diff --git a/drm/mediadrm/plugins/mock/Android.mk b/drm/mediadrm/plugins/mock/Android.mk
index ada23a2..26c245b 100644
--- a/drm/mediadrm/plugins/mock/Android.mk
+++ b/drm/mediadrm/plugins/mock/Android.mk
@@ -21,7 +21,8 @@
 
 LOCAL_MODULE := libmockdrmcryptoplugin
 
-LOCAL_MODULE_PATH := $(TARGET_OUT_VENDOR_SHARED_LIBRARIES)/mediadrm
+LOCAL_PROPRIETARY_MODULE := true
+LOCAL_MODULE_RELATIVE_PATH := mediadrm
 
 LOCAL_SHARED_LIBRARIES := \
     libutils liblog
diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
index fd86737..28fdfd4 100644
--- a/include/media/AudioSystem.h
+++ b/include/media/AudioSystem.h
@@ -118,6 +118,7 @@
 
     static bool routedToA2dpOutput(audio_stream_type_t streamType);
 
+    // return status NO_ERROR implies *buffSize > 0
     static status_t getInputBufferSize(uint32_t sampleRate, audio_format_t format,
         audio_channel_mask_t channelMask, size_t* buffSize);
 
diff --git a/include/media/IMediaHTTPConnection.h b/include/media/IMediaHTTPConnection.h
index e048b64..2a63eb7 100644
--- a/include/media/IMediaHTTPConnection.h
+++ b/include/media/IMediaHTTPConnection.h
@@ -38,6 +38,7 @@
     virtual ssize_t readAt(off64_t offset, void *data, size_t size) = 0;
     virtual off64_t getSize() = 0;
     virtual status_t getMIMEType(String8 *mimeType) = 0;
+    virtual status_t getUri(String8 *uri) = 0;
 
 private:
     DISALLOW_EVIL_CONSTRUCTORS(IMediaHTTPConnection);
diff --git a/libvideoeditor/lvpp/Android.mk b/libvideoeditor/lvpp/Android.mk
index 2286827..860d351 100755
--- a/libvideoeditor/lvpp/Android.mk
+++ b/libvideoeditor/lvpp/Android.mk
@@ -71,7 +71,6 @@
     $(TOP)/frameworks/av/media/libstagefright \
     $(TOP)/frameworks/av/media/libstagefright/include \
     $(TOP)/frameworks/av/media/libstagefright/rtsp \
-    $(call include-path-for, corecg graphics) \
     $(TOP)/frameworks/av/libvideoeditor/osal/inc \
     $(TOP)/frameworks/av/libvideoeditor/vss/common/inc \
     $(TOP)/frameworks/av/libvideoeditor/vss/mcs/inc \
diff --git a/libvideoeditor/vss/stagefrightshells/src/Android.mk b/libvideoeditor/vss/stagefrightshells/src/Android.mk
index e30b85d..9188942 100755
--- a/libvideoeditor/vss/stagefrightshells/src/Android.mk
+++ b/libvideoeditor/vss/stagefrightshells/src/Android.mk
@@ -33,7 +33,6 @@
     $(TOP)/frameworks/av/media/libstagefright \
     $(TOP)/frameworks/av/media/libstagefright/include \
     $(TOP)/frameworks/av/media/libstagefright/rtsp \
-    $(call include-path-for, corecg graphics) \
     $(TOP)/frameworks/av/libvideoeditor/lvpp \
     $(TOP)/frameworks/av/libvideoeditor/osal/inc \
     $(TOP)/frameworks/av/libvideoeditor/vss/inc \
diff --git a/media/libeffects/downmix/EffectDownmix.c b/media/libeffects/downmix/EffectDownmix.c
index 4ee05f2..d25dc9b 100644
--- a/media/libeffects/downmix/EffectDownmix.c
+++ b/media/libeffects/downmix/EffectDownmix.c
@@ -629,7 +629,9 @@
         return -EINVAL;
     }
 
-    memcpy(&pDwmModule->config, pConfig, sizeof(effect_config_t));
+    if (&pDwmModule->config != pConfig) {
+        memcpy(&pDwmModule->config, pConfig, sizeof(effect_config_t));
+    }
 
     if (init) {
         pDownmixer->type = DOWNMIX_TYPE_FOLD;
diff --git a/media/libeffects/visualizer/Android.mk b/media/libeffects/visualizer/Android.mk
index dd2d306..c92c543 100644
--- a/media/libeffects/visualizer/Android.mk
+++ b/media/libeffects/visualizer/Android.mk
@@ -17,7 +17,6 @@
 LOCAL_MODULE:= libvisualizer
 
 LOCAL_C_INCLUDES := \
-	$(call include-path-for, graphics corecg) \
 	$(call include-path-for, audio-effects)
 
 
diff --git a/media/libmedia/Android.mk b/media/libmedia/Android.mk
index e0acae6..f3770e4 100644
--- a/media/libmedia/Android.mk
+++ b/media/libmedia/Android.mk
@@ -72,7 +72,6 @@
 LOCAL_MODULE:= libmedia
 
 LOCAL_C_INCLUDES := \
-    $(call include-path-for, graphics corecg) \
     $(TOP)/frameworks/native/include/media/openmax \
     external/icu4c/common \
     external/icu4c/i18n \
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 700718d..4438dfd 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -41,30 +41,22 @@
         return BAD_VALUE;
     }
 
-    // default to 0 in case of error
-    *frameCount = 0;
-
-    size_t size = 0;
+    size_t size;
     status_t status = AudioSystem::getInputBufferSize(sampleRate, format, channelMask, &size);
     if (status != NO_ERROR) {
-        ALOGE("AudioSystem could not query the input buffer size; status %d", status);
-        return NO_INIT;
+        ALOGE("AudioSystem could not query the input buffer size for sampleRate %u, format %#x, "
+              "channelMask %#x; status %d", sampleRate, format, channelMask, status);
+        return status;
     }
 
-    if (size == 0) {
+    // We double the size of input buffer for ping pong use of record buffer.
+    // Assumes audio_is_linear_pcm(format)
+    if ((*frameCount = (size * 2) / (popcount(channelMask) * audio_bytes_per_sample(format))) == 0) {
         ALOGE("Unsupported configuration: sampleRate %u, format %#x, channelMask %#x",
             sampleRate, format, channelMask);
         return BAD_VALUE;
     }
 
-    // We double the size of input buffer for ping pong use of record buffer.
-    size <<= 1;
-
-    // Assumes audio_is_linear_pcm(format)
-    uint32_t channelCount = popcount(channelMask);
-    size /= channelCount * audio_bytes_per_sample(format);
-
-    *frameCount = size;
     return NO_ERROR;
 }
 
@@ -110,10 +102,8 @@
             mAudioRecordThread->requestExitAndWait();
             mAudioRecordThread.clear();
         }
-        if (mAudioRecord != 0) {
-            mAudioRecord->asBinder()->unlinkToDeath(mDeathNotifier, this);
-            mAudioRecord.clear();
-        }
+        mAudioRecord->asBinder()->unlinkToDeath(mDeathNotifier, this);
+        mAudioRecord.clear();
         IPCThreadState::self()->flushCommands();
         AudioSystem::releaseAudioSessionId(mSessionId, -1);
     }
@@ -133,6 +123,11 @@
         transfer_type transferType,
         audio_input_flags_t flags)
 {
+    ALOGV("set(): inputSource %d, sampleRate %u, format %#x, channelMask %#x, frameCount %d, "
+          "notificationFrames %d, sessionId %d, transferType %d, flags %#x",
+          inputSource, sampleRate, format, channelMask, frameCountInt, notificationFrames,
+          sessionId, transferType, flags);
+
     switch (transferType) {
     case TRANSFER_DEFAULT:
         if (cbf == NULL || threadCanCallJava) {
@@ -163,16 +158,15 @@
     }
     size_t frameCount = frameCountInt;
 
