diff --git a/camera/Camera.cpp b/camera/Camera.cpp
index 8d7a107..c52e581 100644
--- a/camera/Camera.cpp
+++ b/camera/Camera.cpp
@@ -98,9 +98,37 @@
         c->mStatus = NO_ERROR;
         camera = c;
     } else {
-        ALOGW("An error occurred while connecting to camera %d: %s", cameraId,
-                (cs != nullptr) ? "Service not available" : ret.toString8().string());
-        status = -EINVAL;
+        switch(ret.serviceSpecificErrorCode()) {
+            case hardware::ICameraService::ERROR_DISCONNECTED:
+                status = -ENODEV;
+                break;
+            case hardware::ICameraService::ERROR_CAMERA_IN_USE:
+                status = -EBUSY;
+                break;
+            case hardware::ICameraService::ERROR_INVALID_OPERATION:
+                status = -EINVAL;
+                break;
+            case hardware::ICameraService::ERROR_MAX_CAMERAS_IN_USE:
+                status = -EUSERS;
+                break;
+            case hardware::ICameraService::ERROR_ILLEGAL_ARGUMENT:
+                status = BAD_VALUE;
+                break;
+            case hardware::ICameraService::ERROR_DEPRECATED_HAL:
+                status = -EOPNOTSUPP;
+                break;
+            case hardware::ICameraService::ERROR_DISABLED:
+                status = -EACCES;
+                break;
+            case hardware::ICameraService::ERROR_PERMISSION_DENIED:
+                status = PERMISSION_DENIED;
+                break;
+            default:
+                status = -EINVAL;
+                ALOGW("An error occurred while connecting to camera %d: %s", cameraId,
+                        (cs != nullptr) ? "Service not available" : ret.toString8().string());
+                break;
+        }
         c.clear();
     }
     return status;
diff --git a/cmds/stagefright/SineSource.cpp b/cmds/stagefright/SineSource.cpp
index 587077a..cad8caf 100644
--- a/cmds/stagefright/SineSource.cpp
+++ b/cmds/stagefright/SineSource.cpp
@@ -53,6 +53,7 @@
     meta->setInt32(kKeyChannelCount, mNumChannels);
     meta->setInt32(kKeySampleRate, mSampleRate);
     meta->setInt32(kKeyMaxInputSize, kBufferSize);
+    meta->setInt32(kKeyPcmEncoding, kAudioEncodingPcm16bit);
 
     return meta;
 }
diff --git a/include/media/stagefright/ACodec.h b/include/media/stagefright/ACodec.h
index 2365323..bc11da2 100644
--- a/include/media/stagefright/ACodec.h
+++ b/include/media/stagefright/ACodec.h
@@ -25,6 +25,7 @@
 #include <media/stagefright/foundation/AHierarchicalStateMachine.h>
 #include <media/stagefright/CodecBase.h>
 #include <media/stagefright/FrameRenderTracker.h>
+#include <media/stagefright/MediaDefs.h>
 #include <media/stagefright/SkipCutBuffer.h>
 #include <utils/NativeHandle.h>
 #include <OMX_Audio.h>
@@ -36,6 +37,7 @@
 struct ABuffer;
 struct MemoryDealer;
 struct DescribeColorFormat2Params;
+struct DataConverter;
 
 struct ACodec : public AHierarchicalStateMachine, public CodecBase {
     ACodec();
@@ -188,8 +190,11 @@
         Status mStatus;
         unsigned mDequeuedAt;
 
-        sp<ABuffer> mData;
-        sp<RefBase> mMemRef;
+        sp<ABuffer> mData;      // the client's buffer; if not using data conversion, this is the
+                                // codec buffer; otherwise, it is allocated separately
+        sp<RefBase> mMemRef;    // and a reference to the IMemory, so it does not go away
+        sp<ABuffer> mCodecData; // the codec's buffer
+        sp<RefBase> mCodecRef;  // and a reference to the IMemory
         sp<GraphicBuffer> mGraphicBuffer;
         sp<NativeHandle> mNativeHandle;
         int mFenceFd;
@@ -248,6 +253,9 @@
     sp<AMessage> mConfigFormat;
     sp<AMessage> mInputFormat;
     sp<AMessage> mOutputFormat;
+
+    // Initial output format + configuration params that is reused as the base for all subsequent
+    // format updates. This will equal to mOutputFormat until the first actual frame is received.
     sp<AMessage> mBaseOutputFormat;
 
     FrameRenderTracker mRenderTracker; // render information for buffers rendered by ACodec
@@ -280,6 +288,7 @@
     bool mLegacyAdaptiveExperiment;
     int32_t mMetadataBuffersToSubmit;
     size_t mNumUndequeuedBuffers;
+    sp<DataConverter> mConverter[2];
 
     int64_t mRepeatFrameDelayUs;
     int64_t mMaxPtsGapUs;
@@ -441,7 +450,8 @@
             bool encoder, int32_t numChannels, int32_t sampleRate, int32_t compressionLevel);
 
     status_t setupRawAudioFormat(
-            OMX_U32 portIndex, int32_t sampleRate, int32_t numChannels);
+            OMX_U32 portIndex, int32_t sampleRate, int32_t numChannels,
+            AudioEncoding encoding = kAudioEncodingPcm16bit);
 
     status_t setPriority(int32_t priority);
     status_t setOperatingRate(float rateFloat, bool isVideo);
diff --git a/include/media/stagefright/CodecBase.h b/include/media/stagefright/CodecBase.h
index 2d28432..be2835d 100644
--- a/include/media/stagefright/CodecBase.h
+++ b/include/media/stagefright/CodecBase.h
@@ -55,6 +55,10 @@
         kWhatOutputFramesRendered = 'outR',
     };
 
+    enum {
+        kMaxCodecBufferSize = 8192 * 4096 * 4, // 8K RGBA
+    };
+
     virtual void setNotificationMessage(const sp<AMessage> &msg) = 0;
 
     virtual void initiateAllocateComponent(const sp<AMessage> &msg) = 0;
diff --git a/include/media/stagefright/MediaDefs.h b/include/media/stagefright/MediaDefs.h
index e5bcec6..5f2a32d 100644
--- a/include/media/stagefright/MediaDefs.h
+++ b/include/media/stagefright/MediaDefs.h
@@ -68,6 +68,15 @@
 extern const char *MEDIA_MIMETYPE_TEXT_CEA_708;
 extern const char *MEDIA_MIMETYPE_DATA_TIMED_ID3;
 
+// These are values exported to JAVA API that need to be in sync with
+// frameworks/base/media/java/android/media/AudioFormat.java. Unfortunately,
+// they are not defined in frameworks/av, so defining them here.
+enum AudioEncoding {
+    kAudioEncodingPcm16bit = 2,
+    kAudioEncodingPcm8bit = 3,
+    kAudioEncodingPcmFloat = 4,
+};
+
 }  // namespace android
 
 #endif  // MEDIA_DEFS_H_
diff --git a/include/media/stagefright/MetaData.h b/include/media/stagefright/MetaData.h
index 7d2208c..a9ae49b 100644
--- a/include/media/stagefright/MetaData.h
+++ b/include/media/stagefright/MetaData.h
@@ -49,6 +49,7 @@
     kKeyChannelCount      = '#chn',  // int32_t
     kKeyChannelMask       = 'chnm',  // int32_t
     kKeySampleRate        = 'srte',  // int32_t (audio sampling rate Hz)
+    kKeyPcmEncoding       = 'PCMe',  // int32_t (audio encoding enum)
     kKeyFrameRate         = 'frmR',  // int32_t (video frame rate fps)
     kKeyBitRate           = 'brte',  // int32_t (bps)
     kKeyESDS              = 'esds',  // raw data
diff --git a/media/audioserver/Android.mk b/media/audioserver/Android.mk
index aa79bc1..5ce1798 100644
--- a/media/audioserver/Android.mk
+++ b/media/audioserver/Android.mk
@@ -31,9 +31,29 @@
 	$(call include-path-for, audio-utils) \
 	external/sonic \
 
+# If AUDIOSERVER_MULTILIB in device.mk is non-empty then it is used to control
+# the LOCAL_MULTILIB for all audioserver exclusive libraries.
+# This is relevant for 64 bit architectures where either or both
+# 32 and 64 bit libraries may be built.
+#
+# AUDIOSERVER_MULTILIB may be set as follows:
+#   32      to build 32 bit audioserver libraries and 32 bit audioserver.
+#   64      to build 64 bit audioserver libraries and 64 bit audioserver.
+#   both    to build both 32 bit and 64 bit libraries,
+#           and use primary target architecture (32 or 64) for audioserver.
+#   first   to build libraries and audioserver for the primary target architecture only.
+#   <empty> to build both 32 and 64 bit libraries and 32 bit audioserver.
+
+ifeq ($(strip $(AUDIOSERVER_MULTILIB)),)
+LOCAL_MULTILIB := 32
+else
+LOCAL_MULTILIB := $(AUDIOSERVER_MULTILIB)
+endif
+
 LOCAL_MODULE := audioserver
-LOCAL_32_BIT_ONLY := true
 
 LOCAL_INIT_RC := audioserver.rc
 
+LOCAL_CFLAGS := -Werror -Wall
+
 include $(BUILD_EXECUTABLE)
diff --git a/media/libcpustats/Android.mk b/media/libcpustats/Android.mk
index ee283a6..57fe527 100644
--- a/media/libcpustats/Android.mk
+++ b/media/libcpustats/Android.mk
@@ -8,6 +8,6 @@
 
 LOCAL_MODULE := libcpustats
 
-LOCAL_CFLAGS := -std=gnu++11 -Werror
+LOCAL_CFLAGS := -std=gnu++11 -Werror -Wall
 
 include $(BUILD_STATIC_LIBRARY)
diff --git a/media/libmedia/Android.mk b/media/libmedia/Android.mk
index 479ccbb..63f9ed7 100644
--- a/media/libmedia/Android.mk
+++ b/media/libmedia/Android.mk
@@ -7,7 +7,7 @@
 LOCAL_MODULE:= libmedia_helper
 LOCAL_MODULE_TAGS := optional
 
-LOCAL_C_FLAGS += -Werror -Wno-error=deprecated-declarations -Wall
+LOCAL_CFLAGS += -Werror -Wno-error=deprecated-declarations -Wall
 LOCAL_CLANG := true
 
 include $(BUILD_STATIC_LIBRARY)
diff --git a/media/libmedia/AudioTrackShared.cpp b/media/libmedia/AudioTrackShared.cpp
index e5d6fc3..f7baa15 100644
--- a/media/libmedia/AudioTrackShared.cpp
+++ b/media/libmedia/AudioTrackShared.cpp
@@ -88,13 +88,14 @@
 
 uint32_t ClientProxy::setBufferSizeInFrames(uint32_t size)
 {
-    // TODO set minimum to 2X the fast mixer buffer size.
     // The minimum should be  greater than zero and less than the size
     // at which underruns will occur.
-    const uint32_t minimum = 128 * 2; // arbitrary
+    const uint32_t minimum = 16; // based on AudioMixer::BLOCKSIZE
     const uint32_t maximum = frameCount();
     uint32_t clippedSize = size;
-    if (clippedSize < minimum) {
+    if (maximum < minimum) {
+        clippedSize = maximum;
+    } else if (clippedSize < minimum) {
         clippedSize = minimum;
     } else if (clippedSize > maximum) {
         clippedSize = maximum;
diff --git a/media/libmedia/IOMX.cpp b/media/libmedia/IOMX.cpp
index ea4a966..61fba35 100644
--- a/media/libmedia/IOMX.cpp
+++ b/media/libmedia/IOMX.cpp
@@ -718,11 +718,12 @@
             void *params = NULL;
             size_t pageSize = 0;
             size_t allocSize = 0;
-            if ((index == (OMX_INDEXTYPE) OMX_IndexParamConsumerUsageBits && size < 4) ||
-                    (code != SET_INTERNAL_OPTION && size < 8)) {
+            bool isUsageBits = (index == (OMX_INDEXTYPE) OMX_IndexParamConsumerUsageBits);
+            if ((isUsageBits && size < 4) ||
+                    (!isUsageBits && code != SET_INTERNAL_OPTION && size < 8)) {
                 // we expect the structure to contain at least the size and
                 // version, 8 bytes total
-                ALOGE("b/27207275 (%zu)", size);
+                ALOGE("b/27207275 (%zu) (%d/%d)", size, int(index), int(code));
                 android_errorWriteLog(0x534e4554, "27207275");
             } else {
                 err = NO_MEMORY;
diff --git a/media/libmedia/mediaplayer.cpp b/media/libmedia/mediaplayer.cpp
index 53b6df6..25c9fb1 100644
--- a/media/libmedia/mediaplayer.cpp
+++ b/media/libmedia/mediaplayer.cpp
@@ -207,7 +207,7 @@
         ALOGV("invoke %zu", request.dataSize());
         return  mPlayer->invoke(request, reply);
     }
-    ALOGE("invoke failed: wrong state %X", mCurrentState);
+    ALOGE("invoke failed: wrong state %X, mPlayer(%p)", mCurrentState, mPlayer.get());
     return INVALID_OPERATION;
 }
 
@@ -252,7 +252,7 @@
         mCurrentState = MEDIA_PLAYER_PREPARING;
         return mPlayer->prepareAsync();
     }
-    ALOGE("prepareAsync called in state %d", mCurrentState);
+    ALOGE("prepareAsync called in state %d, mPlayer(%p)", mCurrentState, mPlayer.get());
     return INVALID_OPERATION;
 }
 
@@ -318,7 +318,7 @@
             }
         }
     } else {
-        ALOGE("start called in state %d", mCurrentState);
+        ALOGE("start called in state %d, mPlayer(%p)", mCurrentState, mPlayer.get());
         ret = INVALID_OPERATION;
     }
 
@@ -342,7 +342,7 @@
         }
         return ret;
     }
-    ALOGE("stop called in state %d", mCurrentState);
+    ALOGE("stop called in state %d, mPlayer(%p)", mCurrentState, mPlayer.get());
     return INVALID_OPERATION;
 }
 
@@ -361,7 +361,7 @@
         }
         return ret;
     }
-    ALOGE("pause called in state %d", mCurrentState);
+    ALOGE("pause called in state %d, mPlayer(%p)", mCurrentState, mPlayer.get());
     return INVALID_OPERATION;
 }
 
@@ -484,7 +484,8 @@
         }
         return ret;
     }
-    ALOGE("Attempt to call getDuration without a valid mediaplayer");
+    ALOGE("Attempt to call getDuration in wrong state: mPlayer=%p, mCurrentState=%u",
+            mPlayer.get(), mCurrentState);
     return INVALID_OPERATION;
 }
 
@@ -691,7 +692,7 @@
     if (mPlayer == 0 ||
         (mCurrentState & MEDIA_PLAYER_IDLE) ||
         (mCurrentState == MEDIA_PLAYER_STATE_ERROR )) {
-        ALOGE("attachAuxEffect called in state %d", mCurrentState);
+        ALOGE("attachAuxEffect called in state %d, mPlayer(%p)", mCurrentState, mPlayer.get());
         return INVALID_OPERATION;
     }
 
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
index 332fef6..06bb53d 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
@@ -45,8 +45,7 @@
       mPlayerFlags(0),
       mAtEOS(false),
       mLooping(false),
-      mAutoLoop(false),
-      mStartupSeekTimeUs(-1) {
+      mAutoLoop(false) {
     ALOGV("NuPlayerDriver(%p)", this);
     mLooper->setName("NuPlayerDriver Looper");
 
@@ -261,25 +260,11 @@
 
         case STATE_PAUSED:
         case STATE_STOPPED_AND_PREPARED:
-        {
-            if (mAtEOS && mStartupSeekTimeUs < 0) {
-                mStartupSeekTimeUs = 0;
-                mPositionUs = -1;
-            }
-
-            // fall through
-        }
-
         case STATE_PREPARED:
         {
-            mAtEOS = false;
             mPlayer->start();
 
-            if (mStartupSeekTimeUs >= 0) {
-                mPlayer->seekToAsync(mStartupSeekTimeUs);
-                mStartupSeekTimeUs = -1;
-            }
-            break;
+            // fall through
         }
 
         case STATE_RUNNING:
@@ -330,6 +315,7 @@
 }
 
 status_t NuPlayerDriver::pause() {
+    ALOGD("pause(%p)", this);
     // The NuPlayerRenderer may get flushed if pause for long enough, e.g. the pause timeout tear
     // down for audio offload mode. If that happens, the NuPlayerRenderer will no longer know the
     // current position. So similar to seekTo, update |mPositionUs| to the pause position by calling
@@ -400,8 +386,6 @@
         case STATE_PREPARED:
         case STATE_STOPPED_AND_PREPARED:
         case STATE_PAUSED:
-            mStartupSeekTimeUs = seekTimeUs;
-            // fall through.
         case STATE_RUNNING:
         {
             mAtEOS = false;
@@ -502,7 +486,6 @@
 
     mDurationUs = -1;
     mPositionUs = -1;
-    mStartupSeekTimeUs = -1;
     mLooping = false;
 
     return OK;
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.h b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.h
index d009fd7..d5b4ba1 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.h
@@ -122,8 +122,6 @@
     bool mLooping;
     bool mAutoLoop;
 
