diff --git a/drm/mediadrm/plugins/mock/MockDrmCryptoPlugin.cpp b/drm/mediadrm/plugins/mock/MockDrmCryptoPlugin.cpp
index 9b786c5..851ad2c 100644
--- a/drm/mediadrm/plugins/mock/MockDrmCryptoPlugin.cpp
+++ b/drm/mediadrm/plugins/mock/MockDrmCryptoPlugin.cpp
@@ -56,7 +56,7 @@
         return true;
     }
 
-    status_t MockDrmFactory::createDrmPlugin(const uint8_t uuid[16], DrmPlugin **plugin)
+    status_t MockDrmFactory::createDrmPlugin(const uint8_t /* uuid */[16], DrmPlugin **plugin)
     {
         *plugin = new MockDrmPlugin();
         return OK;
@@ -68,8 +68,9 @@
         return (!memcmp(uuid, mock_uuid, sizeof(mock_uuid)));
     }
 
-    status_t MockCryptoFactory::createPlugin(const uint8_t uuid[16], const void *data,
-                                             size_t size, CryptoPlugin **plugin)
+    status_t MockCryptoFactory::createPlugin(const uint8_t /* uuid */[16],
+                                             const void * /* data */,
+                                             size_t /* size */, CryptoPlugin **plugin)
     {
         *plugin = new MockCryptoPlugin();
         return OK;
@@ -150,7 +151,7 @@
         // Properties used in mock test, set by cts test app returned from mock plugin
         //   byte[] mock-request       -> request
         //   string mock-default-url   -> defaultUrl
-        //   string mock-key-request-type -> keyRequestType
+        //   string mock-keyRequestType -> keyRequestType
 
         index = mByteArrayProperties.indexOfKey(String8("mock-request"));
         if (index < 0) {
@@ -266,8 +267,8 @@
         return OK;
     }
 
-    status_t MockDrmPlugin::getProvisionRequest(String8 const &certType,
-                                                String8 const &certAuthority,
+    status_t MockDrmPlugin::getProvisionRequest(String8 const & /* certType */,
+                                                String8 const & /* certAuthority */,
                                                 Vector<uint8_t> &request,
                                                 String8 &defaultUrl)
     {
@@ -297,8 +298,8 @@
     }
 
     status_t MockDrmPlugin::provideProvisionResponse(Vector<uint8_t> const &response,
-                                                     Vector<uint8_t> &certificate,
-                                                     Vector<uint8_t> &wrappedKey)
+                                                     Vector<uint8_t> & /* certificate */,
+                                                     Vector<uint8_t> & /* wrappedKey */)
     {
         Mutex::Autolock lock(mLock);
         ALOGD("MockDrmPlugin::provideProvisionResponse(%s)",
@@ -317,7 +318,8 @@
         return OK;
     }
 
-    status_t MockDrmPlugin::getSecureStop(Vector<uint8_t> const &ssid, Vector<uint8_t> &secureStop)
+    status_t MockDrmPlugin::getSecureStop(Vector<uint8_t> const & /* ssid */,
+                                          Vector<uint8_t> & secureStop)
     {
         Mutex::Autolock lock(mLock);
         ALOGD("MockDrmPlugin::getSecureStop()");
@@ -439,6 +441,63 @@
                   pData ? vectorToString(*pData) : "{}");
 
             sendEvent(eventType, extra, pSessionId, pData);
+        } else if (name == "mock-send-expiration-update") {
+            int64_t expiryTimeMS;
+            sscanf(value.string(), "%jd", &expiryTimeMS);
+
+            Vector<uint8_t> const *pSessionId = NULL;
+            ssize_t index = mByteArrayProperties.indexOfKey(String8("mock-event-session-id"));
+            if (index >= 0) {
+                pSessionId = &mByteArrayProperties[index];
+            }
+
+            ALOGD("sending expiration-update from mock drm plugin: %jd %s",
+                  expiryTimeMS, pSessionId ? vectorToString(*pSessionId) : "{}");
+
+            sendExpirationUpdate(pSessionId, expiryTimeMS);
+        } else if (name == "mock-send-keys-change") {
+            Vector<uint8_t> const *pSessionId = NULL;
+            ssize_t index = mByteArrayProperties.indexOfKey(String8("mock-event-session-id"));
+            if (index >= 0) {
+                pSessionId = &mByteArrayProperties[index];
+            }
+
+            ALOGD("sending keys-change from mock drm plugin: %s",
+                  pSessionId ? vectorToString(*pSessionId) : "{}");
+
+            Vector<DrmPlugin::KeyStatus> keyStatusList;
+            DrmPlugin::KeyStatus keyStatus;
+            uint8_t keyId1[] = {'k', 'e', 'y', '1'};
+            keyStatus.mKeyId.clear();
+            keyStatus.mKeyId.appendArray(keyId1, sizeof(keyId1));
+            keyStatus.mType = DrmPlugin::kKeyStatusType_Usable;
+            keyStatusList.add(keyStatus);
+
+            uint8_t keyId2[] = {'k', 'e', 'y', '2'};
+            keyStatus.mKeyId.clear();
+            keyStatus.mKeyId.appendArray(keyId2, sizeof(keyId2));
+            keyStatus.mType = DrmPlugin::kKeyStatusType_Expired;
+            keyStatusList.add(keyStatus);
+
+            uint8_t keyId3[] = {'k', 'e', 'y', '3'};
+            keyStatus.mKeyId.clear();
+            keyStatus.mKeyId.appendArray(keyId3, sizeof(keyId3));
+            keyStatus.mType = DrmPlugin::kKeyStatusType_OutputNotAllowed;
+            keyStatusList.add(keyStatus);
+
+            uint8_t keyId4[] = {'k', 'e', 'y', '4'};
+            keyStatus.mKeyId.clear();
+            keyStatus.mKeyId.appendArray(keyId4, sizeof(keyId4));
+            keyStatus.mType = DrmPlugin::kKeyStatusType_StatusPending;
+            keyStatusList.add(keyStatus);
+
+            uint8_t keyId5[] = {'k', 'e', 'y', '5'};
+            keyStatus.mKeyId.clear();
+            keyStatus.mKeyId.appendArray(keyId5, sizeof(keyId5));
+            keyStatus.mType = DrmPlugin::kKeyStatusType_InternalError;
+            keyStatusList.add(keyStatus);
+
+            sendKeysChange(pSessionId, &keyStatusList, true);
         } else {
             mStringProperties.add(name, value);
         }
@@ -740,7 +799,7 @@
     ssize_t
     MockCryptoPlugin::decrypt(bool secure, const uint8_t key[16], const uint8_t iv[16],
                               Mode mode, const void *srcPtr, const SubSample *subSamples,
-                              size_t numSubSamples, void *dstPtr, AString *errorDetailMsg)
+                              size_t numSubSamples, void *dstPtr, AString * /* errorDetailMsg */)
     {
         ALOGD("MockCryptoPlugin::decrypt(secure=%d, key=%s, iv=%s, mode=%d, src=%p, "
               "subSamples=%s, dst=%p)",
@@ -769,7 +828,7 @@
     {
         String8 result;
         for (size_t i = 0; i < numSubSamples; i++) {
-            result.appendFormat("[%zu] {clear:%zu, encrypted:%zu} ", i,
+            result.appendFormat("[%zu] {clear:%u, encrypted:%u} ", i,
                                 subSamples[i].mNumBytesOfClearData,
                                 subSamples[i].mNumBytesOfEncryptedData);
         }
diff --git a/include/media/AudioResamplerPublic.h b/include/media/AudioResamplerPublic.h
index b705efa..07d946d 100644
--- a/include/media/AudioResamplerPublic.h
+++ b/include/media/AudioResamplerPublic.h
@@ -17,6 +17,8 @@
 #ifndef ANDROID_AUDIO_RESAMPLER_PUBLIC_H
 #define ANDROID_AUDIO_RESAMPLER_PUBLIC_H
 
+#include <stdint.h>
+
 // AUDIO_RESAMPLER_DOWN_RATIO_MAX is the maximum ratio between the original
 // audio sample rate and the target rate when downsampling,
 // as permitted in the audio framework, e.g. AudioTrack and AudioFlinger.
@@ -26,6 +28,22 @@
 // TODO: replace with an API
 #define AUDIO_RESAMPLER_DOWN_RATIO_MAX 256
 
+// AUDIO_RESAMPLER_UP_RATIO_MAX is the maximum suggested ratio between the original
+// audio sample rate and the target rate when upsampling.  It is loosely enforced by
+// the system. One issue with large upsampling ratios is the approximation by
+// an int32_t of the phase increments, making the resulting sample rate inexact.
+#define AUDIO_RESAMPLER_UP_RATIO_MAX 65536
+
+#define AUDIO_TIMESTRETCH_SPEED_MIN    0.5f
+#define AUDIO_TIMESTRETCH_SPEED_MAX    2.0f
+#define AUDIO_TIMESTRETCH_SPEED_NORMAL 1.0f
+
+#define AUDIO_TIMESTRETCH_PITCH_MIN    0.5f
+#define AUDIO_TIMESTRETCH_PITCH_MAX    2.0f
+#define AUDIO_TIMESTRETCH_PITCH_NORMAL 1.0f
+
+// TODO: Consider putting these inlines into a class scope
+
 // Returns the source frames needed to resample to destination frames.  This is not a precise
 // value and depends on the resampler (and possibly how it handles rounding internally).
 // Nevertheless, this should be an upper bound on the requirements of the resampler.
@@ -39,4 +57,24 @@
             size_t((uint64_t)dstFramesRequired * srcSampleRate / dstSampleRate + 1 + 1);
 }
 
+// An upper bound for the number of destination frames possible from srcFrames
+// after sample rate conversion.  This may be used for buffer sizing.
+static inline size_t destinationFramesPossible(size_t srcFrames, uint32_t srcSampleRate,
+        uint32_t dstSampleRate) {
+    if (srcSampleRate == dstSampleRate) {
+        return srcFrames;
+    }
+    uint64_t dstFrames = (uint64_t)srcFrames * dstSampleRate / srcSampleRate;
+    return dstFrames > 2 ? dstFrames - 2 : 0;
+}
+
+static inline size_t sourceFramesNeededWithTimestretch(
+        uint32_t srcSampleRate, size_t dstFramesRequired, uint32_t dstSampleRate,
+        float speed) {
+    // required is the number of input frames the resampler needs
+    size_t required = sourceFramesNeeded(srcSampleRate, dstFramesRequired, dstSampleRate);
+    // to deliver this, the time stretcher requires:
+    return required * (double)speed + 1 + 1; // accounting for rounding dependencies
+}
+
 #endif // ANDROID_AUDIO_RESAMPLER_PUBLIC_H
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index e7e0703..a06197f 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -359,6 +359,21 @@
     /* Return current source sample rate in Hz */
             uint32_t    getSampleRate() const;
 
+    /* Set source playback rate for timestretch
+     * 1.0 is normal speed: < 1.0 is slower, > 1.0 is faster
+     * 1.0 is normal pitch: < 1.0 is lower pitch, > 1.0 is higher pitch
+     *
+     * AUDIO_TIMESTRETCH_SPEED_MIN <= speed <= AUDIO_TIMESTRETCH_SPEED_MAX
+     * AUDIO_TIMESTRETCH_PITCH_MIN <= pitch <= AUDIO_TIMESTRETCH_PITCH_MAX
+     *
+     * Speed increases the playback rate of media, but does not alter pitch.
+     * Pitch increases the "tonal frequency" of media, but does not affect the playback rate.
+     */
+            status_t    setPlaybackRate(float speed, float pitch);
+
+    /* Return current playback rate */
+            void        getPlaybackRate(float *speed, float *pitch) const;
+
     /* Enables looping and sets the start and end points of looping.
      * Only supported for static buffer mode.
      *
@@ -719,6 +734,9 @@
             // increment mPosition by the delta of mServer, and return new value of mPosition
             uint32_t updateAndGetPosition_l();
 
+            // check sample rate and speed is compatible with AudioTrack
+            bool     isSampleRateSpeedAllowed_l(uint32_t sampleRate, float speed) const;
+
     // Next 4 fields may be changed if IAudioTrack is re-created, but always != 0
     sp<IAudioTrack>         mAudioTrack;
     sp<IMemory>             mCblkMemory;
@@ -730,6 +748,8 @@
     float                   mVolume[2];
     float                   mSendLevel;
     mutable uint32_t        mSampleRate;            // mutable because getSampleRate() can update it
+    float                   mSpeed;                 // timestretch: 1.0f for normal speed.
+    float                   mPitch;                 // timestretch: 1.0f for normal pitch.
     size_t                  mFrameCount;            // corresponds to current IAudioTrack, value is
                                                     // reported back by AudioFlinger to the client
     size_t                  mReqFrameCount;         // frame count to request the first or next time
diff --git a/include/media/ICrypto.h b/include/media/ICrypto.h
index 07742ca..aa04dbe 100644
--- a/include/media/ICrypto.h
+++ b/include/media/ICrypto.h
@@ -25,6 +25,7 @@
 namespace android {
 
 struct AString;
+struct IMemory;
 
 struct ICrypto : public IInterface {
     DECLARE_META_INTERFACE(Crypto);
@@ -43,12 +44,14 @@
 
     virtual void notifyResolution(uint32_t width, uint32_t height) = 0;
 
+    virtual status_t setMediaDrmSession(const Vector<uint8_t> &sessionId) = 0;
+
     virtual ssize_t decrypt(
             bool secure,
             const uint8_t key[16],
             const uint8_t iv[16],
             CryptoPlugin::Mode mode,
-            const void *srcPtr,
+            const sp<IMemory> &sharedBuffer, size_t offset,
             const CryptoPlugin::SubSample *subSamples, size_t numSubSamples,
             void *dstPtr,
             AString *errorDetailMsg) = 0;
@@ -61,6 +64,9 @@
     virtual status_t onTransact(
             uint32_t code, const Parcel &data, Parcel *reply,
             uint32_t flags = 0);
+private:
+    void readVector(const Parcel &data, Vector<uint8_t> &vector) const;
+    void writeVector(Parcel *reply, Vector<uint8_t> const &vector) const;
 };
 
 }  // namespace android
diff --git a/include/media/stagefright/MediaClock.h b/include/media/stagefright/MediaClock.h
index e9c09a1..dd1a809 100644
--- a/include/media/stagefright/MediaClock.h
+++ b/include/media/stagefright/MediaClock.h
@@ -42,6 +42,7 @@
     void updateMaxTimeMedia(int64_t maxTimeMediaUs);
 
     void setPlaybackRate(float rate);
+    float getPlaybackRate() const;
 
     // query media time corresponding to real time |realUs|, and save the
     // result in |outMediaUs|.
diff --git a/include/media/stagefright/MediaCodec.h b/include/media/stagefright/MediaCodec.h
index d055341..0786fb9 100644
--- a/include/media/stagefright/MediaCodec.h
+++ b/include/media/stagefright/MediaCodec.h
@@ -30,8 +30,10 @@
 struct AReplyToken;
 struct AString;
 struct CodecBase;
-struct ICrypto;
 struct IBatteryStats;
+struct ICrypto;
+struct IMemory;
+struct MemoryDealer;
 struct SoftwareRenderer;
 struct Surface;
 
@@ -51,7 +53,13 @@
         CB_OUTPUT_AVAILABLE = 2,
         CB_ERROR = 3,
         CB_OUTPUT_FORMAT_CHANGED = 4,
-        CB_RESOURCE_RECLAIMED = 5,
+        CB_CODEC_RELEASED = 5,
+    };
+
+    // used by CB_CODEC_RELEASED to tell the upper layer the cause of the release.
+    enum ReleaseReason {
+        REASON_UNKNOWN = 0,
+        REASON_RECLAIMED,  // resources reclaimed by resource manager
     };
 
     struct BatteryNotifier;
@@ -214,6 +222,7 @@
         uint32_t mBufferID;
         sp<ABuffer> mData;
         sp<ABuffer> mEncryptedData;
+        sp<IMemory> mSharedEncryptedBuffer;
         sp<AMessage> mNotify;
         sp<AMessage> mFormat;
         bool mOwnedByClient;
@@ -232,6 +241,7 @@
     sp<AMessage> mOutputFormat;
     sp<AMessage> mInputFormat;
     sp<AMessage> mCallback;
+    sp<MemoryDealer> mDealer;
 
     bool mBatteryStatNotified;
     bool mIsVideo;
diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index 5644428..6cc2e2b 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -25,6 +25,7 @@
 #include <utils/Log.h>
 #include <utils/RefBase.h>
 #include <audio_utils/roundup.h>
+#include <media/AudioResamplerPublic.h>
 #include <media/SingleStateQueue.h>
 
 namespace android {
@@ -113,6 +114,14 @@
                     mPosLoopQueue;
 };
 
+
+struct AudioTrackPlaybackRate {
+    float mSpeed;
+    float mPitch;
+};
+
+typedef SingleStateQueue<AudioTrackPlaybackRate> AudioTrackPlaybackRateQueue;
+
 // ----------------------------------------------------------------------------
 
 // Important: do not add any virtual methods, including ~
@@ -159,6 +168,8 @@
                 uint32_t    mSampleRate;    // AudioTrack only: client's requested sample rate in Hz
                                             // or 0 == default. Write-only client, read-only server.
 
+                AudioTrackPlaybackRateQueue::Shared mPlaybackRateQueue;
+
                 // client write-only, server read-only
                 uint16_t    mSendLevel;      // Fixed point U4.12 so 0x1000 means 1.0
 
@@ -313,7 +324,8 @@
     AudioTrackClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount,
             size_t frameSize, bool clientInServer = false)
         : ClientProxy(cblk, buffers, frameCount, frameSize, true /*isOut*/,
-          clientInServer) { }
+          clientInServer),
+          mPlaybackRateMutator(&cblk->mPlaybackRateQueue) { }
     virtual ~AudioTrackClientProxy() { }
 
     // No barriers on the following operations, so the ordering of loads/stores
@@ -333,6 +345,13 @@
         mCblk->mSampleRate = sampleRate;
     }
 
+    void        setPlaybackRate(float speed, float pitch) {
+        AudioTrackPlaybackRate playbackRate;
+        playbackRate.mSpeed = speed;
+        playbackRate.mPitch = pitch;
+        mPlaybackRateMutator.push(playbackRate);
+    }
+
     virtual void flush();
 
     virtual uint32_t    getUnderrunFrames() const {
@@ -344,6 +363,9 @@
     bool        getStreamEndDone() const;
 
     status_t    waitStreamEndDone(const struct timespec *requested);
+
+private:
+    AudioTrackPlaybackRateQueue::Mutator   mPlaybackRateMutator;
 };
 
 class StaticAudioTrackClientProxy : public AudioTrackClientProxy {
@@ -458,8 +480,11 @@
 public:
     AudioTrackServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount,
             size_t frameSize, bool clientInServer = false, uint32_t sampleRate = 0)
-        : ServerProxy(cblk, buffers, frameCount, frameSize, true /*isOut*/, clientInServer) {
+        : ServerProxy(cblk, buffers, frameCount, frameSize, true /*isOut*/, clientInServer),
+          mPlaybackRateObserver(&cblk->mPlaybackRateQueue) {
         mCblk->mSampleRate = sampleRate;
+        mPlaybackRate.mSpeed = AUDIO_TIMESTRETCH_SPEED_NORMAL;
+        mPlaybackRate.mPitch = AUDIO_TIMESTRETCH_PITCH_NORMAL;
     }
 protected:
     virtual ~AudioTrackServerProxy() { }
@@ -493,6 +518,13 @@
 
     // Return the total number of frames that AudioFlinger has obtained and released
     virtual size_t      framesReleased() const { return mCblk->mServer; }
+
+    // Return the playback speed and pitch read atomically. Not multi-thread safe on server side.
+    void                getPlaybackRate(float *speed, float *pitch);
+
+private:
+    AudioTrackPlaybackRate                  mPlaybackRate;  // last observed playback rate
+    AudioTrackPlaybackRateQueue::Observer   mPlaybackRateObserver;
 };
 
 class StaticAudioTrackServerProxy : public AudioTrackServerProxy {
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 100a914..f4cdde2 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -189,13 +189,9 @@
     }
 
     // validate parameters
-    if (!audio_is_valid_format(format)) {
-        ALOGE("Invalid format %#x", format);
-        return BAD_VALUE;
-    }
-    // Temporary restriction: AudioFlinger currently supports 16-bit PCM only
-    if (format != AUDIO_FORMAT_PCM_16_BIT) {
-        ALOGE("Format %#x is not supported", format);
+    // AudioFlinger capture only supports linear PCM
+    if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) {
+        ALOGE("Format %#x is not linear pcm", format);
         return BAD_VALUE;
     }
     mFormat = format;
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 9e9ec5b..89138e2 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -56,6 +56,24 @@
     return convertTimespecToUs(tv);
 }
 
+// Must match similar computation in createTrack_l in Threads.cpp.
+// TODO: Move to a common library
+static size_t calculateMinFrameCount(
+        uint32_t afLatencyMs, uint32_t afFrameCount, uint32_t afSampleRate,
+        uint32_t sampleRate, float speed)
+{
+    // Ensure that buffer depth covers at least audio hardware latency
+    uint32_t minBufCount = afLatencyMs / ((1000 * afFrameCount) / afSampleRate);
+    if (minBufCount < 2) {
+        minBufCount = 2;
+    }
+    ALOGV("calculateMinFrameCount afLatency %u  afFrameCount %u  afSampleRate %u  "
+            "sampleRate %u  speed %f  minBufCount: %u",
+            afLatencyMs, afFrameCount, afSampleRate, sampleRate, speed, minBufCount);
+    return minBufCount * sourceFramesNeededWithTimestretch(
+            sampleRate, afFrameCount, afSampleRate, speed);
+}
+
 // static
 status_t AudioTrack::getMinFrameCount(
         size_t* frameCount,
@@ -94,13 +112,10 @@
         return status;
     }
 
-    // Ensure that buffer depth covers at least audio hardware latency
-    uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate);
-    if (minBufCount < 2) {
-        minBufCount = 2;
-    }
+    // When called from createTrack, speed is 1.0f (normal speed).
+    // This is rechecked again on setting playback rate (TODO: on setting sample rate, too).
+    *frameCount = calculateMinFrameCount(afLatency, afFrameCount, afSampleRate, sampleRate, 1.0f);
 
-    *frameCount = minBufCount * sourceFramesNeeded(sampleRate, afFrameCount, afSampleRate);
     // The formula above should always produce a non-zero value under normal circumstances:
     // AudioTrack.SAMPLE_RATE_HZ_MIN <= sampleRate <= AudioTrack.SAMPLE_RATE_HZ_MAX.
     // Return error in the unlikely event that it does not, as that's part of the API contract.
@@ -109,8 +124,8 @@
                 streamType, sampleRate);
         return BAD_VALUE;
     }
-    ALOGV("getMinFrameCount=%zu: afFrameCount=%zu, minBufCount=%u, afSampleRate=%u, afLatency=%u",
-            *frameCount, afFrameCount, minBufCount, afSampleRate, afLatency);
+    ALOGV("getMinFrameCount=%zu: afFrameCount=%zu, afSampleRate=%u, afLatency=%u",
+            *frameCount, afFrameCount, afSampleRate, afLatency);
     return NO_ERROR;
 }
 
@@ -360,6 +375,8 @@
         return BAD_VALUE;
     }
     mSampleRate = sampleRate;
+    mSpeed = AUDIO_TIMESTRETCH_SPEED_NORMAL;
+    mPitch = AUDIO_TIMESTRETCH_PITCH_NORMAL;
 
     // Make copy of input parameter offloadInfo so that in the future:
     //  (a) createTrack_l doesn't need it as an input parameter
@@ -689,6 +706,7 @@
     if (rate == 0 || rate > afSamplingRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
         return BAD_VALUE;
     }
+    // TODO: Should we also check if the buffer size is compatible?
 
     mSampleRate = rate;
     mProxy->setSampleRate(rate);
@@ -719,6 +737,42 @@
     return mSampleRate;
 }
 
+status_t AudioTrack::setPlaybackRate(float speed, float pitch)
+{
+    if (speed < AUDIO_TIMESTRETCH_SPEED_MIN
+            || speed > AUDIO_TIMESTRETCH_SPEED_MAX
+            || pitch < AUDIO_TIMESTRETCH_PITCH_MIN
+            || pitch > AUDIO_TIMESTRETCH_PITCH_MAX) {
+        return BAD_VALUE;
+    }
+    AutoMutex lock(mLock);
+    if (speed == mSpeed && pitch == mPitch) {
+        return NO_ERROR;
+    }
+    if (mIsTimed || isOffloadedOrDirect_l()) {
+        return INVALID_OPERATION;
+    }
+    if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
+        return INVALID_OPERATION;
+    }
+    // Check if the buffer size is compatible.
+    if (!isSampleRateSpeedAllowed_l(mSampleRate, speed)) {
+        ALOGV("setPlaybackRate(%f, %f) failed", speed, pitch);
+        return BAD_VALUE;
+    }
+    mSpeed = speed;
+    mPitch = pitch;
+    mProxy->setPlaybackRate(speed, pitch);
+    return NO_ERROR;
+}
+
+void AudioTrack::getPlaybackRate(float *speed, float *pitch) const
+{
+    AutoMutex lock(mLock);
+    *speed = mSpeed;
+    *pitch = mPitch;
+}
+
 status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount)
 {
     if (mSharedBuffer == 0 || mIsTimed || isOffloadedOrDirect()) {
@@ -1086,8 +1140,16 @@
         // there _is_ a frameCount parameter.  We silently ignore it.
         frameCount = mSharedBuffer->size() / mFrameSize;
     } else {
-        // For fast and normal streaming tracks,
-        // the frame count calculations and checks are done by server
+        // For fast tracks the frame count calculations and checks are done by server
+
+        if ((mFlags & AUDIO_OUTPUT_FLAG_FAST) == 0) {
+            // for normal tracks precompute the frame count based on speed.
+            const size_t minFrameCount = calculateMinFrameCount(
+                    afLatency, afFrameCount, afSampleRate, mSampleRate, mSpeed);
+            if (frameCount < minFrameCount) {
+                frameCount = minFrameCount;
+            }
+        }
     }
 
     IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT;
@@ -1230,6 +1292,7 @@
     }
 
     mAudioTrack->attachAuxEffect(mAuxEffectId);
+    // FIXME doesn't take into account speed or future sample rate changes (until restoreTrack)
     // FIXME don't believe this lie
     mLatency = afLatency + (1000*frameCount) / mSampleRate;
 
@@ -1255,6 +1318,7 @@
 
     mProxy->setSendLevel(mSendLevel);
     mProxy->setSampleRate(mSampleRate);
+    mProxy->setPlaybackRate(mSpeed, mPitch);
     mProxy->setMinimum(mNotificationFramesAct);
 
     mDeathNotifier = new DeathNotifier(this);
@@ -1617,6 +1681,7 @@
 
     // Cache other fields that will be needed soon
     uint32_t sampleRate = mSampleRate;
+    float speed = mSpeed;
     uint32_t notificationFrames = mNotificationFramesAct;
     if (mRefreshRemaining) {
         mRefreshRemaining = false;
@@ -1745,7 +1810,7 @@
     if (minFrames != (uint32_t) ~0) {
         // This "fudge factor" avoids soaking CPU, and compensates for late progress by server
         static const nsecs_t kFudgeNs = 10000000LL; // 10 ms
-        ns = ((minFrames * 1000000000LL) / sampleRate) + kFudgeNs;
+        ns = ((double)minFrames * 1000000000) / ((double)sampleRate * speed) + kFudgeNs;
     }
 
     // If not supplying data by EVENT_MORE_DATA, then we're done
@@ -1786,7 +1851,8 @@
         if (mRetryOnPartialBuffer && !isOffloaded()) {
             mRetryOnPartialBuffer = false;
             if (avail < mRemainingFrames) {
-                int64_t myns = ((mRemainingFrames - avail) * 1100000000LL) / sampleRate;
+                int64_t myns = ((double)(mRemainingFrames - avail) * 1100000000)
+                        / ((double)sampleRate * speed);
                 if (ns < 0 || myns < ns) {
                     ns = myns;
                 }
@@ -1841,7 +1907,7 @@
         // that total to a sum == notificationFrames.
         if (0 < misalignment && misalignment <= mRemainingFrames) {
             mRemainingFrames = misalignment;
-            return (mRemainingFrames * 1100000000LL) / sampleRate;
+            return ((double)mRemainingFrames * 1100000000) / ((double)sampleRate * speed);
         }
 #endif
 
@@ -1936,6 +2002,41 @@
     return mPosition += (uint32_t) delta;
 }
 
+bool AudioTrack::isSampleRateSpeedAllowed_l(uint32_t sampleRate, float speed) const
+{
+    // applicable for mixing tracks only (not offloaded or direct)
+    if (mStaticProxy != 0) {
+        return true; // static tracks do not have issues with buffer sizing.
+    }
+    status_t status;
+    uint32_t afLatency;
+    status = AudioSystem::getLatency(mOutput, &afLatency);
+    if (status != NO_ERROR) {
+        ALOGE("getLatency(%d) failed status %d", mOutput, status);
+        return false;
+    }
+
+    size_t afFrameCount;
+    status = AudioSystem::getFrameCount(mOutput, &afFrameCount);
+    if (status != NO_ERROR) {
+        ALOGE("getFrameCount(output=%d) status %d", mOutput, status);
+        return false;
+    }
+
+    uint32_t afSampleRate;
+    status = AudioSystem::getSamplingRate(mOutput, &afSampleRate);
+    if (status != NO_ERROR) {
+        ALOGE("getSamplingRate(output=%d) status %d", mOutput, status);
+        return false;
+    }
+
+    const size_t minFrameCount =
+            calculateMinFrameCount(afLatency, afFrameCount, afSampleRate, sampleRate, speed);
+    ALOGV("isSampleRateSpeedAllowed_l mFrameCount %zu  minFrameCount %zu",
+            mFrameCount, minFrameCount);
+    return mFrameCount >= minFrameCount;
+}
+
 status_t AudioTrack::setParameters(const String8& keyValuePairs)
 {
     AutoMutex lock(mLock);
@@ -2001,7 +2102,8 @@
                     return WOULD_BLOCK;  // stale timestamp time, occurs before start.
                 }
                 const int64_t deltaTimeUs = timestampTimeUs - mStartUs;
-                const int64_t deltaPositionByUs = timestamp.mPosition * 1000000LL / mSampleRate;
+                const int64_t deltaPositionByUs = (double)timestamp.mPosition * 1000000
+                        / ((double)mSampleRate * mSpeed);
 
                 if (deltaPositionByUs > deltaTimeUs + kTimeJitterUs) {
                     // Verify that the counter can't count faster than the sample rate
@@ -2088,7 +2190,8 @@
     snprintf(buffer, 255, "  format(%d), channel count(%d), frame count(%zu)\n", mFormat,
             mChannelCount, mFrameCount);
     result.append(buffer);
-    snprintf(buffer, 255, "  sample rate(%u), status(%d)\n", mSampleRate, mStatus);
+    snprintf(buffer, 255, "  sample rate(%u), speed(%f), status(%d)\n",
+            mSampleRate, mSpeed, mStatus);
     result.append(buffer);
     snprintf(buffer, 255, "  state(%d), latency (%d)\n", mState, mLatency);
     result.append(buffer);
diff --git a/media/libmedia/AudioTrackShared.cpp b/media/libmedia/AudioTrackShared.cpp
index 6d5f1af..ba67b40 100644
--- a/media/libmedia/AudioTrackShared.cpp
+++ b/media/libmedia/AudioTrackShared.cpp
@@ -793,6 +793,16 @@
     (void) android_atomic_or(CBLK_UNDERRUN, &cblk->mFlags);
 }
 
+void AudioTrackServerProxy::getPlaybackRate(float *speed, float *pitch)
+{   // do not call from multiple threads without holding lock
+    AudioTrackPlaybackRate playbackRate;
+    if (mPlaybackRateObserver.poll(playbackRate)) {
+        mPlaybackRate = playbackRate;
+    }
+    *speed = mPlaybackRate.mSpeed;
+    *pitch = mPlaybackRate.mPitch;
+}
+
 // ---------------------------------------------------------------------------
 
 StaticAudioTrackServerProxy::StaticAudioTrackServerProxy(audio_track_cblk_t* cblk, void *buffers,
diff --git a/media/libmedia/ICrypto.cpp b/media/libmedia/ICrypto.cpp
index c26c5bf..9246a7c 100644
--- a/media/libmedia/ICrypto.cpp
+++ b/media/libmedia/ICrypto.cpp
@@ -19,6 +19,7 @@
 #include <utils/Log.h>
 
 #include <binder/Parcel.h>
+#include <binder/IMemory.h>
 #include <media/ICrypto.h>
 #include <media/stagefright/MediaErrors.h>
 #include <media/stagefright/foundation/ADebug.h>
@@ -34,6 +35,7 @@
     REQUIRES_SECURE_COMPONENT,
     DECRYPT,
     NOTIFY_RESOLUTION,
+    SET_MEDIADRM_SESSION,
 };
 
 struct BpCrypto : public BpInterface<ICrypto> {
@@ -97,7 +99,7 @@
             const uint8_t key[16],
             const uint8_t iv[16],
             CryptoPlugin::Mode mode,
-            const void *srcPtr,
+            const sp<IMemory> &sharedBuffer, size_t offset,
             const CryptoPlugin::SubSample *subSamples, size_t numSubSamples,
             void *dstPtr,
             AString *errorDetailMsg) {
@@ -126,7 +128,8 @@
         }
 
         data.writeInt32(totalSize);
-        data.write(srcPtr, totalSize);
+        data.writeStrongBinder(IInterface::asBinder(sharedBuffer));
+        data.writeInt32(offset);
 
         data.writeInt32(numSubSamples);
         data.write(subSamples, sizeof(CryptoPlugin::SubSample) * numSubSamples);
@@ -159,7 +162,28 @@
         remote()->transact(NOTIFY_RESOLUTION, data, &reply);
     }
 
+    virtual status_t setMediaDrmSession(const Vector<uint8_t> &sessionId) {
+        Parcel data, reply;
+        data.writeInterfaceToken(ICrypto::getInterfaceDescriptor());
+
+        writeVector(data, sessionId);
+        remote()->transact(SET_MEDIADRM_SESSION, data, &reply);
+
+        return reply.readInt32();
+    }
+
 private:
+    void readVector(Parcel &reply, Vector<uint8_t> &vector) const {
+        uint32_t size = reply.readInt32();
+        vector.insertAt((size_t)0, size);
+        reply.read(vector.editArray(), size);
+    }
+
+    void writeVector(Parcel &data, Vector<uint8_t> const &vector) const {
+        data.writeInt32(vector.size());
+        data.write(vector.array(), vector.size());
+    }
+
     DISALLOW_EVIL_CONSTRUCTORS(BpCrypto);
 };
 
@@ -167,6 +191,17 @@
 
 ////////////////////////////////////////////////////////////////////////////////
 
+void BnCrypto::readVector(const Parcel &data, Vector<uint8_t> &vector) const {
+    uint32_t size = data.readInt32();
+    vector.insertAt((size_t)0, size);
+    data.read(vector.editArray(), size);
+}
+
+void BnCrypto::writeVector(Parcel *reply, Vector<uint8_t> const &vector) const {
+    reply->writeInt32(vector.size());
+    reply->write(vector.array(), vector.size());
+}
+
 status_t BnCrypto::onTransact(
     uint32_t code, const Parcel &data, Parcel *reply, uint32_t flags) {
     switch (code) {
@@ -245,8 +280,9 @@
             data.read(iv, sizeof(iv));
 
             size_t totalSize = data.readInt32();
-            void *srcData = malloc(totalSize);
-            data.read(srcData, totalSize);
+            sp<IMemory> sharedBuffer =
+                interface_cast<IMemory>(data.readStrongBinder());
+            int32_t offset = data.readInt32();
 
             int32_t numSubSamples = data.readInt32();
 
@@ -265,15 +301,21 @@
             }
 
             AString errorDetailMsg;
-            ssize_t result = decrypt(
+            ssize_t result;
+
+            if (offset + totalSize > sharedBuffer->size()) {
+                result = -EINVAL;
+            } else {
+                result = decrypt(
                     secure,
                     key,
                     iv,
                     mode,
-                    srcData,
+                    sharedBuffer, offset,
                     subSamples, numSubSamples,
                     dstPtr,
                     &errorDetailMsg);
+            }
 
             reply->writeInt32(result);
 
@@ -294,9 +336,6 @@
             delete[] subSamples;
             subSamples = NULL;
 
-            free(srcData);
-            srcData = NULL;
-
             return OK;
         }
 
@@ -311,6 +350,15 @@
             return OK;
         }
 
+        case SET_MEDIADRM_SESSION:
+        {
+            CHECK_INTERFACE(IDrm, data, reply);
+            Vector<uint8_t> sessionId;
+            readVector(data, sessionId);
+            reply->writeInt32(setMediaDrmSession(sessionId));
+            return OK;
+        }
+
         default:
             return BBinder::onTransact(code, data, reply, flags);
     }
diff --git a/media/libmediaplayerservice/Crypto.cpp b/media/libmediaplayerservice/Crypto.cpp
index 8ee7c0b..f639193 100644
--- a/media/libmediaplayerservice/Crypto.cpp
+++ b/media/libmediaplayerservice/Crypto.cpp
@@ -22,6 +22,7 @@
 
 #include "Crypto.h"
 
+#include <binder/IMemory.h>
 #include <media/hardware/CryptoAPI.h>
 #include <media/stagefright/foundation/ADebug.h>
 #include <media/stagefright/foundation/AString.h>
@@ -238,7 +239,7 @@
         const uint8_t key[16],
         const uint8_t iv[16],
         CryptoPlugin::Mode mode,
-        const void *srcPtr,
+        const sp<IMemory> &sharedBuffer, size_t offset,
         const CryptoPlugin::SubSample *subSamples, size_t numSubSamples,
         void *dstPtr,
         AString *errorDetailMsg) {
@@ -252,6 +253,8 @@
         return -EINVAL;
     }
 
+    const void *srcPtr = static_cast<uint8_t *>(sharedBuffer->pointer()) + offset;
+
     return mPlugin->decrypt(
             secure, key, iv, mode, srcPtr, subSamples, numSubSamples, dstPtr,
             errorDetailMsg);
@@ -265,4 +268,14 @@
     }
 }
 
+status_t Crypto::setMediaDrmSession(const Vector<uint8_t> &sessionId) {
+    Mutex::Autolock autoLock(mLock);
+
+    status_t result = NO_INIT;
+    if (mInitCheck == OK && mPlugin != NULL) {
+        result = mPlugin->setMediaDrmSession(sessionId);
+    }
+    return result;
+}
+
 }  // namespace android
diff --git a/media/libmediaplayerservice/Crypto.h b/media/libmediaplayerservice/Crypto.h
index 0037c2e..99ea95d 100644
--- a/media/libmediaplayerservice/Crypto.h
+++ b/media/libmediaplayerservice/Crypto.h
@@ -47,12 +47,14 @@
 
     virtual void notifyResolution(uint32_t width, uint32_t height);
 
+    virtual status_t setMediaDrmSession(const Vector<uint8_t> &sessionId);
+
     virtual ssize_t decrypt(
             bool secure,
             const uint8_t key[16],
             const uint8_t iv[16],
             CryptoPlugin::Mode mode,
-            const void *srcPtr,
+            const sp<IMemory> &sharedBuffer, size_t offset,
             const CryptoPlugin::SubSample *subSamples, size_t numSubSamples,
             void *dstPtr,
             AString *errorDetailMsg);
diff --git a/media/libmediaplayerservice/Drm.cpp b/media/libmediaplayerservice/Drm.cpp
index 49e01d1..62cf3e5 100644
--- a/media/libmediaplayerservice/Drm.cpp
+++ b/media/libmediaplayerservice/Drm.cpp
@@ -136,25 +136,57 @@
 
     if (listener != NULL) {
         Parcel obj;
-        if (sessionId && sessionId->size()) {
-            obj.writeInt32(sessionId->size());
-            obj.write(sessionId->array(), sessionId->size());
-        } else {
-            obj.writeInt32(0);
-        }
-
-        if (data && data->size()) {
-            obj.writeInt32(data->size());
-            obj.write(data->array(), data->size());
-        } else {
-            obj.writeInt32(0);
-        }
+        writeByteArray(obj, sessionId);
+        writeByteArray(obj, data);
 
