EFFECT_FLAG_HW_ACC_TUNNEL: s.append("tunneled"); break;
+        default: s.append("unknown/reserved"); break;
+        }
+        s.append(", ");
+    }
+
+    uint32_t modeind = flags & EFFECT_FLAG_AUDIO_MODE_MASK;
+    if (modeind) {
+        s.append("mode indication: ");
+        switch (modeind) {
+        case EFFECT_FLAG_AUDIO_MODE_IND: s.append("required"); break;
+        default: s.append("unknown/reserved"); break;
+        }
+        s.append(", ");
+    }
+
+    uint32_t srcind = flags & EFFECT_FLAG_AUDIO_SOURCE_MASK;
+    if (srcind) {
+        s.append("source indication: ");
+        switch (srcind) {
+        case EFFECT_FLAG_AUDIO_SOURCE_IND: s.append("required"); break;
+        default: s.append("unknown/reserved"); break;
+        }
+        s.append(", ");
+    }
+
+    if (flags & EFFECT_FLAG_OFFLOAD_MASK) {
+        s.append("offloadable, ");
+    }
+
+    int len = s.length();
+    if (s.length() > 2) {
+        char *str = s.lockBuffer(len);
+        s.unlockBuffer(len - 2);
+    }
+    return s;
+}
+
+
+void AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args __unused)
 {
     const size_t SIZE = 256;
     char buffer[SIZE];
@@ -821,8 +926,8 @@
     }
 
     result.append("\t\tSession Status State Engine:\n");
-    snprintf(buffer, SIZE, "\t\t%05d   %03d    %03d   0x%08x\n",
-            mSessionId, mStatus, mState, (uint32_t)mEffectInterface);
+    snprintf(buffer, SIZE, "\t\t%05d   %03d    %03d   %p\n",
+            mSessionId, mStatus, mState, mEffectInterface);
     result.append(buffer);
 
     result.append("\t\tDescriptor:\n");
@@ -839,9 +944,10 @@
                     mDescriptor.type.node[2],
                 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]);
     result.append(buffer);
-    snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n",
+    snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X (%s)\n",
             mDescriptor.apiVersion,
-            mDescriptor.flags);
+            mDescriptor.flags,
+            effectFlagsToString(mDescriptor.flags).string());
     result.append(buffer);
     snprintf(buffer, SIZE, "\t\t- name: %s\n",
             mDescriptor.name);
@@ -851,38 +957,38 @@
     result.append(buffer);
 
     result.append("\t\t- Input configuration:\n");
-    result.append("\t\t\tBuffer     Frames  Smp rate Channels Format\n");
-    snprintf(buffer, SIZE, "\t\t\t0x%08x %05d   %05d    %08x %d\n",
-            (uint32_t)mConfig.inputCfg.buffer.raw,
+    result.append("\t\t\tFrames  Smp rate Channels Format Buffer\n");
+    snprintf(buffer, SIZE, "\t\t\t%05zu   %05d    %08x %6d (%s) %p\n",
             mConfig.inputCfg.buffer.frameCount,
             mConfig.inputCfg.samplingRate,
             mConfig.inputCfg.channels,
-            mConfig.inputCfg.format);
+            mConfig.inputCfg.format,
+            formatToString((audio_format_t)mConfig.inputCfg.format),
+            mConfig.inputCfg.buffer.raw);
     result.append(buffer);
 
     result.append("\t\t- Output configuration:\n");
     result.append("\t\t\tBuffer     Frames  Smp rate Channels Format\n");
-    snprintf(buffer, SIZE, "\t\t\t0x%08x %05d   %05d    %08x %d\n",
-            (uint32_t)mConfig.outputCfg.buffer.raw,
+    snprintf(buffer, SIZE, "\t\t\t%p %05zu   %05d    %08x %d (%s)\n",
+            mConfig.outputCfg.buffer.raw,
             mConfig.outputCfg.buffer.frameCount,
             mConfig.outputCfg.samplingRate,
             mConfig.outputCfg.channels,
-            mConfig.outputCfg.format);
+            mConfig.outputCfg.format,
+            formatToString((audio_format_t)mConfig.outputCfg.format));
     result.append(buffer);
 
-    snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size());
+    snprintf(buffer, SIZE, "\t\t%zu Clients:\n", mHandles.size());
     result.append(buffer);
-    result.append("\t\t\tPid   Priority Ctrl Locked client server\n");
+    result.append("\t\t\t  Pid Priority Ctrl Locked client server\n");
     for (size_t i = 0; i < mHandles.size(); ++i) {
         EffectHandle *handle = mHandles[i];
         if (handle != NULL && !handle->destroyed_l()) {
-            handle->dump(buffer, SIZE);
+            handle->dumpToBuffer(buffer, SIZE);
             result.append(buffer);
         }
     }
 
-    result.append("\n");
-
     write(fd, result.string(), result.length());
 
     if (locked) {
@@ -1182,15 +1288,15 @@
 }
 
 
-void AudioFlinger::EffectHandle::dump(char* buffer, size_t size)
+void AudioFlinger::EffectHandle::dumpToBuffer(char* buffer, size_t size)
 {
     bool locked = mCblk != NULL && AudioFlinger::dumpTryLock(mCblk->lock);
 
-    snprintf(buffer, size, "\t\t\t%05d %05d    %01u    %01u      %05u  %05u\n",
+    snprintf(buffer, size, "\t\t\t%5d    %5d  %3s    %3s  %5u  %5u\n",
             (mClient == 0) ? getpid_cached : mClient->pid(),
             mPriority,
-            mHasControl,
-            !locked,
+            mHasControl ? "yes" : "no",
+            locked ? "yes" : "no",
             mCblk ? mCblk->clientIndex : 0,
             mCblk ? mCblk->serverIndex : 0
             );
@@ -1571,33 +1677,35 @@
     char buffer[SIZE];
     String8 result;
 
-    snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId);
+    size_t numEffects = mEffects.size();
+    snprintf(buffer, SIZE, "    %d effects for session %d\n", numEffects, mSessionId);
     result.append(buffer);
 
-    bool locked = AudioFlinger::dumpTryLock(mLock);
-    // failed to lock - AudioFlinger is probably deadlocked
-    if (!locked) {
-        result.append("\tCould not lock mutex:\n");
-    }
-
-    result.append("\tNum fx In buffer   Out buffer   Active tracks:\n");
-    snprintf(buffer, SIZE, "\t%02d     0x%08x  0x%08x   %d\n",
-            mEffects.size(),
-            (uint32_t)mInBuffer,
-            (uint32_t)mOutBuffer,
-            mActiveTrackCnt);
-    result.append(buffer);
-    write(fd, result.string(), result.size());
-
-    for (size_t i = 0; i < mEffects.size(); ++i) {
-        sp<EffectModule> effect = mEffects[i];
-        if (effect != 0) {
-            effect->dump(fd, args);
+    if (numEffects) {
+        bool locked = AudioFlinger::dumpTryLock(mLock);
+        // failed to lock - AudioFlinger is probably deadlocked
+        if (!locked) {
+            result.append("\tCould not lock mutex:\n");
         }
-    }
 
-    if (locked) {
-        mLock.unlock();
+        result.append("\tIn buffer   Out buffer   Active tracks:\n");
+        snprintf(buffer, SIZE, "\t%p  %p   %d\n",
+                mInBuffer,
+                mOutBuffer,
+                mActiveTrackCnt);
+        result.append(buffer);
+        write(fd, result.string(), result.size());
+
+        for (size_t i = 0; i < numEffects; ++i) {
+            sp<EffectModule> effect = mEffects[i];
+            if (effect != 0) {
+                effect->dump(fd, args);
+            }
+        }
+
+        if (locked) {
+            mLock.unlock();
+        }
     }
 }
 
diff --git a/services/audioflinger/Effects.h b/services/audioflinger/Effects.h
index 50535a2..ccc4825 100644
--- a/services/audioflinger/Effects.h
+++ b/services/audioflinger/Effects.h
@@ -209,7 +209,7 @@
     // destroyed_l() must be called with the associated EffectModule mLock held
     bool destroyed_l() const { return mDestroyed; }
 
-    void dump(char* buffer, size_t size);
+    void dumpToBuffer(char* buffer, size_t size);
 
 protected:
     friend class AudioFlinger;          // for mEffect, mHasControl, mEnabled
diff --git a/services/audioflinger/FastMixer.cpp b/services/audioflinger/FastMixer.cpp
index 7126e92..90122e0 100644
--- a/services/audioflinger/FastMixer.cpp
+++ b/services/audioflinger/FastMixer.cpp
@@ -236,10 +236,9 @@
                     sampleRate = Format_sampleRate(format);
                     ALOG_ASSERT(Format_channelCount(format) == FCC_2);
                 }
-                dumpState->mSampleRate = sampleRate;
             }
 
-            if ((format != previousFormat) || (frameCount != previous->mFrameCount)) {
+            if ((!Format_isEqual(format, previousFormat)) || (frameCount != previous->mFrameCount)) {
                 // FIXME to avoid priority inversion, don't delete here
                 delete mixer;
                 mixer = NULL;
@@ -321,12 +320,8 @@
                         mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::MAIN_BUFFER,
                                 (void *) mixBuffer);
                         // newly allocated track names default to full scale volume
-                        if (fastTrack->mSampleRate != 0 && fastTrack->mSampleRate != sampleRate) {
-                            mixer->setParameter(name, AudioMixer::RESAMPLE,
-                                    AudioMixer::SAMPLE_RATE, (void*) fastTrack->mSampleRate);
-                        }
                         mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::CHANNEL_MASK,
-                                (void *) fastTrack->mChannelMask);
+                                (void *)(uintptr_t)fastTrack->mChannelMask);
                         mixer->enable(name);
                     }
                     generations[i] = fastTrack->mGeneration;
@@ -353,16 +348,10 @@
                                 mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME1,
                                         (void *)0x1000);
                             }
-                            if (fastTrack->mSampleRate != 0 &&
-                                    fastTrack->mSampleRate != sampleRate) {
-                                mixer->setParameter(name, AudioMixer::RESAMPLE,
-                                        AudioMixer::SAMPLE_RATE, (void*) fastTrack->mSampleRate);
-                            } else {
-                                mixer->setParameter(name, AudioMixer::RESAMPLE,
-                                        AudioMixer::REMOVE, NULL);
-                            }
+                            mixer->setParameter(name, AudioMixer::RESAMPLE,
+                                    AudioMixer::REMOVE, NULL);
                             mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::CHANNEL_MASK,
-                                    (void *) fastTrack->mChannelMask);
+                                    (void *)(uintptr_t) fastTrack->mChannelMask);
                             // already enabled
                         }
                         generations[i] = fastTrack->mGeneration;
@@ -392,16 +381,8 @@
 
