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/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/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/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..e8f75a4 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -799,16 +799,26 @@
                 bufSize = max(bufSize, (int32_t)sizeof(VideoNativeMetadata));
             }
 
+            size_t alignment = MemoryDealer::getAllocationAlignment();
+
             ALOGV("[%s] Allocating %u buffers of size %d/%d (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) {
+            if (bufSize == 0 || bufSize > 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);
+            if (def.nBufferCountActual > SIZE_MAX / alignedSize) {
+                ALOGE("b/22885421");
+                return NO_MEMORY;
+            }
+
+            size_t totalSize = def.nBufferCountActual * alignedSize;
             mDealer[portIndex] = new MemoryDealer(totalSize, "ACodec");
 
             for (OMX_U32 i = 0; i < def.nBufferCountActual && err == OK; ++i) {
@@ -1116,7 +1126,7 @@
 
     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
@@ -4658,6 +4668,7 @@
                     notify->setString("mime", MEDIA_MIMETYPE_AUDIO_RAW);
                     notify->setInt32("channel-count", params.nChannels);
                     notify->setInt32("sample-rate", params.nSamplingRate);
+                    notify->setInt32("pcm-encoding", kAudioEncodingPcm16bit);
 
                     if (mChannelMaskPresent) {
                         notify->setInt32("channel-mask", mChannelMask);
@@ -4840,6 +4851,7 @@
                     notify->setString("mime", mime);
                     notify->setInt32("channel-count", params.nChannels);
                     notify->setInt32("sample-rate", params.nSamplingRate);
+                    notify->setInt32("pcm-encoding", kAudioEncodingPcm16bit);
                     break;
                 }
 
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/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/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..a034368 100644
--- a/media/libstagefright/Utils.cpp
+++ b/media/libstagefright/Utils.cpp
@@ -239,6 +239,11 @@
         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/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/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/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/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/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index d07ca85..f0c5a21 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);
@@ -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,
@@ -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 {
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..93768ca 100644
--- a/services/audioflinger/Effects.cpp
+++ b/services/audioflinger/Effects.cpp
@@ -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;
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..2332f3a 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;
     }
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..658b820 100644
--- a/services/audioflinger/PatchPanel.cpp
+++ b/services/audioflinger/PatchPanel.cpp
@@ -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..040cb93 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);
 
@@ -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;
@@ -3489,7 +3487,12 @@
     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 +3527,9 @@
 
         // 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 +3541,12 @@
         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 +4351,6 @@
         }
 
         }   // local variable scope to avoid goto warning
-track_is_ready: ;
 
     }
 
@@ -4578,10 +4587,6 @@
 
 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 +4667,6 @@
 
 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 +4755,9 @@
         }
 
         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
@@ -4996,6 +5002,10 @@
     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) {
@@ -5276,7 +5286,9 @@
             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 +5740,12 @@
     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 +5823,7 @@
         // 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 +6095,10 @@
             }
         // 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 {
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);