         Mutex::Autolock lock(mNotifyLock);
         listener->notify(eventType, extra, &obj);
     }
 }
 
+void Drm::sendExpirationUpdate(Vector<uint8_t> const *sessionId,
+                               int64_t expiryTimeInMS)
+{
+    mEventLock.lock();
+    sp<IDrmClient> listener = mListener;
+    mEventLock.unlock();
+
+    if (listener != NULL) {
+        Parcel obj;
+        writeByteArray(obj, sessionId);
+        obj.writeInt64(expiryTimeInMS);
+
+        Mutex::Autolock lock(mNotifyLock);
+        listener->notify(DrmPlugin::kDrmPluginEventExpirationUpdate, 0, &obj);
+    }
+}
+
+void Drm::sendKeysChange(Vector<uint8_t> const *sessionId,
+                         Vector<DrmPlugin::KeyStatus> const *keyStatusList,
+                         bool hasNewUsableKey)
+{
+    mEventLock.lock();
+    sp<IDrmClient> listener = mListener;
+    mEventLock.unlock();
+
+    if (listener != NULL) {
+        Parcel obj;
+        writeByteArray(obj, sessionId);
+
+        size_t nkeys = keyStatusList->size();
+        obj.writeInt32(keyStatusList->size());
+        for (size_t i = 0; i < nkeys; ++i) {
+            const DrmPlugin::KeyStatus *keyStatus = &keyStatusList->itemAt(i);
+            writeByteArray(obj, &keyStatus->mKeyId);
+            obj.writeInt32(keyStatus->mType);
+        }
+        obj.writeInt32(hasNewUsableKey);
+
+        Mutex::Autolock lock(mNotifyLock);
+        listener->notify(DrmPlugin::kDrmPluginEventKeysChange, 0, &obj);
+    }
+}
+
 /*
  * Search the plugins directory for a plugin that supports the scheme
  * specified by uuid
@@ -756,4 +788,14 @@
     closeFactory();
 }
 
+void Drm::writeByteArray(Parcel &obj, Vector<uint8_t> const *array)
+{
+    if (array && array->size()) {
+        obj.writeInt32(array->size());
+        obj.write(array->array(), array->size());
+    } else {
+        obj.writeInt32(0);
+    }
+}
+
 }  // namespace android
diff --git a/media/libmediaplayerservice/Drm.h b/media/libmediaplayerservice/Drm.h
index 7e8f246..1591738 100644
--- a/media/libmediaplayerservice/Drm.h
+++ b/media/libmediaplayerservice/Drm.h
@@ -133,6 +133,13 @@
                            Vector<uint8_t> const *sessionId,
                            Vector<uint8_t> const *data);
 
+    virtual void sendExpirationUpdate(Vector<uint8_t> const *sessionId,
+                                      int64_t expiryTimeInMS);
+
+    virtual void sendKeysChange(Vector<uint8_t> const *sessionId,
+                                Vector<DrmPlugin::KeyStatus> const *keyStatusList,
+                                bool hasNewUsableKey);
+
     virtual void binderDied(const wp<IBinder> &the_late_who);
 
 private:
@@ -157,7 +164,7 @@
     void findFactoryForScheme(const uint8_t uuid[16]);
     bool loadLibraryForScheme(const String8 &path, const uint8_t uuid[16]);
     void closeFactory();
-
+    void writeByteArray(Parcel &obj, Vector<uint8_t> const *array);
 
     DISALLOW_EVIL_CONSTRUCTORS(Drm);
 };
diff --git a/media/libstagefright/ESDS.cpp b/media/libstagefright/ESDS.cpp
index 427bf7b..8fbb57c 100644
--- a/media/libstagefright/ESDS.cpp
+++ b/media/libstagefright/ESDS.cpp
@@ -136,6 +136,8 @@
     --size;
 
     if (streamDependenceFlag) {
+        if (size < 2)
+            return ERROR_MALFORMED;
         offset += 2;
         size -= 2;
     }
@@ -145,11 +147,15 @@
             return ERROR_MALFORMED;
         }
         unsigned URLlength = mData[offset];
+        if (URLlength >= size)
+            return ERROR_MALFORMED;
         offset += URLlength + 1;
         size -= URLlength + 1;
     }
 
     if (OCRstreamFlag) {
+        if (size < 2)
+            return ERROR_MALFORMED;
         offset += 2;
         size -= 2;
 
diff --git a/media/libstagefright/MPEG4Extractor.cpp b/media/libstagefright/MPEG4Extractor.cpp
index d0f42cc..f7fa2b6 100644
--- a/media/libstagefright/MPEG4Extractor.cpp
+++ b/media/libstagefright/MPEG4Extractor.cpp
@@ -874,6 +874,9 @@
                     }
                 }
 
+                if (mLastTrack == NULL)
+                    return ERROR_MALFORMED;
+
                 mLastTrack->sampleTable = new SampleTable(mDataSource);
             }
 
@@ -1028,6 +1031,10 @@
             }
             original_fourcc = ntohl(original_fourcc);
             ALOGV("read original format: %d", original_fourcc);
+
+            if (mLastTrack == NULL)
+                return ERROR_MALFORMED;
+
             mLastTrack->meta->setCString(kKeyMIMEType, FourCC2MIME(original_fourcc));
             uint32_t num_channels = 0;
             uint32_t sample_rate = 0;
@@ -1083,6 +1090,9 @@
                 return ERROR_IO;
             }
 
+            if (mLastTrack == NULL)
+                return ERROR_MALFORMED;
+
             mLastTrack->meta->setInt32(kKeyCryptoMode, defaultAlgorithmId);
             mLastTrack->meta->setInt32(kKeyCryptoDefaultIVSize, defaultIVSize);
             mLastTrack->meta->setData(kKeyCryptoKey, 'tenc', defaultKeyId, 16);
@@ -1168,6 +1178,11 @@
                 return ERROR_IO;
             }
 
+            if (!timescale) {
+                ALOGE("timescale should not be ZERO.");
+                return ERROR_MALFORMED;
+            }
+
             mLastTrack->timescale = ntohl(timescale);
 
             // 14496-12 says all ones means indeterminate, but some files seem to use
@@ -1193,7 +1208,7 @@
                     duration = ntohl(duration32);
                 }
             }
-            if (duration != 0) {
+            if (duration != 0 && mLastTrack->timescale != 0) {
                 mLastTrack->meta->setInt64(
                         kKeyDuration, (duration * 1000000) / mLastTrack->timescale);
             }
@@ -1257,6 +1272,10 @@
                 // display the timed text.
                 // For encrypted files, there may also be more than one entry.
                 const char *mime;
+
+                if (mLastTrack == NULL)
+                    return ERROR_MALFORMED;
+
                 CHECK(mLastTrack->meta->findCString(kKeyMIMEType, &mime));
                 if (strcasecmp(mime, MEDIA_MIMETYPE_TEXT_3GPP) &&
                         strcasecmp(mime, "application/octet-stream")) {
@@ -1303,6 +1322,9 @@
             uint16_t sample_size = U16_AT(&buffer[18]);
             uint32_t sample_rate = U32_AT(&buffer[24]) >> 16;
 
+            if (mLastTrack == NULL)
+                return ERROR_MALFORMED;
+
             if (chunk_type != FOURCC('e', 'n', 'c', 'a')) {
                 // if the chunk type is enca, we'll get the type from the sinf/frma box later
                 mLastTrack->meta->setCString(kKeyMIMEType, FourCC2MIME(chunk_type));
@@ -1364,6 +1386,9 @@
             // printf("*** coding='%s' width=%d height=%d\n",
             //        chunk, width, height);
 
+            if (mLastTrack == NULL)
+                return ERROR_MALFORMED;
+
             if (chunk_type != FOURCC('e', 'n', 'c', 'v')) {
                 // if the chunk type is encv, we'll get the type from the sinf/frma box later
                 mLastTrack->meta->setCString(kKeyMIMEType, FourCC2MIME(chunk_type));
@@ -1389,6 +1414,9 @@
         case FOURCC('s', 't', 'c', 'o'):
         case FOURCC('c', 'o', '6', '4'):
         {
+            if ((mLastTrack == NULL) || (mLastTrack->sampleTable == NULL))
+                return ERROR_MALFORMED;
+
             status_t err =
                 mLastTrack->sampleTable->setChunkOffsetParams(
                         chunk_type, data_offset, chunk_data_size);
@@ -1404,6 +1432,9 @@
 
         case FOURCC('s', 't', 's', 'c'):
         {
+            if ((mLastTrack == NULL) || (mLastTrack->sampleTable == NULL))
+                return ERROR_MALFORMED;
+
             status_t err =
                 mLastTrack->sampleTable->setSampleToChunkParams(
                         data_offset, chunk_data_size);
@@ -1420,6 +1451,9 @@
         case FOURCC('s', 't', 's', 'z'):
         case FOURCC('s', 't', 'z', '2'):
         {
+            if ((mLastTrack == NULL) || (mLastTrack->sampleTable == NULL))
+                return ERROR_MALFORMED;
+
             status_t err =
                 mLastTrack->sampleTable->setSampleSizeParams(
                         chunk_type, data_offset, chunk_data_size);
@@ -1489,6 +1523,9 @@
 
         case FOURCC('s', 't', 't', 's'):
         {
+            if ((mLastTrack == NULL) || (mLastTrack->sampleTable == NULL))
+                return ERROR_MALFORMED;
+
             *offset += chunk_size;
 
             status_t err =
@@ -1504,6 +1541,9 @@
 
         case FOURCC('c', 't', 't', 's'):
         {
+            if ((mLastTrack == NULL) || (mLastTrack->sampleTable == NULL))
+                return ERROR_MALFORMED;
+
             *offset += chunk_size;
 
             status_t err =
@@ -1519,6 +1559,9 @@
 
         case FOURCC('s', 't', 's', 's'):
         {
+            if ((mLastTrack == NULL) || (mLastTrack->sampleTable == NULL))
+                return ERROR_MALFORMED;
+
             *offset += chunk_size;
 
             status_t err =
@@ -1591,6 +1634,9 @@
                 return ERROR_MALFORMED;
             }
 
+            if (mLastTrack == NULL)
+                return ERROR_MALFORMED;
+
             mLastTrack->meta->setData(
                     kKeyESDS, kTypeESDS, &buffer[4], chunk_data_size - 4);
 
@@ -1623,6 +1669,9 @@
                 return ERROR_IO;
             }
 
+            if (mLastTrack == NULL)
+                return ERROR_MALFORMED;
+
             mLastTrack->meta->setData(
                     kKeyAVCC, kTypeAVCC, buffer->data(), chunk_data_size);
 
@@ -1637,6 +1686,9 @@
                 return ERROR_IO;
             }
 
+            if (mLastTrack == NULL)
+                return ERROR_MALFORMED;
+
             mLastTrack->meta->setData(
                     kKeyHVCC, kTypeHVCC, buffer->data(), chunk_data_size);
 
@@ -1670,6 +1722,9 @@
                 return ERROR_IO;
             }
 
+            if (mLastTrack == NULL)
+                return ERROR_MALFORMED;
+
             mLastTrack->meta->setData(kKeyD263, kTypeD263, buffer, chunk_data_size);
 
             break;
@@ -1767,7 +1822,7 @@
                 }
                 duration = d32;
             }
-            if (duration != 0) {
+            if (duration != 0 && mHeaderTimescale != 0) {
                 mFileMetaData->setInt64(kKeyDuration, duration * 1000000 / mHeaderTimescale);
             }
 
@@ -1816,7 +1871,7 @@
                 return ERROR_MALFORMED;
             }
 
-            if (duration != 0) {
+            if (duration != 0 && mHeaderTimescale != 0) {
                 mFileMetaData->setInt64(kKeyDuration, duration * 1000000 / mHeaderTimescale);
             }
 
@@ -1851,6 +1906,9 @@
                 return ERROR_IO;
             }
 
+            if (mLastTrack == NULL)
+                return ERROR_MALFORMED;
+
             uint32_t type = ntohl(buffer);
             // For the 3GPP file format, the handler-type within the 'hdlr' box
             // shall be 'text'. We also want to support 'sbtl' handler type
@@ -1883,6 +1941,9 @@
 
         case FOURCC('t', 'x', '3', 'g'):
         {
+            if (mLastTrack == NULL)
+                return ERROR_MALFORMED;
+
             uint32_t type;
             const void *data;
             size_t size = 0;
@@ -2024,6 +2085,8 @@
         return ERROR_MALFORMED;
     }
     ALOGV("sidx refid/timescale: %d/%d", referenceId, timeScale);
+    if (timeScale == 0)
+        return ERROR_MALFORMED;
 
     uint64_t earliestPresentationTime;
     uint64_t firstOffset;
@@ -2107,6 +2170,9 @@
 
     uint64_t sidxDuration = total_duration * 1000000 / timeScale;
 
+    if (mLastTrack == NULL)
+        return ERROR_MALFORMED;
+
     int64_t metaDuration;
     if (!mLastTrack->meta->findInt64(kKeyDuration, &metaDuration) || metaDuration == 0) {
         mLastTrack->meta->setInt64(kKeyDuration, sidxDuration);
@@ -2157,6 +2223,9 @@
         return ERROR_UNSUPPORTED;
     }
 
+    if (mLastTrack == NULL)
+        return ERROR_MALFORMED;
+
     mLastTrack->meta->setInt32(kKeyTrackID, id);
 
     size_t matrixOffset = dynSize + 16;
@@ -2339,6 +2408,9 @@
                     int32_t delay, padding;
                     if (sscanf(mLastCommentData,
                                " %*x %x %x %*x", &delay, &padding) == 2) {
+                        if (mLastTrack == NULL)
+                            return ERROR_MALFORMED;
+
                         mLastTrack->meta->setInt32(kKeyEncoderDelay, delay);
                         mLastTrack->meta->setInt32(kKeyEncoderPadding, padding);
                     }
@@ -2635,6 +2707,11 @@
         return ERROR_MALFORMED;
     }
 
+    if (track->timescale == 0) {
+        ALOGE("timescale invalid.");
+        return ERROR_MALFORMED;
+    }
+
     return OK;
 }
 
@@ -2701,6 +2778,9 @@
 
     if (objectTypeIndication == 0xe1) {
         // This isn't MPEG4 audio at all, it's QCELP 14k...
+        if (mLastTrack == NULL)
+            return ERROR_MALFORMED;
+
         mLastTrack->meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_QCELP);
         return OK;
     }
@@ -2749,6 +2829,9 @@
         objectType = 32 + br.getBits(6);
     }
 
+    if (mLastTrack == NULL)
+        return ERROR_MALFORMED;
+
     //keep AOT type
     mLastTrack->meta->setInt32(kKeyAACAOT, objectType);
 
@@ -2919,6 +3002,9 @@
         return ERROR_UNSUPPORTED;
     }
 
+    if (mLastTrack == NULL)
+        return ERROR_MALFORMED;
+
     int32_t prevSampleRate;
     CHECK(mLastTrack->meta->findInt32(kKeySampleRate, &prevSampleRate));
 
diff --git a/media/libstagefright/MediaClock.cpp b/media/libstagefright/MediaClock.cpp
index 433f555..2641e4e 100644
--- a/media/libstagefright/MediaClock.cpp
+++ b/media/libstagefright/MediaClock.cpp
@@ -92,6 +92,11 @@
     mPlaybackRate = rate;
 }
 
+float MediaClock::getPlaybackRate() const {
+    Mutex::Autolock autoLock(mLock);
+    return mPlaybackRate;
+}
+
 status_t MediaClock::getMediaTime(
         int64_t realUs, int64_t *outMediaUs, bool allowPastMaxTime) const {
     if (outMediaUs == NULL) {
diff --git a/media/libstagefright/MediaCodec.cpp b/media/libstagefright/MediaCodec.cpp
index 0597f1d..8186f63 100644
--- a/media/libstagefright/MediaCodec.cpp
+++ b/media/libstagefright/MediaCodec.cpp
@@ -22,7 +22,9 @@
 #include "include/SoftwareRenderer.h"
 
 #include <binder/IBatteryStats.h>
+#include <binder/IMemory.h>
 #include <binder/IServiceManager.h>
+#include <binder/MemoryDealer.h>
 #include <gui/Surface.h>
 #include <media/ICrypto.h>
 #include <media/stagefright/foundation/ABuffer.h>
@@ -969,6 +971,17 @@
 
                     size_t numBuffers = portDesc->countBuffers();
 
+                    size_t totalSize = 0;
+                    for (size_t i = 0; i < numBuffers; ++i) {
+                        if (portIndex == kPortIndexInput && mCrypto != NULL) {
+                            totalSize += portDesc->bufferAt(i)->capacity();
+                        }
+                    }
+
+                    if (totalSize) {
+                        mDealer = new MemoryDealer(totalSize, "MediaCodec");
+                    }
+
                     for (size_t i = 0; i < numBuffers; ++i) {
                         BufferInfo info;
                         info.mBufferID = portDesc->bufferIDAt(i);
@@ -976,8 +989,10 @@
                         info.mData = portDesc->bufferAt(i);
 
                         if (portIndex == kPortIndexInput && mCrypto != NULL) {
+                            sp<IMemory> mem = mDealer->allocate(info.mData->capacity());
                             info.mEncryptedData =
-                                new ABuffer(info.mData->capacity());
+                                new ABuffer(mem->pointer(), info.mData->capacity());
+                            info.mSharedEncryptedBuffer = mem;
                         }
 
                         buffers->push_back(info);
@@ -1953,7 +1968,8 @@
                 key,
                 iv,
                 mode,
-                info->mEncryptedData->base() + offset,
+                info->mSharedEncryptedBuffer,
+                offset,
                 subSamples,
                 numSubSamples,
                 info->mData->base(),
diff --git a/media/libstagefright/SampleTable.cpp b/media/libstagefright/SampleTable.cpp
index bdd6d56..aba64d5 100644
--- a/media/libstagefright/SampleTable.cpp
+++ b/media/libstagefright/SampleTable.cpp
@@ -230,8 +230,13 @@
         return ERROR_MALFORMED;
     }
 
+    if (SIZE_MAX / sizeof(SampleToChunkEntry) <= mNumSampleToChunkOffsets)
+        return ERROR_OUT_OF_RANGE;
+
     mSampleToChunkEntries =
-        new SampleToChunkEntry[mNumSampleToChunkOffsets];
+        new (std::nothrow) SampleToChunkEntry[mNumSampleToChunkOffsets];
+    if (!mSampleToChunkEntries)
+        return ERROR_OUT_OF_RANGE;
 
     for (uint32_t i = 0; i < mNumSampleToChunkOffsets; ++i) {
         uint8_t buffer[12];
@@ -330,11 +335,13 @@
     }
 
     mTimeToSampleCount = U32_AT(&header[4]);
-    uint64_t allocSize = mTimeToSampleCount * 2 * sizeof(uint32_t);
+    uint64_t allocSize = mTimeToSampleCount * 2 * (uint64_t)sizeof(uint32_t);
     if (allocSize > SIZE_MAX) {
         return ERROR_OUT_OF_RANGE;
     }
-    mTimeToSample = new uint32_t[mTimeToSampleCount * 2];
+    mTimeToSample = new (std::nothrow) uint32_t[mTimeToSampleCount * 2];
+    if (!mTimeToSample)
+        return ERROR_OUT_OF_RANGE;
 
     size_t size = sizeof(uint32_t) * mTimeToSampleCount * 2;
     if (mDataSource->readAt(
@@ -376,12 +383,14 @@
     }
 
     mNumCompositionTimeDeltaEntries = numEntries;
-    uint64_t allocSize = numEntries * 2 * sizeof(uint32_t);
+    uint64_t allocSize = numEntries * 2 * (uint64_t)sizeof(uint32_t);
     if (allocSize > SIZE_MAX) {
         return ERROR_OUT_OF_RANGE;
     }
 
-    mCompositionTimeDeltaEntries = new uint32_t[2 * numEntries];
+    mCompositionTimeDeltaEntries = new (std::nothrow) uint32_t[2 * numEntries];
+    if (!mCompositionTimeDeltaEntries)
+        return ERROR_OUT_OF_RANGE;
 
     if (mDataSource->readAt(
                 data_offset + 8, mCompositionTimeDeltaEntries, numEntries * 8)
@@ -426,12 +435,15 @@
         ALOGV("Table of sync samples is empty or has only a single entry!");
     }
 
-    uint64_t allocSize = mNumSyncSamples * sizeof(uint32_t);
+    uint64_t allocSize = mNumSyncSamples * (uint64_t)sizeof(uint32_t);
     if (allocSize > SIZE_MAX) {
         return ERROR_OUT_OF_RANGE;
     }
 
-    mSyncSamples = new uint32_t[mNumSyncSamples];
+    mSyncSamples = new (std::nothrow) uint32_t[mNumSyncSamples];
+    if (!mSyncSamples)
+        return ERROR_OUT_OF_RANGE;
+
     size_t size = mNumSyncSamples * sizeof(uint32_t);
     if (mDataSource->readAt(mSyncSampleOffset + 8, mSyncSamples, size)
             != (ssize_t)size) {
@@ -499,7 +511,9 @@
         return;
     }
 
-    mSampleTimeEntries = new SampleTimeEntry[mNumSampleSizes];
+    mSampleTimeEntries = new (std::nothrow) SampleTimeEntry[mNumSampleSizes];
+    if (!mSampleTimeEntries)
+        return;
 
     uint32_t sampleIndex = 0;
     uint32_t sampleTime = 0;
diff --git a/media/libstagefright/httplive/LiveSession.cpp b/media/libstagefright/httplive/LiveSession.cpp
index f7a4a0d..74f58e9 100644
--- a/media/libstagefright/httplive/LiveSession.cpp
+++ b/media/libstagefright/httplive/LiveSession.cpp
@@ -498,16 +498,15 @@
 
         case kWhatSeek:
         {
-            sp<AReplyToken> seekReplyID;
-            CHECK(msg->senderAwaitsResponse(&seekReplyID));
-            mSeekReplyID = seekReplyID;
+            if (mReconfigurationInProgress) {
+                msg->post(50000);
+                break;
+            }
+
+            CHECK(msg->senderAwaitsResponse(&mSeekReplyID));
             mSeekReply = new AMessage;
 
-            status_t err = onSeek(msg);
-
-            if (err != OK) {
-                msg->post(50000);
-            }
+            onSeek(msg);
             break;
         }
 
@@ -1372,16 +1371,10 @@
     return audioTime < videoTime ? videoTime : audioTime;
 }
 
-status_t LiveSession::onSeek(const sp<AMessage> &msg) {
+void LiveSession::onSeek(const sp<AMessage> &msg) {
     int64_t timeUs;
     CHECK(msg->findInt64("timeUs", &timeUs));
-
-    if (!mReconfigurationInProgress) {
-        changeConfiguration(timeUs);
-        return OK;
-    } else {
-        return -EWOULDBLOCK;
-    }
+    changeConfiguration(timeUs);
 }
 
 status_t LiveSession::getDuration(int64_t *durationUs) const {
@@ -1462,6 +1455,10 @@
     if (bandwidthIndex >= 0) {
         mOrigBandwidthIndex = mCurBandwidthIndex;
         mCurBandwidthIndex = bandwidthIndex;
+        if (mOrigBandwidthIndex != mCurBandwidthIndex) {
+            ALOGI("#### Starting Bandwidth Switch: %zd => %zd",
+                    mOrigBandwidthIndex, mCurBandwidthIndex);
+        }
     }
     CHECK_LT(mCurBandwidthIndex, mBandwidthItems.size());
     const BandwidthItem &item = mBandwidthItems.itemAt(mCurBandwidthIndex);
@@ -1581,6 +1578,7 @@
 
     if (timeUs >= 0) {
         mLastSeekTimeUs = timeUs;
+        mLastDequeuedTimeUs = timeUs;
 
         for (size_t i = 0; i < mPacketSources.size(); i++) {
             mPacketSources.editValueAt(i)->clear();
@@ -1633,8 +1631,10 @@
             ALOGV("stream %zu changed: oldURI %s, newURI %s", i,
                     mStreams[i].mUri.c_str(), URIs[i].c_str());
             sp<AnotherPacketSource> source = mPacketSources.valueFor(indexToType(i));
-            source->queueDiscontinuity(
-                    ATSParser::DISCONTINUITY_FORMATCHANGE, NULL, true);
+            if (source->getLatestDequeuedMeta() != NULL) {
+                source->queueDiscontinuity(
+                        ATSParser::DISCONTINUITY_FORMATCHANGE, NULL, true);
+            }
         }
         // Determine which decoders to shutdown on the player side,
         // a decoder has to be shutdown if its streamtype was active
@@ -1694,10 +1694,6 @@
             // and resume audio.
             mSwapMask =  mNewStreamMask & mStreamMask & ~resumeMask;
             switching = (mSwapMask != 0);
-            if (!switching) {
-                ALOGV("#### Finishing Bandwidth Switch Early: %zd => %zd",
-                        mOrigBandwidthIndex, mCurBandwidthIndex);
-            }
         }
         mRealTimeBaseUs = ALooper::GetNowUs() - mLastDequeuedTimeUs;
     } else {
@@ -1850,7 +1846,11 @@
         mSwitchInProgress = true;
     } else {
         mStreamMask = mNewStreamMask;
-        mOrigBandwidthIndex = mCurBandwidthIndex;
+        if (mOrigBandwidthIndex != mCurBandwidthIndex) {
+            ALOGV("#### Finished Bandwidth Switch Early: %zd => %zd",
+                    mOrigBandwidthIndex, mCurBandwidthIndex);
+            mOrigBandwidthIndex = mCurBandwidthIndex;
+        }
     }
 
     ALOGV("onChangeConfiguration3: mSwitchInProgress %d, mStreamMask 0x%x",
@@ -1977,11 +1977,19 @@
 
     bool underflow, ready, down, up;
     if (checkBuffering(underflow, ready, down, up)) {
-        if (mInPreparationPhase && ready) {
-            postPrepared(OK);
+        if (mInPreparationPhase) {
+            // Allow down switch even if we're still preparing.
+            //
+            // Some streams have a high bandwidth index as default,
+            // when bandwidth is low, it takes a long time to buffer
+            // to ready mark, then it immediately pauses after start
+            // as we have to do a down switch. It's better experience
+            // to restart from a lower index, if we detect low bw.
+            if (!switchBandwidthIfNeeded(false /* up */, down) && ready) {
+                postPrepared(OK);
+            }
         }
 
-        // don't switch before we report prepared
         if (!mInPreparationPhase) {
             if (ready) {
                 stopBufferingIfNecessary();
@@ -1989,8 +1997,7 @@
                 startBufferingIfNecessary();
             }
             switchBandwidthIfNeeded(up, down);
-       }
-
+        }
     }
 
     schedulePollBuffering();
@@ -2082,7 +2089,8 @@
             if (mPacketSources[i]->isFinished(0 /* duration */)) {
                 percent = 100;
             } else {
-                percent = (int32_t)(100.0 * (mLastDequeuedTimeUs + bufferedDurationUs) / durationUs);
+                percent = (int32_t)(100.0 *
+                        (mLastDequeuedTimeUs + bufferedDurationUs) / durationUs);
             }
             if (minBufferPercent < 0 || percent < minBufferPercent) {
                 minBufferPercent = percent;
@@ -2165,10 +2173,14 @@
     notify->post();
 }
 
-void LiveSession::switchBandwidthIfNeeded(bool bufferHigh, bool bufferLow) {
+/*
+ * returns true if a bandwidth switch is actually needed (and started),
+ * returns false otherwise
+ */
+bool LiveSession::switchBandwidthIfNeeded(bool bufferHigh, bool bufferLow) {
     // no need to check bandwidth if we only have 1 bandwidth settings
     if (mSwitchInProgress || mBandwidthItems.size() < 2) {
-        return;
+        return false;
     }
 
     int32_t bandwidthBps;
@@ -2177,7 +2189,7 @@
         mLastBandwidthBps = bandwidthBps;
     } else {
         ALOGV("no bandwidth estimate.");
-        return;
+        return false;
     }
 
     int32_t curBandwidth = mBandwidthItems.itemAt(mCurBandwidthIndex).mBandwidth;
@@ -2196,16 +2208,16 @@
         // bandwidthIndex is < mCurBandwidthIndex, as getBandwidthIndex() only uses 70%
         // of measured bw. In that case we don't want to do anything, since we have
         // both enough buffer and enough bw.
-        if (bandwidthIndex == mCurBandwidthIndex
-                || (canSwitchUp && bandwidthIndex < mCurBandwidthIndex)
-                || (canSwithDown && bandwidthIndex > mCurBandwidthIndex)) {
-            return;
+        if ((canSwitchUp && bandwidthIndex > mCurBandwidthIndex)
+         || (canSwithDown && bandwidthIndex < mCurBandwidthIndex)) {
+            // if not yet prepared, just restart again with new bw index.
+            // this is faster and playback experience is cleaner.
+            changeConfiguration(
+                    mInPreparationPhase ? 0 : -1ll, bandwidthIndex);
+            return true;
         }
-
-        ALOGI("#### Starting Bandwidth Switch: %zd => %zd",
-                mCurBandwidthIndex, bandwidthIndex);
-        changeConfiguration(-1, bandwidthIndex, false);
     }
+    return false;
 }
 
 void LiveSession::postError(status_t err) {
diff --git a/media/libstagefright/httplive/LiveSession.h b/media/libstagefright/httplive/LiveSession.h
index e4f1b97..9117bb1 100644
--- a/media/libstagefright/httplive/LiveSession.h
+++ b/media/libstagefright/httplive/LiveSession.h
@@ -237,7 +237,7 @@
     sp<PlaylistFetcher> addFetcher(const char *uri);
 
     void onConnect(const sp<AMessage> &msg);
-    status_t onSeek(const sp<AMessage> &msg);
+    void onSeek(const sp<AMessage> &msg);
     void onFinishDisconnect2();
 
     // If given a non-zero block_size (default 0), it is used to cap the number of
@@ -292,7 +292,7 @@
     bool checkSwitchProgress(
             sp<AMessage> &msg, int64_t delayUs, bool *needResumeUntil);
 
-    void switchBandwidthIfNeeded(bool bufferHigh, bool bufferLow);
+    bool switchBandwidthIfNeeded(bool bufferHigh, bool bufferLow);
 
     void schedulePollBuffering();
     void cancelPollBuffering();
diff --git a/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp b/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp
index 0676a33..c7912c0 100644
--- a/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp
+++ b/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp
@@ -355,10 +355,15 @@
     int64_t time2 = -1;
     int64_t durationUs = 0;
 
-    List<sp<ABuffer> >::iterator it = mBuffers.begin();
-    while (it != mBuffers.end()) {
+    List<sp<ABuffer> >::iterator it;
+    for (it = mBuffers.begin(); it != mBuffers.end(); it++) {
         const sp<ABuffer> &buffer = *it;
 
+        int32_t discard;
+        if (buffer->meta()->findInt32("discard", &discard) && discard) {
+            continue;
+        }
+
         int64_t timeUs;
         if (buffer->meta()->findInt64("timeUs", &timeUs)) {
             if (time1 < 0 || timeUs < time1) {
@@ -373,8 +378,6 @@
             durationUs += time2 - time1;
             time1 = time2 = -1;
         }
-
-        ++it;
     }
 
     return durationUs + (time2 - time1);
@@ -393,11 +396,19 @@
         return getBufferedDurationUs_l(&finalResult);
     }
 
-    List<sp<ABuffer> >::iterator it = mBuffers.begin();
-    sp<ABuffer> buffer = *it;
+    sp<ABuffer> buffer;
+    int32_t discard;
+    int64_t startTimeUs = -1ll;
+    List<sp<ABuffer> >::iterator it;
+    for (it = mBuffers.begin(); it != mBuffers.end(); it++) {
+        buffer = *it;
+        if (buffer->meta()->findInt32("discard", &discard) && discard) {
+            continue;
+        }
+        buffer->meta()->findInt64("timeUs", &startTimeUs);
+        break;
+    }
 
-    int64_t startTimeUs;
-    buffer->meta()->findInt64("timeUs", &startTimeUs);
     if (startTimeUs < 0) {
         return 0;
     }
diff --git a/media/libstagefright/mpeg2ts/MPEG2TSExtractor.cpp b/media/libstagefright/mpeg2ts/MPEG2TSExtractor.cpp
index 1f43d6d..33cfd1d 100644
--- a/media/libstagefright/mpeg2ts/MPEG2TSExtractor.cpp
+++ b/media/libstagefright/mpeg2ts/MPEG2TSExtractor.cpp
@@ -85,12 +85,6 @@
         MediaBuffer **out, const ReadOptions *options) {
     *out = NULL;
 
-    int64_t seekTimeUs;
-    ReadOptions::SeekMode seekMode;
-    if (mSeekable && options && options->getSeekTo(&seekTimeUs, &seekMode)) {
-        return ERROR_UNSUPPORTED;
-    }
-
     status_t finalResult;
     while (!mImpl->hasBufferAvailable(&finalResult)) {
         if (finalResult != OK) {
@@ -103,6 +97,17 @@
         }
     }
 
+    int64_t seekTimeUs;
+    ReadOptions::SeekMode seekMode;
+    if (mSeekable && options && options->getSeekTo(&seekTimeUs, &seekMode)) {
+        // A seek was requested, but we don't actually support seeking and so can only "seek" to
+        // the current position
+        int64_t nextBufTimeUs;
+        if (mImpl->nextBufferTime(&nextBufTimeUs) != OK || seekTimeUs != nextBufTimeUs) {
+            return ERROR_UNSUPPORTED;
+        }
+    }
+
     return mImpl->read(out, options);
 }
 
diff --git a/media/libstagefright/omx/SoftOMXPlugin.cpp b/media/libstagefright/omx/SoftOMXPlugin.cpp
index 9b6958a..3ab241a 100644
--- a/media/libstagefright/omx/SoftOMXPlugin.cpp
+++ b/media/libstagefright/omx/SoftOMXPlugin.cpp
@@ -85,7 +85,7 @@
         void *libHandle = dlopen(libName.c_str(), RTLD_NOW);
 
         if (libHandle == NULL) {
-            ALOGE("unable to dlopen %s", libName.c_str());
+            ALOGE("unable to dlopen %s: %s", libName.c_str(), dlerror());
 
             return OMX_ErrorComponentNotFound;
         }
diff --git a/services/audioflinger/Android.mk b/services/audioflinger/Android.mk
index fee2347..f8446ac 100644
--- a/services/audioflinger/Android.mk
+++ b/services/audioflinger/Android.mk
@@ -44,9 +44,9 @@
     SpdifStreamOut.cpp          \
     Effects.cpp                 \
     AudioMixer.cpp.arm          \
-    PatchPanel.cpp
-
-LOCAL_SRC_FILES += StateQueue.cpp
+    BufferProviders.cpp         \
+    PatchPanel.cpp              \
+    StateQueue.cpp
 
 LOCAL_C_INCLUDES := \
     $(TOPDIR)frameworks/av/services/audiopolicy \
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index f3206cb..5002099 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -45,6 +45,8 @@
 #include "AudioFlinger.h"
 #include "ServiceUtilities.h"
 
+#include <media/AudioResamplerPublic.h>
+
 #include <media/EffectsFactoryApi.h>
 #include <audio_effects/effect_visualizer.h>
 #include <audio_effects/effect_ns.h>
@@ -1140,19 +1142,46 @@
     if (ret != NO_ERROR) {
         return 0;
     }
+    if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) {
+        return 0;
+    }
 
     AutoMutex lock(mHardwareLock);
     mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE;
-    audio_config_t config;
-    memset(&config, 0, sizeof(config));
-    config.sample_rate = sampleRate;
-    config.channel_mask = channelMask;
-    config.format = format;
+    audio_config_t config, proposed;
+    memset(&proposed, 0, sizeof(proposed));
+    proposed.sample_rate = sampleRate;
+    proposed.channel_mask = channelMask;
+    proposed.format = format;
 
     audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
-    size_t size = dev->get_input_buffer_size(dev, &config);
+    size_t frames;
+    for (;;) {
+        // Note: config is currently a const parameter for get_input_buffer_size()
+        // but we use a copy from proposed in case config changes from the call.
+        config = proposed;
+        frames = dev->get_input_buffer_size(dev, &config);
+        if (frames != 0) {
+            break; // hal success, config is the result
+        }
+        // change one parameter of the configuration each iteration to a more "common" value
+        // to see if the device will support it.
+        if (proposed.format != AUDIO_FORMAT_PCM_16_BIT) {
+            proposed.format = AUDIO_FORMAT_PCM_16_BIT;
+        } else if (proposed.sample_rate != 44100) { // 44.1 is claimed as must in CDD as well as
+            proposed.sample_rate = 44100;           // legacy AudioRecord.java. TODO: Query hw?
+        } else {
+            ALOGW("getInputBufferSize failed with minimum buffer size sampleRate %u, "
+                    "format %#x, channelMask 0x%X",
+                    sampleRate, format, channelMask);
+            break; // retries failed, break out of loop with frames == 0.
+        }
+    }
     mHardwareStatus = AUDIO_HW_IDLE;
-    return size;
+    if (frames > 0 && config.sample_rate != sampleRate) {
+        frames = destinationFramesPossible(frames, sampleRate, config.sample_rate);
+    }
+    return frames; // may be converted to bytes at the Java level.
 }
 
 uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const
@@ -1419,9 +1448,8 @@
         goto Exit;
     }
 
-    // we don't yet support anything other than 16-bit PCM
-    if (!(audio_is_valid_format(format) &&
-            audio_is_linear_pcm(format) && format == AUDIO_FORMAT_PCM_16_BIT)) {
+    // we don't yet support anything other than linear PCM
+    if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) {
         ALOGE("openRecord() invalid format %#x", format);
         lStatus = BAD_VALUE;
         goto Exit;
@@ -2002,11 +2030,11 @@
             status, address.string());
 
     // If the input could not be opened with the requested parameters and we can handle the
-    // conversion internally, try to open again with the proposed parameters. The AudioFlinger can
-    // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs.
+    // conversion internally, try to open again with the proposed parameters.
     if (status == BAD_VALUE &&
-            config->format == halconfig.format && halconfig.format == AUDIO_FORMAT_PCM_16_BIT &&
-        (halconfig.sample_rate <= 2 * config->sample_rate) &&
+        audio_is_linear_pcm(config->format) &&
+        audio_is_linear_pcm(halconfig.format) &&
+        (halconfig.sample_rate <= AUDIO_RESAMPLER_DOWN_RATIO_MAX * config->sample_rate) &&
         (audio_channel_count_from_in_mask(halconfig.channel_mask) <= FCC_2) &&
         (audio_channel_count_from_in_mask(config->channel_mask) <= FCC_2)) {
         // FIXME describe the change proposed by HAL (save old values so we can log them here)
diff --git a/services/audioflinger/AudioHwDevice.cpp b/services/audioflinger/AudioHwDevice.cpp
index 09d86ea..3191598 100644
--- a/services/audioflinger/AudioHwDevice.cpp
+++ b/services/audioflinger/AudioHwDevice.cpp
@@ -44,7 +44,7 @@
     AudioStreamOut *outputStream = new AudioStreamOut(this, flags);
 
     // Try to open the HAL first using the current format.
-    ALOGV("AudioHwDevice::openOutputStream(), try "
+    ALOGV("openOutputStream(), try "
             " sampleRate %d, Format %#x, "
             "channelMask %#x",
             config->sample_rate,
@@ -59,7 +59,7 @@
         // FIXME Look at any modification to the config.
         // The HAL might modify the config to suggest a wrapped format.
         // Log this so we can see what the HALs are doing.
-        ALOGI("AudioHwDevice::openOutputStream(), HAL returned"
+        ALOGI("openOutputStream(), HAL returned"
             " sampleRate %d, Format %#x, "
             "channelMask %#x, status %d",
             config->sample_rate,
@@ -72,16 +72,19 @@
                 && ((flags & AUDIO_OUTPUT_FLAG_DIRECT) != 0)
                 && ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0);
 