                 // Refresh the per-track timestamp
                 if (timestampStatus == NO_ERROR) {
-                    uint32_t trackFramesWrittenButNotPresented;
-                    uint32_t trackSampleRate = fastTrack->mSampleRate;
-                    // There is currently no sample rate conversion for fast tracks currently
-                    if (trackSampleRate != 0 && trackSampleRate != sampleRate) {
-                        trackFramesWrittenButNotPresented =
-                                ((int64_t) nativeFramesWrittenButNotPresented * trackSampleRate) /
-                                sampleRate;
-                    } else {
-                        trackFramesWrittenButNotPresented = nativeFramesWrittenButNotPresented;
-                    }
+                    uint32_t trackFramesWrittenButNotPresented =
+                        nativeFramesWrittenButNotPresented;
                     uint32_t trackFramesWritten = fastTrack->mBufferProvider->framesReleased();
                     // Can't provide an AudioTimestamp before first frame presented,
                     // or during the brief 32-bit wraparound window
@@ -419,9 +400,9 @@
                 if (fastTrack->mVolumeProvider != NULL) {
                     uint32_t vlr = fastTrack->mVolumeProvider->getVolumeLR();
                     mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME0,
-                            (void *)(vlr & 0xFFFF));
+                            (void *)(uintptr_t)(vlr & 0xFFFF));
                     mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME1,
-                            (void *)(vlr >> 16));
+                            (void *)(uintptr_t)(vlr >> 16));
                 }
                 // FIXME The current implementation of framesReady() for fast tracks
                 // takes a tryLock, which can block
@@ -715,7 +696,7 @@
 void FastMixerDumpState::dump(int fd) const
 {
     if (mCommand == FastMixerState::INITIAL) {
-        fdprintf(fd, "FastMixer not initialized\n");
+        fdprintf(fd, "  FastMixer not initialized\n");
         return;
     }
 #define COMMAND_MAX 32
@@ -749,10 +730,10 @@
     double measuredWarmupMs = (mMeasuredWarmupTs.tv_sec * 1000.0) +
             (mMeasuredWarmupTs.tv_nsec / 1000000.0);
     double mixPeriodSec = (double) mFrameCount / (double) mSampleRate;
-    fdprintf(fd, "FastMixer command=%s writeSequence=%u framesWritten=%u\n"
-                 "          numTracks=%u writeErrors=%u underruns=%u overruns=%u\n"
-                 "          sampleRate=%u frameCount=%u measuredWarmup=%.3g ms, warmupCycles=%u\n"
-                 "          mixPeriod=%.2f ms\n",
+    fdprintf(fd, "  FastMixer command=%s writeSequence=%u framesWritten=%u\n"
+                 "            numTracks=%u writeErrors=%u underruns=%u overruns=%u\n"
+                 "            sampleRate=%u frameCount=%zu measuredWarmup=%.3g ms, warmupCycles=%u\n"
+                 "            mixPeriod=%.2f ms\n",
                  string, mWriteSequence, mFramesWritten,
                  mNumTracks, mWriteErrors, mUnderruns, mOverruns,
                  mSampleRate, mFrameCount, measuredWarmupMs, mWarmupCycles,
@@ -803,14 +784,20 @@
         previousCpukHz = sampleCpukHz;
 #endif
     }
-    fdprintf(fd, "Simple moving statistics over last %.1f seconds:\n", wall.n() * mixPeriodSec);
-    fdprintf(fd, "  wall clock time in ms per mix cycle:\n"
-                 "    mean=%.2f min=%.2f max=%.2f stddev=%.2f\n",
-                 wall.mean()*1e-6, wall.minimum()*1e-6, wall.maximum()*1e-6, wall.stddev()*1e-6);
-    fdprintf(fd, "  raw CPU load in us per mix cycle:\n"
-                 "    mean=%.0f min=%.0f max=%.0f stddev=%.0f\n",
-                 loadNs.mean()*1e-3, loadNs.minimum()*1e-3, loadNs.maximum()*1e-3,
-                 loadNs.stddev()*1e-3);
+    if (n) {
+        fdprintf(fd, "  Simple moving statistics over last %.1f seconds:\n",
+                     wall.n() * mixPeriodSec);
+        fdprintf(fd, "    wall clock time in ms per mix cycle:\n"
+                     "      mean=%.2f min=%.2f max=%.2f stddev=%.2f\n",
+                     wall.mean()*1e-6, wall.minimum()*1e-6, wall.maximum()*1e-6,
+                     wall.stddev()*1e-6);
+        fdprintf(fd, "    raw CPU load in us per mix cycle:\n"
+                     "      mean=%.0f min=%.0f max=%.0f stddev=%.0f\n",
+                     loadNs.mean()*1e-3, loadNs.minimum()*1e-3, loadNs.maximum()*1e-3,
+                     loadNs.stddev()*1e-3);
+    } else {
+        fdprintf(fd, "  No FastMixer statistics available currently\n");
+    }
 #ifdef CPU_FREQUENCY_STATISTICS
     fdprintf(fd, "  CPU clock frequency in MHz:\n"
                  "    mean=%.0f min=%.0f max=%.0f stddev=%.0f\n",
@@ -828,9 +815,9 @@
             left.sample(tail[i]);
             right.sample(tail[n - (i + 1)]);
         }
-        fdprintf(fd, "Distribution of mix cycle times in ms for the tails (> ~3 stddev outliers):\n"
-                     "  left tail: mean=%.2f min=%.2f max=%.2f stddev=%.2f\n"
-                     "  right tail: mean=%.2f min=%.2f max=%.2f stddev=%.2f\n",
+        fdprintf(fd, "  Distribution of mix cycle times in ms for the tails (> ~3 stddev outliers):\n"
+                     "    left tail: mean=%.2f min=%.2f max=%.2f stddev=%.2f\n"
+                     "    right tail: mean=%.2f min=%.2f max=%.2f stddev=%.2f\n",
                      left.mean()*1e-6, left.minimum()*1e-6, left.maximum()*1e-6, left.stddev()*1e-6,
                      right.mean()*1e-6, right.minimum()*1e-6, right.maximum()*1e-6,
                      right.stddev()*1e-6);
@@ -843,9 +830,9 @@
     // Instead we always display all tracks, with an indication
     // of whether we think the track is active.
     uint32_t trackMask = mTrackMask;
-    fdprintf(fd, "Fast tracks: kMaxFastTracks=%u activeMask=%#x\n",
+    fdprintf(fd, "  Fast tracks: kMaxFastTracks=%u activeMask=%#x\n",
             FastMixerState::kMaxFastTracks, trackMask);
-    fdprintf(fd, "Index Active Full Partial Empty  Recent Ready\n");
+    fdprintf(fd, "  Index Active Full Partial Empty  Recent Ready\n");
     for (uint32_t i = 0; i < FastMixerState::kMaxFastTracks; ++i, trackMask >>= 1) {
         bool isActive = trackMask & 1;
         const FastTrackDump *ftDump = &mTracks[i];
@@ -865,7 +852,7 @@
             mostRecent = "?";
             break;
         }
-        fdprintf(fd, "%5u %6s %4u %7u %5u %7s %5u\n", i, isActive ? "yes" : "no",
+        fdprintf(fd, "  %5u %6s %4u %7u %5u %7s %5zu\n", i, isActive ? "yes" : "no",
                 (underruns.mBitFields.mFull) & UNDERRUN_MASK,
                 (underruns.mBitFields.mPartial) & UNDERRUN_MASK,
                 (underruns.mBitFields.mEmpty) & UNDERRUN_MASK,
diff --git a/services/audioflinger/FastMixerState.cpp b/services/audioflinger/FastMixerState.cpp
index 737de97..43ff233 100644
--- a/services/audioflinger/FastMixerState.cpp
+++ b/services/audioflinger/FastMixerState.cpp
@@ -20,7 +20,7 @@
 namespace android {
 
 FastTrack::FastTrack() :
-    mBufferProvider(NULL), mVolumeProvider(NULL), mSampleRate(0),
+    mBufferProvider(NULL), mVolumeProvider(NULL),
     mChannelMask(AUDIO_CHANNEL_OUT_STEREO), mGeneration(0)
 {
 }
diff --git a/services/audioflinger/FastMixerState.h b/services/audioflinger/FastMixerState.h
index f6e7903..9739fe9 100644
--- a/services/audioflinger/FastMixerState.h
+++ b/services/audioflinger/FastMixerState.h
@@ -43,7 +43,6 @@
 
     ExtendedAudioBufferProvider* mBufferProvider; // must be NULL if inactive, or non-NULL if active
     VolumeProvider*         mVolumeProvider; // optional; if NULL then full-scale
-    unsigned                mSampleRate;     // optional; if zero then use mixer sample rate
     audio_channel_mask_t    mChannelMask;    // AUDIO_CHANNEL_OUT_MONO or AUDIO_CHANNEL_OUT_STEREO
     int                     mGeneration;     // increment when any field is assigned
 };
diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h
index 4b6c74d..b5e763d 100644
--- a/services/audioflinger/PlaybackTracks.h
+++ b/services/audioflinger/PlaybackTracks.h
@@ -37,7 +37,7 @@
     virtual status_t    initCheck() const;
 
     static  void        appendDumpHeader(String8& result);
-            void        dump(char* buffer, size_t size);
+            void        dump(char* buffer, size_t size, bool active);
     virtual status_t    start(AudioSystem::sync_event_t event =
                                     AudioSystem::SYNC_EVENT_NONE,
                              int triggerSession = 0);
@@ -94,6 +94,8 @@
     bool isReady() const;
     void setPaused() { mState = PAUSED; }
     void reset();
+    bool isFlushPending() const { return mFlushHwPending; }
+    void flushAck();
 
     bool isOutputTrack() const {
         return (mStreamType == AUDIO_STREAM_CNT);
@@ -155,6 +157,7 @@
     bool                mIsInvalid; // non-resettable latch, set by invalidate()
     AudioTrackServerProxy*  mAudioTrackServerProxy;
     bool                mResumeToStopping; // track was paused in stopping state.
+    bool                mFlushHwPending; // track requests for thread flush
 };  // end of Track
 
 class TimedTrack : public Track {
diff --git a/services/audioflinger/RecordTracks.h b/services/audioflinger/RecordTracks.h
index 5ef6f58..fc3171f 100644
--- a/services/audioflinger/RecordTracks.h
+++ b/services/audioflinger/RecordTracks.h
@@ -45,7 +45,7 @@
                                                 return tmp; }
 
     static  void        appendDumpHeader(String8& result);
-            void        dump(char* buffer, size_t size);
+            void        dump(char* buffer, size_t size, bool active);
 
 private:
     friend class AudioFlinger;  // for mState
diff --git a/services/audioflinger/StateQueue.cpp b/services/audioflinger/StateQueue.cpp
index c2d3bbd..48399c0 100644
--- a/services/audioflinger/StateQueue.cpp
+++ b/services/audioflinger/StateQueue.cpp
@@ -58,7 +58,11 @@
 
 template<typename T> const T* StateQueue<T>::poll()
 {
+#ifdef __LP64__
+    const T *next = (const T *) android_atomic_acquire_load64((volatile int64_t *) &mNext);
+#else
     const T *next = (const T *) android_atomic_acquire_load((volatile int32_t *) &mNext);
+#endif
     if (next != mCurrent) {
         mAck = next;    // no additional barrier needed
         mCurrent = next;
@@ -140,7 +144,11 @@
         }
 
         // publish
+#ifdef __LP64__
+        android_atomic_release_store64((int64_t) mMutating, (volatile int64_t *) &mNext);
+#else
         android_atomic_release_store((int32_t) mMutating, (volatile int32_t *) &mNext);
+#endif
         mExpecting = mMutating;
 
         // copy with circular wraparound
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 35b8575..b064e89 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -185,7 +185,11 @@
 {
 }
 
-void CpuStats::sample(const String8 &title) {
+void CpuStats::sample(const String8 &title
+#ifndef DEBUG_CPU_USAGE
+                __unused
+#endif
+        ) {
 #ifdef DEBUG_CPU_USAGE
     // get current thread's delta CPU time in wall clock ns
     double wcNs;
@@ -421,7 +425,54 @@
     }
 }
 
-void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args)
+String8 channelMaskToString(audio_channel_mask_t mask, bool output) {
+    String8 s;
+    if (output) {
+        if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, ");
+        if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, ");
+        if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, ");
+        if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, ");
+        if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, ");
+        if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, ");
+        if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, ");
+        if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, ");
+        if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, ");
+        if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, ");
+        if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, ");
+        if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,");
+        if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, ");
+        if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, ");
+        if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, ");
+        if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, ");
+        if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " );
+        if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " );
+        if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown,  ");
+    } else {
+        if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, ");
+        if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, ");
+        if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, ");
+        if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, ");
+        if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, ");
+        if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, ");
+        if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, ");
+        if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, ");
+        if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, ");
+        if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, ");
+        if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, ");
+        if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, ");
+        if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, ");
+        if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, ");
+        if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown,  ");
+    }
+    int len = s.length();
+    if (s.length() > 2) {
+        char *str = s.lockBuffer(len);
+        s.unlockBuffer(len - 2);
+    }
+    return s;
+}
+
+void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused)
 {
     const size_t SIZE = 256;
     char buffer[SIZE];
@@ -429,49 +480,43 @@
 
     bool locked = AudioFlinger::dumpTryLock(mLock);
     if (!locked) {
-        snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this);
-        write(fd, buffer, strlen(buffer));
+        fdprintf(fd, "thread %p maybe dead locked\n", this);
     }
 