-    ALOGV("set(): sampleRate %u, channelMask %#x, frameCount %u", sampleRate, channelMask,
-            frameCount);
-
     AutoMutex lock(mLock);
 
+    // invariant that mAudioRecord != 0 is true only after set() returns successfully
     if (mAudioRecord != 0) {
         ALOGE("Track already in use");
         return INVALID_OPERATION;
     }
 
+    // handle default values first.
     if (inputSource == AUDIO_SOURCE_DEFAULT) {
         inputSource = AUDIO_SOURCE_MIC;
     }
@@ -209,15 +203,19 @@
     uint32_t channelCount = popcount(channelMask);
     mChannelCount = channelCount;
 
-    // Assumes audio_is_linear_pcm(format), else sizeof(uint8_t)
-    mFrameSize = channelCount * audio_bytes_per_sample(format);
+    if (audio_is_linear_pcm(format)) {
+        mFrameSize = channelCount * audio_bytes_per_sample(format);
+    } else {
+        mFrameSize = sizeof(uint8_t);
+    }
 
     // validate framecount
-    size_t minFrameCount = 0;
+    size_t minFrameCount;
     status_t status = AudioRecord::getMinFrameCount(&minFrameCount,
             sampleRate, format, channelMask);
     if (status != NO_ERROR) {
-        ALOGE("getMinFrameCount() failed; status %d", status);
+        ALOGE("getMinFrameCount() failed for sampleRate %u, format %#x, channelMask %#x; status %d",
+                sampleRate, format, channelMask, status);
         return status;
     }
     ALOGV("AudioRecord::set() minFrameCount = %d", minFrameCount);
@@ -258,7 +256,6 @@
 
     mActive = false;
     mCbf = cbf;
-    mRefreshRemaining = true;
     mUserData = user;
     // TODO: add audio hardware input latency here
     mLatency = (1000*mFrameCount) / sampleRate;
@@ -485,10 +482,12 @@
     ALOGE_IF(originalSessionId != AUDIO_SESSION_ALLOCATE && mSessionId != originalSessionId,
             "session ID changed from %d to %d", originalSessionId, mSessionId);
 
-    if (record == 0 || status != NO_ERROR) {
+    if (status != NO_ERROR) {
         ALOGE("AudioFlinger could not create record track, status: %d", status);
         goto release;
     }
+    ALOG_ASSERT(record != 0);
+
     // AudioFlinger now owns the reference to the I/O handle,
     // so we are no longer responsible for releasing it.
 
@@ -502,27 +501,21 @@
         ALOGE("Could not get control block pointer");
         return NO_INIT;
     }
+    // invariant that mAudioRecord != 0 is true only after set() returns successfully
     if (mAudioRecord != 0) {
         mAudioRecord->asBinder()->unlinkToDeath(mDeathNotifier, this);
         mDeathNotifier.clear();
     }
-
-    // We retain a copy of the I/O handle, but don't own the reference
-    mInput = input;
     mAudioRecord = record;
+
     mCblkMemory = iMem;
     audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer);
     mCblk = cblk;
-    // note that temp is the (possibly revised) value of mFrameCount
+    // note that temp is the (possibly revised) value of frameCount
     if (temp < frameCount || (frameCount == 0 && temp == 0)) {
         ALOGW("Requested frameCount %u but received frameCount %u", frameCount, temp);
     }
     frameCount = temp;
-    // If IAudioRecord is re-created, don't let the requested frameCount
-    // decrease.  This can confuse clients that cache frameCount().
-    if (frameCount > mReqFrameCount) {
-        mReqFrameCount = frameCount;
-    }
 
     // FIXME missing fast track frameCount logic
     mAwaitBoost = false;
@@ -544,10 +537,21 @@
         }
     }
 
-    // starting address of buffers in shared memory
+    // We retain a copy of the I/O handle, but don't own the reference
+    mInput = input;
+    mRefreshRemaining = true;
+
+    // Starting address of buffers in shared memory, immediately after the control block.  This
+    // address is for the mapping within client address space.  AudioFlinger::TrackBase::mBuffer
+    // is for the server address space.
     void *buffers = (char*)cblk + sizeof(audio_track_cblk_t);
 
     mFrameCount = frameCount;
+    // If IAudioRecord is re-created, don't let the requested frameCount
+    // decrease.  This can confuse clients that cache frameCount().
+    if (frameCount > mReqFrameCount) {
+        mReqFrameCount = frameCount;
+    }
 
     // update proxy
     mProxy = new AudioRecordClientProxy(cblk, buffers, mFrameCount, mFrameSize);
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 5c62260..adf3847 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -195,6 +195,11 @@
         int uid,
         pid_t pid)
 {
+    ALOGV("set(): streamType %d, sampleRate %u, format %#x, channelMask %#x, frameCount %d, "
+          "flags #%x, notificationFrames %d, sessionId %d, transferType %d",
+          streamType, sampleRate, format, channelMask, frameCountInt, flags, notificationFrames,
+          sessionId, transferType);
+
     switch (transferType) {
     case TRANSFER_DEFAULT:
         if (sharedBuffer != 0) {
@@ -288,6 +293,9 @@
         ALOGE("Invalid channel mask %#x", channelMask);
         return BAD_VALUE;
     }
+    mChannelMask = channelMask;
+    uint32_t channelCount = popcount(channelMask);
+    mChannelCount = channelCount;
 
     // AudioFlinger does not currently support 8-bit data in shared memory
     if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) {
@@ -311,10 +319,6 @@
         flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
     }
 
-    mChannelMask = channelMask;
-    uint32_t channelCount = popcount(channelMask);
-    mChannelCount = channelCount;
-
     if (audio_is_linear_pcm(format)) {
         mFrameSize = channelCount * audio_bytes_per_sample(format);
         mFrameSizeAF = channelCount * sizeof(int16_t);
@@ -1012,10 +1016,12 @@
                                                       mClientUid,
                                                       &status);
 
-    if (track == 0) {
+    if (status != NO_ERROR) {
         ALOGE("AudioFlinger could not create track, status: %d", status);
         goto release;
     }
+    ALOG_ASSERT(track != 0);
+
     // AudioFlinger now owns the reference to the I/O handle,
     // so we are no longer responsible for releasing it.
 