-        // FIXME - Add isEncodingSupported() query to SPDIF wrapper then
-        // call it from here.
         if (wrapperNeeded) {
-            outputStream = new SpdifStreamOut(this, flags);
-            status = outputStream->open(handle, devices, &originalConfig, address);
-            if (status != NO_ERROR) {
-                ALOGE("ERROR - AudioHwDevice::openOutputStream(), SPDIF open returned %d",
-                    status);
-                delete outputStream;
-                outputStream = NULL;
+            if (SPDIFEncoder::isFormatSupported(originalConfig.format)) {
+                outputStream = new SpdifStreamOut(this, flags, originalConfig.format);
+                status = outputStream->open(handle, devices, &originalConfig, address);
+                if (status != NO_ERROR) {
+                    ALOGE("ERROR - openOutputStream(), SPDIF open returned %d",
+                        status);
+                    delete outputStream;
+                    outputStream = NULL;
+                }
+            } else {
+                ALOGE("ERROR - openOutputStream(), SPDIFEncoder does not support format 0x%08x",
+                    originalConfig.format);
             }
         }
     }
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index dddca02..c2c791f 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -38,9 +38,7 @@
 #include <audio_utils/format.h>
 #include <common_time/local_clock.h>
 #include <common_time/cc_helper.h>
-
-#include <media/EffectsFactoryApi.h>
-#include <audio_effects/effect_downmix.h>
+#include <media/AudioResamplerPublic.h>
 
 #include "AudioMixerOps.h"
 #include "AudioMixer.h"
@@ -91,323 +89,6 @@
     return a < b ? a : b;
 }
 
-AudioMixer::CopyBufferProvider::CopyBufferProvider(size_t inputFrameSize,
-        size_t outputFrameSize, size_t bufferFrameCount) :
-        mInputFrameSize(inputFrameSize),
-        mOutputFrameSize(outputFrameSize),
-        mLocalBufferFrameCount(bufferFrameCount),
-        mLocalBufferData(NULL),
-        mConsumed(0)
-{
-    ALOGV("CopyBufferProvider(%p)(%zu, %zu, %zu)", this,
-            inputFrameSize, outputFrameSize, bufferFrameCount);
-    LOG_ALWAYS_FATAL_IF(inputFrameSize < outputFrameSize && bufferFrameCount == 0,
-            "Requires local buffer if inputFrameSize(%zu) < outputFrameSize(%zu)",
-            inputFrameSize, outputFrameSize);
-    if (mLocalBufferFrameCount) {
-        (void)posix_memalign(&mLocalBufferData, 32, mLocalBufferFrameCount * mOutputFrameSize);
-    }
-    mBuffer.frameCount = 0;
-}
-
-AudioMixer::CopyBufferProvider::~CopyBufferProvider()
-{
-    ALOGV("~CopyBufferProvider(%p)", this);
-    if (mBuffer.frameCount != 0) {
-        mTrackBufferProvider->releaseBuffer(&mBuffer);
-    }
-    free(mLocalBufferData);
-}
-
-status_t AudioMixer::CopyBufferProvider::getNextBuffer(AudioBufferProvider::Buffer *pBuffer,
-        int64_t pts)
-{
-    //ALOGV("CopyBufferProvider(%p)::getNextBuffer(%p (%zu), %lld)",
-    //        this, pBuffer, pBuffer->frameCount, pts);
-    if (mLocalBufferFrameCount == 0) {
-        status_t res = mTrackBufferProvider->getNextBuffer(pBuffer, pts);
-        if (res == OK) {
-            copyFrames(pBuffer->raw, pBuffer->raw, pBuffer->frameCount);
-        }
-        return res;
-    }
-    if (mBuffer.frameCount == 0) {
-        mBuffer.frameCount = pBuffer->frameCount;
-        status_t res = mTrackBufferProvider->getNextBuffer(&mBuffer, pts);
-        // At one time an upstream buffer provider had
-        // res == OK and mBuffer.frameCount == 0, doesn't seem to happen now 7/18/2014.
-        //
-        // By API spec, if res != OK, then mBuffer.frameCount == 0.
-        // but there may be improper implementations.
-        ALOG_ASSERT(res == OK || mBuffer.frameCount == 0);
-        if (res != OK || mBuffer.frameCount == 0) { // not needed by API spec, but to be safe.
-            pBuffer->raw = NULL;
-            pBuffer->frameCount = 0;
-            return res;
-        }
-        mConsumed = 0;
-    }
-    ALOG_ASSERT(mConsumed < mBuffer.frameCount);
-    size_t count = min(mLocalBufferFrameCount, mBuffer.frameCount - mConsumed);
-    count = min(count, pBuffer->frameCount);
-    pBuffer->raw = mLocalBufferData;
-    pBuffer->frameCount = count;
-    copyFrames(pBuffer->raw, (uint8_t*)mBuffer.raw + mConsumed * mInputFrameSize,
-            pBuffer->frameCount);
-    return OK;
-}
-
-void AudioMixer::CopyBufferProvider::releaseBuffer(AudioBufferProvider::Buffer *pBuffer)
-{
-    //ALOGV("CopyBufferProvider(%p)::releaseBuffer(%p(%zu))",
-    //        this, pBuffer, pBuffer->frameCount);
-    if (mLocalBufferFrameCount == 0) {
-        mTrackBufferProvider->releaseBuffer(pBuffer);
-        return;
-    }
-    // LOG_ALWAYS_FATAL_IF(pBuffer->frameCount == 0, "Invalid framecount");
-    mConsumed += pBuffer->frameCount; // TODO: update for efficiency to reuse existing content
-    if (mConsumed != 0 && mConsumed >= mBuffer.frameCount) {
-        mTrackBufferProvider->releaseBuffer(&mBuffer);
-        ALOG_ASSERT(mBuffer.frameCount == 0);
-    }
-    pBuffer->raw = NULL;
-    pBuffer->frameCount = 0;
-}
-
-void AudioMixer::CopyBufferProvider::reset()
-{
-    if (mBuffer.frameCount != 0) {
-        mTrackBufferProvider->releaseBuffer(&mBuffer);
-    }
-    mConsumed = 0;
-}
-
-AudioMixer::DownmixerBufferProvider::DownmixerBufferProvider(
-        audio_channel_mask_t inputChannelMask,
-        audio_channel_mask_t outputChannelMask, audio_format_t format,
-        uint32_t sampleRate, int32_t sessionId, size_t bufferFrameCount) :
-        CopyBufferProvider(
-            audio_bytes_per_sample(format) * audio_channel_count_from_out_mask(inputChannelMask),
-            audio_bytes_per_sample(format) * audio_channel_count_from_out_mask(outputChannelMask),
-            bufferFrameCount)  // set bufferFrameCount to 0 to do in-place
-{
-    ALOGV("DownmixerBufferProvider(%p)(%#x, %#x, %#x %u %d)",
-            this, inputChannelMask, outputChannelMask, format,
-            sampleRate, sessionId);
-    if (!sIsMultichannelCapable
-            || EffectCreate(&sDwnmFxDesc.uuid,
-                    sessionId,
-                    SESSION_ID_INVALID_AND_IGNORED,
-                    &mDownmixHandle) != 0) {
-         ALOGE("DownmixerBufferProvider() error creating downmixer effect");
-         mDownmixHandle = NULL;
-         return;
-     }
-     // channel input configuration will be overridden per-track
-     mDownmixConfig.inputCfg.channels = inputChannelMask;   // FIXME: Should be bits
-     mDownmixConfig.outputCfg.channels = outputChannelMask; // FIXME: should be bits
-     mDownmixConfig.inputCfg.format = format;
-     mDownmixConfig.outputCfg.format = format;
-     mDownmixConfig.inputCfg.samplingRate = sampleRate;
-     mDownmixConfig.outputCfg.samplingRate = sampleRate;
-     mDownmixConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ;
-     mDownmixConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE;
-     // input and output buffer provider, and frame count will not be used as the downmix effect
-     // process() function is called directly (see DownmixerBufferProvider::getNextBuffer())
-     mDownmixConfig.inputCfg.mask = EFFECT_CONFIG_SMP_RATE | EFFECT_CONFIG_CHANNELS |
-             EFFECT_CONFIG_FORMAT | EFFECT_CONFIG_ACC_MODE;
-     mDownmixConfig.outputCfg.mask = mDownmixConfig.inputCfg.mask;
-
-     int cmdStatus;
-     uint32_t replySize = sizeof(int);
-
-     // Configure downmixer
-     status_t status = (*mDownmixHandle)->command(mDownmixHandle,
-             EFFECT_CMD_SET_CONFIG /*cmdCode*/, sizeof(effect_config_t) /*cmdSize*/,
-             &mDownmixConfig /*pCmdData*/,
-             &replySize, &cmdStatus /*pReplyData*/);
-     if (status != 0 || cmdStatus != 0) {
-         ALOGE("DownmixerBufferProvider() error %d cmdStatus %d while configuring downmixer",
-                 status, cmdStatus);
-         EffectRelease(mDownmixHandle);
-         mDownmixHandle = NULL;
-         return;
-     }
-
-     // Enable downmixer
-     replySize = sizeof(int);
-     status = (*mDownmixHandle)->command(mDownmixHandle,
-             EFFECT_CMD_ENABLE /*cmdCode*/, 0 /*cmdSize*/, NULL /*pCmdData*/,
-             &replySize, &cmdStatus /*pReplyData*/);
-     if (status != 0 || cmdStatus != 0) {
-         ALOGE("DownmixerBufferProvider() error %d cmdStatus %d while enabling downmixer",
-                 status, cmdStatus);
-         EffectRelease(mDownmixHandle);
-         mDownmixHandle = NULL;
-         return;
-     }
-
-     // Set downmix type
-     // parameter size rounded for padding on 32bit boundary
-     const int psizePadded = ((sizeof(downmix_params_t) - 1)/sizeof(int) + 1) * sizeof(int);
-     const int downmixParamSize =
-             sizeof(effect_param_t) + psizePadded + sizeof(downmix_type_t);
-     effect_param_t * const param = (effect_param_t *) malloc(downmixParamSize);
-     param->psize = sizeof(downmix_params_t);
-     const downmix_params_t downmixParam = DOWNMIX_PARAM_TYPE;
-     memcpy(param->data, &downmixParam, param->psize);
-     const downmix_type_t downmixType = DOWNMIX_TYPE_FOLD;
-     param->vsize = sizeof(downmix_type_t);
-     memcpy(param->data + psizePadded, &downmixType, param->vsize);
-     replySize = sizeof(int);
-     status = (*mDownmixHandle)->command(mDownmixHandle,
-             EFFECT_CMD_SET_PARAM /* cmdCode */, downmixParamSize /* cmdSize */,
-             param /*pCmdData*/, &replySize, &cmdStatus /*pReplyData*/);
-     free(param);
-     if (status != 0 || cmdStatus != 0) {
-         ALOGE("DownmixerBufferProvider() error %d cmdStatus %d while setting downmix type",
-                 status, cmdStatus);
-         EffectRelease(mDownmixHandle);
-         mDownmixHandle = NULL;
-         return;
-     }
-     ALOGV("DownmixerBufferProvider() downmix type set to %d", (int) downmixType);
-}
-
-AudioMixer::DownmixerBufferProvider::~DownmixerBufferProvider()
-{
-    ALOGV("~DownmixerBufferProvider (%p)", this);
-    EffectRelease(mDownmixHandle);
-    mDownmixHandle = NULL;
-}
-
-void AudioMixer::DownmixerBufferProvider::copyFrames(void *dst, const void *src, size_t frames)
-{
-    mDownmixConfig.inputCfg.buffer.frameCount = frames;
-    mDownmixConfig.inputCfg.buffer.raw = const_cast<void *>(src);
-    mDownmixConfig.outputCfg.buffer.frameCount = frames;
-    mDownmixConfig.outputCfg.buffer.raw = dst;
-    // may be in-place if src == dst.
-    status_t res = (*mDownmixHandle)->process(mDownmixHandle,
-            &mDownmixConfig.inputCfg.buffer, &mDownmixConfig.outputCfg.buffer);
-    ALOGE_IF(res != OK, "DownmixBufferProvider error %d", res);
-}
-
-/* call once in a pthread_once handler. */
-/*static*/ status_t AudioMixer::DownmixerBufferProvider::init()
-{
-    // find multichannel downmix effect if we have to play multichannel content
-    uint32_t numEffects = 0;
-    int ret = EffectQueryNumberEffects(&numEffects);
-    if (ret != 0) {
-        ALOGE("AudioMixer() error %d querying number of effects", ret);
-        return NO_INIT;
-    }
-    ALOGV("EffectQueryNumberEffects() numEffects=%d", numEffects);
-
-    for (uint32_t i = 0 ; i < numEffects ; i++) {
-        if (EffectQueryEffect(i, &sDwnmFxDesc) == 0) {
-            ALOGV("effect %d is called %s", i, sDwnmFxDesc.name);
-            if (memcmp(&sDwnmFxDesc.type, EFFECT_UIID_DOWNMIX, sizeof(effect_uuid_t)) == 0) {
-                ALOGI("found effect \"%s\" from %s",
-                        sDwnmFxDesc.name, sDwnmFxDesc.implementor);
-                sIsMultichannelCapable = true;
-                break;
-            }
-        }
-    }
-    ALOGW_IF(!sIsMultichannelCapable, "unable to find downmix effect");
-    return NO_INIT;
-}
-
-/*static*/ bool AudioMixer::DownmixerBufferProvider::sIsMultichannelCapable = false;
-/*static*/ effect_descriptor_t AudioMixer::DownmixerBufferProvider::sDwnmFxDesc;
-
-AudioMixer::RemixBufferProvider::RemixBufferProvider(audio_channel_mask_t inputChannelMask,
-        audio_channel_mask_t outputChannelMask, audio_format_t format,
-        size_t bufferFrameCount) :
-        CopyBufferProvider(
-                audio_bytes_per_sample(format)
-                    * audio_channel_count_from_out_mask(inputChannelMask),
-                audio_bytes_per_sample(format)
-                    * audio_channel_count_from_out_mask(outputChannelMask),
-                bufferFrameCount),
-        mFormat(format),
-        mSampleSize(audio_bytes_per_sample(format)),
-        mInputChannels(audio_channel_count_from_out_mask(inputChannelMask)),
-        mOutputChannels(audio_channel_count_from_out_mask(outputChannelMask))
-{
-    ALOGV("RemixBufferProvider(%p)(%#x, %#x, %#x) %zu %zu",
-            this, format, inputChannelMask, outputChannelMask,
-            mInputChannels, mOutputChannels);
-
-    const audio_channel_representation_t inputRepresentation =
-            audio_channel_mask_get_representation(inputChannelMask);
-    const audio_channel_representation_t outputRepresentation =
-            audio_channel_mask_get_representation(outputChannelMask);
-    const uint32_t inputBits = audio_channel_mask_get_bits(inputChannelMask);
-    const uint32_t outputBits = audio_channel_mask_get_bits(outputChannelMask);
-
-    switch (inputRepresentation) {
-    case AUDIO_CHANNEL_REPRESENTATION_POSITION:
-        switch (outputRepresentation) {
-        case AUDIO_CHANNEL_REPRESENTATION_POSITION:
-            memcpy_by_index_array_initialization(mIdxAry, ARRAY_SIZE(mIdxAry),
-                    outputBits, inputBits);
-            return;
-        case AUDIO_CHANNEL_REPRESENTATION_INDEX:
-            // TODO: output channel index mask not currently allowed
-            // fall through
-        default:
-            break;
-        }
-        break;
-    case AUDIO_CHANNEL_REPRESENTATION_INDEX:
-        switch (outputRepresentation) {
-        case AUDIO_CHANNEL_REPRESENTATION_POSITION:
-            memcpy_by_index_array_initialization_src_index(mIdxAry, ARRAY_SIZE(mIdxAry),
-                    outputBits, inputBits);
-            return;
-        case AUDIO_CHANNEL_REPRESENTATION_INDEX:
-            // TODO: output channel index mask not currently allowed
-            // fall through
-        default:
-            break;
-        }
-        break;
-    default:
-        break;
-    }
-    LOG_ALWAYS_FATAL("invalid channel mask conversion from %#x to %#x",
-            inputChannelMask, outputChannelMask);
-}
-
-void AudioMixer::RemixBufferProvider::copyFrames(void *dst, const void *src, size_t frames)
-{
-    memcpy_by_index_array(dst, mOutputChannels,
-            src, mInputChannels, mIdxAry, mSampleSize, frames);
-}
-
-AudioMixer::ReformatBufferProvider::ReformatBufferProvider(int32_t channels,
-        audio_format_t inputFormat, audio_format_t outputFormat,
-        size_t bufferFrameCount) :
-        CopyBufferProvider(
-            channels * audio_bytes_per_sample(inputFormat),
-            channels * audio_bytes_per_sample(outputFormat),
-            bufferFrameCount),
-        mChannels(channels),
-        mInputFormat(inputFormat),
-        mOutputFormat(outputFormat)
-{
-    ALOGV("ReformatBufferProvider(%p)(%d, %#x, %#x)", this, channels, inputFormat, outputFormat);
-}
-
-void AudioMixer::ReformatBufferProvider::copyFrames(void *dst, const void *src, size_t frames)
-{
-    memcpy_by_audio_format(dst, mOutputFormat, src, mInputFormat, frames * mChannels);
-}
-
 // ----------------------------------------------------------------------------
 
 // Ensure mConfiguredNames bitmask is initialized properly on all architectures.
@@ -442,6 +123,7 @@
         t->resampler = NULL;
         t->downmixerBufferProvider = NULL;
         t->mReformatBufferProvider = NULL;
+        t->mTimestretchBufferProvider = NULL;
         t++;
     }
 
@@ -454,6 +136,7 @@
         delete t->resampler;
         delete t->downmixerBufferProvider;
         delete t->mReformatBufferProvider;
+        delete t->mTimestretchBufferProvider;
         t++;
     }
     delete [] mState.outputTemp;
@@ -532,6 +215,7 @@
         t->mReformatBufferProvider = NULL;
         t->downmixerBufferProvider = NULL;
         t->mPostDownmixReformatBufferProvider = NULL;
+        t->mTimestretchBufferProvider = NULL;
         t->mMixerFormat = AUDIO_FORMAT_PCM_16_BIT;
         t->mFormat = format;
         t->mMixerInFormat = selectMixerInFormat(format);
@@ -539,6 +223,8 @@
         t->mMixerChannelMask = audio_channel_mask_from_representation_and_bits(
                 AUDIO_CHANNEL_REPRESENTATION_POSITION, AUDIO_CHANNEL_OUT_STEREO);
         t->mMixerChannelCount = audio_channel_count_from_out_mask(t->mMixerChannelMask);
+        t->mSpeed = AUDIO_TIMESTRETCH_SPEED_NORMAL;
+        t->mPitch = AUDIO_TIMESTRETCH_PITCH_NORMAL;
         // Check the downmixing (or upmixing) requirements.
         status_t status = t->prepareForDownmix();
         if (status != OK) {
@@ -731,6 +417,10 @@
         mPostDownmixReformatBufferProvider->setBufferProvider(bufferProvider);
         bufferProvider = mPostDownmixReformatBufferProvider;
     }
+    if (mTimestretchBufferProvider) {
+        mTimestretchBufferProvider->setBufferProvider(bufferProvider);
+        bufferProvider = mTimestretchBufferProvider;
+    }
 }
 
 void AudioMixer::deleteTrackName(int name)
@@ -751,7 +441,9 @@
     mState.tracks[name].unprepareForDownmix();
     // delete the reformatter
     mState.tracks[name].unprepareForReformat();
-
+    // delete the timestretch provider
+    delete track.mTimestretchBufferProvider;
+    track.mTimestretchBufferProvider = NULL;
     mTrackNames &= ~(1<<name);
 }
 
@@ -973,6 +665,26 @@
             }
         }
         break;
+        case TIMESTRETCH:
+            switch (param) {
+            case PLAYBACK_RATE: {
+                const float speed = reinterpret_cast<float*>(value)[0];
+                const float pitch = reinterpret_cast<float*>(value)[1];
+                ALOG_ASSERT(AUDIO_TIMESTRETCH_SPEED_MIN <= speed
+                        && speed <= AUDIO_TIMESTRETCH_SPEED_MAX,
+                        "bad speed %f", speed);
+                ALOG_ASSERT(AUDIO_TIMESTRETCH_PITCH_MIN <= pitch
+                        && pitch <= AUDIO_TIMESTRETCH_PITCH_MAX,
+                        "bad pitch %f", pitch);
+                if (track.setPlaybackRate(speed, pitch)) {
+                    ALOGV("setParameter(TIMESTRETCH, PLAYBACK_RATE, %f %f", speed, pitch);
+                    // invalidateState(1 << name);
+                }
+                } break;
+            default:
+                LOG_ALWAYS_FATAL("setParameter timestretch: bad param %d", param);
+            }
+            break;
 
     default:
         LOG_ALWAYS_FATAL("setParameter: bad target %d", target);
@@ -1018,6 +730,28 @@
     return false;
 }
 
+bool AudioMixer::track_t::setPlaybackRate(float speed, float pitch)
+{
+    if (speed == mSpeed && pitch == mPitch) {
+        return false;
+    }
+    mSpeed = speed;
+    mPitch = pitch;
+    if (mTimestretchBufferProvider == NULL) {
+        // TODO: Remove MONO_HACK. Resampler sees #channels after the downmixer
+        // but if none exists, it is the channel count (1 for mono).
+        const int timestretchChannelCount = downmixerBufferProvider != NULL
+                ? mMixerChannelCount : channelCount;
+        mTimestretchBufferProvider = new TimestretchBufferProvider(timestretchChannelCount,
+                mMixerInFormat, sampleRate, speed, pitch);
+        reconfigureBufferProviders();
+    } else {
+        reinterpret_cast<TimestretchBufferProvider*>(mTimestretchBufferProvider)
+                ->setPlaybackRate(speed, pitch);
+    }
+    return true;
+}
+
 /* Checks to see if the volume ramp has completed and clears the increment
  * variables appropriately.
  *
@@ -1096,6 +830,8 @@
         mState.tracks[name].downmixerBufferProvider->reset();
     } else if (mState.tracks[name].mPostDownmixReformatBufferProvider != NULL) {
         mState.tracks[name].mPostDownmixReformatBufferProvider->reset();
+    } else if (mState.tracks[name].mTimestretchBufferProvider != NULL) {
+        mState.tracks[name].mTimestretchBufferProvider->reset();
     }
 
     mState.tracks[name].mInputBufferProvider = bufferProvider;
diff --git a/services/audioflinger/AudioMixer.h b/services/audioflinger/AudioMixer.h
index 381036b..e27a0d1 100644
--- a/services/audioflinger/AudioMixer.h
+++ b/services/audioflinger/AudioMixer.h
@@ -29,6 +29,7 @@
 #include <utils/threads.h>
 
 #include "AudioResampler.h"
+#include "BufferProviders.h"
 
 // FIXME This is actually unity gain, which might not be max in future, expressed in U.12
 #define MAX_GAIN_INT AudioMixer::UNITY_GAIN_INT
@@ -72,6 +73,7 @@
         RESAMPLE        = 0x3001,
         RAMP_VOLUME     = 0x3002, // ramp to new volume
         VOLUME          = 0x3003, // don't ramp
+        TIMESTRETCH     = 0x3004,
 
         // set Parameter names
         // for target TRACK
@@ -99,6 +101,9 @@
         VOLUME0         = 0x4200,
         VOLUME1         = 0x4201,
         AUXLEVEL        = 0x4210,
+        // for target TIMESTRETCH
+        PLAYBACK_RATE   = 0x4300, // Configure timestretch on this track name;
+                                  // parameter 'value' is a pointer to the new playback rate.
     };
 
 
@@ -159,7 +164,6 @@
 
     struct state_t;
     struct track_t;
-    class CopyBufferProvider;
 
     typedef void (*hook_t)(track_t* t, int32_t* output, size_t numOutFrames, int32_t* temp,
                            int32_t* aux);
@@ -214,6 +218,9 @@
 
         /* Buffer providers are constructed to translate the track input data as needed.
          *
+         * TODO: perhaps make a single PlaybackConverterProvider class to move
+         * all pre-mixer track buffer conversions outside the AudioMixer class.
+         *
          * 1) mInputBufferProvider: The AudioTrack buffer provider.
          * 2) mReformatBufferProvider: If not NULL, performs the audio reformat to
          *    match either mMixerInFormat or mDownmixRequiresFormat, if the downmixer
@@ -223,13 +230,14 @@
          *    the number of channels required by the mixer sink.
          * 4) mPostDownmixReformatBufferProvider: If not NULL, performs reformatting from
          *    the downmixer requirements to the mixer engine input requirements.
+         * 5) mTimestretchBufferProvider: Adds timestretching for playback rate
          */
         AudioBufferProvider*     mInputBufferProvider;    // externally provided buffer provider.
-        CopyBufferProvider*      mReformatBufferProvider; // provider wrapper for reformatting.
-        CopyBufferProvider*      downmixerBufferProvider; // wrapper for channel conversion.
-        CopyBufferProvider*      mPostDownmixReformatBufferProvider;
+        PassthruBufferProvider*  mReformatBufferProvider; // provider wrapper for reformatting.
+        PassthruBufferProvider*  downmixerBufferProvider; // wrapper for channel conversion.
+        PassthruBufferProvider*  mPostDownmixReformatBufferProvider;
+        PassthruBufferProvider*  mTimestretchBufferProvider;
 
-        // 16-byte boundary
         int32_t     sessionId;
 
         audio_format_t mMixerFormat;     // output mix format: AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
@@ -251,6 +259,9 @@
         audio_channel_mask_t mMixerChannelMask;
         uint32_t             mMixerChannelCount;
 
+        float          mSpeed;
+        float          mPitch;
+
         bool        needsRamp() { return (volumeInc[0] | volumeInc[1] | auxInc) != 0; }
         bool        setResampler(uint32_t trackSampleRate, uint32_t devSampleRate);
         bool        doesResample() const { return resampler != NULL; }
@@ -263,6 +274,7 @@
         void        unprepareForDownmix();
         status_t    prepareForReformat();
         void        unprepareForReformat();
+        bool        setPlaybackRate(float speed, float pitch);
         void        reconfigureBufferProviders();
     };
 
@@ -282,112 +294,6 @@
         track_t         tracks[MAX_NUM_TRACKS] __attribute__((aligned(32)));
     };
 
-    // Base AudioBufferProvider class used for DownMixerBufferProvider, RemixBufferProvider,
-    // and ReformatBufferProvider.
-    // It handles a private buffer for use in converting format or channel masks from the
-    // input data to a form acceptable by the mixer.
-    // TODO: Make a ResamplerBufferProvider when integers are entirely removed from the
-    // processing pipeline.
-    class CopyBufferProvider : public AudioBufferProvider {
-    public:
-        // Use a private buffer of bufferFrameCount frames (each frame is outputFrameSize bytes).
-        // If bufferFrameCount is 0, no private buffer is created and in-place modification of
-        // the upstream buffer provider's buffers is performed by copyFrames().
-        CopyBufferProvider(size_t inputFrameSize, size_t outputFrameSize,
-                size_t bufferFrameCount);
-        virtual ~CopyBufferProvider();
-
-        // Overrides AudioBufferProvider methods
-        virtual status_t getNextBuffer(Buffer* buffer, int64_t pts);
-        virtual void releaseBuffer(Buffer* buffer);
-
-        // Other public methods
-
-        // call this to release the buffer to the upstream provider.
-        // treat it as an audio discontinuity for future samples.
-        virtual void reset();
-
-        // this function should be supplied by the derived class.  It converts
-        // #frames in the *src pointer to the *dst pointer.  It is public because
-        // some providers will allow this to work on arbitrary buffers outside
-        // of the internal buffers.
-        virtual void copyFrames(void *dst, const void *src, size_t frames) = 0;
-
-        // set the upstream buffer provider. Consider calling "reset" before this function.
-        void setBufferProvider(AudioBufferProvider *p) {
-            mTrackBufferProvider = p;
-        }
-
-    protected:
-        AudioBufferProvider* mTrackBufferProvider;
-        const size_t         mInputFrameSize;
-        const size_t         mOutputFrameSize;
-    private:
-        AudioBufferProvider::Buffer mBuffer;
-        const size_t         mLocalBufferFrameCount;
-        void*                mLocalBufferData;
-        size_t               mConsumed;
-    };
-
-    // DownmixerBufferProvider wraps a track AudioBufferProvider to provide
-    // position dependent downmixing by an Audio Effect.
-    class DownmixerBufferProvider : public CopyBufferProvider {
-    public:
-        DownmixerBufferProvider(audio_channel_mask_t inputChannelMask,
-                audio_channel_mask_t outputChannelMask, audio_format_t format,
-                uint32_t sampleRate, int32_t sessionId, size_t bufferFrameCount);
-        virtual ~DownmixerBufferProvider();
-        virtual void copyFrames(void *dst, const void *src, size_t frames);
-        bool isValid() const { return mDownmixHandle != NULL; }
-
-        static status_t init();
-        static bool isMultichannelCapable() { return sIsMultichannelCapable; }
-
-    protected:
-        effect_handle_t    mDownmixHandle;
-        effect_config_t    mDownmixConfig;
-
-        // effect descriptor for the downmixer used by the mixer
-        static effect_descriptor_t sDwnmFxDesc;
-        // indicates whether a downmix effect has been found and is usable by this mixer
-        static bool                sIsMultichannelCapable;
-        // FIXME: should we allow effects outside of the framework?
-        // We need to here. A special ioId that must be <= -2 so it does not map to a session.
-        static const int32_t SESSION_ID_INVALID_AND_IGNORED = -2;
-    };
-
-    // RemixBufferProvider wraps a track AudioBufferProvider to perform an
-    // upmix or downmix to the proper channel count and mask.
-    class RemixBufferProvider : public CopyBufferProvider {
-    public:
-        RemixBufferProvider(audio_channel_mask_t inputChannelMask,
-                audio_channel_mask_t outputChannelMask, audio_format_t format,
-                size_t bufferFrameCount);
-        virtual void copyFrames(void *dst, const void *src, size_t frames);
-
-    protected:
-        const audio_format_t mFormat;
-        const size_t         mSampleSize;
-        const size_t         mInputChannels;
-        const size_t         mOutputChannels;
-        int8_t               mIdxAry[sizeof(uint32_t)*8]; // 32 bits => channel indices
-    };
-
-    // ReformatBufferProvider wraps a track AudioBufferProvider to convert the input data
-    // to an acceptable mixer input format type.
-    class ReformatBufferProvider : public CopyBufferProvider {
-    public:
-        ReformatBufferProvider(int32_t channels,
-                audio_format_t inputFormat, audio_format_t outputFormat,
-                size_t bufferFrameCount);
-        virtual void copyFrames(void *dst, const void *src, size_t frames);
-
-    protected:
-        const int32_t        mChannels;
-        const audio_format_t mInputFormat;
-        const audio_format_t mOutputFormat;
-    };
-
     // bitmask of allocated track names, where bit 0 corresponds to TRACK0 etc.
     uint32_t        mTrackNames;
 
diff --git a/services/audioflinger/AudioResampler.cpp b/services/audioflinger/AudioResampler.cpp
index 46e3d6c..e49b7b1 100644
--- a/services/audioflinger/AudioResampler.cpp
+++ b/services/audioflinger/AudioResampler.cpp
@@ -41,7 +41,7 @@
     AudioResamplerOrder1(int inChannelCount, int32_t sampleRate) :
         AudioResampler(inChannelCount, sampleRate, LOW_QUALITY), mX0L(0), mX0R(0) {
     }
-    virtual void resample(int32_t* out, size_t outFrameCount,
+    virtual size_t resample(int32_t* out, size_t outFrameCount,
             AudioBufferProvider* provider);
 private:
     // number of bits used in interpolation multiply - 15 bits avoids overflow
@@ -51,9 +51,9 @@
     static const int kPreInterpShift = kNumPhaseBits - kNumInterpBits;
 
     void init() {}
-    void resampleMono16(int32_t* out, size_t outFrameCount,
+    size_t resampleMono16(int32_t* out, size_t outFrameCount,
             AudioBufferProvider* provider);
-    void resampleStereo16(int32_t* out, size_t outFrameCount,
+    size_t resampleStereo16(int32_t* out, size_t outFrameCount,
             AudioBufferProvider* provider);
 #ifdef ASM_ARM_RESAMP1  // asm optimisation for ResamplerOrder1
     void AsmMono16Loop(int16_t *in, int32_t* maxOutPt, int32_t maxInIdx,
@@ -329,7 +329,7 @@
 
 // ----------------------------------------------------------------------------
 
-void AudioResamplerOrder1::resample(int32_t* out, size_t outFrameCount,
+size_t AudioResamplerOrder1::resample(int32_t* out, size_t outFrameCount,
         AudioBufferProvider* provider) {
 
     // should never happen, but we overflow if it does
@@ -338,15 +338,16 @@
     // select the appropriate resampler
     switch (mChannelCount) {
     case 1:
-        resampleMono16(out, outFrameCount, provider);
-        break;
+        return resampleMono16(out, outFrameCount, provider);
     case 2:
-        resampleStereo16(out, outFrameCount, provider);
-        break;
+        return resampleStereo16(out, outFrameCount, provider);
+    default:
+        LOG_ALWAYS_FATAL("invalid channel count: %d", mChannelCount);
+        return 0;
     }
 }
 
-void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount,
+size_t AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount,
         AudioBufferProvider* provider) {
 
     int32_t vl = mVolume[0];
@@ -442,9 +443,10 @@
     // save state
     mInputIndex = inputIndex;
     mPhaseFraction = phaseFraction;
+    return outputIndex / 2 /* channels for stereo */;
 }
 
-void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount,
+size_t AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount,
         AudioBufferProvider* provider) {
 
     int32_t vl = mVolume[0];
@@ -538,6 +540,7 @@
     // save state
     mInputIndex = inputIndex;
     mPhaseFraction = phaseFraction;
+    return outputIndex;
 }
 
 #ifdef ASM_ARM_RESAMP1  // asm optimisation for ResamplerOrder1
diff --git a/services/audioflinger/AudioResampler.h b/services/audioflinger/AudioResampler.h
index 863614a..a8e3e6f 100644
--- a/services/audioflinger/AudioResampler.h
+++ b/services/audioflinger/AudioResampler.h
@@ -67,12 +67,18 @@
     // Resample int16_t samples from provider and accumulate into 'out'.
     // A mono provider delivers a sequence of samples.
     // A stereo provider delivers a sequence of interleaved pairs of samples.
-    // Multi-channel providers are not supported.
+    //
     // In either case, 'out' holds interleaved pairs of fixed-point Q4.27.
     // That is, for a mono provider, there is an implicit up-channeling.
     // Since this method accumulates, the caller is responsible for clearing 'out' initially.
-    // FIXME assumes provider is always successful; it should return the actual frame count.
-    virtual void resample(int32_t* out, size_t outFrameCount,
+    //
+    // For a float resampler, 'out' holds interleaved pairs of float samples.
+    //
+    // Multichannel interleaved frames for n > 2 is supported for quality DYN_LOW_QUALITY,
+    // DYN_MED_QUALITY, and DYN_HIGH_QUALITY.
+    //
+    // Returns the number of frames resampled into the out buffer.
+    virtual size_t resample(int32_t* out, size_t outFrameCount,
             AudioBufferProvider* provider) = 0;
 
     virtual void reset();
diff --git a/services/audioflinger/AudioResamplerCubic.cpp b/services/audioflinger/AudioResamplerCubic.cpp
index d3cbd1c..172c2a5 100644
--- a/services/audioflinger/AudioResamplerCubic.cpp
+++ b/services/audioflinger/AudioResamplerCubic.cpp
@@ -14,7 +14,7 @@
  * limitations under the License.
  */
 
-#define LOG_TAG "AudioSRC"
+#define LOG_TAG "AudioResamplerCubic"
 
 #include <stdint.h>
 #include <string.h>
@@ -32,7 +32,7 @@
     memset(&right, 0, sizeof(state));
 }
 
-void AudioResamplerCubic::resample(int32_t* out, size_t outFrameCount,
+size_t AudioResamplerCubic::resample(int32_t* out, size_t outFrameCount,
         AudioBufferProvider* provider) {
 
     // should never happen, but we overflow if it does
@@ -41,15 +41,16 @@
     // select the appropriate resampler
     switch (mChannelCount) {
     case 1:
-        resampleMono16(out, outFrameCount, provider);
-        break;
+        return resampleMono16(out, outFrameCount, provider);
     case 2:
-        resampleStereo16(out, outFrameCount, provider);
-        break;
+        return resampleStereo16(out, outFrameCount, provider);
+    default:
+        LOG_ALWAYS_FATAL("invalid channel count: %d", mChannelCount);
+        return 0;
     }
 }
 
-void AudioResamplerCubic::resampleStereo16(int32_t* out, size_t outFrameCount,
+size_t AudioResamplerCubic::resampleStereo16(int32_t* out, size_t outFrameCount,
         AudioBufferProvider* provider) {
 
     int32_t vl = mVolume[0];
@@ -67,7 +68,7 @@
         mBuffer.frameCount = inFrameCount;
         provider->getNextBuffer(&mBuffer, mPTS);
         if (mBuffer.raw == NULL) {
-            return;
+            return 0;
         }
         // ALOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount);
     }
@@ -115,9 +116,10 @@
     // ALOGW("Done: index=%d, fraction=%u", inputIndex, phaseFraction);
     mInputIndex = inputIndex;
     mPhaseFraction = phaseFraction;
+    return outputIndex / 2 /* channels for stereo */;
 }
 
-void AudioResamplerCubic::resampleMono16(int32_t* out, size_t outFrameCount,
+size_t AudioResamplerCubic::resampleMono16(int32_t* out, size_t outFrameCount,
         AudioBufferProvider* provider) {
 
     int32_t vl = mVolume[0];
@@ -135,7 +137,7 @@
         mBuffer.frameCount = inFrameCount;
         provider->getNextBuffer(&mBuffer, mPTS);
         if (mBuffer.raw == NULL) {
-            return;
+            return 0;
         }
         // ALOGW("New buffer: offset=%p, frames=%d", mBuffer.raw, mBuffer.frameCount);
     }
@@ -182,6 +184,7 @@
     // ALOGW("Done: index=%d, fraction=%u", inputIndex, phaseFraction);
     mInputIndex = inputIndex;
     mPhaseFraction = phaseFraction;
+    return outputIndex;
 }
 