-    snprintf(buffer, SIZE, "io handle: %d\n", mId);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "TID: %d\n", getTid());
-    result.append(buffer);
-    snprintf(buffer, SIZE, "standby: %d\n", mStandby);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "HAL buffer size: %u bytes\n", mBufferSize);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "Channel Count: %u\n", mChannelCount);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "Format: %d\n", mFormat);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize);
-    result.append(buffer);
-
-    snprintf(buffer, SIZE, "\nPending setParameters commands: \n");
-    result.append(buffer);
-    result.append(" Index Command");
-    for (size_t i = 0; i < mNewParameters.size(); ++i) {
-        snprintf(buffer, SIZE, "\n %02d    ", i);
-        result.append(buffer);
-        result.append(mNewParameters[i]);
+    fdprintf(fd, "  I/O handle: %d\n", mId);
+    fdprintf(fd, "  TID: %d\n", getTid());
+    fdprintf(fd, "  Standby: %s\n", mStandby ? "yes" : "no");
+    fdprintf(fd, "  Sample rate: %u\n", mSampleRate);
+    fdprintf(fd, "  HAL frame count: %zu\n", mFrameCount);
+    fdprintf(fd, "  HAL buffer size: %u bytes\n", mBufferSize);
+    fdprintf(fd, "  Channel Count: %u\n", mChannelCount);
+    fdprintf(fd, "  Channel Mask: 0x%08x (%s)\n", mChannelMask,
+            channelMaskToString(mChannelMask, mType != RECORD).string());
+    fdprintf(fd, "  Format: 0x%x (%s)\n", mFormat, formatToString(mFormat));
+    fdprintf(fd, "  Frame size: %zu\n", mFrameSize);
+    fdprintf(fd, "  Pending setParameters commands:");
+    size_t numParams = mNewParameters.size();
+    if (numParams) {
+        fdprintf(fd, "\n   Index Command");
+        for (size_t i = 0; i < numParams; ++i) {
+            fdprintf(fd, "\n   %02zu    ", i);
+            fdprintf(fd, mNewParameters[i]);
+        }
+        fdprintf(fd, "\n");
+    } else {
+        fdprintf(fd, " none\n");
     }
-
-    snprintf(buffer, SIZE, "\n\nPending config events: \n");
-    result.append(buffer);
-    for (size_t i = 0; i < mConfigEvents.size(); i++) {
-        mConfigEvents[i]->dump(buffer, SIZE);
-        result.append(buffer);
+    fdprintf(fd, "  Pending config events:");
+    size_t numConfig = mConfigEvents.size();
+    if (numConfig) {
+        for (size_t i = 0; i < numConfig; i++) {
+            mConfigEvents[i]->dump(buffer, SIZE);
+            fdprintf(fd, "\n    %s", buffer);
+        }
+        fdprintf(fd, "\n");
+    } else {
+        fdprintf(fd, " none\n");
     }
-    result.append("\n");
-
-    write(fd, result.string(), result.size());
 
     if (locked) {
         mLock.unlock();
@@ -484,10 +529,11 @@
     char buffer[SIZE];
     String8 result;
 
-    snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size());
+    size_t numEffectChains = mEffectChains.size();
+    snprintf(buffer, SIZE, "  %zu Effect Chains\n", numEffectChains);
     write(fd, buffer, strlen(buffer));
 
-    for (size_t i = 0; i < mEffectChains.size(); ++i) {
+    for (size_t i = 0; i < numEffectChains; ++i) {
         sp<EffectChain> chain = mEffectChains[i];
         if (chain != 0) {
             chain->dump(fd, args);
@@ -604,7 +650,7 @@
     mPowerManager.clear();
 }
 
-void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who)
+void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused)
 {
     sp<ThreadBase> thread = mThread.promote();
     if (thread != 0) {
@@ -1088,13 +1134,13 @@
     dumpEffectChains(fd, args);
 }
 
-void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args)
+void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused)
 {
     const size_t SIZE = 256;
     char buffer[SIZE];
     String8 result;
 
-    result.appendFormat("Output thread %p stream volumes in dB:\n    ", this);
+    result.appendFormat("  Stream volumes in dB: ");
     for (int i = 0; i < AUDIO_STREAM_CNT; ++i) {
         const stream_type_t *st = &mStreamTypes[i];
         if (i > 0) {
@@ -1109,60 +1155,62 @@
     write(fd, result.string(), result.length());
     result.clear();
 
-    snprintf(buffer, SIZE, "Output thread %p tracks\n", this);
-    result.append(buffer);
-    Track::appendDumpHeader(result);
-    for (size_t i = 0; i < mTracks.size(); ++i) {
-        sp<Track> track = mTracks[i];
-        if (track != 0) {
-            track->dump(buffer, SIZE);
-            result.append(buffer);
-        }
-    }
-
-    snprintf(buffer, SIZE, "Output thread %p active tracks\n", this);
-    result.append(buffer);
-    Track::appendDumpHeader(result);
-    for (size_t i = 0; i < mActiveTracks.size(); ++i) {
-        sp<Track> track = mActiveTracks[i].promote();
-        if (track != 0) {
-            track->dump(buffer, SIZE);
-            result.append(buffer);
-        }
-    }
-    write(fd, result.string(), result.size());
-
     // These values are "raw"; they will wrap around.  See prepareTracks_l() for a better way.
     FastTrackUnderruns underruns = getFastTrackUnderruns(0);
-    fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n",
+    fdprintf(fd, "  Normal mixer raw underrun counters: partial=%u empty=%u\n",
             underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty);
+
+    size_t numtracks = mTracks.size();
+    size_t numactive = mActiveTracks.size();
+    fdprintf(fd, "  %d Tracks", numtracks);
+    size_t numactiveseen = 0;
+    if (numtracks) {
+        fdprintf(fd, " of which %d are active\n", numactive);
+        Track::appendDumpHeader(result);
+        for (size_t i = 0; i < numtracks; ++i) {
+            sp<Track> track = mTracks[i];
+            if (track != 0) {
+                bool active = mActiveTracks.indexOf(track) >= 0;
+                if (active) {
+                    numactiveseen++;
+                }
+                track->dump(buffer, SIZE, active);
+                result.append(buffer);
+            }
+        }
+    } else {
+        result.append("\n");
+    }
+    if (numactiveseen != numactive) {
+        // some tracks in the active list were not in the tracks list
+        snprintf(buffer, SIZE, "  The following tracks are in the active list but"
+                " not in the track list\n");
+        result.append(buffer);
+        Track::appendDumpHeader(result);
+        for (size_t i = 0; i < numactive; ++i) {
+            sp<Track> track = mActiveTracks[i].promote();
+            if (track != 0 && mTracks.indexOf(track) < 0) {
+                track->dump(buffer, SIZE, true);
+                result.append(buffer);
+            }
+        }
+    }
+
+    write(fd, result.string(), result.size());
+
 }
 
 void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
 {
-    const size_t SIZE = 256;
-    char buffer[SIZE];
-    String8 result;
-
-    snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n",
-            ns2ms(systemTime() - mLastWriteTime));
-    result.append(buffer);
-    snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer);
-    result.append(buffer);
-    write(fd, result.string(), result.size());
-    fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask);
+    fdprintf(fd, "\nOutput thread %p:\n", this);
+    fdprintf(fd, "  Normal frame count: %zu\n", mNormalFrameCount);
+    fdprintf(fd, "  Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime));
+    fdprintf(fd, "  Total writes: %d\n", mNumWrites);
+    fdprintf(fd, "  Delayed writes: %d\n", mNumDelayedWrites);
+    fdprintf(fd, "  Blocked in write: %s\n", mInWrite ? "yes" : "no");
+    fdprintf(fd, "  Suspend count: %d\n", mSuspended);
+    fdprintf(fd, "  Mix buffer : %p\n", mMixBuffer);
+    fdprintf(fd, "  Fast track availMask=%#x\n", mFastTrackAvailMask);
 
     dumpBase(fd, args);
 }
@@ -1190,7 +1238,7 @@
         uint32_t sampleRate,
         audio_format_t format,
         audio_channel_mask_t channelMask,
-        size_t frameCount,
+        size_t *pFrameCount,
         const sp<IMemory>& sharedBuffer,
         int sessionId,
         IAudioFlinger::track_flags_t *flags,
@@ -1198,6 +1246,7 @@
         int uid,
         status_t *status)
 {
+    size_t frameCount = *pFrameCount;
     sp<Track> track;
     status_t lStatus;
 
@@ -1226,10 +1275,8 @@
             // mono or stereo
             ( (channelMask == AUDIO_CHANNEL_OUT_MONO) ||
               (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) &&
-#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE
             // hardware sample rate
             (sampleRate == mSampleRate) &&
-#endif
             // normal mixer has an associated fast mixer
             hasFastMixer() &&
             // there are sufficient fast track slots available
@@ -1266,12 +1313,13 @@
         }
       }
     }
+    *pFrameCount = frameCount;
 
     if (mType == DIRECT) {
         if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) {
             if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
-                ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x "
-                        "for output %p with format %d",
+                ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x "
+                        "for output %p with format %#x",
                         sampleRate, format, channelMask, mOutput, mFormat);
                 lStatus = BAD_VALUE;
                 goto Exit;
@@ -1279,16 +1327,16 @@
         }
     } else if (mType == OFFLOAD) {
         if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
-            ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \""
-                    "for output %p with format %d",
+            ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \""
+                    "for output %p with format %#x",
                     sampleRate, format, channelMask, mOutput, mFormat);
             lStatus = BAD_VALUE;
             goto Exit;
         }
     } else {
         if ((format & AUDIO_FORMAT_MAIN_MASK) != AUDIO_FORMAT_PCM) {
-                ALOGE("createTrack_l() Bad parameter: format %d \""
-                        "for output %p with format %d",
+                ALOGE("createTrack_l() Bad parameter: format %#x \""
+                        "for output %p with format %#x",
                         format, mOutput, mFormat);
                 lStatus = BAD_VALUE;
                 goto Exit;
@@ -1339,7 +1387,8 @@
         // but TimedTrack::create() is a factory that could fail by returning NULL
         lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY;
         if (lStatus != NO_ERROR) {
-            track.clear();
+            ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus);
+            // track must be cleared from the caller as the caller has the AF lock
             goto Exit;
         }
 
@@ -1483,9 +1532,7 @@
         status = NO_ERROR;
     }
 
-    ALOGV("signal playback thread");
-    broadcast_l();
-
+    onAddNewTrack_l();
     return status;
 }
 
@@ -1611,7 +1658,7 @@
 
 // static
 int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event,
-                                                void *param,
+                                                void *param __unused,
                                                 void *cookie)
 {
     AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie;
@@ -1645,10 +1692,10 @@
     mChannelCount = popcount(mChannelMask);
     mFormat = mOutput->stream->common.get_format(&mOutput->stream->common);
     if (!audio_is_valid_format(mFormat)) {
-        LOG_FATAL("HAL format %d not valid for output", mFormat);
+        LOG_FATAL("HAL format %#x not valid for output", mFormat);
     }
     if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) {
-        LOG_FATAL("HAL format %d not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT",
+        LOG_FATAL("HAL format %#x not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT",
                 mFormat);
     }
     mFrameSize = audio_stream_frame_size(&mOutput->stream->common);
@@ -1729,7 +1776,7 @@
 }
 
 
-status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames)
+status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames)
 {
     if (halFrames == NULL || dspFrames == NULL) {
         return BAD_VALUE;
@@ -1747,7 +1794,11 @@
         *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0;
         return NO_ERROR;
     } else {
-        return mOutput->stream->get_render_position(mOutput->stream, dspFrames);
+        status_t status;
+        uint32_t frames;
+        status = mOutput->stream->get_render_position(mOutput->stream, &frames);
+        *dspFrames = (size_t)frames;
+        return status;
     }
 }
 
@@ -2355,20 +2406,20 @@
                         (mMixerStatus == MIXER_DRAIN_ALL)) {
                     threadLoop_drain();
                 }
-if (mType == MIXER) {
-                // write blocked detection
-                nsecs_t now = systemTime();
-                nsecs_t delta = now - mLastWriteTime;
-                if (!mStandby && delta > maxPeriod) {
-                    mNumDelayedWrites++;
-                    if ((now - lastWarning) > kWarningThrottleNs) {
-                        ATRACE_NAME("underrun");
-                        ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p",
-                                ns2ms(delta), mNumDelayedWrites, this);
-                        lastWarning = now;
+                if (mType == MIXER) {
+                    // write blocked detection
+                    nsecs_t now = systemTime();
+                    nsecs_t delta = now - mLastWriteTime;
+                    if (!mStandby && delta > maxPeriod) {
+                        mNumDelayedWrites++;
+                        if ((now - lastWarning) > kWarningThrottleNs) {
+                            ATRACE_NAME("underrun");
+                            ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p",
+                                    ns2ms(delta), mNumDelayedWrites, this);
+                            lastWarning = now;
+                        }
                     }
                 }
-}
 
             } else {
                 usleep(sleepTime);
@@ -2720,12 +2771,6 @@
     PlaybackThread::threadLoop_standby();
 }
 
-// Empty implementation for standard mixer
-// Overridden for offloaded playback
-void AudioFlinger::PlaybackThread::flushOutput_l()
-{
-}
-
 bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l()
 {
     return false;
@@ -2757,6 +2802,12 @@
     }
 }
 