@@ -1035,6 +1041,7 @@
         mDeathNotifier.clear();
     }
     mAudioTrack = track;
+
     mCblkMemory = iMem;
     audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer);
     mCblk = cblk;
@@ -1046,6 +1053,7 @@
         ALOGW("Requested frameCount %u but received frameCount %u", frameCount, temp);
     }
     frameCount = temp;
+
     mAwaitBoost = false;
     if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
         if (trackFlags & IAudioFlinger::TRACK_FAST) {
@@ -1099,6 +1107,7 @@
     mAudioTrack->attachAuxEffect(mAuxEffectId);
     // FIXME don't believe this lie
     mLatency = afLatency + (1000*frameCount) / mSampleRate;
+
     mFrameCount = frameCount;
     // If IAudioTrack is re-created, don't let the requested frameCount
     // decrease.  This can confuse clients that cache frameCount().
diff --git a/media/libmedia/IMediaHTTPConnection.cpp b/media/libmedia/IMediaHTTPConnection.cpp
index 622d9cf..22c470a 100644
--- a/media/libmedia/IMediaHTTPConnection.cpp
+++ b/media/libmedia/IMediaHTTPConnection.cpp
@@ -33,6 +33,7 @@
     READ_AT,
     GET_SIZE,
     GET_MIME_TYPE,
+    GET_URI
 };
 
 struct BpMediaHTTPConnection : public BpInterface<IMediaHTTPConnection> {
@@ -147,6 +148,26 @@
         return OK;
     }
 
+    virtual status_t getUri(String8 *uri) {
+        *uri = String8("");
+
+        Parcel data, reply;
+        data.writeInterfaceToken(
+                IMediaHTTPConnection::getInterfaceDescriptor());
+
+        remote()->transact(GET_URI, data, &reply);
+
+        int32_t exceptionCode = reply.readExceptionCode();
+
+        if (exceptionCode) {
+            return UNKNOWN_ERROR;
+        }
+
+        *uri = String8(reply.readString16());
+
+        return OK;
+    }
+
 private:
     sp<IMemory> mMemory;
 };
diff --git a/media/libmediaplayerservice/Android.mk b/media/libmediaplayerservice/Android.mk
index 8f21632..4189a5e 100644
--- a/media/libmediaplayerservice/Android.mk
+++ b/media/libmediaplayerservice/Android.mk
@@ -45,7 +45,6 @@
     libstagefright_rtsp         \
 
 LOCAL_C_INCLUDES :=                                                 \
-    $(call include-path-for, graphics corecg)                       \
     $(TOP)/frameworks/av/media/libstagefright/include               \
     $(TOP)/frameworks/av/media/libstagefright/rtsp                  \
     $(TOP)/frameworks/av/media/libstagefright/wifi-display          \
diff --git a/media/libmediaplayerservice/StagefrightRecorder.cpp b/media/libmediaplayerservice/StagefrightRecorder.cpp
index 845a589..5b7a236 100644
--- a/media/libmediaplayerservice/StagefrightRecorder.cpp
+++ b/media/libmediaplayerservice/StagefrightRecorder.cpp
@@ -748,7 +748,7 @@
     return OK;
 }
 
-status_t StagefrightRecorder::prepare() {
+status_t StagefrightRecorder::prepareInternal() {
     ALOGV("prepare");
     if (mOutputFd < 0) {
         ALOGE("Output file descriptor is invalid");
@@ -794,6 +794,13 @@
     return status;
 }
 
+status_t StagefrightRecorder::prepare() {
+    if (mVideoSource == VIDEO_SOURCE_SURFACE) {
+        return prepareInternal();
+    }
+    return OK;
+}
+
 status_t StagefrightRecorder::start() {
     ALOGV("start");
     if (mOutputFd < 0) {
@@ -801,15 +808,20 @@
         return INVALID_OPERATION;
     }
 
-    // Get UID here for permission checking
-    mClientUid = IPCThreadState::self()->getCallingUid();
+    status_t status = OK;
+
+    if (mVideoSource != VIDEO_SOURCE_SURFACE) {
+        status = prepareInternal();
+        if (status != OK) {
+            return status;
+        }
+    }
+
     if (mWriter == NULL) {
         ALOGE("File writer is not avaialble");
         return UNKNOWN_ERROR;
     }
 
-    status_t status = OK;
-
     switch (mOutputFormat) {
         case OUTPUT_FORMAT_DEFAULT:
         case OUTPUT_FORMAT_THREE_GPP:
diff --git a/media/libmediaplayerservice/StagefrightRecorder.h b/media/libmediaplayerservice/StagefrightRecorder.h
index 7d6abd3..377d168 100644
--- a/media/libmediaplayerservice/StagefrightRecorder.h
+++ b/media/libmediaplayerservice/StagefrightRecorder.h
@@ -127,6 +127,7 @@
     sp<IGraphicBufferProducer> mGraphicBufferProducer;
     sp<ALooper> mLooper;
 
+    status_t prepareInternal();
     status_t setupMPEG4Recording();
     void setupMPEG4MetaData(sp<MetaData> *meta);
     status_t setupAMRRecording();
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
index d47ac98..a750ad0 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
@@ -1006,7 +1006,14 @@
                                     &NuPlayer::performScanSources));
                     }
 
-                    flushDecoder(audio, formatChange);
+                    sp<AMessage> newFormat = mSource->getFormat(audio);
+                    sp<Decoder> &decoder = audio ? mAudioDecoder : mVideoDecoder;
+                    if (formatChange && !decoder->supportsSeamlessFormatChange(newFormat)) {
+                        flushDecoder(audio, /* needShutdown = */ true);
+                    } else {
+                        flushDecoder(audio, /* needShutdown = */ false);
+                        err = OK;
+                    }
                 } else {
                     // This stream is unaffected by the discontinuity
 
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp
index 22f699e..2423fd5 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp
@@ -67,6 +67,7 @@
     // queue.
     bool needDedicatedLooper = !strncasecmp(mime.c_str(), "video/", 6);
 
+    mFormat = format;
     mCodec = new ACodec;
 
     if (needDedicatedLooper && mCodecLooper == NULL) {
@@ -147,5 +148,65 @@
     }
 }
 
+bool NuPlayer::Decoder::supportsSeamlessAudioFormatChange(const sp<AMessage> &targetFormat) const {
+    if (targetFormat == NULL) {
+        return true;
+    }
+
+    AString mime;
+    if (!targetFormat->findString("mime", &mime)) {
+        return false;
+    }
+
+    if (!strcasecmp(mime.c_str(), MEDIA_MIMETYPE_AUDIO_AAC)) {
+        // field-by-field comparison
+        const char * keys[] = { "channel-count", "sample-rate", "is-adts" };
+        for (unsigned int i = 0; i < sizeof(keys) / sizeof(keys[0]); i++) {
+            int32_t oldVal, newVal;
+            if (!mFormat->findInt32(keys[i], &oldVal) || !targetFormat->findInt32(keys[i], &newVal)
+                    || oldVal != newVal) {
+                return false;
+            }
+        }
+
+        sp<ABuffer> oldBuf, newBuf;
+        if (mFormat->findBuffer("csd-0", &oldBuf) && targetFormat->findBuffer("csd-0", &newBuf)) {
+            if (oldBuf->size() != newBuf->size()) {
+                return false;
+            }
+            return !memcmp(oldBuf->data(), newBuf->data(), oldBuf->size());
+        }
+    }
+    return false;
+}
+
+bool NuPlayer::Decoder::supportsSeamlessFormatChange(const sp<AMessage> &targetFormat) const {
+    if (mFormat == NULL) {
+        return false;
+    }
+
+    if (targetFormat == NULL) {
+        return true;
+    }
+
+    AString oldMime, newMime;
+    if (!mFormat->findString("mime", &oldMime)
+            || !targetFormat->findString("mime", &newMime)
+            || !(oldMime == newMime)) {
+        return false;
+    }
+
+    bool audio = !strncasecmp(oldMime.c_str(), "audio/", strlen("audio/"));
+    bool seamless;
+    if (audio) {
+        seamless = supportsSeamlessAudioFormatChange(targetFormat);
+    } else {
+        seamless = mCodec != NULL && mCodec->isConfiguredForAdaptivePlayback();
+    }
+
+    ALOGV("%s seamless support for %s", seamless ? "yes" : "no", oldMime.c_str());
+    return seamless;
+}
+
 }  // namespace android
 
