diff --git a/media/libaaudio/src/client/AudioEndpoint.cpp b/media/libaaudio/src/client/AudioEndpoint.cpp
index e780f4f..cd7679c 100644
--- a/media/libaaudio/src/client/AudioEndpoint.cpp
+++ b/media/libaaudio/src/client/AudioEndpoint.cpp
@@ -278,3 +278,9 @@
         mDataQueue->eraseMemory();
     }
 }
+
+void AudioEndpoint::eraseEmptyDataMemory(int32_t numFrames) {
+    if (mDataQueue != nullptr) {
+        mDataQueue->eraseEmptyMemory(numFrames);
+    }
+}
diff --git a/media/libaaudio/src/client/AudioEndpoint.h b/media/libaaudio/src/client/AudioEndpoint.h
index b117572..7e97c6a 100644
--- a/media/libaaudio/src/client/AudioEndpoint.h
+++ b/media/libaaudio/src/client/AudioEndpoint.h
@@ -107,6 +107,8 @@
      */
     void eraseDataMemory();
 
+    void eraseEmptyDataMemory(int32_t numFrames);
+
     void freeDataQueue() { mDataQueue.reset(); }
 
     void dump() const;
diff --git a/media/libaaudio/src/client/AudioStreamInternal.cpp b/media/libaaudio/src/client/AudioStreamInternal.cpp
index 7648e25..b2e93f0 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternal.cpp
@@ -575,10 +575,20 @@
         return AAUDIO_ERROR_INVALID_STATE;
     }
 
+    // For playback, sleep until all the audio data has played.
+    // Then clear the buffer to prevent noise.
+    prepareBuffersForStop();
+
     mClockModel.stop(AudioClock::getNanoseconds());
     setState(AAUDIO_STREAM_STATE_STOPPING);
     mAtomicInternalTimestamp.clear();
 
+#if 0
+    // Simulate very slow CPU, force race condition where the
+    // DSP keeps playing after we stop writing.
+    AudioClock::sleepForNanos(800 * AAUDIO_NANOS_PER_MILLISECOND);
+#endif
+
     result = mServiceInterface.stopStream(mServiceStreamHandleInfo);
     if (result == AAUDIO_ERROR_INVALID_HANDLE) {
         ALOGD("%s() INVALID_HANDLE, stream was probably stolen", __func__);
diff --git a/media/libaaudio/src/client/AudioStreamInternal.h b/media/libaaudio/src/client/AudioStreamInternal.h
index a5981b1..20d55f9 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.h
+++ b/media/libaaudio/src/client/AudioStreamInternal.h
@@ -123,6 +123,8 @@
 
     virtual void prepareBuffersForStart() {}
 
+    virtual void prepareBuffersForStop() {}
+
     virtual void advanceClientToMatchServerPosition(int32_t serverMargin) = 0;
 
     virtual void onFlushFromServer() {}
diff --git a/media/libaaudio/src/client/AudioStreamInternalPlay.cpp b/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
index 5d4c3d4..0427777 100644
--- a/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
@@ -19,6 +19,8 @@
 
 #define ATRACE_TAG ATRACE_TAG_AUDIO
 
+#include <algorithm>
+
 #include <media/MediaMetricsItem.h>
 #include <utils/Trace.h>
 