+void AudioFlinger::PlaybackThread::onAddNewTrack_l()
+{
+    ALOGV("signal playback thread");
+    broadcast_l();
+}
+
 void AudioFlinger::MixerThread::threadLoop_mix()
 {
     // obtain the presentation timestamp of the next output buffer
@@ -2984,7 +3035,6 @@
                     VolumeProvider *vp = track;
                     fastTrack->mBufferProvider = eabp;
                     fastTrack->mVolumeProvider = vp;
-                    fastTrack->mSampleRate = track->mSampleRate;
                     fastTrack->mChannelMask = track->mChannelMask;
                     fastTrack->mGeneration++;
                     state->mTrackMask |= 1 << j;
@@ -3037,25 +3087,20 @@
             // add frames already consumed but not yet released by the resampler
             // because mAudioTrackServerProxy->framesReady() will include these frames
             desiredFrames += mAudioMixer->getUnreleasedFrames(track->name());
+#if 0
             // the minimum track buffer size is normally twice the number of frames necessary
             // to fill one buffer and the resampler should not leave more than one buffer worth
             // of unreleased frames after each pass, but just in case...
             ALOG_ASSERT(desiredFrames <= cblk->frameCount_);
+#endif
         }
         uint32_t minFrames = 1;
         if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() &&
                 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) {
             minFrames = desiredFrames;
         }
-        // It's not safe to call framesReady() for a static buffer track, so assume it's ready
-        size_t framesReady;
-        if (track->sharedBuffer() == 0) {
-            framesReady = track->framesReady();
-        } else if (track->isStopped()) {
-            framesReady = 0;
-        } else {
-            framesReady = 1;
-        }
+
+        size_t framesReady = track->framesReady();
         if ((framesReady >= minFrames) && track->isReady() &&
                 !track->isPaused() && !track->isTerminated())
         {
@@ -3168,9 +3213,9 @@
             mAudioMixer->setBufferProvider(name, track);
             mAudioMixer->enable(name);
 
-            mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl);
-            mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr);
-            mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va);
+            mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)(uintptr_t)vl);
+            mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)(uintptr_t)vr);
+            mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)(uintptr_t)va);
             mAudioMixer->setParameter(
                 name,
                 AudioMixer::TRACK,
@@ -3178,7 +3223,7 @@
             mAudioMixer->setParameter(
                 name,
                 AudioMixer::TRACK,
-                AudioMixer::CHANNEL_MASK, (void *)track->channelMask());
+                AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask());
             // limit track sample rate to 2 x output sample rate, which changes at re-configuration
             uint32_t maxSampleRate = mSampleRate * 2;
             uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate();
@@ -3191,7 +3236,7 @@
                 name,
                 AudioMixer::RESAMPLE,
                 AudioMixer::SAMPLE_RATE,
-                (void *)reqSampleRate);
+                (void *)(uintptr_t)reqSampleRate);
             mAudioMixer->setParameter(
                 name,
                 AudioMixer::TRACK,
@@ -3485,9 +3530,7 @@
 
     PlaybackThread::dumpInternals(fd, args);
 
-    snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames());
-    result.append(buffer);
-    write(fd, result.string(), result.size());
+    fdprintf(fd, "  AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames());
 
     // Make a non-atomic copy of fast mixer dump state so it won't change underneath us
     const FastMixerDumpState copy(mFastMixerDumpState);
@@ -3741,14 +3784,14 @@
 }
 
 // getTrackName_l() must be called with ThreadBase::mLock held
-int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask,
-        int sessionId)
+int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused,
+        int sessionId __unused)
 {
     return 0;
 }
 
 // deleteTrackName_l() must be called with ThreadBase::mLock held
-void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name)
+void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused)
 {
 }
 
@@ -4001,6 +4044,17 @@
         sp<Track> l = mLatestActiveTrack.promote();
         bool last = l.get() == track;
 
+        if (track->isInvalid()) {
+            ALOGW("An invalidated track shouldn't be in active list");
+            tracksToRemove->add(track);
+            continue;
+        }
+
+        if (track->mState == TrackBase::IDLE) {
+            ALOGW("An idle track shouldn't be in active list");
+            continue;
+        }
+
         if (track->isPausing()) {
             track->setPaused();
             if (last) {
@@ -4019,6 +4073,11 @@
                 mBytesRemaining = 0;    // stop writing
             }
             tracksToRemove->add(track);
+        } else if (track->isFlushPending()) {
+            track->flushAck();
+            if (last) {
+                mFlushPending = true;
+            }
         } else if (track->framesReady() && track->isReady() &&
                 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) {
             ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer);
@@ -4068,7 +4127,6 @@
                         // seek when resuming.
                         if (previousTrack->sessionId() != track->sessionId()) {
                             previousTrack->invalidate();
-                            mFlushPending = true;
                         }
                     }
                 }
@@ -4144,9 +4202,6 @@
     // if resume is received before pause is executed.
     if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) {
         mOutput->stream->pause(mOutput->stream);
-        if (!doHwPause) {
-            doHwResume = true;
-        }
     }
     if (mFlushPending) {
         flushHw_l();
@@ -4162,11 +4217,6 @@
     return mixerStatus;
 }
 
-void AudioFlinger::OffloadThread::flushOutput_l()
-{
-    mFlushPending = true;
-}
-
 // must be called with thread mutex locked
 bool AudioFlinger::OffloadThread::waitingAsyncCallback_l()
 {
@@ -4207,6 +4257,8 @@
     mBytesRemaining = 0;
     mPausedWriteLength = 0;
     mPausedBytesRemaining = 0;
+    mHwPaused = false;
+
     if (mUseAsyncWrite) {
         // discard any pending drain or write ack by incrementing sequence
         mWriteAckSequence = (mWriteAckSequence + 2) & ~1;
@@ -4217,6 +4269,18 @@
     }
 }
 
+void AudioFlinger::OffloadThread::onAddNewTrack_l()
+{
+    sp<Track> previousTrack = mPreviousTrack.promote();
+    sp<Track> latestTrack = mLatestActiveTrack.promote();
+
+    if (previousTrack != 0 && latestTrack != 0 &&
+        (previousTrack->sessionId() != latestTrack->sessionId())) {
+        mFlushPending = true;
+    }
+    PlaybackThread::onAddNewTrack_l();
+}
+
 // ----------------------------------------------------------------------------
 
 AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger,
@@ -4476,13 +4540,17 @@
 
         { // scope for mLock
             Mutex::Autolock _l(mLock);
-            if (exitPending()) {
-                break;
-            }
+
             processConfigEvents_l();
             // return value 'reconfig' is currently unused
             bool reconfig = checkForNewParameters_l();
 
+            // check exitPending here because checkForNewParameters_l() and
+            // checkForNewParameters_l() can temporarily release mLock
+            if (exitPending()) {
+                break;
+            }
+
             // if no active track(s), then standby and release wakelock
             size_t size = mActiveTracks.size();
             if (size == 0) {
@@ -4606,8 +4674,7 @@
                             readInto = mRsmpInBuffer;
                             mRsmpInIndex = 0;
                         }
-                        mBytesRead = mInput->stream->read(mInput->stream, readInto,
-                                mBufferSize);
+                        mBytesRead = mInput->stream->read(mInput->stream, readInto, mBufferSize);
                         if (mBytesRead <= 0) {
                             // TODO: verify that it's benign to use a stale track state
                             if ((mBytesRead < 0) && (activeTrackState == TrackBase::ACTIVE))
@@ -4778,13 +4845,14 @@
         uint32_t sampleRate,
         audio_format_t format,
         audio_channel_mask_t channelMask,
-        size_t frameCount,
+        size_t *pFrameCount,
         int sessionId,
         int uid,
         IAudioFlinger::track_flags_t *flags,
         pid_t tid,
         status_t *status)
 {
+    size_t frameCount = *pFrameCount;
     sp<RecordTrack> track;
     status_t lStatus;
 
@@ -4793,6 +4861,7 @@
         ALOGE("createRecordTrack_l() audio driver not initialized");
         goto Exit;
     }
+
     // client expresses a preference for FAST, but we get the final say
     if (*flags & IAudioFlinger::TRACK_FAST) {
       if (
@@ -4843,6 +4912,7 @@
         }
       }
     }
+    *pFrameCount = frameCount;
 
     // FIXME use flags and tid similar to createTrack_l()
 
@@ -4855,7 +4925,7 @@
         lStatus = track->initCheck();
         if (lStatus != NO_ERROR) {
             ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus);
-            track.clear();
+            // track must be cleared from the caller as the caller has the AF lock
             goto Exit;
         }
         mTracks.add(track);
@@ -5022,12 +5092,12 @@
     return false;
 }
 
-bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const
+bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const
 {
     return false;
 }
 
-status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event)
+status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused)
 {
 #if 0   // This branch is currently dead code, but is preserved in case it will be needed in future
     if (!isValidSyncEvent(event)) {
@@ -5078,59 +5148,60 @@
 
 void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args)
 {
-    const size_t SIZE = 256;
-    char buffer[SIZE];
-    String8 result;
-
-    snprintf(buffer, SIZE, "\nInput thread %p internals\n", this);
-    result.append(buffer);
+    fdprintf(fd, "\nInput thread %p:\n", this);
 
     if (mActiveTracks.size() > 0) {
-        snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex);
-        result.append(buffer);
-        snprintf(buffer, SIZE, "Buffer size: %u bytes\n", mBufferSize);
-        result.append(buffer);
-        snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL));
-        result.append(buffer);
-        snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount);
-        result.append(buffer);
-        snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate);
-        result.append(buffer);
+        fdprintf(fd, "  In index: %zu\n", mRsmpInIndex);
+        fdprintf(fd, "  Buffer size: %zu bytes\n", mBufferSize);
+        fdprintf(fd, "  Resampling: %d\n", (mResampler != NULL));
+        fdprintf(fd, "  Out channel count: %u\n", mReqChannelCount);
+        fdprintf(fd, "  Out sample rate: %u\n", mReqSampleRate);
     } else {
-        result.append("No active record client\n");
+        fdprintf(fd, "  No active record client\n");
     }
 
-    write(fd, result.string(), result.size());
-
     dumpBase(fd, args);
 }
 
-void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args)
+void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused)
 {
     const size_t SIZE = 256;
     char buffer[SIZE];
     String8 result;
 
-    snprintf(buffer, SIZE, "Input thread %p tracks\n", this);
-    result.append(buffer);
-    RecordTrack::appendDumpHeader(result);
-    for (size_t i = 0; i < mTracks.size(); ++i) {
-        sp<RecordTrack> track = mTracks[i];
-        if (track != 0) {
-            track->dump(buffer, SIZE);
-            result.append(buffer);
+    size_t numtracks = mTracks.size();
+    size_t numactive = mActiveTracks.size();
+    size_t numactiveseen = 0;
+    fdprintf(fd, "  %d Tracks", numtracks);
+    if (numtracks) {
+        fdprintf(fd, " of which %d are active\n", numactive);
+        RecordTrack::appendDumpHeader(result);
+        for (size_t i = 0; i < numtracks ; ++i) {
+            sp<RecordTrack> track = mTracks[i];
+            if (track != 0) {
+                bool active = mActiveTracks.indexOf(track) >= 0;
+                if (active) {
+                    numactiveseen++;
+                }
+                track->dump(buffer, SIZE, active);
+                result.append(buffer);
+            }
         }
+    } else {
+        fdprintf(fd, "\n");
     }
 
-    size_t size = mActiveTracks.size();
-    if (size > 0) {
-        snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this);
+    if (numactiveseen != numactive) {
+        snprintf(buffer, SIZE, "  The following tracks are in the active list but"
+                " not in the track list\n");
         result.append(buffer);
         RecordTrack::appendDumpHeader(result);
-        for (size_t i = 0; i < size; ++i) {
+        for (size_t i = 0; i < numactive; ++i) {
             sp<RecordTrack> track = mActiveTracks[i];
-            track->dump(buffer, SIZE);
-            result.append(buffer);
+            if (mTracks.indexOf(track) < 0) {
+                track->dump(buffer, SIZE, true);
+                result.append(buffer);
+            }
         }
 
     }
@@ -5138,7 +5209,7 @@
 }
 
 // AudioBufferProvider interface
-status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts)
+status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts __unused)
 {
     int32_t rear = mRsmpInRear;
     int32_t front = mRsmpInFront;
@@ -5317,7 +5388,7 @@
     return out_s8;
 }
 