-    int64_t mStartupSeekTimeUs;
-
     status_t prepare_l();
     void notifyListener_l(int msg, int ext1 = 0, int ext2 = 0, const Parcel *in = NULL);
 
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerStreamListener.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerStreamListener.cpp
index f53afbd..ee70306 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerStreamListener.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerStreamListener.cpp
@@ -144,8 +144,17 @@
         copy = size;
     }
 
+    if (entry->mIndex >= mBuffers.size()) {
+        return ERROR_MALFORMED;
+    }
+
+    sp<IMemory> mem = mBuffers.editItemAt(entry->mIndex);
+    if (mem == NULL || mem->size() < copy || mem->size() - copy < entry->mOffset) {
+        return ERROR_MALFORMED;
+    }
+
     memcpy(data,
-           (const uint8_t *)mBuffers.editItemAt(entry->mIndex)->pointer()
+           (const uint8_t *)mem->pointer()
             + entry->mOffset,
            copy);
 
diff --git a/media/libnbaio/Android.mk b/media/libnbaio/Android.mk
index 16c5040..e2f416b 100644
--- a/media/libnbaio/Android.mk
+++ b/media/libnbaio/Android.mk
@@ -20,9 +20,6 @@
 #LOCAL_C_INCLUDES += path/to/libsndfile/src
 #LOCAL_STATIC_LIBRARIES += libsndfile
 
-# uncomment for systrace
-# LOCAL_CFLAGS += -DATRACE_TAG=ATRACE_TAG_AUDIO
-
 LOCAL_MODULE := libnbaio
 
 LOCAL_SHARED_LIBRARIES := \
@@ -34,4 +31,6 @@
 
 LOCAL_C_INCLUDES := $(call include-path-for, audio-utils)
 
+LOCAL_CFLAGS := -Werror -Wall
+
 include $(BUILD_SHARED_LIBRARY)
diff --git a/media/libnbaio/PipeReader.cpp b/media/libnbaio/PipeReader.cpp
index b096903..a879647 100644
--- a/media/libnbaio/PipeReader.cpp
+++ b/media/libnbaio/PipeReader.cpp
@@ -36,7 +36,12 @@
 
 PipeReader::~PipeReader()
 {
-    int32_t readers = android_atomic_dec(&mPipe.mReaders);
+#if !LOG_NDEBUG
+    int32_t readers =
+#else
+    (void)
+#endif
+            android_atomic_dec(&mPipe.mReaders);
     ALOG_ASSERT(readers > 0);
 }
 
diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index 520f599..87625d5 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -52,6 +52,7 @@
 #include <OMX_AsString.h>
 
 #include "include/avc_utils.h"
+#include "include/DataConverter.h"
 #include "omx/OMXUtils.h"
 
 namespace android {
@@ -114,6 +115,13 @@
     DISALLOW_EVIL_CONSTRUCTORS(MessageList);
 };
 
+static sp<DataConverter> getCopyConverter() {
+    static pthread_once_t once = PTHREAD_ONCE_INIT; // const-inited
+    static sp<DataConverter> sCopyConverter;        // zero-inited
+    pthread_once(&once, [](){ sCopyConverter = new DataConverter(); });
+    return sCopyConverter;
+}
+
 struct CodecObserver : public BnOMXObserver {
     CodecObserver() {}
 
@@ -505,6 +513,7 @@
       mOutputMetadataType(kMetadataBufferTypeInvalid),
       mLegacyAdaptiveExperiment(false),
       mMetadataBuffersToSubmit(0),
+      mNumUndequeuedBuffers(0),
       mRepeatFrameDelayUs(-1ll),
       mMaxPtsGapUs(-1ll),
       mMaxFps(-1),
@@ -781,7 +790,7 @@
         if (err == OK) {
             MetadataBufferType type =
                 portIndex == kPortIndexOutput ? mOutputMetadataType : mInputMetadataType;
-            int32_t bufSize = def.nBufferSize;
+            size_t bufSize = def.nBufferSize;
             if (type == kMetadataBufferTypeGrallocSource) {
                 bufSize = sizeof(VideoGrallocMetadata);
             } else if (type == kMetadataBufferTypeANWBuffer) {
@@ -792,23 +801,47 @@
             // metadata size as we prefer to generate native source metadata, but component
             // may require gralloc source. For camera source, allocate at least enough
             // size for native metadata buffers.
-            int32_t allottedSize = bufSize;
+            size_t allottedSize = bufSize;
             if (portIndex == kPortIndexInput && type >= kMetadataBufferTypeGrallocSource) {
                 bufSize = max(sizeof(VideoGrallocMetadata), sizeof(VideoNativeMetadata));
             } else if (portIndex == kPortIndexInput && type == kMetadataBufferTypeCameraSource) {
-                bufSize = max(bufSize, (int32_t)sizeof(VideoNativeMetadata));
+                bufSize = max(bufSize, sizeof(VideoNativeMetadata));
             }
 
-            ALOGV("[%s] Allocating %u buffers of size %d/%d (from %u using %s) on %s port",
+            size_t conversionBufferSize = 0;
+
+            sp<DataConverter> converter = mConverter[portIndex];
+            if (converter != NULL) {
+                // here we assume sane conversions of max 4:1, so result fits in int32
+                if (portIndex == kPortIndexInput) {
+                    conversionBufferSize = converter->sourceSize(bufSize);
+                } else {
+                    conversionBufferSize = converter->targetSize(bufSize);
+                }
+            }
+
+            size_t alignment = MemoryDealer::getAllocationAlignment();
+
+            ALOGV("[%s] Allocating %u buffers of size %zu/%zu (from %u using %s) on %s port",
                     mComponentName.c_str(),
                     def.nBufferCountActual, bufSize, allottedSize, def.nBufferSize, asString(type),
                     portIndex == kPortIndexInput ? "input" : "output");
 
-            if (bufSize == 0 || def.nBufferCountActual > SIZE_MAX / bufSize) {
+            // verify buffer sizes to avoid overflow in align()
+            if (bufSize == 0 || max(bufSize, conversionBufferSize) > kMaxCodecBufferSize) {
                 ALOGE("b/22885421");
                 return NO_MEMORY;
             }
-            size_t totalSize = def.nBufferCountActual * bufSize;
+
+            // don't modify bufSize as OMX may not expect it to increase after negotiation
+            size_t alignedSize = align(bufSize, alignment);
+            size_t alignedConvSize = align(conversionBufferSize, alignment);
+            if (def.nBufferCountActual > SIZE_MAX / (alignedSize + alignedConvSize)) {
+                ALOGE("b/22885421");
+                return NO_MEMORY;
+            }
+
+            size_t totalSize = def.nBufferCountActual * (alignedSize + alignedConvSize);
             mDealer[portIndex] = new MemoryDealer(totalSize, "ACodec");
 
             for (OMX_U32 i = 0; i < def.nBufferCountActual && err == OK; ++i) {
@@ -847,6 +880,7 @@
                     // because Widevine source only receives these base addresses.
                     info.mData = new ABuffer(ptr != NULL ? ptr : (void *)native_handle, bufSize);
                     info.mNativeHandle = NativeHandle::create(native_handle, true /* ownsHandle */);
+                    info.mCodecData = info.mData;
                 } else if (mQuirks & requiresAllocateBufferBit) {
                     err = mOMX->allocateBufferWithBackup(
                             mNode, portIndex, mem, &info.mBufferID, allottedSize);
@@ -855,11 +889,27 @@
                 }
 
                 if (mem != NULL) {
-                    info.mData = new ABuffer(mem->pointer(), bufSize);
+                    info.mCodecData = new ABuffer(mem->pointer(), bufSize);
+                    info.mCodecRef = mem;
+
                     if (type == kMetadataBufferTypeANWBuffer) {
                         ((VideoNativeMetadata *)mem->pointer())->nFenceFd = -1;
                     }
-                    info.mMemRef = mem;
+
+                    // if we require conversion, allocate conversion buffer for client use;
+                    // otherwise, reuse codec buffer
+                    if (mConverter[portIndex] != NULL) {
+                        CHECK_GT(conversionBufferSize, (size_t)0);
+                        mem = mDealer[portIndex]->allocate(conversionBufferSize);
+                        if (mem == NULL|| mem->pointer() == NULL) {
+                            return NO_MEMORY;
+                        }
+                        info.mData = new ABuffer(mem->pointer(), conversionBufferSize);
+                        info.mMemRef = mem;
+                    } else {
+                        info.mData = info.mCodecData;
+                        info.mMemRef = info.mCodecRef;
+                    }
                 }
 
                 mBuffers[portIndex].push(info);
@@ -1052,6 +1102,7 @@
         info.mIsReadFence = false;
         info.mRenderInfo = NULL;
         info.mData = new ABuffer(NULL /* data */, bufferSize /* capacity */);
+        info.mCodecData = info.mData;
         info.mGraphicBuffer = graphicBuffer;
         mBuffers[kPortIndexOutput].push(info);
 
@@ -1116,7 +1167,7 @@
 
     size_t bufSize = mOutputMetadataType == kMetadataBufferTypeANWBuffer ?
             sizeof(struct VideoNativeMetadata) : sizeof(struct VideoGrallocMetadata);
-    size_t totalSize = bufferCount * bufSize;
+    size_t totalSize = bufferCount * align(bufSize, MemoryDealer::getAllocationAlignment());
     mDealer[kPortIndexOutput] = new MemoryDealer(totalSize, "ACodec");
 
     // Dequeue buffers and send them to OMX
@@ -1136,11 +1187,13 @@
             ((VideoNativeMetadata *)mem->pointer())->nFenceFd = -1;
         }
         info.mData = new ABuffer(mem->pointer(), mem->size());
+        info.mMemRef = mem;
+        info.mCodecData = info.mData;
+        info.mCodecRef = mem;
 
         // we use useBuffer for metadata regardless of quirks
         err = mOMX->useBuffer(
                 mNode, kPortIndexOutput, mem, &info.mBufferID, mem->size());
-        info.mMemRef = mem;
         mBuffers[kPortIndexOutput].push(info);
 
         ALOGV("[%s] allocated meta buffer with ID %u (pointer = %p)",
@@ -1934,6 +1987,10 @@
         }
     }
 
+    AudioEncoding pcmEncoding = kAudioEncodingPcm16bit;
+    (void)msg->findInt32("pcm-encoding", (int32_t*)&pcmEncoding);
+    // invalid encodings will default to PCM-16bit in setupRawAudioFormat.
+
     if (video) {
         // determine need for software renderer
         bool usingSwRenderer = false;
@@ -2138,7 +2195,7 @@
                 || !msg->findInt32("sample-rate", &sampleRate)) {
             err = INVALID_OPERATION;
         } else {
-            err = setupRawAudioFormat(kPortIndexInput, sampleRate, numChannels);
+            err = setupRawAudioFormat(kPortIndexInput, sampleRate, numChannels, pcmEncoding);
         }
     } else if (!strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_AC3)) {
         int32_t numChannels;
@@ -2200,6 +2257,7 @@
         err = setOperatingRate(rateFloat, video);
     }
 
+    // NOTE: both mBaseOutputFormat and mOutputFormat are outputFormat to signal first frame.
     mBaseOutputFormat = outputFormat;
     // trigger a kWhatOutputFormatChanged msg on first buffer
     mLastOutputFormat.clear();
@@ -2212,6 +2270,25 @@
             mOutputFormat = outputFormat;
         }
     }
+
+    // create data converters if needed
+    if (!video && err == OK) {
+        AudioEncoding codecPcmEncoding = kAudioEncodingPcm16bit;
+        if (encoder) {
+            (void)mInputFormat->findInt32("pcm-encoding", (int32_t*)&codecPcmEncoding);
+            mConverter[kPortIndexInput] = AudioConverter::Create(pcmEncoding, codecPcmEncoding);
+            if (mConverter[kPortIndexInput] != NULL) {
+                mInputFormat->setInt32("pcm-encoding", pcmEncoding);
+            }
+        } else {
+            (void)mOutputFormat->findInt32("pcm-encoding", (int32_t*)&codecPcmEncoding);
+            mConverter[kPortIndexOutput] = AudioConverter::Create(codecPcmEncoding, pcmEncoding);
+            if (mConverter[kPortIndexOutput] != NULL) {
+                mOutputFormat->setInt32("pcm-encoding", pcmEncoding);
+            }
+        }
+    }
+
     return err;
 }
 
@@ -2762,7 +2839,7 @@
 }
 
 status_t ACodec::setupRawAudioFormat(
-        OMX_U32 portIndex, int32_t sampleRate, int32_t numChannels) {
+        OMX_U32 portIndex, int32_t sampleRate, int32_t numChannels, AudioEncoding encoding) {
     OMX_PARAM_PORTDEFINITIONTYPE def;
     InitOMXParams(&def);
     def.nPortIndex = portIndex;
@@ -2795,9 +2872,23 @@
     }
 
     pcmParams.nChannels = numChannels;
-    pcmParams.eNumData = OMX_NumericalDataSigned;
+    switch (encoding) {
+        case kAudioEncodingPcm8bit:
+            pcmParams.eNumData = OMX_NumericalDataUnsigned;
+            pcmParams.nBitPerSample = 8;
+            break;
+        case kAudioEncodingPcmFloat:
+            pcmParams.eNumData = OMX_NumericalDataFloat;
+            pcmParams.nBitPerSample = 32;
+            break;
+        case kAudioEncodingPcm16bit:
+            pcmParams.eNumData = OMX_NumericalDataSigned;
+            pcmParams.nBitPerSample = 16;
+            break;
+        default:
+            return BAD_VALUE;
+    }
     pcmParams.bInterleaved = OMX_TRUE;
-    pcmParams.nBitPerSample = 16;
     pcmParams.nSamplingRate = sampleRate;
     pcmParams.ePCMMode = OMX_AUDIO_PCMModeLinear;
 
@@ -2805,8 +2896,17 @@
         return OMX_ErrorNone;
     }
 
-    return mOMX->setParameter(
+    err = mOMX->setParameter(
             mNode, OMX_IndexParamAudioPcm, &pcmParams, sizeof(pcmParams));
+    // if we could not set up raw format to non-16-bit, try with 16-bit
+    // NOTE: we will also verify this via readback, in case codec ignores these fields
+    if (err != OK && encoding != kAudioEncodingPcm16bit) {
+        pcmParams.eNumData = OMX_NumericalDataSigned;
+        pcmParams.nBitPerSample = 16;
+        err = mOMX->setParameter(
+                mNode, OMX_IndexParamAudioPcm, &pcmParams, sizeof(pcmParams));
+    }
+    return err;
 }
 
 status_t ACodec::configureTunneledVideoPlayback(
@@ -3847,6 +3947,13 @@
         h264type.eLevel = static_cast<OMX_VIDEO_AVCLEVELTYPE>(level);
     }
 
+    // XXX
+    if (h264type.eProfile != OMX_VIDEO_AVCProfileBaseline) {
+        ALOGW("Use baseline profile instead of %d for AVC recording",
+            h264type.eProfile);
+        h264type.eProfile = OMX_VIDEO_AVCProfileBaseline;
+    }
+
     if (h264type.eProfile == OMX_VIDEO_AVCProfileBaseline) {
         h264type.nSliceHeaderSpacing = 0;
         h264type.bUseHadamard = OMX_TRUE;
@@ -3864,23 +3971,6 @@
         h264type.bDirect8x8Inference = OMX_FALSE;
         h264type.bDirectSpatialTemporal = OMX_FALSE;
         h264type.nCabacInitIdc = 0;
-    } else if (h264type.eProfile == OMX_VIDEO_AVCProfileMain ||
-            h264type.eProfile == OMX_VIDEO_AVCProfileHigh) {
-        h264type.nSliceHeaderSpacing = 0;
-        h264type.bUseHadamard = OMX_TRUE;
-        h264type.nRefFrames = 2;
-        h264type.nBFrames = 1;
-        h264type.nPFrames = setPFramesSpacing(iFrameInterval, frameRate);
-        h264type.nAllowedPictureTypes =
-            OMX_VIDEO_PictureTypeI | OMX_VIDEO_PictureTypeP | OMX_VIDEO_PictureTypeB;
-        h264type.nRefIdx10ActiveMinus1 = 0;
-        h264type.nRefIdx11ActiveMinus1 = 0;
-        h264type.bEntropyCodingCABAC = OMX_TRUE;
-        h264type.bWeightedPPrediction = OMX_TRUE;
-        h264type.bconstIpred = OMX_TRUE;
-        h264type.bDirect8x8Inference = OMX_TRUE;
-        h264type.bDirectSpatialTemporal = OMX_TRUE;
-        h264type.nCabacInitIdc = 1;
     }
 
     if (h264type.nBFrames != 0) {
@@ -4643,15 +4733,11 @@
 
                     if (params.nChannels <= 0
                             || (params.nChannels != 1 && !params.bInterleaved)
-                            || params.nBitPerSample != 16u
-                            || params.eNumData != OMX_NumericalDataSigned
                             || params.ePCMMode != OMX_AUDIO_PCMModeLinear) {
-                        ALOGE("unsupported PCM port: %u channels%s, %u-bit, %s(%d), %s(%d) mode ",
+                        ALOGE("unsupported PCM port: %u channels%s, %u-bit",
                                 params.nChannels,
                                 params.bInterleaved ? " interleaved" : "",
-                                params.nBitPerSample,
-                                asString(params.eNumData), params.eNumData,
-                                asString(params.ePCMMode), params.ePCMMode);
+                                params.nBitPerSample);
                         return FAILED_TRANSACTION;
                     }
 