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h
index a876148..78ea74a 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h
@@ -36,6 +36,8 @@
     void signalResume();
     void initiateShutdown();
 
+    bool supportsSeamlessFormatChange(const sp<AMessage> &to) const;
+
 protected:
     virtual ~Decoder();
 
@@ -49,6 +51,7 @@
     sp<AMessage> mNotify;
     sp<NativeWindowWrapper> mNativeWindow;
 
+    sp<AMessage> mFormat;
     sp<ACodec> mCodec;
     sp<ALooper> mCodecLooper;
 
@@ -59,6 +62,8 @@
 
     void onFillThisBuffer(const sp<AMessage> &msg);
 
+    bool supportsSeamlessAudioFormatChange(const sp<AMessage> &targetFormat) const;
+
     DISALLOW_EVIL_CONSTRUCTORS(Decoder);
 };
 
diff --git a/media/libstagefright/MPEG4Extractor.cpp b/media/libstagefright/MPEG4Extractor.cpp
index f80772a..2a3fa04 100644
--- a/media/libstagefright/MPEG4Extractor.cpp
+++ b/media/libstagefright/MPEG4Extractor.cpp
@@ -913,6 +913,8 @@
 
         case FOURCC('e', 'l', 's', 't'):
         {
+            *offset += chunk_size;
+
             // See 14496-12 8.6.6
             uint8_t version;
             if (mDataSource->readAt(data_offset, &version, 1) < 1) {
@@ -975,12 +977,13 @@
                     mLastTrack->meta->setInt32(kKeyEncoderPadding, paddingsamples);
                 }
             }
-            *offset += chunk_size;
             break;
         }
 
         case FOURCC('f', 'r', 'm', 'a'):
         {
+            *offset += chunk_size;
+
             uint32_t original_fourcc;
             if (mDataSource->readAt(data_offset, &original_fourcc, 4) < 4) {
                 return ERROR_IO;
@@ -994,12 +997,13 @@
                 mLastTrack->meta->setInt32(kKeyChannelCount, num_channels);
                 mLastTrack->meta->setInt32(kKeySampleRate, sample_rate);
             }
-            *offset += chunk_size;
             break;
         }
 
         case FOURCC('t', 'e', 'n', 'c'):
         {
+            *offset += chunk_size;
+
             if (chunk_size < 32) {
                 return ERROR_MALFORMED;
             }
@@ -1044,23 +1048,25 @@
             mLastTrack->meta->setInt32(kKeyCryptoMode, defaultAlgorithmId);
             mLastTrack->meta->setInt32(kKeyCryptoDefaultIVSize, defaultIVSize);
             mLastTrack->meta->setData(kKeyCryptoKey, 'tenc', defaultKeyId, 16);
-            *offset += chunk_size;
             break;
         }
 
         case FOURCC('t', 'k', 'h', 'd'):
         {
+            *offset += chunk_size;
+
             status_t err;
             if ((err = parseTrackHeader(data_offset, chunk_data_size)) != OK) {
                 return err;
             }
 
-            *offset += chunk_size;
             break;
         }
 
         case FOURCC('p', 's', 's', 'h'):
         {
+            *offset += chunk_size;
+
             PsshInfo pssh;
 
             if (mDataSource->readAt(data_offset + 4, &pssh.uuid, 16) < 16) {
@@ -1086,12 +1092,13 @@
             }
             mPssh.push_back(pssh);
 
-            *offset += chunk_size;
             break;
         }
 
         case FOURCC('m', 'd', 'h', 'd'):
         {
+            *offset += chunk_size;
+
             if (chunk_data_size < 4) {
                 return ERROR_MALFORMED;
             }
@@ -1172,7 +1179,6 @@
             mLastTrack->meta->setCString(
                     kKeyMediaLanguage, lang_code);
 
-            *offset += chunk_size;
             break;
         }
 
@@ -1339,11 +1345,12 @@
                 mLastTrack->sampleTable->setChunkOffsetParams(
                         chunk_type, data_offset, chunk_data_size);
 
+            *offset += chunk_size;
+
             if (err != OK) {
                 return err;
             }
 
-            *offset += chunk_size;
             break;
         }
 
@@ -1353,11 +1360,12 @@
                 mLastTrack->sampleTable->setSampleToChunkParams(
                         data_offset, chunk_data_size);
 
+            *offset += chunk_size;
+
             if (err != OK) {
                 return err;
             }
 
-            *offset += chunk_size;
             break;
         }
 
@@ -1368,6 +1376,8 @@
                 mLastTrack->sampleTable->setSampleSizeParams(
                         chunk_type, data_offset, chunk_data_size);
 
+            *offset += chunk_size;
+
             if (err != OK) {
                 return err;
             }
@@ -1408,7 +1418,6 @@
                 }
                 mLastTrack->meta->setInt32(kKeyMaxInputSize, max_size);
             }
-            *offset += chunk_size;
 
             // NOTE: setting another piece of metadata invalidates any pointers (such as the
             // mimetype) previously obtained, so don't cache them.
@@ -1432,6 +1441,8 @@
 
         case FOURCC('s', 't', 't', 's'):
         {
+            *offset += chunk_size;
+
             status_t err =
                 mLastTrack->sampleTable->setTimeToSampleParams(
                         data_offset, chunk_data_size);
@@ -1440,12 +1451,13 @@
                 return err;
             }
 
-            *offset += chunk_size;
             break;
         }
 
         case FOURCC('c', 't', 't', 's'):
         {
+            *offset += chunk_size;
+
             status_t err =
                 mLastTrack->sampleTable->setCompositionTimeToSampleParams(
                         data_offset, chunk_data_size);
@@ -1454,12 +1466,13 @@
                 return err;
             }
 
-            *offset += chunk_size;
             break;
         }
 
         case FOURCC('s', 't', 's', 's'):
         {
+            *offset += chunk_size;
+
             status_t err =
                 mLastTrack->sampleTable->setSyncSampleParams(
                         data_offset, chunk_data_size);
@@ -1468,13 +1481,14 @@
                 return err;
             }
 
-            *offset += chunk_size;
             break;
         }
 
         // @xyz
         case FOURCC('\xA9', 'x', 'y', 'z'):
         {
+            *offset += chunk_size;
+
             // Best case the total data length inside "@xyz" box
             // would be 8, for instance "@xyz" + "\x00\x04\x15\xc7" + "0+0/",
             // where "\x00\x04" is the text string length with value = 4,
@@ -1503,12 +1517,13 @@
 
             buffer[location_length] = '\0';
             mFileMetaData->setCString(kKeyLocation, buffer);
-            *offset += chunk_size;
             break;
         }
 
         case FOURCC('e', 's', 'd', 's'):
         {
+            *offset += chunk_size;
+
             if (chunk_data_size < 4) {
                 return ERROR_MALFORMED;
             }
@@ -1546,12 +1561,13 @@
                 }
             }
 
-            *offset += chunk_size;
             break;
         }
 
         case FOURCC('a', 'v', 'c', 'C'):
         {
+            *offset += chunk_size;
+
             sp<ABuffer> buffer = new ABuffer(chunk_data_size);
 
             if (mDataSource->readAt(
@@ -1562,12 +1578,12 @@
             mLastTrack->meta->setData(
                     kKeyAVCC, kTypeAVCC, buffer->data(), chunk_data_size);
 
