diff --git a/include/media/AudioBufferProvider.h b/include/media/AudioBufferProvider.h
index 7be449c..458d170 100644
--- a/include/media/AudioBufferProvider.h
+++ b/include/media/AudioBufferProvider.h
@@ -40,12 +40,6 @@
 
     virtual ~AudioBufferProvider() {}
 
-    // value representing an invalid presentation timestamp
-    static const int64_t kInvalidPTS = 0x7FFFFFFFFFFFFFFFLL;    // <stdint.h> is too painful
-
-    // pts is the local time when the next sample yielded by getNextBuffer
-    // will be rendered.
-    // Pass kInvalidPTS if the PTS is unknown or not applicable.
     // On entry:
     //  buffer              != NULL
     //  buffer->raw         unused
@@ -59,7 +53,7 @@
     //  status              != NO_ERROR
     //  buffer->raw         NULL
     //  buffer->frameCount  0
-    virtual status_t getNextBuffer(Buffer* buffer, int64_t pts = kInvalidPTS) = 0;
+    virtual status_t getNextBuffer(Buffer* buffer) = 0;
 
     // Release (a portion of) the buffer previously obtained by getNextBuffer().
     // It is permissible to call releaseBuffer() multiple times per getNextBuffer().
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index 38adb03..a4b8571 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -954,7 +954,6 @@
 
     mutable Mutex           mLock;
 
-    bool                    mIsTimed;
     int                     mPreviousPriority;          // before start()
     SchedPolicy             mPreviousSchedulingGroup;
     bool                    mAwaitBoost;    // thread should wait for priority boost before running
@@ -992,29 +991,6 @@
     sp<AudioSystem::AudioDeviceCallback> mDeviceCallback;
 };
 
-class TimedAudioTrack : public AudioTrack
-{
-public:
-    TimedAudioTrack();
-
-    /* allocate a shared memory buffer that can be passed to queueTimedBuffer */
-    status_t allocateTimedBuffer(size_t size, sp<IMemory>* buffer);
-
-    /* queue a buffer obtained via allocateTimedBuffer for playback at the
-       given timestamp.  PTS units are microseconds on the media time timeline.
-       The media time transform (set with setMediaTimeTransform) set by the
-       audio producer will handle converting from media time to local time
-       (perhaps going through the common time timeline in the case of
-       synchronized multiroom audio case) */
-    status_t queueTimedBuffer(const sp<IMemory>& buffer, int64_t pts);
-
-    /* define a transform between media time and either common time or
-       local time */
-    enum TargetTimeline {LOCAL_TIME, COMMON_TIME};
-    status_t setMediaTimeTransform(const LinearTransform& xform,
-                                   TargetTimeline target);
-};
-
 }; // namespace android
 
 #endif // ANDROID_AUDIOTRACK_H
diff --git a/include/media/IAudioFlinger.h b/include/media/IAudioFlinger.h
index 5051aff..3b69ecf 100644
--- a/include/media/IAudioFlinger.h
+++ b/include/media/IAudioFlinger.h
@@ -47,7 +47,8 @@
     // or-able bits shared by createTrack and openRecord, but not all combinations make sense
     enum {
         TRACK_DEFAULT = 0,  // client requests a default AudioTrack
-        TRACK_TIMED   = 1,  // client requests a TimedAudioTrack
+        // FIXME: obsolete
+        // TRACK_TIMED= 1,  // client requests a TimedAudioTrack
         TRACK_FAST    = 2,  // client requests a fast AudioTrack or AudioRecord
         TRACK_OFFLOAD = 4,  // client requests offload to hw codec
         TRACK_DIRECT = 8,   // client requests a direct output
diff --git a/include/media/IAudioTrack.h b/include/media/IAudioTrack.h
index 619ac78..a31cec6 100644
--- a/include/media/IAudioTrack.h
+++ b/include/media/IAudioTrack.h
@@ -24,7 +24,6 @@
 #include <utils/Errors.h>
 #include <binder/IInterface.h>
 #include <binder/IMemory.h>
-#include <utils/LinearTransform.h>
 #include <utils/String8.h>
 #include <media/AudioTimestamp.h>
 
@@ -67,24 +66,6 @@
      */
     virtual status_t    attachAuxEffect(int effectId) = 0;
 