 // ----------------------------------------------------------------------------
diff --git a/services/audioflinger/AudioResamplerCubic.h b/services/audioflinger/AudioResamplerCubic.h
index 1ddc5f9..4b45b0b 100644
--- a/services/audioflinger/AudioResamplerCubic.h
+++ b/services/audioflinger/AudioResamplerCubic.h
@@ -31,7 +31,7 @@
     AudioResamplerCubic(int inChannelCount, int32_t sampleRate) :
         AudioResampler(inChannelCount, sampleRate, MED_QUALITY) {
     }
-    virtual void resample(int32_t* out, size_t outFrameCount,
+    virtual size_t resample(int32_t* out, size_t outFrameCount,
             AudioBufferProvider* provider);
 private:
     // number of bits used in interpolation multiply - 14 bits avoids overflow
@@ -43,9 +43,9 @@
         int32_t a, b, c, y0, y1, y2, y3;
     } state;
     void init();
-    void resampleMono16(int32_t* out, size_t outFrameCount,
+    size_t resampleMono16(int32_t* out, size_t outFrameCount,
             AudioBufferProvider* provider);
-    void resampleStereo16(int32_t* out, size_t outFrameCount,
+    size_t resampleStereo16(int32_t* out, size_t outFrameCount,
             AudioBufferProvider* provider);
     static inline int32_t interp(state* p, int32_t x) {
         return (((((p->a * x >> 14) + p->b) * x >> 14) + p->c) * x >> 14) + p->y1;
diff --git a/services/audioflinger/AudioResamplerDyn.cpp b/services/audioflinger/AudioResamplerDyn.cpp
index c21d4ca..6481b85 100644
--- a/services/audioflinger/AudioResamplerDyn.cpp
+++ b/services/audioflinger/AudioResamplerDyn.cpp
@@ -477,15 +477,15 @@
 }
 
 template<typename TC, typename TI, typename TO>
-void AudioResamplerDyn<TC, TI, TO>::resample(int32_t* out, size_t outFrameCount,
+size_t AudioResamplerDyn<TC, TI, TO>::resample(int32_t* out, size_t outFrameCount,
             AudioBufferProvider* provider)
 {
-    (this->*mResampleFunc)(reinterpret_cast<TO*>(out), outFrameCount, provider);
+    return (this->*mResampleFunc)(reinterpret_cast<TO*>(out), outFrameCount, provider);
 }
 
 template<typename TC, typename TI, typename TO>
 template<int CHANNELS, bool LOCKED, int STRIDE>
-void AudioResamplerDyn<TC, TI, TO>::resample(TO* out, size_t outFrameCount,
+size_t AudioResamplerDyn<TC, TI, TO>::resample(TO* out, size_t outFrameCount,
         AudioBufferProvider* provider)
 {
     // TODO Mono -> Mono is not supported. OUTPUT_CHANNELS reflects minimum of stereo out.
@@ -610,6 +610,7 @@
     ALOG_ASSERT(mBuffer.frameCount == 0); // there must be no frames in the buffer
     mInBuffer.setImpulse(impulse);
     mPhaseFraction = phaseFraction;
+    return outputIndex / OUTPUT_CHANNELS;
 }
 
 /* instantiate templates used by AudioResampler::create */
diff --git a/services/audioflinger/AudioResamplerDyn.h b/services/audioflinger/AudioResamplerDyn.h
index 238b163..3b1c381 100644
--- a/services/audioflinger/AudioResamplerDyn.h
+++ b/services/audioflinger/AudioResamplerDyn.h
@@ -52,7 +52,7 @@
 
     virtual void setVolume(float left, float right);
 
-    virtual void resample(int32_t* out, size_t outFrameCount,
+    virtual size_t resample(int32_t* out, size_t outFrameCount,
             AudioBufferProvider* provider);
 
 private:
@@ -111,10 +111,10 @@
             int inSampleRate, int outSampleRate, double tbwCheat);
 
     template<int CHANNELS, bool LOCKED, int STRIDE>
-    void resample(TO* out, size_t outFrameCount, AudioBufferProvider* provider);
+    size_t resample(TO* out, size_t outFrameCount, AudioBufferProvider* provider);
 
     // define a pointer to member function type for resample
-    typedef void (AudioResamplerDyn<TC, TI, TO>::*resample_ABP_t)(TO* out,
+    typedef size_t (AudioResamplerDyn<TC, TI, TO>::*resample_ABP_t)(TO* out,
             size_t outFrameCount, AudioBufferProvider* provider);
 
     // data - the contiguous storage and layout of these is important.
diff --git a/services/audioflinger/AudioResamplerSinc.cpp b/services/audioflinger/AudioResamplerSinc.cpp
index ba9a356..41730ee 100644
--- a/services/audioflinger/AudioResamplerSinc.cpp
+++ b/services/audioflinger/AudioResamplerSinc.cpp
@@ -256,7 +256,7 @@
     mVolumeSIMD[1] = u4_28_from_float(clampFloatVol(right));
 }
 
-void AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount,
+size_t AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount,
             AudioBufferProvider* provider)
 {
     // FIXME store current state (up or down sample) and only load the coefs when the state
@@ -272,17 +272,18 @@
     // select the appropriate resampler
     switch (mChannelCount) {
     case 1:
-        resample<1>(out, outFrameCount, provider);
-        break;
+        return resample<1>(out, outFrameCount, provider);
     case 2:
-        resample<2>(out, outFrameCount, provider);
-        break;
+        return resample<2>(out, outFrameCount, provider);
+    default:
+        LOG_ALWAYS_FATAL("invalid channel count: %d", mChannelCount);
+        return 0;
     }
 }
 
 
 template<int CHANNELS>
-void AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount,
+size_t AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount,
         AudioBufferProvider* provider)
 {
     const Constants& c(*mConstants);
@@ -357,6 +358,7 @@
     mImpulse = impulse;
     mInputIndex = inputIndex;
     mPhaseFraction = phaseFraction;
+    return outputIndex / CHANNELS;
 }
 
 template<int CHANNELS>
diff --git a/services/audioflinger/AudioResamplerSinc.h b/services/audioflinger/AudioResamplerSinc.h
index 6d8e85d..0fbeac8 100644
--- a/services/audioflinger/AudioResamplerSinc.h
+++ b/services/audioflinger/AudioResamplerSinc.h
@@ -39,7 +39,7 @@
 
     virtual ~AudioResamplerSinc();
 
-    virtual void resample(int32_t* out, size_t outFrameCount,
+    virtual size_t resample(int32_t* out, size_t outFrameCount,
             AudioBufferProvider* provider);
 private:
     void init();
@@ -47,7 +47,7 @@
     virtual void setVolume(float left, float right);
 
     template<int CHANNELS>
-    void resample(int32_t* out, size_t outFrameCount,
+    size_t resample(int32_t* out, size_t outFrameCount,
             AudioBufferProvider* provider);
 
     template<int CHANNELS>
diff --git a/services/audioflinger/BufferProviders.cpp b/services/audioflinger/BufferProviders.cpp
new file mode 100644
index 0000000..e058e6c
--- /dev/null
+++ b/services/audioflinger/BufferProviders.cpp
@@ -0,0 +1,524 @@
+/*
+ * Copyright (C) 2015 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_TAG "BufferProvider"
+//#define LOG_NDEBUG 0
+
+#include <audio_effects/effect_downmix.h>
+#include <audio_utils/primitives.h>
+#include <audio_utils/format.h>
+#include <media/AudioResamplerPublic.h>
+#include <media/EffectsFactoryApi.h>
+
+#include <utils/Log.h>
+
+#include "Configuration.h"
+#include "BufferProviders.h"
+
+#ifndef ARRAY_SIZE
+#define ARRAY_SIZE(x) (sizeof(x)/sizeof((x)[0]))
+#endif
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+template <typename T>
+static inline T min(const T& a, const T& b)
+{
+    return a < b ? a : b;
+}
+
+CopyBufferProvider::CopyBufferProvider(size_t inputFrameSize,
+        size_t outputFrameSize, size_t bufferFrameCount) :
+        mInputFrameSize(inputFrameSize),
+        mOutputFrameSize(outputFrameSize),
+        mLocalBufferFrameCount(bufferFrameCount),
+        mLocalBufferData(NULL),
+        mConsumed(0)
+{
+    ALOGV("CopyBufferProvider(%p)(%zu, %zu, %zu)", this,
+            inputFrameSize, outputFrameSize, bufferFrameCount);
+    LOG_ALWAYS_FATAL_IF(inputFrameSize < outputFrameSize && bufferFrameCount == 0,
+            "Requires local buffer if inputFrameSize(%zu) < outputFrameSize(%zu)",
+            inputFrameSize, outputFrameSize);
+    if (mLocalBufferFrameCount) {
+        (void)posix_memalign(&mLocalBufferData, 32, mLocalBufferFrameCount * mOutputFrameSize);
+    }
+    mBuffer.frameCount = 0;
+}
+
+CopyBufferProvider::~CopyBufferProvider()
+{
+    ALOGV("~CopyBufferProvider(%p)", this);
+    if (mBuffer.frameCount != 0) {
+        mTrackBufferProvider->releaseBuffer(&mBuffer);
+    }
+    free(mLocalBufferData);
+}
+
+status_t CopyBufferProvider::getNextBuffer(AudioBufferProvider::Buffer *pBuffer,
+        int64_t pts)
+{
+    //ALOGV("CopyBufferProvider(%p)::getNextBuffer(%p (%zu), %lld)",
+    //        this, pBuffer, pBuffer->frameCount, pts);
+    if (mLocalBufferFrameCount == 0) {
+        status_t res = mTrackBufferProvider->getNextBuffer(pBuffer, pts);
+        if (res == OK) {
+            copyFrames(pBuffer->raw, pBuffer->raw, pBuffer->frameCount);
+        }
+        return res;
+    }
+    if (mBuffer.frameCount == 0) {
+        mBuffer.frameCount = pBuffer->frameCount;
+        status_t res = mTrackBufferProvider->getNextBuffer(&mBuffer, pts);
+        // At one time an upstream buffer provider had
+        // res == OK and mBuffer.frameCount == 0, doesn't seem to happen now 7/18/2014.
+        //
+        // By API spec, if res != OK, then mBuffer.frameCount == 0.
+        // but there may be improper implementations.
+        ALOG_ASSERT(res == OK || mBuffer.frameCount == 0);
+        if (res != OK || mBuffer.frameCount == 0) { // not needed by API spec, but to be safe.
+            pBuffer->raw = NULL;
+            pBuffer->frameCount = 0;
+            return res;
+        }
+        mConsumed = 0;
+    }
+    ALOG_ASSERT(mConsumed < mBuffer.frameCount);
+    size_t count = min(mLocalBufferFrameCount, mBuffer.frameCount - mConsumed);
+    count = min(count, pBuffer->frameCount);
+    pBuffer->raw = mLocalBufferData;
+    pBuffer->frameCount = count;
+    copyFrames(pBuffer->raw, (uint8_t*)mBuffer.raw + mConsumed * mInputFrameSize,
+            pBuffer->frameCount);
+    return OK;
+}
+
+void CopyBufferProvider::releaseBuffer(AudioBufferProvider::Buffer *pBuffer)
+{
+    //ALOGV("CopyBufferProvider(%p)::releaseBuffer(%p(%zu))",
+    //        this, pBuffer, pBuffer->frameCount);
+    if (mLocalBufferFrameCount == 0) {
+        mTrackBufferProvider->releaseBuffer(pBuffer);
+        return;
+    }
+    // LOG_ALWAYS_FATAL_IF(pBuffer->frameCount == 0, "Invalid framecount");
+    mConsumed += pBuffer->frameCount; // TODO: update for efficiency to reuse existing content
+    if (mConsumed != 0 && mConsumed >= mBuffer.frameCount) {
+        mTrackBufferProvider->releaseBuffer(&mBuffer);
+        ALOG_ASSERT(mBuffer.frameCount == 0);
+    }
+    pBuffer->raw = NULL;
+    pBuffer->frameCount = 0;
+}
+
+void CopyBufferProvider::reset()
+{
+    if (mBuffer.frameCount != 0) {
+        mTrackBufferProvider->releaseBuffer(&mBuffer);
+    }
+    mConsumed = 0;
+}
+
+DownmixerBufferProvider::DownmixerBufferProvider(
+        audio_channel_mask_t inputChannelMask,
+        audio_channel_mask_t outputChannelMask, audio_format_t format,
+        uint32_t sampleRate, int32_t sessionId, size_t bufferFrameCount) :
+        CopyBufferProvider(
+            audio_bytes_per_sample(format) * audio_channel_count_from_out_mask(inputChannelMask),
+            audio_bytes_per_sample(format) * audio_channel_count_from_out_mask(outputChannelMask),
+            bufferFrameCount)  // set bufferFrameCount to 0 to do in-place
+{
+    ALOGV("DownmixerBufferProvider(%p)(%#x, %#x, %#x %u %d)",
+            this, inputChannelMask, outputChannelMask, format,
+            sampleRate, sessionId);
+    if (!sIsMultichannelCapable
+            || EffectCreate(&sDwnmFxDesc.uuid,
+                    sessionId,
+                    SESSION_ID_INVALID_AND_IGNORED,
+                    &mDownmixHandle) != 0) {
+         ALOGE("DownmixerBufferProvider() error creating downmixer effect");
+         mDownmixHandle = NULL;
+         return;
+     }
+     // channel input configuration will be overridden per-track
+     mDownmixConfig.inputCfg.channels = inputChannelMask;   // FIXME: Should be bits
+     mDownmixConfig.outputCfg.channels = outputChannelMask; // FIXME: should be bits
+     mDownmixConfig.inputCfg.format = format;
+     mDownmixConfig.outputCfg.format = format;
+     mDownmixConfig.inputCfg.samplingRate = sampleRate;
+     mDownmixConfig.outputCfg.samplingRate = sampleRate;
+     mDownmixConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ;
+     mDownmixConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE;
+     // input and output buffer provider, and frame count will not be used as the downmix effect
+     // process() function is called directly (see DownmixerBufferProvider::getNextBuffer())
+     mDownmixConfig.inputCfg.mask = EFFECT_CONFIG_SMP_RATE | EFFECT_CONFIG_CHANNELS |
+             EFFECT_CONFIG_FORMAT | EFFECT_CONFIG_ACC_MODE;
+     mDownmixConfig.outputCfg.mask = mDownmixConfig.inputCfg.mask;
+
+     int cmdStatus;
+     uint32_t replySize = sizeof(int);
+
+     // Configure downmixer
+     status_t status = (*mDownmixHandle)->command(mDownmixHandle,
+             EFFECT_CMD_SET_CONFIG /*cmdCode*/, sizeof(effect_config_t) /*cmdSize*/,
+             &mDownmixConfig /*pCmdData*/,
+             &replySize, &cmdStatus /*pReplyData*/);
+     if (status != 0 || cmdStatus != 0) {
+         ALOGE("DownmixerBufferProvider() error %d cmdStatus %d while configuring downmixer",
+                 status, cmdStatus);
+         EffectRelease(mDownmixHandle);
+         mDownmixHandle = NULL;
+         return;
+     }
+
+     // Enable downmixer
+     replySize = sizeof(int);
+     status = (*mDownmixHandle)->command(mDownmixHandle,
+             EFFECT_CMD_ENABLE /*cmdCode*/, 0 /*cmdSize*/, NULL /*pCmdData*/,
+             &replySize, &cmdStatus /*pReplyData*/);
+     if (status != 0 || cmdStatus != 0) {
+         ALOGE("DownmixerBufferProvider() error %d cmdStatus %d while enabling downmixer",
+                 status, cmdStatus);
+         EffectRelease(mDownmixHandle);
+         mDownmixHandle = NULL;
+         return;
+     }
+
+     // Set downmix type
+     // parameter size rounded for padding on 32bit boundary
+     const int psizePadded = ((sizeof(downmix_params_t) - 1)/sizeof(int) + 1) * sizeof(int);
+     const int downmixParamSize =
+             sizeof(effect_param_t) + psizePadded + sizeof(downmix_type_t);
+     effect_param_t * const param = (effect_param_t *) malloc(downmixParamSize);
+     param->psize = sizeof(downmix_params_t);
+     const downmix_params_t downmixParam = DOWNMIX_PARAM_TYPE;
+     memcpy(param->data, &downmixParam, param->psize);
+     const downmix_type_t downmixType = DOWNMIX_TYPE_FOLD;
+     param->vsize = sizeof(downmix_type_t);
+     memcpy(param->data + psizePadded, &downmixType, param->vsize);
+     replySize = sizeof(int);
+     status = (*mDownmixHandle)->command(mDownmixHandle,
+             EFFECT_CMD_SET_PARAM /* cmdCode */, downmixParamSize /* cmdSize */,
+             param /*pCmdData*/, &replySize, &cmdStatus /*pReplyData*/);
+     free(param);
+     if (status != 0 || cmdStatus != 0) {
+         ALOGE("DownmixerBufferProvider() error %d cmdStatus %d while setting downmix type",
+                 status, cmdStatus);
+         EffectRelease(mDownmixHandle);
+         mDownmixHandle = NULL;
+         return;
+     }
+     ALOGV("DownmixerBufferProvider() downmix type set to %d", (int) downmixType);
+}
+
+DownmixerBufferProvider::~DownmixerBufferProvider()
+{
+    ALOGV("~DownmixerBufferProvider (%p)", this);
+    EffectRelease(mDownmixHandle);
+    mDownmixHandle = NULL;
+}
+
+void DownmixerBufferProvider::copyFrames(void *dst, const void *src, size_t frames)
+{
+    mDownmixConfig.inputCfg.buffer.frameCount = frames;
+    mDownmixConfig.inputCfg.buffer.raw = const_cast<void *>(src);
+    mDownmixConfig.outputCfg.buffer.frameCount = frames;
+    mDownmixConfig.outputCfg.buffer.raw = dst;
+    // may be in-place if src == dst.
+    status_t res = (*mDownmixHandle)->process(mDownmixHandle,
+            &mDownmixConfig.inputCfg.buffer, &mDownmixConfig.outputCfg.buffer);
+    ALOGE_IF(res != OK, "DownmixBufferProvider error %d", res);
+}
+
+/* call once in a pthread_once handler. */
+/*static*/ status_t DownmixerBufferProvider::init()
+{
+    // find multichannel downmix effect if we have to play multichannel content
+    uint32_t numEffects = 0;
+    int ret = EffectQueryNumberEffects(&numEffects);
+    if (ret != 0) {
+        ALOGE("AudioMixer() error %d querying number of effects", ret);
+        return NO_INIT;
+    }
+    ALOGV("EffectQueryNumberEffects() numEffects=%d", numEffects);
+
+    for (uint32_t i = 0 ; i < numEffects ; i++) {
+        if (EffectQueryEffect(i, &sDwnmFxDesc) == 0) {
+            ALOGV("effect %d is called %s", i, sDwnmFxDesc.name);
+            if (memcmp(&sDwnmFxDesc.type, EFFECT_UIID_DOWNMIX, sizeof(effect_uuid_t)) == 0) {
+                ALOGI("found effect \"%s\" from %s",
+                        sDwnmFxDesc.name, sDwnmFxDesc.implementor);
+                sIsMultichannelCapable = true;
+                break;
+            }
+        }
+    }
+    ALOGW_IF(!sIsMultichannelCapable, "unable to find downmix effect");
+    return NO_INIT;
+}
+
+/*static*/ bool DownmixerBufferProvider::sIsMultichannelCapable = false;
+/*static*/ effect_descriptor_t DownmixerBufferProvider::sDwnmFxDesc;
+
+RemixBufferProvider::RemixBufferProvider(audio_channel_mask_t inputChannelMask,
+        audio_channel_mask_t outputChannelMask, audio_format_t format,
+        size_t bufferFrameCount) :
+        CopyBufferProvider(
+                audio_bytes_per_sample(format)
+                    * audio_channel_count_from_out_mask(inputChannelMask),
+                audio_bytes_per_sample(format)
+                    * audio_channel_count_from_out_mask(outputChannelMask),
+                bufferFrameCount),
+        mFormat(format),
+        mSampleSize(audio_bytes_per_sample(format)),
+        mInputChannels(audio_channel_count_from_out_mask(inputChannelMask)),
+        mOutputChannels(audio_channel_count_from_out_mask(outputChannelMask))
+{
+    ALOGV("RemixBufferProvider(%p)(%#x, %#x, %#x) %zu %zu",
+            this, format, inputChannelMask, outputChannelMask,
+            mInputChannels, mOutputChannels);
+
+    const audio_channel_representation_t inputRepresentation =
+            audio_channel_mask_get_representation(inputChannelMask);
+    const audio_channel_representation_t outputRepresentation =
+            audio_channel_mask_get_representation(outputChannelMask);
+    const uint32_t inputBits = audio_channel_mask_get_bits(inputChannelMask);
+    const uint32_t outputBits = audio_channel_mask_get_bits(outputChannelMask);
+
+    switch (inputRepresentation) {
+    case AUDIO_CHANNEL_REPRESENTATION_POSITION:
+        switch (outputRepresentation) {
+        case AUDIO_CHANNEL_REPRESENTATION_POSITION:
+            memcpy_by_index_array_initialization(mIdxAry, ARRAY_SIZE(mIdxAry),
+                    outputBits, inputBits);
+            return;
+        case AUDIO_CHANNEL_REPRESENTATION_INDEX:
+            // TODO: output channel index mask not currently allowed
+            // fall through
+        default:
+            break;
+        }
+        break;
+    case AUDIO_CHANNEL_REPRESENTATION_INDEX:
+        switch (outputRepresentation) {
+        case AUDIO_CHANNEL_REPRESENTATION_POSITION:
+            memcpy_by_index_array_initialization_src_index(mIdxAry, ARRAY_SIZE(mIdxAry),
+                    outputBits, inputBits);
+            return;
+        case AUDIO_CHANNEL_REPRESENTATION_INDEX:
+            // TODO: output channel index mask not currently allowed
+            // fall through
+        default:
+            break;
+        }
+        break;
+    default:
+        break;
+    }
+    LOG_ALWAYS_FATAL("invalid channel mask conversion from %#x to %#x",
+            inputChannelMask, outputChannelMask);
+}
+
+void RemixBufferProvider::copyFrames(void *dst, const void *src, size_t frames)
+{
+    memcpy_by_index_array(dst, mOutputChannels,
+            src, mInputChannels, mIdxAry, mSampleSize, frames);
+}
+
+ReformatBufferProvider::ReformatBufferProvider(int32_t channelCount,
+        audio_format_t inputFormat, audio_format_t outputFormat,
+        size_t bufferFrameCount) :
+        CopyBufferProvider(
+                channelCount * audio_bytes_per_sample(inputFormat),
+                channelCount * audio_bytes_per_sample(outputFormat),
+                bufferFrameCount),
+        mChannelCount(channelCount),
+        mInputFormat(inputFormat),
+        mOutputFormat(outputFormat)
+{
+    ALOGV("ReformatBufferProvider(%p)(%u, %#x, %#x)",
+            this, channelCount, inputFormat, outputFormat);
+}
+
+void ReformatBufferProvider::copyFrames(void *dst, const void *src, size_t frames)
+{
+    memcpy_by_audio_format(dst, mOutputFormat, src, mInputFormat, frames * mChannelCount);
+}
+
+TimestretchBufferProvider::TimestretchBufferProvider(int32_t channelCount,
+        audio_format_t format, uint32_t sampleRate, float speed, float pitch) :
+        mChannelCount(channelCount),
+        mFormat(format),
+        mSampleRate(sampleRate),
+        mFrameSize(channelCount * audio_bytes_per_sample(format)),
+        mSpeed(speed),
+        mPitch(pitch),
+        mLocalBufferFrameCount(0),
+        mLocalBufferData(NULL),
+        mRemaining(0)
+{
+    ALOGV("TimestretchBufferProvider(%p)(%u, %#x, %u %f %f)",
+            this, channelCount, format, sampleRate, speed, pitch);
+    mBuffer.frameCount = 0;
+}
+
+TimestretchBufferProvider::~TimestretchBufferProvider()
+{
+    ALOGV("~TimestretchBufferProvider(%p)", this);
+    if (mBuffer.frameCount != 0) {
+        mTrackBufferProvider->releaseBuffer(&mBuffer);
+    }
+    free(mLocalBufferData);
+}
+
+status_t TimestretchBufferProvider::getNextBuffer(
+        AudioBufferProvider::Buffer *pBuffer, int64_t pts)
+{
+    ALOGV("TimestretchBufferProvider(%p)::getNextBuffer(%p (%zu), %lld)",
+            this, pBuffer, pBuffer->frameCount, pts);
+
+    // BYPASS
+    //return mTrackBufferProvider->getNextBuffer(pBuffer, pts);
+
+    // check if previously processed data is sufficient.
+    if (pBuffer->frameCount <= mRemaining) {
+        ALOGV("previous sufficient");
+        pBuffer->raw = mLocalBufferData;
+        return OK;
+    }
+
+    // do we need to resize our buffer?
+    if (pBuffer->frameCount > mLocalBufferFrameCount) {
+        void *newmem;
+        if (posix_memalign(&newmem, 32, pBuffer->frameCount * mFrameSize) == OK) {
+            if (mRemaining != 0) {
+                memcpy(newmem, mLocalBufferData, mRemaining * mFrameSize);
+            }
+            free(mLocalBufferData);
+            mLocalBufferData = newmem;
+            mLocalBufferFrameCount = pBuffer->frameCount;
+        }
+    }
+
+    // need to fetch more data
+    const size_t outputDesired = pBuffer->frameCount - mRemaining;
+    mBuffer.frameCount = mSpeed == AUDIO_TIMESTRETCH_SPEED_NORMAL
+            ? outputDesired : outputDesired * mSpeed + 1;
+
+    status_t res = mTrackBufferProvider->getNextBuffer(&mBuffer, pts);
+
+    ALOG_ASSERT(res == OK || mBuffer.frameCount == 0);
+    if (res != OK || mBuffer.frameCount == 0) { // not needed by API spec, but to be safe.
+        ALOGD("buffer error");
+        if (mRemaining == 0) {
+            pBuffer->raw = NULL;
+            pBuffer->frameCount = 0;
+            return res;
+        } else { // return partial count
+            pBuffer->raw = mLocalBufferData;
+            pBuffer->frameCount = mRemaining;
+            return OK;
+        }
+    }
+
+    // time-stretch the data
+    size_t dstAvailable = min(mLocalBufferFrameCount - mRemaining, outputDesired);
+    size_t srcAvailable = mBuffer.frameCount;
+    processFrames((uint8_t*)mLocalBufferData + mRemaining * mFrameSize, &dstAvailable,
+            mBuffer.raw, &srcAvailable);
+
+    // release all data consumed
+    mBuffer.frameCount = srcAvailable;
+    mTrackBufferProvider->releaseBuffer(&mBuffer);
+
+    // update buffer vars with the actual data processed and return with buffer
+    mRemaining += dstAvailable;
+
+    pBuffer->raw = mLocalBufferData;
+    pBuffer->frameCount = mRemaining;
+
+    return OK;
+}
+
+void TimestretchBufferProvider::releaseBuffer(AudioBufferProvider::Buffer *pBuffer)
+{
+    ALOGV("TimestretchBufferProvider(%p)::releaseBuffer(%p (%zu))",
+       this, pBuffer, pBuffer->frameCount);
+
+    // BYPASS
+    //return mTrackBufferProvider->releaseBuffer(pBuffer);
+
+    // LOG_ALWAYS_FATAL_IF(pBuffer->frameCount == 0, "Invalid framecount");
+    if (pBuffer->frameCount < mRemaining) {
+        memcpy(mLocalBufferData,
+                (uint8_t*)mLocalBufferData + pBuffer->frameCount * mFrameSize,
+                (mRemaining - pBuffer->frameCount) * mFrameSize);
+        mRemaining -= pBuffer->frameCount;
+    } else if (pBuffer->frameCount == mRemaining) {
+        mRemaining = 0;
+    } else {
+        LOG_ALWAYS_FATAL("Releasing more frames(%zu) than available(%zu)",
+                pBuffer->frameCount, mRemaining);
+    }
+
+    pBuffer->raw = NULL;
+    pBuffer->frameCount = 0;
+}
+
+void TimestretchBufferProvider::reset()
+{
+    mRemaining = 0;
+}
+
+status_t TimestretchBufferProvider::setPlaybackRate(float speed, float pitch)
+{
+    mSpeed = speed;
+    mPitch = pitch;
+    return OK;
+}
+
+void TimestretchBufferProvider::processFrames(void *dstBuffer, size_t *dstFrames,
+        const void *srcBuffer, size_t *srcFrames)
+{
+    ALOGV("processFrames(%zu %zu)  remaining(%zu)", *dstFrames, *srcFrames, mRemaining);
+    // Note dstFrames is the required number of frames.
+
+    // Ensure consumption from src is as expected.
+    const size_t targetSrc = *dstFrames * mSpeed;
+    if (*srcFrames < targetSrc) { // limit dst frames to that possible
+        *dstFrames = *srcFrames / mSpeed;
+    } else if (*srcFrames > targetSrc + 1) {
+        *srcFrames = targetSrc + 1;
+    }
+
+    // Do the time stretch by memory copy without any local buffer.
+    if (*dstFrames <= *srcFrames) {
+        size_t copySize = mFrameSize * *dstFrames;
+        memcpy(dstBuffer, srcBuffer, copySize);
+    } else {
+        // cyclically repeat the source.
+        for (size_t count = 0; count < *dstFrames; count += *srcFrames) {
+            size_t remaining = min(*srcFrames, *dstFrames - count);
+            memcpy((uint8_t*)dstBuffer + mFrameSize * count,
+                    srcBuffer, mFrameSize * *srcFrames);
+        }
+    }
+}
+
+// ----------------------------------------------------------------------------
+} // namespace android
diff --git a/services/audioflinger/BufferProviders.h b/services/audioflinger/BufferProviders.h
new file mode 100644
index 0000000..2b6ea47
--- /dev/null
+++ b/services/audioflinger/BufferProviders.h
@@ -0,0 +1,191 @@
+/*
+ * Copyright (C) 2015 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 ANDROID_BUFFER_PROVIDERS_H
+#define ANDROID_BUFFER_PROVIDERS_H
+
+#include <stdint.h>
+#include <sys/types.h>
+
+#include <hardware/audio_effect.h>
+#include <media/AudioBufferProvider.h>
+#include <system/audio.h>
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+class PassthruBufferProvider : public AudioBufferProvider {
+public:
+    PassthruBufferProvider() : mTrackBufferProvider(NULL) { }
+
+    virtual ~PassthruBufferProvider() { }
+
+    // call this to release the buffer to the upstream provider.
+    // treat it as an audio discontinuity for future samples.
+    virtual void reset() { }
+
+    // set the upstream buffer provider. Consider calling "reset" before this function.
+    virtual void setBufferProvider(AudioBufferProvider *p) {
+        mTrackBufferProvider = p;
+    }
+
+protected:
+    AudioBufferProvider *mTrackBufferProvider;
+};
+
+// Base AudioBufferProvider class used for DownMixerBufferProvider, RemixBufferProvider,
+// and ReformatBufferProvider.
+// It handles a private buffer for use in converting format or channel masks from the
+// input data to a form acceptable by the mixer.
+// TODO: Make a ResamplerBufferProvider when integers are entirely removed from the
+// processing pipeline.
+class CopyBufferProvider : public PassthruBufferProvider {
+public:
+    // Use a private buffer of bufferFrameCount frames (each frame is outputFrameSize bytes).
+    // If bufferFrameCount is 0, no private buffer is created and in-place modification of
+    // the upstream buffer provider's buffers is performed by copyFrames().
+    CopyBufferProvider(size_t inputFrameSize, size_t outputFrameSize,
+            size_t bufferFrameCount);
+    virtual ~CopyBufferProvider();
+
+    // Overrides AudioBufferProvider methods
+    virtual status_t getNextBuffer(Buffer *buffer, int64_t pts);
+    virtual void releaseBuffer(Buffer *buffer);
+
+    // Overrides PassthruBufferProvider
+    virtual void reset();
+
+    // this function should be supplied by the derived class.  It converts
+    // #frames in the *src pointer to the *dst pointer.  It is public because
+    // some providers will allow this to work on arbitrary buffers outside
+    // of the internal buffers.
+    virtual void copyFrames(void *dst, const void *src, size_t frames) = 0;
+
+protected:
+    const size_t         mInputFrameSize;
+    const size_t         mOutputFrameSize;
+private:
+    AudioBufferProvider::Buffer mBuffer;
+    const size_t         mLocalBufferFrameCount;
+    void                *mLocalBufferData;
+    size_t               mConsumed;
+};
+
+// DownmixerBufferProvider derives from CopyBufferProvider to provide
+// position dependent downmixing by an Audio Effect.
+class DownmixerBufferProvider : public CopyBufferProvider {
+public:
+    DownmixerBufferProvider(audio_channel_mask_t inputChannelMask,
+            audio_channel_mask_t outputChannelMask, audio_format_t format,
+            uint32_t sampleRate, int32_t sessionId, size_t bufferFrameCount);
+    virtual ~DownmixerBufferProvider();
+    //Overrides
+    virtual void copyFrames(void *dst, const void *src, size_t frames);
+
+    bool isValid() const { return mDownmixHandle != NULL; }
+    static status_t init();
+    static bool isMultichannelCapable() { return sIsMultichannelCapable; }
+
+protected:
+    effect_handle_t    mDownmixHandle;
+    effect_config_t    mDownmixConfig;
+
+    // effect descriptor for the downmixer used by the mixer
+    static effect_descriptor_t sDwnmFxDesc;
+    // indicates whether a downmix effect has been found and is usable by this mixer
+    static bool                sIsMultichannelCapable;
+    // FIXME: should we allow effects outside of the framework?
+    // We need to here. A special ioId that must be <= -2 so it does not map to a session.
+    static const int32_t SESSION_ID_INVALID_AND_IGNORED = -2;
+};
+
+// RemixBufferProvider derives from CopyBufferProvider to perform an
+// upmix or downmix to the proper channel count and mask.
+class RemixBufferProvider : public CopyBufferProvider {
+public:
+    RemixBufferProvider(audio_channel_mask_t inputChannelMask,
+            audio_channel_mask_t outputChannelMask, audio_format_t format,
+            size_t bufferFrameCount);
+    //Overrides
+    virtual void copyFrames(void *dst, const void *src, size_t frames);
+
+protected:
+    const audio_format_t mFormat;
+    const size_t         mSampleSize;
+    const size_t         mInputChannels;
+    const size_t         mOutputChannels;
+    int8_t               mIdxAry[sizeof(uint32_t) * 8]; // 32 bits => channel indices
+};
+
+// ReformatBufferProvider derives from CopyBufferProvider to convert the input data
+// to an acceptable mixer input format type.
+class ReformatBufferProvider : public CopyBufferProvider {
+public:
+    ReformatBufferProvider(int32_t channelCount,
+            audio_format_t inputFormat, audio_format_t outputFormat,
+            size_t bufferFrameCount);
+    virtual void copyFrames(void *dst, const void *src, size_t frames);
+
+protected:
+    const uint32_t       mChannelCount;
+    const audio_format_t mInputFormat;
+    const audio_format_t mOutputFormat;
+};
+
+// TimestretchBufferProvider derives from PassthruBufferProvider for time stretching
+class TimestretchBufferProvider : public PassthruBufferProvider {
+public:
+    TimestretchBufferProvider(int32_t channelCount,
+            audio_format_t format, uint32_t sampleRate, float speed, float pitch);
+    virtual ~TimestretchBufferProvider();
+
+    // Overrides AudioBufferProvider methods
+    virtual status_t getNextBuffer(Buffer* buffer, int64_t pts);
+    virtual void releaseBuffer(Buffer* buffer);
+
+    // Overrides PassthruBufferProvider
+    virtual void reset();
+
+    virtual status_t setPlaybackRate(float speed, float pitch);
+
+    // processes frames
+    // dstBuffer is where to place the data
+    // dstFrames [in/out] is the desired frames (return with actual placed in buffer)
+    // srcBuffer is the source data
+    // srcFrames [in/out] is the available source frames (return with consumed)
+    virtual void processFrames(void *dstBuffer, size_t *dstFrames,
+            const void *srcBuffer, size_t *srcFrames);
+
+protected:
+    const uint32_t       mChannelCount;
+    const audio_format_t mFormat;
+    const uint32_t       mSampleRate; // const for now (TODO change this)
+    const size_t         mFrameSize;
+    float                mSpeed;
+    float                mPitch;
+
+private:
+    AudioBufferProvider::Buffer mBuffer;
+    size_t               mLocalBufferFrameCount;
+    void                *mLocalBufferData;
+    size_t               mRemaining;
+};
+
+// ----------------------------------------------------------------------------
+} // namespace android
+
+#endif // ANDROID_BUFFER_PROVIDERS_H
diff --git a/services/audioflinger/RecordTracks.h b/services/audioflinger/RecordTracks.h
index 204a9d6..25d6d95 100644
--- a/services/audioflinger/RecordTracks.h
+++ b/services/audioflinger/RecordTracks.h
@@ -34,6 +34,7 @@
                                 IAudioFlinger::track_flags_t flags,
                                 track_type type);
     virtual             ~RecordTrack();
+    virtual status_t    initCheck() const;
 
     virtual status_t    start(AudioSystem::sync_event_t event, int triggerSession);
     virtual void        stop();
@@ -66,21 +67,6 @@
 
     bool                mOverflow;  // overflow on most recent attempt to fill client buffer
 
-           // updated by RecordThread::readInputParameters_l()
-            AudioResampler                      *mResampler;
-
-            // interleaved stereo pairs of fixed-point Q4.27
-            int32_t                             *mRsmpOutBuffer;
-            // current allocated frame count for the above, which may be larger than needed
-            size_t                              mRsmpOutFrameCount;
-
-            size_t                              mRsmpInUnrel;   // unreleased frames remaining from
-                                                                // most recent getNextBuffer
-                                                                // for debug only
-
-            // rolling counter that is never cleared
-            int32_t                             mRsmpInFront;   // next available frame
-
             AudioBufferProvider::Buffer mSink;  // references client's buffer sink in shared memory
 
             // sync event triggering actual audio capture. Frames read before this event will
@@ -93,7 +79,10 @@
             ssize_t                             mFramesToDrop;
 
             // used by resampler to find source frames
-            ResamplerBufferProvider *mResamplerBufferProvider;
+            ResamplerBufferProvider            *mResamplerBufferProvider;
+
+            // used by the record thread to convert frames to proper destination format
+            RecordBufferConverter              *mRecordBufferConverter;
 };
 
 // playback track, used by PatchPanel
diff --git a/services/audioflinger/SpdifStreamOut.cpp b/services/audioflinger/SpdifStreamOut.cpp
index d23588e..45b541a 100644
--- a/services/audioflinger/SpdifStreamOut.cpp
+++ b/services/audioflinger/SpdifStreamOut.cpp
@@ -32,10 +32,12 @@
  * If the AudioFlinger is processing encoded data and the HAL expects
  * PCM then we need to wrap the data in an SPDIF wrapper.
  */
-SpdifStreamOut::SpdifStreamOut(AudioHwDevice *dev, audio_output_flags_t flags)
+SpdifStreamOut::SpdifStreamOut(AudioHwDevice *dev,
+            audio_output_flags_t flags,
+            audio_format_t format)
         : AudioStreamOut(dev,flags)
         , mRateMultiplier(1)
-        , mSpdifEncoder(this)
+        , mSpdifEncoder(this, format)
         , mRenderPositionHal(0)
         , mPreviousHalPosition32(0)
 {
@@ -49,15 +51,15 @@
 {
     struct audio_config customConfig = *config;
 
-    customConfig.format = AUDIO_FORMAT_PCM_16_BIT;
-    customConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
-
     // Some data bursts run at a higher sample rate.
+    // TODO Move this into the audio_utils as a static method.
     switch(config->format) {
         case AUDIO_FORMAT_E_AC3:
             mRateMultiplier = 4;
             break;
         case AUDIO_FORMAT_AC3:
+        case AUDIO_FORMAT_DTS:
+        case AUDIO_FORMAT_DTS_HD:
             mRateMultiplier = 1;
             break;
         default:
@@ -67,6 +69,9 @@
     }
     customConfig.sample_rate = config->sample_rate * mRateMultiplier;
 
+    customConfig.format = AUDIO_FORMAT_PCM_16_BIT;
+    customConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+
     // Always print this because otherwise it could be very confusing if the
     // HAL and AudioFlinger are using different formats.
     // Print before open() because HAL may modify customConfig.
diff --git a/services/audioflinger/SpdifStreamOut.h b/services/audioflinger/SpdifStreamOut.h
index cb82ac7..d81c064 100644
--- a/services/audioflinger/SpdifStreamOut.h
+++ b/services/audioflinger/SpdifStreamOut.h
@@ -38,7 +38,8 @@
 class SpdifStreamOut : public AudioStreamOut {
 public:
 