@@ -108,6 +110,61 @@
     mAudioEndpoint->eraseDataMemory();
 }
 
+void AudioStreamInternalPlay::prepareBuffersForStop() {
+    // If this is a shared stream and the FIFO is being read by the mixer then
+    // we don't have to worry about the DSP reading past the valid data. We can skip all this.
+    if(!mAudioEndpoint->isFreeRunning()) {
+        return;
+    }
+    // Sleep until the DSP has read all of the data written.
+    int64_t validFramesInBuffer = getFramesWritten() - getFramesRead();
+    if (validFramesInBuffer >= 0) {
+        int64_t emptyFramesInBuffer = ((int64_t) getBufferCapacity()) - validFramesInBuffer;
+
+        // Prevent stale data from being played if the DSP is still running.
+        // Erase some of the FIFO memory in front of the DSP read cursor.
+        // Subtract one burst so we do not accidentally erase data that the DSP might be using.
+        int64_t framesToErase = std::max((int64_t) 0,
+                                         emptyFramesInBuffer - getFramesPerBurst());
+        mAudioEndpoint->eraseEmptyDataMemory(framesToErase);
+
+        // Sleep until we are confident the DSP has consumed all of the valid data.
+        // Sleep for one extra burst as a safety margin because the IsochronousClockModel
+        // is not perfectly accurate.
+        int64_t positionInEmptyMemory = getFramesWritten() + getFramesPerBurst();
+        int64_t timeAllConsumed = mClockModel.convertPositionToTime(positionInEmptyMemory);
+        int64_t durationAllConsumed = timeAllConsumed - AudioClock::getNanoseconds();
+        // Prevent sleeping for too long.
+        durationAllConsumed = std::min(200 * AAUDIO_NANOS_PER_MILLISECOND, durationAllConsumed);
+        AudioClock::sleepForNanos(durationAllConsumed);
+    }
+
+    // Erase all of the memory in case the DSP keeps going and wraps around.
+    mAudioEndpoint->eraseDataMemory();
+
+    // Wait for the last buffer to reach the DAC.
+    // This is because the expected behavior of stop() is that all data written to the stream
+    // should be played before the hardware actually shuts down.
+    // This is different than pause(), where we just end as soon as possible.
+    // This can be important when, for example, playing car navigation and
+    // you want the user to hear the complete instruction.
+    if (mAtomicInternalTimestamp.isValid()) {
+        // Use timestamps to calculate the latency between the DSP reading
+        // a frame and when it reaches the DAC.
+        // This code assumes that timestamps are accurate.
+        Timestamp timestamp = mAtomicInternalTimestamp.read();
+        int64_t dacPosition = timestamp.getPosition();
+        int64_t hardwareReadTime = mClockModel.convertPositionToTime(dacPosition);
+        int64_t hardwareLatencyNanos = timestamp.getNanoseconds() - hardwareReadTime;
+        ALOGD("%s() hardwareLatencyNanos = %lld", __func__,
+              (long long) hardwareLatencyNanos);
+        // Prevent sleeping for too long.
+        hardwareLatencyNanos = std::min(30 * AAUDIO_NANOS_PER_MILLISECOND,
+                                        hardwareLatencyNanos);
+        AudioClock::sleepForNanos(hardwareLatencyNanos);
+    }
+}
+
 void AudioStreamInternalPlay::advanceClientToMatchServerPosition(int32_t serverMargin) {
     int64_t readCounter = mAudioEndpoint->getDataReadCounter() + serverMargin;
     int64_t writeCounter = mAudioEndpoint->getDataWriteCounter();
@@ -353,20 +410,26 @@
         // Call application using the AAudio callback interface.
         callbackResult = maybeCallDataCallback(mCallbackBuffer.get(), mCallbackFrames);
 
