diff --git a/camera/ndk/include/camera/NdkCameraMetadataTags.h b/camera/ndk/include/camera/NdkCameraMetadataTags.h
index c7d2545..ee3e98e 100644
--- a/camera/ndk/include/camera/NdkCameraMetadataTags.h
+++ b/camera/ndk/include/camera/NdkCameraMetadataTags.h
@@ -7100,7 +7100,7 @@
 
     /**
      * <p>The camera device is a monochrome camera that doesn't contain a color filter array,
-     * and the pixel values on U and Y planes are all 128.</p>
+     * and the pixel values on U and V planes are all 128.</p>
      */
     ACAMERA_REQUEST_AVAILABLE_CAPABILITIES_MONOCHROME                = 12,
 
diff --git a/media/extractors/mp4/ItemTable.cpp b/media/extractors/mp4/ItemTable.cpp
index 26b8251..a1f6e9a 100644
--- a/media/extractors/mp4/ItemTable.cpp
+++ b/media/extractors/mp4/ItemTable.cpp
@@ -1591,10 +1591,9 @@
 
     ssize_t thumbItemIndex = mItemIdToItemMap.indexOfKey(masterImage.thumbnails[0]);
     if (thumbItemIndex < 0) {
-        ALOGW("%s: Thumbnail item id %d not found, use master instead",
-                __FUNCTION__, masterImage.thumbnails[0]);
-        *itemIndex = masterItemIndex;
-        return OK;
+        // Do not return the master image in this case, fail it so that the
+        // thumbnail extraction code knows we really don't have it.
+        return INVALID_OPERATION;
     }
 
     *itemIndex = thumbItemIndex;
diff --git a/media/libaaudio/examples/input_monitor/src/input_monitor.cpp b/media/libaaudio/examples/input_monitor/src/input_monitor.cpp
index e5ad2d9..c1ff34b 100644
--- a/media/libaaudio/examples/input_monitor/src/input_monitor.cpp
+++ b/media/libaaudio/examples/input_monitor/src/input_monitor.cpp
@@ -26,23 +26,22 @@
 #include "AAudioExampleUtils.h"
 #include "AAudioSimpleRecorder.h"
 
-// TODO support FLOAT
-#define REQUIRED_FORMAT    AAUDIO_FORMAT_PCM_I16
 #define MIN_FRAMES_TO_READ 48  /* arbitrary, 1 msec at 48000 Hz */
 
 static const int FRAMES_PER_LINE = 20000;
 
 int main(int argc, const char **argv)
 {
-    AAudioArgsParser   argParser;
-    aaudio_result_t result;
-    AAudioSimpleRecorder recorder;
-    int actualSamplesPerFrame;
-    int actualSampleRate;
-    aaudio_format_t       actualDataFormat;
+    AAudioArgsParser      argParser;
+    AAudioSimpleRecorder  recorder;
+    AAudioStream         *aaudioStream = nullptr;
 
-    AAudioStream *aaudioStream = nullptr;
+    aaudio_result_t       result;
+    aaudio_format_t       actualDataFormat;
     aaudio_stream_state_t state;
+
+    int32_t actualSamplesPerFrame;
+    int32_t actualSampleRate;
     int32_t framesPerBurst = 0;
     int32_t framesPerRead = 0;
     int32_t framesToRecord = 0;
@@ -50,18 +49,18 @@
     int32_t nextFrameCount = 0;
     int32_t frameCount = 0;
     int32_t xRunCount = 0;
-    int64_t previousFramePosition = -1;
-    int16_t *data = nullptr;
-    float peakLevel = 0.0;
     int32_t deviceId;
 
+    int16_t *shortData = nullptr;
+    float   *floatData = nullptr;
+    float    peakLevel = 0.0;
+
     // Make printf print immediately so that debug info is not stuck
     // in a buffer if we hang or crash.
     setvbuf(stdout, nullptr, _IONBF, (size_t) 0);
 
-    printf("%s - Monitor input level using AAudio read, V0.1.2\n", argv[0]);
+    printf("%s - Monitor input level using AAudio read, V0.1.3\n", argv[0]);
 
-    argParser.setFormat(REQUIRED_FORMAT);
     if (argParser.parseArgs(argc, argv)) {
         return EXIT_FAILURE;
     }
@@ -69,6 +68,7 @@
     result = recorder.open(argParser);
     if (result != AAUDIO_OK) {
         fprintf(stderr, "ERROR -  recorder.open() returned %d\n", result);
+        printf("IMPORTANT - Did you remember to enter:   adb root\n");
         goto finish;
     }
     aaudioStream = recorder.getStream();
@@ -96,17 +96,18 @@
     printf("DataFormat: framesPerRead  = %d\n",framesPerRead);
 
     actualDataFormat = AAudioStream_getFormat(aaudioStream);
-    printf("DataFormat: requested      = %d, actual = %d\n",
-           REQUIRED_FORMAT, actualDataFormat);
-    // TODO handle other data formats
-    assert(actualDataFormat == REQUIRED_FORMAT);
 
     // Allocate a buffer for the PCM_16 audio data.
-    data = new(std::nothrow) int16_t[framesPerRead * actualSamplesPerFrame];
-    if (data == nullptr) {
-        fprintf(stderr, "ERROR - could not allocate data buffer\n");
-        result = AAUDIO_ERROR_NO_MEMORY;
-        goto finish;
+    switch (actualDataFormat) {
+        case AAUDIO_FORMAT_PCM_I16:
+            shortData = new int16_t[framesPerRead * actualSamplesPerFrame];
+            break;
+        case AAUDIO_FORMAT_PCM_FLOAT:
+            floatData = new float[framesPerRead * actualSamplesPerFrame];
+            break;
+        default:
+            fprintf(stderr, "UNEXPECTED FORMAT! %d", actualDataFormat);
+            goto finish;
     }
 
     // Start the stream.
@@ -126,7 +127,12 @@
         // Read audio data from the stream.
         const int64_t timeoutNanos = 1000 * NANOS_PER_MILLISECOND;
         int minFrames = (framesToRecord < framesPerRead) ? framesToRecord : framesPerRead;
-        int actual = AAudioStream_read(aaudioStream, data, minFrames, timeoutNanos);
+        int actual = 0;
+        if (actualDataFormat == AAUDIO_FORMAT_PCM_I16) {
+            actual = AAudioStream_read(aaudioStream, shortData, minFrames, timeoutNanos);
+        } else if (actualDataFormat == AAUDIO_FORMAT_PCM_FLOAT) {
+            actual = AAudioStream_read(aaudioStream, floatData, minFrames, timeoutNanos);
+        }
         if (actual < 0) {
             fprintf(stderr, "ERROR - AAudioStream_read() returned %d\n", actual);
             result = actual;
@@ -140,7 +146,12 @@
 
         // Peak finder.
         for (int frameIndex = 0; frameIndex < actual; frameIndex++) {
-            float sample = data[frameIndex * actualSamplesPerFrame] * (1.0/32768);
+            float sample = 0.0f;
+            if (actualDataFormat == AAUDIO_FORMAT_PCM_I16) {
+                sample = shortData[frameIndex * actualSamplesPerFrame] * (1.0/32768);
+            } else if (actualDataFormat == AAUDIO_FORMAT_PCM_FLOAT) {
+                sample = floatData[frameIndex * actualSamplesPerFrame];
+            }
             if (sample > peakLevel) {
                 peakLevel = sample;
             }
@@ -151,17 +162,15 @@
             displayPeakLevel(peakLevel);
             peakLevel = 0.0;
             nextFrameCount += FRAMES_PER_LINE;
-        }
 
-        // Print timestamps.
-        int64_t framePosition = 0;
-        int64_t frameTime = 0;
-        aaudio_result_t timeResult;
-        timeResult = AAudioStream_getTimestamp(aaudioStream, CLOCK_MONOTONIC,
-                                               &framePosition, &frameTime);
+            // Print timestamps.
+            int64_t framePosition = 0;
+            int64_t frameTime = 0;
+            aaudio_result_t timeResult;
+            timeResult = AAudioStream_getTimestamp(aaudioStream, CLOCK_MONOTONIC,
+                                                   &framePosition, &frameTime);
 
-        if (timeResult == AAUDIO_OK) {
-            if (framePosition > (previousFramePosition + FRAMES_PER_LINE)) {
+            if (timeResult == AAUDIO_OK) {
                 int64_t realTime = getNanoseconds();
                 int64_t framesRead = AAudioStream_getFramesRead(aaudioStream);
 
@@ -175,11 +184,15 @@
                        (long long) framePosition,
                        (long long) frameTime,
                        latencyMillis);
-                previousFramePosition = framePosition;
+            } else {
+                printf("WARNING - AAudioStream_getTimestamp() returned %d\n", timeResult);
             }
         }
     }
 
+    state = AAudioStream_getState(aaudioStream);
+    printf("after loop, state = %s\n", AAudio_convertStreamStateToText(state));
+
     xRunCount = AAudioStream_getXRunCount(aaudioStream);
     printf("AAudioStream_getXRunCount %d\n", xRunCount);
 
@@ -192,7 +205,8 @@
 
 finish:
     recorder.close();
-    delete[] data;
+    delete[] shortData;
+    delete[] floatData;
     printf("exiting - AAudio result = %d = %s\n", result, AAudio_convertResultToText(result));
     return (result != AAUDIO_OK) ? EXIT_FAILURE : EXIT_SUCCESS;
 }
diff --git a/media/libaaudio/examples/input_monitor/src/input_monitor_callback.cpp b/media/libaaudio/examples/input_monitor/src/input_monitor_callback.cpp
index 893795b..d10f812 100644
--- a/media/libaaudio/examples/input_monitor/src/input_monitor_callback.cpp
+++ b/media/libaaudio/examples/input_monitor/src/input_monitor_callback.cpp
@@ -26,29 +26,39 @@
 #include "AAudioExampleUtils.h"
 #include "AAudioSimpleRecorder.h"
 
-#define NUM_SECONDS           5
-
-int main(int argc, char **argv)
+int main(int argc, const char **argv)
 {
-    (void)argc; // unused
-    AAudioSimpleRecorder recorder;
-    PeakTrackerData_t myData = {0.0};
-    aaudio_result_t result;
+    AAudioArgsParser      argParser;
+    AAudioSimpleRecorder  recorder;
+    PeakTrackerData_t     myData = {0.0};
+    AAudioStream         *aaudioStream = nullptr;
+    aaudio_result_t       result;
     aaudio_stream_state_t state;
+
+    int       loopsNeeded = 0;
     const int displayRateHz = 20; // arbitrary
-    const int loopsNeeded = NUM_SECONDS * displayRateHz;
 
     // Make printf print immediately so that debug info is not stuck
     // in a buffer if we hang or crash.
     setvbuf(stdout, nullptr, _IONBF, (size_t) 0);
-    printf("%s - Display audio input using an AAudio callback, V0.1.2\n", argv[0]);
+    printf("%s - Display audio input using an AAudio callback, V0.1.3\n", argv[0]);
 
-    result = recorder.open(2, 48000, AAUDIO_FORMAT_PCM_I16,
-                       SimpleRecorderDataCallbackProc, SimpleRecorderErrorCallbackProc, &myData);
+    if (argParser.parseArgs(argc, argv)) {
+        return EXIT_FAILURE;
+    }
+
+    result = recorder.open(argParser,
+                           SimpleRecorderDataCallbackProc,
+                           SimpleRecorderErrorCallbackProc,
+                           &myData);
     if (result != AAUDIO_OK) {
         fprintf(stderr, "ERROR -  recorder.open() returned %d\n", result);
+        printf("IMPORTANT - Did you remember to enter:   adb root\n");
         goto error;
     }
+    aaudioStream = recorder.getStream();
+    argParser.compareWithStream(aaudioStream);
+
     printf("recorder.getFramesPerSecond() = %d\n", recorder.getFramesPerSecond());
     printf("recorder.getSamplesPerFrame() = %d\n", recorder.getSamplesPerFrame());
 
@@ -58,7 +68,9 @@
         goto error;
     }
 
-    printf("Sleep for %d seconds while audio record in a callback thread.\n", NUM_SECONDS);
+    printf("Sleep for %d seconds while audio record in a callback thread.\n",
+           argParser.getDurationSeconds());
+    loopsNeeded = argParser.getDurationSeconds() * displayRateHz;
     for (int i = 0; i < loopsNeeded; i++)
     {
         const struct timespec request = { .tv_sec = 0,
@@ -67,7 +79,7 @@
         printf("%08d: ", (int)recorder.getFramesRead());
         displayPeakLevel(myData.peakLevel);
 
-        result = AAudioStream_waitForStateChange(recorder.getStream(),
+        result = AAudioStream_waitForStateChange(aaudioStream,
                                                  AAUDIO_STREAM_STATE_CLOSED,
                                                  &state,
                                                  0);
@@ -93,7 +105,8 @@
         goto error;
     }
 
-    printf("Sleep for %d seconds while audio records in a callback thread.\n", NUM_SECONDS);
+    printf("Sleep for %d seconds while audio records in a callback thread.\n",
+           argParser.getDurationSeconds());
     for (int i = 0; i < loopsNeeded; i++)
     {
         const struct timespec request = { .tv_sec = 0,
@@ -102,13 +115,14 @@
         printf("%08d: ", (int)recorder.getFramesRead());
         displayPeakLevel(myData.peakLevel);
 
-        state = AAudioStream_getState(recorder.getStream());
+        state = AAudioStream_getState(aaudioStream);
         if (state != AAUDIO_STREAM_STATE_STARTING && state != AAUDIO_STREAM_STATE_STARTED) {
             printf("Stream state is %d %s!\n", state, AAudio_convertStreamStateToText(state));
             break;
         }
     }
     printf("Woke up now.\n");
+    argParser.compareWithStream(aaudioStream);
 
     result = recorder.stop();
     if (result != AAUDIO_OK) {
diff --git a/media/libaaudio/examples/loopback/src/loopback.cpp b/media/libaaudio/examples/loopback/src/loopback.cpp
index 39d079e..026ff0f 100644
--- a/media/libaaudio/examples/loopback/src/loopback.cpp
+++ b/media/libaaudio/examples/loopback/src/loopback.cpp
@@ -151,8 +151,7 @@
 static void MyErrorCallbackProc(
         AAudioStream *stream __unused,
         void *userData __unused,
-        aaudio_result_t error)
-{
+        aaudio_result_t error) {
     printf("Error Callback, error: %d\n",(int)error);
     LoopbackData *myData = (LoopbackData *) userData;
     myData->outputError = error;
diff --git a/media/libaaudio/examples/utils/AAudioArgsParser.h b/media/libaaudio/examples/utils/AAudioArgsParser.h
index eb6925a..88d7401 100644
--- a/media/libaaudio/examples/utils/AAudioArgsParser.h
+++ b/media/libaaudio/examples/utils/AAudioArgsParser.h
@@ -87,7 +87,6 @@
     return;
 }
 
-// TODO use this as a base class within AAudio
 class AAudioParameters {
 public:
 