-            *offset += chunk_size;
             break;
         }
 
         case FOURCC('d', '2', '6', '3'):
         {
+            *offset += chunk_size;
             /*
              * d263 contains a fixed 7 bytes part:
              *   vendor - 4 bytes
@@ -1593,7 +1609,6 @@
 
             mLastTrack->meta->setData(kKeyD263, kTypeD263, buffer, chunk_data_size);
 
-            *offset += chunk_size;
             break;
         }
 
@@ -1601,11 +1616,13 @@
         {
             uint8_t buffer[4];
             if (chunk_data_size < (off64_t)sizeof(buffer)) {
+                *offset += chunk_size;
                 return ERROR_MALFORMED;
             }
 
             if (mDataSource->readAt(
                         data_offset, buffer, 4) < 4) {
+                *offset += chunk_size;
                 return ERROR_IO;
             }
 
@@ -1639,6 +1656,8 @@
         case FOURCC('n', 'a', 'm', 'e'):
         case FOURCC('d', 'a', 't', 'a'):
         {
+            *offset += chunk_size;
+
             if (mPath.size() == 6 && underMetaDataPath(mPath)) {
                 status_t err = parseITunesMetaData(data_offset, chunk_data_size);
 
@@ -1647,12 +1666,13 @@
                 }
             }
 
-            *offset += chunk_size;
             break;
         }
 
         case FOURCC('m', 'v', 'h', 'd'):
         {
+            *offset += chunk_size;
+
             if (chunk_data_size < 24) {
                 return ERROR_MALFORMED;
             }
@@ -1680,7 +1700,6 @@
 
             mFileMetaData->setCString(kKeyDate, s.string());
 
-            *offset += chunk_size;
             break;
         }
 
@@ -1701,6 +1720,8 @@
 
         case FOURCC('h', 'd', 'l', 'r'):
         {
+            *offset += chunk_size;
+
             uint32_t buffer;
             if (mDataSource->readAt(
                         data_offset + 8, &buffer, 4) < 4) {
@@ -1715,7 +1736,6 @@
                 mLastTrack->meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_TEXT_3GPP);
             }
 
-            *offset += chunk_size;
             break;
         }
 
@@ -1740,6 +1760,8 @@
                 delete[] buffer;
                 buffer = NULL;
 
+                // advance read pointer so we don't end up reading this again
+                *offset += chunk_size;
                 return ERROR_IO;
             }
 
@@ -1754,6 +1776,8 @@
 
         case FOURCC('c', 'o', 'v', 'r'):
         {
+            *offset += chunk_size;
+
             if (mFileMetaData != NULL) {
                 ALOGV("chunk_data_size = %lld and data_offset = %lld",
                         chunk_data_size, data_offset);
@@ -1768,7 +1792,6 @@
                     buffer->data() + kSkipBytesOfDataBox, chunk_data_size - kSkipBytesOfDataBox);
             }
 
-            *offset += chunk_size;
             break;
         }
 
@@ -1779,25 +1802,27 @@
         case FOURCC('a', 'l', 'b', 'm'):
         case FOURCC('y', 'r', 'r', 'c'):
         {
+            *offset += chunk_size;
+
             status_t err = parse3GPPMetaData(data_offset, chunk_data_size, depth);
 
             if (err != OK) {
                 return err;
             }
 
-            *offset += chunk_size;
             break;
         }
 
         case FOURCC('I', 'D', '3', '2'):
         {
+            *offset += chunk_size;
+
             if (chunk_data_size < 6) {
                 return ERROR_MALFORMED;
             }
 
             parseID3v2MetaData(data_offset + 6);
 
-            *offset += chunk_size;
             break;
         }
 
diff --git a/media/libstagefright/codecs/on2/h264dec/source/h264bsd_intra_prediction.c b/media/libstagefright/codecs/on2/h264dec/source/h264bsd_intra_prediction.c
index 15eabfb..52c85e5 100755
--- a/media/libstagefright/codecs/on2/h264dec/source/h264bsd_intra_prediction.c
+++ b/media/libstagefright/codecs/on2/h264dec/source/h264bsd_intra_prediction.c
@@ -1110,7 +1110,7 @@
 
 /* Variables */
 
-    u32 i, j;
+    i32 i, j;
     i32 a, b, c;
     i32 tmp;
 
@@ -1123,20 +1123,20 @@
     a = 16 * (above[15] + left[15]);
 
     for (i = 0, b = 0; i < 8; i++)
-        b += ((i32)i + 1) * (above[8+i] - above[6-i]);
+        b += (i + 1) * (above[8+i] - above[6-i]);
     b = (5 * b + 32) >> 6;
 
     for (i = 0, c = 0; i < 7; i++)
-        c += ((i32)i + 1) * (left[8+i] - left[6-i]);
+        c += (i + 1) * (left[8+i] - left[6-i]);
     /* p[-1,-1] has to be accessed through above pointer */
-    c += ((i32)i + 1) * (left[8+i] - above[-1]);
+    c += (i + 1) * (left[8+i] - above[-1]);
     c = (5 * c + 32) >> 6;
 
     for (i = 0; i < 16; i++)
     {
         for (j = 0; j < 16; j++)
         {
-            tmp = (a + b * ((i32)j - 7) + c * ((i32)i - 7) + 16) >> 5;
+            tmp = (a + b * (j - 7) + c * (i - 7) + 16) >> 5;
             data[i*16+j] = (u8)CLIP1(tmp);
         }
     }
diff --git a/media/libstagefright/http/MediaHTTP.cpp b/media/libstagefright/http/MediaHTTP.cpp
index 157d967..2d29913 100644
--- a/media/libstagefright/http/MediaHTTP.cpp
+++ b/media/libstagefright/http/MediaHTTP.cpp
@@ -171,6 +171,10 @@
 }
 
 String8 MediaHTTP::getUri() {
+    String8 uri;
+    if (OK == mHTTPConnection->getUri(&uri)) {
+        return uri;
+    }
     return String8(mLastURI.c_str());
 }
 
diff --git a/media/libstagefright/httplive/LiveSession.cpp b/media/libstagefright/httplive/LiveSession.cpp
index bc26de1..95779c4 100644
--- a/media/libstagefright/httplive/LiveSession.cpp
+++ b/media/libstagefright/httplive/LiveSession.cpp
@@ -1007,8 +1007,7 @@
         uint32_t resumeMask = 0;
 
         sp<AnotherPacketSource> sources[kMaxStreams];
-        // TRICKY: looping from i as earlier streams are already removed from streamMask
-        for (size_t j = i; j < kMaxStreams; ++j) {
+        for (size_t j = 0; j < kMaxStreams; ++j) {
             if ((streamMask & indexToType(j)) && uri == mStreams[j].mUri) {
                 sources[j] = mPacketSources.valueFor(indexToType(j));
                 resumeMask |= indexToType(j);
diff --git a/media/libstagefright/httplive/M3UParser.cpp b/media/libstagefright/httplive/M3UParser.cpp
index 4dc5db0..587a6d5 100644
--- a/media/libstagefright/httplive/M3UParser.cpp
+++ b/media/libstagefright/httplive/M3UParser.cpp
@@ -24,6 +24,7 @@
 #include <media/stagefright/foundation/ADebug.h>
 #include <media/stagefright/foundation/AMessage.h>
 #include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/Utils.h>
 #include <media/mediaplayer.h>
 
 namespace android {
@@ -352,9 +353,27 @@
     if (!meta->findString(key, &groupID)) {
         *uri = mItems.itemAt(index).mURI;
 
