diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index eaaef4a..3b13194 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -201,6 +201,10 @@
                            binder to AudioFlinger.
                            It will return an error instead.  The application will recreate
                            the track based on offloading or different channel configuration, etc.
+     * maxRequiredSpeed:   For PCM tracks, this creates an appropriate buffer size that will allow
+     *                     maxRequiredSpeed playback. Values less than 1.0f and greater than
+     *                     AUDIO_TIMESTRETCH_SPEED_MAX will be clamped.  For non-PCM tracks
+     *                     and direct or offloaded tracks, this parameter is ignored.
      * threadCanCallJava:  Not present in parameter list, and so is fixed at false.
      */
 
@@ -219,7 +223,8 @@
                                     int uid = -1,
                                     pid_t pid = -1,
                                     const audio_attributes_t* pAttributes = NULL,
-                                    bool doNotReconnect = false);
+                                    bool doNotReconnect = false,
+                                    float maxRequiredSpeed = 1.0f);
 
     /* Creates an audio track and registers it with AudioFlinger.
      * With this constructor, the track is configured for static buffer mode.
@@ -248,7 +253,8 @@
                                     int uid = -1,
                                     pid_t pid = -1,
                                     const audio_attributes_t* pAttributes = NULL,
-                                    bool doNotReconnect = false);
+                                    bool doNotReconnect = false,
+                                    float maxRequiredSpeed = 1.0f);
 
     /* Terminates the AudioTrack and unregisters it from AudioFlinger.
      * Also destroys all resources associated with the AudioTrack.
@@ -293,7 +299,8 @@
                             int uid = -1,
                             pid_t pid = -1,
                             const audio_attributes_t* pAttributes = NULL,
-                            bool doNotReconnect = false);
+                            bool doNotReconnect = false,
+                            float maxRequiredSpeed = 1.0f);
 
     /* Result of constructing the AudioTrack. This must be checked for successful initialization
      * before using any AudioTrack API (except for set()), because using
@@ -916,6 +923,7 @@
     mutable uint32_t        mSampleRate;            // mutable because getSampleRate() can update it
     uint32_t                mOriginalSampleRate;
     AudioPlaybackRate       mPlaybackRate;
+    float                   mMaxRequiredSpeed;      // use PCM buffer size to allow this speed
 
     // Corresponds to current IAudioTrack, value is reported back by AudioFlinger to the client.
     // This allocated buffer size is maintained by the proxy.
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index a4cc3d7..27db7b3 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -191,7 +191,8 @@
         int uid,
         pid_t pid,
         const audio_attributes_t* pAttributes,
-        bool doNotReconnect)
+        bool doNotReconnect,
+        float maxRequiredSpeed)
     : mStatus(NO_INIT),
       mState(STATE_STOPPED),
       mPreviousPriority(ANDROID_PRIORITY_NORMAL),
@@ -202,7 +203,7 @@
     mStatus = set(streamType, sampleRate, format, channelMask,
             frameCount, flags, cbf, user, notificationFrames,
             0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType,
-            offloadInfo, uid, pid, pAttributes, doNotReconnect);
+            offloadInfo, uid, pid, pAttributes, doNotReconnect, maxRequiredSpeed);
 }
 
 AudioTrack::AudioTrack(
@@ -221,7 +222,8 @@
         int uid,
         pid_t pid,
         const audio_attributes_t* pAttributes,
-        bool doNotReconnect)
+        bool doNotReconnect,
+        float maxRequiredSpeed)
     : mStatus(NO_INIT),
       mState(STATE_STOPPED),
       mPreviousPriority(ANDROID_PRIORITY_NORMAL),
@@ -232,7 +234,7 @@
     mStatus = set(streamType, sampleRate, format, channelMask,
             0 /*frameCount*/, flags, cbf, user, notificationFrames,
             sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo,
-            uid, pid, pAttributes, doNotReconnect);
+            uid, pid, pAttributes, doNotReconnect, maxRequiredSpeed);
 }
 
 AudioTrack::~AudioTrack()
@@ -281,7 +283,8 @@
         int uid,
         pid_t pid,
         const audio_attributes_t* pAttributes,
-        bool doNotReconnect)
+        bool doNotReconnect,
+        float maxRequiredSpeed)
 {
     ALOGV("set(): streamType %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, "
           "flags #%x, notificationFrames %u, sessionId %d, transferType %d, uid %d, pid %d",
@@ -422,6 +425,8 @@
     mSampleRate = sampleRate;
     mOriginalSampleRate = sampleRate;
     mPlaybackRate = AUDIO_PLAYBACK_RATE_DEFAULT;
+    // 1.0 <= mMaxRequiredSpeed <= AUDIO_TIMESTRETCH_SPEED_MAX
+    mMaxRequiredSpeed = min(max(maxRequiredSpeed, 1.0f), AUDIO_TIMESTRETCH_SPEED_MAX);
 