@@ -262,6 +261,9 @@
                 case 'd':
                     setDeviceId(atoi(&arg[2]));
                     break;
+                case 'f':
+                    setFormat(atoi(&arg[2]));
+                    break;
                 case 'i':
                     setInputPreset(atoi(&arg[2]));
                     break;
@@ -326,6 +328,10 @@
         printf("      -b{bufferCapacity} frames\n");
         printf("      -c{channels} for example 2 for stereo\n");
         printf("      -d{deviceId} default is %d\n", AAUDIO_UNSPECIFIED);
+        printf("      -f{0|1|2} set format\n");
+        printf("          0 = UNSPECIFIED\n");
+        printf("          1 = PCM_I16\n");
+        printf("          2 = FLOAT\n");
         printf("      -i{inputPreset} eg. 5 for AAUDIO_INPUT_PRESET_CAMCORDER\n");
         printf("      -m{0|1|2|3} set MMAP policy\n");
         printf("          0 = _UNSPECIFIED, use aaudio.mmap_policy system property, default\n");
diff --git a/media/libaaudio/examples/write_sine/src/write_sine.cpp b/media/libaaudio/examples/write_sine/src/write_sine.cpp
index 38e1e4c..8e33a31 100644
--- a/media/libaaudio/examples/write_sine/src/write_sine.cpp
+++ b/media/libaaudio/examples/write_sine/src/write_sine.cpp
@@ -57,7 +57,7 @@
     // in a buffer if we hang or crash.
     setvbuf(stdout, nullptr, _IONBF, (size_t) 0);
 
-    printf("%s - Play a sine wave using AAudio V0.1.2\n", argv[0]);
+    printf("%s - Play a sine wave using AAudio V0.1.3\n", argv[0]);
 
     if (argParser.parseArgs(argc, argv)) {
         return EXIT_FAILURE;
diff --git a/media/libaaudio/examples/write_sine/src/write_sine_callback.cpp b/media/libaaudio/examples/write_sine/src/write_sine_callback.cpp
index e167773..e33e9f8 100644
--- a/media/libaaudio/examples/write_sine/src/write_sine_callback.cpp
+++ b/media/libaaudio/examples/write_sine/src/write_sine_callback.cpp
@@ -204,7 +204,7 @@
     AAudioArgsParser::usage();
     printf("      -l{count} loopCount start/stop, every other one is silent\n");
     printf("      -t{msec}  play a high pitched tone at the beginning\n");
-    printf("      -f        force periodic underruns by sleeping in callback\n");
+    printf("      -z        force periodic underruns by sleeping in callback\n");
 }
 
 int main(int argc, const char **argv)
@@ -219,7 +219,7 @@
     // in a buffer if we hang or crash.
     setvbuf(stdout, nullptr, _IONBF, (size_t) 0);
 
-    printf("%s - Play a sine sweep using an AAudio callback V0.1.3\n", argv[0]);
+    printf("%s - Play a sine sweep using an AAudio callback V0.1.4\n", argv[0]);
 
     for (int i = 1; i < argc; i++) {
         const char *arg = argv[i];
@@ -234,8 +234,8 @@
                     case 't':
                         prefixToneMsec = atoi(&arg[2]);
                         break;
-                    case 'f':
-                        forceUnderruns = true;
+                    case 'z':
+                        forceUnderruns = true;  // Zzzzzzz
                         break;
                     default:
                         usage();
diff --git a/media/libaaudio/include/aaudio/AAudio.h b/media/libaaudio/include/aaudio/AAudio.h
index e40a6cd..2207cb8c 100644
--- a/media/libaaudio/include/aaudio/AAudio.h
+++ b/media/libaaudio/include/aaudio/AAudio.h
@@ -146,6 +146,8 @@
  * to make more refined volume or routing decisions.
  *
  * Note that these match the equivalent values in AudioAttributes in the Android Java API.
+ *
+ * Added in API level 28.
  */
 enum {
     /**
@@ -220,6 +222,8 @@
  * enforce audio focus.
  *
  * Note that these match the equivalent values in AudioAttributes in the Android Java API.
+ *
+ * Added in API level 28.
  */
 enum {
 
@@ -252,6 +256,8 @@
  * configuration.
  *
  * Note that these match the equivalent values in MediaRecorder.AudioSource in the Android Java API.
+ *
+ * Added in API level 28.
  */
 enum {
     /**
@@ -288,6 +294,8 @@
      * Do not allocate a session ID.
      * Effects cannot be used with this stream.
      * Default.
+     *
+     * Added in API level 28.
      */
     AAUDIO_SESSION_ID_NONE = -1,
 
@@ -297,6 +305,8 @@
      * Note that the use of this flag may result in higher latency.
      *
      * Note that this matches the value of AudioManager.AUDIO_SESSION_ID_GENERATE.
+     *
+     * Added in API level 28.
      */
     AAUDIO_SESSION_ID_ALLOCATE = 0,
 };
@@ -481,6 +491,8 @@
  *
  * The default, if you do not call this function, is AAUDIO_USAGE_MEDIA.
  *
+ * Added in API level 28.
+ *
  * @param builder reference provided by AAudio_createStreamBuilder()
  * @param usage the desired usage, eg. AAUDIO_USAGE_GAME
  */
@@ -496,6 +508,8 @@
  *
  * The default, if you do not call this function, is AAUDIO_CONTENT_TYPE_MUSIC.
  *
+ * Added in API level 28.
+ *
  * @param builder reference provided by AAudio_createStreamBuilder()
  * @param contentType the type of audio data, eg. AAUDIO_CONTENT_TYPE_SPEECH
  */
@@ -514,6 +528,8 @@
  * That is because VOICE_RECOGNITION is the preset with the lowest latency
  * on many platforms.
  *
+ * Added in API level 28.
+ *
  * @param builder reference provided by AAudio_createStreamBuilder()
  * @param inputPreset the desired configuration for recording
  */
@@ -540,6 +556,8 @@
  *
  * Allocated session IDs will always be positive and nonzero.
  *
+ * Added in API level 28.
+ *
  * @param builder reference provided by AAudio_createStreamBuilder()
  * @param sessionId an allocated sessionID or AAUDIO_SESSION_ID_ALLOCATE
  */
@@ -1059,6 +1077,8 @@
  *
  * The sessionID for a stream should not change once the stream has been opened.
  *
+ * Added in API level 28.
+ *
  * @param stream reference provided by AAudioStreamBuilder_openStream()
  * @return session ID or AAUDIO_SESSION_ID_NONE
  */
@@ -1094,6 +1114,8 @@
 /**
  * Return the use case for the stream.
  *
+ * Added in API level 28.
+ *
  * @param stream reference provided by AAudioStreamBuilder_openStream()
  * @return frames read
  */
@@ -1102,6 +1124,8 @@
 /**
  * Return the content type for the stream.
  *
+ * Added in API level 28.
+ *
  * @param stream reference provided by AAudioStreamBuilder_openStream()
  * @return content type, for example AAUDIO_CONTENT_TYPE_MUSIC
  */
@@ -1110,6 +1134,8 @@
 /**
  * Return the input preset for the stream.
  *
+ * Added in API level 28.
+ *
  * @param stream reference provided by AAudioStreamBuilder_openStream()
  * @return input preset, for example AAUDIO_INPUT_PRESET_CAMCORDER
  */
diff --git a/media/libaaudio/src/Android.bp b/media/libaaudio/src/Android.bp
index 788833b..b9e28a0 100644
--- a/media/libaaudio/src/Android.bp
+++ b/media/libaaudio/src/Android.bp
@@ -57,6 +57,7 @@
 
     shared_libs: [
         "libaudioclient",
+        "libaudioutils",
         "liblog",
         "libcutils",
         "libutils",
diff --git a/media/libaaudio/src/client/AudioStreamInternal.cpp b/media/libaaudio/src/client/AudioStreamInternal.cpp
index 2a3e668..0bdfeac 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternal.cpp
@@ -156,7 +156,7 @@
     setInputPreset(configurationOutput.getInputPreset());
 
     // Save device format so we can do format conversion and volume scaling together.
-    mDeviceFormat = configurationOutput.getFormat();
+    setDeviceFormat(configurationOutput.getFormat());
 
     result = mServiceInterface.getStreamDescription(mServiceStreamHandle, mEndPointParcelable);
     if (result != AAUDIO_OK) {
@@ -501,9 +501,9 @@
             ALOGW("%s - AAUDIO_SERVICE_EVENT_DISCONNECTED - FIFO cleared", __func__);
             break;
         case AAUDIO_SERVICE_EVENT_VOLUME:
+            ALOGD("%s - AAUDIO_SERVICE_EVENT_VOLUME %lf", __func__, message->event.dataDouble);
             mStreamVolume = (float)message->event.dataDouble;
             doSetVolume();
-            ALOGD("%s - AAUDIO_SERVICE_EVENT_VOLUME %lf", __func__, message->event.dataDouble);
             break;
         case AAUDIO_SERVICE_EVENT_XRUN:
             mXRunCount = static_cast<int32_t>(message->event.dataLong);
diff --git a/media/libaaudio/src/client/AudioStreamInternal.h b/media/libaaudio/src/client/AudioStreamInternal.h
index 0e0724b..0425cd5 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.h
+++ b/media/libaaudio/src/client/AudioStreamInternal.h
@@ -138,8 +138,6 @@
     // Calculate timeout for an operation involving framesPerOperation.
     int64_t calculateReasonableTimeout(int32_t framesPerOperation);
 
-    aaudio_format_t getDeviceFormat() const { return mDeviceFormat; }
-
     int32_t getDeviceChannelCount() const { return mDeviceChannelCount; }
 
     /**
@@ -195,9 +193,6 @@
 
     int64_t                  mServiceLatencyNanos = 0;
 
-    // Sometimes the hardware is operating with a different format or channel count from the app.
-    // Then we require conversion in AAudio.
-    aaudio_format_t          mDeviceFormat = AAUDIO_FORMAT_UNSPECIFIED;
     int32_t                  mDeviceChannelCount = 0;
 };
 
diff --git a/media/libaaudio/src/client/AudioStreamInternalPlay.cpp b/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
index 0c3b1fa..795ba2c 100644
--- a/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
@@ -117,7 +117,7 @@
         // Still haven't got any timestamps from server.
         // Keep waiting until we get some valid timestamps then start writing to the
         // current buffer position.
-        ALOGD("%s() wait for valid timestamps", __func__);
+        ALOGV("%s() wait for valid timestamps", __func__);
         // Sleep very briefly and hope we get a timestamp soon.
         *wakeTimePtr = currentNanoTime + (2000 * AAUDIO_NANOS_PER_MICROSECOND);
         ATRACE_END();
@@ -310,6 +310,9 @@
 //------------------------------------------------------------------------------
 // Implementation of PlayerBase
 status_t AudioStreamInternalPlay::doSetVolume() {
-    mVolumeRamp.setTarget(mStreamVolume * getDuckAndMuteVolume());
+    float combinedVolume = mStreamVolume * getDuckAndMuteVolume();
+    ALOGD("%s() mStreamVolume * duckAndMuteVolume = %f * %f = %f",
+          __func__, mStreamVolume, getDuckAndMuteVolume(), combinedVolume);
+    mVolumeRamp.setTarget(combinedVolume);
     return android::NO_ERROR;
 }
diff --git a/media/libaaudio/src/core/AudioStream.cpp b/media/libaaudio/src/core/AudioStream.cpp
index 61e03db..358021b 100644
--- a/media/libaaudio/src/core/AudioStream.cpp
+++ b/media/libaaudio/src/core/AudioStream.cpp
@@ -367,7 +367,6 @@
     return err ? AAudioConvert_androidToAAudioResult(-errno) : mThreadRegistrationResult;
 }
 
-
 aaudio_data_callback_result_t AudioStream::maybeCallDataCallback(void *audioData,
                                                                  int32_t numFrames) {
     aaudio_data_callback_result_t result = AAUDIO_CALLBACK_RESULT_STOP;
@@ -429,6 +428,12 @@
 }
 #endif
 
+void AudioStream::setDuckAndMuteVolume(float duckAndMuteVolume) {
+    ALOGD("%s() to %f", __func__, duckAndMuteVolume);
+    mDuckAndMuteVolume = duckAndMuteVolume;
+    doSetVolume(); // apply this change
+}
+
 AudioStream::MyPlayerBase::MyPlayerBase(AudioStream *parent) : mParent(parent) {
 }
 
@@ -450,7 +455,6 @@
     }
 }
 
-
 void AudioStream::MyPlayerBase::destroy() {
     unregisterWithAudioManager();
 }
diff --git a/media/libaaudio/src/core/AudioStream.h b/media/libaaudio/src/core/AudioStream.h
index 5273e36..31b895c 100644
--- a/media/libaaudio/src/core/AudioStream.h
+++ b/media/libaaudio/src/core/AudioStream.h
@@ -252,6 +252,20 @@
         return AAudioConvert_formatToSizeInBytes(mFormat);
     }
 
+    /**
+     * This is only valid after setSamplesPerFrame() and setDeviceFormat() have been called.
+     */
+    int32_t getBytesPerDeviceFrame() const {
+        return mSamplesPerFrame * getBytesPerDeviceSample();
+    }
+
+    /**
+     * This is only valid after setDeviceFormat() has been called.
+     */
+    int32_t getBytesPerDeviceSample() const {
+        return AAudioConvert_formatToSizeInBytes(getDeviceFormat());
+    }
+
     virtual int64_t getFramesWritten() = 0;
 
     virtual int64_t getFramesRead() = 0;
@@ -314,10 +328,7 @@
     }
 
     // This is used by the AudioManager to duck and mute the stream when changing audio focus.
-    void setDuckAndMuteVolume(float duckAndMuteVolume) {
-        mDuckAndMuteVolume = duckAndMuteVolume;
-        doSetVolume(); // apply this change
-    }
+    void setDuckAndMuteVolume(float duckAndMuteVolume);
 
     float getDuckAndMuteVolume() const {
         return mDuckAndMuteVolume;
@@ -471,6 +482,17 @@
         mFormat = format;
     }
 
+    /**
+     * This should not be called after the open() call.
+     */
+    void setDeviceFormat(aaudio_format_t format) {
+        mDeviceFormat = format;
+    }
+
+    aaudio_format_t getDeviceFormat() const {
+        return mDeviceFormat;
+    }
+
     void setState(aaudio_stream_state_t state);
 
     void setDeviceId(int32_t deviceId) {
@@ -485,9 +507,23 @@
 
     float                mDuckAndMuteVolume = 1.0f;
 
-
 protected:
 
+    /**
+     * Either convert the data from device format to app format and return a pointer
+     * to the conversion buffer,
+     * OR just pass back the original pointer.
+     *
+     * Note that this is only used for the INPUT path.
+     *
+     * @param audioData
+     * @param numFrames
+     * @return original pointer or the conversion buffer
+     */
+    virtual const void * maybeConvertDeviceData(const void *audioData, int32_t numFrames) {
+        return audioData;
+    }
+
     void setPeriodNanoseconds(int64_t periodNanoseconds) {
         mPeriodNanoseconds.store(periodNanoseconds, std::memory_order_release);
     }
@@ -539,6 +575,10 @@
 
     int32_t                     mSessionId = AAUDIO_UNSPECIFIED;
 
+    // Sometimes the hardware is operating with a different format from the app.
+    // Then we require conversion in AAudio.
+    aaudio_format_t             mDeviceFormat = AAUDIO_FORMAT_UNSPECIFIED;
+
     // callback ----------------------------------
 