-        // Assume media without any more specific attribute contains
-        // audio and video, but no subtitles.
-        return !strcmp("audio", key) || !strcmp("video", key);
+        AString codecs;
+        if (!meta->findString("codecs", &codecs)) {
+            // Assume media without any more specific attribute contains
+            // audio and video, but no subtitles.
+            return !strcmp("audio", key) || !strcmp("video", key);
+        } else {
+            // Split the comma separated list of codecs.
+            size_t offset = 0;
+            ssize_t commaPos = -1;
+            codecs.append(',');
+            while ((commaPos = codecs.find(",", offset)) >= 0) {
+                AString codec(codecs, offset, commaPos - offset);
+                // return true only if a codec of type `key` ("audio"/"video")
+                // is found.
+                if (codecIsType(codec, key)) {
+                    return true;
+                }
+                offset = commaPos + 1;
+            }
+            return false;
+        }
     }
 
     sp<MediaGroup> group = mMediaGroups.valueFor(groupID);
@@ -682,12 +701,22 @@
                 *meta = new AMessage;
             }
             (*meta)->setInt32("bandwidth", x);
+        } else if (!strcasecmp("codecs", key.c_str())) {
+            if (!isQuotedString(val)) {
+                ALOGE("Expected quoted string for %s attribute, "
+                      "got '%s' instead.",
+                      key.c_str(), val.c_str());;
+
+                return ERROR_MALFORMED;
+            }
+
+            key.tolower();
+            const AString &codecs = unquoteString(val);
+            (*meta)->setString(key.c_str(), codecs.c_str());
         } else if (!strcasecmp("audio", key.c_str())
                 || !strcasecmp("video", key.c_str())
                 || !strcasecmp("subtitles", key.c_str())) {
-            if (val.size() < 2
-                    || val.c_str()[0] != '"'
-                    || val.c_str()[val.size() - 1] != '"') {
+            if (!isQuotedString(val)) {
                 ALOGE("Expected quoted string for %s attribute, "
                       "got '%s' instead.",
                       key.c_str(), val.c_str());
@@ -695,7 +724,7 @@
                 return ERROR_MALFORMED;
             }
 
-            AString groupID(val, 1, val.size() - 2);
+            const AString &groupID = unquoteString(val);
             ssize_t groupIndex = mMediaGroups.indexOfKey(groupID);
 
             if (groupIndex < 0) {
@@ -1084,4 +1113,121 @@
     return OK;
 }
 
+// static
+bool M3UParser::isQuotedString(const AString &str) {
+    if (str.size() < 2
+            || str.c_str()[0] != '"'
+            || str.c_str()[str.size() - 1] != '"') {
+        return false;
+    }
+    return true;
+}
+
+// static
+AString M3UParser::unquoteString(const AString &str) {
+     if (!isQuotedString(str)) {
+         return str;
+     }
+     return AString(str, 1, str.size() - 2);
+}
+
+// static
+bool M3UParser::codecIsType(const AString &codec, const char *type) {
+    if (codec.size() < 4) {
+        return false;
+    }
+    const char *c = codec.c_str();
+    switch (FOURCC(c[0], c[1], c[2], c[3])) {
+        // List extracted from http://www.mp4ra.org/codecs.html
+        case 'ac-3':
+        case 'alac':
+        case 'dra1':
+        case 'dtsc':
+        case 'dtse':
+        case 'dtsh':
+        case 'dtsl':
+        case 'ec-3':
+        case 'enca':
+        case 'g719':
+        case 'g726':
+        case 'm4ae':
+        case 'mlpa':
+        case 'mp4a':
+        case 'raw ':
+        case 'samr':
+        case 'sawb':
+        case 'sawp':
+        case 'sevc':
+        case 'sqcp':
+        case 'ssmv':
+        case 'twos':
+        case 'agsm':
+        case 'alaw':
+        case 'dvi ':
+        case 'fl32':
+        case 'fl64':
+        case 'ima4':
+        case 'in24':
+        case 'in32':
+        case 'lpcm':
+        case 'Qclp':
+        case 'QDM2':
+        case 'QDMC':
+        case 'ulaw':
+        case 'vdva':
+            return !strcmp("audio", type);
+
+        case 'avc1':
+        case 'avc2':
+        case 'avcp':
+        case 'drac':
+        case 'encv':
+        case 'mjp2':
+        case 'mp4v':
+        case 'mvc1':
+        case 'mvc2':
+        case 'resv':
+        case 's263':
+        case 'svc1':
+        case 'vc-1':
+        case 'CFHD':
+        case 'civd':
+        case 'DV10':
+        case 'dvh5':
+        case 'dvh6':
+        case 'dvhp':
+        case 'DVOO':
+        case 'DVOR':
+        case 'DVTV':
+        case 'DVVT':
+        case 'flic':
+        case 'gif ':
+        case 'h261':
+        case 'h263':
+        case 'HD10':
+        case 'jpeg':
+        case 'M105':
+        case 'mjpa':
+        case 'mjpb':
+        case 'png ':
+        case 'PNTG':
+        case 'rle ':
+        case 'rpza':
+        case 'Shr0':
+        case 'Shr1':
+        case 'Shr2':
+        case 'Shr3':
+        case 'Shr4':
+        case 'SVQ1':
+        case 'SVQ3':
+        case 'tga ':
+        case 'tiff':
+        case 'WRLE':
+            return !strcmp("video", type);
+
+        default:
+            return false;
+    }
+}
+
 }  // namespace android
diff --git a/media/libstagefright/httplive/M3UParser.h b/media/libstagefright/httplive/M3UParser.h
index 2051e41..ccd6556 100644
--- a/media/libstagefright/httplive/M3UParser.h
+++ b/media/libstagefright/httplive/M3UParser.h
@@ -96,6 +96,10 @@
     static status_t ParseInt32(const char *s, int32_t *x);
     static status_t ParseDouble(const char *s, double *x);
 
+    static bool isQuotedString(const AString &str);
+    static AString unquoteString(const AString &str);
+    static bool codecIsType(const AString &codec, const char *type);
+
     DISALLOW_EVIL_CONSTRUCTORS(M3UParser);
 };
 
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index 67d83b1..f00b82a 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -193,6 +193,7 @@
         t->mainBuffer = NULL;
         t->auxBuffer = NULL;
         t->downmixerBufferProvider = NULL;
+        t->mSinkFormat = AUDIO_FORMAT_PCM_16_BIT;
 
         status_t status = initTrackDownmix(&mState.tracks[n], n, channelMask);
         if (status == OK) {
@@ -440,6 +441,13 @@
         //         for a specific track? or per mixer?
         /* case DOWNMIX_TYPE:
             break          */
+        case SINK_FORMAT: {
+            audio_format_t format = static_cast<audio_format_t>(valueInt);
+            if (track.mSinkFormat != format) {
+                track.mSinkFormat = format;
+                ALOGV("setParameter(TRACK, SINK_FORMAT, %#x)", format);
+            }
+            } break;
         default:
             LOG_FATAL("bad param");
         }
@@ -1043,7 +1051,7 @@
 void AudioMixer::process__nop(state_t* state, int64_t pts)
 {
     uint32_t e0 = state->enabledTracks;
-    size_t bufSize = state->frameCount * sizeof(int16_t) * MAX_NUM_CHANNELS;
+    size_t sampleCount = state->frameCount * MAX_NUM_CHANNELS;
     while (e0) {
         // process by group of tracks with same output buffer to
         // avoid multiple memset() on same buffer
@@ -1062,7 +1070,8 @@
             }
             e0 &= ~(e1);
 