@@ -4659,6 +4745,22 @@
                     notify->setInt32("channel-count", params.nChannels);
                     notify->setInt32("sample-rate", params.nSamplingRate);
 
+                    AudioEncoding encoding = kAudioEncodingPcm16bit;
+                    if (params.eNumData == OMX_NumericalDataUnsigned
+                            && params.nBitPerSample == 8u) {
+                        encoding = kAudioEncodingPcm8bit;
+                    } else if (params.eNumData == OMX_NumericalDataFloat
+                            && params.nBitPerSample == 32u) {
+                        encoding = kAudioEncodingPcmFloat;
+                    } else if (params.nBitPerSample != 16u
+                            || params.eNumData != OMX_NumericalDataSigned) {
+                        ALOGE("unsupported PCM port: %s(%d), %s(%d) mode ",
+                                asString(params.eNumData), params.eNumData,
+                                asString(params.ePCMMode), params.ePCMMode);
+                        return FAILED_TRANSACTION;
+                    }
+                    notify->setInt32("pcm-encoding", encoding);
+
                     if (mChannelMaskPresent) {
                         notify->setInt32("channel-mask", mChannelMask);
                     }
@@ -4840,6 +4942,7 @@
                     notify->setString("mime", mime);
                     notify->setInt32("channel-count", params.nChannels);
                     notify->setInt32("sample-rate", params.nSamplingRate);
+                    notify->setInt32("pcm-encoding", kAudioEncodingPcm16bit);
                     break;
                 }
 
@@ -4917,7 +5020,7 @@
 }
 
 void ACodec::onOutputFormatChanged() {
-    // store new output format
+    // store new output format, at the same time mark that this is no longer the first frame
     mOutputFormat = mBaseOutputFormat->dup();
 
     if (getPortFormat(kPortIndexOutput, mOutputFormat) != OK) {
@@ -4925,6 +5028,18 @@
         return;
     }
 
+    if (!mIsVideo && !mIsEncoder) {
+        AudioEncoding pcmEncoding = kAudioEncodingPcm16bit;
+        (void)mConfigFormat->findInt32("pcm-encoding", (int32_t*)&pcmEncoding);
+        AudioEncoding codecPcmEncoding = kAudioEncodingPcm16bit;
+        (void)mOutputFormat->findInt32("pcm-encoding", (int32_t*)&pcmEncoding);
+
+        mConverter[kPortIndexOutput] = AudioConverter::Create(codecPcmEncoding, pcmEncoding);
+        if (mConverter[kPortIndexOutput] != NULL) {
+            mOutputFormat->setInt32("pcm-encoding", pcmEncoding);
+        }
+    }
+
     if (mTunneled) {
         sendFormatChange();
     }
@@ -5452,20 +5567,21 @@
                     flags |= OMX_BUFFERFLAG_EOS;
                 }
 
-                if (buffer != info->mData) {
+                if (buffer != info->mCodecData) {
                     ALOGV("[%s] Needs to copy input data for buffer %u. (%p != %p)",
                          mCodec->mComponentName.c_str(),
                          bufferID,
-                         buffer.get(), info->mData.get());
+                         buffer.get(), info->mCodecData.get());
 
-                    if (buffer->size() > info->mData->capacity()) {
-                        ALOGE("data size (%zu) is greated than buffer capacity (%zu)",
-                                buffer->size(),           // this is the data received
-                                info->mData->capacity()); // this is out buffer size
-                        mCodec->signalError(OMX_ErrorUndefined, FAILED_TRANSACTION);
+                    sp<DataConverter> converter = mCodec->mConverter[kPortIndexInput];
+                    if (converter == NULL) {
+                        converter = getCopyConverter();
+                    }
+                    status_t err = converter->convert(buffer, info->mCodecData);
+                    if (err != OK) {
+                        mCodec->signalError(OMX_ErrorUndefined, err);
                         return;
                     }
-                    memcpy(info->mData->data(), buffer->data(), buffer->size());
                 }
 
                 if (flags & OMX_BUFFERFLAG_CODECCONFIG) {
@@ -5508,7 +5624,7 @@
                     mCodec->mNode,
                     bufferID,
                     0,
-                    buffer->size(),
+                    info->mCodecData->size(),
                     flags,
                     timeUs,
                     info->mFenceFd);
@@ -5687,6 +5803,10 @@
                 new AMessage(kWhatOutputBufferDrained, mCodec);
 
             if (mCodec->mOutputFormat != mCodec->mLastOutputFormat && rangeLength > 0) {
+                // pretend that output format has changed on the first frame (we used to do this)
+                if (mCodec->mBaseOutputFormat == mCodec->mOutputFormat) {
+                    mCodec->onOutputFormatChanged();
+                }
                 mCodec->addKeyFormatChangesToRenderBufferNotification(reply);
                 mCodec->sendFormatChange();
             }
@@ -5710,8 +5830,17 @@
                 info->mData->meta()->setPointer("handle", handle);
                 info->mData->meta()->setInt32("rangeOffset", rangeOffset);
                 info->mData->meta()->setInt32("rangeLength", rangeLength);
-            } else {
+            } else if (info->mData == info->mCodecData) {
                 info->mData->setRange(rangeOffset, rangeLength);
+            } else {
+                info->mCodecData->setRange(rangeOffset, rangeLength);
+                // in this case we know that mConverter is not null
+                status_t err = mCodec->mConverter[kPortIndexOutput]->convert(
+                        info->mCodecData, info->mData);
+                if (err != OK) {
+                    mCodec->signalError(OMX_ErrorUndefined, makeNoSideEffectStatus(err));
+                    return true;
+                }
             }
 #if 0
             if (mCodec->mNativeWindow == NULL) {
@@ -5927,6 +6056,8 @@
     mCodec->mFlags = 0;
     mCodec->mInputMetadataType = kMetadataBufferTypeInvalid;
     mCodec->mOutputMetadataType = kMetadataBufferTypeInvalid;
+    mCodec->mConverter[0].clear();
+    mCodec->mConverter[1].clear();
     mCodec->mComponentName.clear();
 }
 
diff --git a/media/libstagefright/Android.mk b/media/libstagefright/Android.mk
index 557971d..2445842 100644
--- a/media/libstagefright/Android.mk
+++ b/media/libstagefright/Android.mk
@@ -14,6 +14,7 @@
         CameraSource.cpp                  \
         CameraSourceTimeLapse.cpp         \
         CodecBase.cpp                     \
+        DataConverter.cpp                 \
         DataSource.cpp                    \
         DataURISource.cpp                 \
         DRMExtractor.cpp                  \
diff --git a/media/libstagefright/AudioPlayer.cpp b/media/libstagefright/AudioPlayer.cpp
index cb42847..b3fb8d4 100644
--- a/media/libstagefright/AudioPlayer.cpp
+++ b/media/libstagefright/AudioPlayer.cpp
@@ -25,6 +25,7 @@
 #include <media/AudioTrack.h>
 #include <media/openmax/OMX_Audio.h>
 #include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/ALookup.h>
 #include <media/stagefright/foundation/ALooper.h>
 #include <media/stagefright/AudioPlayer.h>
 #include <media/stagefright/MediaDefs.h>
@@ -71,6 +72,14 @@
     mSource = source;
 }
 
+ALookup<audio_format_t, int32_t> sAudioFormatToPcmEncoding {
+    {
+        { AUDIO_FORMAT_PCM_16_BIT, kAudioEncodingPcm16bit },
+        { AUDIO_FORMAT_PCM_8_BIT,  kAudioEncodingPcm8bit  },
+        { AUDIO_FORMAT_PCM_FLOAT,  kAudioEncodingPcmFloat },
+    }
+};
+
 status_t AudioPlayer::start(bool sourceAlreadyStarted) {
     CHECK(!mStarted);
     CHECK(mSource != NULL);
@@ -129,6 +138,10 @@
     }
 
     audio_format_t audioFormat = AUDIO_FORMAT_PCM_16_BIT;
+    int32_t pcmEncoding;
+    if (format->findInt32(kKeyPcmEncoding, &pcmEncoding)) {
+        sAudioFormatToPcmEncoding.map(pcmEncoding, &audioFormat);
+    }
 
     if (useOffload()) {
         if (mapMimeToAudioFormat(audioFormat, mime) != OK) {
diff --git a/media/libstagefright/AudioSource.cpp b/media/libstagefright/AudioSource.cpp
index 6e4a1dd..f28ac58 100644
--- a/media/libstagefright/AudioSource.cpp
+++ b/media/libstagefright/AudioSource.cpp
@@ -184,6 +184,7 @@
     meta->setInt32(kKeySampleRate, mSampleRate);
     meta->setInt32(kKeyChannelCount, mRecord->channelCount());
     meta->setInt32(kKeyMaxInputSize, kMaxBufferSize);
+    meta->setInt32(kKeyPcmEncoding, kAudioEncodingPcm16bit);
 
     return meta;
 }
diff --git a/media/libstagefright/DataConverter.cpp b/media/libstagefright/DataConverter.cpp
new file mode 100644
index 0000000..aea47f3
--- /dev/null
+++ b/media/libstagefright/DataConverter.cpp
@@ -0,0 +1,123 @@
+/*
+ * Copyright (C) 2016 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 "DataConverter"
+
+#include "include/DataConverter.h"
+
+#include <audio_utils/primitives.h>
+
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AUtils.h>
+
+namespace android {
+
+status_t DataConverter::convert(const sp<ABuffer> &source, sp<ABuffer> &target) {
+    CHECK(source->base() != target->base());
+    size_t size = targetSize(source->size());
+    status_t err = OK;
+    if (size > target->capacity()) {
+        ALOGE("data size (%zu) is greater than buffer capacity (%zu)",
+                size,          // this is the data received/to be converted
+                target->capacity()); // this is out buffer size
+        err = FAILED_TRANSACTION;
+    } else {
+        err = safeConvert(source, target);
+    }
+    target->setRange(0, err == OK ? size : 0);
+    return err;
+}
+
+status_t DataConverter::safeConvert(const sp<ABuffer> &source, sp<ABuffer> &target) {
+    memcpy(target->base(), source->data(), source->size());
+    return OK;
+}
+
+size_t DataConverter::sourceSize(size_t targetSize) {
+    return targetSize;
+}
+
+size_t DataConverter::targetSize(size_t sourceSize) {
+    return sourceSize;
+}
+
+DataConverter::~DataConverter() { }
+
+
+size_t SampleConverterBase::sourceSize(size_t targetSize) {
+    size_t numSamples = targetSize / mTargetSampleSize;
+    if (numSamples > SIZE_MAX / mSourceSampleSize) {
+        ALOGW("limiting source size due to overflow (%zu*%zu/%zu)",
+                targetSize, mSourceSampleSize, mTargetSampleSize);
+        return SIZE_MAX;
+    }
+    return numSamples * mSourceSampleSize;
+}
+
+size_t SampleConverterBase::targetSize(size_t sourceSize) {
+    // we round up on conversion
+    size_t numSamples = divUp(sourceSize, (size_t)mSourceSampleSize);
+    if (numSamples > SIZE_MAX / mTargetSampleSize) {
+        ALOGW("limiting target size due to overflow (%zu*%zu/%zu)",
+                sourceSize, mTargetSampleSize, mSourceSampleSize);
+        return SIZE_MAX;
+    }
+    return numSamples * mTargetSampleSize;
+}
+
+
+static size_t getAudioSampleSize(AudioEncoding e) {
+    switch (e) {
+        case kAudioEncodingPcm16bit: return 2;
+        case kAudioEncodingPcm8bit:  return 1;
+        case kAudioEncodingPcmFloat: return 4;
+        default: return 0;
+    }
+}
+
+
+// static
+AudioConverter* AudioConverter::Create(AudioEncoding source, AudioEncoding target) {
+    uint32_t sourceSampleSize = getAudioSampleSize(source);
+    uint32_t targetSampleSize = getAudioSampleSize(target);
+    if (sourceSampleSize && targetSampleSize && sourceSampleSize != targetSampleSize) {
+        return new AudioConverter(source, sourceSampleSize, target, targetSampleSize);
+    }
+    return NULL;
+}
+
+status_t AudioConverter::safeConvert(const sp<ABuffer> &src, sp<ABuffer> &tgt) {
+    if (mTo == kAudioEncodingPcm8bit && mFrom == kAudioEncodingPcm16bit) {
+        memcpy_to_u8_from_i16((uint8_t*)tgt->base(), (const int16_t*)src->data(), src->size() / 2);
+    } else if (mTo == kAudioEncodingPcm8bit && mFrom == kAudioEncodingPcmFloat) {
+        memcpy_to_u8_from_float((uint8_t*)tgt->base(), (const float*)src->data(), src->size() / 4);
+    } else if (mTo == kAudioEncodingPcm16bit && mFrom == kAudioEncodingPcm8bit) {
+        memcpy_to_i16_from_u8((int16_t*)tgt->base(), (const uint8_t*)src->data(), src->size());
+    } else if (mTo == kAudioEncodingPcm16bit && mFrom == kAudioEncodingPcmFloat) {
+        memcpy_to_i16_from_float((int16_t*)tgt->base(), (const float*)src->data(), src->size() / 4);
+    } else if (mTo == kAudioEncodingPcmFloat && mFrom == kAudioEncodingPcm8bit) {
+        memcpy_to_float_from_u8((float*)tgt->base(), (const uint8_t*)src->data(), src->size());
+    } else if (mTo == kAudioEncodingPcmFloat && mFrom == kAudioEncodingPcm16bit) {
+        memcpy_to_float_from_i16((float*)tgt->base(), (const int16_t*)src->data(), src->size() / 2);
+    } else {
+        return INVALID_OPERATION;
+    }
+    return OK;
+}
+
+} // namespace android
diff --git a/media/libstagefright/FLACExtractor.cpp b/media/libstagefright/FLACExtractor.cpp
index 6e99d02..13b66f3 100644
--- a/media/libstagefright/FLACExtractor.cpp
+++ b/media/libstagefright/FLACExtractor.cpp
@@ -615,6 +615,7 @@
             mTrackMetadata->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_RAW);
             mTrackMetadata->setInt32(kKeyChannelCount, getChannels());
             mTrackMetadata->setInt32(kKeySampleRate, getSampleRate());
+            mTrackMetadata->setInt32(kKeyPcmEncoding, kAudioEncodingPcm16bit);
             // sample rate is non-zero, so division by zero not possible
             mTrackMetadata->setInt64(kKeyDuration,
                     (getTotalSamples() * 1000000LL) / getSampleRate());
diff --git a/media/libstagefright/MPEG4Extractor.cpp b/media/libstagefright/MPEG4Extractor.cpp
index 322eab9..f5549e4 100644
--- a/media/libstagefright/MPEG4Extractor.cpp
+++ b/media/libstagefright/MPEG4Extractor.cpp
@@ -756,13 +756,22 @@
 }
 
 // Given a time in seconds since Jan 1 1904, produce a human-readable string.
-static void convertTimeToDate(int64_t time_1904, String8 *s) {
-    time_t time_1970 = time_1904 - (((66 * 365 + 17) * 24) * 3600);
+static bool convertTimeToDate(int64_t time_1904, String8 *s) {
+    // delta between mpeg4 time and unix epoch time
+    static const int64_t delta = (((66 * 365 + 17) * 24) * 3600);
+    if (time_1904 < INT64_MIN + delta) {
+        return false;
+    }
+    time_t time_1970 = time_1904 - delta;
 
     char tmp[32];
-    strftime(tmp, sizeof(tmp), "%Y%m%dT%H%M%S.000Z", gmtime(&time_1970));
-
-    s->setTo(tmp);
+    struct tm* tm = gmtime(&time_1970);
+    if (tm != NULL &&
+            strftime(tmp, sizeof(tmp), "%Y%m%dT%H%M%S.000Z", tm) > 0) {
+        s->setTo(tmp);
+        return true;
+    }
+    return false;
 }
 
 status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) {
@@ -1880,14 +1889,15 @@
                 }
                 duration = d32;
             }
-            if (duration != 0 && mHeaderTimescale != 0) {
+            if (duration != 0 && mHeaderTimescale != 0 && duration < UINT64_MAX / 1000000) {
                 mFileMetaData->setInt64(kKeyDuration, duration * 1000000 / mHeaderTimescale);
             }
 