     AAudioStream_dataCallback   mDataCallbackProc = nullptr;  // external callback functions
diff --git a/media/libaaudio/src/core/AudioStreamBuilder.cpp b/media/libaaudio/src/core/AudioStreamBuilder.cpp
index 293a6a8..3a7a578 100644
--- a/media/libaaudio/src/core/AudioStreamBuilder.cpp
+++ b/media/libaaudio/src/core/AudioStreamBuilder.cpp
@@ -87,7 +87,7 @@
             break;
 
         default:
-            ALOGE("bad direction = %d", direction);
+            ALOGE("%s() bad direction = %d", __func__, direction);
             result = AAUDIO_ERROR_ILLEGAL_ARGUMENT;
     }
     return result;
@@ -99,7 +99,7 @@
 aaudio_result_t AudioStreamBuilder::build(AudioStream** streamPtr) {
     AudioStream *audioStream = nullptr;
     if (streamPtr == nullptr) {
-        ALOGE("build() streamPtr is null");
+        ALOGE("%s() streamPtr is null", __func__);
         return AAUDIO_ERROR_NULL;
     }
     *streamPtr = nullptr;
@@ -124,13 +124,11 @@
     if (mapExclusivePolicy == AAUDIO_UNSPECIFIED) {
         mapExclusivePolicy = AAUDIO_MMAP_EXCLUSIVE_POLICY_DEFAULT;
     }
-    ALOGD("mmapPolicy = %d, mapExclusivePolicy = %d",
-          mmapPolicy, mapExclusivePolicy);
 
     aaudio_sharing_mode_t sharingMode = getSharingMode();
     if ((sharingMode == AAUDIO_SHARING_MODE_EXCLUSIVE)
         && (mapExclusivePolicy == AAUDIO_POLICY_NEVER)) {
-        ALOGW("EXCLUSIVE sharing mode not supported. Use SHARED.");
+        ALOGD("%s() EXCLUSIVE sharing mode not supported. Use SHARED.", __func__);
         sharingMode = AAUDIO_SHARING_MODE_SHARED;
         setSharingMode(sharingMode);
     }
@@ -141,13 +139,14 @@
     // TODO Support other performance settings in MMAP mode.
     // Disable MMAP if low latency not requested.
     if (getPerformanceMode() != AAUDIO_PERFORMANCE_MODE_LOW_LATENCY) {
-        ALOGD("build() MMAP not available because AAUDIO_PERFORMANCE_MODE_LOW_LATENCY not used.");
+        ALOGD("%s() MMAP not available because AAUDIO_PERFORMANCE_MODE_LOW_LATENCY not used.",
+              __func__);
         allowMMap = false;
     }
 
     // SessionID and Effects are only supported in Legacy mode.
     if (getSessionId() != AAUDIO_SESSION_ID_NONE) {
-        ALOGD("build() MMAP not available because sessionId used.");
+        ALOGD("%s() MMAP not available because sessionId used.", __func__);
         allowMMap = false;
     }
 
@@ -163,7 +162,7 @@
             audioStream = nullptr;
 
             if (isMMap && allowLegacy) {
-                ALOGD("build() MMAP stream did not open so try Legacy path");
+                ALOGV("%s() MMAP stream did not open so try Legacy path", __func__);
                 // If MMAP stream failed to open then TRY using a legacy stream.
                 result = builder_createStream(getDirection(), sharingMode,
                                               false, &audioStream);
diff --git a/media/libaaudio/src/fifo/FifoBuffer.cpp b/media/libaaudio/src/fifo/FifoBuffer.cpp
index e6e7c8e..9b9744e 100644
--- a/media/libaaudio/src/fifo/FifoBuffer.cpp
+++ b/media/libaaudio/src/fifo/FifoBuffer.cpp
@@ -43,7 +43,7 @@
     int32_t bytesPerBuffer = bytesPerFrame * capacityInFrames;
     mStorage = new uint8_t[bytesPerBuffer];
     mStorageOwned = true;
-    ALOGD("capacityInFrames = %d, bytesPerFrame = %d",
+    ALOGV("capacityInFrames = %d, bytesPerFrame = %d",
           capacityInFrames, bytesPerFrame);
 }
 
diff --git a/media/libaaudio/src/legacy/AudioStreamLegacy.cpp b/media/libaaudio/src/legacy/AudioStreamLegacy.cpp
index 3352b33..a6b9f5d 100644
--- a/media/libaaudio/src/legacy/AudioStreamLegacy.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamLegacy.cpp
@@ -19,10 +19,12 @@
 #include <utils/Log.h>
 
 #include <stdint.h>
-#include <utils/String16.h>
+
+#include <aaudio/AAudio.h>
+#include <audio_utils/primitives.h>
 #include <media/AudioTrack.h>
 #include <media/AudioTimestamp.h>
-#include <aaudio/AAudio.h>
+#include <utils/String16.h>
 
 #include "core/AudioStream.h"
 #include "legacy/AudioStreamLegacy.h"
@@ -48,14 +50,17 @@
     return AudioStreamLegacy_callback;
 }
 
-aaudio_data_callback_result_t AudioStreamLegacy::callDataCallbackFrames(uint8_t *buffer, int32_t numFrames) {
+aaudio_data_callback_result_t AudioStreamLegacy::callDataCallbackFrames(uint8_t *buffer,
+                                                                        int32_t numFrames) {
+    void *finalAudioData = buffer;
     if (getDirection() == AAUDIO_DIRECTION_INPUT) {
         // Increment before because we already got the data from the device.
         incrementFramesRead(numFrames);
+        finalAudioData = (void *) maybeConvertDeviceData(buffer, numFrames);
     }
 
     // Call using the AAudio callback interface.
-    aaudio_data_callback_result_t callbackResult = maybeCallDataCallback(buffer, numFrames);
+    aaudio_data_callback_result_t callbackResult = maybeCallDataCallback(finalAudioData, numFrames);
 
     if (callbackResult == AAUDIO_CALLBACK_RESULT_CONTINUE
             && getDirection() == AAUDIO_DIRECTION_OUTPUT) {
@@ -67,15 +72,15 @@
 
 // Implement FixedBlockProcessor
 int32_t AudioStreamLegacy::onProcessFixedBlock(uint8_t *buffer, int32_t numBytes) {
-    int32_t numFrames = numBytes / getBytesPerFrame();
+    int32_t numFrames = numBytes / getBytesPerDeviceFrame();
     return (int32_t) callDataCallbackFrames(buffer, numFrames);
 }
 
 void AudioStreamLegacy::processCallbackCommon(aaudio_callback_operation_t opcode, void *info) {
     aaudio_data_callback_result_t callbackResult;
-    // This illegal size can be used to AudioFlinger to stop calling us.
+    // This illegal size can be used to tell AudioFlinger to stop calling us.
     // This takes advantage of AudioFlinger killing the stream.
-    // TODO need API change in AudioRecord and AudioTrack
+    // TODO add to API in AudioRecord and AudioTrack
     const size_t SIZE_STOP_CALLBACKS = SIZE_MAX;
 
     switch (opcode) {
@@ -100,7 +105,7 @@
 
                 // If the caller specified an exact size then use a block size adapter.
                 if (mBlockAdapter != nullptr) {
-                    int32_t byteCount = audioBuffer->frameCount * getBytesPerFrame();
+                    int32_t byteCount = audioBuffer->frameCount * getBytesPerDeviceFrame();
                     callbackResult = mBlockAdapter->processVariableBlock(
                             (uint8_t *) audioBuffer->raw, byteCount);
                 } else {
@@ -109,7 +114,7 @@
                                                             audioBuffer->frameCount);
                 }
                 if (callbackResult == AAUDIO_CALLBACK_RESULT_CONTINUE) {
-                    audioBuffer->size = audioBuffer->frameCount * getBytesPerFrame();
+                    audioBuffer->size = audioBuffer->frameCount * getBytesPerDeviceFrame();
                 } else { // STOP or invalid result
                     ALOGW("%s() callback requested stop, fake an error", __func__);
                     audioBuffer->size = SIZE_STOP_CALLBACKS;
@@ -179,19 +184,17 @@
     int64_t localPosition;
     status_t status = extendedTimestamp->getBestTimestamp(&localPosition, timeNanoseconds,
                                                           timebase, &location);
-    // use MonotonicCounter to prevent retrograde motion.
-    mTimestampPosition.update32((int32_t)localPosition);
-    *framePosition = mTimestampPosition.get();
+    if (status == OK) {
+        // use MonotonicCounter to prevent retrograde motion.
+        mTimestampPosition.update32((int32_t) localPosition);
+        *framePosition = mTimestampPosition.get();
+    }
 
 //    ALOGD("getBestTimestamp() fposition: server = %6lld, kernel = %6lld, location = %d",
 //          (long long) extendedTimestamp->mPosition[ExtendedTimestamp::Location::LOCATION_SERVER],
 //          (long long) extendedTimestamp->mPosition[ExtendedTimestamp::Location::LOCATION_KERNEL],
 //          (int)location);
-    if (status == WOULD_BLOCK) {
-        return AAUDIO_ERROR_INVALID_STATE;
-    } else {
-        return AAudioConvert_androidToAAudioResult(status);
-    }
+    return AAudioConvert_androidToAAudioResult(status);
 }
 
 void AudioStreamLegacy::onAudioDeviceUpdate(audio_port_handle_t deviceId)
diff --git a/media/libaaudio/src/legacy/AudioStreamRecord.cpp b/media/libaaudio/src/legacy/AudioStreamRecord.cpp
index 28158e2..1981ba3 100644
--- a/media/libaaudio/src/legacy/AudioStreamRecord.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamRecord.cpp
@@ -19,13 +19,15 @@
 #include <utils/Log.h>
 
 #include <stdint.h>
-#include <utils/String16.h>
-#include <media/AudioRecord.h>
-#include <aaudio/AAudio.h>
 
-#include "AudioClock.h"
+#include <aaudio/AAudio.h>
+#include <audio_utils/primitives.h>
+#include <media/AudioRecord.h>
+#include <utils/String16.h>
+
 #include "legacy/AudioStreamLegacy.h"
 #include "legacy/AudioStreamRecord.h"
+#include "utility/AudioClock.h"
 #include "utility/FixedBlockWriter.h"
 
 using namespace android;
@@ -63,10 +65,6 @@
     size_t frameCount = (builder.getBufferCapacity() == AAUDIO_UNSPECIFIED) ? 0
                         : builder.getBufferCapacity();
 
-    // TODO implement an unspecified Android format then use that.
-    audio_format_t format = (getFormat() == AAUDIO_FORMAT_UNSPECIFIED)
-            ? AUDIO_FORMAT_PCM_FLOAT
-            : AAudioConvert_aaudioToAndroidDataFormat(getFormat());
 
     audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE;
     aaudio_performance_mode_t perfMode = getPerformanceMode();
@@ -82,6 +80,35 @@
             break;
     }
 
+    // Preserve behavior of API 26
+    if (getFormat() == AAUDIO_FORMAT_UNSPECIFIED) {
+        setFormat(AAUDIO_FORMAT_PCM_FLOAT);
+    }
+
+    // Maybe change device format to get a FAST path.
+    // AudioRecord does not support FAST mode for FLOAT data.
+    // TODO AudioRecord should allow FLOAT data paths for FAST tracks.
+    // So IF the user asks for low latency FLOAT
+    // AND the sampleRate is likely to be compatible with FAST
+    // THEN request I16 and convert to FLOAT when passing to user.
+    // Note that hard coding 48000 Hz is not ideal because the sampleRate
+    // for a FAST path might not be 48000 Hz.
+    // It normally is but there is a chance that it is not.
+    // And there is no reliable way to know that in advance.
+    // Luckily the consequences of a wrong guess are minor.
+    // We just may not get a FAST track.
+    // But we wouldn't have anyway without this hack.
+    constexpr int32_t kMostLikelySampleRateForFast = 48000;
+    if (getFormat() == AAUDIO_FORMAT_PCM_FLOAT
+            && perfMode == AAUDIO_PERFORMANCE_MODE_LOW_LATENCY
+            && (samplesPerFrame <= 2) // FAST only for mono and stereo
+            && (getSampleRate() == kMostLikelySampleRateForFast
+                || getSampleRate() == AAUDIO_UNSPECIFIED)) {
+        setDeviceFormat(AAUDIO_FORMAT_PCM_I16);
+    } else {
+        setDeviceFormat(getFormat());
+    }
+
     uint32_t notificationFrames = 0;
 
     // Setup the callback if there is one.
@@ -96,9 +123,6 @@
     }
     mCallbackBufferSize = builder.getFramesPerDataCallback();
 
-    ALOGD("open(), request notificationFrames = %u, frameCount = %u",
-          notificationFrames, (uint)frameCount);
-
     // Don't call mAudioRecord->setInputDevice() because it will be overwritten by set()!
     audio_port_handle_t selectedDeviceId = (getDeviceId() == AAUDIO_UNSPECIFIED)
                                            ? AUDIO_PORT_HANDLE_NONE
@@ -120,39 +144,59 @@
     aaudio_session_id_t requestedSessionId = builder.getSessionId();
     audio_session_t sessionId = AAudioConvert_aaudioToAndroidSessionId(requestedSessionId);
 
-    mAudioRecord = new AudioRecord(
-            mOpPackageName // const String16& opPackageName TODO does not compile
-            );
-    mAudioRecord->set(
-            AUDIO_SOURCE_DEFAULT, // ignored because we pass attributes below
-            getSampleRate(),
-            format,
-            channelMask,
-            frameCount,
-            callback,
-            callbackData,
-            notificationFrames,
-            false /*threadCanCallJava*/,
-            sessionId,
-            streamTransferType,
-            flags,
-            AUDIO_UID_INVALID, // DEFAULT uid
-            -1,                // DEFAULT pid
-            &attributes,
-            selectedDeviceId
-            );
+    // ----------- open the AudioRecord ---------------------
+    // Might retry, but never more than once.
+    for (int i = 0; i < 2; i ++) {
+        audio_format_t requestedInternalFormat =
+                AAudioConvert_aaudioToAndroidDataFormat(getDeviceFormat());
 
-    // Did we get a valid track?
-    status_t status = mAudioRecord->initCheck();
-    if (status != OK) {
-        close();
-        ALOGE("open(), initCheck() returned %d", status);
-        return AAudioConvert_androidToAAudioResult(status);
+        mAudioRecord = new AudioRecord(
+                mOpPackageName // const String16& opPackageName TODO does not compile
+        );
+        mAudioRecord->set(
+                AUDIO_SOURCE_DEFAULT, // ignored because we pass attributes below
+                getSampleRate(),
+                requestedInternalFormat,
+                channelMask,
+                frameCount,
+                callback,
+                callbackData,
+                notificationFrames,
+                false /*threadCanCallJava*/,
+                sessionId,
+                streamTransferType,
+                flags,
+                AUDIO_UID_INVALID, // DEFAULT uid
+                -1,                // DEFAULT pid
+                &attributes,
+                selectedDeviceId
+        );
+
+        // Did we get a valid track?
+        status_t status = mAudioRecord->initCheck();
+        if (status != OK) {
+            close();
+            ALOGE("open(), initCheck() returned %d", status);
+            return AAudioConvert_androidToAAudioResult(status);
+        }
+
+        // Check to see if it was worth hacking the deviceFormat.
+        bool gotFastPath = (mAudioRecord->getFlags() & AUDIO_INPUT_FLAG_FAST)
+                           == AUDIO_INPUT_FLAG_FAST;
+        if (getFormat() != getDeviceFormat() && !gotFastPath) {
+            // We tried to get a FAST path by switching the device format.
+            // But it didn't work. So we might as well reopen using the same
+            // format for device and for app.
+            ALOGD("%s() used a different device format but no FAST path, reopen", __func__);
+            mAudioRecord.clear();
+            setDeviceFormat(getFormat());
+        } else {
+            break; // Keep the one we just opened.
+        }
     }
 