-            memset(t1.mainBuffer, 0, bufSize);
+            memset(t1.mainBuffer, 0, sampleCount
+                    * audio_bytes_per_sample(t1.mSinkFormat));
         }
 
         while (e1) {
@@ -1170,8 +1179,18 @@
                     }
                 }
             }
-            ditherAndClamp(out, outTemp, BLOCKSIZE);
-            out += BLOCKSIZE;
+            switch (t1.mSinkFormat) {
+            case AUDIO_FORMAT_PCM_FLOAT:
+                memcpy_to_float_from_q19_12(reinterpret_cast<float *>(out), outTemp, BLOCKSIZE * 2);
+                out += BLOCKSIZE * 2; // output is 2 floats/frame.
+                break;
+            case AUDIO_FORMAT_PCM_16_BIT:
+                ditherAndClamp(out, outTemp, BLOCKSIZE);
+                out += BLOCKSIZE; // output is 1 int32_t (2 int16_t samples)/frame
+                break;
+            default:
+                LOG_ALWAYS_FATAL("bad sink format: %d", t1.mSinkFormat);
+            }
             numFrames += BLOCKSIZE;
         } while (numFrames < state->frameCount);
     }
@@ -1253,7 +1272,16 @@
                 }
             }
         }
-        ditherAndClamp(out, outTemp, numFrames);
+        switch (t1.mSinkFormat) {
+        case AUDIO_FORMAT_PCM_FLOAT:
+            memcpy_to_float_from_q19_12(reinterpret_cast<float*>(out), outTemp, numFrames*2);
+            break;
+        case AUDIO_FORMAT_PCM_16_BIT:
+            ditherAndClamp(out, outTemp, numFrames);
+            break;
+        default:
+            LOG_ALWAYS_FATAL("bad sink format: %d", t1.mSinkFormat);
+        }
     }
 }
 
@@ -1294,27 +1322,45 @@
         }
         size_t outFrames = b.frameCount;
 
-        if (CC_UNLIKELY(uint32_t(vl) > UNITY_GAIN || uint32_t(vr) > UNITY_GAIN)) {
-            // volume is boosted, so we might need to clamp even though
-            // we process only one track.
+        switch (t.mSinkFormat) {
+        case AUDIO_FORMAT_PCM_FLOAT: {
+            float *fout = reinterpret_cast<float*>(out);
+            static float scale = 1. / (32768. * 4096.); // exact when inverted
             do {
                 uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
                 in += 2;
-                int32_t l = mulRL(1, rl, vrl) >> 12;
-                int32_t r = mulRL(0, rl, vrl) >> 12;
-                // clamping...
-                l = clamp16(l);
-                r = clamp16(r);
-                *out++ = (r<<16) | (l & 0xFFFF);
+                int32_t l = mulRL(1, rl, vrl);
+                int32_t r = mulRL(0, rl, vrl);
+                *fout++ = static_cast<float>(l) * scale;
+                *fout++ = static_cast<float>(r) * scale;
             } while (--outFrames);
-        } else {
-            do {
-                uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
-                in += 2;
-                int32_t l = mulRL(1, rl, vrl) >> 12;
-                int32_t r = mulRL(0, rl, vrl) >> 12;
-                *out++ = (r<<16) | (l & 0xFFFF);
-            } while (--outFrames);
+            } break;
+        case AUDIO_FORMAT_PCM_16_BIT:
+            if (CC_UNLIKELY(uint32_t(vl) > UNITY_GAIN || uint32_t(vr) > UNITY_GAIN)) {
+                // volume is boosted, so we might need to clamp even though
+                // we process only one track.
+                do {
+                    uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
+                    in += 2;
+                    int32_t l = mulRL(1, rl, vrl) >> 12;
+                    int32_t r = mulRL(0, rl, vrl) >> 12;
+                    // clamping...
+                    l = clamp16(l);
+                    r = clamp16(r);
+                    *out++ = (r<<16) | (l & 0xFFFF);
+                } while (--outFrames);
+            } else {
+                do {
+                    uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
+                    in += 2;
+                    int32_t l = mulRL(1, rl, vrl) >> 12;
+                    int32_t r = mulRL(0, rl, vrl) >> 12;
+                    *out++ = (r<<16) | (l & 0xFFFF);
+                } while (--outFrames);
+            }
+            break;
+        default:
+            LOG_ALWAYS_FATAL("bad sink format: %d", t.mSinkFormat);
         }
         numFrames -= b.frameCount;
         t.bufferProvider->releaseBuffer(&b);
diff --git a/services/audioflinger/AudioMixer.h b/services/audioflinger/AudioMixer.h
index d286986..3355db4 100644
--- a/services/audioflinger/AudioMixer.h
+++ b/services/audioflinger/AudioMixer.h
@@ -77,6 +77,7 @@
         MAIN_BUFFER     = 0x4002,
         AUX_BUFFER      = 0x4003,
         DOWNMIX_TYPE    = 0X4004,
+        SINK_FORMAT     = 0x4005, // AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
         // for target RESAMPLE
         SAMPLE_RATE     = 0x4100, // Configure sample rate conversion on this track name;
                                   // parameter 'value' is the new sample rate in Hz.
@@ -193,7 +194,9 @@
 
         int32_t     sessionId;
 
-        int32_t     padding[2];
+        audio_format_t mSinkFormat; // at this time: AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
+
+        int32_t     padding[1];
 
         // 16-byte boundary
 
diff --git a/services/audioflinger/AudioResamplerDyn.cpp b/services/audioflinger/AudioResamplerDyn.cpp
index 939b128..7e4ca0c 100644
--- a/services/audioflinger/AudioResamplerDyn.cpp
+++ b/services/audioflinger/AudioResamplerDyn.cpp
@@ -165,6 +165,10 @@
     mCoefBuffer(NULL)
 {
     mVolumeSimd[0] = mVolumeSimd[1] = 0;
+    // The AudioResampler base class assumes we are always ready for 1:1 resampling.
+    // We reset mInSampleRate to 0, so setSampleRate() will calculate filters for
+    // setSampleRate() for 1:1. (May be removed if precalculated filters are used.)
+    mInSampleRate = 0;
     mConstants.set(128, 8, mSampleRate, mSampleRate); // TODO: set better
 }
 
diff --git a/services/audioflinger/test-resample.cpp b/services/audioflinger/test-resample.cpp
index 74d5702..3ab3ba9 100644
--- a/services/audioflinger/test-resample.cpp
+++ b/services/audioflinger/test-resample.cpp
@@ -27,6 +27,7 @@
 #include <time.h>
 #include <math.h>
 #include <audio_utils/sndfile.h>
+#include <utils/Vector.h>
 
 using namespace android;
 