-    SpdifStreamOut(AudioHwDevice *dev, audio_output_flags_t flags);
+    SpdifStreamOut(AudioHwDevice *dev, audio_output_flags_t flags,
+            audio_format_t format);
 
     virtual ~SpdifStreamOut() { }
 
@@ -77,8 +78,9 @@
     class MySPDIFEncoder : public SPDIFEncoder
     {
     public:
-        MySPDIFEncoder(SpdifStreamOut *spdifStreamOut)
-          : mSpdifStreamOut(spdifStreamOut)
+        MySPDIFEncoder(SpdifStreamOut *spdifStreamOut, audio_format_t format)
+          :  SPDIFEncoder(format)
+          , mSpdifStreamOut(spdifStreamOut)
         {
         }
 
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 4efb3d7..b30fd20 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -86,7 +86,13 @@
 #define ALOGVV(a...) do { } while(0)
 #endif
 
+// TODO: Move these macro/inlines to a header file.
 #define max(a, b) ((a) > (b) ? (a) : (b))
+template <typename T>
+static inline T min(const T& a, const T& b)
+{
+    return a < b ? a : b;
+}
 
 namespace android {
 
@@ -1602,13 +1608,19 @@
     // If you change this calculation, also review the start threshold which is related.
     if (!(*flags & IAudioFlinger::TRACK_FAST)
             && audio_is_linear_pcm(format) && sharedBuffer == 0) {
+        // this must match AudioTrack.cpp calculateMinFrameCount().
+        // TODO: Move to a common library
         uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream);
         uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate);
         if (minBufCount < 2) {
             minBufCount = 2;
         }
+        // For normal mixing tracks, if speed is > 1.0f (normal), AudioTrack
+        // or the client should compute and pass in a larger buffer request.
         size_t minFrameCount =
-                minBufCount * sourceFramesNeeded(sampleRate, mNormalFrameCount, mSampleRate);
+                minBufCount * sourceFramesNeededWithTimestretch(
+                        sampleRate, mNormalFrameCount,
+                        mSampleRate, AUDIO_TIMESTRETCH_SPEED_NORMAL /*speed*/);
         if (frameCount < minFrameCount) { // including frameCount == 0
             frameCount = minFrameCount;
         }
@@ -3586,21 +3598,17 @@
         // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed
         // during last round
         size_t desiredFrames;
-        uint32_t sr = track->sampleRate();
-        if (sr == mSampleRate) {
-            desiredFrames = mNormalFrameCount;
-        } else {
-            desiredFrames = sourceFramesNeeded(sr, mNormalFrameCount, mSampleRate);
-            // add frames already consumed but not yet released by the resampler
-            // because mAudioTrackServerProxy->framesReady() will include these frames
-            desiredFrames += mAudioMixer->getUnreleasedFrames(track->name());
-#if 0
-            // the minimum track buffer size is normally twice the number of frames necessary
-            // to fill one buffer and the resampler should not leave more than one buffer worth
-            // of unreleased frames after each pass, but just in case...
-            ALOG_ASSERT(desiredFrames <= cblk->frameCount_);
-#endif
-        }
+        const uint32_t sampleRate = track->mAudioTrackServerProxy->getSampleRate();
+        float speed, pitch;
+        track->mAudioTrackServerProxy->getPlaybackRate(&speed, &pitch);
+
+        desiredFrames = sourceFramesNeededWithTimestretch(
+                sampleRate, mNormalFrameCount, mSampleRate, speed);
+        // TODO: ONLY USED FOR LEGACY RESAMPLERS, remove when they are removed.
+        // add frames already consumed but not yet released by the resampler
+        // because mAudioTrackServerProxy->framesReady() will include these frames
+        desiredFrames += mAudioMixer->getUnreleasedFrames(track->name());
+
         uint32_t minFrames = 1;
         if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() &&
                 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) {
@@ -3763,6 +3771,17 @@
                 AudioMixer::RESAMPLE,
                 AudioMixer::SAMPLE_RATE,
                 (void *)(uintptr_t)reqSampleRate);
+
+            // set the playback rate as an float array {speed, pitch}
+            float playbackRate[2];
+            track->mAudioTrackServerProxy->getPlaybackRate(
+                    &playbackRate[0] /*speed*/, &playbackRate[1] /*pitch*/);
+            mAudioMixer->setParameter(
+                name,
+                AudioMixer::TIMESTRETCH,
+                AudioMixer::PLAYBACK_RATE,
+                playbackRate);
+
             /*
              * Select the appropriate output buffer for the track.
              *
@@ -5290,7 +5309,6 @@
     // FIXME mNormalSource
 }
 
-
 AudioFlinger::RecordThread::~RecordThread()
 {
     if (mFastCapture != 0) {
@@ -5594,6 +5612,9 @@
                 continue;
             }
 
+            // TODO: This code probably should be moved to RecordTrack.
+            // TODO: Update the activeTrack buffer converter in case of reconfigure.
+
             enum {
                 OVERRUN_UNKNOWN,
                 OVERRUN_TRUE,
@@ -5608,131 +5629,28 @@
                 size_t framesOut = activeTrack->mSink.frameCount;
                 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0));
 
-                int32_t front = activeTrack->mRsmpInFront;
-                ssize_t filled = rear - front;
+                // check available frames and handle overrun conditions
+                // if the record track isn't draining fast enough.
+                bool hasOverrun;
                 size_t framesIn;
-
-                if (filled < 0) {
-                    // should not happen, but treat like a massive overrun and re-sync
-                    framesIn = 0;
-                    activeTrack->mRsmpInFront = rear;
-                    overrun = OVERRUN_TRUE;
-                } else if ((size_t) filled <= mRsmpInFrames) {
-                    framesIn = (size_t) filled;
-                } else {
-                    // client is not keeping up with server, but give it latest data
-                    framesIn = mRsmpInFrames;
-                    activeTrack->mRsmpInFront = front = rear - framesIn;
+                activeTrack->mResamplerBufferProvider->sync(&framesIn, &hasOverrun);
+                if (hasOverrun) {
                     overrun = OVERRUN_TRUE;
                 }
-
                 if (framesOut == 0 || framesIn == 0) {
                     break;
                 }
 
-                if (activeTrack->mResampler == NULL) {
-                    // no resampling
-                    if (framesIn > framesOut) {
-                        framesIn = framesOut;
-                    } else {
-                        framesOut = framesIn;
-                    }
-                    int8_t *dst = activeTrack->mSink.i8;
-                    while (framesIn > 0) {
-                        front &= mRsmpInFramesP2 - 1;
-                        size_t part1 = mRsmpInFramesP2 - front;
-                        if (part1 > framesIn) {
-                            part1 = framesIn;
-                        }
-                        int8_t *src = (int8_t *)mRsmpInBuffer + (front * mFrameSize);
-                        if (mChannelCount == activeTrack->mChannelCount) {
-                            memcpy(dst, src, part1 * mFrameSize);
-                        } else if (mChannelCount == 1) {
-                            upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (const int16_t *)src,
-                                    part1);
-                        } else {
-                            downmix_to_mono_i16_from_stereo_i16((int16_t *)dst,
-                                    (const int16_t *)src, part1);
-                        }
-                        dst += part1 * activeTrack->mFrameSize;
-                        front += part1;
-                        framesIn -= part1;
-                    }
-                    activeTrack->mRsmpInFront += framesOut;
-
-                } else {
-                    // resampling
-                    // FIXME framesInNeeded should really be part of resampler API, and should
-                    //       depend on the SRC ratio
-                    //       to keep mRsmpInBuffer full so resampler always has sufficient input
-                    size_t framesInNeeded;
-                    // FIXME only re-calculate when it changes, and optimize for common ratios
-                    // Do not precompute in/out because floating point is not associative
-                    // e.g. a*b/c != a*(b/c).
-                    const double in(mSampleRate);
-                    const double out(activeTrack->mSampleRate);
-                    framesInNeeded = ceil(framesOut * in / out) + 1;
-                    ALOGV("need %u frames in to produce %u out given in/out ratio of %.4g",
-                                framesInNeeded, framesOut, in / out);
-                    // Although we theoretically have framesIn in circular buffer, some of those are
-                    // unreleased frames, and thus must be discounted for purpose of budgeting.
-                    size_t unreleased = activeTrack->mRsmpInUnrel;
-                    framesIn = framesIn > unreleased ? framesIn - unreleased : 0;
-                    if (framesIn < framesInNeeded) {
-                        ALOGV("not enough to resample: have %u frames in but need %u in to "
-                                "produce %u out given in/out ratio of %.4g",
-                                framesIn, framesInNeeded, framesOut, in / out);
-                        size_t newFramesOut = framesIn > 0 ? floor((framesIn - 1) * out / in) : 0;
-                        LOG_ALWAYS_FATAL_IF(newFramesOut >= framesOut);
-                        if (newFramesOut == 0) {
-                            break;
-                        }
-                        framesInNeeded = ceil(newFramesOut * in / out) + 1;
-                        ALOGV("now need %u frames in to produce %u out given out/in ratio of %.4g",
-                                framesInNeeded, newFramesOut, out / in);
-                        LOG_ALWAYS_FATAL_IF(framesIn < framesInNeeded);
-                        ALOGV("success 2: have %u frames in and need %u in to produce %u out "
-                              "given in/out ratio of %.4g",
-                              framesIn, framesInNeeded, newFramesOut, in / out);
-                        framesOut = newFramesOut;
-                    } else {
-                        ALOGV("success 1: have %u in and need %u in to produce %u out "
-                            "given in/out ratio of %.4g",
-                            framesIn, framesInNeeded, framesOut, in / out);
-                    }
-
-                    // reallocate mRsmpOutBuffer as needed; we will grow but never shrink
-                    if (activeTrack->mRsmpOutFrameCount < framesOut) {
-                        // FIXME why does each track need it's own mRsmpOutBuffer? can't they share?
-                        delete[] activeTrack->mRsmpOutBuffer;
-                        // resampler always outputs stereo
-                        activeTrack->mRsmpOutBuffer = new int32_t[framesOut * FCC_2];
-                        activeTrack->mRsmpOutFrameCount = framesOut;
-                    }
-
-                    // resampler accumulates, but we only have one source track
-                    memset(activeTrack->mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t));
-                    activeTrack->mResampler->resample(activeTrack->mRsmpOutBuffer, framesOut,
-                            // FIXME how about having activeTrack implement this interface itself?
-                            activeTrack->mResamplerBufferProvider
-                            /*this*/ /* AudioBufferProvider* */);
-                    // ditherAndClamp() works as long as all buffers returned by
-                    // activeTrack->getNextBuffer() are 32 bit aligned which should be always true.
-                    if (activeTrack->mChannelCount == 1) {
-                        // temporarily type pun mRsmpOutBuffer from Q4.27 to int16_t
-                        ditherAndClamp(activeTrack->mRsmpOutBuffer, activeTrack->mRsmpOutBuffer,
-                                framesOut);
-                        // the resampler always outputs stereo samples:
-                        // do post stereo to mono conversion
-                        downmix_to_mono_i16_from_stereo_i16(activeTrack->mSink.i16,
-                                (const int16_t *)activeTrack->mRsmpOutBuffer, framesOut);
-                    } else {
-                        ditherAndClamp((int32_t *)activeTrack->mSink.raw,
-                                activeTrack->mRsmpOutBuffer, framesOut);
-                    }
-                    // now done with mRsmpOutBuffer
-
-                }
+                // Don't allow framesOut to be larger than what is possible with resampling
+                // from framesIn.
+                // This isn't strictly necessary but helps limit buffer resizing in
+                // RecordBufferConverter.  TODO: remove when no longer needed.
+                framesOut = min(framesOut,
+                        destinationFramesPossible(
+                                framesIn, mSampleRate, activeTrack->mSampleRate));
+                // process frames from the RecordThread buffer provider to the RecordTrack buffer
+                framesOut = activeTrack->mRecordBufferConverter->convert(
+                        activeTrack->mSink.raw, activeTrack->mResamplerBufferProvider, framesOut);
 
                 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) {
                     overrun = OVERRUN_FALSE;
@@ -6041,12 +5959,9 @@
         // was initialized to some value closer to the thread's mRsmpInFront, then the track could
         // see previously buffered data before it called start(), but with greater risk of overrun.
 
-        recordTrack->mRsmpInFront = mRsmpInRear;
-        recordTrack->mRsmpInUnrel = 0;
-        // FIXME why reset?
-        if (recordTrack->mResampler != NULL) {
-            recordTrack->mResampler->reset();
-        }
+        recordTrack->mResamplerBufferProvider->reset();
+        // clear any converter state as new data will be discontinuous
+        recordTrack->mRecordBufferConverter->reset();
         recordTrack->mState = TrackBase::STARTING_2;
         // signal thread to start
         mWaitWorkCV.broadcast();
@@ -6222,12 +6137,52 @@
     write(fd, result.string(), result.size());
 }
 
+
+void AudioFlinger::RecordThread::ResamplerBufferProvider::reset()
+{
+    sp<ThreadBase> threadBase = mRecordTrack->mThread.promote();
+    RecordThread *recordThread = (RecordThread *) threadBase.get();
+    mRsmpInFront = recordThread->mRsmpInRear;
+    mRsmpInUnrel = 0;
+}
+
+void AudioFlinger::RecordThread::ResamplerBufferProvider::sync(
+        size_t *framesAvailable, bool *hasOverrun)
+{
+    sp<ThreadBase> threadBase = mRecordTrack->mThread.promote();
+    RecordThread *recordThread = (RecordThread *) threadBase.get();
+    const int32_t rear = recordThread->mRsmpInRear;
+    const int32_t front = mRsmpInFront;
+    const ssize_t filled = rear - front;
+
+    size_t framesIn;
+    bool overrun = false;
+    if (filled < 0) {
+        // should not happen, but treat like a massive overrun and re-sync
+        framesIn = 0;
+        mRsmpInFront = rear;
+        overrun = true;
+    } else if ((size_t) filled <= recordThread->mRsmpInFrames) {
+        framesIn = (size_t) filled;
+    } else {
+        // client is not keeping up with server, but give it latest data
+        framesIn = recordThread->mRsmpInFrames;
+        mRsmpInFront = /* front = */ rear - framesIn;
+        overrun = true;
+    }
+    if (framesAvailable != NULL) {
+        *framesAvailable = framesIn;
+    }
+    if (hasOverrun != NULL) {
+        *hasOverrun = overrun;
+    }
+}
+
 // AudioBufferProvider interface
 status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer(
         AudioBufferProvider::Buffer* buffer, int64_t pts __unused)
 {
-    RecordTrack *activeTrack = mRecordTrack;
-    sp<ThreadBase> threadBase = activeTrack->mThread.promote();
+    sp<ThreadBase> threadBase = mRecordTrack->mThread.promote();
     if (threadBase == 0) {
         buffer->frameCount = 0;
         buffer->raw = NULL;
@@ -6235,7 +6190,7 @@
     }
     RecordThread *recordThread = (RecordThread *) threadBase.get();
     int32_t rear = recordThread->mRsmpInRear;
-    int32_t front = activeTrack->mRsmpInFront;
+    int32_t front = mRsmpInFront;
     ssize_t filled = rear - front;
     // FIXME should not be P2 (don't want to increase latency)
     // FIXME if client not keeping up, discard
@@ -6252,17 +6207,16 @@
         part1 = ask;
     }
     if (part1 == 0) {
-        // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty
-        LOG_ALWAYS_FATAL("RecordThread::getNextBuffer() starved");
+        // out of data is fine since the resampler will return a short-count.
         buffer->raw = NULL;
         buffer->frameCount = 0;
-        activeTrack->mRsmpInUnrel = 0;
+        mRsmpInUnrel = 0;
         return NOT_ENOUGH_DATA;
     }
 
     buffer->raw = recordThread->mRsmpInBuffer + front * recordThread->mChannelCount;
     buffer->frameCount = part1;
-    activeTrack->mRsmpInUnrel = part1;
+    mRsmpInUnrel = part1;
     return NO_ERROR;
 }
 
@@ -6270,18 +6224,197 @@
 void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer(
         AudioBufferProvider::Buffer* buffer)
 {
-    RecordTrack *activeTrack = mRecordTrack;
     size_t stepCount = buffer->frameCount;
     if (stepCount == 0) {
         return;
     }
-    ALOG_ASSERT(stepCount <= activeTrack->mRsmpInUnrel);
-    activeTrack->mRsmpInUnrel -= stepCount;
-    activeTrack->mRsmpInFront += stepCount;
+    ALOG_ASSERT(stepCount <= mRsmpInUnrel);
+    mRsmpInUnrel -= stepCount;
+    mRsmpInFront += stepCount;
     buffer->raw = NULL;
     buffer->frameCount = 0;
 }
 
+AudioFlinger::RecordThread::RecordBufferConverter::RecordBufferConverter(
+        audio_channel_mask_t srcChannelMask, audio_format_t srcFormat,
+        uint32_t srcSampleRate,
+        audio_channel_mask_t dstChannelMask, audio_format_t dstFormat,
+        uint32_t dstSampleRate) :
+            mSrcChannelMask(AUDIO_CHANNEL_INVALID), // updateParameters will set following vars
+            // mSrcFormat
+            // mSrcSampleRate
+            // mDstChannelMask
+            // mDstFormat
+            // mDstSampleRate
+            // mSrcChannelCount
+            // mDstChannelCount
+            // mDstFrameSize
+            mBuf(NULL), mBufFrames(0), mBufFrameSize(0),
+            mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpOutFrameCount(0)
+{
+    (void)updateParameters(srcChannelMask, srcFormat, srcSampleRate,
+            dstChannelMask, dstFormat, dstSampleRate);
+}
+
+AudioFlinger::RecordThread::RecordBufferConverter::~RecordBufferConverter() {
+    free(mBuf);
+    delete mResampler;
+    free(mRsmpOutBuffer);
+}
+
+size_t AudioFlinger::RecordThread::RecordBufferConverter::convert(void *dst,
+        AudioBufferProvider *provider, size_t frames)
+{
+    if (mSrcSampleRate == mDstSampleRate) {
+        ALOGVV("NO RESAMPLING sampleRate:%u mSrcFormat:%#x mDstFormat:%#x",
+                mSrcSampleRate, mSrcFormat, mDstFormat);
+
+        AudioBufferProvider::Buffer buffer;
+        for (size_t i = frames; i > 0; ) {
+            buffer.frameCount = i;
+            status_t status = provider->getNextBuffer(&buffer, 0);
+            if (status != OK || buffer.frameCount == 0) {
+                frames -= i; // cannot fill request.
+                break;
+            }
+            // convert to destination buffer
+            convert(dst, buffer.raw, buffer.frameCount);
+
+            dst = (int8_t*)dst + buffer.frameCount * mDstFrameSize;
+            i -= buffer.frameCount;
+            provider->releaseBuffer(&buffer);
+        }
+    } else {
+         ALOGVV("RESAMPLING mSrcSampleRate:%u mDstSampleRate:%u mSrcFormat:%#x mDstFormat:%#x",
+                 mSrcSampleRate, mDstSampleRate, mSrcFormat, mDstFormat);
+
+        // reallocate mRsmpOutBuffer as needed; we will grow but never shrink
+        if (mRsmpOutFrameCount < frames) {
+            // FIXME why does each track need it's own mRsmpOutBuffer? can't they share?
+            free(mRsmpOutBuffer);
+            // resampler always outputs stereo (FOR NOW)
+            (void)posix_memalign(&mRsmpOutBuffer, 32, frames * FCC_2 * sizeof(int32_t) /*Q4.27*/);
+            mRsmpOutFrameCount = frames;
+        }
+        // resampler accumulates, but we only have one source track
+        memset(mRsmpOutBuffer, 0, frames * FCC_2 * sizeof(int32_t));
+        frames = mResampler->resample((int32_t*)mRsmpOutBuffer, frames, provider);
+
+        // convert to destination buffer
+        convert(dst, mRsmpOutBuffer, frames);
+    }
+    return frames;
+}
+
+status_t AudioFlinger::RecordThread::RecordBufferConverter::updateParameters(
+        audio_channel_mask_t srcChannelMask, audio_format_t srcFormat,
+        uint32_t srcSampleRate,
+        audio_channel_mask_t dstChannelMask, audio_format_t dstFormat,
+        uint32_t dstSampleRate)
+{
+    // quick evaluation if there is any change.
+    if (mSrcFormat == srcFormat
+            && mSrcChannelMask == srcChannelMask
+            && mSrcSampleRate == srcSampleRate
+            && mDstFormat == dstFormat
+            && mDstChannelMask == dstChannelMask
+            && mDstSampleRate == dstSampleRate) {
+        return NO_ERROR;
+    }
+
+    const bool valid =
+            audio_is_input_channel(srcChannelMask)
+            && audio_is_input_channel(dstChannelMask)
+            && audio_is_valid_format(srcFormat) && audio_is_linear_pcm(srcFormat)
+            && audio_is_valid_format(dstFormat) && audio_is_linear_pcm(dstFormat)
+            && (srcSampleRate <= dstSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX)
+            ; // no upsampling checks for now
+    if (!valid) {
+        return BAD_VALUE;
+    }
+
+    mSrcFormat = srcFormat;
+    mSrcChannelMask = srcChannelMask;
+    mSrcSampleRate = srcSampleRate;
+    mDstFormat = dstFormat;
+    mDstChannelMask = dstChannelMask;
+    mDstSampleRate = dstSampleRate;
+
+    // compute derived parameters
+    mSrcChannelCount = audio_channel_count_from_in_mask(srcChannelMask);
+    mDstChannelCount = audio_channel_count_from_in_mask(dstChannelMask);
+    mDstFrameSize = mDstChannelCount * audio_bytes_per_sample(mDstFormat);
+
+    // do we need a format buffer?
+    if (mSrcFormat != mDstFormat && mDstChannelCount != mSrcChannelCount) {
+        mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(mSrcFormat);
+    } else {
+        mBufFrameSize = 0;
+    }
+    mBufFrames = 0; // force the buffer to be resized.
+
+    // do we need to resample?
+    if (mSrcSampleRate != mDstSampleRate) {
+        if (mResampler != NULL) {
+            delete mResampler;
+        }
+        mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_16_BIT,
+                mSrcChannelCount, mDstSampleRate); // may seem confusing...
+        mResampler->setSampleRate(mSrcSampleRate);
+        mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT);
+    }
+    return NO_ERROR;
+}
+
+void AudioFlinger::RecordThread::RecordBufferConverter::convert(
+        void *dst, /*const*/ void *src, size_t frames)
+{
+    // check if a memcpy will do
+    if (mResampler == NULL
+            && mSrcChannelCount == mDstChannelCount
+            && mSrcFormat == mDstFormat) {
+        memcpy(dst, src,
+                frames * mDstChannelCount * audio_bytes_per_sample(mDstFormat));
+        return;
+    }
+    // reallocate buffer if needed
+    if (mBufFrameSize != 0 && mBufFrames < frames) {
+        free(mBuf);
+        mBufFrames = frames;
+        (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize);
+    }
+    // do processing
+    if (mResampler != NULL) {
+        // src channel count is always >= 2.
+        void *dstBuf = mBuf != NULL ? mBuf : dst;
+        // ditherAndClamp() works as long as all buffers returned by
+        // activeTrack->getNextBuffer() are 32 bit aligned which should be always true.
+        if (mDstChannelCount == 1) {
+            // the resampler always outputs stereo samples.
+            // FIXME: this rewrites back into src
+            ditherAndClamp((int32_t *)src, (const int32_t *)src, frames);
+            downmix_to_mono_i16_from_stereo_i16((int16_t *)dstBuf,
+                    (const int16_t *)src, frames);
+        } else {
+            ditherAndClamp((int32_t *)dstBuf, (const int32_t *)src, frames);
+        }
+    } else if (mSrcChannelCount != mDstChannelCount) {
+        void *dstBuf = mBuf != NULL ? mBuf : dst;
+        if (mSrcChannelCount == 1) {
+            upmix_to_stereo_i16_from_mono_i16((int16_t *)dstBuf, (const int16_t *)src,
+                    frames);
+        } else {
+            downmix_to_mono_i16_from_stereo_i16((int16_t *)dstBuf,
+                    (const int16_t *)src, frames);
+        }
+    }
+    if (mSrcFormat != mDstFormat) {
+        void *srcBuf = mBuf != NULL ? mBuf : src;
+        memcpy_by_audio_format(dst, mDstFormat, srcBuf, mSrcFormat,
+                frames * mDstChannelCount);
+    }
+}
+
 bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair,
                                                         status_t& status)
 {
@@ -6303,7 +6436,7 @@
         reconfig = true;
     }
     if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
-        if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) {
+        if (!audio_is_linear_pcm((audio_format_t) value)) {
             status = BAD_VALUE;
         } else {
             reqFormat = (audio_format_t) value;
@@ -6377,10 +6510,10 @@
         }
         if (reconfig) {
             if (status == BAD_VALUE &&
-                reqFormat == mInput->stream->common.get_format(&mInput->stream->common) &&
-                reqFormat == AUDIO_FORMAT_PCM_16_BIT &&
+                audio_is_linear_pcm(mInput->stream->common.get_format(&mInput->stream->common)) &&
+                audio_is_linear_pcm(reqFormat) &&
                 (mInput->stream->common.get_sample_rate(&mInput->stream->common)
-                        <= (2 * samplingRate)) &&
+                        <= (AUDIO_RESAMPLER_DOWN_RATIO_MAX * samplingRate)) &&
                 audio_channel_count_from_in_mask(
                         mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 &&
                 (channelMask == AUDIO_CHANNEL_IN_MONO ||
@@ -6451,6 +6584,8 @@
     // The value is somewhat arbitrary, and could probably be even larger.
     // A larger value should allow more old data to be read after a track calls start(),
     // without increasing latency.
+    //
+    // Note this is independent of the maximum downsampling ratio permitted for capture.
     mRsmpInFrames = mFrameCount * 7;
     mRsmpInFramesP2 = roundup(mRsmpInFrames);
     delete[] mRsmpInBuffer;
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index d600ea9..27bc56b 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -1036,17 +1036,127 @@
 public:
 
     class RecordTrack;
+
+    /* The ResamplerBufferProvider is used to retrieve recorded input data from the
+     * RecordThread.  It maintains local state on the relative position of the read
+     * position of the RecordTrack compared with the RecordThread.
+     */
     class ResamplerBufferProvider : public AudioBufferProvider
-                        // derives from AudioBufferProvider interface for use by resampler
     {
     public:
-        ResamplerBufferProvider(RecordTrack* recordTrack) : mRecordTrack(recordTrack) { }
+        ResamplerBufferProvider(RecordTrack* recordTrack) :
+            mRecordTrack(recordTrack),
+            mRsmpInUnrel(0), mRsmpInFront(0) { }
         virtual ~ResamplerBufferProvider() { }
+
+        // called to set the ResamplerBufferProvider to head of the RecordThread data buffer,
+        // skipping any previous data read from the hal.
+        virtual void reset();
+
+        /* Synchronizes RecordTrack position with the RecordThread.
+         * Calculates available frames and handle overruns if the RecordThread
+         * has advanced faster than the ResamplerBufferProvider has retrieved data.
+         * TODO: why not do this for every getNextBuffer?
+         *
+         * Parameters
+         * framesAvailable:  pointer to optional output size_t to store record track
+         *                   frames available.
+         *      hasOverrun:  pointer to optional boolean, returns true if track has overrun.
+         */
+
+        virtual void sync(size_t *framesAvailable = NULL, bool *hasOverrun = NULL);
+
         // AudioBufferProvider interface
         virtual status_t    getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts);
         virtual void        releaseBuffer(AudioBufferProvider::Buffer* buffer);
     private:
         RecordTrack * const mRecordTrack;
+        size_t              mRsmpInUnrel;   // unreleased frames remaining from
+                                            // most recent getNextBuffer
+                                            // for debug only
+        int32_t             mRsmpInFront;   // next available frame
+                                            // rolling counter that is never cleared
+    };
+
+    /* The RecordBufferConverter is used for format, channel, and sample rate
+     * conversion for a RecordTrack.
+     *
+     * TODO: Self contained, so move to a separate file later.
+     *
+     * RecordBufferConverter uses the convert() method rather than exposing a
+     * buffer provider interface; this is to save a memory copy.
+     */
+    class RecordBufferConverter
+    {
+    public:
+        RecordBufferConverter(
+                audio_channel_mask_t srcChannelMask, audio_format_t srcFormat,
+                uint32_t srcSampleRate,
+                audio_channel_mask_t dstChannelMask, audio_format_t dstFormat,
+                uint32_t dstSampleRate);
+
+        ~RecordBufferConverter();
+
+        /* Converts input data from an AudioBufferProvider by format, channelMask,
+         * and sampleRate to a destination buffer.
+         *
+         * Parameters
+         *      dst:  buffer to place the converted data.
+         * provider:  buffer provider to obtain source data.
+         *   frames:  number of frames to convert
+         *
+         * Returns the number of frames converted.
+         */
+        size_t convert(void *dst, AudioBufferProvider *provider, size_t frames);
+
+        // returns NO_ERROR if constructor was successful
+        status_t initCheck() const {
+            // mSrcChannelMask set on successful updateParameters
+            return mSrcChannelMask != AUDIO_CHANNEL_INVALID ? NO_ERROR : NO_INIT;
+        }
+
+        // allows dynamic reconfigure of all parameters
+        status_t updateParameters(
+                audio_channel_mask_t srcChannelMask, audio_format_t srcFormat,
+                uint32_t srcSampleRate,
+                audio_channel_mask_t dstChannelMask, audio_format_t dstFormat,
+                uint32_t dstSampleRate);
+
+        // called to reset resampler buffers on record track discontinuity
+        void reset() {
+            if (mResampler != NULL) {
+                mResampler->reset();
+            }
+        }
+
+    private:
+        // internal convert function for format and channel mask.
+        void convert(void *dst, /*const*/ void *src, size_t frames);
+
+        // user provided information
+        audio_channel_mask_t mSrcChannelMask;
+        audio_format_t       mSrcFormat;
+        uint32_t             mSrcSampleRate;
+        audio_channel_mask_t mDstChannelMask;
+        audio_format_t       mDstFormat;
+        uint32_t             mDstSampleRate;
+
+        // derived information
+        uint32_t             mSrcChannelCount;
+        uint32_t             mDstChannelCount;
+        size_t               mDstFrameSize;
+
+        // format conversion buffer
+        void                *mBuf;
+        size_t               mBufFrames;
+        size_t               mBufFrameSize;
+
+        // resampler info
+        AudioResampler      *mResampler;
+        // interleaved stereo pairs of fixed-point Q4.27 or float depending on resampler
+        void                *mRsmpOutBuffer;
+        // current allocated frame count for the above, which may be larger than needed
+        size_t               mRsmpOutFrameCount;
     };
 
 #include "RecordTracks.h"
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 5625661..da2d634 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -903,9 +903,14 @@
             mPreviousTimestampValid = false;
             return INVALID_OPERATION;
         }
+        // FIXME Not accurate under dynamic changes of sample rate and speed.
+        // Do not use track's mSampleRate as it is not current for mixer tracks.
+        uint32_t sampleRate = mAudioTrackServerProxy->getSampleRate();
+        float speed, pitch;
+        mAudioTrackServerProxy->getPlaybackRate(&speed, &pitch);
         uint32_t unpresentedFrames =
-                ((int64_t) playbackThread->mLatchQ.mUnpresentedFrames * mSampleRate) /
-                playbackThread->mSampleRate;
+                ((double) playbackThread->mLatchQ.mUnpresentedFrames * sampleRate * speed)
+                / playbackThread->mSampleRate;
         // FIXME Since we're using a raw pointer as the key, it is theoretically possible
         //       for a brand new track to share the same address as a recently destroyed
         //       track, and thus for us to get the frames released of the wrong track.
@@ -1990,29 +1995,30 @@
                           ((flags & IAudioFlinger::TRACK_FAST) ? ALLOC_PIPE : ALLOC_CBLK) :
                           ((buffer == NULL) ? ALLOC_LOCAL : ALLOC_NONE),
                   type),
-        mOverflow(false), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpOutFrameCount(0),
-        // See real initialization of mRsmpInFront at RecordThread::start()
-        mRsmpInUnrel(0), mRsmpInFront(0), mFramesToDrop(0), mResamplerBufferProvider(NULL)
+        mOverflow(false),
+        mFramesToDrop(0)
 {
     if (mCblk == NULL) {
         return;
     }
 
+    mRecordBufferConverter = new RecordBufferConverter(
+            thread->mChannelMask, thread->mFormat, thread->mSampleRate,
+            channelMask, format, sampleRate);
+    // Check if the RecordBufferConverter construction was successful.
+    // If not, don't continue with construction.
+    //
+    // NOTE: It would be extremely rare that the record track cannot be created
+    // for the current device, but a pending or future device change would make
+    // the record track configuration valid.
+    if (mRecordBufferConverter->initCheck() != NO_ERROR) {
+        ALOGE("RecordTrack unable to create record buffer converter");
+        return;
+    }
+
     mServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount,
                                               mFrameSize, !isExternalTrack());
-
-    uint32_t channelCount = audio_channel_count_from_in_mask(channelMask);
-    // FIXME I don't understand either of the channel count checks
-    if (thread->mSampleRate != sampleRate && thread->mChannelCount <= FCC_2 &&
-            channelCount <= FCC_2) {
-        // sink SR
-        mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_16_BIT,
-                thread->mChannelCount, sampleRate);
-        // source SR
-        mResampler->setSampleRate(thread->mSampleRate);
-        mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT);
-        mResamplerBufferProvider = new ResamplerBufferProvider(this);
-    }
+    mResamplerBufferProvider = new ResamplerBufferProvider(this);
 
     if (flags & IAudioFlinger::TRACK_FAST) {
         ALOG_ASSERT(thread->mFastTrackAvail);
@@ -2023,11 +2029,19 @@
 AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
 {
     ALOGV("%s", __func__);
-    delete mResampler;
-    delete[] mRsmpOutBuffer;
+    delete mRecordBufferConverter;
     delete mResamplerBufferProvider;
 }
 
+status_t AudioFlinger::RecordThread::RecordTrack::initCheck() const
+{
+    status_t status = TrackBase::initCheck();
+    if (status == NO_ERROR && mServerProxy == 0) {
+        status = BAD_VALUE;
+    }
+    return status;
+}
+
 // AudioBufferProvider interface
 status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer,
         int64_t pts __unused)
diff --git a/services/audioflinger/tests/Android.mk b/services/audioflinger/tests/Android.mk
index 8604ef5..76997be 100644
--- a/services/audioflinger/tests/Android.mk
+++ b/services/audioflinger/tests/Android.mk
@@ -39,6 +39,7 @@
 LOCAL_SRC_FILES:= \
 	test-mixer.cpp \
 	../AudioMixer.cpp.arm \
+	../BufferProviders.cpp
 
 LOCAL_C_INCLUDES := \
 	$(call include-path-for, audio-effects) \
diff --git a/services/audioflinger/tests/resampler_tests.cpp b/services/audioflinger/tests/resampler_tests.cpp
index d6217ba..9e375db 100644
--- a/services/audioflinger/tests/resampler_tests.cpp
+++ b/services/audioflinger/tests/resampler_tests.cpp
@@ -48,7 +48,10 @@
         if (thisFrames == 0 || thisFrames > outputFrames - i) {
             thisFrames = outputFrames - i;
         }
-        resampler->resample((int32_t*) output + channels*i, thisFrames, provider);
+        size_t framesResampled = resampler->resample(
+                (int32_t*) output + channels*i, thisFrames, provider);
+        // we should have enough buffer space, so there is no short count.
+        ASSERT_EQ(thisFrames, framesResampled);
         i += thisFrames;
     }
 }
diff --git a/services/audiopolicy/common/include/Volume.h b/services/audiopolicy/common/include/Volume.h
index a4cc759..4205589 100755
--- a/services/audiopolicy/common/include/Volume.h
+++ b/services/audiopolicy/common/include/Volume.h
@@ -18,6 +18,10 @@
 
 #include <system/audio.h>
 #include <utils/Log.h>
+#include <math.h>
+
+// Absolute min volume in dB (can be represented in single precision normal float value)
+#define VOLUME_MIN_DB (-758)
 
 class VolumeCurvePoint
 {
@@ -32,7 +36,7 @@
     /**
      * 4 points to define the volume attenuation curve, each characterized by the volume
      * index (from 0 to 100) at which they apply, and the attenuation in dB at that index.
-     * we use 100 steps to avoid rounding errors when computing the volume in volIndexToAmpl()
+     * we use 100 steps to avoid rounding errors when computing the volume in volIndexToDb()
      *
      * @todo shall become configurable
      */
@@ -134,4 +138,20 @@
         }
     }
 
+    static inline float DbToAmpl(float decibels)
+    {
+        if (decibels <= VOLUME_MIN_DB) {
+            return 0.0f;
+        }
+        return exp( decibels * 0.115129f); // exp( dB * ln(10) / 20 )
+    }
+
+    static inline float AmplToDb(float amplification)
+    {
+        if (amplification == 0) {
+            return VOLUME_MIN_DB;
+        }
+        return 20 * log10(amplification);
+    }
+
 };
diff --git a/services/audiopolicy/common/managerdefinitions/Android.mk b/services/audiopolicy/common/managerdefinitions/Android.mk
index 71ba1cb..7c265aa 100644
--- a/services/audiopolicy/common/managerdefinitions/Android.mk
+++ b/services/audiopolicy/common/managerdefinitions/Android.mk
@@ -25,6 +25,7 @@
 LOCAL_C_INCLUDES += \
     $(LOCAL_PATH)/include \
     $(TOPDIR)frameworks/av/services/audiopolicy/common/include \
+    $(TOPDIR)frameworks/av/services/audiopolicy
 
 LOCAL_EXPORT_C_INCLUDE_DIRS := \
     $(LOCAL_PATH)/include
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
index cc2a3bd..f1aee46 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
@@ -27,25 +27,36 @@
 
 class IOProfile;
 class AudioMix;
+class AudioPolicyClientInterface;
 
 // descriptor for audio outputs. Used to maintain current configuration of each opened audio output
 // and keep track of the usage of this output by each audio stream type.
 class AudioOutputDescriptor: public AudioPortConfig
 {
 public:
-    AudioOutputDescriptor(const sp<IOProfile>& profile);
+    AudioOutputDescriptor(const sp<AudioPort>& port,
+                          AudioPolicyClientInterface *clientInterface);
+    virtual ~AudioOutputDescriptor() {}
 
     status_t    dump(int fd);
     void        log(const char* indent);
 
-    audio_devices_t device() const;
-    void changeRefCount(audio_stream_type_t stream, int delta);
     audio_port_handle_t getId() const;
-    void setIoHandle(audio_io_handle_t ioHandle);
-    bool isDuplicated() const { return (mOutput1 != NULL && mOutput2 != NULL); }
-    audio_devices_t supportedDevices();
-    uint32_t latency();
-    bool sharesHwModuleWith(const sp<AudioOutputDescriptor> outputDesc);
+    virtual audio_devices_t device() const;
+    virtual bool sharesHwModuleWith(const sp<AudioOutputDescriptor> outputDesc);
+    virtual audio_devices_t supportedDevices();
+    virtual bool isDuplicated() const { return false; }
+    virtual uint32_t latency() { return 0; }
+    virtual bool isFixedVolume(audio_devices_t device);
+    virtual sp<AudioOutputDescriptor> subOutput1() { return 0; }
+    virtual sp<AudioOutputDescriptor> subOutput2() { return 0; }
+    virtual bool setVolume(float volume,
+                           audio_stream_type_t stream,
+                           audio_devices_t device,
+                           uint32_t delayMs,
+                           bool force);
+    virtual void changeRefCount(audio_stream_type_t stream, int delta);
+
     bool isActive(uint32_t inPastMs = 0) const;
     bool isStreamActive(audio_stream_type_t stream,
                         uint32_t inPastMs = 0,
@@ -53,34 +64,69 @@
 
     virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
                            const struct audio_port_config *srcConfig = NULL) const;
-    virtual sp<AudioPort> getAudioPort() const { return mProfile; }
-    void toAudioPort(struct audio_port *port) const;
+    virtual sp<AudioPort> getAudioPort() const { return mPort; }
+    virtual void toAudioPort(struct audio_port *port) const;
 
     audio_module_handle_t getModuleHandle() const;
 