-
-    /* Allocate a shared memory buffer suitable for holding timed audio
-       samples */
-    virtual status_t    allocateTimedBuffer(size_t size,
-                                            sp<IMemory>* buffer) = 0;
-
-    /* Queue a buffer obtained via allocateTimedBuffer for playback at the given
-       timestamp */
-    virtual status_t    queueTimedBuffer(const sp<IMemory>& buffer,
-                                         int64_t pts) = 0;
-
-    /* Define the linear transform that will be applied to the timestamps
-       given to queueTimedBuffer (which are expressed in media time).
-       Target specifies whether this transform converts media time to local time
-       or Tungsten time. The values for target are defined in AudioTrack.h */
-    virtual status_t    setMediaTimeTransform(const LinearTransform& xform,
-                                              int target) = 0;
-
     /* Send parameters to the audio hardware */
     virtual status_t    setParameters(const String8& keyValuePairs) = 0;
 
diff --git a/include/media/nbaio/AudioBufferProviderSource.h b/include/media/nbaio/AudioBufferProviderSource.h
index b16e20a..4747dcf 100644
--- a/include/media/nbaio/AudioBufferProviderSource.h
+++ b/include/media/nbaio/AudioBufferProviderSource.h
@@ -42,9 +42,8 @@
     //virtual size_t framesOverrun();
     //virtual size_t overruns();
     virtual ssize_t availableToRead();
-    virtual ssize_t read(void *buffer, size_t count, int64_t readPTS);
-    virtual ssize_t readVia(readVia_t via, size_t total, void *user,
-                            int64_t readPTS, size_t block);
+    virtual ssize_t read(void *buffer, size_t count);
+    virtual ssize_t readVia(readVia_t via, size_t total, void *user, size_t block);
 
 private:
     AudioBufferProvider * const mProvider;
diff --git a/include/media/nbaio/AudioStreamInSource.h b/include/media/nbaio/AudioStreamInSource.h
index 5169f1e..eaea63c 100644
--- a/include/media/nbaio/AudioStreamInSource.h
+++ b/include/media/nbaio/AudioStreamInSource.h
@@ -45,7 +45,7 @@
     // FIXME Use an audio HAL API to query the buffer filling status when it's available.
     virtual ssize_t availableToRead() { return mStreamBufferSizeBytes / mFrameSize; }
 
-    virtual ssize_t read(void *buffer, size_t count, int64_t readPTS);
+    virtual ssize_t read(void *buffer, size_t count);
 
     // NBAIO_Sink end
 
diff --git a/include/media/nbaio/AudioStreamOutSink.h b/include/media/nbaio/AudioStreamOutSink.h
index 9949b88..0998d45 100644
--- a/include/media/nbaio/AudioStreamOutSink.h
+++ b/include/media/nbaio/AudioStreamOutSink.h
@@ -47,11 +47,6 @@
 
     virtual ssize_t write(const void *buffer, size_t count);
 
-    // AudioStreamOutSink wraps a HAL's output stream.  Its
-    // getNextWriteTimestamp method is simply a passthru to the HAL's underlying
-    // implementation of GNWT (if any)
-    virtual status_t getNextWriteTimestamp(int64_t *timestamp);
-
     virtual status_t getTimestamp(AudioTimestamp& timestamp);
 
     // NBAIO_Sink end
diff --git a/include/media/nbaio/MonoPipe.h b/include/media/nbaio/MonoPipe.h
index b09b35f..df9cafe 100644
--- a/include/media/nbaio/MonoPipe.h
+++ b/include/media/nbaio/MonoPipe.h
@@ -18,7 +18,6 @@
 #define ANDROID_AUDIO_MONO_PIPE_H
 
 #include <time.h>
-#include <utils/LinearTransform.h>
 #include "NBAIO.h"
 #include <media/SingleStateQueue.h>
 
@@ -60,20 +59,6 @@
     virtual ssize_t write(const void *buffer, size_t count);
     //virtual ssize_t writeVia(writeVia_t via, size_t total, void *user, size_t block);
 
-    // MonoPipe's implementation of getNextWriteTimestamp works in conjunction
-    // with MonoPipeReader.  Every time a MonoPipeReader reads from the pipe, it
-    // receives a "readPTS" indicating the point in time for which the reader
-    // would like to read data.  This "last read PTS" is offset by the amt of
-    // data the reader is currently mixing and then cached cached along with the
-    // updated read pointer.  This cached value is the local time for which the
-    // reader is going to request data next time it reads data (assuming we are
-    // in steady state and operating with no underflows).  Writers to the
-    // MonoPipe who would like to know when their next write operation will hit
-    // the speakers can call getNextWriteTimestamp which will return the value
-    // of the last read PTS plus the duration of the amt of data waiting to be
-    // read in the MonoPipe.
-    virtual status_t getNextWriteTimestamp(int64_t *timestamp);
-
             // average number of frames present in the pipe under normal conditions.
             // See throttling mechanism in MonoPipe::write()
             size_t  getAvgFrames() const { return mSetpoint; }
@@ -95,43 +80,21 @@
             status_t getTimestamp(AudioTimestamp& timestamp);
 