-void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) {
+void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param __unused) {
     AudioSystem::OutputDescriptor desc;
     const void *param2 = NULL;
 
@@ -5353,7 +5424,7 @@
     mChannelCount = popcount(mChannelMask);
     mFormat = mInput->stream->common.get_format(&mInput->stream->common);
     if (mFormat != AUDIO_FORMAT_PCM_16_BIT) {
-        ALOGE("HAL format %d not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat);
+        ALOGE("HAL format %#x not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat);
     }
     mFrameSize = audio_stream_frame_size(&mInput->stream->common);
     mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common);
@@ -5378,7 +5449,7 @@
     mRsmpInIndex = mFrameCount;
 }
 
-unsigned int AudioFlinger::RecordThread::getInputFramesLost()
+uint32_t AudioFlinger::RecordThread::getInputFramesLost()
 {
     Mutex::Autolock _l(mLock);
     if (initCheck() != NO_ERROR) {
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 6b81c38..999fea3 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -65,7 +65,7 @@
     class IoConfigEvent : public ConfigEvent {
     public:
         IoConfigEvent(int event, int param) :
-            ConfigEvent(CFG_EVENT_IO), mEvent(event), mParam(event) {}
+            ConfigEvent(CFG_EVENT_IO), mEvent(event), mParam(param) {}
         virtual ~IoConfigEvent() {}
 
                 int event() const { return mEvent; }
@@ -201,13 +201,13 @@
                 // effect
                 void removeEffect_l(const sp< EffectModule>& effect);
                 // detach all tracks connected to an auxiliary effect
-    virtual     void detachAuxEffect_l(int effectId) {}
+    virtual     void detachAuxEffect_l(int effectId __unused) {}
                 // returns either EFFECT_SESSION if effects on this audio session exist in one
                 // chain, or TRACK_SESSION if tracks on this audio session exist, or both
                 virtual uint32_t hasAudioSession(int sessionId) const = 0;
                 // the value returned by default implementation is not important as the
                 // strategy is only meaningful for PlaybackThread which implements this method
-                virtual uint32_t getStrategyForSession_l(int sessionId) { return 0; }
+                virtual uint32_t getStrategyForSession_l(int sessionId __unused) { return 0; }
 
                 // suspend or restore effect according to the type of effect passed. a NULL
                 // type pointer means suspend all effects in the session
@@ -307,12 +307,12 @@
                 Vector<ConfigEvent *>     mConfigEvents;
 
                 // These fields are written and read by thread itself without lock or barrier,
-                // and read by other threads without lock or barrier via standby() , outDevice()
+                // and read by other threads without lock or barrier via standby(), outDevice()
                 // and inDevice().
                 // Because of the absence of a lock or barrier, any other thread that reads
                 // these fields must use the information in isolation, or be prepared to deal
                 // with possibility that it might be inconsistent with other information.
-                bool                    mStandby;   // Whether thread is currently in standby.
+                bool                    mStandby;     // Whether thread is currently in standby.
                 audio_devices_t         mOutDevice;   // output device
                 audio_devices_t         mInDevice;    // input device
                 audio_source_t          mAudioSource; // (see audio.h, audio_source_t)
@@ -394,7 +394,7 @@
     virtual     bool        waitingAsyncCallback();
     virtual     bool        waitingAsyncCallback_l();
     virtual     bool        shouldStandby_l();
-
+    virtual     void        onAddNewTrack_l();
 
     // ThreadBase virtuals
     virtual     void        preExit();
@@ -422,7 +422,7 @@
                                 uint32_t sampleRate,
                                 audio_format_t format,
                                 audio_channel_mask_t channelMask,
-                                size_t frameCount,
+                                size_t *pFrameCount,
                                 const sp<IMemory>& sharedBuffer,
                                 int sessionId,
                                 IAudioFlinger::track_flags_t *flags,
@@ -449,7 +449,7 @@
 
     virtual     String8     getParameters(const String8& keys);
     virtual     void        audioConfigChanged_l(int event, int param = 0);
-                status_t    getRenderPosition(size_t *halFrames, size_t *dspFrames);
+                status_t    getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames);
                 int16_t     *mixBuffer() const { return mMixBuffer; };
 
     virtual     void detachAuxEffect_l(int effectId);
@@ -625,13 +625,12 @@
     sp<NBLog::Writer>       mFastMixerNBLogWriter;
 public:
     virtual     bool        hasFastMixer() const = 0;
-    virtual     FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const
+    virtual     FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex __unused) const
                                 { FastTrackUnderruns dummy; return dummy; }
 
 protected:
                 // accessed by both binder threads and within threadLoop(), lock on mutex needed
                 unsigned    mFastTrackAvailMask;    // bit i set if fast track [i] is available
-    virtual     void        flushOutput_l();
 
 private:
     // timestamp latch:
@@ -750,11 +749,11 @@
     // threadLoop snippets
     virtual     mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove);
     virtual     void        threadLoop_exit();
-    virtual     void        flushOutput_l();
 
     virtual     bool        waitingAsyncCallback();
     virtual     bool        waitingAsyncCallback_l();
     virtual     bool        shouldStandby_l();
+    virtual     void        onAddNewTrack_l();
 
 private:
                 void        flushHw_l();
@@ -880,7 +879,7 @@
                     uint32_t sampleRate,
                     audio_format_t format,
                     audio_channel_mask_t channelMask,
-                    size_t frameCount,
+                    size_t *pFrameCount,
                     int sessionId,
                     int uid,
                     IAudioFlinger::track_flags_t *flags,
@@ -907,7 +906,7 @@
     virtual String8     getParameters(const String8& keys);
     virtual void        audioConfigChanged_l(int event, int param = 0);
             void        readInputParameters();
-    virtual unsigned int  getInputFramesLost();
+    virtual uint32_t    getInputFramesLost();
 
     virtual status_t addEffectChain_l(const sp<EffectChain>& chain);
     virtual size_t removeEffectChain_l(const sp<EffectChain>& chain);
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 53196c8..e5152b8 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -133,7 +133,6 @@
     if (mCblk != NULL) {
         new(mCblk) audio_track_cblk_t();
         // clear all buffers
-        mCblk->frameCount_ = frameCount;
         if (sharedBuffer == 0) {
             mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
             memset(mBuffer, 0, bufferSize);
@@ -147,7 +146,7 @@
 #ifdef TEE_SINK
         if (mTeeSinkTrackEnabled) {
             NBAIO_Format pipeFormat = Format_from_SR_C(mSampleRate, mChannelCount);
-            if (pipeFormat != Format_Invalid) {
+            if (Format_isValid(pipeFormat)) {
                 Pipe *pipe = new Pipe(mTeeSinkTrackFrames, pipeFormat);
                 size_t numCounterOffers = 0;
                 const NBAIO_Format offers[1] = {pipeFormat};
@@ -348,7 +347,8 @@
     mCachedVolume(1.0),
     mIsInvalid(false),
     mAudioTrackServerProxy(NULL),
-    mResumeToStopping(false)
+    mResumeToStopping(false),
+    mFlushHwPending(false)
 {
     if (mCblk != NULL) {
         if (sharedBuffer == 0) {
@@ -435,17 +435,19 @@
 
 /*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result)
 {
-    result.append("   Name Client Type      Fmt Chn mask Session fCount S F SRate  "
+    result.append("    Name Active Client Type      Fmt Chn mask Session fCount S F SRate  "
                   "L dB  R dB    Server Main buf  Aux Buf Flags UndFrmCnt\n");
 }
 
-void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size)
+void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size, bool active)
 {
     uint32_t vlr = mAudioTrackServerProxy->getVolumeLR();
     if (isFastTrack()) {
-        sprintf(buffer, "   F %2d", mFastIndex);
+        sprintf(buffer, "    F %2d", mFastIndex);
+    } else if (mName >= AudioMixer::TRACK0) {
+        sprintf(buffer, "    %4d", mName - AudioMixer::TRACK0);
     } else {
-        sprintf(buffer, "   %4d", mName - AudioMixer::TRACK0);
+        sprintf(buffer, "    none");
     }
     track_state state = mState;
     char stateChar;
@@ -500,8 +502,9 @@
         nowInUnderrun = '?';
         break;
     }
-    snprintf(&buffer[7], size-7, " %6u %4u %08X %08X %7u %6u %1c %1d %5u %5.2g %5.2g  "
-                                 "%08X %08X %08X 0x%03X %9u%c\n",
+    snprintf(&buffer[8], size-8, " %6s %6u %4u %08X %08X %7u %6zu %1c %1d %5u %5.2g %5.2g  "
+                                 "%08X %p %p 0x%03X %9u%c\n",
+            active ? "yes" : "no",
             (mClient == 0) ? getpid_cached : mClient->pid(),
             mStreamType,
             mFormat,
@@ -514,8 +517,8 @@
             20.0 * log10((vlr & 0xFFFF) / 4096.0),
             20.0 * log10((vlr >> 16) / 4096.0),
             mCblk->mServer,
-            (int)mMainBuffer,
-            (int)mAuxBuffer,
+            mMainBuffer,
+            mAuxBuffer,
             mCblk->mFlags,
             mAudioTrackServerProxy->getUnderrunFrames(),
             nowInUnderrun);
@@ -527,7 +530,7 @@
 
 // AudioBufferProvider interface
 status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(
-        AudioBufferProvider::Buffer* buffer, int64_t pts)
+        AudioBufferProvider::Buffer* buffer, int64_t pts __unused)
 {
     ServerProxy::Buffer buf;
     size_t desiredFrames = buffer->frameCount;
@@ -564,7 +567,7 @@
 
 // Don't call for fast tracks; the framesReady() could result in priority inversion
 bool AudioFlinger::PlaybackThread::Track::isReady() const {
-    if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) {
+    if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing() || isStopping()) {
         return true;
     }
 
@@ -577,8 +580,8 @@
     return false;
 }
 
-status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event,
-                                                    int triggerSession)
+status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event __unused,
+                                                    int triggerSession __unused)
 {
     status_t status = NO_ERROR;
     ALOGV("start(%d), calling pid %d session %d",
@@ -732,6 +735,7 @@
                 mRetryCount = PlaybackThread::kMaxTrackRetriesOffload;
             }
 
+            mFlushHwPending = true;
             mResumeToStopping = false;
         } else {
             if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED &&
@@ -752,11 +756,19 @@
         // Prevent flush being lost if the track is flushed and then resumed
         // before mixer thread can run. This is important when offloading
         // because the hardware buffer could hold a large amount of audio
-        playbackThread->flushOutput_l();
         playbackThread->broadcast_l();
     }
 }
 
+// must be called with thread lock held
+void AudioFlinger::PlaybackThread::Track::flushAck()
+{
+    if (!isOffloaded())
+        return;
+
+    mFlushHwPending = false;
+}
+
 void AudioFlinger::PlaybackThread::Track::reset()
 {
     // Do not reset twice to avoid discarding data written just after a flush and before
@@ -992,7 +1004,8 @@
             size_t frameCount,
             const sp<IMemory>& sharedBuffer,
             int sessionId,
-            int uid) {
+            int uid)
+{
     if (!client->reserveTimedTrack())
         return 0;
 
@@ -1164,7 +1177,7 @@
 
 void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l(
         const TimedBuffer& buf,
-        const char* logTag) {
+        const char* logTag __unused) {
     uint32_t bufBytes        = buf.buffer()->size();
     uint32_t consumedAlready = buf.position();
 
@@ -1516,9 +1529,9 @@
         mOutBuffer.frameCount = 0;
         playbackThread->mTracks.add(this);
         ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, "
-                "mCblk->frameCount_ %u, mChannelMask 0x%08x",
+                "frameCount %u, mChannelMask 0x%08x",
                 mCblk, mBuffer,
-                mCblk->frameCount_, mChannelMask);
+                frameCount, mChannelMask);
         // since client and server are in the same process,
         // the buffer has the same virtual address on both sides
         mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize);
@@ -1787,7 +1800,7 @@
 
 // AudioBufferProvider interface
 status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer,
-        int64_t pts)
+        int64_t pts __unused)
 {
     ServerProxy::Buffer buf;
     buf.mFrameCount = buffer->frameCount;
@@ -1855,12 +1868,13 @@
 
 /*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result)
 {
-    result.append("Client Fmt Chn mask Session S   Server fCount\n");
+    result.append("    Active Client Fmt Chn mask Session S   Server fCount\n");
 }
 