     // Get the actual values from the AudioRecord.
     setSamplesPerFrame(mAudioRecord->channelCount());
-    setFormat(AAudioConvert_androidToAAudioDataFormat(mAudioRecord->format()));
 
     int32_t actualSampleRate = mAudioRecord->getSampleRate();
     ALOGW_IF(actualSampleRate != getSampleRate(),
@@ -169,6 +213,29 @@
         mBlockAdapter = nullptr;
     }
 
+    // Allocate format conversion buffer if needed.
+    if (getDeviceFormat() == AAUDIO_FORMAT_PCM_I16
+        && getFormat() == AAUDIO_FORMAT_PCM_FLOAT) {
+
+        if (builder.getDataCallbackProc() != nullptr) {
+            // If we have a callback then we need to convert the data into an internal float
+            // array and then pass that entire array to the app.
+            mFormatConversionBufferSizeInFrames =
+                    (mCallbackBufferSize != AAUDIO_UNSPECIFIED)
+                    ? mCallbackBufferSize : getFramesPerBurst();
+            int32_t numSamples = mFormatConversionBufferSizeInFrames * getSamplesPerFrame();
+            mFormatConversionBufferFloat = std::make_unique<float[]>(numSamples);
+        } else {
+            // If we don't have a callback then we will read into an internal short array
+            // and then convert into the app float array in read().
+            mFormatConversionBufferSizeInFrames = getFramesPerBurst();
+            int32_t numSamples = mFormatConversionBufferSizeInFrames * getSamplesPerFrame();
+            mFormatConversionBufferI16 = std::make_unique<int16_t[]>(numSamples);
+        }
+        ALOGD("%s() setup I16>FLOAT conversion buffer with %d frames",
+              __func__, mFormatConversionBufferSizeInFrames);
+    }
+
     // Update performance mode based on the actual stream.
     // For example, if the sample rate does not match native then you won't get a FAST track.
     audio_input_flags_t actualFlags = mAudioRecord->getFlags();
@@ -216,6 +283,24 @@
     return AudioStream::close();
 }
 
+const void * AudioStreamRecord::maybeConvertDeviceData(const void *audioData, int32_t numFrames) {
+    if (mFormatConversionBufferFloat.get() != nullptr) {
+        LOG_ALWAYS_FATAL_IF(numFrames > mFormatConversionBufferSizeInFrames,
+                            "%s() conversion size %d too large for buffer %d",
+                            __func__, numFrames, mFormatConversionBufferSizeInFrames);
+
+        int32_t numSamples = numFrames * getSamplesPerFrame();
+        // Only conversion supported is I16 to FLOAT
+        memcpy_to_float_from_i16(
+                    mFormatConversionBufferFloat.get(),
+                    (const int16_t *) audioData,
+                    numSamples);
+        return mFormatConversionBufferFloat.get();
+    } else {
+        return audioData;
+    }
+}
+
 void AudioStreamRecord::processCallback(int event, void *info) {
     switch (event) {
         case AudioRecord::EVENT_MORE_DATA:
@@ -302,9 +387,10 @@
                                       int32_t numFrames,
                                       int64_t timeoutNanoseconds)
 {
-    int32_t bytesPerFrame = getBytesPerFrame();
+    int32_t bytesPerDeviceFrame = getBytesPerDeviceFrame();
     int32_t numBytes;
-    aaudio_result_t result = AAudioConvert_framesToBytes(numFrames, bytesPerFrame, &numBytes);
+    // This will detect out of range values for numFrames.
+    aaudio_result_t result = AAudioConvert_framesToBytes(numFrames, bytesPerDeviceFrame, &numBytes);
     if (result != AAUDIO_OK) {
         return result;
     }
@@ -315,19 +401,49 @@
 
     // TODO add timeout to AudioRecord
     bool blocking = (timeoutNanoseconds > 0);
-    ssize_t bytesRead = mAudioRecord->read(buffer, numBytes, blocking);
-    if (bytesRead == WOULD_BLOCK) {
+
+    ssize_t bytesActuallyRead = 0;
+    ssize_t totalBytesRead = 0;
+    if (mFormatConversionBufferI16.get() != nullptr) {
+        // Convert I16 data to float using an intermediate buffer.
+        float *floatBuffer = (float *) buffer;
+        int32_t framesLeft = numFrames;
+        // Perform conversion using multiple read()s if necessary.
+        while (framesLeft > 0) {
+            // Read into short internal buffer.
+            int32_t framesToRead = std::min(framesLeft, mFormatConversionBufferSizeInFrames);
+            size_t bytesToRead = framesToRead * bytesPerDeviceFrame;
+            bytesActuallyRead = mAudioRecord->read(mFormatConversionBufferI16.get(), bytesToRead, blocking);
+            if (bytesActuallyRead <= 0) {
+                break;
+            }
+            totalBytesRead += bytesActuallyRead;
+            int32_t framesToConvert = bytesActuallyRead / bytesPerDeviceFrame;
+            // Convert into app float buffer.
+            size_t numSamples = framesToConvert * getSamplesPerFrame();
+            memcpy_to_float_from_i16(
+                    floatBuffer,
+                    mFormatConversionBufferI16.get(),
+                    numSamples);
+            floatBuffer += numSamples;
+            framesLeft -= framesToConvert;
+        }
+    } else {
+        bytesActuallyRead = mAudioRecord->read(buffer, numBytes, blocking);
+        totalBytesRead = bytesActuallyRead;
+    }
+    if (bytesActuallyRead == WOULD_BLOCK) {
         return 0;
-    } else if (bytesRead < 0) {
-        // in this context, a DEAD_OBJECT is more likely to be a disconnect notification due to
-        // AudioRecord invalidation
-        if (bytesRead == DEAD_OBJECT) {
+    } else if (bytesActuallyRead < 0) {
+        // In this context, a DEAD_OBJECT is more likely to be a disconnect notification due to
+        // AudioRecord invalidation.
+        if (bytesActuallyRead == DEAD_OBJECT) {
             setState(AAUDIO_STREAM_STATE_DISCONNECTED);
             return AAUDIO_ERROR_DISCONNECTED;
         }
-        return AAudioConvert_androidToAAudioResult(bytesRead);
+        return AAudioConvert_androidToAAudioResult(bytesActuallyRead);
     }
-    int32_t framesRead = (int32_t)(bytesRead / bytesPerFrame);
+    int32_t framesRead = (int32_t)(totalBytesRead / bytesPerDeviceFrame);
     incrementFramesRead(framesRead);
 
     result = updateStateMachine();
diff --git a/media/libaaudio/src/legacy/AudioStreamRecord.h b/media/libaaudio/src/legacy/AudioStreamRecord.h
index c1723ba..2f41d34 100644
--- a/media/libaaudio/src/legacy/AudioStreamRecord.h
+++ b/media/libaaudio/src/legacy/AudioStreamRecord.h
@@ -76,6 +76,8 @@
         return incrementFramesRead(frames);
     }
 
+    const void * maybeConvertDeviceData(const void *audioData, int32_t numFrames) override;
+
 private:
     android::sp<android::AudioRecord> mAudioRecord;
     // adapts between variable sized blocks and fixed size blocks
@@ -83,6 +85,11 @@
 
     // TODO add 64-bit position reporting to AudioRecord and use it.
     android::String16                mOpPackageName;
+
+    // Only one type of conversion buffer is used.
+    std::unique_ptr<float[]>         mFormatConversionBufferFloat;
+    std::unique_ptr<int16_t[]>       mFormatConversionBufferI16;
+    int32_t                          mFormatConversionBufferSizeInFrames = 0;
 };
 
 } /* namespace aaudio */
diff --git a/media/libaaudio/src/legacy/AudioStreamTrack.cpp b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
index 023e8af..9653601 100644
--- a/media/libaaudio/src/legacy/AudioStreamTrack.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
@@ -181,6 +181,7 @@
     aaudio_format_t aaudioFormat =
             AAudioConvert_androidToAAudioDataFormat(mAudioTrack->format());
     setFormat(aaudioFormat);
+    setDeviceFormat(aaudioFormat);
 
     int32_t actualSampleRate = mAudioTrack->getSampleRate();
     ALOGW_IF(actualSampleRate != getSampleRate(),
diff --git a/media/libaudioclient/IAudioPolicyService.cpp b/media/libaudioclient/IAudioPolicyService.cpp
index a49b2cb..f87fcc4 100644
--- a/media/libaudioclient/IAudioPolicyService.cpp
+++ b/media/libaudioclient/IAudioPolicyService.cpp
@@ -872,7 +872,9 @@
         case INIT_STREAM_VOLUME:
         case SET_STREAM_VOLUME:
         case REGISTER_POLICY_MIXES:
-        case SET_MASTER_MONO: {
+        case SET_MASTER_MONO:
+        case START_AUDIO_SOURCE:
+        case STOP_AUDIO_SOURCE: {
             uid_t multiUserClientUid = IPCThreadState::self()->getCallingUid() % AID_USER_OFFSET;
             if (multiUserClientUid >= AID_APP_START) {
                 ALOGW("%s: transaction %d received from PID %d unauthorized UID %d",
@@ -989,7 +991,7 @@
 
         case GET_OUTPUT_FOR_ATTR: {
             CHECK_INTERFACE(IAudioPolicyService, data, reply);
-            audio_attributes_t attr;
+            audio_attributes_t attr = {};
             bool hasAttributes = data.readInt32() != 0;
             if (hasAttributes) {
                 data.read(&attr, sizeof(audio_attributes_t));
@@ -1058,7 +1060,7 @@
 
         case GET_INPUT_FOR_ATTR: {
             CHECK_INTERFACE(IAudioPolicyService, data, reply);
-            audio_attributes_t attr;
+            audio_attributes_t attr = {};
             data.read(&attr, sizeof(audio_attributes_t));
             sanetizeAudioAttributes(&attr);
             audio_io_handle_t input = (audio_io_handle_t)data.readInt32();
@@ -1160,8 +1162,11 @@
 
         case GET_OUTPUT_FOR_EFFECT: {
             CHECK_INTERFACE(IAudioPolicyService, data, reply);
-            effect_descriptor_t desc;
-            data.read(&desc, sizeof(effect_descriptor_t));
+            effect_descriptor_t desc = {};
+            if (data.read(&desc, sizeof(desc)) != NO_ERROR) {
+                android_errorWriteLog(0x534e4554, "73126106");
+            }
+            (void)sanitizeEffectDescriptor(&desc);
             audio_io_handle_t output = getOutputForEffect(&desc);
             reply->writeInt32(static_cast <int>(output));
             return NO_ERROR;
@@ -1169,8 +1174,11 @@
 
         case REGISTER_EFFECT: {
             CHECK_INTERFACE(IAudioPolicyService, data, reply);
-            effect_descriptor_t desc;
-            data.read(&desc, sizeof(effect_descriptor_t));
+            effect_descriptor_t desc = {};
+            if (data.read(&desc, sizeof(desc)) != NO_ERROR) {
+                android_errorWriteLog(0x534e4554, "73126106");
+            }
+            (void)sanitizeEffectDescriptor(&desc);
             audio_io_handle_t io = data.readInt32();
             uint32_t strategy = data.readInt32();
             audio_session_t session = (audio_session_t) data.readInt32();
@@ -1229,7 +1237,7 @@
                 count = AudioEffect::kMaxPreProcessing;
             }
             uint32_t retCount = count;
-            effect_descriptor_t *descriptors = new effect_descriptor_t[count];
+            effect_descriptor_t *descriptors = new effect_descriptor_t[count]{};
             status_t status = queryDefaultPreProcessing(audioSession, descriptors, &retCount);
             reply->writeInt32(status);
             if (status != NO_ERROR && status != NO_MEMORY) {
@@ -1248,7 +1256,7 @@
 
         case IS_OFFLOAD_SUPPORTED: {
             CHECK_INTERFACE(IAudioPolicyService, data, reply);
-            audio_offload_info_t info;
+            audio_offload_info_t info = {};
             data.read(&info, sizeof(audio_offload_info_t));
             bool isSupported = isOffloadSupported(info);
             reply->writeInt32(isSupported);
@@ -1303,7 +1311,7 @@
 
         case CREATE_AUDIO_PATCH: {
             CHECK_INTERFACE(IAudioPolicyService, data, reply);
-            struct audio_patch patch;
+            struct audio_patch patch = {};
             data.read(&patch, sizeof(struct audio_patch));
             audio_patch_handle_t handle = AUDIO_PATCH_HANDLE_NONE;
             if (data.read(&handle, sizeof(audio_patch_handle_t)) != NO_ERROR) {
@@ -1319,7 +1327,7 @@
 
         case RELEASE_AUDIO_PATCH: {
             CHECK_INTERFACE(IAudioPolicyService, data, reply);
-            audio_patch_handle_t handle;
+            audio_patch_handle_t handle = AUDIO_PATCH_HANDLE_NONE;
             data.read(&handle, sizeof(audio_patch_handle_t));
             status_t status = releaseAudioPatch(handle);
             reply->writeInt32(status);
@@ -1358,8 +1366,9 @@
 
         case SET_AUDIO_PORT_CONFIG: {
             CHECK_INTERFACE(IAudioPolicyService, data, reply);
-            struct audio_port_config config;
+            struct audio_port_config config = {};
             data.read(&config, sizeof(struct audio_port_config));
+            (void)sanitizeAudioPortConfig(&config);
             status_t status = setAudioPortConfig(&config);
             reply->writeInt32(status);
             return NO_ERROR;
@@ -1433,9 +1442,10 @@
 
         case START_AUDIO_SOURCE: {
             CHECK_INTERFACE(IAudioPolicyService, data, reply);
-            struct audio_port_config source;
+            struct audio_port_config source = {};
             data.read(&source, sizeof(struct audio_port_config));
-            audio_attributes_t attributes;
+            (void)sanitizeAudioPortConfig(&source);
+            audio_attributes_t attributes = {};
             data.read(&attributes, sizeof(audio_attributes_t));
             sanetizeAudioAttributes(&attributes);
             audio_patch_handle_t handle = AUDIO_PATCH_HANDLE_NONE;
@@ -1488,6 +1498,14 @@
     }
 }
 