@@ -34,7 +35,7 @@
 
 static int usage(const char* name) {
     fprintf(stderr,"Usage: %s [-p] [-h] [-v] [-s] [-q {dq|lq|mq|hq|vhq|dlq|dmq|dhq}]"
-                   " [-i input-sample-rate] [-o output-sample-rate] [-O #] [<input-file>]"
+                   " [-i input-sample-rate] [-o output-sample-rate] [-O csv] [-P csv] [<input-file>]"
                    " <output-file>\n", name);
     fprintf(stderr,"    -p    enable profiling\n");
     fprintf(stderr,"    -h    create wav file\n");
@@ -51,10 +52,50 @@
     fprintf(stderr,"              dhq : dynamic high quality\n");
     fprintf(stderr,"    -i    input file sample rate (ignored if input file is specified)\n");
     fprintf(stderr,"    -o    output file sample rate\n");
-    fprintf(stderr,"    -O    # frames output per call to resample()\n");
+    fprintf(stderr,"    -O    # frames output per call to resample() in CSV format\n");
+    fprintf(stderr,"    -P    # frames provided per call to resample() in CSV format\n");
     return -1;
 }
 
+// Convert a list of integers in CSV format to a Vector of those values.
+// Returns the number of elements in the list, or -1 on error.
+int parseCSV(const char *string, Vector<int>& values)
+{
+    // pass 1: count the number of values and do syntax check
+    size_t numValues = 0;
+    bool hadDigit = false;
+    for (const char *p = string; ; ) {
+        switch (*p++) {
+        case '0': case '1': case '2': case '3': case '4':
+        case '5': case '6': case '7': case '8': case '9':
+            hadDigit = true;
+            break;
+        case '\0':
+            if (hadDigit) {
+                // pass 2: allocate and initialize vector of values
+                values.resize(++numValues);
+                values.editItemAt(0) = atoi(p = optarg);
+                for (size_t i = 1; i < numValues; ) {
+                    if (*p++ == ',') {
+                        values.editItemAt(i++) = atoi(p);
+                    }
+                }
+                return numValues;
+            }
+            // fall through
+        case ',':
+            if (hadDigit) {
+                hadDigit = false;
+                numValues++;
+                break;
+            }
+            // fall through
+        default:
+            return -1;
+        }
+    }
+}
+
 int main(int argc, char* argv[]) {
 
     const char* const progname = argv[0];
@@ -65,10 +106,11 @@
     int input_freq = 0;
     int output_freq = 0;
     AudioResampler::src_quality quality = AudioResampler::DEFAULT_QUALITY;
-    size_t framesPerCall = 0;
+    Vector<int> Ovalues;
+    Vector<int> Pvalues;
 
     int ch;
-    while ((ch = getopt(argc, argv, "pfhvsq:i:o:O:")) != -1) {
+    while ((ch = getopt(argc, argv, "pfhvsq:i:o:O:P:")) != -1) {
         switch (ch) {
         case 'p':
             profileResample = true;
@@ -114,7 +156,16 @@
             output_freq = atoi(optarg);
             break;
         case 'O':
-            framesPerCall = atoi(optarg);
+            if (parseCSV(optarg, Ovalues) < 0) {
+                fprintf(stderr, "incorrect syntax for -O option\n");
+                return -1;
+            }
+            break;
+        case 'P':
+            if (parseCSV(optarg, Pvalues) < 0) {
+                fprintf(stderr, "incorrect syntax for -P option\n");
+                return -1;
+            }
             break;
         case '?':
         default:
@@ -182,12 +233,14 @@
         const int       mChannels;
         size_t          mNextFrame; // index of next frame to provide
         size_t          mUnrel;     // number of frames not yet released
+        const Vector<int> mPvalues; // number of frames provided per call
+        size_t          mNextPidx;  // index of next entry in mPvalues to use
     public:
-        Provider(const void* addr, size_t size, int channels)
+        Provider(const void* addr, size_t size, int channels, const Vector<int>& Pvalues)
           : mAddr((int16_t*) addr),
             mNumFrames(size / (channels*sizeof(int16_t))),
             mChannels(channels),
-            mNextFrame(0), mUnrel(0) {
+            mNextFrame(0), mUnrel(0), mPvalues(Pvalues), mNextPidx(0) {
         }
         virtual status_t getNextBuffer(Buffer* buffer,
                 int64_t pts = kInvalidPTS) {
@@ -196,6 +249,16 @@
             if (requestedFrames > mNumFrames - mNextFrame) {
                 buffer->frameCount = mNumFrames - mNextFrame;
             }
+            if (!mPvalues.isEmpty()) {
+                size_t provided = mPvalues[mNextPidx++];
+                printf("mPvalue[%d]=%u not %u\n", mNextPidx-1, provided, buffer->frameCount);
+                if (provided < buffer->frameCount) {
+                    buffer->frameCount = provided;
+                }
+                if (mNextPidx >= mPvalues.size()) {
+                    mNextPidx = 0;
+                }
+            }
             if (gVerbose) {
                 printf("getNextBuffer() requested %u frames out of %u frames available,"
                         " and returned %u frames\n",
@@ -230,7 +293,7 @@
         void reset() {
             mNextFrame = 0;
         }
-    } provider(input_vaddr, input_size, channels);
+    } provider(input_vaddr, input_size, channels, Pvalues);
 
     size_t input_frames = input_size / (channels * sizeof(int16_t));
     if (gVerbose) {
@@ -348,11 +411,17 @@
     if (gVerbose) {
         printf("resample() %u output frames\n", out_frames);
     }
-    if (framesPerCall == 0 || framesPerCall > out_frames) {
-        framesPerCall = out_frames;
+    if (Ovalues.isEmpty()) {
+        Ovalues.push(out_frames);
     }
-    for (size_t i = 0; i < out_frames; ) {
-        size_t thisFrames = framesPerCall <= out_frames - i ? framesPerCall : out_frames - i;
+    for (size_t i = 0, j = 0; i < out_frames; ) {
+        size_t thisFrames = Ovalues[j++];
+        if (j >= Ovalues.size()) {
+            j = 0;
+        }
+        if (thisFrames == 0 || thisFrames > out_frames - i) {
+            thisFrames = out_frames - i;
+        }
         resampler->resample((int*) output_vaddr + 2*i, thisFrames, &provider);
         i += thisFrames;
     }
diff --git a/services/camera/libcameraservice/api1/Camera2Client.cpp b/services/camera/libcameraservice/api1/Camera2Client.cpp
index 0a88a75..80b7cd4 100644
--- a/services/camera/libcameraservice/api1/Camera2Client.cpp
+++ b/services/camera/libcameraservice/api1/Camera2Client.cpp
@@ -407,12 +407,6 @@
         l.mParameters.state = Parameters::DISCONNECTED;
     }
 
-    mStreamingProcessor->deletePreviewStream();
-    mStreamingProcessor->deleteRecordingStream();
-    mJpegProcessor->deleteStream();
-    mCallbackProcessor->deleteStream();
-    mZslProcessor->deleteStream();
-
     mStreamingProcessor->requestExit();
     mFrameProcessor->requestExit();
     mCaptureSequencer->requestExit();
@@ -429,6 +423,14 @@
     mZslProcessorThread->join();
     mCallbackProcessor->join();
 
+    ALOGV("Camera %d: Deleting streams", mCameraId);
+
+    mStreamingProcessor->deletePreviewStream();
+    mStreamingProcessor->deleteRecordingStream();
+    mJpegProcessor->deleteStream();
+    mCallbackProcessor->deleteStream();
+    mZslProcessor->deleteStream();
+
     ALOGV("Camera %d: Disconnecting device", mCameraId);
 
     mDevice->disconnect();