             String8 s;
-            convertTimeToDate(creationTime, &s);
+            if (convertTimeToDate(creationTime, &s)) {
+                mFileMetaData->setCString(kKeyDate, s.string());
+            }
 
-            mFileMetaData->setCString(kKeyDate, s.string());
 
             break;
         }
diff --git a/media/libstagefright/MidiExtractor.cpp b/media/libstagefright/MidiExtractor.cpp
index 7525f57..7930bbb 100644
--- a/media/libstagefright/MidiExtractor.cpp
+++ b/media/libstagefright/MidiExtractor.cpp
@@ -178,6 +178,7 @@
         mEasConfig = EAS_Config();
         trackMetadata->setInt32(kKeySampleRate, mEasConfig->sampleRate);
         trackMetadata->setInt32(kKeyChannelCount, mEasConfig->numChannels);
+        trackMetadata->setInt32(kKeyPcmEncoding, kAudioEncodingPcm16bit);
     }
     mIsInitialized = true;
 }
diff --git a/media/libstagefright/Utils.cpp b/media/libstagefright/Utils.cpp
index 34deaad..dc4acda 100644
--- a/media/libstagefright/Utils.cpp
+++ b/media/libstagefright/Utils.cpp
@@ -232,13 +232,18 @@
 
         int32_t isADTS;
         if (meta->findInt32(kKeyIsADTS, &isADTS)) {
-            msg->setInt32("is-adts", true);
+            msg->setInt32("is-adts", isADTS);
         }
 
         int32_t aacProfile = -1;
         if (meta->findInt32(kKeyAACAOT, &aacProfile)) {
             msg->setInt32("aac-profile", aacProfile);
         }
+
+        int32_t pcmEncoding;
+        if (meta->findInt32(kKeyPcmEncoding, &pcmEncoding)) {
+            msg->setInt32("pcm-encoding", pcmEncoding);
+        }
     }
 
     int32_t maxInputSize;
@@ -794,6 +799,11 @@
         if (msg->findInt32("is-adts", &isADTS)) {
             meta->setInt32(kKeyIsADTS, isADTS);
         }
+
+        int32_t pcmEncoding;
+        if (msg->findInt32("pcm-encoding", &pcmEncoding)) {
+            meta->setInt32(kKeyPcmEncoding, pcmEncoding);
+        }
     }
 
     int32_t maxInputSize;
diff --git a/media/libstagefright/VideoFrameScheduler.cpp b/media/libstagefright/VideoFrameScheduler.cpp
index 5564926..03226c7 100644
--- a/media/libstagefright/VideoFrameScheduler.cpp
+++ b/media/libstagefright/VideoFrameScheduler.cpp
@@ -156,12 +156,12 @@
         lastTime = time;
     }
 
-    int64_t div   = numSamplesToUse * sumXX - sumX * sumX;
+    int64_t div   = (int64_t)numSamplesToUse * sumXX - sumX * sumX;
     if (div == 0) {
         return false;
     }
 
-    int64_t a_nom = numSamplesToUse * sumXY - sumX * sumY;
+    int64_t a_nom = (int64_t)numSamplesToUse * sumXY - sumX * sumY;
     int64_t b_nom = sumXX * sumY            - sumX * sumXY;
     *a = divRound(a_nom, div);
     *b = divRound(b_nom, div);
@@ -437,10 +437,10 @@
                 (renderTime + mTimeCorrection + videoPeriod * i - mVsyncTime) % mVsyncPeriod;
             edgeRemainder += (videoPeriod * i) % mVsyncPeriod;
         }
-        mTimeCorrection += mVsyncPeriod / 2 - offset / N;
+        mTimeCorrection += mVsyncPeriod / 2 - offset / (nsecs_t)N;
         renderTime += mTimeCorrection;
         nsecs_t correctionLimit = mVsyncPeriod * 3 / 5;
-        edgeRemainder = abs(edgeRemainder / N - mVsyncPeriod / 2);
+        edgeRemainder = abs(edgeRemainder / (nsecs_t)N - mVsyncPeriod / 2);
         if (edgeRemainder <= mVsyncPeriod / 3) {
             correctionLimit /= 2;
         }
diff --git a/media/libstagefright/WAVExtractor.cpp b/media/libstagefright/WAVExtractor.cpp
index 15e3845..38a2a06 100644
--- a/media/libstagefright/WAVExtractor.cpp
+++ b/media/libstagefright/WAVExtractor.cpp
@@ -308,6 +308,7 @@
                 mTrackMeta->setInt32(kKeyChannelCount, mNumChannels);
                 mTrackMeta->setInt32(kKeyChannelMask, mChannelMask);
                 mTrackMeta->setInt32(kKeySampleRate, mSampleRate);
+                mTrackMeta->setInt32(kKeyPcmEncoding, kAudioEncodingPcm16bit);
 
                 int64_t durationUs = 0;
                 if (mWaveFormat == WAVE_FORMAT_MSGSM) {
diff --git a/media/libstagefright/codecs/amrnb/dec/SoftAMR.cpp b/media/libstagefright/codecs/amrnb/dec/SoftAMR.cpp
index ac19ee3..be22098 100644
--- a/media/libstagefright/codecs/amrnb/dec/SoftAMR.cpp
+++ b/media/libstagefright/codecs/amrnb/dec/SoftAMR.cpp
@@ -306,6 +306,13 @@
         BufferInfo *inInfo = *inQueue.begin();
         OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
 
+        if (inHeader->nFilledLen == 0) {
+            inInfo->mOwnedByUs = false;
+            inQueue.erase(inQueue.begin());
+            notifyEmptyBufferDone(inHeader);
+            continue;
+        }
+
         BufferInfo *outInfo = *outQueue.begin();
         OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
 
@@ -332,6 +339,26 @@
         int32_t numBytesRead;
 
         if (mMode == MODE_NARROW) {
+            if (outHeader->nAllocLen < kNumSamplesPerFrameNB * sizeof(int16_t)) {
+                ALOGE("b/27662364: NB expected output buffer %zu bytes vs %u",
+                       kNumSamplesPerFrameNB * sizeof(int16_t), outHeader->nAllocLen);
+                android_errorWriteLog(0x534e4554, "27662364");
+                notify(OMX_EventError, OMX_ErrorOverflow, 0, NULL);
+                mSignalledError = true;
+                return;
+            }
+
+            int16 mode = ((inputPtr[0] >> 3) & 0x0f);
+            // for WMF since MIME_IETF is used when calling AMRDecode.
+            size_t frameSize = WmfDecBytesPerFrame[mode] + 1;
+
+            if (inHeader->nFilledLen < frameSize) {
+                ALOGE("b/27662364: expected %zu bytes vs %u", frameSize, inHeader->nFilledLen);
+                notify(OMX_EventError, OMX_ErrorStreamCorrupt, 0, NULL);
+                mSignalledError = true;
+                return;
+            }
+
             numBytesRead =
                 AMRDecode(mState,
                   (Frame_Type_3GPP)((inputPtr[0] >> 3) & 0x0f),
@@ -359,6 +386,15 @@
                 return;
             }
         } else {
+            if (outHeader->nAllocLen < kNumSamplesPerFrameWB * sizeof(int16_t)) {
+                ALOGE("b/27662364: WB expected output buffer %zu bytes vs %u",
+                       kNumSamplesPerFrameWB * sizeof(int16_t), outHeader->nAllocLen);
+                android_errorWriteLog(0x534e4554, "27662364");
+                notify(OMX_EventError, OMX_ErrorOverflow, 0, NULL);
+                mSignalledError = true;
+                return;
+            }
+
             int16 mode = ((inputPtr[0] >> 3) & 0x0f);
 
             if (mode >= 10 && mode <= 13) {
@@ -372,7 +408,12 @@
             }
 
             size_t frameSize = getFrameSize(mode);
-            CHECK_GE(inHeader->nFilledLen, frameSize);
+            if (inHeader->nFilledLen < frameSize) {
+                ALOGE("b/27662364: expected %zu bytes vs %u", frameSize, inHeader->nFilledLen);
+                notify(OMX_EventError, OMX_ErrorStreamCorrupt, 0, NULL);
+                mSignalledError = true;
+                return;
+            }
 
             int16_t *outPtr = (int16_t *)outHeader->pBuffer;
 
diff --git a/media/libstagefright/codecs/amrnb/dec/src/amrdecode.h b/media/libstagefright/codecs/amrnb/dec/src/amrdecode.h
index 0988e17f..f224fb6 100644
--- a/media/libstagefright/codecs/amrnb/dec/src/amrdecode.h
+++ b/media/libstagefright/codecs/amrnb/dec/src/amrdecode.h
@@ -104,7 +104,6 @@
 ; INCLUDES
 ----------------------------------------------------------------------------*/
 #include    "typedef.h"
-#include    "mode.h"
 #include    "frame_type_3gpp.h"
 
 /*--------------------------------------------------------------------------*/
diff --git a/media/libstagefright/codecs/amrnb/dec/src/gsmamr_dec.h b/media/libstagefright/codecs/amrnb/dec/src/gsmamr_dec.h
index 8f54ee8..dc64d67 100644
--- a/media/libstagefright/codecs/amrnb/dec/src/gsmamr_dec.h
+++ b/media/libstagefright/codecs/amrnb/dec/src/gsmamr_dec.h
@@ -87,6 +87,7 @@
 
 #include "gsm_amr_typedefs.h"
 #include "frame_type_3gpp.h"
+#include "amrdecode.h"
 
 /*--------------------------------------------------------------------------*/
 #ifdef __cplusplus
@@ -136,19 +137,6 @@
                          Word8 *id);
 
     /*
-     * AMRDecode steps into the part of the library that decodes the raw data
-     * speech bits for the decoding process. It returns the address offset of
-     * the next frame to be decoded.
-     */
-    Word16 AMRDecode(
-        void                      *state_data,
-        enum Frame_Type_3GPP      frame_type,
-        UWord8                    *speech_bits_ptr,
-        Word16                    *raw_pcm_buffer,
-        Word16                    input_format
-    );
-
-    /*
      * This function resets the state memory used by the GSM AMR decoder. This
      * function returns zero. It will return negative one if there is an error.
      */
diff --git a/media/libstagefright/codecs/on2/dec/SoftVPX.cpp b/media/libstagefright/codecs/on2/dec/SoftVPX.cpp
index 6106a93..2a56ed5 100644
--- a/media/libstagefright/codecs/on2/dec/SoftVPX.cpp
+++ b/media/libstagefright/codecs/on2/dec/SoftVPX.cpp
@@ -156,15 +156,20 @@
         outHeader->nFlags = 0;
         outHeader->nFilledLen = (outputBufferWidth() * outputBufferHeight() * 3) / 2;
         outHeader->nTimeStamp = *(OMX_TICKS *)mImg->user_priv;
-
-        uint8_t *dst = outHeader->pBuffer;
-        const uint8_t *srcY = (const uint8_t *)mImg->planes[VPX_PLANE_Y];
-        const uint8_t *srcU = (const uint8_t *)mImg->planes[VPX_PLANE_U];
-        const uint8_t *srcV = (const uint8_t *)mImg->planes[VPX_PLANE_V];
-        size_t srcYStride = mImg->stride[VPX_PLANE_Y];
-        size_t srcUStride = mImg->stride[VPX_PLANE_U];
-        size_t srcVStride = mImg->stride[VPX_PLANE_V];
-        copyYV12FrameToOutputBuffer(dst, srcY, srcU, srcV, srcYStride, srcUStride, srcVStride);
+        if (outHeader->nAllocLen >= outHeader->nFilledLen) {
+            uint8_t *dst = outHeader->pBuffer;
+            const uint8_t *srcY = (const uint8_t *)mImg->planes[VPX_PLANE_Y];
+            const uint8_t *srcU = (const uint8_t *)mImg->planes[VPX_PLANE_U];
+            const uint8_t *srcV = (const uint8_t *)mImg->planes[VPX_PLANE_V];
+            size_t srcYStride = mImg->stride[VPX_PLANE_Y];
+            size_t srcUStride = mImg->stride[VPX_PLANE_U];
+            size_t srcVStride = mImg->stride[VPX_PLANE_V];
+            copyYV12FrameToOutputBuffer(dst, srcY, srcU, srcV, srcYStride, srcUStride, srcVStride);
+        } else {
+            ALOGE("b/27597103, buffer too small");
+            android_errorWriteLog(0x534e4554, "27597103");
+            outHeader->nFilledLen = 0;
+        }
 
         mImg = NULL;
         outInfo->mOwnedByUs = false;
diff --git a/media/libstagefright/codecs/on2/enc/SoftVPXEncoder.cpp b/media/libstagefright/codecs/on2/enc/SoftVPXEncoder.cpp
index a354690..5edfbb5 100644
--- a/media/libstagefright/codecs/on2/enc/SoftVPXEncoder.cpp
+++ b/media/libstagefright/codecs/on2/enc/SoftVPXEncoder.cpp
@@ -734,9 +734,10 @@
         const uint8_t *source =
             inputBufferHeader->pBuffer + inputBufferHeader->nOffset;
 
+        size_t frameSize = mWidth * mHeight * 3 / 2;
         if (mInputDataIsMeta) {
             source = extractGraphicBuffer(
-                    mConversionBuffer, mWidth * mHeight * 3 / 2,
+                    mConversionBuffer, frameSize,
                     source, inputBufferHeader->nFilledLen,
                     mWidth, mHeight);
             if (source == NULL) {
@@ -744,11 +745,21 @@
                 notify(OMX_EventError, OMX_ErrorUndefined, 0, 0);
                 return;
             }
-        } else if (mColorFormat == OMX_COLOR_FormatYUV420SemiPlanar) {
-            ConvertYUV420SemiPlanarToYUV420Planar(
-                    source, mConversionBuffer, mWidth, mHeight);
+        } else {
+            if (inputBufferHeader->nFilledLen < frameSize) {
+                android_errorWriteLog(0x534e4554, "27569635");
+                notify(OMX_EventError, OMX_ErrorUndefined, 0, 0);
+                return;
+            } else if (inputBufferHeader->nFilledLen > frameSize) {
+                ALOGW("Input buffer contains too many pixels");
+            }
 
-            source = mConversionBuffer;
+            if (mColorFormat == OMX_COLOR_FormatYUV420SemiPlanar) {
+                ConvertYUV420SemiPlanarToYUV420Planar(
+                        source, mConversionBuffer, mWidth, mHeight);
+
+                source = mConversionBuffer;
+            }
         }
         vpx_image_t raw_frame;
         vpx_img_wrap(&raw_frame, VPX_IMG_FMT_I420, mWidth, mHeight,
@@ -810,9 +821,14 @@
                 outputBufferHeader->nTimeStamp = encoded_packet->data.frame.pts;
                 outputBufferHeader->nFlags = 0;
                 if (encoded_packet->data.frame.flags & VPX_FRAME_IS_KEY)
-                  outputBufferHeader->nFlags |= OMX_BUFFERFLAG_SYNCFRAME;
+                    outputBufferHeader->nFlags |= OMX_BUFFERFLAG_SYNCFRAME;
                 outputBufferHeader->nOffset = 0;
                 outputBufferHeader->nFilledLen = encoded_packet->data.frame.sz;
+                if (outputBufferHeader->nFilledLen > outputBufferHeader->nAllocLen) {
+                    android_errorWriteLog(0x534e4554, "27569635");
+                    notify(OMX_EventError, OMX_ErrorUndefined, 0, 0);
+                    return;
+                }
                 memcpy(outputBufferHeader->pBuffer,
                        encoded_packet->data.frame.buf,
                        encoded_packet->data.frame.sz);
diff --git a/media/libstagefright/codecs/raw/SoftRaw.cpp b/media/libstagefright/codecs/raw/SoftRaw.cpp
index c4e0659..acb2b37 100644
--- a/media/libstagefright/codecs/raw/SoftRaw.cpp
+++ b/media/libstagefright/codecs/raw/SoftRaw.cpp
@@ -42,7 +42,9 @@
     : SimpleSoftOMXComponent(name, callbacks, appData, component),
       mSignalledError(false),
       mChannelCount(2),
-      mSampleRate(44100) {
+      mSampleRate(44100),
+      mNumericalData(OMX_NumericalDataSigned),
+      mBitsPerSample(16) {
     initPorts();
     CHECK_EQ(initDecoder(), (status_t)OK);
 }
@@ -111,10 +113,10 @@
                 return OMX_ErrorUndefined;
             }
 