+/** returns true if string overflow was prevented by zero termination */
+template <size_t size>
+static bool preventStringOverflow(char (&s)[size]) {
+    if (strnlen(s, size) < size) return false;
+    s[size - 1] = '\0';
+    return true;
+}
+
 void BnAudioPolicyService::sanetizeAudioAttributes(audio_attributes_t* attr)
 {
     const size_t tagsMaxSize = AUDIO_ATTRIBUTES_TAGS_MAX_SIZE;
@@ -1497,6 +1515,27 @@
     attr->tags[tagsMaxSize - 1] = '\0';
 }
 
+/** returns BAD_VALUE if sanitization was required. */
+status_t BnAudioPolicyService::sanitizeEffectDescriptor(effect_descriptor_t* desc)
+{
+    if (preventStringOverflow(desc->name)
+        | /* always */ preventStringOverflow(desc->implementor)) {
+        android_errorWriteLog(0x534e4554, "73126106"); // SafetyNet logging
+        return BAD_VALUE;
+    }
+    return NO_ERROR;
+}
+
+/** returns BAD_VALUE if sanitization was required. */
+status_t BnAudioPolicyService::sanitizeAudioPortConfig(struct audio_port_config* config)
+{
+    if (config->type == AUDIO_PORT_TYPE_DEVICE &&
+        preventStringOverflow(config->ext.device.address)) {
+        return BAD_VALUE;
+    }
+    return NO_ERROR;
+}
+
 // ----------------------------------------------------------------------------
 
 } // namespace android
diff --git a/media/libaudioclient/include/media/IAudioPolicyService.h b/media/libaudioclient/include/media/IAudioPolicyService.h
index 949d593..1114eeb 100644
--- a/media/libaudioclient/include/media/IAudioPolicyService.h
+++ b/media/libaudioclient/include/media/IAudioPolicyService.h
@@ -180,6 +180,8 @@
                                     uint32_t flags = 0);
 private:
     void sanetizeAudioAttributes(audio_attributes_t* attr);
+    status_t sanitizeEffectDescriptor(effect_descriptor_t* desc);
+    status_t sanitizeAudioPortConfig(struct audio_port_config* config);
 };
 
 // ----------------------------------------------------------------------------
diff --git a/media/libaudiohal/2.0/DevicesFactoryHalHybrid.cpp b/media/libaudiohal/2.0/DevicesFactoryHalHybrid.cpp
index 77df6b5..1c4be74 100644
--- a/media/libaudiohal/2.0/DevicesFactoryHalHybrid.cpp
+++ b/media/libaudiohal/2.0/DevicesFactoryHalHybrid.cpp
@@ -32,7 +32,8 @@
 }
 
 status_t DevicesFactoryHalHybrid::openDevice(const char *name, sp<DeviceHalInterface> *device) {
-    if (mHidlFactory != 0 && strcmp(AUDIO_HARDWARE_MODULE_ID_A2DP, name) != 0) {
+    if (mHidlFactory != 0 && strcmp(AUDIO_HARDWARE_MODULE_ID_A2DP, name) != 0 &&
+        strcmp(AUDIO_HARDWARE_MODULE_ID_HEARING_AID, name) != 0) {
         return mHidlFactory->openDevice(name, device);
     }
     return mLocalFactory->openDevice(name, device);
diff --git a/media/libmedia/IMediaMetadataRetriever.cpp b/media/libmedia/IMediaMetadataRetriever.cpp
index f725c97..214117b 100644
--- a/media/libmedia/IMediaMetadataRetriever.cpp
+++ b/media/libmedia/IMediaMetadataRetriever.cpp
@@ -166,15 +166,16 @@
         return interface_cast<IMemory>(reply.readStrongBinder());
     }
 
-    sp<IMemory> getImageAtIndex(int index, int colorFormat, bool metaOnly)
+    sp<IMemory> getImageAtIndex(int index, int colorFormat, bool metaOnly, bool thumbnail)
     {
-        ALOGV("getImageAtIndex: index %d, colorFormat(%d) metaOnly(%d)",
-                index, colorFormat, metaOnly);
+        ALOGV("getImageAtIndex: index %d, colorFormat(%d) metaOnly(%d) thumbnail(%d)",
+                index, colorFormat, metaOnly, thumbnail);
         Parcel data, reply;
         data.writeInterfaceToken(IMediaMetadataRetriever::getInterfaceDescriptor());
         data.writeInt32(index);
         data.writeInt32(colorFormat);
         data.writeInt32(metaOnly);
+        data.writeInt32(thumbnail);
 #ifndef DISABLE_GROUP_SCHEDULE_HACK
         sendSchedPolicy(data);
 #endif
@@ -356,12 +357,13 @@
             int index = data.readInt32();
             int colorFormat = data.readInt32();
             bool metaOnly = (data.readInt32() != 0);
-            ALOGV("getImageAtIndex: index(%d), colorFormat(%d), metaOnly(%d)",
-                    index, colorFormat, metaOnly);
+            bool thumbnail = (data.readInt32() != 0);
+            ALOGV("getImageAtIndex: index(%d), colorFormat(%d), metaOnly(%d), thumbnail(%d)",
+                    index, colorFormat, metaOnly, thumbnail);
 #ifndef DISABLE_GROUP_SCHEDULE_HACK
             setSchedPolicy(data);
 #endif
-            sp<IMemory> bitmap = getImageAtIndex(index, colorFormat, metaOnly);
+            sp<IMemory> bitmap = getImageAtIndex(index, colorFormat, metaOnly, thumbnail);
             if (bitmap != 0) {  // Don't send NULL across the binder interface
                 reply->writeInt32(NO_ERROR);
                 reply->writeStrongBinder(IInterface::asBinder(bitmap));
diff --git a/media/libmedia/include/media/IMediaMetadataRetriever.h b/media/libmedia/include/media/IMediaMetadataRetriever.h
index 5491535..1a04552 100644
--- a/media/libmedia/include/media/IMediaMetadataRetriever.h
+++ b/media/libmedia/include/media/IMediaMetadataRetriever.h
@@ -45,7 +45,7 @@
     virtual sp<IMemory>     getFrameAtTime(
             int64_t timeUs, int option, int colorFormat, bool metaOnly) = 0;
     virtual sp<IMemory>     getImageAtIndex(
-            int index, int colorFormat, bool metaOnly) = 0;
+            int index, int colorFormat, bool metaOnly, bool thumbnail) = 0;
     virtual status_t        getFrameAtIndex(
             std::vector<sp<IMemory> > *frames,
             int frameIndex, int numFrames, int colorFormat, bool metaOnly) = 0;
diff --git a/media/libmedia/include/media/MediaMetadataRetrieverInterface.h b/media/libmedia/include/media/MediaMetadataRetrieverInterface.h
index 116b548..c45a964 100644
--- a/media/libmedia/include/media/MediaMetadataRetrieverInterface.h
+++ b/media/libmedia/include/media/MediaMetadataRetrieverInterface.h
@@ -46,7 +46,7 @@
     virtual VideoFrame* getFrameAtTime(
             int64_t timeUs, int option, int colorFormat, bool metaOnly) = 0;
     virtual VideoFrame* getImageAtIndex(
-            int index, int colorFormat, bool metaOnly) = 0;
+            int index, int colorFormat, bool metaOnly, bool thumbnail) = 0;
     virtual status_t getFrameAtIndex(
             std::vector<VideoFrame*>* frames,
             int frameIndex, int numFrames, int colorFormat, bool metaOnly) = 0;
@@ -65,7 +65,7 @@
             int64_t /*timeUs*/, int /*option*/, int /*colorFormat*/, bool /*metaOnly*/)
     { return NULL; }
     virtual VideoFrame* getImageAtIndex(
-            int /*index*/, int /*colorFormat*/, bool /*metaOnly*/)
+            int /*index*/, int /*colorFormat*/, bool /*metaOnly*/, bool /*thumbnail*/)
     { return NULL; }
     virtual status_t getFrameAtIndex(
             std::vector<VideoFrame*>* /*frames*/,
diff --git a/media/libmedia/include/media/mediametadataretriever.h b/media/libmedia/include/media/mediametadataretriever.h
index b41da80..4cdeeb7 100644
--- a/media/libmedia/include/media/mediametadataretriever.h
+++ b/media/libmedia/include/media/mediametadataretriever.h
@@ -90,7 +90,7 @@
     sp<IMemory> getFrameAtTime(int64_t timeUs, int option,
             int colorFormat = HAL_PIXEL_FORMAT_RGB_565, bool metaOnly = false);
     sp<IMemory> getImageAtIndex(int index,
-            int colorFormat = HAL_PIXEL_FORMAT_RGB_565, bool metaOnly = false);
+            int colorFormat = HAL_PIXEL_FORMAT_RGB_565, bool metaOnly = false, bool thumbnail = false);
     status_t getFrameAtIndex(
             std::vector<sp<IMemory> > *frames, int frameIndex, int numFrames = 1,
             int colorFormat = HAL_PIXEL_FORMAT_RGB_565, bool metaOnly = false);
diff --git a/media/libmedia/mediametadataretriever.cpp b/media/libmedia/mediametadataretriever.cpp
index 6a4204b..c10a907 100644
--- a/media/libmedia/mediametadataretriever.cpp
+++ b/media/libmedia/mediametadataretriever.cpp
@@ -155,15 +155,15 @@
 }
 
 sp<IMemory> MediaMetadataRetriever::getImageAtIndex(
-        int index, int colorFormat, bool metaOnly) {
-    ALOGV("getImageAtIndex: index(%d) colorFormat(%d) metaOnly(%d)",
-            index, colorFormat, metaOnly);
+        int index, int colorFormat, bool metaOnly, bool thumbnail) {
+    ALOGV("getImageAtIndex: index(%d) colorFormat(%d) metaOnly(%d) thumbnail(%d)",
+            index, colorFormat, metaOnly, thumbnail);
     Mutex::Autolock _l(mLock);
     if (mRetriever == 0) {
         ALOGE("retriever is not initialized");
         return NULL;
     }
-    return mRetriever->getImageAtIndex(index, colorFormat, metaOnly);
+    return mRetriever->getImageAtIndex(index, colorFormat, metaOnly, thumbnail);
 }
 
 status_t MediaMetadataRetriever::getFrameAtIndex(
diff --git a/media/libmediaplayerservice/MetadataRetrieverClient.cpp b/media/libmediaplayerservice/MetadataRetrieverClient.cpp
index 16ed530..3b3ac29 100644
--- a/media/libmediaplayerservice/MetadataRetrieverClient.cpp
+++ b/media/libmediaplayerservice/MetadataRetrieverClient.cpp
@@ -234,9 +234,9 @@
 }
 
 sp<IMemory> MetadataRetrieverClient::getImageAtIndex(
-        int index, int colorFormat, bool metaOnly) {
-    ALOGV("getFrameAtTime: index(%d) colorFormat(%d), metaOnly(%d)",
-            index, colorFormat, metaOnly);
+        int index, int colorFormat, bool metaOnly, bool thumbnail) {
+    ALOGV("getFrameAtTime: index(%d) colorFormat(%d), metaOnly(%d) thumbnail(%d)",
+            index, colorFormat, metaOnly, thumbnail);
     Mutex::Autolock lock(mLock);
     Mutex::Autolock glock(sLock);
     mThumbnail.clear();
@@ -244,7 +244,7 @@
         ALOGE("retriever is not initialized");
         return NULL;
     }
-    VideoFrame *frame = mRetriever->getImageAtIndex(index, colorFormat, metaOnly);
+    VideoFrame *frame = mRetriever->getImageAtIndex(index, colorFormat, metaOnly, thumbnail);
     if (frame == NULL) {
         ALOGE("failed to extract image");
         return NULL;
diff --git a/media/libmediaplayerservice/MetadataRetrieverClient.h b/media/libmediaplayerservice/MetadataRetrieverClient.h
index f71891a..e774c8f 100644
--- a/media/libmediaplayerservice/MetadataRetrieverClient.h
+++ b/media/libmediaplayerservice/MetadataRetrieverClient.h
@@ -53,7 +53,7 @@
     virtual sp<IMemory>             getFrameAtTime(
             int64_t timeUs, int option, int colorFormat, bool metaOnly);
     virtual sp<IMemory>             getImageAtIndex(
-            int index, int colorFormat, bool metaOnly);
+            int index, int colorFormat, bool metaOnly, bool thumbnail);
     virtual status_t getFrameAtIndex(
                 std::vector<sp<IMemory> > *frames,
                 int frameIndex, int numFrames, int colorFormat, bool metaOnly);
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
index 14ffb1d..0a1bdfe 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
@@ -1289,7 +1289,8 @@
                 ALOGV("Tear down audio with reason %d.", reason);
                 if (reason == Renderer::kDueToTimeout && !(mPaused && mOffloadAudio)) {
                     // TimeoutWhenPaused is only for offload mode.
-                    ALOGW("Receive a stale message for teardown.");
+                    ALOGW("Received a stale message for teardown, mPaused(%d), mOffloadAudio(%d)",
+                          mPaused, mOffloadAudio);
                     break;
                 }
                 int64_t positionUs;
@@ -1789,6 +1790,8 @@
 
 void NuPlayer::restartAudio(
         int64_t currentPositionUs, bool forceNonOffload, bool needsToCreateAudioDecoder) {
+    ALOGD("restartAudio timeUs(%lld), dontOffload(%d), createDecoder(%d)",
+          (long long)currentPositionUs, forceNonOffload, needsToCreateAudioDecoder);
     if (mAudioDecoder != NULL) {
         mAudioDecoder->pause();
         mAudioDecoder.clear();
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
index 63c887b..3e5bdd6 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
@@ -669,6 +669,11 @@
         notifyListener_l(MEDIA_STOPPED);
     }
 
+    if (property_get_bool("persist.debug.sf.stats", false)) {
+        Vector<String16> args;
+        dump(-1, args);
+    }
+
     mState = STATE_RESET_IN_PROGRESS;
     mPlayer->resetAsync();
 
diff --git a/media/libstagefright/FrameDecoder.cpp b/media/libstagefright/FrameDecoder.cpp
index 3d0aad1..a00d13a 100644
--- a/media/libstagefright/FrameDecoder.cpp
+++ b/media/libstagefright/FrameDecoder.cpp
@@ -42,30 +42,30 @@
 static const int64_t kBufferTimeOutUs = 30000ll; // 30 msec
 static const size_t kRetryCount = 20; // must be >0
 