-void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size)
+void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size, bool active)
 {
-    snprintf(buffer, size, "%6u %3u %08X %7u %1d %08X %6u\n",
+    snprintf(buffer, size, "    %6s %6u %3u %08X %7u %1d %08X %6zu\n",
+            active ? "yes" : "no",
             (mClient == 0) ? getpid_cached : mClient->pid(),
             mFormat,
             mChannelMask,
diff --git a/services/audioflinger/test-resample.cpp b/services/audioflinger/test-resample.cpp
index 3f2ce55..66fcd90 100644
--- a/services/audioflinger/test-resample.cpp
+++ b/services/audioflinger/test-resample.cpp
@@ -57,7 +57,8 @@
 int main(int argc, char* argv[]) {
 
     const char* const progname = argv[0];
-    bool profiling = false;
+    bool profileResample = false;
+    bool profileFilter = false;
     bool writeHeader = false;
     int channels = 1;
     int input_freq = 0;
@@ -65,10 +66,13 @@
     AudioResampler::src_quality quality = AudioResampler::DEFAULT_QUALITY;
 
     int ch;
-    while ((ch = getopt(argc, argv, "phvsq:i:o:")) != -1) {
+    while ((ch = getopt(argc, argv, "pfhvsq:i:o:")) != -1) {
         switch (ch) {
         case 'p':
-            profiling = true;
+            profileResample = true;
+            break;
+        case 'f':
+            profileFilter = true;
             break;
         case 'h':
             writeHeader = true;
@@ -230,27 +234,108 @@
     size_t output_size = 2 * 4 * ((int64_t) input_frames * output_freq) / input_freq;
     output_size &= ~7; // always stereo, 32-bits
 
-    void* output_vaddr = malloc(output_size);
-    AudioResampler* resampler = AudioResampler::create(16, channels,
-            output_freq, quality);
-    size_t out_frames = output_size/8;
-    resampler->setSampleRate(input_freq);
-    resampler->setVolume(0x1000, 0x1000);
-
-    if (profiling) {
-        const int looplimit = 100;
+    if (profileFilter) {
+        // Check how fast sample rate changes are that require filter changes.
+        // The delta sample rate changes must indicate a downsampling ratio,
+        // and must be larger than 10% changes.
+        //
+        // On fast devices, filters should be generated between 0.1ms - 1ms.
+        // (single threaded).
+        AudioResampler* resampler = AudioResampler::create(16, channels,
+                8000, quality);
+        int looplimit = 100;
         timespec start, end;
         clock_gettime(CLOCK_MONOTONIC, &start);
         for (int i = 0; i < looplimit; ++i) {
-            resampler->resample((int*) output_vaddr, out_frames, &provider);
-            provider.reset(); //  reset only provider as benchmarking
+            resampler->setSampleRate(9000);
+            resampler->setSampleRate(12000);
+            resampler->setSampleRate(20000);
+            resampler->setSampleRate(30000);
         }
         clock_gettime(CLOCK_MONOTONIC, &end);
         int64_t start_ns = start.tv_sec * 1000000000LL + start.tv_nsec;
         int64_t end_ns = end.tv_sec * 1000000000LL + end.tv_nsec;
         int64_t time = end_ns - start_ns;
-        printf("time(ns):%lld  channels:%d  quality:%d\n", time, channels, quality);
-        printf("%f Mspl/s\n", out_frames * looplimit / (time / 1e9) / 1e6);
+        printf("%.2f sample rate changes with filter calculation/sec\n",
+                looplimit * 4 / (time / 1e9));
+
+        // Check how fast sample rate changes are without filter changes.
+        // This should be very fast, probably 0.1us - 1us per sample rate
+        // change.
+        resampler->setSampleRate(1000);
+        looplimit = 1000;
+        clock_gettime(CLOCK_MONOTONIC, &start);
+        for (int i = 0; i < looplimit; ++i) {
+            resampler->setSampleRate(1000+i);
+        }
+        clock_gettime(CLOCK_MONOTONIC, &end);
+        start_ns = start.tv_sec * 1000000000LL + start.tv_nsec;
+        end_ns = end.tv_sec * 1000000000LL + end.tv_nsec;
+        time = end_ns - start_ns;
+        printf("%.2f sample rate changes without filter calculation/sec\n",
+                looplimit / (time / 1e9));
+        resampler->reset();
+        delete resampler;
+    }
+
+    void* output_vaddr = malloc(output_size);
+    AudioResampler* resampler = AudioResampler::create(16, channels,
+            output_freq, quality);
+    size_t out_frames = output_size/8;
+
+    /* set volume precision to 12 bits, so the volume scale is 1<<12.
+     * This means the "integer" part fits in the Q19.12 precision
+     * representation of output int32_t.
+     *
+     * Generally 0 < volumePrecision <= 14 (due to the limits of
+     * int16_t values for Volume). volumePrecision cannot be 0 due
+     * to rounding and shifts.
+     */
+    const int volumePrecision = 12; // in bits
+
+    resampler->setSampleRate(input_freq);
+    resampler->setVolume(1 << volumePrecision, 1 << volumePrecision);
+
+    if (profileResample) {
+        /*
+         * For profiling on mobile devices, upon experimentation
+         * it is better to run a few trials with a shorter loop limit,
+         * and take the minimum time.
+         *
+         * Long tests can cause CPU temperature to build up and thermal throttling
+         * to reduce CPU frequency.
+         *
+         * For frequency checks (index=0, or 1, etc.):
+         * "cat /sys/devices/system/cpu/cpu${index}/cpufreq/scaling_*_freq"
+         *
+         * For temperature checks (index=0, or 1, etc.):
+         * "cat /sys/class/thermal/thermal_zone${index}/temp"
+         *
+         * Another way to avoid thermal throttling is to fix the CPU frequency
+         * at a lower level which prevents excessive temperatures.
+         */
+        const int trials = 4;
+        const int looplimit = 4;
+        timespec start, end;
+        int64_t time;
+
+        for (int n = 0; n < trials; ++n) {
+            clock_gettime(CLOCK_MONOTONIC, &start);
+            for (int i = 0; i < looplimit; ++i) {
+                resampler->resample((int*) output_vaddr, out_frames, &provider);
+                provider.reset(); //  during benchmarking reset only the provider
+            }
+            clock_gettime(CLOCK_MONOTONIC, &end);
+            int64_t start_ns = start.tv_sec * 1000000000LL + start.tv_nsec;
+            int64_t end_ns = end.tv_sec * 1000000000LL + end.tv_nsec;
+            int64_t diff_ns = end_ns - start_ns;
+            if (n == 0 || diff_ns < time) {
+                time = diff_ns;   // save the best out of our trials.
+            }
+        }
+        // Mfrms/s is "Millions of output frames per second".
+        printf("quality: %d  channels: %d  msec: %lld  Mfrms/s: %.2lf\n",
+                quality, channels, time/1000000, out_frames * looplimit / (time / 1e9) / 1e6);
         resampler->reset();
     }
 
@@ -266,19 +351,31 @@
     if (gVerbose) {
         printf("reset() complete\n");
     }
+    delete resampler;
+    resampler = NULL;
 
     // mono takes left channel only
     // stereo right channel is half amplitude of stereo left channel (due to input creation)
     int32_t* out = (int32_t*) output_vaddr;
     int16_t* convert = (int16_t*) malloc(out_frames * channels * sizeof(int16_t));
 
+    // round to half towards zero and saturate at int16 (non-dithered)
+    const int roundVal = (1<<(volumePrecision-1)) - 1; // volumePrecision > 0
+
     for (size_t i = 0; i < out_frames; i++) {
         for (int j = 0; j < channels; j++) {
-            int32_t s = out[i * 2 + j] >> 12;
-            if (s > 32767)
-                s = 32767;
-            else if (s < -32768)
-                s = -32768;
+            int32_t s = out[i * 2 + j] + roundVal; // add offset here
+            if (s < 0) {
+                s = (s + 1) >> volumePrecision; // round to 0
+                if (s < -32768) {
+                    s = -32768;
+                }
+            } else {
+                s = s >> volumePrecision;
+                if (s > 32767) {
+                    s = 32767;
+                }
+            }
             convert[i * channels + j] = int16_t(s);
         }
     }
diff --git a/services/camera/libcameraservice/CameraService.cpp b/services/camera/libcameraservice/CameraService.cpp
index eeedfc9..79fbf76 100644
--- a/services/camera/libcameraservice/CameraService.cpp
+++ b/services/camera/libcameraservice/CameraService.cpp
@@ -32,6 +32,7 @@
 #include <gui/Surface.h>
 #include <hardware/hardware.h>
 #include <media/AudioSystem.h>
+#include <media/IMediaHTTPService.h>
 #include <media/mediaplayer.h>
 #include <utils/Errors.h>
 #include <utils/Log.h>
@@ -866,7 +867,7 @@
 
 MediaPlayer* CameraService::newMediaPlayer(const char *file) {
     MediaPlayer* mp = new MediaPlayer();
-    if (mp->setDataSource(file, NULL) == NO_ERROR) {
+    if (mp->setDataSource(NULL /* httpService */, file, NULL) == NO_ERROR) {
         mp->setAudioStreamType(AUDIO_STREAM_ENFORCED_AUDIBLE);
         mp->prepare();
     } else {
@@ -1042,13 +1043,13 @@
 // ----------------------------------------------------------------------------
 
 Mutex* CameraService::Client::getClientLockFromCookie(void* user) {
-    return gCameraService->getClientLockById((int) user);
+    return gCameraService->getClientLockById((int)(intptr_t) user);
 }
 
 // Provide client pointer for callbacks. Client lock returned from getClientLockFromCookie should
 // be acquired for this to be safe
 CameraService::Client* CameraService::Client::getClientFromCookie(void* user) {
-    BasicClient *basicClient = gCameraService->getClientByIdUnsafe((int) user);
+    BasicClient *basicClient = gCameraService->getClientByIdUnsafe((int)(intptr_t) user);
     // OK: only CameraClient calls this, and they already cast anyway.
     Client* client = static_cast<Client*>(basicClient);
 
diff --git a/services/camera/libcameraservice/api1/Camera2Client.cpp b/services/camera/libcameraservice/api1/Camera2Client.cpp
index 0b6ca5c..0a88a75 100644
--- a/services/camera/libcameraservice/api1/Camera2Client.cpp
+++ b/services/camera/libcameraservice/api1/Camera2Client.cpp
@@ -18,6 +18,7 @@
 #define ATRACE_TAG ATRACE_TAG_CAMERA
 //#define LOG_NDEBUG 0
 
+#include <inttypes.h>
 #include <utils/Log.h>
 #include <utils/Trace.h>
 
@@ -193,7 +194,7 @@
         result.appendFormat("    GPS lat x long x alt: %f x %f x %f\n",
                 p.gpsCoordinates[0], p.gpsCoordinates[1],
                 p.gpsCoordinates[2]);
-        result.appendFormat("    GPS timestamp: %lld\n",
+        result.appendFormat("    GPS timestamp: %" PRId64 "\n",
                 p.gpsTimestamp);
         result.appendFormat("    GPS processing method: %s\n",
                 p.gpsProcessingMethod.string());
@@ -237,7 +238,7 @@
 
     result.append("    Scene mode: ");
     switch (p.sceneMode) {
-        case ANDROID_CONTROL_SCENE_MODE_UNSUPPORTED:
+        case ANDROID_CONTROL_SCENE_MODE_DISABLED:
             result.append("AUTO\n"); break;
         CASE_APPEND_ENUM(ANDROID_CONTROL_SCENE_MODE_ACTION)
         CASE_APPEND_ENUM(ANDROID_CONTROL_SCENE_MODE_PORTRAIT)
@@ -731,6 +732,7 @@
         return OK;
     }
     params.state = Parameters::STOPPED;
+    int lastPreviewStreamId = mStreamingProcessor->getPreviewStreamId();
 
     res = mStreamingProcessor->updatePreviewStream(params);
     if (res != OK) {
@@ -739,6 +741,8 @@
         return res;
     }
 