-            pcmParams->eNumData = OMX_NumericalDataSigned;
+            pcmParams->eNumData = (OMX_NUMERICALDATATYPE)mNumericalData;
             pcmParams->eEndian = OMX_EndianBig;
             pcmParams->bInterleaved = OMX_TRUE;
-            pcmParams->nBitPerSample = 16;
+            pcmParams->nBitPerSample = mBitsPerSample;
             pcmParams->ePCMMode = OMX_AUDIO_PCMModeLinear;
             pcmParams->eChannelMapping[0] = OMX_AUDIO_ChannelLF;
             pcmParams->eChannelMapping[1] = OMX_AUDIO_ChannelRF;
@@ -166,6 +168,8 @@
 
             mChannelCount = pcmParams->nChannels;
             mSampleRate = pcmParams->nSamplingRate;
+            mNumericalData = pcmParams->eNumData;
+            mBitsPerSample = pcmParams->nBitPerSample;
 
             return OMX_ErrorNone;
         }
diff --git a/media/libstagefright/codecs/raw/SoftRaw.h b/media/libstagefright/codecs/raw/SoftRaw.h
index 015c4a3..80906b4 100644
--- a/media/libstagefright/codecs/raw/SoftRaw.h
+++ b/media/libstagefright/codecs/raw/SoftRaw.h
@@ -50,6 +50,8 @@
 
     int32_t mChannelCount;
     int32_t mSampleRate;
+    int32_t mNumericalData;
+    int32_t mBitsPerSample;
 
     void initPorts();
     status_t initDecoder();
diff --git a/media/libstagefright/include/DataConverter.h b/media/libstagefright/include/DataConverter.h
new file mode 100644
index 0000000..8d67921
--- /dev/null
+++ b/media/libstagefright/include/DataConverter.h
@@ -0,0 +1,79 @@
+/*
+ * Copyright (C) 2016 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 STAGEFRIGHT_DATACONVERTER_H_
+#define STAGEFRIGHT_DATACONVERTER_H_
+
+#include <utils/Errors.h>
+#include <utils/RefBase.h>
+
+#include <media/stagefright/MediaDefs.h>
+
+namespace android {
+
+struct ABuffer;
+
+// DataConverter base class, defaults to memcpy
+struct DataConverter : public RefBase {
+    virtual size_t sourceSize(size_t targetSize); // will clamp to SIZE_MAX
+    virtual size_t targetSize(size_t sourceSize); // will clamp to SIZE_MAX
+
+    status_t convert(const sp<ABuffer> &source, sp<ABuffer> &target);
+    virtual ~DataConverter();
+
+protected:
+    virtual status_t safeConvert(const sp<ABuffer> &source, sp<ABuffer> &target);
+};
+
+// SampleConverterBase uses a ratio to calculate the source and target sizes
+// based on source and target sample sizes.
+struct SampleConverterBase : public DataConverter {
+    virtual size_t sourceSize(size_t targetSize);
+    virtual size_t targetSize(size_t sourceSize);
+
+protected:
+    virtual status_t safeConvert(const sp<ABuffer> &source, sp<ABuffer> &target) = 0;
+
+    // sourceSize = sourceSampleSize / targetSampleSize * targetSize
+    SampleConverterBase(uint32_t sourceSampleSize, uint32_t targetSampleSize)
+        : mSourceSampleSize(sourceSampleSize),
+          mTargetSampleSize(targetSampleSize) { }
+    size_t mSourceSampleSize;
+    size_t mTargetSampleSize;
+};
+
+// AudioConverter converts between audio PCM formats
+struct AudioConverter : public SampleConverterBase {
+    // return nullptr if conversion is not needed or not supported
+    static AudioConverter *Create(AudioEncoding source, AudioEncoding target);
+
+protected:
+    virtual status_t safeConvert(const sp<ABuffer> &src, sp<ABuffer> &tgt);
+
+private:
+    AudioConverter(
+            AudioEncoding source, size_t sourceSample,
+            AudioEncoding target, size_t targetSample)
+        : SampleConverterBase(sourceSample, targetSample),
+          mFrom(source),
+          mTo(target) { }
+    AudioEncoding mFrom;
+    AudioEncoding mTo;
+};
+
+} // namespace android
+
+#endif
diff --git a/media/libstagefright/include/MPEG2TSExtractor.h b/media/libstagefright/include/MPEG2TSExtractor.h
index e5c24ca..34b9606 100644
--- a/media/libstagefright/include/MPEG2TSExtractor.h
+++ b/media/libstagefright/include/MPEG2TSExtractor.h
@@ -25,6 +25,8 @@
 #include <utils/KeyedVector.h>
 #include <utils/Vector.h>
 
+#include "mpeg2ts/ATSParser.h"
+
 namespace android {
 
 struct AMessage;
@@ -55,6 +57,10 @@
 
     sp<ATSParser> mParser;
 
+    // Used to remember SyncEvent occurred in feedMore() when called from init(),
+    // because init() needs to update |mSourceImpls| before adding SyncPoint.
+    ATSParser::SyncEvent mLastSyncEvent;
+
     Vector<sp<AnotherPacketSource> > mSourceImpls;
 
     Vector<KeyedVector<int64_t, off64_t> > mSyncPoints;
@@ -65,7 +71,14 @@
     off64_t mOffset;
 
     void init();
-    status_t feedMore();
+    // Try to feed more data from source to parser.
+    // |isInit| means this function is called inside init(). This is a signal to
+    // save SyncEvent so that init() can add SyncPoint after it updates |mSourceImpls|.
+    // This function returns OK if expected amount of data is fed from DataSource to
+    // parser and is successfully parsed. Otherwise, various error codes could be
+    // returned, e.g., ERROR_END_OF_STREAM, or no data availalbe from DataSource, or
+    // the data has syntax error during parsing, etc.
+    status_t feedMore(bool isInit = false);
     status_t seek(int64_t seekTimeUs,
             const MediaSource::ReadOptions::SeekMode& seekMode);
     status_t queueDiscontinuityForSeek(int64_t actualSeekTimeUs);
@@ -73,6 +86,9 @@
 
     status_t feedUntilBufferAvailable(const sp<AnotherPacketSource> &impl);
 
+    // Add a SynPoint derived from |event|.
+    void addSyncPoint_l(const ATSParser::SyncEvent &event);
+
     DISALLOW_EVIL_CONSTRUCTORS(MPEG2TSExtractor);
 };
 
diff --git a/media/libstagefright/mpeg2ts/ATSParser.cpp b/media/libstagefright/mpeg2ts/ATSParser.cpp
index 2790a0e..fb43a38 100644
--- a/media/libstagefright/mpeg2ts/ATSParser.cpp
+++ b/media/libstagefright/mpeg2ts/ATSParser.cpp
@@ -122,7 +122,7 @@
     void setPID(unsigned pid) { mElementaryPID = pid; }
 
     // Parse the payload and set event when PES with a sync frame is detected.
-    // This method knows when a PES starts; so record mPesStartOffset in that
+    // This method knows when a PES starts; so record mPesStartOffsets in that
     // case.
     status_t parse(
             unsigned continuity_counter,
@@ -157,7 +157,7 @@
     bool mEOSReached;
 
     uint64_t mPrevPTS;
-    off64_t mPesStartOffset;
+    List<off64_t> mPesStartOffsets;
 
     ElementaryStreamQueue *mQueue;
 
@@ -205,16 +205,19 @@
 };
 
 ATSParser::SyncEvent::SyncEvent(off64_t offset)
-    : mInit(false), mOffset(offset), mTimeUs(0) {}
+    : mHasReturnedData(false), mOffset(offset), mTimeUs(0) {}
 
 void ATSParser::SyncEvent::init(off64_t offset, const sp<MediaSource> &source,
         int64_t timeUs) {
-    mInit = true;
+    mHasReturnedData = true;
     mOffset = offset;
     mMediaSource = source;
     mTimeUs = timeUs;
 }
 
+void ATSParser::SyncEvent::reset() {
+    mHasReturnedData = false;
+}
 ////////////////////////////////////////////////////////////////////////////////
 
 ATSParser::Program::Program(
@@ -661,6 +664,7 @@
         ALOGI("discontinuity on stream pid 0x%04x", mElementaryPID);
 
         mPayloadStarted = false;
+        mPesStartOffsets.clear();
         mBuffer->setRange(0, 0);
         mExpectedContinuityCounter = -1;
 
@@ -697,7 +701,7 @@
         }
 
         mPayloadStarted = true;
-        mPesStartOffset = offset;
+        mPesStartOffsets.push_back(offset);
     }
 
     if (!mPayloadStarted) {
@@ -772,6 +776,7 @@
     }
 
     mPayloadStarted = false;
+    mPesStartOffsets.clear();
     mEOSReached = false;
     mBuffer->setRange(0, 0);
 
@@ -1105,7 +1110,9 @@
                 int64_t timeUs;
                 if (accessUnit->meta()->findInt64("timeUs", &timeUs)) {
                     found = true;
-                    event->init(mPesStartOffset, mSource, timeUs);
+                    off64_t pesStartOffset = *mPesStartOffsets.begin();
+                    event->init(pesStartOffset, mSource, timeUs);
+                    mPesStartOffsets.erase(mPesStartOffsets.begin());
                 }
             }
         }
diff --git a/media/libstagefright/mpeg2ts/ATSParser.h b/media/libstagefright/mpeg2ts/ATSParser.h
index 430a8d5..fb03cd6 100644
--- a/media/libstagefright/mpeg2ts/ATSParser.h
+++ b/media/libstagefright/mpeg2ts/ATSParser.h
@@ -69,16 +69,18 @@
         void init(off64_t offset, const sp<MediaSource> &source,
                 int64_t timeUs);
 
-        bool isInit() { return mInit; }
-        off64_t getOffset() { return mOffset; }
-        const sp<MediaSource> &getMediaSource() { return mMediaSource; }
-        int64_t getTimeUs() { return mTimeUs; }
+        bool hasReturnedData() const { return mHasReturnedData; }
+        void reset();
+        off64_t getOffset() const { return mOffset; }
+        const sp<MediaSource> &getMediaSource() const { return mMediaSource; }
+        int64_t getTimeUs() const { return mTimeUs; }
 
     private:
-        bool mInit;
+        bool mHasReturnedData;
         /*
-         * mInit == false: the current offset
-         * mInit == true: the start offset of sync payload
+         * mHasReturnedData == false: the current offset (or undefined if the returned data
+                                      has been invalidated via reset())
+         * mHasReturnedData == true: the start offset of sync payload
          */
         off64_t mOffset;
         /* The media source object for this event. */
diff --git a/media/libstagefright/mpeg2ts/ESQueue.cpp b/media/libstagefright/mpeg2ts/ESQueue.cpp
index daf6b3d..96ca405 100644
--- a/media/libstagefright/mpeg2ts/ESQueue.cpp
+++ b/media/libstagefright/mpeg2ts/ESQueue.cpp
@@ -592,6 +592,7 @@
         mFormat->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_RAW);
         mFormat->setInt32(kKeyChannelCount, 2);
         mFormat->setInt32(kKeySampleRate, 48000);
+        mFormat->setInt32(kKeyPcmEncoding, kAudioEncodingPcm16bit);
     }
 
     static const size_t kFramesPerAU = 80;
diff --git a/media/libstagefright/mpeg2ts/MPEG2TSExtractor.cpp b/media/libstagefright/mpeg2ts/MPEG2TSExtractor.cpp
index 0b456c3..fb5e079 100644
--- a/media/libstagefright/mpeg2ts/MPEG2TSExtractor.cpp
+++ b/media/libstagefright/mpeg2ts/MPEG2TSExtractor.cpp
@@ -112,6 +112,7 @@
 MPEG2TSExtractor::MPEG2TSExtractor(const sp<DataSource> &source)
     : mDataSource(source),
       mParser(new ATSParser),
+      mLastSyncEvent(0),
       mOffset(0) {
     init();
 }
@@ -149,8 +150,10 @@
     bool haveVideo = false;
     int64_t startTime = ALooper::GetNowUs();
 
-    while (feedMore() == OK) {
+    while (feedMore(true /* isInit */) == OK) {
         if (haveAudio && haveVideo) {
+            addSyncPoint_l(mLastSyncEvent);
+            mLastSyncEvent.reset();
             break;
         }
         if (!haveVideo) {
@@ -181,6 +184,9 @@
             }
         }
 
+        addSyncPoint_l(mLastSyncEvent);
+        mLastSyncEvent.reset();
+
         // Wait only for 2 seconds to detect audio/video streams.
         if (ALooper::GetNowUs() - startTime > 2000000ll) {
             break;
@@ -245,7 +251,7 @@
             haveAudio, haveVideo, ALooper::GetNowUs() - startTime);
 }
 
-status_t MPEG2TSExtractor::feedMore() {
+status_t MPEG2TSExtractor::feedMore(bool isInit) {
     Mutex::Autolock autoLock(mLock);
 
     uint8_t packet[kTSPacketSize];
@@ -261,29 +267,41 @@
     ATSParser::SyncEvent event(mOffset);
     mOffset += n;
     status_t err = mParser->feedTSPacket(packet, kTSPacketSize, &event);
-    if (event.isInit()) {
-        for (size_t i = 0; i < mSourceImpls.size(); ++i) {
-            if (mSourceImpls[i].get() == event.getMediaSource().get()) {
-                KeyedVector<int64_t, off64_t> *syncPoints = &mSyncPoints.editItemAt(i);
-                syncPoints->add(event.getTimeUs(), event.getOffset());
-                // We're keeping the size of the sync points at most 5mb per a track.
-                size_t size = syncPoints->size();
-                if (size >= 327680) {
-                    int64_t firstTimeUs = syncPoints->keyAt(0);
-                    int64_t lastTimeUs = syncPoints->keyAt(size - 1);
-                    if (event.getTimeUs() - firstTimeUs > lastTimeUs - event.getTimeUs()) {
-                        syncPoints->removeItemsAt(0, 4096);
-                    } else {
-                        syncPoints->removeItemsAt(size - 4096, 4096);
-                    }
-                }
-                break;
-            }
+    if (event.hasReturnedData()) {
+        if (isInit) {
+            mLastSyncEvent = event;
+        } else {
+            addSyncPoint_l(event);
         }
     }
     return err;
 }
 
+void MPEG2TSExtractor::addSyncPoint_l(const ATSParser::SyncEvent &event) {
+    if (!event.hasReturnedData()) {
+        return;
+    }
+
+    for (size_t i = 0; i < mSourceImpls.size(); ++i) {
+        if (mSourceImpls[i].get() == event.getMediaSource().get()) {
+            KeyedVector<int64_t, off64_t> *syncPoints = &mSyncPoints.editItemAt(i);
+            syncPoints->add(event.getTimeUs(), event.getOffset());
+            // We're keeping the size of the sync points at most 5mb per a track.
+            size_t size = syncPoints->size();
+            if (size >= 327680) {
+                int64_t firstTimeUs = syncPoints->keyAt(0);
+                int64_t lastTimeUs = syncPoints->keyAt(size - 1);
+                if (event.getTimeUs() - firstTimeUs > lastTimeUs - event.getTimeUs()) {
+                    syncPoints->removeItemsAt(0, 4096);
+                } else {
+                    syncPoints->removeItemsAt(size - 4096, 4096);
+                }
+            }
+            break;
+        }
+    }
+}
+
 uint32_t MPEG2TSExtractor::flags() const {
     return CAN_PAUSE | CAN_SEEK_BACKWARD | CAN_SEEK_FORWARD;
 }
diff --git a/media/libstagefright/omx/OMX.cpp b/media/libstagefright/omx/OMX.cpp
index 759648b..4d89ba1 100644
--- a/media/libstagefright/omx/OMX.cpp
+++ b/media/libstagefright/omx/OMX.cpp
@@ -183,7 +183,12 @@
         Mutex::Autolock autoLock(mLock);
 
         ssize_t index = mLiveNodes.indexOfKey(the_late_who);
-        CHECK(index >= 0);
+
+        if (index < 0) {
+            ALOGE("b/27597103, nonexistent observer on binderDied");
+            android_errorWriteLog(0x534e4554, "27597103");
+            return;
+        }
 
         instance = mLiveNodes.editValueAt(index);
         mLiveNodes.removeItemsAt(index);
diff --git a/media/libstagefright/wifi-display/source/PlaybackSession.cpp b/media/libstagefright/wifi-display/source/PlaybackSession.cpp
index ed5a404..3ecb52b 100644
--- a/media/libstagefright/wifi-display/source/PlaybackSession.cpp
+++ b/media/libstagefright/wifi-display/source/PlaybackSession.cpp
@@ -957,10 +957,12 @@
         format->setInt32("level-idc", levelIdc);
         format->setInt32("constraint-set", constraintSet);
     } else {
-        format->setString(
-                "mime",
-                usePCMAudio
-                    ? MEDIA_MIMETYPE_AUDIO_RAW : MEDIA_MIMETYPE_AUDIO_AAC);
+        if (usePCMAudio) {
+            format->setInt32("pcm-encoding", kAudioEncodingPcm16bit);
+            format->setString("mime", MEDIA_MIMETYPE_AUDIO_RAW);
+        } else {
+            format->setString("mime", MEDIA_MIMETYPE_AUDIO_AAC);
+        }
     }
 
     notify = new AMessage(kWhatConverterNotify, this);
diff --git a/media/mtp/MtpDevice.cpp b/media/mtp/MtpDevice.cpp
index a398aca..d2a93a7 100644
--- a/media/mtp/MtpDevice.cpp
+++ b/media/mtp/MtpDevice.cpp
@@ -679,11 +679,6 @@
         return false;
     }
 