-VideoFrame *FrameDecoder::allocVideoFrame(
-        int32_t width, int32_t height, bool metaOnly) {
+//static
+VideoFrame *allocVideoFrame(const sp<MetaData> &trackMeta,
+        int32_t width, int32_t height, int32_t dstBpp, bool metaOnly = false) {
     int32_t rotationAngle;
-    if (!mTrackMeta->findInt32(kKeyRotation, &rotationAngle)) {
+    if (!trackMeta->findInt32(kKeyRotation, &rotationAngle)) {
         rotationAngle = 0;  // By default, no rotation
     }
-
     uint32_t type;
     const void *iccData;
     size_t iccSize;
-    if (!mTrackMeta->findData(kKeyIccProfile, &type, &iccData, &iccSize)){
+    if (!trackMeta->findData(kKeyIccProfile, &type, &iccData, &iccSize)){
         iccData = NULL;
         iccSize = 0;
     }
 
     int32_t sarWidth, sarHeight;
     int32_t displayWidth, displayHeight;
-    if (mTrackMeta->findInt32(kKeySARWidth, &sarWidth)
-            && mTrackMeta->findInt32(kKeySARHeight, &sarHeight)
+    if (trackMeta->findInt32(kKeySARWidth, &sarWidth)
+            && trackMeta->findInt32(kKeySARHeight, &sarHeight)
             && sarHeight != 0) {
         displayWidth = (width * sarWidth) / sarHeight;
         displayHeight = height;
-    } else if (mTrackMeta->findInt32(kKeyDisplayWidth, &displayWidth)
-                && mTrackMeta->findInt32(kKeyDisplayHeight, &displayHeight)
+    } else if (trackMeta->findInt32(kKeyDisplayWidth, &displayWidth)
+                && trackMeta->findInt32(kKeyDisplayHeight, &displayHeight)
                 && displayWidth > 0 && displayHeight > 0
                 && width > 0 && height > 0) {
         ALOGV("found display size %dx%d", displayWidth, displayHeight);
@@ -75,27 +75,66 @@
     }
 
     return new VideoFrame(width, height, displayWidth, displayHeight,
-            rotationAngle, mDstBpp, !metaOnly, iccData, iccSize);
+            rotationAngle, dstBpp, !metaOnly, iccData, iccSize);
 }
 
-bool FrameDecoder::setDstColorFormat(android_pixel_format_t colorFormat) {
+//static
+bool findThumbnailInfo(
+        const sp<MetaData> &trackMeta, int32_t *width, int32_t *height,
+        uint32_t *type = NULL, const void **data = NULL, size_t *size = NULL) {
+    uint32_t dummyType;
+    const void *dummyData;
+    size_t dummySize;
+    return trackMeta->findInt32(kKeyThumbnailWidth, width)
+        && trackMeta->findInt32(kKeyThumbnailHeight, height)
+        && trackMeta->findData(kKeyThumbnailHVCC,
+                type ?: &dummyType, data ?: &dummyData, size ?: &dummySize);
+}
+
+//static
+VideoFrame* FrameDecoder::getMetadataOnly(
+        const sp<MetaData> &trackMeta, int colorFormat, bool thumbnail) {
+    OMX_COLOR_FORMATTYPE dstFormat;
+    int32_t dstBpp;
+    if (!getDstColorFormat(
+            (android_pixel_format_t)colorFormat, &dstFormat, &dstBpp)) {
+        return NULL;
+    }
+
+    int32_t width, height;
+    if (thumbnail) {
+        if (!findThumbnailInfo(trackMeta, &width, &height)) {
+            return NULL;
+        }
+    } else {
+        CHECK(trackMeta->findInt32(kKeyWidth, &width));
+        CHECK(trackMeta->findInt32(kKeyHeight, &height));
+    }
+    return allocVideoFrame(trackMeta, width, height, dstBpp, true /*metaOnly*/);
+}
+
+//static
+bool FrameDecoder::getDstColorFormat(
+        android_pixel_format_t colorFormat,
+        OMX_COLOR_FORMATTYPE *dstFormat,
+        int32_t *dstBpp) {
     switch (colorFormat) {
         case HAL_PIXEL_FORMAT_RGB_565:
         {
-            mDstFormat = OMX_COLOR_Format16bitRGB565;
-            mDstBpp = 2;
+            *dstFormat = OMX_COLOR_Format16bitRGB565;
+            *dstBpp = 2;
             return true;
         }
         case HAL_PIXEL_FORMAT_RGBA_8888:
         {
-            mDstFormat = OMX_COLOR_Format32BitRGBA8888;
-            mDstBpp = 4;
+            *dstFormat = OMX_COLOR_Format32BitRGBA8888;
+            *dstBpp = 4;
             return true;
         }
         case HAL_PIXEL_FORMAT_BGRA_8888:
         {
-            mDstFormat = OMX_COLOR_Format32bitBGRA8888;
-            mDstBpp = 4;
+            *dstFormat = OMX_COLOR_Format32bitBGRA8888;
+            *dstBpp = 4;
             return true;
         }
         default:
@@ -108,18 +147,12 @@
 }
 
 VideoFrame* FrameDecoder::extractFrame(
-        int64_t frameTimeUs, int option, int colorFormat, bool metaOnly) {
-    if (!setDstColorFormat((android_pixel_format_t)colorFormat)) {
+        int64_t frameTimeUs, int option, int colorFormat) {
+    if (!getDstColorFormat(
+            (android_pixel_format_t)colorFormat, &mDstFormat, &mDstBpp)) {
         return NULL;
     }
 
-    if (metaOnly) {
-        int32_t width, height;
-        CHECK(trackMeta()->findInt32(kKeyWidth, &width));
-        CHECK(trackMeta()->findInt32(kKeyHeight, &height));
-        return allocVideoFrame(width, height, true);
-    }
-
     status_t err = extractInternal(frameTimeUs, 1, option);
     if (err != OK) {
         return NULL;
@@ -131,7 +164,8 @@
 status_t FrameDecoder::extractFrames(
         int64_t frameTimeUs, size_t numFrames, int option, int colorFormat,
         std::vector<VideoFrame*>* frames) {
-    if (!setDstColorFormat((android_pixel_format_t)colorFormat)) {
+    if (!getDstColorFormat(
+            (android_pixel_format_t)colorFormat, &mDstFormat, &mDstBpp)) {
         return ERROR_UNSUPPORTED;
     }
 
@@ -435,9 +469,10 @@
     }
 
     VideoFrame *frame = allocVideoFrame(
+            trackMeta(),
             (crop_right - crop_left + 1),
             (crop_bottom - crop_top + 1),
-            false /*metaOnly*/);
+            dstBpp());
     addFrame(frame);
 
     int32_t srcFormat;
@@ -466,28 +501,28 @@
         int64_t frameTimeUs, size_t /*numFrames*/,
         int /*seekMode*/, MediaSource::ReadOptions *options) {
     sp<MetaData> overrideMeta;
+    mThumbnail = false;
     if (frameTimeUs < 0) {
         uint32_t type;
         const void *data;
         size_t size;
-        int64_t thumbNailTime = 0;
-        int32_t thumbnailWidth, thumbnailHeight;
+        int32_t thumbWidth, thumbHeight;
 
         // if we have a stand-alone thumbnail, set up the override meta,
         // and set seekTo time to -1.
-        if (trackMeta()->findInt32(kKeyThumbnailWidth, &thumbnailWidth)
-         && trackMeta()->findInt32(kKeyThumbnailHeight, &thumbnailHeight)
-         && trackMeta()->findData(kKeyThumbnailHVCC, &type, &data, &size)){
-            overrideMeta = new MetaData(*(trackMeta()));
-            overrideMeta->remove(kKeyDisplayWidth);
-            overrideMeta->remove(kKeyDisplayHeight);
-            overrideMeta->setInt32(kKeyWidth, thumbnailWidth);
-            overrideMeta->setInt32(kKeyHeight, thumbnailHeight);
-            overrideMeta->setData(kKeyHVCC, type, data, size);
-            thumbNailTime = -1ll;
-            ALOGV("thumbnail: %dx%d", thumbnailWidth, thumbnailHeight);
+        if (!findThumbnailInfo(trackMeta(),
+                &thumbWidth, &thumbHeight, &type, &data, &size)) {
+            ALOGE("Thumbnail not available");
+            return NULL;
         }
-        options->setSeekTo(thumbNailTime);
+        overrideMeta = new MetaData(*(trackMeta()));
+        overrideMeta->remove(kKeyDisplayWidth);
+        overrideMeta->remove(kKeyDisplayHeight);
+        overrideMeta->setInt32(kKeyWidth, thumbWidth);
+        overrideMeta->setInt32(kKeyHeight, thumbHeight);
+        overrideMeta->setData(kKeyHVCC, type, data, size);
+        options->setSeekTo(-1);
+        mThumbnail = true;
     } else {
         options->setSeekTo(frameTimeUs);
     }
@@ -552,11 +587,16 @@
     CHECK(outputFormat->findInt32("height", &height));
 
     int32_t imageWidth, imageHeight;
-    CHECK(trackMeta()->findInt32(kKeyWidth, &imageWidth));
-    CHECK(trackMeta()->findInt32(kKeyHeight, &imageHeight));
+    if (mThumbnail) {
+        CHECK(trackMeta()->findInt32(kKeyThumbnailWidth, &imageWidth));
+        CHECK(trackMeta()->findInt32(kKeyThumbnailHeight, &imageHeight));
+    } else {
+        CHECK(trackMeta()->findInt32(kKeyWidth, &imageWidth));
+        CHECK(trackMeta()->findInt32(kKeyHeight, &imageHeight));
+    }
 
     if (mFrame == NULL) {
-        mFrame = allocVideoFrame(imageWidth, imageHeight, false /*metaOnly*/);
+        mFrame = allocVideoFrame(trackMeta(), imageWidth, imageHeight, dstBpp());
 
         addFrame(mFrame);
     }
diff --git a/media/libstagefright/StagefrightMetadataRetriever.cpp b/media/libstagefright/StagefrightMetadataRetriever.cpp
index 5ae5644..6ad6004 100644
--- a/media/libstagefright/StagefrightMetadataRetriever.cpp
+++ b/media/libstagefright/StagefrightMetadataRetriever.cpp
@@ -125,10 +125,10 @@
 }
 
 VideoFrame* StagefrightMetadataRetriever::getImageAtIndex(
-        int index, int colorFormat, bool metaOnly) {
+        int index, int colorFormat, bool metaOnly, bool thumbnail) {
 
-    ALOGV("getImageAtIndex: index: %d colorFormat: %d, metaOnly: %d",
-            index, colorFormat, metaOnly);
+    ALOGV("getImageAtIndex: index(%d) colorFormat(%d) metaOnly(%d) thumbnail(%d)",
+            index, colorFormat, metaOnly, thumbnail);
 
     if (mExtractor.get() == NULL) {
         ALOGE("no extractor.");
@@ -163,6 +163,10 @@
 
     sp<MetaData> trackMeta = mExtractor->getTrackMetaData(i);
 
+    if (metaOnly) {
+        return FrameDecoder::getMetadataOnly(trackMeta, colorFormat, thumbnail);
+    }
+
     sp<IMediaSource> source = mExtractor->getTrack(i);
 
     if (source.get() == NULL) {
@@ -190,7 +194,7 @@
         const AString &componentName = matchingCodecs[i];
         ImageDecoder decoder(componentName, trackMeta, source);
         VideoFrame* frame = decoder.extractFrame(
-                0 /*frameTimeUs*/, 0 /*seekMode*/, colorFormat, metaOnly);
+                thumbnail ? -1 : 0 /*frameTimeUs*/, 0 /*seekMode*/, colorFormat);
 
         if (frame != NULL) {
             return frame;
@@ -265,6 +269,16 @@
     sp<MetaData> trackMeta = mExtractor->getTrackMetaData(
             i, MediaExtractor::kIncludeExtensiveMetaData);
 
+    if (metaOnly) {
+        if (outFrame != NULL) {
+            *outFrame = FrameDecoder::getMetadataOnly(trackMeta, colorFormat);
+            if (*outFrame != NULL) {
+                return OK;
+            }
+        }
+        return UNKNOWN_ERROR;
+    }
+
     sp<IMediaSource> source = mExtractor->getTrack(i);
 
     if (source.get() == NULL) {
@@ -294,8 +308,7 @@
         const AString &componentName = matchingCodecs[i];
         VideoFrameDecoder decoder(componentName, trackMeta, source);
         if (outFrame != NULL) {
-            *outFrame = decoder.extractFrame(
-                    timeUs, option, colorFormat, metaOnly);
+            *outFrame = decoder.extractFrame(timeUs, option, colorFormat);
             if (*outFrame != NULL) {
                 return OK;
             }
diff --git a/media/libstagefright/VideoFrameScheduler.cpp b/media/libstagefright/VideoFrameScheduler.cpp
index 03226c7..6819bba 100644
--- a/media/libstagefright/VideoFrameScheduler.cpp
+++ b/media/libstagefright/VideoFrameScheduler.cpp
@@ -129,6 +129,11 @@
         numSamplesToUse = mNumSamples;
     }
 
+    if ((period >> kPrecision) == 0 ) {
+        ALOGW("Period is 0, or after including precision is 0 - would cause div0, returning");
+        return false;
+    }
+
     int64_t sumX = 0;
     int64_t sumXX = 0;
     int64_t sumXY = 0;
diff --git a/media/libstagefright/include/FrameDecoder.h b/media/libstagefright/include/FrameDecoder.h
index dfbe2cd..b67e928 100644
--- a/media/libstagefright/include/FrameDecoder.h
+++ b/media/libstagefright/include/FrameDecoder.h
@@ -44,11 +44,7 @@
                 mDstFormat(OMX_COLOR_Format16bitRGB565),
                 mDstBpp(2) {}
 
-    VideoFrame* extractFrame(
-            int64_t frameTimeUs,
-            int option,
-            int colorFormat,
-            bool metaOnly);
+    VideoFrame* extractFrame(int64_t frameTimeUs, int option, int colorFormat);
 
     status_t extractFrames(
             int64_t frameTimeUs,
@@ -57,6 +53,9 @@
             int colorFormat,
             std::vector<VideoFrame*>* frames);
 
+    static VideoFrame* getMetadataOnly(
+            const sp<MetaData> &trackMeta, int colorFormat, bool thumbnail = false);
+
 protected:
     virtual ~FrameDecoder() {}
 
@@ -78,8 +77,6 @@
             int64_t timeUs,
             bool *done) = 0;
 
-    VideoFrame *allocVideoFrame(int32_t width, int32_t height, bool metaOnly);
-
     sp<MetaData> trackMeta()     const      { return mTrackMeta; }
     OMX_COLOR_FORMATTYPE dstFormat() const  { return mDstFormat; }
     int32_t dstBpp()             const      { return mDstBpp; }
@@ -96,7 +93,11 @@
     int32_t mDstBpp;
     std::vector<std::unique_ptr<VideoFrame> > mFrames;
 
-    bool setDstColorFormat(android_pixel_format_t colorFormat);
+    static bool getDstColorFormat(
+            android_pixel_format_t colorFormat,
+            OMX_COLOR_FORMATTYPE *dstFormat,
+            int32_t *dstBpp);
+
     status_t extractInternal(int64_t frameTimeUs, size_t numFrames, int option);
 
     DISALLOW_EVIL_CONSTRUCTORS(FrameDecoder);
@@ -147,7 +148,8 @@
             const sp<MetaData> &trackMeta,
             const sp<IMediaSource> &source) :
                 FrameDecoder(componentName, trackMeta, source),
-                mFrame(NULL), mGridRows(1), mGridCols(1), mTilesDecoded(0) {}
+                mFrame(NULL), mGridRows(1), mGridCols(1),
+                mTilesDecoded(0), mThumbnail(false) {}
 
 protected:
     virtual sp<AMessage> onGetFormatAndSeekOptions(
@@ -173,6 +175,7 @@
     int32_t mGridRows;
     int32_t mGridCols;
     int32_t mTilesDecoded;
+    bool mThumbnail;
 };
 