+    bool previewStreamChanged = mStreamingProcessor->getPreviewStreamId() != lastPreviewStreamId;
+
     // We could wait to create the JPEG output stream until first actual use
     // (first takePicture call). However, this would substantially increase the
     // first capture latency on HAL3 devices, and potentially on some HAL2
@@ -788,6 +792,19 @@
             return res;
         }
         outputStreams.push(getCallbackStreamId());
+    } else if (previewStreamChanged && mCallbackProcessor->getStreamId() != NO_STREAM) {
+        /**
+         * Delete the unused callback stream when preview stream is changed and
+         * preview is not enabled. Don't need stop preview stream as preview is in
+         * STOPPED state now.
+         */
+        ALOGV("%s: Camera %d: Delete unused preview callback stream.",  __FUNCTION__, mCameraId);
+        res = mCallbackProcessor->deleteStream();
+        if (res != OK) {
+            ALOGE("%s: Camera %d: Unable to delete callback stream %s (%d)",
+                    __FUNCTION__, mCameraId, strerror(-res), res);
+            return res;
+        }
     }
     if (params.zslMode && !params.recordingHint) {
         res = updateProcessorStream(mZslProcessor, params);
@@ -797,6 +814,8 @@
             return res;
         }
         outputStreams.push(getZslStreamId());
+    } else {
+        mZslProcessor->deleteStream();
     }
 
     outputStreams.push(getPreviewStreamId());
@@ -1143,7 +1162,7 @@
          * Handle quirk mode for AF in scene modes
          */
         if (l.mParameters.quirks.triggerAfWithAuto &&
-                l.mParameters.sceneMode != ANDROID_CONTROL_SCENE_MODE_UNSUPPORTED &&
+                l.mParameters.sceneMode != ANDROID_CONTROL_SCENE_MODE_DISABLED &&
                 l.mParameters.focusMode != Parameters::FOCUS_MODE_AUTO &&
                 !l.mParameters.focusingAreas[0].isEmpty()) {
             ALOGV("%s: Quirk: Switching from focusMode %d to AUTO",
diff --git a/services/camera/libcameraservice/api1/CameraClient.cpp b/services/camera/libcameraservice/api1/CameraClient.cpp
index bd6805d..30b7bb8 100644
--- a/services/camera/libcameraservice/api1/CameraClient.cpp
+++ b/services/camera/libcameraservice/api1/CameraClient.cpp
@@ -85,7 +85,7 @@
     mHardware->setCallbacks(notifyCallback,
             dataCallback,
             dataCallbackTimestamp,
-            (void *)mCameraId);
+            (void *)(uintptr_t)mCameraId);
 
     // Enable zoom, error, focus, and metadata messages by default
     enableMsgType(CAMERA_MSG_ERROR | CAMERA_MSG_ZOOM | CAMERA_MSG_FOCUS |
diff --git a/services/camera/libcameraservice/api1/client2/Parameters.cpp b/services/camera/libcameraservice/api1/client2/Parameters.cpp
index 08af566..2cf0d29 100644
--- a/services/camera/libcameraservice/api1/client2/Parameters.cpp
+++ b/services/camera/libcameraservice/api1/client2/Parameters.cpp
@@ -466,7 +466,7 @@
                 supportedAntibanding);
     }
 
-    sceneMode = ANDROID_CONTROL_SCENE_MODE_UNSUPPORTED;
+    sceneMode = ANDROID_CONTROL_SCENE_MODE_DISABLED;
     params.set(CameraParameters::KEY_SCENE_MODE,
             CameraParameters::SCENE_MODE_AUTO);
 
@@ -482,7 +482,7 @@
             if (addComma) supportedSceneModes += ",";
             addComma = true;
             switch (availableSceneModes.data.u8[i]) {
-                case ANDROID_CONTROL_SCENE_MODE_UNSUPPORTED:
+                case ANDROID_CONTROL_SCENE_MODE_DISABLED:
                     noSceneModes = true;
                     break;
                 case ANDROID_CONTROL_SCENE_MODE_FACE_PRIORITY:
@@ -664,13 +664,13 @@
     focusState = ANDROID_CONTROL_AF_STATE_INACTIVE;
     shadowFocusMode = FOCUS_MODE_INVALID;
 
-    camera_metadata_ro_entry_t max3aRegions =
-        staticInfo(ANDROID_CONTROL_MAX_REGIONS, 1, 1);
-    if (!max3aRegions.count) return NO_INIT;
+    camera_metadata_ro_entry_t max3aRegions = staticInfo(ANDROID_CONTROL_MAX_REGIONS,
+            Parameters::NUM_REGION, Parameters::NUM_REGION);
+    if (max3aRegions.count != Parameters::NUM_REGION) return NO_INIT;
 
     int32_t maxNumFocusAreas = 0;
     if (focusMode != Parameters::FOCUS_MODE_FIXED) {
-        maxNumFocusAreas = max3aRegions.data.i32[0];
+        maxNumFocusAreas = max3aRegions.data.i32[Parameters::REGION_AF];
     }
     params.set(CameraParameters::KEY_MAX_NUM_FOCUS_AREAS, maxNumFocusAreas);
     params.set(CameraParameters::KEY_FOCUS_AREAS,
@@ -730,7 +730,7 @@
 
     meteringAreas.add(Parameters::Area(0, 0, 0, 0, 0));
     params.set(CameraParameters::KEY_MAX_NUM_METERING_AREAS,
-            max3aRegions.data.i32[0]);
+            max3aRegions.data.i32[Parameters::REGION_AE]);
     params.set(CameraParameters::KEY_METERING_AREAS,
             "(0,0,0,0,0)");
 
@@ -1446,7 +1446,7 @@
         newParams.get(CameraParameters::KEY_SCENE_MODE) );
     if (validatedParams.sceneMode != sceneMode &&
             validatedParams.sceneMode !=
-            ANDROID_CONTROL_SCENE_MODE_UNSUPPORTED) {
+            ANDROID_CONTROL_SCENE_MODE_DISABLED) {
         camera_metadata_ro_entry_t availableSceneModes =
             staticInfo(ANDROID_CONTROL_AVAILABLE_SCENE_MODES);
         for (i = 0; i < availableSceneModes.count; i++) {
@@ -1461,7 +1461,7 @@
         }
     }
     bool sceneModeSet =
-            validatedParams.sceneMode != ANDROID_CONTROL_SCENE_MODE_UNSUPPORTED;
+            validatedParams.sceneMode != ANDROID_CONTROL_SCENE_MODE_DISABLED;
 
     // FLASH_MODE
     if (sceneModeSet) {
@@ -1591,10 +1591,11 @@
     // FOCUS_AREAS
     res = parseAreas(newParams.get(CameraParameters::KEY_FOCUS_AREAS),
             &validatedParams.focusingAreas);
-    size_t max3aRegions =
-        (size_t)staticInfo(ANDROID_CONTROL_MAX_REGIONS, 1, 1).data.i32[0];
+    size_t maxAfRegions = (size_t)staticInfo(ANDROID_CONTROL_MAX_REGIONS,
+              Parameters::NUM_REGION, Parameters::NUM_REGION).
+              data.i32[Parameters::REGION_AF];
     if (res == OK) res = validateAreas(validatedParams.focusingAreas,
-            max3aRegions, AREA_KIND_FOCUS);
+            maxAfRegions, AREA_KIND_FOCUS);
     if (res != OK) {
         ALOGE("%s: Requested focus areas are malformed: %s",
                 __FUNCTION__, newParams.get(CameraParameters::KEY_FOCUS_AREAS));
@@ -1624,10 +1625,13 @@
         newParams.get(CameraParameters::KEY_AUTO_WHITEBALANCE_LOCK));
 
     // METERING_AREAS
+    size_t maxAeRegions = (size_t)staticInfo(ANDROID_CONTROL_MAX_REGIONS,
+            Parameters::NUM_REGION, Parameters::NUM_REGION).
+            data.i32[Parameters::REGION_AE];
     res = parseAreas(newParams.get(CameraParameters::KEY_METERING_AREAS),
             &validatedParams.meteringAreas);
     if (res == OK) {
-        res = validateAreas(validatedParams.meteringAreas, max3aRegions,
+        res = validateAreas(validatedParams.meteringAreas, maxAeRegions,
                             AREA_KIND_METERING);
     }
     if (res != OK) {
@@ -1776,7 +1780,7 @@
     // (face detection statistics and face priority scene mode). Map from other
     // to the other.
     bool sceneModeActive =
-            sceneMode != (uint8_t)ANDROID_CONTROL_SCENE_MODE_UNSUPPORTED;
+            sceneMode != (uint8_t)ANDROID_CONTROL_SCENE_MODE_DISABLED;
     uint8_t reqControlMode = ANDROID_CONTROL_MODE_AUTO;
     if (enableFaceDetect || sceneModeActive) {
         reqControlMode = ANDROID_CONTROL_MODE_USE_SCENE_MODE;
@@ -1788,7 +1792,7 @@
     uint8_t reqSceneMode =
             sceneModeActive ? sceneMode :
             enableFaceDetect ? (uint8_t)ANDROID_CONTROL_SCENE_MODE_FACE_PRIORITY :
-            (uint8_t)ANDROID_CONTROL_SCENE_MODE_UNSUPPORTED;
+            (uint8_t)ANDROID_CONTROL_SCENE_MODE_DISABLED;
     res = request->update(ANDROID_CONTROL_SCENE_MODE,
             &reqSceneMode, 1);
     if (res != OK) return res;
@@ -1858,23 +1862,23 @@
 
     size_t reqFocusingAreasSize = focusingAreas.size() * 5;
     int32_t *reqFocusingAreas = new int32_t[reqFocusingAreasSize];
-    for (size_t i = 0; i < reqFocusingAreasSize; i += 5) {
-        if (focusingAreas[i].weight != 0) {
+    for (size_t i = 0, j = 0; i < reqFocusingAreasSize; i += 5, j++) {
+        if (focusingAreas[j].weight != 0) {
             reqFocusingAreas[i + 0] =
-                    normalizedXToArray(focusingAreas[i].left);
+                    normalizedXToArray(focusingAreas[j].left);
             reqFocusingAreas[i + 1] =
-                    normalizedYToArray(focusingAreas[i].top);
+                    normalizedYToArray(focusingAreas[j].top);
             reqFocusingAreas[i + 2] =
-                    normalizedXToArray(focusingAreas[i].right);
+                    normalizedXToArray(focusingAreas[j].right);
             reqFocusingAreas[i + 3] =
-                    normalizedYToArray(focusingAreas[i].bottom);
+                    normalizedYToArray(focusingAreas[j].bottom);
         } else {
             reqFocusingAreas[i + 0] = 0;
             reqFocusingAreas[i + 1] = 0;
             reqFocusingAreas[i + 2] = 0;
             reqFocusingAreas[i + 3] = 0;
         }
-        reqFocusingAreas[i + 4] = focusingAreas[i].weight;
+        reqFocusingAreas[i + 4] = focusingAreas[j].weight;
     }
     res = request->update(ANDROID_CONTROL_AF_REGIONS,
             reqFocusingAreas, reqFocusingAreasSize);
@@ -1887,28 +1891,45 @@
 
     size_t reqMeteringAreasSize = meteringAreas.size() * 5;
     int32_t *reqMeteringAreas = new int32_t[reqMeteringAreasSize];
-    for (size_t i = 0; i < reqMeteringAreasSize; i += 5) {
-        if (meteringAreas[i].weight != 0) {
+    for (size_t i = 0, j = 0; i < reqMeteringAreasSize; i += 5, j++) {
+        if (meteringAreas[j].weight != 0) {
             reqMeteringAreas[i + 0] =
-                normalizedXToArray(meteringAreas[i].left);
+                normalizedXToArray(meteringAreas[j].left);
             reqMeteringAreas[i + 1] =
-                normalizedYToArray(meteringAreas[i].top);
+                normalizedYToArray(meteringAreas[j].top);
             reqMeteringAreas[i + 2] =
-                normalizedXToArray(meteringAreas[i].right);
+                normalizedXToArray(meteringAreas[j].right);
             reqMeteringAreas[i + 3] =
-                normalizedYToArray(meteringAreas[i].bottom);
+                normalizedYToArray(meteringAreas[j].bottom);
         } else {
             reqMeteringAreas[i + 0] = 0;
             reqMeteringAreas[i + 1] = 0;
             reqMeteringAreas[i + 2] = 0;
             reqMeteringAreas[i + 3] = 0;
         }
-        reqMeteringAreas[i + 4] = meteringAreas[i].weight;
+        reqMeteringAreas[i + 4] = meteringAreas[j].weight;
     }
     res = request->update(ANDROID_CONTROL_AE_REGIONS,
             reqMeteringAreas, reqMeteringAreasSize);
     if (res != OK) return res;
 