-    if (mData.getContainerType() == MTP_CONTAINER_TYPE_RESPONSE) {
-        mResponse.copyFrom(mData);
-        return mResponse.getResponseCode() == MTP_RESPONSE_OK ? 0 : -1;
-    }
-
     // If object size 0 byte, the remote device can reply response packet
     // without sending any data packets.
     if (mData.getContainerType() == MTP_CONTAINER_TYPE_RESPONSE) {
diff --git a/services/audioflinger/Android.mk b/services/audioflinger/Android.mk
index 302e4dc..6700f6e 100644
--- a/services/audioflinger/Android.mk
+++ b/services/audioflinger/Android.mk
@@ -60,8 +60,9 @@
     libcpustats \
     libmedia_helper
 
+LOCAL_MULTILIB := $(AUDIOSERVER_MULTILIB)
+
 LOCAL_MODULE:= libaudioflinger
-LOCAL_32_BIT_ONLY := true
 
 LOCAL_SRC_FILES += \
     AudioWatchdog.cpp        \
@@ -79,6 +80,8 @@
 
 LOCAL_CFLAGS += -fvisibility=hidden
 
+LOCAL_CFLAGS += -Werror -Wall
+
 include $(BUILD_SHARED_LIBRARY)
 
 #
@@ -107,6 +110,8 @@
 
 LOCAL_MODULE_TAGS := optional
 
+LOCAL_CFLAGS := -Werror -Wall
+
 include $(BUILD_EXECUTABLE)
 
 include $(CLEAR_VARS)
@@ -127,6 +132,8 @@
 
 LOCAL_MODULE := libaudioresampler
 
+LOCAL_CFLAGS := -Werror -Wall
+
 # uncomment to disable NEON on architectures that actually do support NEON, for benchmarking
 #LOCAL_CFLAGS += -DUSE_NEON=false
 
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index d07ca85..016c25e 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -219,8 +219,6 @@
 
 void AudioFlinger::onFirstRef()
 {
-    int rc = 0;
-
     Mutex::Autolock _l(mLock);
 
     /* TODO: move all this work into an Init() function */
@@ -1246,8 +1244,6 @@
 status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
         audio_io_handle_t output) const
 {
-    status_t status;
-
     Mutex::Autolock _l(mLock);
 
     PlaybackThread *playbackThread = checkPlaybackThread_l(output);
@@ -1306,7 +1302,7 @@
     bool removed = false;
     for (size_t i = 0; i< num; ) {
         AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
-        ALOGV(" pid %d @ %d", ref->mPid, i);
+        ALOGV(" pid %d @ %zu", ref->mPid, i);
         if (ref->mPid == pid) {
             ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid);
             mAudioSessionRefs.removeAt(i);
@@ -1410,10 +1406,6 @@
 
 // ----------------------------------------------------------------------------
 
-static bool deviceRequiresCaptureAudioOutputPermission(audio_devices_t inDevice) {
-    return audio_is_remote_submix_device(inDevice);
-}
-
 sp<IAudioRecord> AudioFlinger::openRecord(
         audio_io_handle_t input,
         uint32_t sampleRate,
@@ -1571,7 +1563,7 @@
 
     int rc = load_audio_interface(name, &dev);
     if (rc) {
-        ALOGI("loadHwModule() error %d loading module %s ", rc, name);
+        ALOGE("loadHwModule() error %d loading module %s", rc, name);
         return 0;
     }
 
@@ -1579,7 +1571,7 @@
     rc = dev->init_check(dev);
     mHardwareStatus = AUDIO_HW_IDLE;
     if (rc) {
-        ALOGI("loadHwModule() init check error %d for module %s ", rc, name);
+        ALOGE("loadHwModule() init check error %d for module %s", rc, name);
         return 0;
     }
 
@@ -1771,8 +1763,6 @@
         return 0;
     }
 
-    audio_hw_device_t *hwDevHal = outHwDev->hwDevice();
-
     if (*output == AUDIO_IO_HANDLE_NONE) {
         *output = nextUniqueId(AUDIO_UNIQUE_ID_USE_OUTPUT);
     } else {
@@ -2867,7 +2857,7 @@
 {
     audio_session_t session = chain->sessionId();
     ssize_t index = mOrphanEffectChains.indexOfKey(session);
-    ALOGV("putOrphanEffectChain_l session %d index %d", session, index);
+    ALOGV("putOrphanEffectChain_l session %d index %zd", session, index);
     if (index >= 0) {
         ALOGW("putOrphanEffectChain_l chain for session %d already present", session);
         return ALREADY_EXISTS;
@@ -2880,7 +2870,7 @@
 {
     sp<EffectChain> chain;
     ssize_t index = mOrphanEffectChains.indexOfKey(session);
-    ALOGV("getOrphanEffectChain_l session %d index %d", session, index);
+    ALOGV("getOrphanEffectChain_l session %d index %zd", session, index);
     if (index >= 0) {
         chain = mOrphanEffectChains.valueAt(index);
         mOrphanEffectChains.removeItemsAt(index);
@@ -2893,11 +2883,11 @@
     Mutex::Autolock _l(mLock);
     audio_session_t session = effect->sessionId();
     ssize_t index = mOrphanEffectChains.indexOfKey(session);
-    ALOGV("updateOrphanEffectChains session %d index %d", session, index);
+    ALOGV("updateOrphanEffectChains session %d index %zd", session, index);
     if (index >= 0) {
         sp<EffectChain> chain = mOrphanEffectChains.valueAt(index);
         if (chain->removeEffect_l(effect) == 0) {
-            ALOGV("updateOrphanEffectChains removing effect chain at index %d", index);
+            ALOGV("updateOrphanEffectChains removing effect chain at index %zd", index);
             mOrphanEffectChains.removeItemsAt(index);
         }
         return true;
diff --git a/services/audioflinger/AudioResamplerCubic.cpp b/services/audioflinger/AudioResamplerCubic.cpp
index 6a324ad..9c3c7cb 100644
--- a/services/audioflinger/AudioResamplerCubic.cpp
+++ b/services/audioflinger/AudioResamplerCubic.cpp
@@ -75,7 +75,6 @@
     int16_t *in = mBuffer.i16;
 
     while (outputIndex < outputSampleCount) {
-        int32_t sample;
         int32_t x;
 
         // calculate output sample
diff --git a/services/audioflinger/AudioResamplerDyn.cpp b/services/audioflinger/AudioResamplerDyn.cpp
index 618b56c..e615700 100644
--- a/services/audioflinger/AudioResamplerDyn.cpp
+++ b/services/audioflinger/AudioResamplerDyn.cpp
@@ -282,7 +282,6 @@
         return;
     }
     int32_t oldSampleRate = mInSampleRate;
-    int32_t oldHalfNumCoefs = mConstants.mHalfNumCoefs;
     uint32_t oldPhaseWrapLimit = mConstants.mL << mConstants.mShift;
     bool useS32 = false;
 
diff --git a/services/audioflinger/AudioResamplerSinc.cpp b/services/audioflinger/AudioResamplerSinc.cpp
index f600d6c..320b8cf 100644
--- a/services/audioflinger/AudioResamplerSinc.cpp
+++ b/services/audioflinger/AudioResamplerSinc.cpp
@@ -141,6 +141,8 @@
 
 // ----------------------------------------------------------------------------
 
+#if !USE_NEON
+
 static inline
 int32_t mulRL(int left, int32_t in, uint32_t vRL)
 {
@@ -202,6 +204,8 @@
 #endif
 }
 
+#endif // !USE_NEON
+
 // ----------------------------------------------------------------------------
 
 AudioResamplerSinc::AudioResamplerSinc(
diff --git a/services/audioflinger/Effects.cpp b/services/audioflinger/Effects.cpp
index 060ffe9..00304b2 100644
--- a/services/audioflinger/Effects.cpp
+++ b/services/audioflinger/Effects.cpp
@@ -138,7 +138,7 @@
     } else {
         status = ALREADY_EXISTS;
     }
-    ALOGV("addHandle() %p added handle %p in position %d", this, handle, i);
+    ALOGV("addHandle() %p added handle %p in position %zu", this, handle, i);
     mHandles.insertAt(handle, i);
     return status;
 }
@@ -156,7 +156,7 @@
     if (i == size) {
         return size;
     }
-    ALOGV("removeHandle() %p removed handle %p in position %d", this, handle, i);
+    ALOGV("removeHandle() %p removed handle %p in position %zu", this, handle, i);
 
     mHandles.removeAt(i);
     // if removed from first place, move effect control from this handle to next in line
@@ -380,7 +380,7 @@
     mConfig.inputCfg.buffer.frameCount = thread->frameCount();
     mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount;
 
-    ALOGV("configure() %p thread %p buffer %p framecount %d",
+    ALOGV("configure() %p thread %p buffer %p framecount %zu",
             this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount);
 
     status_t cmdStatus;
@@ -677,7 +677,6 @@
     if (isProcessEnabled() &&
             ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL ||
             (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) {
-        status_t cmdStatus;
         uint32_t volume[2];
         uint32_t *pVolume = NULL;
         uint32_t size = sizeof(volume);
@@ -934,7 +933,7 @@
 
     int len = s.length();
     if (s.length() > 2) {
-        char *str = s.lockBuffer(len);
+        (void) s.lockBuffer(len);
         s.unlockBuffer(len - 2);
     }
     return s;
@@ -1051,7 +1050,7 @@
     mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset);
     if (mCblkMemory == 0 ||
             (mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer())) == NULL) {
-        ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE +
+        ALOGE("not enough memory for Effect size=%zu", EFFECT_PARAM_BUFFER_SIZE +
                 sizeof(effect_param_cblk_t));
         mCblkMemory.clear();
         return;
@@ -1580,7 +1579,7 @@
         }
         mEffects.insertAt(effect, idx_insert);
 
-        ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this,
+        ALOGV("addEffect_l() effect %p, added in chain %p at rank %zu", effect.get(), this,
                 idx_insert);
     }
     effect->configure();
@@ -1612,7 +1611,7 @@
                 }
             }
             mEffects.removeAt(i);
-            ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(),
+            ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %zu", effect.get(),
                     this, i);
             break;
         }
@@ -1727,7 +1726,7 @@
     String8 result;
 
     size_t numEffects = mEffects.size();
-    snprintf(buffer, SIZE, "    %d effects for session %d\n", numEffects, mSessionId);
+    snprintf(buffer, SIZE, "    %zu effects for session %d\n", numEffects, mSessionId);
     result.append(buffer);
 
     if (numEffects) {
diff --git a/services/audioflinger/FastCapture.cpp b/services/audioflinger/FastCapture.cpp
index bb83858..d202169 100644
--- a/services/audioflinger/FastCapture.cpp
+++ b/services/audioflinger/FastCapture.cpp
@@ -104,8 +104,10 @@
         } else {
             mFormat = mInputSource->format();
             mSampleRate = Format_sampleRate(mFormat);
+#if !LOG_NDEBUG
             unsigned channelCount = Format_channelCount(mFormat);
             ALOG_ASSERT(channelCount >= 1 && channelCount <= FCC_8);
+#endif
         }
         dumpState->mSampleRate = mSampleRate;
         eitherChanged = true;
@@ -186,7 +188,6 @@
         ALOG_ASSERT(mPipeSink != NULL);
         ALOG_ASSERT(mReadBuffer != NULL);
         if (mReadBufferState < 0) {
-            unsigned channelCount = Format_channelCount(mFormat);
             memset(mReadBuffer, 0, frameCount * Format_frameSize(mFormat));
             mReadBufferState = frameCount;
         }
diff --git a/services/audioflinger/FastMixer.cpp b/services/audioflinger/FastMixer.cpp
index d31b8d3..26cd1f9 100644
--- a/services/audioflinger/FastMixer.cpp
+++ b/services/audioflinger/FastMixer.cpp
@@ -47,7 +47,6 @@
 /*static*/ const FastMixerState FastMixer::sInitial;
 
 FastMixer::FastMixer() : FastThread(),
-    mSlopNs(0),
     // mFastTrackNames
     // mGenerations
     mOutputSink(NULL),
@@ -338,6 +337,11 @@
 
     if ((command & FastMixerState::MIX) && (mMixer != NULL) && mIsWarm) {
         ALOG_ASSERT(mMixerBuffer != NULL);
+
+        // AudioMixer::mState.enabledTracks is undefined if mState.hook == process__validate,
+        // so we keep a side copy of enabledTracks
+        bool anyEnabledTracks = false;
+
         // for each track, update volume and check for underrun
         unsigned currentTrackMask = current->mTrackMask;
         while (currentTrackMask != 0) {
@@ -398,19 +402,26 @@
                     underruns.mBitFields.mPartial++;
                     underruns.mBitFields.mMostRecent = UNDERRUN_PARTIAL;
                     mMixer->enable(name);
+                    anyEnabledTracks = true;
                 }
             } else {
                 underruns.mBitFields.mFull++;
                 underruns.mBitFields.mMostRecent = UNDERRUN_FULL;
                 mMixer->enable(name);
+                anyEnabledTracks = true;
             }
             ftDump->mUnderruns = underruns;
             ftDump->mFramesReady = framesReady;
         }
 
-        // process() is CPU-bound
-        mMixer->process();
-        mMixerBufferState = MIXED;
+        if (anyEnabledTracks) {
+            // process() is CPU-bound
+            mMixer->process();
+            mMixerBufferState = MIXED;
+        } else if (mMixerBufferState != ZEROED) {
+            mMixerBufferState = UNDEFINED;
+        }
+
     } else if (mMixerBufferState == MIXED) {
         mMixerBufferState = UNDEFINED;
     }
@@ -422,7 +433,8 @@
         }
 
         if (mMasterMono.load()) {  // memory_order_seq_cst
-            mono_blend(mMixerBuffer, mMixerBufferFormat, Format_channelCount(mFormat), frameCount, true /*limit*/);
+            mono_blend(mMixerBuffer, mMixerBufferFormat, Format_channelCount(mFormat), frameCount,
+                    true /*limit*/);
         }
         // prepare the buffer used to write to sink
         void *buffer = mSinkBuffer != NULL ? mSinkBuffer : mMixerBuffer;
diff --git a/services/audioflinger/FastMixer.h b/services/audioflinger/FastMixer.h
index 3cc7c9f..bdfd8a0 100644
--- a/services/audioflinger/FastMixer.h
+++ b/services/audioflinger/FastMixer.h
@@ -57,7 +57,6 @@
     static const FastMixerState sInitial;
 
     FastMixerState  mPreIdle;   // copy of state before we went into idle
-    long            mSlopNs;    // accumulated time we've woken up too early (> 0) or too late (< 0)
     int             mFastTrackNames[FastMixerState::kMaxFastTracks];
                                 // handles used by mixer to identify tracks
     int             mGenerations[FastMixerState::kMaxFastTracks];
diff --git a/services/audioflinger/PatchPanel.cpp b/services/audioflinger/PatchPanel.cpp
index a99becf..bf6763f 100644
--- a/services/audioflinger/PatchPanel.cpp
+++ b/services/audioflinger/PatchPanel.cpp
@@ -401,7 +401,7 @@
         shift = playbackShift;
     }
     size_t frameCount = (playbackFrameCount * recordFramecount) >> shift;
-    ALOGV("createPatchConnections() playframeCount %d recordFramecount %d frameCount %d ",
+    ALOGV("createPatchConnections() playframeCount %zu recordFramecount %zu frameCount %zu",
           playbackFrameCount, recordFramecount, frameCount);
 
     // create a special record track to capture from record thread
@@ -614,7 +614,6 @@
 status_t AudioFlinger::PatchPanel::setAudioPortConfig(const struct audio_port_config *config)
 {
     ALOGV("setAudioPortConfig");
-    status_t status = NO_ERROR;
 
     sp<AudioFlinger> audioflinger = mAudioFlinger.promote();
     if (audioflinger == 0) {
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index e2932f1..973c983 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -707,8 +707,6 @@
 
 status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs)
 {
-    status_t status;
-
     ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string());
     Mutex::Autolock _l(mLock);
 