-    audio_io_handle_t mIoHandle;              // output handle
-    uint32_t mLatency;                  //
-    audio_output_flags_t mFlags;   //
+    sp<AudioPort>       mPort;
     audio_devices_t mDevice;                   // current device this output is routed to
-    AudioMix *mPolicyMix;             // non NULL when used by a dynamic policy
     audio_patch_handle_t mPatchHandle;
     uint32_t mRefCount[AUDIO_STREAM_CNT]; // number of streams of each type using this output
     nsecs_t mStopTime[AUDIO_STREAM_CNT];
-    sp<AudioOutputDescriptor> mOutput1;    // used by duplicated outputs: first output
-    sp<AudioOutputDescriptor> mOutput2;    // used by duplicated outputs: second output
-    float mCurVolume[AUDIO_STREAM_CNT];   // current stream volume
+    float mCurVolume[AUDIO_STREAM_CNT];   // current stream volume in dB
     int mMuteCount[AUDIO_STREAM_CNT];     // mute request counter
-    const sp<IOProfile> mProfile;          // I/O profile this output derives from
     bool mStrategyMutedByDevice[NUM_STRATEGIES]; // strategies muted because of incompatible
                                         // device selection. See checkDeviceMuteStrategies()
-    uint32_t mDirectOpenCount; // number of clients using this output (direct outputs only)
+    AudioPolicyClientInterface *mClientInterface;
 
-private:
+protected:
     audio_port_handle_t mId;
 };
 
-class AudioOutputCollection :
-        public DefaultKeyedVector< audio_io_handle_t, sp<AudioOutputDescriptor> >
+// Audio output driven by a software mixer in audio flinger.
+class SwAudioOutputDescriptor: public AudioOutputDescriptor
+{
+public:
+    SwAudioOutputDescriptor(const sp<IOProfile>& profile,
+                            AudioPolicyClientInterface *clientInterface);
+    virtual ~SwAudioOutputDescriptor() {}
+
+    status_t    dump(int fd);
+
+    void setIoHandle(audio_io_handle_t ioHandle);
+
+    virtual audio_devices_t device() const;
+    virtual bool sharesHwModuleWith(const sp<AudioOutputDescriptor> outputDesc);
+    virtual audio_devices_t supportedDevices();
+    virtual uint32_t latency();
+    virtual bool isDuplicated() const { return (mOutput1 != NULL && mOutput2 != NULL); }
+    virtual bool isFixedVolume(audio_devices_t device);
+    virtual sp<AudioOutputDescriptor> subOutput1() { return mOutput1; }
+    virtual sp<AudioOutputDescriptor> subOutput2() { return mOutput2; }
+    virtual void changeRefCount(audio_stream_type_t stream, int delta);
+    virtual bool setVolume(float volume,
+                           audio_stream_type_t stream,
+                           audio_devices_t device,
+                           uint32_t delayMs,
+                           bool force);
+
+    virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
+                           const struct audio_port_config *srcConfig = NULL) const;
+    virtual void toAudioPort(struct audio_port *port) const;
+
+    const sp<IOProfile> mProfile;          // I/O profile this output derives from
+    audio_io_handle_t mIoHandle;           // output handle
+    uint32_t mLatency;                  //
+    audio_output_flags_t mFlags;   //
+    AudioMix *mPolicyMix;             // non NULL when used by a dynamic policy
+    sp<SwAudioOutputDescriptor> mOutput1;    // used by duplicated outputs: first output
+    sp<SwAudioOutputDescriptor> mOutput2;    // used by duplicated outputs: second output
+    uint32_t mDirectOpenCount; // number of clients using this output (direct outputs only)
+};
+
+class SwAudioOutputCollection :
+        public DefaultKeyedVector< audio_io_handle_t, sp<SwAudioOutputDescriptor> >
 {
 public:
     bool isStreamActive(audio_stream_type_t stream, uint32_t inPastMs = 0) const;
@@ -99,9 +145,9 @@
      */
     audio_io_handle_t getA2dpOutput() const;
 
-    sp<AudioOutputDescriptor> getOutputFromId(audio_port_handle_t id) const;
+    sp<SwAudioOutputDescriptor> getOutputFromId(audio_port_handle_t id) const;
 
-    sp<AudioOutputDescriptor> getPrimaryOutput() const;
+    sp<SwAudioOutputDescriptor> getPrimaryOutput() const;
 
     /**
      * return true if any output is playing anything besides the stream to ignore
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioPolicyMix.h b/services/audiopolicy/common/managerdefinitions/include/AudioPolicyMix.h
index 988aed6..d51f4e1 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioPolicyMix.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioPolicyMix.h
@@ -24,7 +24,7 @@
 
 namespace android {
 
-class AudioOutputDescriptor;
+class SwAudioOutputDescriptor;
 
 /**
  * custom mix entry in mPolicyMixes
@@ -33,19 +33,19 @@
 public:
     AudioPolicyMix() {}
 
-    const sp<AudioOutputDescriptor> &getOutput() const;
+    const sp<SwAudioOutputDescriptor> &getOutput() const;
 
-    void setOutput(sp<AudioOutputDescriptor> &output);
+    void setOutput(sp<SwAudioOutputDescriptor> &output);
 
     void clearOutput();
 
-    android::AudioMix &getMix();
+    android::AudioMix *getMix();
 
     void setMix(AudioMix &mix);
 
 private:
     AudioMix    mMix;                   // Audio policy mix descriptor
-    sp<AudioOutputDescriptor> mOutput;  // Corresponding output stream
+    sp<SwAudioOutputDescriptor> mOutput;  // Corresponding output stream
 };
 
 
@@ -58,24 +58,24 @@
 
     status_t unregisterMix(String8 address);
 
-    void closeOutput(sp<AudioOutputDescriptor> &desc);
+    void closeOutput(sp<SwAudioOutputDescriptor> &desc);
 
     /**
      * Try to find an output descriptor for the given attributes.
      *
-     * @param[in] attributes to consider for the research of output descriptor.
+     * @param[in] attributes to consider fowr the research of output descriptor.
      * @param[out] desc to return if an output could be found.
      *
      * @return NO_ERROR if an output was found for the given attribute (in this case, the
      *                  descriptor output param is initialized), error code otherwise.
      */
-    status_t getOutputForAttr(audio_attributes_t attributes, sp<AudioOutputDescriptor> &desc);
+    status_t getOutputForAttr(audio_attributes_t attributes, sp<SwAudioOutputDescriptor> &desc);
 
     audio_devices_t getDeviceAndMixForInputSource(audio_source_t inputSource,
                                                   audio_devices_t availableDeviceTypes,
                                                   AudioMix **policyMix);
 
-    status_t getInputMixForAttr(audio_attributes_t attr, AudioMix *&policyMix);
+    status_t getInputMixForAttr(audio_attributes_t attr, AudioMix **policyMix);
 };
 
 }; // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioPort.h b/services/audiopolicy/common/managerdefinitions/include/AudioPort.h
index dea1b8a..1c2c27e 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioPort.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioPort.h
@@ -62,8 +62,12 @@
     // searches for an exact match
     status_t checkExactChannelMask(audio_channel_mask_t channelMask) const;
     // searches for a compatible match, currently implemented for input channel masks only
-    status_t checkCompatibleChannelMask(audio_channel_mask_t channelMask) const;
-    status_t checkFormat(audio_format_t format) const;
+    status_t checkCompatibleChannelMask(audio_channel_mask_t channelMask,
+            audio_channel_mask_t *updatedChannelMask) const;
+
+    status_t checkExactFormat(audio_format_t format) const;
+    // searches for a compatible match, currently implemented for input formats only
+    status_t checkCompatibleFormat(audio_format_t format, audio_format_t *updatedFormat) const;
     status_t checkGain(const struct audio_gain_config *gainConfig, int index) const;
 
     uint32_t pickSamplingRate() const;
@@ -71,6 +75,11 @@
     audio_format_t pickFormat() const;
 
     static const audio_format_t sPcmFormatCompareTable[];
+    static int compareFormatsGoodToBad(
+            const audio_format_t *format1, const audio_format_t *format2) {
+        // compareFormats sorts from bad to good, we reverse it here
+        return compareFormats(*format2, *format1);
+    }
     static int compareFormats(audio_format_t format1, audio_format_t format2);
 
     audio_module_handle_t getModuleHandle() const;
diff --git a/services/audiopolicy/common/managerdefinitions/include/IOProfile.h b/services/audiopolicy/common/managerdefinitions/include/IOProfile.h
index 022257e..ab6fcc1 100644
--- a/services/audiopolicy/common/managerdefinitions/include/IOProfile.h
+++ b/services/audiopolicy/common/managerdefinitions/include/IOProfile.h
@@ -45,7 +45,9 @@
                              uint32_t samplingRate,
                              uint32_t *updatedSamplingRate,
                              audio_format_t format,
+                             audio_format_t *updatedFormat,
                              audio_channel_mask_t channelMask,
+                             audio_channel_mask_t *updatedChannelMask,
                              uint32_t flags) const;
 
     void dump(int fd);
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
index 8de8cd8..596aa1d 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
@@ -17,9 +17,11 @@
 #define LOG_TAG "APM::AudioOutputDescriptor"
 //#define LOG_NDEBUG 0
 
+#include <AudioPolicyInterface.h>
 #include "AudioOutputDescriptor.h"
 #include "IOProfile.h"
 #include "AudioGain.h"
+#include "Volume.h"
 #include "HwModule.h"
 #include <media/AudioPolicy.h>
 
@@ -29,12 +31,10 @@
 
 namespace android {
 
-AudioOutputDescriptor::AudioOutputDescriptor(const sp<IOProfile>& profile)
-    : mIoHandle(0), mLatency(0),
-    mFlags((audio_output_flags_t)0), mDevice(AUDIO_DEVICE_NONE), mPolicyMix(NULL),
-    mPatchHandle(0),
-    mOutput1(0), mOutput2(0), mProfile(profile), mDirectOpenCount(0),
-    mId(0)
+AudioOutputDescriptor::AudioOutputDescriptor(const sp<AudioPort>& port,
+                                             AudioPolicyClientInterface *clientInterface)
+    : mPort(port), mDevice(AUDIO_DEVICE_NONE),
+      mPatchHandle(0), mClientInterface(clientInterface), mId(0)
 {
     // clear usage count for all stream types
     for (int i = 0; i < AUDIO_STREAM_CNT; i++) {
@@ -46,23 +46,19 @@
     for (int i = 0; i < NUM_STRATEGIES; i++) {
         mStrategyMutedByDevice[i] = false;
     }
-    if (profile != NULL) {
-        mFlags = (audio_output_flags_t)profile->mFlags;
-        mSamplingRate = profile->pickSamplingRate();
-        mFormat = profile->pickFormat();
-        mChannelMask = profile->pickChannelMask();
-        if (profile->mGains.size() > 0) {
-            profile->mGains[0]->getDefaultConfig(&mGain);
+    if (port != NULL) {
+        mSamplingRate = port->pickSamplingRate();
+        mFormat = port->pickFormat();
+        mChannelMask = port->pickChannelMask();
+        if (port->mGains.size() > 0) {
+            port->mGains[0]->getDefaultConfig(&mGain);
         }
     }
 }
 
 audio_module_handle_t AudioOutputDescriptor::getModuleHandle() const
 {
-    if (mProfile == 0) {
-        return 0;
-    }
-    return mProfile->getModuleHandle();
+    return mPort->getModuleHandle();
 }
 
 audio_port_handle_t AudioOutputDescriptor::getId() const
@@ -72,35 +68,20 @@
 
 audio_devices_t AudioOutputDescriptor::device() const
 {
-    if (isDuplicated()) {
-        return (audio_devices_t)(mOutput1->mDevice | mOutput2->mDevice);
-    } else {
-        return mDevice;
-    }
+    return mDevice;
 }
 
-void AudioOutputDescriptor::setIoHandle(audio_io_handle_t ioHandle)
+audio_devices_t AudioOutputDescriptor::supportedDevices()
 {
-    mId = AudioPort::getNextUniqueId();
-    mIoHandle = ioHandle;
-}
-
-uint32_t AudioOutputDescriptor::latency()
-{
-    if (isDuplicated()) {
-        return (mOutput1->mLatency > mOutput2->mLatency) ? mOutput1->mLatency : mOutput2->mLatency;
-    } else {
-        return mLatency;
-    }
+    return mDevice;
 }
 
 bool AudioOutputDescriptor::sharesHwModuleWith(
         const sp<AudioOutputDescriptor> outputDesc)
 {
-    if (isDuplicated()) {
-        return mOutput1->sharesHwModuleWith(outputDesc) || mOutput2->sharesHwModuleWith(outputDesc);
-    } else if (outputDesc->isDuplicated()){
-        return sharesHwModuleWith(outputDesc->mOutput1) || sharesHwModuleWith(outputDesc->mOutput2);
+    if (outputDesc->isDuplicated()) {
+        return sharesHwModuleWith(outputDesc->subOutput1()) ||
+                    sharesHwModuleWith(outputDesc->subOutput2());
     } else {
         return (getModuleHandle() == outputDesc->getModuleHandle());
     }
@@ -109,11 +90,6 @@
 void AudioOutputDescriptor::changeRefCount(audio_stream_type_t stream,
                                                                    int delta)
 {
-    // forward usage count change to attached outputs
-    if (isDuplicated()) {
-        mOutput1->changeRefCount(stream, delta);
-        mOutput2->changeRefCount(stream, delta);
-    }
     if ((delta + (int)mRefCount[stream]) < 0) {
         ALOGW("changeRefCount() invalid delta %d for stream %d, refCount %d",
               delta, stream, mRefCount[stream]);
@@ -124,15 +100,6 @@
     ALOGV("changeRefCount() stream %d, count %d", stream, mRefCount[stream]);
 }
 
-audio_devices_t AudioOutputDescriptor::supportedDevices()
-{
-    if (isDuplicated()) {
-        return (audio_devices_t)(mOutput1->supportedDevices() | mOutput2->supportedDevices());
-    } else {
-        return mProfile->mSupportedDevices.types() ;
-    }
-}
-
 bool AudioOutputDescriptor::isActive(uint32_t inPastMs) const
 {
     nsecs_t sysTime = 0;
@@ -169,12 +136,33 @@
     return false;
 }
 
+
+bool AudioOutputDescriptor::isFixedVolume(audio_devices_t device __unused)
+{
+    return false;
+}
+
+bool AudioOutputDescriptor::setVolume(float volume,
+                                      audio_stream_type_t stream,
+                                      audio_devices_t device __unused,
+                                      uint32_t delayMs,
+                                      bool force)
+{
+    // We actually change the volume if:
+    // - the float value returned by computeVolume() changed
+    // - the force flag is set
+    if (volume != mCurVolume[stream] || force) {
+        ALOGV("setVolume() for stream %d, volume %f, delay %d", stream, volume, delayMs);
+        mCurVolume[stream] = volume;
+        return true;
+    }
+    return false;
+}
+
 void AudioOutputDescriptor::toAudioPortConfig(
                                                  struct audio_port_config *dstConfig,
                                                  const struct audio_port_config *srcConfig) const
 {
-    ALOG_ASSERT(!isDuplicated(), "toAudioPortConfig() called on duplicated output %d", mIoHandle);
-
     dstConfig->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK|
                             AUDIO_PORT_CONFIG_FORMAT|AUDIO_PORT_CONFIG_GAIN;
     if (srcConfig != NULL) {
@@ -186,21 +174,15 @@
     dstConfig->role = AUDIO_PORT_ROLE_SOURCE;
     dstConfig->type = AUDIO_PORT_TYPE_MIX;
     dstConfig->ext.mix.hw_module = getModuleHandle();
-    dstConfig->ext.mix.handle = mIoHandle;
     dstConfig->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT;
 }
 
 void AudioOutputDescriptor::toAudioPort(
                                                     struct audio_port *port) const
 {
-    ALOG_ASSERT(!isDuplicated(), "toAudioPort() called on duplicated output %d", mIoHandle);
-    mProfile->toAudioPort(port);
+    mPort->toAudioPort(port);
     port->id = mId;
-    toAudioPortConfig(&port->active_config);
     port->ext.mix.hw_module = getModuleHandle();
-    port->ext.mix.handle = mIoHandle;
-    port->ext.mix.latency_class =
-            mFlags & AUDIO_OUTPUT_FLAG_FAST ? AUDIO_LATENCY_LOW : AUDIO_LATENCY_NORMAL;
 }
 
 status_t AudioOutputDescriptor::dump(int fd)
@@ -209,7 +191,7 @@
     char buffer[SIZE];
     String8 result;
 
-    snprintf(buffer, SIZE, " ID: %d\n", getId());
+    snprintf(buffer, SIZE, " ID: %d\n", mId);
     result.append(buffer);
     snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate);
     result.append(buffer);
@@ -217,10 +199,6 @@
     result.append(buffer);
     snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask);
     result.append(buffer);
-    snprintf(buffer, SIZE, " Latency: %d\n", mLatency);
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Flags %08x\n", mFlags);
-    result.append(buffer);
     snprintf(buffer, SIZE, " Devices %08x\n", device());
     result.append(buffer);
     snprintf(buffer, SIZE, " Stream volume refCount muteCount\n");
@@ -237,15 +215,163 @@
 
 void AudioOutputDescriptor::log(const char* indent)
 {
-    ALOGI("%sID: %d,0x%X, [rt:%d fmt:0x%X ch:0x%X] hndl:%d",
-          indent, mId, mId, mSamplingRate, mFormat, mChannelMask, mIoHandle);
+    ALOGI("%sID: %d,0x%X, [rt:%d fmt:0x%X ch:0x%X]",
+          indent, mId, mId, mSamplingRate, mFormat, mChannelMask);
 }
 
-bool AudioOutputCollection::isStreamActive(audio_stream_type_t stream, uint32_t inPastMs) const
+// SwAudioOutputDescriptor implementation
+SwAudioOutputDescriptor::SwAudioOutputDescriptor(
+        const sp<IOProfile>& profile, AudioPolicyClientInterface *clientInterface)
+    : AudioOutputDescriptor(profile, clientInterface),
+    mProfile(profile), mIoHandle(0), mLatency(0),
+    mFlags((audio_output_flags_t)0), mPolicyMix(NULL),
+    mOutput1(0), mOutput2(0), mDirectOpenCount(0)
+{
+    if (profile != NULL) {
+        mFlags = (audio_output_flags_t)profile->mFlags;
+    }
+}
+
+void SwAudioOutputDescriptor::setIoHandle(audio_io_handle_t ioHandle)
+{
+    mId = AudioPort::getNextUniqueId();
+    mIoHandle = ioHandle;
+}
+
+
+status_t SwAudioOutputDescriptor::dump(int fd)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, " Latency: %d\n", mLatency);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Flags %08x\n", mFlags);
+    result.append(buffer);
+    write(fd, result.string(), result.size());
+
+    AudioOutputDescriptor::dump(fd);
+
+    return NO_ERROR;
+}
+
+audio_devices_t SwAudioOutputDescriptor::device() const
+{
+    if (isDuplicated()) {
+        return (audio_devices_t)(mOutput1->mDevice | mOutput2->mDevice);
+    } else {
+        return mDevice;
+    }
+}
+
+bool SwAudioOutputDescriptor::sharesHwModuleWith(
+        const sp<AudioOutputDescriptor> outputDesc)
+{
+    if (isDuplicated()) {
+        return mOutput1->sharesHwModuleWith(outputDesc) || mOutput2->sharesHwModuleWith(outputDesc);
+    } else if (outputDesc->isDuplicated()){
+        return sharesHwModuleWith(outputDesc->subOutput1()) ||
+                    sharesHwModuleWith(outputDesc->subOutput2());
+    } else {
+        return AudioOutputDescriptor::sharesHwModuleWith(outputDesc);
+    }
+}
+
+audio_devices_t SwAudioOutputDescriptor::supportedDevices()
+{
+    if (isDuplicated()) {
+        return (audio_devices_t)(mOutput1->supportedDevices() | mOutput2->supportedDevices());
+    } else {
+        return mProfile->mSupportedDevices.types() ;
+    }
+}
+
+uint32_t SwAudioOutputDescriptor::latency()
+{
+    if (isDuplicated()) {
+        return (mOutput1->mLatency > mOutput2->mLatency) ? mOutput1->mLatency : mOutput2->mLatency;
+    } else {
+        return mLatency;
+    }
+}
+
+void SwAudioOutputDescriptor::changeRefCount(audio_stream_type_t stream,
+                                                                   int delta)
+{
+    // forward usage count change to attached outputs
+    if (isDuplicated()) {
+        mOutput1->changeRefCount(stream, delta);
+        mOutput2->changeRefCount(stream, delta);
+    }
+    AudioOutputDescriptor::changeRefCount(stream, delta);
+}
+
+
+bool SwAudioOutputDescriptor::isFixedVolume(audio_devices_t device)
+{
+    // unit gain if rerouting to external policy
+    if (device == AUDIO_DEVICE_OUT_REMOTE_SUBMIX) {
+        if (mPolicyMix != NULL) {
+            ALOGV("max gain when rerouting for output=%d", mIoHandle);
+            return true;
+        }
+    }
+    return false;
+}
+
+void SwAudioOutputDescriptor::toAudioPortConfig(
+                                                 struct audio_port_config *dstConfig,
+                                                 const struct audio_port_config *srcConfig) const
+{
+
+    ALOG_ASSERT(!isDuplicated(), "toAudioPortConfig() called on duplicated output %d", mIoHandle);
+    AudioOutputDescriptor::toAudioPortConfig(dstConfig, srcConfig);
+
+    dstConfig->ext.mix.handle = mIoHandle;
+}
+
+void SwAudioOutputDescriptor::toAudioPort(
+                                                    struct audio_port *port) const
+{
+    ALOG_ASSERT(!isDuplicated(), "toAudioPort() called on duplicated output %d", mIoHandle);
+
+    AudioOutputDescriptor::toAudioPort(port);
+
+    toAudioPortConfig(&port->active_config);
+    port->ext.mix.handle = mIoHandle;
+    port->ext.mix.latency_class =
+            mFlags & AUDIO_OUTPUT_FLAG_FAST ? AUDIO_LATENCY_LOW : AUDIO_LATENCY_NORMAL;
+}
+
+bool SwAudioOutputDescriptor::setVolume(float volume,
+                                        audio_stream_type_t stream,
+                                        audio_devices_t device,
+                                        uint32_t delayMs,
+                                        bool force)
+{
+    bool changed = AudioOutputDescriptor::setVolume(volume, stream, device, delayMs, force);
+
+    if (changed) {
+        // Force VOICE_CALL to track BLUETOOTH_SCO stream volume when bluetooth audio is
+        // enabled
+        float volume = Volume::DbToAmpl(mCurVolume[stream]);
+        if (stream == AUDIO_STREAM_BLUETOOTH_SCO) {
+            mClientInterface->setStreamVolume(
+                    AUDIO_STREAM_VOICE_CALL, volume, mIoHandle, delayMs);
+        }
+        mClientInterface->setStreamVolume(stream, volume, mIoHandle, delayMs);
+    }
+    return changed;
+}
+
+// SwAudioOutputCollection implementation
+
+bool SwAudioOutputCollection::isStreamActive(audio_stream_type_t stream, uint32_t inPastMs) const
 {
     nsecs_t sysTime = systemTime();
     for (size_t i = 0; i < this->size(); i++) {
-        const sp<AudioOutputDescriptor> outputDesc = this->valueAt(i);
+        const sp<SwAudioOutputDescriptor> outputDesc = this->valueAt(i);
         if (outputDesc->isStreamActive(stream, inPastMs, sysTime)) {
             return true;
         }
@@ -253,12 +379,12 @@
     return false;
 }
 
-bool AudioOutputCollection::isStreamActiveRemotely(audio_stream_type_t stream,
+bool SwAudioOutputCollection::isStreamActiveRemotely(audio_stream_type_t stream,
                                                    uint32_t inPastMs) const
 {
     nsecs_t sysTime = systemTime();
     for (size_t i = 0; i < size(); i++) {
-        const sp<AudioOutputDescriptor> outputDesc = valueAt(i);
+        const sp<SwAudioOutputDescriptor> outputDesc = valueAt(i);
         if (((outputDesc->device() & APM_AUDIO_OUT_DEVICE_REMOTE_ALL) != 0) &&
                 outputDesc->isStreamActive(stream, inPastMs, sysTime)) {
             // do not consider re routing (when the output is going to a dynamic policy)
@@ -271,10 +397,10 @@
     return false;
 }
 
-audio_io_handle_t AudioOutputCollection::getA2dpOutput() const
+audio_io_handle_t SwAudioOutputCollection::getA2dpOutput() const
 {
     for (size_t i = 0; i < size(); i++) {
-        sp<AudioOutputDescriptor> outputDesc = valueAt(i);
+        sp<SwAudioOutputDescriptor> outputDesc = valueAt(i);
         if (!outputDesc->isDuplicated() && outputDesc->device() & AUDIO_DEVICE_OUT_ALL_A2DP) {
             return this->keyAt(i);
         }
@@ -282,10 +408,10 @@
     return 0;
 }
 
-sp<AudioOutputDescriptor> AudioOutputCollection::getPrimaryOutput() const
+sp<SwAudioOutputDescriptor> SwAudioOutputCollection::getPrimaryOutput() const
 {
     for (size_t i = 0; i < size(); i++) {
-        const sp<AudioOutputDescriptor> outputDesc = valueAt(i);
+        const sp<SwAudioOutputDescriptor> outputDesc = valueAt(i);
         if (outputDesc->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) {
             return outputDesc;
         }
@@ -293,9 +419,9 @@
     return NULL;
 }
 
-sp<AudioOutputDescriptor> AudioOutputCollection::getOutputFromId(audio_port_handle_t id) const
+sp<SwAudioOutputDescriptor> SwAudioOutputCollection::getOutputFromId(audio_port_handle_t id) const
 {
-    sp<AudioOutputDescriptor> outputDesc = NULL;
+    sp<SwAudioOutputDescriptor> outputDesc = NULL;
     for (size_t i = 0; i < size(); i++) {
         outputDesc = valueAt(i);
         if (outputDesc->getId() == id) {
@@ -305,14 +431,14 @@
     return outputDesc;
 }
 
-bool AudioOutputCollection::isAnyOutputActive(audio_stream_type_t streamToIgnore) const
+bool SwAudioOutputCollection::isAnyOutputActive(audio_stream_type_t streamToIgnore) const
 {
     for (size_t s = 0 ; s < AUDIO_STREAM_CNT ; s++) {
         if (s == (size_t) streamToIgnore) {
             continue;
         }
         for (size_t i = 0; i < size(); i++) {
-            const sp<AudioOutputDescriptor> outputDesc = valueAt(i);
+            const sp<SwAudioOutputDescriptor> outputDesc = valueAt(i);
             if (outputDesc->mRefCount[s] != 0) {
                 return true;
             }
@@ -321,15 +447,15 @@
     return false;
 }
 
-audio_devices_t AudioOutputCollection::getSupportedDevices(audio_io_handle_t handle) const
+audio_devices_t SwAudioOutputCollection::getSupportedDevices(audio_io_handle_t handle) const
 {
-    sp<AudioOutputDescriptor> outputDesc = valueFor(handle);
+    sp<SwAudioOutputDescriptor> outputDesc = valueFor(handle);
     audio_devices_t devices = outputDesc->mProfile->mSupportedDevices.types();
     return devices;
 }
 
 
-status_t AudioOutputCollection::dump(int fd) const
+status_t SwAudioOutputCollection::dump(int fd) const
 {
     const size_t SIZE = 256;
     char buffer[SIZE];
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp
index 84a53ebd..77fc0b9 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp
@@ -26,12 +26,12 @@
 
 namespace android {
 
-void AudioPolicyMix::setOutput(sp<AudioOutputDescriptor> &output)
+void AudioPolicyMix::setOutput(sp<SwAudioOutputDescriptor> &output)
 {
     mOutput = output;
 }
 
-const sp<AudioOutputDescriptor> &AudioPolicyMix::getOutput() const
+const sp<SwAudioOutputDescriptor> &AudioPolicyMix::getOutput() const
 {
     return mOutput;
 }
@@ -46,9 +46,9 @@
     mMix = mix;
 }
 
-android::AudioMix &AudioPolicyMix::getMix()
+android::AudioMix *AudioPolicyMix::getMix()
 {
-    return mMix;
+    return &mMix;
 }
 
 status_t AudioPolicyMixCollection::registerMix(String8 address, AudioMix mix)
@@ -88,7 +88,7 @@
     return NO_ERROR;
 }
 
-void AudioPolicyMixCollection::closeOutput(sp<AudioOutputDescriptor> &desc)
+void AudioPolicyMixCollection::closeOutput(sp<SwAudioOutputDescriptor> &desc)
 {
     for (size_t i = 0; i < size(); i++) {
         sp<AudioPolicyMix> policyMix = valueAt(i);
@@ -99,40 +99,40 @@
 }
 
 status_t AudioPolicyMixCollection::getOutputForAttr(audio_attributes_t attributes,
-                                                    sp<AudioOutputDescriptor> &desc)
+                                                    sp<SwAudioOutputDescriptor> &desc)
 {
     for (size_t i = 0; i < size(); i++) {
         sp<AudioPolicyMix> policyMix = valueAt(i);
-        AudioMix mix = policyMix->getMix();
+        AudioMix *mix = policyMix->getMix();
 
-        if (mix.mMixType == MIX_TYPE_PLAYERS) {
-            for (size_t j = 0; j < mix.mCriteria.size(); j++) {
-                if ((RULE_MATCH_ATTRIBUTE_USAGE == mix.mCriteria[j].mRule &&
-                     mix.mCriteria[j].mAttr.mUsage == attributes.usage) ||
-                        (RULE_EXCLUDE_ATTRIBUTE_USAGE == mix.mCriteria[j].mRule &&
-                         mix.mCriteria[j].mAttr.mUsage != attributes.usage)) {
+        if (mix->mMixType == MIX_TYPE_PLAYERS) {
+            for (size_t j = 0; j < mix->mCriteria.size(); j++) {
+                if ((RULE_MATCH_ATTRIBUTE_USAGE == mix->mCriteria[j].mRule &&
+                     mix->mCriteria[j].mAttr.mUsage == attributes.usage) ||
+                        (RULE_EXCLUDE_ATTRIBUTE_USAGE == mix->mCriteria[j].mRule &&
+                         mix->mCriteria[j].mAttr.mUsage != attributes.usage)) {
                     desc = policyMix->getOutput();
                     break;
                 }
                 if (strncmp(attributes.tags, "addr=", strlen("addr=")) == 0 &&
                         strncmp(attributes.tags + strlen("addr="),
-                                mix.mRegistrationId.string(),
+                                mix->mRegistrationId.string(),
                                 AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - strlen("addr=") - 1) == 0) {
                     desc = policyMix->getOutput();
                     break;
                 }
             }
-        } else if (mix.mMixType == MIX_TYPE_RECORDERS) {
+        } else if (mix->mMixType == MIX_TYPE_RECORDERS) {
             if (attributes.usage == AUDIO_USAGE_VIRTUAL_SOURCE &&
                     strncmp(attributes.tags, "addr=", strlen("addr=")) == 0 &&
                     strncmp(attributes.tags + strlen("addr="),
-                            mix.mRegistrationId.string(),
+                            mix->mRegistrationId.string(),
                             AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - strlen("addr=") - 1) == 0) {
                 desc = policyMix->getOutput();
             }
         }
         if (desc != 0) {
-            desc->mPolicyMix = &mix;
+            desc->mPolicyMix = mix;
             return NO_ERROR;
         }
     }
@@ -144,19 +144,19 @@
                                                                         AudioMix **policyMix)
 {
     for (size_t i = 0; i < size(); i++) {
-        AudioMix mix = valueAt(i)->getMix();
+        AudioMix *mix = valueAt(i)->getMix();
 
-        if (mix.mMixType != MIX_TYPE_RECORDERS) {
+        if (mix->mMixType != MIX_TYPE_RECORDERS) {
             continue;
         }
-        for (size_t j = 0; j < mix.mCriteria.size(); j++) {
-            if ((RULE_MATCH_ATTRIBUTE_CAPTURE_PRESET == mix.mCriteria[j].mRule &&
-                    mix.mCriteria[j].mAttr.mSource == inputSource) ||
-               (RULE_EXCLUDE_ATTRIBUTE_CAPTURE_PRESET == mix.mCriteria[j].mRule &&
-                    mix.mCriteria[j].mAttr.mSource != inputSource)) {
+        for (size_t j = 0; j < mix->mCriteria.size(); j++) {
+            if ((RULE_MATCH_ATTRIBUTE_CAPTURE_PRESET == mix->mCriteria[j].mRule &&
+                    mix->mCriteria[j].mAttr.mSource == inputSource) ||
+               (RULE_EXCLUDE_ATTRIBUTE_CAPTURE_PRESET == mix->mCriteria[j].mRule &&
+                    mix->mCriteria[j].mAttr.mSource != inputSource)) {
                 if (availDevices & AUDIO_DEVICE_IN_REMOTE_SUBMIX) {
                     if (policyMix != NULL) {
-                        *policyMix = &mix;
+                        *policyMix = mix;
                     }
                     return AUDIO_DEVICE_IN_REMOTE_SUBMIX;
                 }
@@ -167,7 +167,7 @@
     return AUDIO_DEVICE_NONE;
 }
 
-status_t AudioPolicyMixCollection::getInputMixForAttr(audio_attributes_t attr, AudioMix *&policyMix)
+status_t AudioPolicyMixCollection::getInputMixForAttr(audio_attributes_t attr, AudioMix **policyMix)
 {
     if (strncmp(attr.tags, "addr=", strlen("addr=")) != 0) {
         return BAD_VALUE;
@@ -180,13 +180,13 @@
         return BAD_VALUE;
     }
     sp<AudioPolicyMix> audioPolicyMix = valueAt(index);
-    AudioMix mix = audioPolicyMix->getMix();
+    AudioMix *mix = audioPolicyMix->getMix();
 
-    if (mix.mMixType != MIX_TYPE_PLAYERS) {
+    if (mix->mMixType != MIX_TYPE_PLAYERS) {
         ALOGW("getInputForAttr() bad policy mix type for address %s", address.string());
         return BAD_VALUE;
     }
-    policyMix = &mix;
+    *policyMix = mix;
     return NO_ERROR;
 }
 
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp
index e8191dd..f3978ec 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp
@@ -16,7 +16,7 @@
 
 #define LOG_TAG "APM::AudioPort"
 //#define LOG_NDEBUG 0
-
+#include <media/AudioResamplerPublic.h>
 #include "AudioPort.h"
 #include "HwModule.h"
 #include "AudioGain.h"
@@ -216,6 +216,7 @@
         }
         str = strtok(NULL, "|");
     }
+    mFormats.sort(compareFormatsGoodToBad);
 }
 
 void AudioPort::loadInChannels(char *name)
@@ -358,6 +359,9 @@
         uint32_t *updatedSamplingRate) const
 {
     if (mSamplingRates.isEmpty()) {
+        if (updatedSamplingRate != NULL) {
+            *updatedSamplingRate = samplingRate;
+        }
         return NO_ERROR;
     }
 
@@ -387,16 +391,11 @@
             }
         }
     }
-    // This uses hard-coded knowledge about AudioFlinger resampling ratios.
-    // TODO Move these assumptions out.
-    static const uint32_t kMaxDownSampleRatio = 6;  // beyond this aliasing occurs
-    static const uint32_t kMaxUpSampleRatio = 256;  // beyond this sample rate inaccuracies occur
-                                                    // due to approximation by an int32_t of the
-                                                    // phase increments
+
     // Prefer to down-sample from a higher sampling rate, as we get the desired frequency spectrum.
     if (minAbove >= 0) {
         candidate = mSamplingRates[minAbove];
-        if (candidate / kMaxDownSampleRatio <= samplingRate) {
+        if (candidate / AUDIO_RESAMPLER_DOWN_RATIO_MAX <= samplingRate) {
             if (updatedSamplingRate != NULL) {
                 *updatedSamplingRate = candidate;
             }
@@ -406,7 +405,7 @@
     // But if we have to up-sample from a lower sampling rate, that's OK.
     if (maxBelow >= 0) {
         candidate = mSamplingRates[maxBelow];
-        if (candidate * kMaxUpSampleRatio >= samplingRate) {
+        if (candidate * AUDIO_RESAMPLER_UP_RATIO_MAX >= samplingRate) {
             if (updatedSamplingRate != NULL) {
                 *updatedSamplingRate = candidate;
             }
@@ -431,10 +430,13 @@
     return BAD_VALUE;
 }
 
-status_t AudioPort::checkCompatibleChannelMask(audio_channel_mask_t channelMask)
-        const
+status_t AudioPort::checkCompatibleChannelMask(audio_channel_mask_t channelMask,
+        audio_channel_mask_t *updatedChannelMask) const
 {
     if (mChannelMasks.isEmpty()) {
+        if (updatedChannelMask != NULL) {
+            *updatedChannelMask = channelMask;
+        }
         return NO_ERROR;
     }
 
@@ -443,6 +445,9 @@
         // FIXME Does not handle multi-channel automatic conversions yet
         audio_channel_mask_t supported = mChannelMasks[i];
         if (supported == channelMask) {
+            if (updatedChannelMask != NULL) {
+                *updatedChannelMask = channelMask;
+            }
             return NO_ERROR;
         }
         if (isRecordThread) {
@@ -452,6 +457,9 @@
                     && channelMask == AUDIO_CHANNEL_IN_MONO) ||
                 (supported == AUDIO_CHANNEL_IN_MONO && (channelMask == AUDIO_CHANNEL_IN_FRONT_BACK
                     || channelMask == AUDIO_CHANNEL_IN_STEREO))) {
+                if (updatedChannelMask != NULL) {
+                    *updatedChannelMask = supported;
+                }
                 return NO_ERROR;
             }
         }
@@ -459,7 +467,7 @@
     return BAD_VALUE;
 }
 
-status_t AudioPort::checkFormat(audio_format_t format) const
+status_t AudioPort::checkExactFormat(audio_format_t format) const
 {
     if (mFormats.isEmpty()) {
         return NO_ERROR;
@@ -473,6 +481,33 @@
     return BAD_VALUE;
 }
 
+status_t AudioPort::checkCompatibleFormat(audio_format_t format, audio_format_t *updatedFormat)
+        const
+{
+    if (mFormats.isEmpty()) {
+        if (updatedFormat != NULL) {
+            *updatedFormat = format;
+        }
+        return NO_ERROR;
+    }
+
+    const bool checkInexact = // when port is input and format is linear pcm
+            mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SINK
+            && audio_is_linear_pcm(format);
+
+    for (size_t i = 0; i < mFormats.size(); ++i) {
+        if (mFormats[i] == format ||
+                (checkInexact && audio_is_linear_pcm(mFormats[i]))) {
+            // for inexact checks we take the first linear pcm format since
+            // mFormats is sorted from best PCM format to worst PCM format.
+            if (updatedFormat != NULL) {
+                *updatedFormat = mFormats[i];
+            }
+            return NO_ERROR;
+        }
+    }
+    return BAD_VALUE;
+}
 
 uint32_t AudioPort::pickSamplingRate() const
 {
@@ -756,7 +791,7 @@
         mChannelMask = config->channel_mask;
     }
     if (config->config_mask & AUDIO_PORT_CONFIG_FORMAT) {
-        status = audioport->checkFormat(config->format);
+        status = audioport->checkExactFormat(config->format);
         if (status != NO_ERROR) {
             goto exit;
         }
diff --git a/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp b/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp
index de6539c..7b6d51d 100644
--- a/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp
@@ -40,7 +40,9 @@
                                     uint32_t samplingRate,
                                     uint32_t *updatedSamplingRate,
                                     audio_format_t format,
+                                    audio_format_t *updatedFormat,
                                     audio_channel_mask_t channelMask,
+                                    audio_channel_mask_t *updatedChannelMask,
                                     uint32_t flags) const
 {
     const bool isPlaybackThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SOURCE;
@@ -71,7 +73,14 @@
          return false;
     }
 