+    // Set awb regions to be the same as the metering regions if allowed
+    size_t maxAwbRegions = (size_t)staticInfo(ANDROID_CONTROL_MAX_REGIONS,
+            Parameters::NUM_REGION, Parameters::NUM_REGION).
+            data.i32[Parameters::REGION_AWB];
+    if (maxAwbRegions > 0) {
+        if (maxAwbRegions >= meteringAreas.size()) {
+            res = request->update(ANDROID_CONTROL_AWB_REGIONS,
+                    reqMeteringAreas, reqMeteringAreasSize);
+        } else {
+            // Ensure the awb regions are zeroed if the region count is too high.
+            int32_t zeroedAwbAreas[5] = {0, 0, 0, 0, 0};
+            res = request->update(ANDROID_CONTROL_AWB_REGIONS,
+                    zeroedAwbAreas, sizeof(zeroedAwbAreas)/sizeof(int32_t));
+        }
+        if (res != OK) return res;
+    }
+
     delete[] reqMeteringAreas;
 
     /* don't include jpeg thumbnail size - it's valid for
@@ -2149,9 +2170,9 @@
 int Parameters::sceneModeStringToEnum(const char *sceneMode) {
     return
         !sceneMode ?
-            ANDROID_CONTROL_SCENE_MODE_UNSUPPORTED :
+            ANDROID_CONTROL_SCENE_MODE_DISABLED :
         !strcmp(sceneMode, CameraParameters::SCENE_MODE_AUTO) ?
-            ANDROID_CONTROL_SCENE_MODE_UNSUPPORTED :
+            ANDROID_CONTROL_SCENE_MODE_DISABLED :
         !strcmp(sceneMode, CameraParameters::SCENE_MODE_ACTION) ?
             ANDROID_CONTROL_SCENE_MODE_ACTION :
         !strcmp(sceneMode, CameraParameters::SCENE_MODE_PORTRAIT) ?
diff --git a/services/camera/libcameraservice/api1/client2/Parameters.h b/services/camera/libcameraservice/api1/client2/Parameters.h
index 32dbd42..60c4687 100644
--- a/services/camera/libcameraservice/api1/client2/Parameters.h
+++ b/services/camera/libcameraservice/api1/client2/Parameters.h
@@ -113,6 +113,14 @@
     bool autoExposureLock;
     bool autoWhiteBalanceLock;
 
+    // 3A region types, for use with ANDROID_CONTROL_MAX_REGIONS
+    enum region_t {
+        REGION_AE = 0,
+        REGION_AWB,
+        REGION_AF,
+        NUM_REGION // Number of region types
+    } region;
+
     Vector<Area> meteringAreas;
 
     int zoom;
diff --git a/services/camera/libcameraservice/api1/client2/ZslProcessor.cpp b/services/camera/libcameraservice/api1/client2/ZslProcessor.cpp
index 4207ba9..453d54c 100644
--- a/services/camera/libcameraservice/api1/client2/ZslProcessor.cpp
+++ b/services/camera/libcameraservice/api1/client2/ZslProcessor.cpp
@@ -540,7 +540,7 @@
             if (entry.count > 0) frameAeState = entry.data.u8[0];
         }
         String8 result =
-                String8::format("   %d: b: %lld\tf: %lld, AE state: %d", i,
+                String8::format("   %zu: b: %lld\tf: %lld, AE state: %d", i,
                         bufferTimestamp, frameTimestamp, frameAeState);
         ALOGV("%s", result.string());
         if (fd != -1) {
diff --git a/services/camera/libcameraservice/api1/client2/ZslProcessor3.cpp b/services/camera/libcameraservice/api1/client2/ZslProcessor3.cpp
index 776ebe2..c1d0496 100644
--- a/services/camera/libcameraservice/api1/client2/ZslProcessor3.cpp
+++ b/services/camera/libcameraservice/api1/client2/ZslProcessor3.cpp
@@ -273,6 +273,15 @@
             return INVALID_OPERATION;
         }
 
+        // Flush device to clear out all in-flight requests pending in HAL.
+        res = client->getCameraDevice()->flush();
+        if (res != OK) {
+            ALOGE("%s: Camera %d: Failed to flush device: "
+                "%s (%d)",
+                __FUNCTION__, client->getCameraId(), strerror(-res), res);
+            return res;
+        }
+
         // Update JPEG settings
         {
             SharedParameters::Lock l(client->getParameters());
@@ -355,7 +364,7 @@
             if (entry.count > 0) frameAeState = entry.data.u8[0];
         }
         String8 result =
-                String8::format("   %d: b: %lld\tf: %lld, AE state: %d", i,
+                String8::format("   %zu: b: %lld\tf: %lld, AE state: %d", i,
                         bufferTimestamp, frameTimestamp, frameAeState);
         ALOGV("%s", result.string());
         if (fd != -1) {
diff --git a/services/camera/libcameraservice/device2/Camera2Device.cpp b/services/camera/libcameraservice/device2/Camera2Device.cpp
index 2bc1a8a..dc97c47 100644
--- a/services/camera/libcameraservice/device2/Camera2Device.cpp
+++ b/services/camera/libcameraservice/device2/Camera2Device.cpp
@@ -25,6 +25,7 @@
 #define ALOGVV(...) ((void)0)
 #endif
 
+#include <inttypes.h>
 #include <utils/Log.h>
 #include <utils/Trace.h>
 #include <utils/Timers.h>
@@ -822,7 +823,7 @@
         result.append("      Stream slot: Empty\n");
         write(fd, result.string(), result.size());
     } else {
-        result.appendFormat("      Stream slot: %d entries\n",
+        result.appendFormat("      Stream slot: %zu entries\n",
                 mStreamSlot.size());
         int i = 0;
         for (List<camera_metadata_t*>::iterator r = mStreamSlot.begin();
@@ -837,7 +838,7 @@
         result = "      Main queue is empty\n";
         write(fd, result.string(), result.size());
     } else {
-        result = String8::format("      Main queue has %d entries:\n",
+        result = String8::format("      Main queue has %zu entries:\n",
                 mEntries.size());
         int i = 0;
         for (List<camera_metadata_t*>::iterator r = mEntries.begin();
@@ -1214,11 +1215,11 @@
     ATRACE_CALL();
     String8 result = String8::format("      Stream %d: %d x %d, format 0x%x\n",
             mId, mWidth, mHeight, mFormat);
-    result.appendFormat("        size %d, usage 0x%x, requested format 0x%x\n",
+    result.appendFormat("        size %zu, usage 0x%x, requested format 0x%x\n",
             mSize, mUsage, mFormatRequested);
     result.appendFormat("        total buffers: %d, dequeued buffers: %d\n",
             mTotalBuffers, mActiveBuffers);
-    result.appendFormat("        frame count: %d, last timestamp %lld\n",
+    result.appendFormat("        frame count: %d, last timestamp %" PRId64 "\n",
             mFrameCount, mLastTimestamp);
     write(fd, result.string(), result.size());
     return OK;
diff --git a/services/camera/libcameraservice/device3/Camera3Device.cpp b/services/camera/libcameraservice/device3/Camera3Device.cpp
index edb77aa..da3e121 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.cpp
+++ b/services/camera/libcameraservice/device3/Camera3Device.cpp
@@ -840,16 +840,20 @@
     }
 
     sp<Camera3StreamInterface> deletedStream;
+    ssize_t outputStreamIdx = mOutputStreams.indexOfKey(id);
     if (mInputStream != NULL && id == mInputStream->getId()) {
         deletedStream = mInputStream;
         mInputStream.clear();
     } else {
-        ssize_t idx = mOutputStreams.indexOfKey(id);
-        if (idx == NAME_NOT_FOUND) {
+        if (outputStreamIdx == NAME_NOT_FOUND) {
             CLOGE("Stream %d does not exist", id);
             return BAD_VALUE;
         }
-        deletedStream = mOutputStreams.editValueAt(idx);
+    }
+
+    // Delete output stream or the output part of a bi-directional stream.
+    if (outputStreamIdx != NAME_NOT_FOUND) {
+        deletedStream = mOutputStreams.editValueAt(outputStreamIdx);
         mOutputStreams.removeItem(id);
     }
 
@@ -1124,7 +1128,14 @@
     Mutex::Autolock l(mLock);
 
     mRequestThread->clear();
-    return mHal3Device->ops->flush(mHal3Device);
+    status_t res;
+    if (mHal3Device->common.version >= CAMERA_DEVICE_API_VERSION_3_1) {
+        res = mHal3Device->ops->flush(mHal3Device);
+    } else {
+        res = waitUntilDrained();
+    }
+
+    return res;
 }
 
 /**
diff --git a/services/camera/libcameraservice/device3/Camera3IOStreamBase.cpp b/services/camera/libcameraservice/device3/Camera3IOStreamBase.cpp
index da51228..42e02d8 100644
--- a/services/camera/libcameraservice/device3/Camera3IOStreamBase.cpp
+++ b/services/camera/libcameraservice/device3/Camera3IOStreamBase.cpp
@@ -70,12 +70,12 @@
     lines.appendFormat("      Dims: %d x %d, format 0x%x\n",
             camera3_stream::width, camera3_stream::height,
             camera3_stream::format);
-    lines.appendFormat("      Max size: %d\n", mMaxSize);
+    lines.appendFormat("      Max size: %zu\n", mMaxSize);
     lines.appendFormat("      Usage: %d, max HAL buffers: %d\n",
             camera3_stream::usage, camera3_stream::max_buffers);
     lines.appendFormat("      Frames produced: %d, last timestamp: %lld ns\n",
             mFrameCount, mLastTimestamp);
-    lines.appendFormat("      Total buffers: %d, currently dequeued: %d\n",
+    lines.appendFormat("      Total buffers: %zu, currently dequeued: %zu\n",
             mTotalBufferCount, mDequeuedBufferCount);
     write(fd, lines.string(), lines.size());
 }
diff --git a/services/camera/libcameraservice/device3/Camera3ZslStream.cpp b/services/camera/libcameraservice/device3/Camera3ZslStream.cpp
index 5f63a6e..e509350 100644
--- a/services/camera/libcameraservice/device3/Camera3ZslStream.cpp
+++ b/services/camera/libcameraservice/device3/Camera3ZslStream.cpp
@@ -271,7 +271,7 @@
     Camera3IOStreamBase::dump(fd, args);
 
     lines = String8();
-    lines.appendFormat("      Input buffers pending: %d, in flight %d\n",
+    lines.appendFormat("      Input buffers pending: %zu, in flight %zu\n",
             mInputBufferQueue.size(), mBuffersInFlight.size());
     write(fd, lines.string(), lines.size());
 }
diff --git a/services/camera/libcameraservice/gui/RingBufferConsumer.cpp b/services/camera/libcameraservice/gui/RingBufferConsumer.cpp
index ebc7ea7..9a6dc28 100644
--- a/services/camera/libcameraservice/gui/RingBufferConsumer.cpp
+++ b/services/camera/libcameraservice/gui/RingBufferConsumer.cpp
@@ -21,11 +21,11 @@
 
 #include <gui/RingBufferConsumer.h>
 
-#define BI_LOGV(x, ...) ALOGV("[%s] "x, mName.string(), ##__VA_ARGS__)
-#define BI_LOGD(x, ...) ALOGD("[%s] "x, mName.string(), ##__VA_ARGS__)
-#define BI_LOGI(x, ...) ALOGI("[%s] "x, mName.string(), ##__VA_ARGS__)
-#define BI_LOGW(x, ...) ALOGW("[%s] "x, mName.string(), ##__VA_ARGS__)
-#define BI_LOGE(x, ...) ALOGE("[%s] "x, mName.string(), ##__VA_ARGS__)
+#define BI_LOGV(x, ...) ALOGV("[%s] " x, mName.string(), ##__VA_ARGS__)
+#define BI_LOGD(x, ...) ALOGD("[%s] " x, mName.string(), ##__VA_ARGS__)
+#define BI_LOGI(x, ...) ALOGI("[%s] " x, mName.string(), ##__VA_ARGS__)
+#define BI_LOGW(x, ...) ALOGW("[%s] " x, mName.string(), ##__VA_ARGS__)
+#define BI_LOGE(x, ...) ALOGE("[%s] " x, mName.string(), ##__VA_ARGS__)
 
 #undef assert
 #define assert(x) ALOG_ASSERT((x), #x)
diff --git a/services/medialog/MediaLogService.cpp b/services/medialog/MediaLogService.cpp
index 683fdf3..0c7fbbd 100644
--- a/services/medialog/MediaLogService.cpp
+++ b/services/medialog/MediaLogService.cpp
@@ -54,7 +54,7 @@
     }
 }
 
-status_t MediaLogService::dump(int fd, const Vector<String16>& args)
+status_t MediaLogService::dump(int fd, const Vector<String16>& args __unused)
 {
     // FIXME merge with similar but not identical code at services/audioflinger/ServiceUtilities.cpp
     static const String16 sDump("android.permission.DUMP");