@@ -727,7 +725,7 @@
         return status;
     }
     mConfigEvents.add(event);
-    ALOGV("sendConfigEvent_l() num events %d event %d", mConfigEvents.size(), event->mType);
+    ALOGV("sendConfigEvent_l() num events %zu event %d", mConfigEvents.size(), event->mType);
     mWaitWorkCV.signal();
     mLock.unlock();
     {
@@ -819,7 +817,7 @@
     bool configChanged = false;
 
     while (!mConfigEvents.isEmpty()) {
-        ALOGV("processConfigEvents_l() remaining events %d", mConfigEvents.size());
+        ALOGV("processConfigEvents_l() remaining events %zu", mConfigEvents.size());
         sp<ConfigEvent> event = mConfigEvents[0];
         mConfigEvents.removeAt(0);
         switch (event->mType) {
@@ -918,7 +916,7 @@
         }
         const int len = s.length();
         if (len > 2) {
-            char *str = s.lockBuffer(len); // needed?
+            (void) s.lockBuffer(len);      // needed?
             s.unlockBuffer(len - 2);       // remove trailing ", "
         }
         return s;
@@ -951,7 +949,7 @@
     dprintf(fd, "  Sample rate: %u Hz\n", mSampleRate);
     dprintf(fd, "  HAL frame count: %zu\n", mFrameCount);
     dprintf(fd, "  HAL format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat));
-    dprintf(fd, "  HAL buffer size: %u bytes\n", mBufferSize);
+    dprintf(fd, "  HAL buffer size: %zu bytes\n", mBufferSize);
     dprintf(fd, "  Channel count: %u\n", mChannelCount);
     dprintf(fd, "  Channel mask: 0x%08x (%s)\n", mChannelMask,
             channelMaskToString(mChannelMask, mType != RECORD).string());
@@ -1687,10 +1685,10 @@
 
     size_t numtracks = mTracks.size();
     size_t numactive = mActiveTracks.size();
-    dprintf(fd, "  %d Tracks", numtracks);
+    dprintf(fd, "  %zu Tracks", numtracks);
     size_t numactiveseen = 0;
     if (numtracks) {
-        dprintf(fd, " of which %d are active\n", numactive);
+        dprintf(fd, " of which %zu are active\n", numactive);
         Track::appendDumpHeader(result);
         for (size_t i = 0; i < numtracks; ++i) {
             sp<Track> track = mTracks[i];
@@ -1731,7 +1729,8 @@
     dumpBase(fd, args);
 
     dprintf(fd, "  Normal frame count: %zu\n", mNormalFrameCount);
-    dprintf(fd, "  Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime));
+    dprintf(fd, "  Last write occurred (msecs): %llu\n",
+            (unsigned long long) ns2ms(systemTime() - mLastWriteTime));
     dprintf(fd, "  Total writes: %d\n", mNumWrites);
     dprintf(fd, "  Delayed writes: %d\n", mNumDelayedWrites);
     dprintf(fd, "  Blocked in write: %s\n", mInWrite ? "yes" : "no");
@@ -1825,11 +1824,11 @@
             }
             frameCount = mFrameCount * sFastTrackMultiplier;
         }
-        ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d",
+        ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%zu mFrameCount=%zu",
                 frameCount, mFrameCount);
       } else {
-        ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: sharedBuffer=%p frameCount=%d "
-                "mFrameCount=%d format=%#x mFormat=%#x isLinear=%d channelMask=%#x "
+        ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: sharedBuffer=%p frameCount=%zu "
+                "mFrameCount=%zu format=%#x mFormat=%#x isLinear=%d channelMask=%#x "
                 "sampleRate=%u mSampleRate=%u "
                 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x",
                 sharedBuffer.get(), frameCount, mFrameCount, format, mFormat,
@@ -2261,7 +2260,7 @@
     mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common);
     mFrameCount = mBufferSize / mFrameSize;
     if (mFrameCount & 15) {
-        ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames",
+        ALOGW("HAL output buffer size is %zu frames but AudioMixer requires multiples of 16 frames",
                 mFrameCount);
     }
 
@@ -2347,7 +2346,7 @@
     if (mType == MIXER || mType == DUPLICATING) {
         mNormalFrameCount = (mNormalFrameCount + 15) & ~15;
     }
-    ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount,
+    ALOGI("HAL output buffer size %zu frames, normal sink buffer size %zu frames", mFrameCount,
             mNormalFrameCount);
 
     // Check if we want to throttle the processing to no more than 2x normal rate
@@ -2684,7 +2683,7 @@
 
 void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType)
 {
-    ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d",
+    ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %zu",
             this,  streamType, mTracks.size());
     Mutex::Autolock _l(mLock);
 
@@ -3128,7 +3127,7 @@
                         if ((now - lastWarning) > kWarningThrottleNs) {
                             ATRACE_NAME("underrun");
                             ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p",
-                                    ns2ms(delta), mNumDelayedWrites, this);
+                                    (unsigned long long) ns2ms(delta), mNumDelayedWrites, this);
                             lastWarning = now;
                         }
                     }
@@ -3473,8 +3472,8 @@
         // mNormalSink below
 {
     ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type);
-    ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, "
-            "mFrameCount=%d, mNormalFrameCount=%d",
+    ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%zu, "
+            "mFrameCount=%zu, mNormalFrameCount=%zu",
             mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount,
             mNormalFrameCount);
     mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate);
@@ -3489,7 +3488,12 @@
     mOutputSink = new AudioStreamOutSink(output->stream);
     size_t numCounterOffers = 0;
     const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)};
-    ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers);
+#if !LOG_NDEBUG
+    ssize_t index =
+#else
+    (void)
+#endif
+            mOutputSink->negotiate(offers, 1, NULL, numCounterOffers);
     ALOG_ASSERT(index == 0);
 
     // initialize fast mixer depending on configuration
@@ -3524,7 +3528,9 @@
 
         // create a MonoPipe to connect our submix to FastMixer
         NBAIO_Format format = mOutputSink->format();
+#ifdef TEE_SINK
         NBAIO_Format origformat = format;
+#endif
         // adjust format to match that of the Fast Mixer
         ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat);
         format.mFormat = fastMixerFormat;
@@ -3536,7 +3542,12 @@
         MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/);
         const NBAIO_Format offers[1] = {format};
         size_t numCounterOffers = 0;
-        ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers);
+#if !LOG_NDEBUG
+        ssize_t index =
+#else
+        (void)
+#endif
+                monoPipe->negotiate(offers, 1, NULL, numCounterOffers);
         ALOG_ASSERT(index == 0);
         monoPipe->setAvgFrames((mScreenState & 1) ?
                 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2);
@@ -4341,7 +4352,6 @@
         }
 
         }   // local variable scope to avoid goto warning
-track_is_ready: ;
 
     }
 
@@ -4578,10 +4588,6 @@
 
 void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args)
 {
-    const size_t SIZE = 256;
-    char buffer[SIZE];
-    String8 result;
-
     PlaybackThread::dumpInternals(fd, args);
     dprintf(fd, "  Thread throttle time (msecs): %u\n", mThreadThrottleTimeMs);
     dprintf(fd, "  AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames());
@@ -4662,7 +4668,6 @@
 
 void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack)
 {
-    audio_track_cblk_t* cblk = track->cblk();
     float left, right;
 
     if (mMasterMute || mStreamTypes[track->streamType()].mute) {
@@ -4751,7 +4756,9 @@
         }
 
         Track* const track = t.get();
+#ifdef VERY_VERY_VERBOSE_LOGGING
         audio_track_cblk_t* cblk = track->cblk();
+#endif
         // Only consider last track started for volume and mixer state control.
         // In theory an older track could underrun and restart after the new one starts
         // but as we only care about the transition phase between two tracks on a
@@ -4959,7 +4966,7 @@
             // For compressed offload, use faster sleep time when underruning until more than an
             // entire buffer was written to the audio HAL
             if (!audio_has_proportional_frames(mFormat) &&
-                    (mType == OFFLOAD) && (mBytesWritten < mBufferSize)) {
+                    (mType == OFFLOAD) && (mBytesWritten < (int64_t) mBufferSize)) {
                 mSleepTimeUs = kDirectMinSleepTimeUs;
             } else {
                 mSleepTimeUs = mActiveSleepTimeUs;
@@ -4996,6 +5003,10 @@
     bool trackPaused = false;
     bool trackStopped = false;
 
+    if ((mType == DIRECT) && audio_is_linear_pcm(mFormat) && !usesHwAvSync()) {
+        return !mStandby;
+    }
+
     // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack
     // after a timeout and we will enter standby then.
     if (mTracks.size() > 0) {
@@ -5266,7 +5277,7 @@
     bool doHwPause = false;
     bool doHwResume = false;
 
-    ALOGV("OffloadThread::prepareTracks_l active tracks %d", count);
+    ALOGV("OffloadThread::prepareTracks_l active tracks %zu", count);
 
     // find out which tracks need to be processed
     for (size_t i = 0; i < count; i++) {
@@ -5276,7 +5287,9 @@
             continue;
         }
         Track* const track = t.get();
+#ifdef VERY_VERY_VERBOSE_LOGGING
         audio_track_cblk_t* cblk = track->cblk();
+#endif
         // Only consider last track started for volume and mixer state control.
         // In theory an older track could underrun and restart after the new one starts
         // but as we only care about the transition phase between two tracks on a
@@ -5728,7 +5741,12 @@
     mInputSource = new AudioStreamInSource(input->stream);
     size_t numCounterOffers = 0;
     const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)};
-    ssize_t index = mInputSource->negotiate(offers, 1, NULL, numCounterOffers);
+#if !LOG_NDEBUG
+    ssize_t index =
+#else
+    (void)
+#endif
+            mInputSource->negotiate(offers, 1, NULL, numCounterOffers);
     ALOG_ASSERT(index == 0);
 
     // initialize fast capture depending on configuration
@@ -5806,7 +5824,7 @@
         // start the fast capture
         mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO);
         pid_t tid = mFastCapture->getTid();
-        sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer);
+        sendPrioConfigEvent(getpid_cached, tid, kPriorityFastCapture);
 #ifdef AUDIO_WATCHDOG
         // FIXME
 #endif
@@ -6078,8 +6096,10 @@
             }
         // otherwise use the HAL / AudioStreamIn directly
         } else {
+            ATRACE_BEGIN("read");
             ssize_t bytesRead = mInput->stream->read(mInput->stream,
                     (uint8_t*)mRsmpInBuffer + rear * mFrameSize, mBufferSize);
+            ATRACE_END();
             if (bytesRead < 0) {
                 framesRead = bytesRead;
             } else {
@@ -6110,7 +6130,7 @@
         // ALOGD("%s", mTimestamp.toString().c_str());
 
         if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) {
-            ALOGE("read failed: framesRead=%d", framesRead);
+            ALOGE("read failed: framesRead=%zd", framesRead);
             // Force input into standby so that it tries to recover at next read attempt
             inputStandBy();
             sleepUs = kRecordThreadSleepUs;
@@ -6346,10 +6366,10 @@
             // there are sufficient fast track slots available
             mFastTrackAvail
         ) {
-        ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%u mFrameCount=%u",
+        ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%zu mFrameCount=%zu",
                 frameCount, mFrameCount);
       } else {
-        ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%u mFrameCount=%u mPipeFramesP2=%u "
+        ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%zu mFrameCount=%zu mPipeFramesP2=%zu "
                 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u "
                 "hasFastCapture=%d tid=%d mFastTrackAvail=%d",
                 frameCount, mFrameCount, mPipeFramesP2,
@@ -6642,9 +6662,9 @@
     size_t numtracks = mTracks.size();
     size_t numactive = mActiveTracks.size();
     size_t numactiveseen = 0;
-    dprintf(fd, "  %d Tracks", numtracks);
+    dprintf(fd, "  %zu Tracks", numtracks);
     if (numtracks) {
-        dprintf(fd, " of which %d are active\n", numactive);
+        dprintf(fd, " of which %zu are active\n", numactive);
         RecordTrack::appendDumpHeader(result);
         for (size_t i = 0; i < numtracks ; ++i) {
             sp<RecordTrack> track = mTracks[i];
@@ -7308,7 +7328,7 @@
 {
     ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this);
     ALOGW_IF(mEffectChains.size() != 1,
-            "removeEffectChain_l() %p invalid chain size %d on thread %p",
+            "removeEffectChain_l() %p invalid chain size %zu on thread %p",
             chain.get(), mEffectChains.size(), this);
     if (mEffectChains.size() == 1) {
         mEffectChains.removeAt(0);
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index f575918..7cbb6b8 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -122,7 +122,7 @@
         mCblkMemory = client->heap()->allocate(size);
         if (mCblkMemory == 0 ||
                 (mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer())) == NULL) {
-            ALOGE("not enough memory for AudioTrack size=%u", size);
+            ALOGE("not enough memory for AudioTrack size=%zu", size);
             client->heap()->dump("AudioTrack");
             mCblkMemory.clear();
             return;
@@ -959,9 +959,9 @@
     if (isOffloaded()) {
         complete = true;
     } else if (isDirect() || isFastTrack()) { // these do not go through linear map
-        complete = framesWritten >= mPresentationCompleteFrames;
+        complete = framesWritten >= (int64_t) mPresentationCompleteFrames;
     } else {  // Normal tracks, OutputTracks, and PatchTracks
-        complete = framesWritten >= mPresentationCompleteFrames
+        complete = framesWritten >= (int64_t) mPresentationCompleteFrames
                 && mAudioTrackServerProxy->isDrained();
     }
 
@@ -1016,7 +1016,7 @@
     if (isTerminated() || mState == PAUSED ||
             ((framesReady() == 0) && ((mSharedBuffer != 0) ||
                                       (mState == STOPPED)))) {
-        ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ",
+        ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %zu",
               mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady());
         event->cancel();
         return INVALID_OPERATION;
@@ -1137,7 +1137,7 @@
         mOutBuffer.frameCount = 0;
         playbackThread->mTracks.add(this);
         ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, "
-                "frameCount %u, mChannelMask 0x%08x",
+                "frameCount %zu, mChannelMask 0x%08x",
                 mCblk, mBuffer,
                 frameCount, mChannelMask);
         // since client and server are in the same process,
@@ -1246,7 +1246,7 @@
                 mBufferQueue.removeAt(0);
                 free(pInBuffer->mBuffer);
                 delete pInBuffer;
-                ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this,
+                ALOGV("OutputTrack::write() %p thread %p released overflow buffer %zu", this,
                         mThread.unsafe_get(), mBufferQueue.size());
             } else {
                 break;
@@ -1265,7 +1265,7 @@
                 pInBuffer->raw = pInBuffer->mBuffer;
                 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * mFrameSize);
                 mBufferQueue.add(pInBuffer);
-                ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this,
+                ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %zu", this,
                         mThread.unsafe_get(), mBufferQueue.size());
             } else {
                 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers",
diff --git a/services/audioflinger/audio-resampler/Android.mk b/services/audioflinger/audio-resampler/Android.mk
index ba37b19..bb2807c 100644
--- a/services/audioflinger/audio-resampler/Android.mk
+++ b/services/audioflinger/audio-resampler/Android.mk
@@ -11,4 +11,6 @@
 
 LOCAL_SHARED_LIBRARIES  := libutils liblog
 
+LOCAL_CFLAGS += -Werror -Wall
+
 include $(BUILD_SHARED_LIBRARY)
diff --git a/services/audioflinger/tests/Android.mk b/services/audioflinger/tests/Android.mk
index 6182de0..3505e0f 100644
--- a/services/audioflinger/tests/Android.mk
+++ b/services/audioflinger/tests/Android.mk
@@ -23,6 +23,8 @@
 LOCAL_MODULE := resampler_tests
 LOCAL_MODULE_TAGS := tests
 
+LOCAL_CFLAGS := -Werror -Wall
+
 include $(BUILD_NATIVE_TEST)
 
 #
@@ -61,4 +63,6 @@
 
 LOCAL_CXX_STL := libc++
 
+LOCAL_CFLAGS := -Werror -Wall
+
 include $(BUILD_EXECUTABLE)
diff --git a/services/audiopolicy/Android.mk b/services/audiopolicy/Android.mk
index 8218edd..8b45adc 100644
--- a/services/audiopolicy/Android.mk
+++ b/services/audiopolicy/Android.mk
@@ -45,13 +45,14 @@
     libmedia_helper \
     libaudiopolicycomponents
 
+LOCAL_MULTILIB := $(AUDIOSERVER_MULTILIB)
+
 LOCAL_MODULE:= libaudiopolicyservice
 
 LOCAL_CFLAGS += -fvisibility=hidden
 
 include $(BUILD_SHARED_LIBRARY)
 
-
 ifneq ($(USE_LEGACY_AUDIO_POLICY), 1)
 
 include $(CLEAR_VARS)
@@ -101,6 +102,8 @@
 LOCAL_CFLAGS += -DUSE_XML_AUDIO_POLICY_CONF
 endif #ifeq ($(USE_XML_AUDIO_POLICY_CONF), 1)
 