 private:
-    // A pair of methods and a helper variable which allows the reader and the
-    // writer to update and observe the values of mFront and mNextRdPTS in an
-    // atomic lock-less fashion.
-    //
-    // :: Important ::
-    // Two assumptions must be true in order for this lock-less approach to
-    // function properly on all systems.  First, there may only be one updater
-    // thread in the system.  Second, the updater thread must be running at a
-    // strictly higher priority than the observer threads.  Currently, both of
-    // these assumptions are true.  The only updater is always a single
-    // FastMixer thread (which runs with SCHED_FIFO/RT priority while the only
-    // observer is always an AudioFlinger::PlaybackThread running with
-    // traditional (non-RT) audio priority.
-    void updateFrontAndNRPTS(int32_t newFront, int64_t newNextRdPTS);
-    void observeFrontAndNRPTS(int32_t *outFront, int64_t *outNextRdPTS);
-    volatile int32_t mUpdateSeq;
-
     const size_t    mReqFrames;     // as requested in constructor, unrounded
     const size_t    mMaxFrames;     // always a power of 2
     void * const    mBuffer;
     // mFront and mRear will never be separated by more than mMaxFrames.
     // 32-bit overflow is possible if the pipe is active for a long time, but if that happens it's
     // safe because we "&" with (mMaxFrames-1) at end of computations to calculate a buffer index.
-    volatile int32_t mFront;        // written by the reader with updateFrontAndNRPTS, observed by
-                                    // the writer with observeFrontAndNRPTS
+    volatile int32_t mFront;        // written by reader with android_atomic_release_store,
+                                    // read by writer with android_atomic_acquire_load
     volatile int32_t mRear;         // written by writer with android_atomic_release_store,
                                     // read by reader with android_atomic_acquire_load
-    volatile int64_t mNextRdPTS;    // written by the reader with updateFrontAndNRPTS, observed by
-                                    // the writer with observeFrontAndNRPTS
     bool            mWriteTsValid;  // whether mWriteTs is valid
     struct timespec mWriteTs;       // time that the previous write() completed
     size_t          mSetpoint;      // target value for pipe fill depth
     const bool      mWriteCanBlock; // whether write() should block if the pipe is full
 
-    int64_t offsetTimestampByAudioFrames(int64_t ts, size_t audFrames);
-    LinearTransform mSamplesToLocalTime;
-
     bool            mIsShutdown;    // whether shutdown(true) was called, no barriers are needed
 
     AudioTimestampSingleStateQueue::Shared      mTimestampShared;
diff --git a/include/media/nbaio/MonoPipeReader.h b/include/media/nbaio/MonoPipeReader.h
index 78fe867..4a7c3c5 100644
--- a/include/media/nbaio/MonoPipeReader.h
+++ b/include/media/nbaio/MonoPipeReader.h
@@ -47,7 +47,7 @@
 
     virtual ssize_t availableToRead();
 
-    virtual ssize_t read(void *buffer, size_t count, int64_t readPTS);
+    virtual ssize_t read(void *buffer, size_t count);
 
     virtual void    onTimestamp(const AudioTimestamp& timestamp);
 
diff --git a/include/media/nbaio/NBAIO.h b/include/media/nbaio/NBAIO.h
index d9bbc8d..2f7e291 100644
--- a/include/media/nbaio/NBAIO.h
+++ b/include/media/nbaio/NBAIO.h
@@ -79,8 +79,7 @@
 
 // Callbacks used by NBAIO_Sink::writeVia() and NBAIO_Source::readVia() below.
 typedef ssize_t (*writeVia_t)(void *user, void *buffer, size_t count);
-typedef ssize_t (*readVia_t)(void *user, const void *buffer,
-                             size_t count, int64_t readPTS);
+typedef ssize_t (*readVia_t)(void *user, const void *buffer, size_t count);
 
 // Check whether an NBAIO_Format is valid
 bool Format_isValid(const NBAIO_Format& format);
@@ -210,21 +209,6 @@
     //  < 0     status_t error occurred prior to the first frame transfer during this callback.
     virtual ssize_t writeVia(writeVia_t via, size_t total, void *user, size_t block = 0);
 