     // Make copy of input parameter offloadInfo so that in the future:
     //  (a) createTrack_l doesn't need it as an input parameter
@@ -824,6 +829,9 @@
     if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
         return INVALID_OPERATION;
     }
+
+    ALOGV("setPlaybackRate (input): mSampleRate:%u  mSpeed:%f  mPitch:%f",
+            mSampleRate, playbackRate.mSpeed, playbackRate.mPitch);
     // pitch is emulated by adjusting speed and sampleRate
     const uint32_t effectiveRate = adjustSampleRate(mSampleRate, playbackRate.mPitch);
     const float effectiveSpeed = adjustSpeed(playbackRate.mSpeed, playbackRate.mPitch);
@@ -832,12 +840,18 @@
     playbackRateTemp.mSpeed = effectiveSpeed;
     playbackRateTemp.mPitch = effectivePitch;
 
+    ALOGV("setPlaybackRate (effective): mSampleRate:%u  mSpeed:%f  mPitch:%f",
+            effectiveRate, effectiveSpeed, effectivePitch);
+
     if (!isAudioPlaybackRateValid(playbackRateTemp)) {
+        ALOGV("setPlaybackRate(%f, %f) failed (effective rate out of bounds)",
+                playbackRate.mSpeed, playbackRate.mPitch);
         return BAD_VALUE;
     }
     // Check if the buffer size is compatible.
     if (!isSampleRateSpeedAllowed_l(effectiveRate, effectiveSpeed)) {
-        ALOGV("setPlaybackRate(%f, %f) failed", playbackRate.mSpeed, playbackRate.mPitch);
+        ALOGV("setPlaybackRate(%f, %f) failed (buffer size)",
+                playbackRate.mSpeed, playbackRate.mPitch);
         return BAD_VALUE;
     }
 
@@ -1275,9 +1289,11 @@
 
         if ((mFlags & AUDIO_OUTPUT_FLAG_FAST) == 0) {
             // for normal tracks precompute the frame count based on speed.
+            const float speed = !isPurePcmData_l() || isOffloadedOrDirect_l() ? 1.0f :
+                            max(mMaxRequiredSpeed, mPlaybackRate.mSpeed);
             const size_t minFrameCount = calculateMinFrameCount(
                     mAfLatency, mAfFrameCount, mAfSampleRate, mSampleRate,
-                    mPlaybackRate.mSpeed);
+                    speed);
             if (frameCount < minFrameCount) {
                 frameCount = minFrameCount;
             }
diff --git a/media/libmediaplayerservice/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp
index 1a3013a..2afbdb4 100644
--- a/media/libmediaplayerservice/MediaPlayerService.cpp
+++ b/media/libmediaplayerservice/MediaPlayerService.cpp
@@ -94,6 +94,8 @@
 // Max number of entries in the filter.
 const int kMaxFilterSize = 64;  // I pulled that out of thin air.
 
+const float kMaxRequiredSpeed = 8.0f; // for PCM tracks allow up to 8x speedup.
+
 // FIXME: Move all the metadata related function in the Metadata.cpp
 
 
@@ -1749,6 +1751,14 @@
                     mAttributes,
                     doNotReconnect);
         } else {
+            // TODO: Due to buffer memory concerns, we use a max target playback speed
+            // based on mPlaybackRate at the time of open (instead of kMaxRequiredSpeed),
+            // also clamping the target speed to 1.0 <= targetSpeed <= kMaxRequiredSpeed.
+            const float targetSpeed =
+                    std::min(std::max(mPlaybackRate.mSpeed, 1.0f), kMaxRequiredSpeed);
+            ALOGW_IF(targetSpeed != mPlaybackRate.mSpeed,
+                    "track target speed:%f clamped from playback speed:%f",
+                    targetSpeed, mPlaybackRate.mSpeed);
             t = new AudioTrack(
                     mStreamType,
                     sampleRate,
@@ -1765,7 +1775,8 @@
                     mUid,
                     mPid,
                     mAttributes,
-                    doNotReconnect);
+                    doNotReconnect,
+                    targetSpeed);
         }
 
         if ((t == 0) || (t->initCheck() != NO_ERROR)) {
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
index cbb9d95..5f36fc1 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
@@ -1876,6 +1876,10 @@
         // NuPlayer a chance to switch from non-offload mode to offload mode.
         // So we only set doNotReconnect when there's no video.
         const bool doNotReconnect = !hasVideo;
+
+        // We should always be able to set our playback settings if the sink is closed.
+        LOG_ALWAYS_FATAL_IF(mAudioSink->setPlaybackRate(mPlaybackSettings) != OK,
+                "onOpenAudioSink: can't set playback rate on closed sink");
         status_t err = mAudioSink->open(
                     sampleRate,
                     numChannels,
@@ -1888,9 +1892,6 @@
                     NULL,
                     doNotReconnect,
                     frameCount);
-        if (err == OK) {
-            err = mAudioSink->setPlaybackRate(mPlaybackSettings);
-        }
         if (err != OK) {
             ALOGW("openAudioSink: non offloaded open failed status: %d", err);
             mAudioSink->close();