+LOCAL_MULTILIB := $(AUDIOSERVER_MULTILIB)
+
 LOCAL_MODULE:= libaudiopolicymanagerdefault
 
 include $(BUILD_SHARED_LIBRARY)
@@ -122,6 +125,8 @@
     $(TOPDIR)frameworks/av/services/audiopolicy/common/include \
     $(TOPDIR)frameworks/av/services/audiopolicy/engine/interface
 
+LOCAL_MULTILIB := $(AUDIOSERVER_MULTILIB)
+
 LOCAL_MODULE:= libaudiopolicymanager
 
 include $(BUILD_SHARED_LIBRARY)
diff --git a/services/audiopolicy/common/managerdefinitions/Android.mk b/services/audiopolicy/common/managerdefinitions/Android.mk
index 5c81410..3b4ae6b 100644
--- a/services/audiopolicy/common/managerdefinitions/Android.mk
+++ b/services/audiopolicy/common/managerdefinitions/Android.mk
@@ -58,6 +58,8 @@
 LOCAL_EXPORT_C_INCLUDE_DIRS := \
     $(LOCAL_PATH)/include
 
+LOCAL_MULTILIB := $(AUDIOSERVER_MULTILIB)
+
 LOCAL_MODULE := libaudiopolicycomponents
 
 include $(BUILD_STATIC_LIBRARY)
diff --git a/services/audiopolicy/engineconfigurable/Android.mk b/services/audiopolicy/engineconfigurable/Android.mk
index e6b5f85..6dba75b 100755
--- a/services/audiopolicy/engineconfigurable/Android.mk
+++ b/services/audiopolicy/engineconfigurable/Android.mk
@@ -35,6 +35,7 @@
     $(call include-path-for, audio-utils) \
     $(TOPDIR)frameworks/av/services/audiopolicy/common/include
 
+LOCAL_MULTILIB := $(AUDIOSERVER_MULTILIB)
 
 LOCAL_MODULE := libaudiopolicyengineconfigurable
 LOCAL_MODULE_TAGS := optional
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/plugin/Android.mk b/services/audiopolicy/engineconfigurable/parameter-framework/plugin/Android.mk
index 6348648..0e44f2c 100755
--- a/services/audiopolicy/engineconfigurable/parameter-framework/plugin/Android.mk
+++ b/services/audiopolicy/engineconfigurable/parameter-framework/plugin/Android.mk
@@ -30,6 +30,8 @@
     libparameter \
     liblog \
 
+LOCAL_MULTILIB := $(AUDIOSERVER_MULTILIB)
+
 LOCAL_STATIC_LIBRARIES := libpfw_utility
 
 LOCAL_MODULE_TAGS := optional
diff --git a/services/audiopolicy/engineconfigurable/wrapper/Android.mk b/services/audiopolicy/engineconfigurable/wrapper/Android.mk
index 096f913..f4283a8 100644
--- a/services/audiopolicy/engineconfigurable/wrapper/Android.mk
+++ b/services/audiopolicy/engineconfigurable/wrapper/Android.mk
@@ -18,6 +18,8 @@
 LOCAL_STATIC_LIBRARIES := \
     libmedia_helper \
 
+LOCAL_MULTILIB := $(AUDIOSERVER_MULTILIB)
+
 LOCAL_MODULE:= libaudiopolicypfwwrapper
 LOCAL_EXPORT_C_INCLUDE_DIRS := $(LOCAL_PATH)/include
 
diff --git a/services/audiopolicy/enginedefault/Android.mk b/services/audiopolicy/enginedefault/Android.mk
index bb12714..85d1822 100755
--- a/services/audiopolicy/enginedefault/Android.mk
+++ b/services/audiopolicy/enginedefault/Android.mk
@@ -29,6 +29,7 @@
     $(call include-path-for, bionic) \
     $(TOPDIR)frameworks/av/services/audiopolicy/common/include
 
+LOCAL_MULTILIB := $(AUDIOSERVER_MULTILIB)
 
 LOCAL_MODULE := libaudiopolicyenginedefault
 LOCAL_MODULE_TAGS := optional
diff --git a/services/audiopolicy/enginedefault/src/Engine.cpp b/services/audiopolicy/enginedefault/src/Engine.cpp
index f8ba3f2..f2224fd 100755
--- a/services/audiopolicy/enginedefault/src/Engine.cpp
+++ b/services/audiopolicy/enginedefault/src/Engine.cpp
@@ -231,20 +231,20 @@
 
 audio_devices_t Engine::getDeviceForStrategy(routing_strategy strategy) const
 {
-    const DeviceVector &availableOutputDevices = mApmObserver->getAvailableOutputDevices();
-    const DeviceVector &availableInputDevices = mApmObserver->getAvailableInputDevices();
+    DeviceVector availableOutputDevices = mApmObserver->getAvailableOutputDevices();
+    DeviceVector availableInputDevices = mApmObserver->getAvailableInputDevices();
 
     const SwAudioOutputCollection &outputs = mApmObserver->getOutputs();
 
-    return getDeviceForStrategyInt(strategy, (DeviceVector&)availableOutputDevices,
+    return getDeviceForStrategyInt(strategy, availableOutputDevices,
                                    availableInputDevices, outputs);
 }
 
 
 
 audio_devices_t Engine::getDeviceForStrategyInt(routing_strategy strategy,
-                                                DeviceVector &availableOutputDevices,
-                                                const DeviceVector &availableInputDevices,
+                                                DeviceVector availableOutputDevices,
+                                                DeviceVector availableInputDevices,
                                                 const SwAudioOutputCollection &outputs) const
 {
     uint32_t device = AUDIO_DEVICE_NONE;
diff --git a/services/audiopolicy/enginedefault/src/Engine.h b/services/audiopolicy/enginedefault/src/Engine.h
index ed93d1c..606ad28 100755
--- a/services/audiopolicy/enginedefault/src/Engine.h
+++ b/services/audiopolicy/enginedefault/src/Engine.h
@@ -126,8 +126,8 @@
     routing_strategy getStrategyForUsage(audio_usage_t usage);
     audio_devices_t getDeviceForStrategy(routing_strategy strategy) const;
     audio_devices_t getDeviceForStrategyInt(routing_strategy strategy,
-                                            DeviceVector &availableOutputDevices,
-                                            const DeviceVector &availableInputDevices,
+                                            DeviceVector availableOutputDevices,
+                                            DeviceVector availableInputDevices,
                                             const SwAudioOutputCollection &outputs) const;
     audio_devices_t getDeviceForInputSource(audio_source_t inputSource) const;
     audio_mode_t mPhoneState;  /**< current phone state. */
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index b2b014a..6f52e35 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -5212,7 +5212,7 @@
     bool supportsIEC61937 = false;
     for (size_t formatIndex = 0; formatIndex < formats.size(); formatIndex++) {
         audio_format_t format = formats[formatIndex];
-        ALOGI("%s: original formats: #%x", __FUNCTION__, format);
+        ALOGI("%s: original formats: %#x", __FUNCTION__, format);
         switch (format) {
             case AUDIO_FORMAT_AC3:
             case AUDIO_FORMAT_E_AC3:
@@ -5243,7 +5243,7 @@
                 case AUDIO_FORMAT_DTS:
                 case AUDIO_FORMAT_DTS_HD:
                 case AUDIO_FORMAT_IEC61937:
-                    ALOGI("%s: remove #%x", __FUNCTION__, format);
+                    ALOGI("%s: remove %#x", __FUNCTION__, format);
                     formats.removeAt(formatIndex);
                     break;
                 default:
@@ -5274,7 +5274,7 @@
     // Just for debugging.
     for (size_t formatIndex = 0; formatIndex < formats.size(); formatIndex++) {
         audio_format_t format = formats[formatIndex];
-        ALOGI("%s: final formats: #%x", __FUNCTION__, format);
+        ALOGI("%s: final formats: %#x", __FUNCTION__, format);
     }
 }
 
diff --git a/services/camera/libcameraservice/CameraService.cpp b/services/camera/libcameraservice/CameraService.cpp
index af8fc74..0c88dad 100644
--- a/services/camera/libcameraservice/CameraService.cpp
+++ b/services/camera/libcameraservice/CameraService.cpp
@@ -912,7 +912,8 @@
 }
 
 Status CameraService::validateConnectLocked(const String8& cameraId,
-        const String8& clientName8, /*inout*/int& clientUid, /*inout*/int& clientPid) const {
+        const String8& clientName8, /*inout*/int& clientUid, /*inout*/int& clientPid,
+        /*out*/int& originalClientPid) const {
 
     int callingPid = getCallingPid();
     int callingUid = getCallingUid();
@@ -954,6 +955,7 @@
 
     // Only use passed in clientPid to check permission. Use calling PID as the client PID that's
     // connected to camera service directly.
+    originalClientPid = clientPid;
     clientPid = callingPid;
 
     if (!mModule) {
diff --git a/services/camera/libcameraservice/CameraService.h b/services/camera/libcameraservice/CameraService.h
index e29b01c..11b1351 100644
--- a/services/camera/libcameraservice/CameraService.h
+++ b/services/camera/libcameraservice/CameraService.h
@@ -488,8 +488,14 @@
     virtual void onFirstRef();
 
     // Check if we can connect, before we acquire the service lock.
+    // The returned originalClientPid is the PID of the original process that wants to connect to
+    // camera.
+    // The returned clientPid is the PID of the client that directly connects to camera.
+    // originalClientPid and clientPid are usually the same except when the application uses
+    // mediaserver to connect to camera (using MediaRecorder to connect to camera). In that case,
+    // clientPid is the PID of mediaserver and originalClientPid is the PID of the application.
     binder::Status validateConnectLocked(const String8& cameraId, const String8& clientName8,
-          /*inout*/int& clientUid, /*inout*/int& clientPid) const;
+          /*inout*/int& clientUid, /*inout*/int& clientPid, /*out*/int& originalClientPid) const;
 
     // Handle active client evictions, and update service state.
     // Only call with with mServiceLock held.
@@ -819,6 +825,8 @@
 
     String8 clientName8(clientPackageName);
 
+    int originalClientPid = 0;
+
     ALOGI("CameraService::connect call (PID %d \"%s\", camera ID %s) for HAL version %s and "
             "Camera API version %d", clientPid, clientName8.string(), cameraId.string(),
             (halVersion == -1) ? "default" : std::to_string(halVersion).c_str(),
@@ -840,7 +848,7 @@
 
         // Enforce client permissions and do basic sanity checks
         if(!(ret = validateConnectLocked(cameraId, clientName8,
-                /*inout*/clientUid, /*inout*/clientPid)).isOk()) {
+                /*inout*/clientUid, /*inout*/clientPid, /*out*/originalClientPid)).isOk()) {
             return ret;
         }
 
@@ -857,7 +865,7 @@
 
         sp<BasicClient> clientTmp = nullptr;
         std::shared_ptr<resource_policy::ClientDescriptor<String8, sp<BasicClient>>> partial;
-        if ((err = handleEvictionsLocked(cameraId, clientPid, effectiveApiLevel,
+        if ((err = handleEvictionsLocked(cameraId, originalClientPid, effectiveApiLevel,
                 IInterface::asBinder(cameraCb), clientName8, /*out*/&clientTmp,
                 /*out*/&partial)) != NO_ERROR) {
             switch (err) {
@@ -909,9 +917,30 @@
 
         if ((err = client->initialize(mModule)) != OK) {
             ALOGE("%s: Could not initialize client from HAL module.", __FUNCTION__);
-            return STATUS_ERROR_FMT(ERROR_INVALID_OPERATION,
-                    "Failed to initialize camera \"%s\": %s (%d)", cameraId.string(),
-                    strerror(-err), err);
+            // Errors could be from the HAL module open call or from AppOpsManager
+            switch(err) {
+                case BAD_VALUE:
+                    return STATUS_ERROR_FMT(ERROR_ILLEGAL_ARGUMENT,
+                            "Illegal argument to HAL module for camera \"%s\"", cameraId.string());
+                case -EBUSY:
+                    return STATUS_ERROR_FMT(ERROR_CAMERA_IN_USE,
+                            "Camera \"%s\" is already open", cameraId.string());
+                case -EUSERS:
+                    return STATUS_ERROR_FMT(ERROR_MAX_CAMERAS_IN_USE,
+                            "Too many cameras already open, cannot open camera \"%s\"",
+                            cameraId.string());
+                case PERMISSION_DENIED:
+                    return STATUS_ERROR_FMT(ERROR_PERMISSION_DENIED,
+                            "No permission to open camera \"%s\"", cameraId.string());
+                case -EACCES:
+                    return STATUS_ERROR_FMT(ERROR_DISABLED,
+                            "Camera \"%s\" disabled by policy", cameraId.string());
+                case -ENODEV:
+                default:
+                    return STATUS_ERROR_FMT(ERROR_INVALID_OPERATION,
+                            "Failed to initialize camera \"%s\": %s (%d)", cameraId.string(),
+                            strerror(-err), err);
+            }
         }
 
         // Update shim paremeters for legacy clients
diff --git a/services/camera/libcameraservice/api2/CameraDeviceClient.cpp b/services/camera/libcameraservice/api2/CameraDeviceClient.cpp
index 316cfda..51c8148 100644
--- a/services/camera/libcameraservice/api2/CameraDeviceClient.cpp
+++ b/services/camera/libcameraservice/api2/CameraDeviceClient.cpp
@@ -342,7 +342,11 @@
     }
 
     status_t err = mDevice->configureStreams(isConstrainedHighSpeed);
-    if (err != OK) {
+    if (err == BAD_VALUE) {
+        res = STATUS_ERROR_FMT(CameraService::ERROR_ILLEGAL_ARGUMENT,
+                "Camera %d: Unsupported set of inputs/outputs provided",
+                mCameraId);
+    } else if (err != OK) {
         res = STATUS_ERROR_FMT(CameraService::ERROR_INVALID_OPERATION,
                 "Camera %d: Error configuring streams: %s (%d)",
                 mCameraId, strerror(-err), err);
diff --git a/services/medialog/Android.mk b/services/medialog/Android.mk
index 03438bf..88f98cf 100644
--- a/services/medialog/Android.mk
+++ b/services/medialog/Android.mk
@@ -6,10 +6,12 @@
 
 LOCAL_SHARED_LIBRARIES := libmedia libbinder libutils liblog libnbaio
 
+LOCAL_MULTILIB := $(AUDIOSERVER_MULTILIB)
+
 LOCAL_MODULE:= libmedialogservice
 
-LOCAL_32_BIT_ONLY := true
-
 LOCAL_C_INCLUDES := $(call include-path-for, audio-utils)
 
+LOCAL_CFLAGS := -Werror -Wall
+
 include $(BUILD_SHARED_LIBRARY)
diff --git a/services/mediaresourcemanager/ResourceManagerService.cpp b/services/mediaresourcemanager/ResourceManagerService.cpp
index e1235b8..64534bf 100644
--- a/services/mediaresourcemanager/ResourceManagerService.cpp
+++ b/services/mediaresourcemanager/ResourceManagerService.cpp
@@ -96,8 +96,11 @@
     if (binder != NULL) {
         sp<IMediaResourceMonitor> service = interface_cast<IMediaResourceMonitor>(binder);
         for (size_t i = 0; i < resources.size(); ++i) {
-            service->notifyResourceGranted(pid, String16(asString(resources[i].mType)),
-                    String16(asString(resources[i].mSubType)), resources[i].mValue);
+            if (resources[i].mSubType == MediaResource::kAudioCodec) {
+                service->notifyResourceGranted(pid, IMediaResourceMonitor::TYPE_AUDIO_CODEC);
+            } else if (resources[i].mSubType == MediaResource::kVideoCodec) {
+                service->notifyResourceGranted(pid, IMediaResourceMonitor::TYPE_VIDEO_CODEC);
+            }
         }
     }
 }
diff --git a/services/radio/Android.mk b/services/radio/Android.mk
index 6aae31d..f5d74d3 100644
--- a/services/radio/Android.mk
+++ b/services/radio/Android.mk
@@ -33,6 +33,8 @@
 
 LOCAL_CFLAGS += -Wall -Wextra -Werror
 
+LOCAL_MULTILIB := $(AUDIOSERVER_MULTILIB)
+
 LOCAL_MODULE:= libradioservice
 
 include $(BUILD_SHARED_LIBRARY)
diff --git a/services/soundtrigger/Android.mk b/services/soundtrigger/Android.mk
index ecc49ae..e8e18b8 100644
--- a/services/soundtrigger/Android.mk
+++ b/services/soundtrigger/Android.mk
@@ -38,6 +38,8 @@
 LOCAL_C_INCLUDES += \
     $(TOPDIR)frameworks/av/services/audioflinger
 
+LOCAL_MULTILIB := $(AUDIOSERVER_MULTILIB)
+
 LOCAL_MODULE:= libsoundtriggerservice
 
 include $(BUILD_SHARED_LIBRARY)