diff --git a/include/media/AudioResamplerPublic.h b/include/media/AudioResamplerPublic.h
index 3c30148..07d946d 100644
--- a/include/media/AudioResamplerPublic.h
+++ b/include/media/AudioResamplerPublic.h
@@ -42,6 +42,8 @@
 #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.
@@ -66,4 +68,13 @@
     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/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/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/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 1a20fae..b30fd20 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -1608,13 +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;
         }
@@ -3592,21 +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)) {
@@ -3769,6 +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.
              *
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 1566b1f..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.