-    // Get the time (on the LocalTime timeline) at which the first frame of audio of the next write
-    // operation to this sink will be eventually rendered by the HAL.
-    // Inputs:
-    //  ts      A pointer pointing to the int64_t which will hold the result.
-    // Return value:
-    //  OK      Everything went well, *ts holds the time at which the first audio frame of the next
-    //          write operation will be rendered, or AudioBufferProvider::kInvalidPTS if this sink
-    //          does not know the answer for some reason.  Sinks which eventually lead to a HAL
-    //          which implements get_next_write_timestamp may return Invalid temporarily if the DMA
-    //          output of the audio driver has not started yet.  Sinks which lead to a HAL which
-    //          does not implement get_next_write_timestamp, or which don't lead to a HAL at all,
-    //          will always return kInvalidPTS.
-    //  <other> Something unexpected happened internally.  Check the logs and start debugging.
-    virtual status_t getNextWriteTimestamp(int64_t *ts) { return INVALID_OPERATION; }
-
     // Returns NO_ERROR if a timestamp is available.  The timestamp includes the total number
     // of frames presented to an external observer, together with the value of CLOCK_MONOTONIC
     // as of this presentation count.  The timestamp parameter is undefined if error is returned.
@@ -271,8 +255,6 @@
     // Inputs:
     //  buffer  Non-NULL destination buffer owned by consumer.
     //  count   Maximum number of frames to transfer.
-    //  readPTS The presentation time (on the LocalTime timeline) for which data
-    //          is being requested, or kInvalidPTS if not known.
     // Return value:
     //  > 0     Number of frames successfully transferred prior to first error.
     //  = 0     Count was zero.
@@ -282,7 +264,7 @@
     //  WOULD_BLOCK No frames can be transferred without blocking.
     //  OVERRUN     read() has not been called frequently enough, or with enough frames to keep up.
     //              One or more frames were lost due to overrun, try again to read more recent data.
-    virtual ssize_t read(void *buffer, size_t count, int64_t readPTS) = 0;
+    virtual ssize_t read(void *buffer, size_t count) = 0;
 
     // Transfer data from source using a series of callbacks.  More suitable for zero-fill,
     // synthesis, and non-contiguous transfers (e.g. circular buffer or readv).
@@ -291,8 +273,6 @@
     //  total   Estimate of the number of frames the consumer desires.  This is an estimate,
     //          and it can consume a different number of frames during the series of callbacks.
     //  user    Arbitrary void * reserved for data consumer.
-    //  readPTS The presentation time (on the LocalTime timeline) for which data
-    //          is being requested, or kInvalidPTS if not known.
     //  block   Number of frames per block, that is a suggested value for 'count' in each callback.
     //          Zero means no preference.  This parameter is a hint only, and may be ignored.
     // Return value:
@@ -315,8 +295,7 @@
     //  > 0     Number of frames successfully transferred during this callback prior to first error.
     //  = 0     Count was zero.
     //  < 0     status_t error occurred prior to the first frame transfer during this callback.
-    virtual ssize_t readVia(readVia_t via, size_t total, void *user,
-                            int64_t readPTS, size_t block = 0);
+    virtual ssize_t readVia(readVia_t via, size_t total, void *user, size_t block = 0);
 
     // Invoked asynchronously by corresponding sink when a new timestamp is available.
     // Default implementation ignores the timestamp.
diff --git a/include/media/nbaio/PipeReader.h b/include/media/nbaio/PipeReader.h
index 350e6ab..398353b 100644
--- a/include/media/nbaio/PipeReader.h
+++ b/include/media/nbaio/PipeReader.h
@@ -45,7 +45,7 @@
 
     virtual ssize_t availableToRead();
 
-    virtual ssize_t read(void *buffer, size_t count, int64_t readPTS);
+    virtual ssize_t read(void *buffer, size_t count);
 
     // NBAIO_Source end
 
diff --git a/include/media/nbaio/SourceAudioBufferProvider.h b/include/media/nbaio/SourceAudioBufferProvider.h
index daf6bc3..29172e1 100644
--- a/include/media/nbaio/SourceAudioBufferProvider.h
+++ b/include/media/nbaio/SourceAudioBufferProvider.h
@@ -31,7 +31,7 @@
     virtual ~SourceAudioBufferProvider();
 
     // AudioBufferProvider interface
-    virtual status_t getNextBuffer(Buffer *buffer, int64_t pts);
+    virtual status_t getNextBuffer(Buffer *buffer);
     virtual void     releaseBuffer(Buffer *buffer);
 
     // ExtendedAudioBufferProvider interface