-    if (!audio_is_valid_format(format) || checkFormat(format) != NO_ERROR) {
+    if (!audio_is_valid_format(format)) {
+        return false;
+    }
+    if (isPlaybackThread && checkExactFormat(format) != NO_ERROR) {
+        return false;
+    }
+    audio_format_t myUpdatedFormat = format;
+    if (isRecordThread && checkCompatibleFormat(format, &myUpdatedFormat) != NO_ERROR) {
         return false;
     }
 
@@ -79,8 +88,9 @@
             checkExactChannelMask(channelMask) != NO_ERROR)) {
         return false;
     }
+    audio_channel_mask_t myUpdatedChannelMask = channelMask;
     if (isRecordThread && (!audio_is_input_channel(channelMask) ||
-            checkCompatibleChannelMask(channelMask) != NO_ERROR)) {
+            checkCompatibleChannelMask(channelMask, &myUpdatedChannelMask) != NO_ERROR)) {
         return false;
     }
 
@@ -99,6 +109,12 @@
     if (updatedSamplingRate != NULL) {
         *updatedSamplingRate = myUpdatedSamplingRate;
     }
+    if (updatedFormat != NULL) {
+        *updatedFormat = myUpdatedFormat;
+    }
+    if (updatedChannelMask != NULL) {
+        *updatedChannelMask = myUpdatedChannelMask;
+    }
     return true;
 }
 
diff --git a/services/audiopolicy/engine/interface/AudioPolicyManagerInterface.h b/services/audiopolicy/engine/interface/AudioPolicyManagerInterface.h
index eadaa77..db0573f 100755
--- a/services/audiopolicy/engine/interface/AudioPolicyManagerInterface.h
+++ b/services/audiopolicy/engine/interface/AudioPolicyManagerInterface.h
@@ -134,16 +134,16 @@
                                               audio_policy_dev_state_t state) = 0;
 
     /**
-     * Translate a volume index given by the UI to an amplification value for a stream type
+     * Translate a volume index given by the UI to an amplification value in dB for a stream type
      * and a device category.
      *
      * @param[in] deviceCategory for which the conversion is requested.
      * @param[in] stream type for which the conversion is requested.
      * @param[in] indexInUi index received from the UI to be translated.
      *
-     * @return amplification value matching the UI index for this given device and stream.
+     * @return amplification value in dB matching the UI index for this given device and stream.
      */
-    virtual float volIndexToAmpl(Volume::device_category deviceCategory, audio_stream_type_t stream,
+    virtual float volIndexToDb(Volume::device_category deviceCategory, audio_stream_type_t stream,
                                  int indexInUi) = 0;
 
     /**
diff --git a/services/audiopolicy/engine/interface/AudioPolicyManagerObserver.h b/services/audiopolicy/engine/interface/AudioPolicyManagerObserver.h
index 4f5427e..6d43df2 100755
--- a/services/audiopolicy/engine/interface/AudioPolicyManagerObserver.h
+++ b/services/audiopolicy/engine/interface/AudioPolicyManagerObserver.h
@@ -43,7 +43,7 @@
 
     virtual const AudioPolicyMixCollection &getAudioPolicyMixCollection() const = 0;
 
-    virtual const AudioOutputCollection &getOutputs() const = 0;
+    virtual const SwAudioOutputCollection &getOutputs() const = 0;
 
     virtual const AudioInputCollection &getInputs() const = 0;
 
diff --git a/services/audiopolicy/enginedefault/src/Engine.cpp b/services/audiopolicy/enginedefault/src/Engine.cpp
index 417eebc..50f1609 100755
--- a/services/audiopolicy/enginedefault/src/Engine.cpp
+++ b/services/audiopolicy/enginedefault/src/Engine.cpp
@@ -63,13 +63,14 @@
     return (mApmObserver != NULL) ?  NO_ERROR : NO_INIT;
 }
 
-float Engine::volIndexToAmpl(Volume::device_category category, audio_stream_type_t streamType,
+float Engine::volIndexToDb(Volume::device_category category, audio_stream_type_t streamType,
                              int indexInUi)
 {
     const StreamDescriptor &streamDesc = mApmObserver->getStreamDescriptors().valueAt(streamType);
-    return Gains::volIndexToAmpl(category, streamDesc, indexInUi);
+    return Gains::volIndexToDb(category, streamDesc, indexInUi);
 }
 
+
 status_t Engine::initStreamVolume(audio_stream_type_t stream, int indexMin, int indexMax)
 {
     ALOGV("initStreamVolume() stream %d, min %d, max %d", stream , indexMin, indexMax);
@@ -243,7 +244,7 @@
 
 routing_strategy Engine::getStrategyForUsage(audio_usage_t usage)
 {
-    const AudioOutputCollection &outputs = mApmObserver->getOutputs();
+    const SwAudioOutputCollection &outputs = mApmObserver->getOutputs();
 
     // usage to strategy mapping
     switch (usage) {
@@ -291,7 +292,7 @@
     const DeviceVector &availableOutputDevices = mApmObserver->getAvailableOutputDevices();
     const DeviceVector &availableInputDevices = mApmObserver->getAvailableInputDevices();
 
-    const AudioOutputCollection &outputs = mApmObserver->getOutputs();
+    const SwAudioOutputCollection &outputs = mApmObserver->getOutputs();
 
     uint32_t device = AUDIO_DEVICE_NONE;
     uint32_t availableOutputDevicesType = availableOutputDevices.types();
@@ -582,7 +583,7 @@
 {
     const DeviceVector &availableOutputDevices = mApmObserver->getAvailableOutputDevices();
     const DeviceVector &availableInputDevices = mApmObserver->getAvailableInputDevices();
-    const AudioOutputCollection &outputs = mApmObserver->getOutputs();
+    const SwAudioOutputCollection &outputs = mApmObserver->getOutputs();
     audio_devices_t availableDeviceTypes = availableInputDevices.types() & ~AUDIO_DEVICE_BIT_IN;
 
     uint32_t device = AUDIO_DEVICE_NONE;
diff --git a/services/audiopolicy/enginedefault/src/Engine.h b/services/audiopolicy/enginedefault/src/Engine.h
index f44556c..56a4748 100755
--- a/services/audiopolicy/enginedefault/src/Engine.h
+++ b/services/audiopolicy/enginedefault/src/Engine.h
@@ -101,10 +101,10 @@
         {
             return mPolicyEngine->initializeVolumeCurves(isSpeakerDrcEnabled);
         }
-        virtual float volIndexToAmpl(Volume::device_category deviceCategory,
+        virtual float volIndexToDb(Volume::device_category deviceCategory,
                                      audio_stream_type_t stream,int indexInUi)
         {
-            return mPolicyEngine->volIndexToAmpl(deviceCategory, stream, indexInUi);
+            return mPolicyEngine->volIndexToDb(deviceCategory, stream, indexInUi);
         }
     private:
         Engine *mPolicyEngine;
@@ -141,7 +141,7 @@
     audio_devices_t getDeviceForStrategy(routing_strategy strategy) const;
     audio_devices_t getDeviceForInputSource(audio_source_t inputSource) const;
 
-    float volIndexToAmpl(Volume::device_category category,
+    float volIndexToDb(Volume::device_category category,
                          audio_stream_type_t stream, int indexInUi);
     status_t initStreamVolume(audio_stream_type_t stream, int indexMin, int indexMax);
     void initializeVolumeCurves(bool isSpeakerDrcEnabled);
diff --git a/services/audiopolicy/enginedefault/src/Gains.cpp b/services/audiopolicy/enginedefault/src/Gains.cpp
index a684fdd..78f2909 100644
--- a/services/audiopolicy/enginedefault/src/Gains.cpp
+++ b/services/audiopolicy/enginedefault/src/Gains.cpp
@@ -197,10 +197,10 @@
 };
 
 //static
-float Gains::volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc,
-        int indexInUi)
+float Gains::volIndexToDb(Volume::device_category deviceCategory,
+                          const StreamDescriptor& streamDesc,
+                          int indexInUi)
 {
-    Volume::device_category deviceCategory = Volume::getDeviceCategory(device);
     const VolumeCurvePoint *curve = streamDesc.getVolumeCurvePoint(deviceCategory);
 
     // the volume index in the UI is relative to the min and max volume indices for this stream type
@@ -212,7 +212,7 @@
     // find what part of the curve this index volume belongs to, or if it's out of bounds
     int segment = 0;
     if (volIdx < curve[Volume::VOLMIN].mIndex) {         // out of bounds
-        return 0.0f;
+        return VOLUME_MIN_DB;
     } else if (volIdx < curve[Volume::VOLKNEE1].mIndex) {
         segment = 0;
     } else if (volIdx < curve[Volume::VOLKNEE2].mIndex) {
@@ -220,7 +220,7 @@
     } else if (volIdx <= curve[Volume::VOLMAX].mIndex) {
         segment = 2;
     } else {                                                               // out of bounds
-        return 1.0f;
+        return 0.0f;
     }
 
     // linear interpolation in the attenuation table in dB
@@ -231,17 +231,25 @@
                     ((float)(curve[segment+1].mIndex -
                             curve[segment].mIndex)) );
 
-    float amplification = exp( decibels * 0.115129f); // exp( dB * ln(10) / 20 )
-
-    ALOGVV("VOLUME vol index=[%d %d %d], dB=[%.1f %.1f %.1f] ampl=%.5f",
+    ALOGVV("VOLUME vol index=[%d %d %d], dB=[%.1f %.1f %.1f]",
             curve[segment].mIndex, volIdx,
             curve[segment+1].mIndex,
             curve[segment].mDBAttenuation,
             decibels,
-            curve[segment+1].mDBAttenuation,
-            amplification);
+            curve[segment+1].mDBAttenuation);
 
-    return amplification;
+    return decibels;
 }
 
+
+//static
+float Gains::volIndexToAmpl(Volume::device_category deviceCategory,
+                            const StreamDescriptor& streamDesc,
+                            int indexInUi)
+{
+    return Volume::DbToAmpl(volIndexToDb(deviceCategory, streamDesc, indexInUi));
+}
+
+
+
 }; // namespace android
diff --git a/services/audiopolicy/enginedefault/src/Gains.h b/services/audiopolicy/enginedefault/src/Gains.h
index b5601ca..7620b7d 100644
--- a/services/audiopolicy/enginedefault/src/Gains.h
+++ b/services/audiopolicy/enginedefault/src/Gains.h
@@ -29,8 +29,13 @@
 class Gains
 {
 public :
-    static float volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc,
-                    int indexInUi);
+    static float volIndexToAmpl(Volume::device_category deviceCategory,
+                                const StreamDescriptor& streamDesc,
+                                int indexInUi);
+
+    static float volIndexToDb(Volume::device_category deviceCategory,
+                              const StreamDescriptor& streamDesc,
+                              int indexInUi);
 
     // default volume curve
     static const VolumeCurvePoint sDefaultVolumeCurve[Volume::VOLCNT];
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index ffa689a..ba9f996 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -157,7 +157,7 @@
         // outputs must be closed after checkOutputForAllStrategies() is executed
         if (!outputs.isEmpty()) {
             for (size_t i = 0; i < outputs.size(); i++) {
-                sp<AudioOutputDescriptor> desc = mOutputs.valueFor(outputs[i]);
+                sp<SwAudioOutputDescriptor> desc = mOutputs.valueFor(outputs[i]);
                 // close unused outputs after device disconnection or direct outputs that have been
                 // opened by checkOutputsForDevice() to query dynamic parameters
                 if ((state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) ||
@@ -176,18 +176,17 @@
             updateCallRouting(newDevice);
         }
         for (size_t i = 0; i < mOutputs.size(); i++) {
-            audio_io_handle_t output = mOutputs.keyAt(i);
-            if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || (output != mPrimaryOutput)) {
-                audio_devices_t newDevice = getNewOutputDevice(mOutputs.keyAt(i),
-                                                               true /*fromCache*/);
+            sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
+            if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || (desc != mPrimaryOutput)) {
+                audio_devices_t newDevice = getNewOutputDevice(desc, true /*fromCache*/);
                 // do not force device change on duplicated output because if device is 0, it will
                 // also force a device 0 for the two outputs it is duplicated to which may override
                 // a valid device selection on those outputs.
-                bool force = !mOutputs.valueAt(i)->isDuplicated()
+                bool force = !desc->isDuplicated()
                         && (!device_distinguishes_on_address(device)
                                 // always force when disconnecting (a non-duplicated device)
                                 || (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE));
-                setOutputDevice(output, newDevice, force, 0);
+                setOutputDevice(desc, newDevice, force, 0);
             }
         }
 
@@ -349,7 +348,7 @@
                                                 AUDIO_OUTPUT_FLAG_NONE,
                                                 AUDIO_FORMAT_INVALID);
         if (output != AUDIO_IO_HANDLE_NONE) {
-            sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
+            sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
             ALOG_ASSERT(!outputDesc->isDuplicated(),
                         "updateCallRouting() RX device output is duplicated");
             outputDesc->toAudioPortConfig(&patch.sources[1]);
@@ -450,13 +449,13 @@
     checkOutputForAllStrategies();
     updateDevicesAndOutputs();
 
-    sp<AudioOutputDescriptor> hwOutputDesc = mOutputs.valueFor(mPrimaryOutput);
+    sp<SwAudioOutputDescriptor> hwOutputDesc = mPrimaryOutput;
 
     int delayMs = 0;
     if (isStateInCall(state)) {
         nsecs_t sysTime = systemTime();
         for (size_t i = 0; i < mOutputs.size(); i++) {
-            sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
+            sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
             // mute media and sonification strategies and delay device switch by the largest
             // latency of any output where either strategy is active.
             // This avoid sending the ring tone or music tail into the earpiece or headset.
@@ -466,14 +465,14 @@
                  isStrategyActive(desc, STRATEGY_SONIFICATION,
                                   SONIFICATION_HEADSET_MUSIC_DELAY,
                                   sysTime)) &&
-                    (delayMs < (int)desc->mLatency*2)) {
-                delayMs = desc->mLatency*2;
+                    (delayMs < (int)desc->latency()*2)) {
+                delayMs = desc->latency()*2;
             }
-            setStrategyMute(STRATEGY_MEDIA, true, mOutputs.keyAt(i));
-            setStrategyMute(STRATEGY_MEDIA, false, mOutputs.keyAt(i), MUTE_TIME_MS,
+            setStrategyMute(STRATEGY_MEDIA, true, desc);
+            setStrategyMute(STRATEGY_MEDIA, false, desc, MUTE_TIME_MS,
                 getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/));
-            setStrategyMute(STRATEGY_SONIFICATION, true, mOutputs.keyAt(i));
-            setStrategyMute(STRATEGY_SONIFICATION, false, mOutputs.keyAt(i), MUTE_TIME_MS,
+            setStrategyMute(STRATEGY_SONIFICATION, true, desc);
+            setStrategyMute(STRATEGY_SONIFICATION, false, desc, MUTE_TIME_MS,
                 getDeviceForStrategy(STRATEGY_SONIFICATION, true /*fromCache*/));
         }
     }
@@ -549,13 +548,13 @@
         updateCallRouting(newDevice);
     }
     for (size_t i = 0; i < mOutputs.size(); i++) {
-        audio_io_handle_t output = mOutputs.keyAt(i);
-        audio_devices_t newDevice = getNewOutputDevice(output, true /*fromCache*/);
-        if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || (output != mPrimaryOutput)) {
-            setOutputDevice(output, newDevice, (newDevice != AUDIO_DEVICE_NONE));
+        sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
+        audio_devices_t newDevice = getNewOutputDevice(outputDesc, true /*fromCache*/);
+        if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || (outputDesc != mPrimaryOutput)) {
+            setOutputDevice(outputDesc, newDevice, (newDevice != AUDIO_DEVICE_NONE));
         }
         if (forceVolumeReeval && (newDevice != AUDIO_DEVICE_NONE)) {
-            applyStreamVolumes(output, newDevice, 0, true);
+            applyStreamVolumes(outputDesc, newDevice, 0, true);
         }
     }
 
@@ -586,8 +585,10 @@
         }
         for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) {
             sp<IOProfile> profile = mHwModules[i]->mOutputProfiles[j];
-            bool found = profile->isCompatibleProfile(device, String8(""), samplingRate,
-                    NULL /*updatedSamplingRate*/, format, channelMask,
+            bool found = profile->isCompatibleProfile(device, String8(""),
+                    samplingRate, NULL /*updatedSamplingRate*/,
+                    format, NULL /*updatedFormat*/,
+                    channelMask, NULL /*updatedChannelMask*/,
                     flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD ?
                         AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD : AUDIO_OUTPUT_FLAG_DIRECT);
             if (found && (mAvailableOutputDevices.types() & profile->mSupportedDevices.types())) {
@@ -642,7 +643,7 @@
         }
         stream_type_to_audio_attributes(*stream, &attributes);
     }
-    sp<AudioOutputDescriptor> desc;
+    sp<SwAudioOutputDescriptor> desc;
     if (mPolicyMixes.getOutputForAttr(attributes, desc) == NO_ERROR) {
         ALOG_ASSERT(desc != 0, "Invalid desc returned by getOutputForAttr");
         if (!audio_is_linear_pcm(format)) {
@@ -713,7 +714,8 @@
 
         if (mTestOutputs[mCurOutput] == 0) {
             ALOGV("getOutput() opening test output");
-            sp<AudioOutputDescriptor> outputDesc = new AudioOutputDescriptor(NULL);
+            sp<AudioOutputDescriptor> outputDesc = new SwAudioOutputDescriptor(NULL,
+                                                                               mpClientInterface);
             outputDesc->mDevice = mTestDevice;
             outputDesc->mLatency = mTestLatencyMs;
             outputDesc->mFlags =
@@ -789,10 +791,10 @@
     }
 
     if (profile != 0) {
-        sp<AudioOutputDescriptor> outputDesc = NULL;
+        sp<SwAudioOutputDescriptor> outputDesc = NULL;
 
         for (size_t i = 0; i < mOutputs.size(); i++) {
-            sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
+            sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
             if (!desc->isDuplicated() && (profile == desc->mProfile)) {
                 outputDesc = desc;
                 // reuse direct output if currently open and configured with same parameters
@@ -809,7 +811,7 @@
         if (outputDesc != NULL) {
             closeOutput(outputDesc->mIoHandle);
         }
-        outputDesc = new AudioOutputDescriptor(profile);
+        outputDesc = new SwAudioOutputDescriptor(profile, mpClientInterface);
         outputDesc->mDevice = device;
         outputDesc->mLatency = 0;
         outputDesc->mFlags =(audio_output_flags_t) (outputDesc->mFlags | flags);
@@ -915,7 +917,7 @@
     audio_io_handle_t outputPrimary = 0;
 
     for (size_t i = 0; i < outputs.size(); i++) {
-        sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(outputs[i]);
+        sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueFor(outputs[i]);
         if (!outputDesc->isDuplicated()) {
             // if a valid format is specified, skip output if not compatible
             if (format != AUDIO_FORMAT_INVALID) {
@@ -962,8 +964,51 @@
         return BAD_VALUE;
     }
 
+    sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(index);
+
+    audio_devices_t newDevice;
+    if (outputDesc->mPolicyMix != NULL) {
+        newDevice = AUDIO_DEVICE_OUT_REMOTE_SUBMIX;
+    } else {
+        newDevice = AUDIO_DEVICE_NONE;
+    }
+
+    uint32_t delayMs = 0;
+
+    // Routing?
+    mOutputRoutes.incRouteActivity(session);
+
+    status_t status = startSource(outputDesc, stream, newDevice, &delayMs);
+
+    if (status != NO_ERROR) {
+        mOutputRoutes.decRouteActivity(session);
+    }
+    // Automatically enable the remote submix input when output is started on a re routing mix
+    // of type MIX_TYPE_RECORDERS
+    if (audio_is_remote_submix_device(newDevice) && outputDesc->mPolicyMix != NULL &&
+            outputDesc->mPolicyMix->mMixType == MIX_TYPE_RECORDERS) {
+            setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
+                    AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
+                    outputDesc->mPolicyMix->mRegistrationId,
+                    "remote-submix");
+    }
+
+    if (delayMs != 0) {
+        usleep(delayMs * 1000);
+    }
+
+    return status;
+}
+
+status_t AudioPolicyManager::startSource(sp<AudioOutputDescriptor> outputDesc,
+                                             audio_stream_type_t stream,
+                                             audio_devices_t device,
+                                             uint32_t *delayMs)
+{
     // cannot start playback of STREAM_TTS if any other output is being used
     uint32_t beaconMuteLatency = 0;
+
+    *delayMs = 0;
     if (stream == AUDIO_STREAM_TTS) {
         ALOGV("\t found BEACON stream");
         if (mOutputs.isAnyOutputActive(AUDIO_STREAM_TTS /*streamToIgnore*/)) {
@@ -976,22 +1021,15 @@
         beaconMuteLatency = handleEventForBeacon(STARTING_OUTPUT);
     }
 
-    sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(index);
     // increment usage count for this stream on the requested output:
     // NOTE that the usage count is the same for duplicated output and hardware output which is
     // necessary for a correct control of hardware output routing by startOutput() and stopOutput()
     outputDesc->changeRefCount(stream, 1);
 
-    // Routing?
-    mOutputRoutes.incRouteActivity(session);
-
     if (outputDesc->mRefCount[stream] == 1) {
         // starting an output being rerouted?
-        audio_devices_t newDevice;
-        if (outputDesc->mPolicyMix != NULL) {
-            newDevice = AUDIO_DEVICE_OUT_REMOTE_SUBMIX;
-        } else {
-            newDevice = getNewOutputDevice(output, false /*fromCache*/);
+        if (device == AUDIO_DEVICE_NONE) {
+            device = getNewOutputDevice(outputDesc, false /*fromCache*/);
         }
         routing_strategy strategy = getStrategy(stream);
         bool shouldWait = (strategy == STRATEGY_SONIFICATION) ||
@@ -1007,7 +1045,7 @@
                 // In this case, the audio HAL must receive the new device selection so that it can
                 // change the device currently selected by the other active output.
                 if (outputDesc->sharesHwModuleWith(desc) &&
-                    desc->device() != newDevice) {
+                    desc->device() != device) {
                     force = true;
                 }
                 // wait for audio on other active outputs to be presented when starting
@@ -1019,7 +1057,7 @@
                 }
             }
         }
-        uint32_t muteWaitMs = setOutputDevice(output, newDevice, force);
+        uint32_t muteWaitMs = setOutputDevice(outputDesc, device, force);
 
         // handle special case for sonification while in call
         if (isInCall()) {
@@ -1028,32 +1066,18 @@
 
         // apply volume rules for current stream and device if necessary
         checkAndSetVolume(stream,
-                          mStreams[stream].getVolumeIndex(newDevice),
-                          output,
-                          newDevice);
+                          mStreams.valueFor(stream).getVolumeIndex(device),
+                          outputDesc,
+                          device);
 
         // update the outputs if starting an output with a stream that can affect notification
         // routing
         handleNotificationRoutingForStream(stream);
 
-        // Automatically enable the remote submix input when output is started on a re routing mix
-        // of type MIX_TYPE_RECORDERS
-        if (audio_is_remote_submix_device(newDevice) && outputDesc->mPolicyMix != NULL &&
-                outputDesc->mPolicyMix->mMixType == MIX_TYPE_RECORDERS) {
-                setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
-                        AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
-                        outputDesc->mPolicyMix->mRegistrationId,
-                        "remote-submix");
-        }
-
         // force reevaluating accessibility routing when ringtone or alarm starts
         if (strategy == STRATEGY_SONIFICATION) {
             mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY);
         }
-
-        if (waitMs > muteWaitMs) {
-            usleep((waitMs - muteWaitMs) * 2 * 1000);
-        }
     }
     return NO_ERROR;
 }
@@ -1070,8 +1094,32 @@
         return BAD_VALUE;
     }
 
-    sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(index);
+    sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(index);
 
+    if (outputDesc->mRefCount[stream] == 1) {
+        // Automatically disable the remote submix input when output is stopped on a
+        // re routing mix of type MIX_TYPE_RECORDERS
+        if (audio_is_remote_submix_device(outputDesc->mDevice) &&
+                outputDesc->mPolicyMix != NULL &&
+                outputDesc->mPolicyMix->mMixType == MIX_TYPE_RECORDERS) {
+            setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
+                    AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
+                    outputDesc->mPolicyMix->mRegistrationId,
+                    "remote-submix");
+        }
+    }
+
+    // Routing?
+    if (outputDesc->mRefCount[stream] > 0) {
+        mOutputRoutes.decRouteActivity(session);
+    }
+
+    return stopSource(outputDesc, stream);
+}
+
+status_t AudioPolicyManager::stopSource(sp<AudioOutputDescriptor> outputDesc,
+                                            audio_stream_type_t stream)
+{
     // always handle stream stop, check which stream type is stopping
     handleEventForBeacon(stream == AUDIO_STREAM_TTS ? STOPPING_BEACON : STOPPING_OUTPUT);
 
@@ -1084,44 +1132,30 @@
         // decrement usage count of this stream on the output
         outputDesc->changeRefCount(stream, -1);
 
-        // Routing?
-        mOutputRoutes.decRouteActivity(session);
-
         // store time at which the stream was stopped - see isStreamActive()
         if (outputDesc->mRefCount[stream] == 0) {
-            // Automatically disable the remote submix input when output is stopped on a
-            // re routing mix of type MIX_TYPE_RECORDERS
-            if (audio_is_remote_submix_device(outputDesc->mDevice) &&
-                    outputDesc->mPolicyMix != NULL &&
-                    outputDesc->mPolicyMix->mMixType == MIX_TYPE_RECORDERS) {
-                setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
-                        AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
-                        outputDesc->mPolicyMix->mRegistrationId,
-                        "remote-submix");
-            }
-
             outputDesc->mStopTime[stream] = systemTime();
-            audio_devices_t newDevice = getNewOutputDevice(output, false /*fromCache*/);
+            audio_devices_t newDevice = getNewOutputDevice(outputDesc, false /*fromCache*/);
             // delay the device switch by twice the latency because stopOutput() is executed when
             // the track stop() command is received and at that time the audio track buffer can
             // still contain data that needs to be drained. The latency only covers the audio HAL
             // and kernel buffers. Also the latency does not always include additional delay in the
             // audio path (audio DSP, CODEC ...)
-            setOutputDevice(output, newDevice, false, outputDesc->mLatency*2);
+            setOutputDevice(outputDesc, newDevice, false, outputDesc->latency()*2);
 
             // force restoring the device selection on other active outputs if it differs from the
             // one being selected for this output
             for (size_t i = 0; i < mOutputs.size(); i++) {
                 audio_io_handle_t curOutput = mOutputs.keyAt(i);
                 sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
-                if (curOutput != output &&
+                if (desc != outputDesc &&
                         desc->isActive() &&
                         outputDesc->sharesHwModuleWith(desc) &&
                         (newDevice != desc->device())) {
-                    setOutputDevice(curOutput,
-                                    getNewOutputDevice(curOutput, false /*fromCache*/),
+                    setOutputDevice(desc,
+                                    getNewOutputDevice(desc, false /*fromCache*/),
                                     true,
-                                    outputDesc->mLatency*2);
+                                    outputDesc->latency()*2);
                 }
             }
             // update the outputs if stopping one with a stream that can affect notification routing
@@ -1129,7 +1163,7 @@
         }
         return NO_ERROR;
     } else {
-        ALOGW("stopOutput() refcount is already 0 for output %d", output);
+        ALOGW("stopOutput() refcount is already 0");
         return INVALID_OPERATION;
     }
 }
@@ -1161,7 +1195,7 @@
     // Routing
     mOutputRoutes.removeRoute(session);
 
-    sp<AudioOutputDescriptor> desc = mOutputs.valueAt(index);
+    sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(index);
     if (desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {
         if (desc->mDirectOpenCount <= 0) {
             ALOGW("releaseOutput() invalid open count %d for output %d",
@@ -1173,8 +1207,9 @@
             // If effects where present on the output, audioflinger moved them to the primary
             // output by default: move them back to the appropriate output.
             audio_io_handle_t dstOutput = getOutputForEffect();
-            if (dstOutput != mPrimaryOutput) {
-                mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, mPrimaryOutput, dstOutput);
+            if (dstOutput != mPrimaryOutput->mIoHandle) {
+                mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX,
+                                               mPrimaryOutput->mIoHandle, dstOutput);
             }
             mpClientInterface->onAudioPortListUpdate();
         }
@@ -1212,7 +1247,7 @@
 
     if (inputSource == AUDIO_SOURCE_REMOTE_SUBMIX &&
             strncmp(attr->tags, "addr=", strlen("addr=")) == 0) {
-        status_t ret = mPolicyMixes.getInputMixForAttr(*attr, policyMix);
+        status_t ret = mPolicyMixes.getInputMixForAttr(*attr, &policyMix);
         if (ret != NO_ERROR) {
             return ret;
         }
@@ -1270,20 +1305,25 @@
         }
     }
 
-    sp<IOProfile> profile = getInputProfile(device, address,
-                                            samplingRate, format, channelMask,
-                                            flags);
-    if (profile == 0) {
-        //retry without flags
-        audio_input_flags_t log_flags = flags;
-        flags = AUDIO_INPUT_FLAG_NONE;
+    // find a compatible input profile (not necessarily identical in parameters)
+    sp<IOProfile> profile;
+    // samplingRate and flags may be updated by getInputProfile
+    uint32_t profileSamplingRate = samplingRate;
+    audio_format_t profileFormat = format;
+    audio_channel_mask_t profileChannelMask = channelMask;
+    audio_input_flags_t profileFlags = flags;
+    for (;;) {
         profile = getInputProfile(device, address,
-                                  samplingRate, format, channelMask,
-                                  flags);
-        if (profile == 0) {
+                                  profileSamplingRate, profileFormat, profileChannelMask,
+                                  profileFlags);
+        if (profile != 0) {
+            break; // success
+        } else if (profileFlags != AUDIO_INPUT_FLAG_NONE) {
+            profileFlags = AUDIO_INPUT_FLAG_NONE; // retry
+        } else { // fail
             ALOGW("getInputForAttr() could not find profile for device 0x%X, samplingRate %u,"
                     "format %#x, channelMask 0x%X, flags %#x",
-                    device, samplingRate, format, channelMask, log_flags);
+                    device, samplingRate, format, channelMask, flags);
             return BAD_VALUE;
         }
     }
@@ -1294,9 +1334,9 @@
     }
 
     audio_config_t config = AUDIO_CONFIG_INITIALIZER;
-    config.sample_rate = samplingRate;
-    config.channel_mask = channelMask;
-    config.format = format;
+    config.sample_rate = profileSamplingRate;
+    config.channel_mask = profileChannelMask;
+    config.format = profileFormat;
 
     status_t status = mpClientInterface->openInput(profile->getModuleHandle(),
                                                    input,
@@ -1304,14 +1344,15 @@
                                                    &device,
                                                    address,
                                                    halInputSource,
-                                                   flags);
+                                                   profileFlags);
 
     // only accept input with the exact requested set of parameters
     if (status != NO_ERROR || *input == AUDIO_IO_HANDLE_NONE ||
-        (samplingRate != config.sample_rate) ||
-        (format != config.format) ||
-        (channelMask != config.channel_mask)) {
-        ALOGW("getInputForAttr() failed opening input: samplingRate %d, format %d, channelMask %x",
+        (profileSamplingRate != config.sample_rate) ||
+        (profileFormat != config.format) ||
+        (profileChannelMask != config.channel_mask)) {
+        ALOGW("getInputForAttr() failed opening input: samplingRate %d, format %d,"
+                " channelMask %x",
                 samplingRate, format, channelMask);
         if (*input != AUDIO_IO_HANDLE_NONE) {
             mpClientInterface->closeInput(*input);
@@ -1323,15 +1364,15 @@
     inputDesc->mInputSource = inputSource;
     inputDesc->mRefCount = 0;
     inputDesc->mOpenRefCount = 1;
-    inputDesc->mSamplingRate = samplingRate;
-    inputDesc->mFormat = format;
-    inputDesc->mChannelMask = channelMask;
+    inputDesc->mSamplingRate = profileSamplingRate;
+    inputDesc->mFormat = profileFormat;
+    inputDesc->mChannelMask = profileChannelMask;
     inputDesc->mDevice = device;
     inputDesc->mSessions.add(session);
     inputDesc->mIsSoundTrigger = isSoundTrigger;
     inputDesc->mPolicyMix = policyMix;
 
-    ALOGV("getInputForAttr() returns input type = %d", inputType);
+    ALOGV("getInputForAttr() returns input type = %d", *inputType);
 
     addInput(*input, inputDesc);
     mpClientInterface->onAudioPortListUpdate();
@@ -1528,8 +1569,8 @@
                                                   audio_devices_t device)
 {
 
-    if ((index < mStreams[stream].getVolumeIndexMin()) ||
-            (index > mStreams[stream].getVolumeIndexMax())) {
+    if ((index < mStreams.valueFor(stream).getVolumeIndexMin()) ||
+            (index > mStreams.valueFor(stream).getVolumeIndexMax())) {
         return BAD_VALUE;
     }
     if (!audio_is_output_device(device)) {
@@ -1537,7 +1578,7 @@
     }
 
     // Force max volume if stream cannot be muted
-    if (!mStreams.canBeMuted(stream)) index = mStreams[stream].getVolumeIndexMax();
+    if (!mStreams.canBeMuted(stream)) index = mStreams.valueFor(stream).getVolumeIndexMax();
 
     ALOGV("setStreamVolumeIndex() stream %d, device %04x, index %d",
           stream, device, index);
@@ -1566,16 +1607,17 @@
     }
     status_t status = NO_ERROR;
     for (size_t i = 0; i < mOutputs.size(); i++) {
-        audio_devices_t curDevice = Volume::getDeviceForVolume(mOutputs.valueAt(i)->device());
+        sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
+        audio_devices_t curDevice = Volume::getDeviceForVolume(desc->device());
         if ((device == AUDIO_DEVICE_OUT_DEFAULT) || ((curDevice & strategyDevice) != 0)) {
-            status_t volStatus = checkAndSetVolume(stream, index, mOutputs.keyAt(i), curDevice);
+            status_t volStatus = checkAndSetVolume(stream, index, desc, curDevice);
             if (volStatus != NO_ERROR) {
                 status = volStatus;
             }
         }
         if ((device == AUDIO_DEVICE_OUT_DEFAULT) || ((curDevice & accessibilityDevice) != 0)) {
             status_t volStatus = checkAndSetVolume(AUDIO_STREAM_ACCESSIBILITY,
-                                                   index, mOutputs.keyAt(i), curDevice);
+                                                   index, desc, curDevice);
         }
     }
     return status;
@@ -1598,7 +1640,7 @@
     }
     device = Volume::getDeviceForVolume(device);
 
-    *index =  mStreams[stream].getVolumeIndex(device);
+    *index =  mStreams.valueFor(stream).getVolumeIndex(device);
     ALOGV("getStreamVolumeIndex() stream %d device %08x index %d", stream, device, *index);
     return NO_ERROR;
 }
@@ -1622,7 +1664,7 @@
     audio_io_handle_t outputDeepBuffer = 0;
 
     for (size_t i = 0; i < outputs.size(); i++) {
-        sp<AudioOutputDescriptor> desc = mOutputs.valueFor(outputs[i]);
+        sp<SwAudioOutputDescriptor> desc = mOutputs.valueFor(outputs[i]);
         ALOGV("selectOutputForEffects outputs[%zu] flags %x", i, desc->mFlags);
         if ((desc->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
             outputOffloaded = outputs[i];
@@ -1676,6 +1718,16 @@
     return mEffects.registerEffect(desc, io, strategy, session, id);
 }
 
+bool AudioPolicyManager::isStreamActive(audio_stream_type_t stream, uint32_t inPastMs) const
+{
+    return mOutputs.isStreamActive(stream, inPastMs);
+}
+
+bool AudioPolicyManager::isStreamActiveRemotely(audio_stream_type_t stream, uint32_t inPastMs) const
+{
+    return mOutputs.isStreamActiveRemotely(stream, inPastMs);
+}
+
 bool AudioPolicyManager::isSourceActive(audio_source_t source) const
 {
     for (size_t i = 0; i < mInputs.size(); i++) {
@@ -1826,7 +1878,7 @@
     snprintf(buffer, SIZE, "\nAudioPolicyManager Dump: %p\n", this);
     result.append(buffer);
 
-    snprintf(buffer, SIZE, " Primary Output: %d\n", mPrimaryOutput);
+    snprintf(buffer, SIZE, " Primary Output: %d\n", mPrimaryOutput->mIoHandle);
     result.append(buffer);
     snprintf(buffer, SIZE, " Phone state: %d\n", mEngine->getPhoneState());
     result.append(buffer);
@@ -2044,7 +2096,7 @@
     }
 
     if (patch->sources[0].type == AUDIO_PORT_TYPE_MIX) {
-        sp<AudioOutputDescriptor> outputDesc = mOutputs.getOutputFromId(patch->sources[0].id);
+        sp<SwAudioOutputDescriptor> outputDesc = mOutputs.getOutputFromId(patch->sources[0].id);
         if (outputDesc == NULL) {
             ALOGV("createAudioPatch() output not found for id %d", patch->sources[0].id);
             return BAD_VALUE;
@@ -2078,9 +2130,12 @@
                                                            patch->sources[0].sample_rate,
                                                            NULL,  // updatedSamplingRate
                                                            patch->sources[0].format,
+                                                           NULL,  // updatedFormat
                                                            patch->sources[0].channel_mask,
+                                                           NULL,  // updatedChannelMask
                                                            AUDIO_OUTPUT_FLAG_NONE /*FIXME*/)) {
-                ALOGV("createAudioPatch() profile not supported for device %08x", devDesc->type());
+                ALOGV("createAudioPatch() profile not supported for device %08x",
+                        devDesc->type());
                 return INVALID_OPERATION;
             }
             devices.add(devDesc);
@@ -2092,7 +2147,7 @@
         // TODO: reconfigure output format and channels here
         ALOGV("createAudioPatch() setting device %08x on output %d",
               devices.types(), outputDesc->mIoHandle);
-        setOutputDevice(outputDesc->mIoHandle, devices.types(), true, 0, handle);
+        setOutputDevice(outputDesc, devices.types(), true, 0, handle);
         index = mAudioPatches.indexOfKey(*handle);
         if (index >= 0) {
             if (patchDesc != 0 && patchDesc != mAudioPatches.valueAt(index)) {
@@ -2132,7 +2187,9 @@
                                                           patch->sinks[0].sample_rate,
                                                           NULL, /*updatedSampleRate*/
                                                           patch->sinks[0].format,
+                                                          NULL, /*updatedFormat*/
                                                           patch->sinks[0].channel_mask,
+                                                          NULL, /*updatedChannelMask*/
                                                           // FIXME for the parameter type,
                                                           // and the NONE
                                                           (audio_output_flags_t)
@@ -2270,14 +2327,14 @@
     struct audio_patch *patch = &patchDesc->mPatch;
     patchDesc->mUid = mUidCached;
     if (patch->sources[0].type == AUDIO_PORT_TYPE_MIX) {
-        sp<AudioOutputDescriptor> outputDesc = mOutputs.getOutputFromId(patch->sources[0].id);
+        sp<SwAudioOutputDescriptor> outputDesc = mOutputs.getOutputFromId(patch->sources[0].id);
         if (outputDesc == NULL) {
             ALOGV("releaseAudioPatch() output not found for id %d", patch->sources[0].id);
             return BAD_VALUE;
         }
 