-        if (callbackResult == AAUDIO_CALLBACK_RESULT_CONTINUE) {
-            // Write audio data to stream. This is a BLOCKING WRITE!
-            result = write(mCallbackBuffer.get(), mCallbackFrames, timeoutNanos);
-            if ((result != mCallbackFrames)) {
-                if (result >= 0) {
-                    // Only wrote some of the frames requested. The stream can be disconnected
-                    // or timed out.
-                    processCommands();
-                    result = isDisconnected() ? AAUDIO_ERROR_DISCONNECTED : AAUDIO_ERROR_TIMEOUT;
-                }
-                maybeCallErrorCallback(result);
-                break;
+        // Write audio data to stream. This is a BLOCKING WRITE!
+        // Write data regardless of the callbackResult because we assume the data
+        // is valid even when the callback returns AAUDIO_CALLBACK_RESULT_STOP.
+        // Imagine a callback that is playing a large sound in menory.
+        // When it gets to the end of the sound it can partially fill
+        // the last buffer with the end of the sound, then zero pad the buffer, then return STOP.
+        // If the callback has no valid data then it should zero-fill the entire buffer.
+        result = write(mCallbackBuffer.get(), mCallbackFrames, timeoutNanos);
+        if ((result != mCallbackFrames)) {
+            if (result >= 0) {
+                // Only wrote some of the frames requested. The stream can be disconnected
+                // or timed out.
+                processCommands();
+                result = isDisconnected() ? AAUDIO_ERROR_DISCONNECTED : AAUDIO_ERROR_TIMEOUT;
             }
-        } else if (callbackResult == AAUDIO_CALLBACK_RESULT_STOP) {
+            maybeCallErrorCallback(result);
+            break;
+        }
+
+        if (callbackResult == AAUDIO_CALLBACK_RESULT_STOP) {
             ALOGD("%s(): callback returned AAUDIO_CALLBACK_RESULT_STOP", __func__);
             result = systemStopInternal();
             break;
diff --git a/media/libaaudio/src/client/AudioStreamInternalPlay.h b/media/libaaudio/src/client/AudioStreamInternalPlay.h
index b51b5d0..4e14f18 100644
--- a/media/libaaudio/src/client/AudioStreamInternalPlay.h
+++ b/media/libaaudio/src/client/AudioStreamInternalPlay.h
@@ -66,6 +66,8 @@
 
     void prepareBuffersForStart() override;
 
+    void prepareBuffersForStop() override;
+
     void advanceClientToMatchServerPosition(int32_t serverMargin) override;
 
     void onFlushFromServer() override;
diff --git a/media/libaaudio/src/fifo/FifoBuffer.cpp b/media/libaaudio/src/fifo/FifoBuffer.cpp
index 5c11882..f3e3bbd 100644
--- a/media/libaaudio/src/fifo/FifoBuffer.cpp
+++ b/media/libaaudio/src/fifo/FifoBuffer.cpp
@@ -150,7 +150,7 @@
 
     getEmptyRoomAvailable(&wrappingBuffer);
 
-    // Read data in one or two parts.
+    // Write data in one or two parts.
     int partIndex = 0;
     while (framesLeft > 0 && partIndex < WrappingBuffer::SIZE) {
         fifo_frames_t framesToWrite = framesLeft;
@@ -192,3 +192,29 @@
         memset(getStorage(), 0, (size_t) numBytes);
     }
 }
+
+fifo_frames_t FifoBuffer::eraseEmptyMemory(fifo_frames_t numFrames) {
+    WrappingBuffer wrappingBuffer;
+    fifo_frames_t framesLeft = numFrames;
+
+    getEmptyRoomAvailable(&wrappingBuffer);
+
+    // Erase data in one or two parts.
+    int partIndex = 0;
+    while (framesLeft > 0 && partIndex < WrappingBuffer::SIZE) {
+        fifo_frames_t framesToWrite = framesLeft;
+        fifo_frames_t framesAvailable = wrappingBuffer.numFrames[partIndex];
+        if (framesAvailable > 0) {
+            if (framesToWrite > framesAvailable) {
+                framesToWrite = framesAvailable;
+            }
+            int32_t numBytes = convertFramesToBytes(framesToWrite);
+            memset(wrappingBuffer.data[partIndex], 0, numBytes);
+            framesLeft -= framesToWrite;
+        } else {
+            break;
+        }
+        partIndex++;
+    }
+    return numFrames - framesLeft; // number erased
+}
diff --git a/media/libaaudio/src/fifo/FifoBuffer.h b/media/libaaudio/src/fifo/FifoBuffer.h
index 7b0aca1..860ccad 100644
--- a/media/libaaudio/src/fifo/FifoBuffer.h
+++ b/media/libaaudio/src/fifo/FifoBuffer.h
@@ -115,6 +115,13 @@
      */
     void eraseMemory();
 
+    /**
+     * Clear some memory after the write pointer.
+     * This can be used to prevent the reader from accidentally reading stale data
+     * in case it is reading asynchronously.
+     */
+    fifo_frames_t eraseEmptyMemory(fifo_frames_t numFrames);
+
 protected:
 
     virtual uint8_t *getStorage() const = 0;