 }  // namespace android
diff --git a/media/libstagefright/include/StagefrightMetadataRetriever.h b/media/libstagefright/include/StagefrightMetadataRetriever.h
index 58442fe..8443fbe 100644
--- a/media/libstagefright/include/StagefrightMetadataRetriever.h
+++ b/media/libstagefright/include/StagefrightMetadataRetriever.h
@@ -43,7 +43,7 @@
     virtual VideoFrame* getFrameAtTime(
             int64_t timeUs, int option, int colorFormat, bool metaOnly);
     virtual VideoFrame* getImageAtIndex(
-            int index, int colorFormat, bool metaOnly);
+            int index, int colorFormat, bool metaOnly, bool thumbnail);
     virtual status_t getFrameAtIndex(
             std::vector<VideoFrame*>* frames,
             int frameIndex, int numFrames, int colorFormat, bool metaOnly);
diff --git a/media/ndk/libmediandk.map.txt b/media/ndk/libmediandk.map.txt
index 28524b0..fb56694 100644
--- a/media/ndk/libmediandk.map.txt
+++ b/media/ndk/libmediandk.map.txt
@@ -108,6 +108,7 @@
     AMediaCodec_queueInputBuffer;
     AMediaCodec_queueSecureInputBuffer;
     AMediaCodec_releaseCrypto; # introduced=28
+    AMediaCodec_releaseName; # introduced=28
     AMediaCodec_releaseOutputBuffer;
     AMediaCodec_releaseOutputBufferAtTime;
     AMediaCodec_setAsyncNotifyCallback; # introduced=28
diff --git a/packages/MediaComponents/Android.mk b/packages/MediaComponents/Android.mk
index b0d8e7d..def9dc5 100644
--- a/packages/MediaComponents/Android.mk
+++ b/packages/MediaComponents/Android.mk
@@ -14,59 +14,59 @@
 # limitations under the License.
 #
 
-LOCAL_PATH := $(call my-dir)
-
-ifneq ($(TARGET_BUILD_PDK),true)
-# Build MediaComponents only if this is not a PDK build.  MediaComponents won't
-# build in PDK builds because frameworks/base/core/java is not available but
-# IMediaSession2.aidl and IMediaController2.aidl are using classes from
-# frameworks/base/core/java.
-
-include $(CLEAR_VARS)
-
-LOCAL_PACKAGE_NAME := MediaComponents
-LOCAL_MODULE_OWNER := google
-
-# TODO: create a separate key for this package.
-LOCAL_CERTIFICATE := platform
-
-# TODO: Use System SDK once public APIs are approved
-# LOCAL_SDK_VERSION := system_current
-LOCAL_PRIVATE_PLATFORM_APIS := true
-
-LOCAL_SRC_FILES := \
-    $(call all-java-files-under, src) \
-    $(call all-Iaidl-files-under, src)
-
-LOCAL_PROGUARD_FLAG_FILES := proguard.cfg
-
-LOCAL_MULTILIB := first
-
-LOCAL_JAVA_LIBRARIES += android-support-annotations
-
-# To embed native libraries in package, uncomment the lines below.
-#LOCAL_MODULE_TAGS := samples
-#LOCAL_JNI_SHARED_LIBRARIES := \
-#    libaacextractor \
-#    libamrextractor \
-#    libflacextractor \
-#    libmidiextractor \
-#    libmkvextractor \
-#    libmp3extractor \
-#    libmp4extractor \
-#    libmpeg2extractor \
-#    liboggextractor \
-#    libwavextractor \
-
-# TODO: Remove dependency with other support libraries.
-LOCAL_STATIC_ANDROID_LIBRARIES += \
-    android-support-v4 \
-    android-support-v7-appcompat \
-    android-support-v7-palette
-LOCAL_USE_AAPT2 := true
-
-include $(BUILD_PACKAGE)
-
-endif  # ifneq ($(TARGET_BUILD_PDK),true)
-
-include $(call all-makefiles-under,$(LOCAL_PATH))
+# This package is excluded from build for now since APIs using this apk became hidden.
+#
+#LOCAL_PATH := $(call my-dir)
+#ifneq ($(TARGET_BUILD_PDK),true) # Build MediaComponents only if this is not a PDK build.  MediaComponents won't
+## build in PDK builds because frameworks/base/core/java is not available but
+## IMediaSession2.aidl and IMediaController2.aidl are using classes from
+## frameworks/base/core/java.
+#
+#include $(CLEAR_VARS)
+#
+#LOCAL_PACKAGE_NAME := MediaComponents
+#LOCAL_MODULE_OWNER := google
+#
+## TODO: create a separate key for this package.
+#LOCAL_CERTIFICATE := platform
+#
+## TODO: Use System SDK once public APIs are approved
+## LOCAL_SDK_VERSION := system_current
+#LOCAL_PRIVATE_PLATFORM_APIS := true
+#
+#LOCAL_SRC_FILES := \
+#    $(call all-java-files-under, src) \
+#    $(call all-Iaidl-files-under, src)
+#
+#LOCAL_PROGUARD_FLAG_FILES := proguard.cfg
+#
+#LOCAL_MULTILIB := first
+#
+#LOCAL_JAVA_LIBRARIES += android-support-annotations
+#
+## To embed native libraries in package, uncomment the lines below.
+##LOCAL_MODULE_TAGS := samples
+##LOCAL_JNI_SHARED_LIBRARIES := \
+##    libaacextractor \
+##    libamrextractor \
+##    libflacextractor \
+##    libmidiextractor \
+##    libmkvextractor \
+##    libmp3extractor \
+##    libmp4extractor \
+##    libmpeg2extractor \
+##    liboggextractor \
+##    libwavextractor \
+#
+## TODO: Remove dependency with other support libraries.
+#LOCAL_STATIC_ANDROID_LIBRARIES += \
+#    android-support-v4 \
+#    android-support-v7-appcompat \
+#    android-support-v7-palette
+#LOCAL_USE_AAPT2 := true
+#
+#include $(BUILD_PACKAGE)
+#
+#endif  # ifneq ($(TARGET_BUILD_PDK),true)
+#
+#include $(call all-makefiles-under,$(LOCAL_PATH))
diff --git a/packages/MediaComponents/AndroidManifest.xml b/packages/MediaComponents/AndroidManifest.xml
index 061ae44..50fdca1 100644
--- a/packages/MediaComponents/AndroidManifest.xml
+++ b/packages/MediaComponents/AndroidManifest.xml
@@ -8,6 +8,7 @@
         android:label="Media Components Update"
         android:multiArch="true"
         android:allowBackup="false"
+        android:hasCode="false"
         android:extractNativeLibs="false">
     </application>
 
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 43a8b50..ab65601 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -4395,7 +4395,7 @@
                 float volume = masterVolume
                         * mStreamTypes[track->streamType()].volume
                         * vh;
-                track->mCachedVolume = masterVolume;
+                track->mCachedVolume = volume;
                 gain_minifloat_packed_t vlr = proxy->getVolumeLR();
                 float vlf = volume * float_from_gain(gain_minifloat_unpack_left(vlr));
                 float vrf = volume * float_from_gain(gain_minifloat_unpack_right(vlr));
@@ -4807,6 +4807,18 @@
         track->reset();
     }
 
+    // Track destruction may occur outside of threadLoop once it is removed from active tracks.
+    // Ensure the AudioMixer doesn't have a raw "buffer provider" pointer to the track if
+    // it ceases to be active, to allow safe removal from the AudioMixer at the start
+    // of prepareTracks_l(); this releases any outstanding buffer back to the track.
+    // See also the implementation of destroyTrack_l().
+    for (const auto &track : *tracksToRemove) {
+        const int name = track->name();
+        if (mAudioMixer->exists(name)) { // Normal tracks here, fast tracks in FastMixer.
+            mAudioMixer->setBufferProvider(name, nullptr /* bufferProvider */);
+        }
+    }
+
     // remove all the tracks that need to be...
     removeTracks_l(*tracksToRemove);
 
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index ee9ce84..3fe41d8 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -1782,14 +1782,14 @@
 
 /*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result)
 {
-    result.append("Active Client Session S  Flags   Format Chn mask  SRate   Server FrmCnt\n");
+    result.append("Active Client Session S  Flags   Format Chn mask  SRate   Server FrmCnt Sil\n");
 }
 
 void AudioFlinger::RecordThread::RecordTrack::appendDump(String8& result, bool active)
 {
     result.appendFormat("%c%5s %6u %7u %2s 0x%03X "
             "%08X %08X %6u "
-            "%08X %6zu\n",
+            "%08X %6zu %3c\n",
             isFastTrack() ? 'F' : ' ',
             active ? "yes" : "no",
             (mClient == 0) ? getpid_cached : mClient->pid(),
@@ -1802,7 +1802,8 @@
             mSampleRate,
 
             mCblk->mServer,
-            mFrameCount
+            mFrameCount,
+            isSilenced() ? 's' : 'n'
             );
 }
 
diff --git a/services/audiopolicy/common/managerdefinitions/src/VolumeCurve.cpp b/services/audiopolicy/common/managerdefinitions/src/VolumeCurve.cpp
index ec861c1..ac3f1bc 100644
--- a/services/audiopolicy/common/managerdefinitions/src/VolumeCurve.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/VolumeCurve.cpp
@@ -29,6 +29,13 @@
     size_t nbCurvePoints = mCurvePoints.size();
     // the volume index in the UI is relative to the min and max volume indices for this stream
     int nbSteps = 1 + mCurvePoints[nbCurvePoints - 1].mIndex - mCurvePoints[0].mIndex;
+    if (indexInUi < volIndexMin) {
+        ALOGV("VOLUME remapping index from %d to min index %d", indexInUi, volIndexMin);
+        indexInUi = volIndexMin;
+    } else if (indexInUi > volIndexMax) {
+        ALOGV("VOLUME remapping index from %d to max index %d", indexInUi, volIndexMax);
+        indexInUi = volIndexMax;
+    }
     int volIdx = (nbSteps * (indexInUi - volIndexMin)) / (volIndexMax - volIndexMin);
 
     // Where would this volume index been inserted in the curve point
diff --git a/services/audiopolicy/enginedefault/src/Engine.cpp b/services/audiopolicy/enginedefault/src/Engine.cpp
index 08bcf4d..83aec3b 100644
--- a/services/audiopolicy/enginedefault/src/Engine.cpp
+++ b/services/audiopolicy/enginedefault/src/Engine.cpp
@@ -613,6 +613,23 @@
 
     uint32_t device = AUDIO_DEVICE_NONE;
 
+    // when a call is active, force device selection to match source VOICE_COMMUNICATION
+    // for most other input sources to avoid rerouting call TX audio
+    if (isInCall()) {
+        switch (inputSource) {
+        case AUDIO_SOURCE_DEFAULT:
+        case AUDIO_SOURCE_MIC:
+        case AUDIO_SOURCE_VOICE_RECOGNITION:
+        case AUDIO_SOURCE_UNPROCESSED:
+        case AUDIO_SOURCE_HOTWORD:
+        case AUDIO_SOURCE_CAMCORDER:
+            inputSource = AUDIO_SOURCE_VOICE_COMMUNICATION;
+            break;
+        default:
+            break;
+        }
+    }
+
     switch (inputSource) {
     case AUDIO_SOURCE_VOICE_UPLINK:
       if (availableDeviceTypes & AUDIO_DEVICE_IN_VOICE_CALL) {
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index 29ec961..947529e 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -4530,10 +4530,13 @@
         }
     }
 
+    // If we are not in call and no client is active on this input, this methods returns
+    // AUDIO_DEVICE_NONE, causing the patch on the input stream to be released.
     audio_source_t source = inputDesc->getHighestPrioritySource(true /*activeOnly*/);
-    if (isInCall()) {
-        device = getDeviceAndMixForInputSource(AUDIO_SOURCE_VOICE_COMMUNICATION);
-    } else if (source != AUDIO_SOURCE_DEFAULT) {
+    if (source == AUDIO_SOURCE_DEFAULT && isInCall()) {
+        source = AUDIO_SOURCE_VOICE_COMMUNICATION;
+    }
+    if (source != AUDIO_SOURCE_DEFAULT) {
         device = getDeviceAndMixForInputSource(source);
     }
 
diff --git a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
index 8f0c846..cf24c13 100644
--- a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
+++ b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
@@ -44,6 +44,7 @@
 
     ALOGV("setDeviceConnectionState()");
     Mutex::Autolock _l(mLock);
+    AutoCallerClear acc;
     return mAudioPolicyManager->setDeviceConnectionState(device, state,
                                                          device_address, device_name);
 }
@@ -55,6 +56,7 @@
     if (mAudioPolicyManager == NULL) {
         return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE;
     }
+    AutoCallerClear acc;
     return mAudioPolicyManager->getDeviceConnectionState(device,
                                                       device_address);
 }
@@ -72,6 +74,7 @@
 
     ALOGV("handleDeviceConfigChange()");
     Mutex::Autolock _l(mLock);
+    AutoCallerClear acc;
     return mAudioPolicyManager->handleDeviceConfigChange(device, device_address,
                                                          device_name);
 }
@@ -94,10 +97,10 @@
     // operation from policy manager standpoint (no other operation (e.g track start or stop)
     // can be interleaved).
     Mutex::Autolock _l(mLock);
-
     // TODO: check if it is more appropriate to do it in platform specific policy manager
     AudioSystem::setMode(state);
 
+    AutoCallerClear acc;
     mAudioPolicyManager->setPhoneState(state);
     mPhoneState = state;
     return NO_ERROR;
@@ -126,6 +129,7 @@
     }
     ALOGV("setForceUse()");
     Mutex::Autolock _l(mLock);
+    AutoCallerClear acc;
     mAudioPolicyManager->setForceUse(usage, config);
     return NO_ERROR;
 }
@@ -138,6 +142,7 @@
     if (usage < 0 || usage >= AUDIO_POLICY_FORCE_USE_CNT) {
         return AUDIO_POLICY_FORCE_NONE;
     }
+    AutoCallerClear acc;
     return mAudioPolicyManager->getForceUse(usage);
 }
 
@@ -151,6 +156,7 @@
     }
     ALOGV("getOutput()");
     Mutex::Autolock _l(mLock);
+    AutoCallerClear acc;
     return mAudioPolicyManager->getOutput(stream);
 }
 
@@ -178,6 +184,7 @@
         uid = callingUid;
     }
     audio_output_flags_t originalFlags = flags;
+    AutoCallerClear acc;
     status_t result = mAudioPolicyManager->getOutputForAttr(attr, output, session, stream, uid,
                                                  config,
                                                  &flags, selectedDeviceId, portId);
@@ -223,6 +230,7 @@
         }
     }
     Mutex::Autolock _l(mLock);
+    AutoCallerClear acc;
     return mAudioPolicyManager->startOutput(output, stream, session);
 }
 
@@ -259,6 +267,7 @@
         }
     }
     Mutex::Autolock _l(mLock);
+    AutoCallerClear acc;
     return mAudioPolicyManager->stopOutput(output, stream, session);
 }
 
@@ -279,6 +288,7 @@
 {
     ALOGV("doReleaseOutput from tid %d", gettid());
     Mutex::Autolock _l(mLock);
+    // called from internal thread: no need to clear caller identity
     mAudioPolicyManager->releaseOutput(output, stream, session);
 }
 
@@ -337,11 +347,14 @@
         AudioPolicyInterface::input_type_t inputType;
 