-        setOutputDevice(outputDesc->mIoHandle,
-                        getNewOutputDevice(outputDesc->mIoHandle, true /*fromCache*/),
+        setOutputDevice(outputDesc,
+                        getNewOutputDevice(outputDesc, true /*fromCache*/),
                        true,
                        0,
                        NULL);
@@ -2336,7 +2393,7 @@
     sp<AudioPortConfig> audioPortConfig;
     if (config->type == AUDIO_PORT_TYPE_MIX) {
         if (config->role == AUDIO_PORT_ROLE_SOURCE) {
-            sp<AudioOutputDescriptor> outputDesc = mOutputs.getOutputFromId(config->id);
+            sp<SwAudioOutputDescriptor> outputDesc = mOutputs.getOutputFromId(config->id);
             if (outputDesc == NULL) {
                 return BAD_VALUE;
             }
@@ -2418,7 +2475,6 @@
 #ifdef AUDIO_POLICY_TEST
     Thread(false),
 #endif //AUDIO_POLICY_TEST
-    mPrimaryOutput((audio_io_handle_t)0),
     mLimitRingtoneVolume(false), mLastVoiceVolume(-1.0f),
     mA2dpSuspended(false),
     mSpeakerDrcEnabled(false),
@@ -2502,7 +2558,8 @@
             if ((profileType & outputDeviceTypes) == 0) {
                 continue;
             }
-            sp<AudioOutputDescriptor> outputDesc = new AudioOutputDescriptor(outProfile);
+            sp<SwAudioOutputDescriptor> outputDesc = new SwAudioOutputDescriptor(outProfile,
+                                                                                 mpClientInterface);
 
             outputDesc->mDevice = profileType;
             audio_config_t config = AUDIO_CONFIG_INITIALIZER;
@@ -2538,10 +2595,10 @@
                 }
                 if (mPrimaryOutput == 0 &&
                         outProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) {
-                    mPrimaryOutput = output;
+                    mPrimaryOutput = outputDesc;
                 }
                 addOutput(output, outputDesc);
-                setOutputDevice(output,
+                setOutputDevice(outputDesc,
                                 outputDesc->mDevice,
                                 true);
             }
@@ -2648,7 +2705,7 @@
     if (mPrimaryOutput != 0) {
         AudioParameter outputCmd = AudioParameter();
         outputCmd.addInt(String8("set_id"), 0);
-        mpClientInterface->setParameters(mPrimaryOutput, outputCmd.toString());
+        mpClientInterface->setParameters(mPrimaryOutput->mIoHandle, outputCmd.toString());
 
         mTestDevice = AUDIO_DEVICE_OUT_SPEAKER;
         mTestSamplingRate = 44100;
@@ -2788,20 +2845,21 @@
             if (param.get(String8("test_cmd_policy_reopen"), value) == NO_ERROR) {
                 param.remove(String8("test_cmd_policy_reopen"));
 
-                sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(mPrimaryOutput);
-                mpClientInterface->closeOutput(mPrimaryOutput);
+                mpClientInterface->closeOutput(mpClientInterface->closeOutput(mPrimaryOutput););
 
-                audio_module_handle_t moduleHandle = outputDesc->getModuleHandle();
+                audio_module_handle_t moduleHandle = mPrimaryOutput->getModuleHandle();
 
-                removeOutput(mPrimaryOutput);
-                sp<AudioOutputDescriptor> outputDesc = new AudioOutputDescriptor(NULL);
+                removeOutput(mPrimaryOutput->mIoHandle);
+                sp<SwAudioOutputDescriptor> outputDesc = new AudioOutputDescriptor(NULL,
+                                                                               mpClientInterface);
                 outputDesc->mDevice = AUDIO_DEVICE_OUT_SPEAKER;
                 audio_config_t config = AUDIO_CONFIG_INITIALIZER;
                 config.sample_rate = outputDesc->mSamplingRate;
                 config.channel_mask = outputDesc->mChannelMask;
                 config.format = outputDesc->mFormat;
+                audio_io_handle_t handle;
                 status_t status = mpClientInterface->openOutput(moduleHandle,
-                                                                &mPrimaryOutput,
+                                                                &handle,
                                                                 &config,
                                                                 &outputDesc->mDevice,
                                                                 String8(""),
@@ -2815,10 +2873,11 @@
                     outputDesc->mSamplingRate = config.sample_rate;
                     outputDesc->mChannelMask = config.channel_mask;
                     outputDesc->mFormat = config.format;
+                    mPrimaryOutput = outputDesc;
                     AudioParameter outputCmd = AudioParameter();
                     outputCmd.addInt(String8("set_id"), 0);
-                    mpClientInterface->setParameters(mPrimaryOutput, outputCmd.toString());
-                    addOutput(mPrimaryOutput, outputDesc);
+                    mpClientInterface->setParameters(handle, outputCmd.toString());
+                    addOutput(handle, outputDesc);
                 }
             }
 
@@ -2850,7 +2909,7 @@
 
 // ---
 
-void AudioPolicyManager::addOutput(audio_io_handle_t output, sp<AudioOutputDescriptor> outputDesc)
+void AudioPolicyManager::addOutput(audio_io_handle_t output, sp<SwAudioOutputDescriptor> outputDesc)
 {
     outputDesc->setIoHandle(output);
     mOutputs.add(output, outputDesc);
@@ -2869,7 +2928,7 @@
     nextAudioPortGeneration();
 }
 
-void AudioPolicyManager::findIoHandlesByAddress(sp<AudioOutputDescriptor> desc /*in*/,
+void AudioPolicyManager::findIoHandlesByAddress(sp<SwAudioOutputDescriptor> desc /*in*/,
         const audio_devices_t device /*in*/,
         const String8 address /*in*/,
         SortedVector<audio_io_handle_t>& outputs /*out*/) {
@@ -2888,7 +2947,7 @@
                                                    const String8 address)
 {
     audio_devices_t device = devDesc->type();
-    sp<AudioOutputDescriptor> desc;
+    sp<SwAudioOutputDescriptor> desc;
     // erase all current sample rates, formats and channel masks
     devDesc->clearCapabilities();
 
@@ -2896,7 +2955,7 @@
         // first list already open outputs that can be routed to this device
         for (size_t i = 0; i < mOutputs.size(); i++) {
             desc = mOutputs.valueAt(i);
-            if (!desc->isDuplicated() && (desc->mProfile->mSupportedDevices.types() & device)) {
+            if (!desc->isDuplicated() && (desc->supportedDevices() & device)) {
                 if (!device_distinguishes_on_address(device)) {
                     ALOGV("checkOutputsForDevice(): adding opened output %d", mOutputs.keyAt(i));
                     outputs.add(mOutputs.keyAt(i));
@@ -2955,7 +3014,7 @@
 
             ALOGV("opening output for device %08x with params %s profile %p",
                                                       device, address.string(), profile.get());
-            desc = new AudioOutputDescriptor(profile);
+            desc = new SwAudioOutputDescriptor(profile, mpClientInterface);
             desc->mDevice = device;
             audio_config_t config = AUDIO_CONFIG_INITIALIZER;
             config.sample_rate = desc->mSamplingRate;
@@ -3060,7 +3119,7 @@
                                   address.string());
                         }
                         policyMix->setOutput(desc);
-                        desc->mPolicyMix = &(policyMix->getMix());
+                        desc->mPolicyMix = policyMix->getMix();
 
                     } else if ((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) == 0) {
                         // no duplicated output for direct outputs and
@@ -3068,28 +3127,29 @@
                         audio_io_handle_t duplicatedOutput = AUDIO_IO_HANDLE_NONE;
 
                         // set initial stream volume for device
-                        applyStreamVolumes(output, device, 0, true);
+                        applyStreamVolumes(desc, device, 0, true);
 
                         //TODO: configure audio effect output stage here
 
                         // open a duplicating output thread for the new output and the primary output
-                        duplicatedOutput = mpClientInterface->openDuplicateOutput(output,
-                                                                                  mPrimaryOutput);
+                        duplicatedOutput =
+                                mpClientInterface->openDuplicateOutput(output,
+                                                                       mPrimaryOutput->mIoHandle);
                         if (duplicatedOutput != AUDIO_IO_HANDLE_NONE) {
                             // add duplicated output descriptor
-                            sp<AudioOutputDescriptor> dupOutputDesc =
-                                    new AudioOutputDescriptor(NULL);
-                            dupOutputDesc->mOutput1 = mOutputs.valueFor(mPrimaryOutput);
-                            dupOutputDesc->mOutput2 = mOutputs.valueFor(output);
+                            sp<SwAudioOutputDescriptor> dupOutputDesc =
+                                    new SwAudioOutputDescriptor(NULL, mpClientInterface);
+                            dupOutputDesc->mOutput1 = mPrimaryOutput;
+                            dupOutputDesc->mOutput2 = desc;
                             dupOutputDesc->mSamplingRate = desc->mSamplingRate;
                             dupOutputDesc->mFormat = desc->mFormat;
                             dupOutputDesc->mChannelMask = desc->mChannelMask;
                             dupOutputDesc->mLatency = desc->mLatency;
                             addOutput(duplicatedOutput, dupOutputDesc);
-                            applyStreamVolumes(duplicatedOutput, device, 0, true);
+                            applyStreamVolumes(dupOutputDesc, device, 0, true);
                         } else {
                             ALOGW("checkOutputsForDevice() could not open dup output for %d and %d",
-                                    mPrimaryOutput, output);
+                                    mPrimaryOutput->mIoHandle, output);
                             mpClientInterface->closeOutput(output);
                             removeOutput(output);
                             nextAudioPortGeneration();
@@ -3111,7 +3171,7 @@
                 if (device_distinguishes_on_address(device)) {
                     ALOGV("checkOutputsForDevice(): setOutputDevice(dev=0x%x, addr=%s)",
                             device, address.string());
-                    setOutputDevice(output, device, true/*force*/, 0/*delay*/,
+                    setOutputDevice(desc, device, true/*force*/, 0/*delay*/,
                             NULL/*patch handle*/, address.string());
                 }
                 ALOGV("checkOutputsForDevice(): adding output %d", output);
@@ -3129,10 +3189,9 @@
             if (!desc->isDuplicated()) {
                 // exact match on device
                 if (device_distinguishes_on_address(device) &&
-                        (desc->mProfile->mSupportedDevices.types() == device)) {
+                        (desc->supportedDevices() == device)) {
                     findIoHandlesByAddress(desc, device, address, outputs);
-                } else if (!(desc->mProfile->mSupportedDevices.types()
-                        & mAvailableOutputDevices.types())) {
+                } else if (!(desc->supportedDevices() & mAvailableOutputDevices.types())) {
                     ALOGV("checkOutputsForDevice(): disconnecting adding output %d",
                             mOutputs.keyAt(i));
                     outputs.add(mOutputs.keyAt(i));
@@ -3367,7 +3426,7 @@
 {
     ALOGV("closeOutput(%d)", output);
 
-    sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
+    sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
     if (outputDesc == NULL) {
         ALOGW("closeOutput() unknown output %d", output);
         return;
@@ -3376,7 +3435,7 @@
 
     // look for duplicated outputs connected to the output being removed.
     for (size_t i = 0; i < mOutputs.size(); i++) {
-        sp<AudioOutputDescriptor> dupOutputDesc = mOutputs.valueAt(i);
+        sp<SwAudioOutputDescriptor> dupOutputDesc = mOutputs.valueAt(i);
         if (dupOutputDesc->isDuplicated() &&
                 (dupOutputDesc->mOutput1 == outputDesc ||
                 dupOutputDesc->mOutput2 == outputDesc)) {
@@ -3445,8 +3504,9 @@
     mInputs.removeItem(input);
 }
 
-SortedVector<audio_io_handle_t> AudioPolicyManager::getOutputsForDevice(audio_devices_t device,
-                                                                        AudioOutputCollection openOutputs)
+SortedVector<audio_io_handle_t> AudioPolicyManager::getOutputsForDevice(
+                                                                audio_devices_t device,
+                                                                SwAudioOutputCollection openOutputs)
 {
     SortedVector<audio_io_handle_t> outputs;
 
@@ -3487,14 +3547,14 @@
     // associated with policies in the "before" and "after" output vectors
     ALOGVV("checkOutputForStrategy(): policy related outputs");
     for (size_t i = 0 ; i < mPreviousOutputs.size() ; i++) {
-        const sp<AudioOutputDescriptor> desc = mPreviousOutputs.valueAt(i);
+        const sp<SwAudioOutputDescriptor> desc = mPreviousOutputs.valueAt(i);
         if (desc != 0 && desc->mPolicyMix != NULL) {
             srcOutputs.add(desc->mIoHandle);
             ALOGVV(" previous outputs: adding %d", desc->mIoHandle);
         }
     }
     for (size_t i = 0 ; i < mOutputs.size() ; i++) {
-        const sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
+        const sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
         if (desc != 0 && desc->mPolicyMix != NULL) {
             dstOutputs.add(desc->mIoHandle);
             ALOGVV(" new outputs: adding %d", desc->mIoHandle);
@@ -3506,10 +3566,10 @@
               strategy, srcOutputs[0], dstOutputs[0]);
         // mute strategy while moving tracks from one output to another
         for (size_t i = 0; i < srcOutputs.size(); i++) {
-            sp<AudioOutputDescriptor> desc = mOutputs.valueFor(srcOutputs[i]);
+            sp<SwAudioOutputDescriptor> desc = mOutputs.valueFor(srcOutputs[i]);
             if (isStrategyActive(desc, strategy)) {
-                setStrategyMute(strategy, true, srcOutputs[i]);
-                setStrategyMute(strategy, false, srcOutputs[i], MUTE_TIME_MS, newDevice);
+                setStrategyMute(strategy, true, desc);
+                setStrategyMute(strategy, false, desc, MUTE_TIME_MS, newDevice);
             }
         }
 
@@ -3606,12 +3666,11 @@
     }
 }
 
-audio_devices_t AudioPolicyManager::getNewOutputDevice(audio_io_handle_t output, bool fromCache)
+audio_devices_t AudioPolicyManager::getNewOutputDevice(const sp<AudioOutputDescriptor>& outputDesc,
+                                                       bool fromCache)
 {
     audio_devices_t device = AUDIO_DEVICE_NONE;
 
-    sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
-
     ssize_t index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle);
     if (index >= 0) {
         sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
@@ -3789,9 +3848,9 @@
         ALOGV("\t muting %d", mute);
         uint32_t maxLatency = 0;
         for (size_t i = 0; i < mOutputs.size(); i++) {
-            sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
+            sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
             setStreamMute(AUDIO_STREAM_TTS, mute/*on*/,
-                    desc->mIoHandle,
+                    desc,
                     0 /*delay*/, AUDIO_DEVICE_NONE);
             const uint32_t latency = desc->latency() * 2;
             if (latency > maxLatency) {
@@ -3855,7 +3914,7 @@
 
     for (size_t i = 0; i < NUM_STRATEGIES; i++) {
         audio_devices_t curDevice = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/);
-        curDevice = curDevice & outputDesc->mProfile->mSupportedDevices.types();
+        curDevice = curDevice & outputDesc->supportedDevices();
         bool mute = shouldMute && (curDevice & device) && (curDevice != device);
         bool doMute = false;
 
@@ -3874,10 +3933,9 @@
                         == AUDIO_DEVICE_NONE) {
                     continue;
                 }
-                audio_io_handle_t curOutput = mOutputs.keyAt(j);
-                ALOGVV("checkDeviceMuteStrategies() %s strategy %d (curDevice %04x) on output %d",
-                      mute ? "muting" : "unmuting", i, curDevice, curOutput);
-                setStrategyMute((routing_strategy)i, mute, curOutput, mute ? 0 : delayMs);
+                ALOGVV("checkDeviceMuteStrategies() %s strategy %d (curDevice %04x)",
+                      mute ? "muting" : "unmuting", i, curDevice);
+                setStrategyMute((routing_strategy)i, mute, desc, mute ? 0 : delayMs);
                 if (isStrategyActive(desc, (routing_strategy)i)) {
                     if (mute) {
                         // FIXME: should not need to double latency if volume could be applied
@@ -3902,9 +3960,9 @@
         }
         for (size_t i = 0; i < NUM_STRATEGIES; i++) {
             if (isStrategyActive(outputDesc, (routing_strategy)i)) {
-                setStrategyMute((routing_strategy)i, true, outputDesc->mIoHandle);
+                setStrategyMute((routing_strategy)i, true, outputDesc);
                 // do tempMute unmute after twice the mute wait time
-                setStrategyMute((routing_strategy)i, false, outputDesc->mIoHandle,
+                setStrategyMute((routing_strategy)i, false, outputDesc,
                                 muteWaitMs *2, device);
             }
         }
@@ -3919,32 +3977,31 @@
     return 0;
 }
 
-uint32_t AudioPolicyManager::setOutputDevice(audio_io_handle_t output,
+uint32_t AudioPolicyManager::setOutputDevice(const sp<AudioOutputDescriptor>& outputDesc,
                                              audio_devices_t device,
                                              bool force,
                                              int delayMs,
                                              audio_patch_handle_t *patchHandle,
                                              const char* address)
 {
-    ALOGV("setOutputDevice() output %d device %04x delayMs %d", output, device, delayMs);
-    sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
+    ALOGV("setOutputDevice() device %04x delayMs %d", device, delayMs);
     AudioParameter param;
     uint32_t muteWaitMs;
 
     if (outputDesc->isDuplicated()) {
-        muteWaitMs = setOutputDevice(outputDesc->mOutput1->mIoHandle, device, force, delayMs);
-        muteWaitMs += setOutputDevice(outputDesc->mOutput2->mIoHandle, device, force, delayMs);
+        muteWaitMs = setOutputDevice(outputDesc->subOutput1(), device, force, delayMs);
+        muteWaitMs += setOutputDevice(outputDesc->subOutput2(), device, force, delayMs);
         return muteWaitMs;
     }
     // no need to proceed if new device is not AUDIO_DEVICE_NONE and not supported by current
     // output profile
-    if (device != AUDIO_DEVICE_NONE &&
-        (device & outputDesc->mProfile->mSupportedDevices.types()) == 0) {
+    if ((device != AUDIO_DEVICE_NONE) &&
+            ((device & outputDesc->supportedDevices()) == 0)) {
         return 0;
     }
 
     // filter devices according to output selected
-    device = (audio_devices_t)(device & outputDesc->mProfile->mSupportedDevices.types());
+    device = (audio_devices_t)(device & outputDesc->supportedDevices());
 
     audio_devices_t prevDevice = outputDesc->mDevice;
 
@@ -3964,8 +4021,7 @@
     if ((device == AUDIO_DEVICE_NONE || device == prevDevice) &&
         !force &&
         outputDesc->mPatchHandle != 0) {
-        ALOGV("setOutputDevice() setting same device 0x%04x or null device for output %d",
-              device, output);
+        ALOGV("setOutputDevice() setting same device 0x%04x or null device", device);
         return muteWaitMs;
     }
 
@@ -3973,7 +4029,7 @@
 
     // do the routing
     if (device == AUDIO_DEVICE_NONE) {
-        resetOutputDevice(output, delayMs, NULL);
+        resetOutputDevice(outputDesc, delayMs, NULL);
     } else {
         DeviceVector deviceList = (address == NULL) ?
                 mAvailableOutputDevices.getDevicesFromType(device)
@@ -4040,16 +4096,15 @@
     }
 
     // update stream volumes according to new device
-    applyStreamVolumes(output, device, delayMs);
+    applyStreamVolumes(outputDesc, device, delayMs);
 
     return muteWaitMs;
 }
 
-status_t AudioPolicyManager::resetOutputDevice(audio_io_handle_t output,
+status_t AudioPolicyManager::resetOutputDevice(const sp<AudioOutputDescriptor>& outputDesc,
                                                int delayMs,
                                                audio_patch_handle_t *patchHandle)
 {
-    sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
     ssize_t index;
     if (patchHandle) {
         index = mAudioPatches.indexOfKey(*patchHandle);
@@ -4159,12 +4214,15 @@
 sp<IOProfile> AudioPolicyManager::getInputProfile(audio_devices_t device,
                                                   String8 address,
                                                   uint32_t& samplingRate,
-                                                  audio_format_t format,
-                                                  audio_channel_mask_t channelMask,
+                                                  audio_format_t& format,
+                                                  audio_channel_mask_t& channelMask,
                                                   audio_input_flags_t flags)
 {
     // Choose an input profile based on the requested capture parameters: select the first available
     // profile supporting all requested parameters.
+    //
+    // TODO: perhaps isCompatibleProfile should return a "matching" score so we can return
+    // the best matching profile, not the first one.
 
     for (size_t i = 0; i < mHwModules.size(); i++)
     {
@@ -4177,7 +4235,11 @@
             // profile->log();
             if (profile->isCompatibleProfile(device, address, samplingRate,
                                              &samplingRate /*updatedSamplingRate*/,
-                                             format, channelMask, (audio_output_flags_t) flags)) {
+                                             format,
+                                             &format /*updatedFormat*/,
+                                             channelMask,
+                                             &channelMask /*updatedChannelMask*/,
+                                             (audio_output_flags_t) flags)) {
 
                 return profile;
             }
@@ -4206,17 +4268,10 @@
 }
 
 float AudioPolicyManager::computeVolume(audio_stream_type_t stream,
-                                        int index,
-                                        audio_io_handle_t output,
-                                        audio_devices_t device)
+                                            int index,
+                                            audio_devices_t device)
 {
-    float volume = 1.0;
-    sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
-
-    if (device == AUDIO_DEVICE_NONE) {
-        device = outputDesc->device();
-    }
-    volume = mEngine->volIndexToAmpl(Volume::getDeviceCategory(device), stream, index);
+    float volumeDb = mEngine->volIndexToDb(Volume::getDeviceCategory(device), stream, index);
 
     // if a headset is connected, apply the following rules to ring tones and notifications
     // to avoid sound level bursts in user's ears:
@@ -4234,41 +4289,39 @@
                 || ((stream_strategy == STRATEGY_ENFORCED_AUDIBLE) &&
                     (mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_NONE))) &&
             mStreams.canBeMuted(stream)) {
-        volume *= SONIFICATION_HEADSET_VOLUME_FACTOR;
+        volumeDb += SONIFICATION_HEADSET_VOLUME_FACTOR_DB;
         // when the phone is ringing we must consider that music could have been paused just before
         // by the music application and behave as if music was active if the last music track was
         // just stopped
         if (isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY) ||
                 mLimitRingtoneVolume) {
             audio_devices_t musicDevice = getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/);
-            float musicVol = computeVolume(AUDIO_STREAM_MUSIC,
-                               mStreams[AUDIO_STREAM_MUSIC].getVolumeIndex(musicDevice),
-                               output,
+            float musicVolDB = computeVolume(AUDIO_STREAM_MUSIC,
+                                 mStreams.valueFor(AUDIO_STREAM_MUSIC).getVolumeIndex(musicDevice),
                                musicDevice);
-            float minVol = (musicVol > SONIFICATION_HEADSET_VOLUME_MIN) ?
-                                musicVol : SONIFICATION_HEADSET_VOLUME_MIN;
-            if (volume > minVol) {
-                volume = minVol;
-                ALOGV("computeVolume limiting volume to %f musicVol %f", minVol, musicVol);
+            float minVolDB = (musicVolDB > SONIFICATION_HEADSET_VOLUME_MIN_DB) ?
+                    musicVolDB : SONIFICATION_HEADSET_VOLUME_MIN_DB;
+            if (volumeDb > minVolDB) {
+                volumeDb = minVolDB;
+                ALOGV("computeVolume limiting volume to %f musicVol %f", minVolDB, musicVolDB);
             }
         }
     }
 
-    return volume;
+    return volumeDb;
 }
 
 status_t AudioPolicyManager::checkAndSetVolume(audio_stream_type_t stream,
-                                               int index,
-                                               audio_io_handle_t output,
-                                               audio_devices_t device,
-                                               int delayMs,
-                                               bool force)
+                                                   int index,
+                                                   const sp<AudioOutputDescriptor>& outputDesc,
+                                                   audio_devices_t device,
+                                                   int delayMs,
+                                                   bool force)
 {
-
     // do not change actual stream volume if the stream is muted
-    if (mOutputs.valueFor(output)->mMuteCount[stream] != 0) {
+    if (outputDesc->mMuteCount[stream] != 0) {
         ALOGVV("checkAndSetVolume() stream %d muted count %d",
-              stream, mOutputs.valueFor(output)->mMuteCount[stream]);
+              stream, outputDesc->mMuteCount[stream]);
         return NO_ERROR;
     }
     audio_policy_forced_cfg_t forceUseForComm =
@@ -4281,45 +4334,28 @@
         return INVALID_OPERATION;
     }
 
-    float volume = computeVolume(stream, index, output, device);
-    // unit gain if rerouting to external policy
-    if (device == AUDIO_DEVICE_OUT_REMOTE_SUBMIX) {
-        ssize_t index = mOutputs.indexOfKey(output);
-        if (index >= 0) {
-            sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(index);
-            if (outputDesc->mPolicyMix != NULL) {
-                ALOGV("max gain when rerouting for output=%d", output);
-                volume = 1.0f;
-            }
-        }
+    if (device == AUDIO_DEVICE_NONE) {
+        device = outputDesc->device();
+    }
 
+    float volumeDb = computeVolume(stream, index, device);
+    if (outputDesc->isFixedVolume(device)) {
+        volumeDb = 0.0f;
     }
-    // We actually change the volume if:
-    // - the float value returned by computeVolume() changed
-    // - the force flag is set
-    if (volume != mOutputs.valueFor(output)->mCurVolume[stream] ||
-            force) {
-        mOutputs.valueFor(output)->mCurVolume[stream] = volume;
-        ALOGVV("checkAndSetVolume() for output %d stream %d, volume %f, delay %d", output, stream, volume, delayMs);
-        // Force VOICE_CALL to track BLUETOOTH_SCO stream volume when bluetooth audio is
-        // enabled
-        if (stream == AUDIO_STREAM_BLUETOOTH_SCO) {
-            mpClientInterface->setStreamVolume(AUDIO_STREAM_VOICE_CALL, volume, output, delayMs);
-        }
-        mpClientInterface->setStreamVolume(stream, volume, output, delayMs);
-    }
+
+    outputDesc->setVolume(volumeDb, stream, device, delayMs, force);
 
     if (stream == AUDIO_STREAM_VOICE_CALL ||
         stream == AUDIO_STREAM_BLUETOOTH_SCO) {
         float voiceVolume;
         // Force voice volume to max for bluetooth SCO as volume is managed by the headset
         if (stream == AUDIO_STREAM_VOICE_CALL) {
-            voiceVolume = (float)index/(float)mStreams[stream].getVolumeIndexMax();
+            voiceVolume = (float)index/(float)mStreams.valueFor(stream).getVolumeIndexMax();
         } else {
             voiceVolume = 1.0;
         }
 
-        if (voiceVolume != mLastVoiceVolume && output == mPrimaryOutput) {
+        if (voiceVolume != mLastVoiceVolume && outputDesc == mPrimaryOutput) {
             mpClientInterface->setVoiceVolume(voiceVolume, delayMs);
             mLastVoiceVolume = voiceVolume;
         }
@@ -4328,20 +4364,20 @@
     return NO_ERROR;
 }
 
-void AudioPolicyManager::applyStreamVolumes(audio_io_handle_t output,
-                                            audio_devices_t device,
-                                            int delayMs,
-                                            bool force)
+void AudioPolicyManager::applyStreamVolumes(const sp<AudioOutputDescriptor>& outputDesc,
+                                                audio_devices_t device,
+                                                int delayMs,
+                                                bool force)
 {
-    ALOGVV("applyStreamVolumes() for output %d and device %x", output, device);
+    ALOGVV("applyStreamVolumes() for device %08x", device);
 
     for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
         if (stream == AUDIO_STREAM_PATCH) {
             continue;
         }
         checkAndSetVolume((audio_stream_type_t)stream,
-                          mStreams[stream].getVolumeIndex(device),
-                          output,
+                          mStreams.valueFor((audio_stream_type_t)stream).getVolumeIndex(device),
+                          outputDesc,
                           device,
                           delayMs,
                           force);
@@ -4349,10 +4385,10 @@
 }
 
 void AudioPolicyManager::setStrategyMute(routing_strategy strategy,
-                                         bool on,
-                                         audio_io_handle_t output,
-                                         int delayMs,
-                                         audio_devices_t device)
+                                             bool on,
+                                             const sp<AudioOutputDescriptor>& outputDesc,
+                                             int delayMs,
+                                             audio_devices_t device)
 {
     ALOGVV("setStrategyMute() strategy %d, mute %d, output %d", strategy, on, output);
     for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
@@ -4360,32 +4396,31 @@
             continue;
         }
         if (getStrategy((audio_stream_type_t)stream) == strategy) {
-            setStreamMute((audio_stream_type_t)stream, on, output, delayMs, device);
+            setStreamMute((audio_stream_type_t)stream, on, outputDesc, delayMs, device);
         }
     }
 }
 
 void AudioPolicyManager::setStreamMute(audio_stream_type_t stream,
-                                       bool on,
-                                       audio_io_handle_t output,
-                                       int delayMs,
-                                       audio_devices_t device)
+                                           bool on,
+                                           const sp<AudioOutputDescriptor>& outputDesc,
+                                           int delayMs,
+                                           audio_devices_t device)
 {
-    const StreamDescriptor &streamDesc = mStreams[stream];
-    sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
+    const StreamDescriptor& streamDesc = mStreams.valueFor(stream);
     if (device == AUDIO_DEVICE_NONE) {
         device = outputDesc->device();
     }
 
-    ALOGVV("setStreamMute() stream %d, mute %d, output %d, mMuteCount %d device %04x",
-          stream, on, output, outputDesc->mMuteCount[stream], device);
+    ALOGVV("setStreamMute() stream %d, mute %d, mMuteCount %d device %04x",
+          stream, on, outputDesc->mMuteCount[stream], device);
 
     if (on) {
         if (outputDesc->mMuteCount[stream] == 0) {
             if (streamDesc.canBeMuted() &&
                     ((stream != AUDIO_STREAM_ENFORCED_AUDIBLE) ||
                      (mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_NONE))) {
-                checkAndSetVolume(stream, 0, output, device, delayMs);
+                checkAndSetVolume(stream, 0, outputDesc, device, delayMs);
             }
         }
         // increment mMuteCount after calling checkAndSetVolume() so that volume change is not ignored
@@ -4398,7 +4433,7 @@
         if (--outputDesc->mMuteCount[stream] == 0) {
             checkAndSetVolume(stream,
                               streamDesc.getVolumeIndex(device),
-                              output,
+                              outputDesc,
                               device,
                               delayMs);
         }
@@ -4417,7 +4452,7 @@
     const routing_strategy stream_strategy = getStrategy(stream);
     if ((stream_strategy == STRATEGY_SONIFICATION) ||
             ((stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL))) {
-        sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(mPrimaryOutput);
+        sp<SwAudioOutputDescriptor> outputDesc = mPrimaryOutput;
         ALOGV("handleIncallSonification() stream %d starting %d device %x stateChange %d",
                 stream, starting, outputDesc->mDevice, stateChange);
         if (outputDesc->mRefCount[stream]) {
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h
index 9fab9ef..fe6b986 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.h
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h
@@ -49,8 +49,11 @@
 
 // Attenuation applied to STRATEGY_SONIFICATION streams when a headset is connected: 6dB
 #define SONIFICATION_HEADSET_VOLUME_FACTOR 0.5
+#define SONIFICATION_HEADSET_VOLUME_FACTOR_DB (-6)
 // Min volume for STRATEGY_SONIFICATION streams when limited by music volume: -36dB
 #define SONIFICATION_HEADSET_VOLUME_MIN  0.016
+#define SONIFICATION_HEADSET_VOLUME_MIN_DB  (-36)
+
 // Time in milliseconds during which we consider that music is still active after a music
 // track was stopped - see computeVolume()
 #define SONIFICATION_HEADSET_MUSIC_DELAY  5000
@@ -173,19 +176,15 @@
             return mEffects.setEffectEnabled(id, enabled);
         }
 
-        virtual bool isStreamActive(audio_stream_type_t stream, uint32_t inPastMs = 0) const
-        {
-            return mOutputs.isStreamActive(stream, inPastMs);
-        }
+        virtual bool isStreamActive(audio_stream_type_t stream, uint32_t inPastMs = 0) const;
         // return whether a stream is playing remotely, override to change the definition of
         //   local/remote playback, used for instance by notification manager to not make
         //   media players lose audio focus when not playing locally
         //   For the base implementation, "remotely" means playing during screen mirroring which
         //   uses an output for playback with a non-empty, non "0" address.
-        virtual bool isStreamActiveRemotely(audio_stream_type_t stream, uint32_t inPastMs = 0) const
-        {
-            return mOutputs.isStreamActiveRemotely(stream, inPastMs);
-        }
+        virtual bool isStreamActiveRemotely(audio_stream_type_t stream,
+                                            uint32_t inPastMs = 0) const;
+
         virtual bool isSourceActive(audio_source_t source) const;
 
         virtual status_t dump(int fd);
@@ -281,7 +280,7 @@
         {
             return mPolicyMixes;
         }
-        virtual const AudioOutputCollection &getOutputs() const
+        virtual const SwAudioOutputCollection &getOutputs() const
         {
             return mOutputs;
         }
@@ -306,7 +305,7 @@
             return mDefaultOutputDevice;
         }
 protected:
-        void addOutput(audio_io_handle_t output, sp<AudioOutputDescriptor> outputDesc);
+        void addOutput(audio_io_handle_t output, sp<SwAudioOutputDescriptor> outputDesc);
         void removeOutput(audio_io_handle_t output);
         void addInput(audio_io_handle_t input, sp<AudioInputDescriptor> inputDesc);
 
@@ -329,13 +328,13 @@
 
         // change the route of the specified output. Returns the number of ms we have slept to
         // allow new routing to take effect in certain cases.
-        virtual uint32_t setOutputDevice(audio_io_handle_t output,
+        virtual uint32_t setOutputDevice(const sp<AudioOutputDescriptor>& outputDesc,
                              audio_devices_t device,
                              bool force = false,
                              int delayMs = 0,
                              audio_patch_handle_t *patchHandle = NULL,
                              const char* address = NULL);
-        status_t resetOutputDevice(audio_io_handle_t output,
+        status_t resetOutputDevice(const sp<AudioOutputDescriptor>& outputDesc,
                                    int delayMs = 0,
                                    audio_patch_handle_t *patchHandle = NULL);
         status_t setInputDevice(audio_io_handle_t input,
@@ -350,29 +349,31 @@
 
         // compute the actual volume for a given stream according to the requested index and a particular
         // device
-        virtual float computeVolume(audio_stream_type_t stream, int index,
-                                    audio_io_handle_t output, audio_devices_t device);
+        virtual float computeVolume(audio_stream_type_t stream,
+                                    int index,
+                                    audio_devices_t device);
 
         // check that volume change is permitted, compute and send new volume to audio hardware
         virtual status_t checkAndSetVolume(audio_stream_type_t stream, int index,
-                                           audio_io_handle_t output,
+                                           const sp<AudioOutputDescriptor>& outputDesc,
                                            audio_devices_t device,
                                            int delayMs = 0, bool force = false);
 
         // apply all stream volumes to the specified output and device
-        void applyStreamVolumes(audio_io_handle_t output, audio_devices_t device, int delayMs = 0, bool force = false);
+        void applyStreamVolumes(const sp<AudioOutputDescriptor>& outputDesc,
+                                audio_devices_t device, int delayMs = 0, bool force = false);
 
         // Mute or unmute all streams handled by the specified strategy on the specified output
         void setStrategyMute(routing_strategy strategy,
                              bool on,
-                             audio_io_handle_t output,
+                             const sp<AudioOutputDescriptor>& outputDesc,
                              int delayMs = 0,
                              audio_devices_t device = (audio_devices_t)0);
 
         // Mute or unmute the stream on the specified output
         void setStreamMute(audio_stream_type_t stream,
                            bool on,
-                           audio_io_handle_t output,
+                           const sp<AudioOutputDescriptor>& outputDesc,
                            int delayMs = 0,
                            audio_devices_t device = (audio_devices_t)0);
 
@@ -425,7 +426,8 @@
         // must be called every time a condition that affects the device choice for a given output is
         // changed: connected device, phone state, force use, output start, output stop..
         // see getDeviceForStrategy() for the use of fromCache parameter
-        audio_devices_t getNewOutputDevice(audio_io_handle_t output, bool fromCache);
+        audio_devices_t getNewOutputDevice(const sp<AudioOutputDescriptor>& outputDesc,
+                                           bool fromCache);
 
         // updates cache of device used by all strategies (mDeviceForStrategy[])
         // must be called every time a condition that affects the device choice for a given strategy is
@@ -453,7 +455,7 @@
 #endif //AUDIO_POLICY_TEST
 
         SortedVector<audio_io_handle_t> getOutputsForDevice(audio_devices_t device,
-                                                            AudioOutputCollection openOutputs);
+                                                            SwAudioOutputCollection openOutputs);
         bool vectorsEqual(SortedVector<audio_io_handle_t>& outputs1,
                                            SortedVector<audio_io_handle_t>& outputs2);
 
@@ -468,12 +470,12 @@
         audio_io_handle_t selectOutput(const SortedVector<audio_io_handle_t>& outputs,
                                        audio_output_flags_t flags,
                                        audio_format_t format);
-        // samplingRate parameter is an in/out and so may be modified
+        // samplingRate, format, channelMask are in/out and so may be modified
         sp<IOProfile> getInputProfile(audio_devices_t device,
                                       String8 address,
                                       uint32_t& samplingRate,
-                                      audio_format_t format,
-                                      audio_channel_mask_t channelMask,
+                                      audio_format_t& format,
+                                      audio_channel_mask_t& channelMask,
                                       audio_input_flags_t flags);
         sp<IOProfile> getProfileForDirectOutput(audio_devices_t device,
                                                        uint32_t samplingRate,
@@ -494,25 +496,33 @@
 
         audio_devices_t availablePrimaryOutputDevices() const
         {
-            return mOutputs.getSupportedDevices(mPrimaryOutput) & mAvailableOutputDevices.types();
+            return mPrimaryOutput->supportedDevices() & mAvailableOutputDevices.types();
         }
         audio_devices_t availablePrimaryInputDevices() const
         {
-            return mAvailableInputDevices.getDevicesFromHwModule(
-                        mOutputs.valueFor(mPrimaryOutput)->getModuleHandle());
+            return mAvailableInputDevices.getDevicesFromHwModule(mPrimaryOutput->getModuleHandle());
         }
 
         void updateCallRouting(audio_devices_t rxDevice, int delayMs = 0);
 
+        status_t startSource(sp<AudioOutputDescriptor> outputDesc,
+                             audio_stream_type_t stream,
+                             audio_devices_t device,
+                             uint32_t *delayMs);
+        status_t stopSource(sp<AudioOutputDescriptor> outputDesc,
+                            audio_stream_type_t stream);
+
         uid_t mUidCached;
         AudioPolicyClientInterface *mpClientInterface;  // audio policy client interface
-        audio_io_handle_t mPrimaryOutput;              // primary output handle
+        sp<SwAudioOutputDescriptor> mPrimaryOutput;     // primary output descriptor
         // list of descriptors for outputs currently opened
-        AudioOutputCollection mOutputs;
+
+        SwAudioOutputCollection mOutputs;
         // copy of mOutputs before setDeviceConnectionState() opens new outputs
         // reset to mOutputs when updateDevicesAndOutputs() is called.
-        AudioOutputCollection mPreviousOutputs;
+        SwAudioOutputCollection mPreviousOutputs;
         AudioInputCollection mInputs;     // list of input descriptors
+
         DeviceVector  mAvailableOutputDevices; // all available output devices
         DeviceVector  mAvailableInputDevices;  // all available input devices
 
@@ -583,7 +593,7 @@
         //   in mProfile->mSupportedDevices) matches the device whose address is to be matched.
         // see deviceDistinguishesOnAddress(audio_devices_t) for whether the device type is one
         //   where addresses are used to distinguish between one connected device and another.
-        void findIoHandlesByAddress(sp<AudioOutputDescriptor> desc /*in*/,
+        void findIoHandlesByAddress(sp<SwAudioOutputDescriptor> desc /*in*/,
                 const audio_devices_t device /*in*/,
                 const String8 address /*in*/,
                 SortedVector<audio_io_handle_t>& outputs /*out*/);