         Mutex::Autolock _l(mLock);
-        // the audio_in_acoustics_t parameter is ignored by get_input()
-        status = mAudioPolicyManager->getInputForAttr(attr, input, session, uid,
-                                                     config,
-                                                     flags, selectedDeviceId,
-                                                     &inputType, portId);
+        {
+            AutoCallerClear acc;
+            // the audio_in_acoustics_t parameter is ignored by get_input()
+            status = mAudioPolicyManager->getInputForAttr(attr, input, session, uid,
+                                                         config,
+                                                         flags, selectedDeviceId,
+                                                         &inputType, portId);
+        }
         audioPolicyEffects = mAudioPolicyEffects;
 
         if (status == NO_ERROR) {
@@ -372,6 +385,7 @@
 
         if (status != NO_ERROR) {
             if (status == PERMISSION_DENIED) {
+                AutoCallerClear acc;
                 mAudioPolicyManager->releaseInput(*input, session);
             }
             return status;
@@ -425,8 +439,12 @@
     AudioPolicyInterface::concurrency_type__mask_t concurrency =
             AudioPolicyInterface::API_INPUT_CONCURRENCY_NONE;
 
-    status_t status = mAudioPolicyManager->startInput(
-            client->input, client->session, *silenced, &concurrency);
+    status_t status;
+    {
+        AutoCallerClear acc;
+        status = mAudioPolicyManager->startInput(
+                    client->input, client->session, *silenced, &concurrency);
+    }
 
     if (status == NO_ERROR) {
         LOG_ALWAYS_FATAL_IF(concurrency & ~AudioPolicyInterface::API_INPUT_CONCURRENCY_ALL,
@@ -461,7 +479,7 @@
 
     // finish the recording app op
     finishRecording(client->opPackageName, client->uid);
-
+    AutoCallerClear acc;
     return mAudioPolicyManager->stopInput(client->input, client->session);
 }
 
@@ -494,6 +512,7 @@
     }
     {
         Mutex::Autolock _l(mLock);
+        AutoCallerClear acc;
         mAudioPolicyManager->releaseInput(client->input, client->session);
     }
 }
@@ -512,6 +531,7 @@
         return BAD_VALUE;
     }
     Mutex::Autolock _l(mLock);
+    AutoCallerClear acc;
     mAudioPolicyManager->initStreamVolume(stream, indexMin, indexMax);
     return NO_ERROR;
 }
@@ -530,6 +550,7 @@
         return BAD_VALUE;
     }
     Mutex::Autolock _l(mLock);
+    AutoCallerClear acc;
     return mAudioPolicyManager->setStreamVolumeIndex(stream,
                                                     index,
                                                     device);
@@ -546,6 +567,7 @@
         return BAD_VALUE;
     }
     Mutex::Autolock _l(mLock);
+    AutoCallerClear acc;
     return mAudioPolicyManager->getStreamVolumeIndex(stream,
                                                     index,
                                                     device);
@@ -559,6 +581,7 @@
     if (mAudioPolicyManager == NULL) {
         return 0;
     }
+    AutoCallerClear acc;
     return mAudioPolicyManager->getStrategyForStream(stream);
 }
 
@@ -573,6 +596,7 @@
         return AUDIO_DEVICE_NONE;
     }
     Mutex::Autolock _l(mLock);
+    AutoCallerClear acc;
     return mAudioPolicyManager->getDevicesForStream(stream);
 }
 
@@ -583,6 +607,7 @@
         return 0;
     }
     Mutex::Autolock _l(mLock);
+    AutoCallerClear acc;
     return mAudioPolicyManager->getOutputForEffect(desc);
 }
 
@@ -596,6 +621,7 @@
         return NO_INIT;
     }
     Mutex::Autolock _l(mEffectsLock);
+    AutoCallerClear acc;
     return mAudioPolicyManager->registerEffect(desc, io, strategy, session, id);
 }
 
@@ -605,6 +631,7 @@
         return NO_INIT;
     }
     Mutex::Autolock _l(mEffectsLock);
+    AutoCallerClear acc;
     return mAudioPolicyManager->unregisterEffect(id);
 }
 
@@ -614,6 +641,7 @@
         return NO_INIT;
     }
     Mutex::Autolock _l(mEffectsLock);
+    AutoCallerClear acc;
     return mAudioPolicyManager->setEffectEnabled(id, enabled);
 }
 
@@ -626,6 +654,7 @@
         return false;
     }
     Mutex::Autolock _l(mLock);
+    AutoCallerClear acc;
     return mAudioPolicyManager->isStreamActive(stream, inPastMs);
 }
 
@@ -638,6 +667,7 @@
         return false;
     }
     Mutex::Autolock _l(mLock);
+    AutoCallerClear acc;
     return mAudioPolicyManager->isStreamActiveRemotely(stream, inPastMs);
 }
 
@@ -647,6 +677,7 @@
         return false;
     }
     Mutex::Autolock _l(mLock);
+    AutoCallerClear acc;
     return mAudioPolicyManager->isSourceActive(source);
 }
 
@@ -680,6 +711,7 @@
     Mutex::Autolock _l(mLock);
     Mutex::Autolock _le(mEffectsLock); // isOffloadSupported queries for
                                       // non-offloadable effects
+    AutoCallerClear acc;
     return mAudioPolicyManager->isOffloadSupported(info);
 }
 
@@ -693,7 +725,7 @@
     if (mAudioPolicyManager == NULL) {
         return NO_INIT;
     }
-
+    AutoCallerClear acc;
     return mAudioPolicyManager->listAudioPorts(role, type, num_ports, ports, generation);
 }
 
@@ -703,7 +735,7 @@
     if (mAudioPolicyManager == NULL) {
         return NO_INIT;
     }
-
+    AutoCallerClear acc;
     return mAudioPolicyManager->getAudioPort(port);
 }
 
@@ -717,6 +749,7 @@
     if (mAudioPolicyManager == NULL) {
         return NO_INIT;
     }
+    AutoCallerClear acc;
     return mAudioPolicyManager->createAudioPatch(patch, handle,
                                                   IPCThreadState::self()->getCallingUid());
 }
@@ -730,7 +763,7 @@
     if (mAudioPolicyManager == NULL) {
         return NO_INIT;
     }
-
+    AutoCallerClear acc;
     return mAudioPolicyManager->releaseAudioPatch(handle,
                                                      IPCThreadState::self()->getCallingUid());
 }
@@ -743,7 +776,7 @@
     if (mAudioPolicyManager == NULL) {
         return NO_INIT;
     }
-
+    AutoCallerClear acc;
     return mAudioPolicyManager->listAudioPatches(num_patches, patches, generation);
 }
 
@@ -756,7 +789,7 @@
     if (mAudioPolicyManager == NULL) {
         return NO_INIT;
     }
-
+    AutoCallerClear acc;
     return mAudioPolicyManager->setAudioPortConfig(config);
 }
 
@@ -768,7 +801,7 @@
     if (mAudioPolicyManager == NULL) {
         return NO_INIT;
     }
-
+    AutoCallerClear acc;
     return mAudioPolicyManager->acquireSoundTriggerSession(session, ioHandle, device);
 }
 
@@ -778,7 +811,7 @@
     if (mAudioPolicyManager == NULL) {
         return NO_INIT;
     }
-
+    AutoCallerClear acc;
     return mAudioPolicyManager->releaseSoundTriggerSession(session);
 }
 
@@ -791,6 +824,7 @@
     if (mAudioPolicyManager == NULL) {
         return NO_INIT;
     }
+    AutoCallerClear acc;
     if (registration) {
         return mAudioPolicyManager->registerPolicyMixes(mixes);
     } else {
@@ -806,7 +840,7 @@
     if (mAudioPolicyManager == NULL) {
         return NO_INIT;
     }
-
+    AutoCallerClear acc;
     return mAudioPolicyManager->startAudioSource(source, attributes, handle,
                                                  IPCThreadState::self()->getCallingUid());
 }
@@ -817,7 +851,7 @@
     if (mAudioPolicyManager == NULL) {
         return NO_INIT;
     }
-
+    AutoCallerClear acc;
     return mAudioPolicyManager->stopAudioSource(handle);
 }
 
@@ -830,6 +864,7 @@
         return PERMISSION_DENIED;
     }
     Mutex::Autolock _l(mLock);
+    AutoCallerClear acc;
     return mAudioPolicyManager->setMasterMono(mono);
 }
 
@@ -839,6 +874,7 @@
         return NO_INIT;
     }
     Mutex::Autolock _l(mLock);
+    AutoCallerClear acc;
     return mAudioPolicyManager->getMasterMono(mono);
 }
 
@@ -850,6 +886,7 @@
         return NAN;
     }
     Mutex::Autolock _l(mLock);
+    AutoCallerClear acc;
     return mAudioPolicyManager->getStreamVolumeDB(stream, index, device);
 }
 
diff --git a/services/audiopolicy/service/AudioPolicyService.cpp b/services/audiopolicy/service/AudioPolicyService.cpp
index 082923a..13bf605 100644
--- a/services/audiopolicy/service/AudioPolicyService.cpp
+++ b/services/audiopolicy/service/AudioPolicyService.cpp
@@ -151,6 +151,7 @@
     {
         Mutex::Autolock _l(mLock);
         if (mAudioPolicyManager) {
+            // called from binder death notification: no need to clear caller identity
             mAudioPolicyManager->releaseResourcesForUid(uid);
         }
     }
@@ -335,6 +336,7 @@
     {
         Mutex::Autolock _l(mLock);
         if (mAudioPolicyManager) {
+            AutoCallerClear acc;
             mAudioPolicyManager->setRecordSilenced(uid, silenced);
         }
     }
diff --git a/services/audiopolicy/service/AudioPolicyService.h b/services/audiopolicy/service/AudioPolicyService.h
index b3bc12b..d8dd797 100644
--- a/services/audiopolicy/service/AudioPolicyService.h
+++ b/services/audiopolicy/service/AudioPolicyService.h
@@ -640,6 +640,22 @@
         bool isVirtualDevice;          // uses vitual device: updated by APM::getInputForAttr()
     };
 
+    // A class automatically clearing and restoring binder caller identity inside
+    // a code block (scoped variable)
+    // Declare one systematically before calling AudioPolicyManager methods so that they are
+    // executed with the same level of privilege as audioserver process.
+    class AutoCallerClear {
+    public:
+            AutoCallerClear() :
+                mToken(IPCThreadState::self()->clearCallingIdentity()) {}
+            ~AutoCallerClear() {
+                IPCThreadState::self()->restoreCallingIdentity(mToken);
+            }
+
+    private:
+        const   int64_t mToken;
+    };
+
     // Internal dump utilities.
     status_t dumpPermissionDenial(int fd);
 
diff --git a/services/mediaextractor/Android.mk b/services/mediaextractor/Android.mk
index 5b7571c..d505cfe 100644
--- a/services/mediaextractor/Android.mk
+++ b/services/mediaextractor/Android.mk
@@ -32,7 +32,6 @@
     libmpeg2extractor \
     liboggextractor \
     libwavextractor \
-    MediaComponents \
 
 LOCAL_SRC_FILES := main_extractorservice.cpp
 LOCAL_SHARED_LIBRARIES := libmedia libmediaextractorservice libbinder libutils \
diff --git a/services/oboeservice/AAudioClientTracker.cpp b/services/oboeservice/AAudioClientTracker.cpp
index 549a4e9..7264a9b 100644
--- a/services/oboeservice/AAudioClientTracker.cpp
+++ b/services/oboeservice/AAudioClientTracker.cpp
@@ -21,6 +21,8 @@
 
 #include <assert.h>
 #include <binder/IPCThreadState.h>
+#include <iomanip>
+#include <iostream>
 #include <map>
 #include <mutex>
 #include <utils/Singleton.h>
@@ -39,7 +41,6 @@
         : Singleton<AAudioClientTracker>() {
 }
 
-
 std::string AAudioClientTracker::dump() const {
     std::stringstream result;
     const bool isLocked = AAudio_tryUntilTrue(
@@ -198,7 +199,9 @@
 
     result << "  client: pid = " << mProcessId << " has " << mStreams.size() << " streams\n";
     for (const auto& serviceStream : mStreams) {
-        result << "     stream: 0x" << std::hex << serviceStream->getHandle() << std::dec << "\n";
+        result << "     stream: 0x" << std::setfill('0') << std::setw(8) << std::hex
+               << serviceStream->getHandle()
+               << std::dec << std::setfill(' ') << "\n";
     }
 
     if (isLocked) {
diff --git a/services/oboeservice/AAudioEndpointManager.cpp b/services/oboeservice/AAudioEndpointManager.cpp
index 11fd9f6..ab8f4ed 100644
--- a/services/oboeservice/AAudioEndpointManager.cpp
+++ b/services/oboeservice/AAudioEndpointManager.cpp
@@ -67,11 +67,17 @@
 
         result << "Exclusive MMAP Endpoints: " << mExclusiveStreams.size() << "\n";
         index = 0;
-        for (const auto &output : mExclusiveStreams) {
+        for (const auto &stream : mExclusiveStreams) {
             result << "  #" << index++ << ":";
-            result << output->dump() << "\n";
+            result << stream->dump() << "\n";
         }
 
+        result << "  ExclusiveSearchCount:  " << mExclusiveSearchCount << "\n";
+        result << "  ExclusiveFoundCount:   " << mExclusiveFoundCount << "\n";
+        result << "  ExclusiveOpenCount:    " << mExclusiveOpenCount << "\n";
+        result << "  ExclusiveCloseCount:   " << mExclusiveCloseCount << "\n";
+        result << "\n";
+
         if (isExclusiveLocked) {
             mExclusiveLock.unlock();
         }
@@ -79,11 +85,17 @@
 
     result << "Shared Endpoints: " << mSharedStreams.size() << "\n";
     index = 0;
-    for (const auto &input : mSharedStreams) {
+    for (const auto &stream : mSharedStreams) {
         result << "  #" << index++ << ":";
-        result << input->dump() << "\n";
+        result << stream->dump() << "\n";
     }
 
+    result << "  SharedSearchCount:     " << mSharedSearchCount << "\n";
+    result << "  SharedFoundCount:      " << mSharedFoundCount << "\n";
+    result << "  SharedOpenCount:       " << mSharedOpenCount << "\n";
+    result << "  SharedCloseCount:      " << mSharedCloseCount << "\n";
+    result << "\n";
+
     if (isSharedLocked) {
         mSharedLock.unlock();
     }
@@ -95,8 +107,10 @@
 sp<AAudioServiceEndpoint> AAudioEndpointManager::findExclusiveEndpoint_l(
         const AAudioStreamConfiguration &configuration) {
     sp<AAudioServiceEndpoint> endpoint;
+    mExclusiveSearchCount++;
     for (const auto ep : mExclusiveStreams) {
         if (ep->matches(configuration)) {
+            mExclusiveFoundCount++;
             endpoint = ep;
             break;
         }
@@ -111,8 +125,10 @@
 sp<AAudioServiceEndpointShared> AAudioEndpointManager::findSharedEndpoint_l(
         const AAudioStreamConfiguration &configuration) {
     sp<AAudioServiceEndpointShared> endpoint;
+    mSharedSearchCount++;
     for (const auto ep  : mSharedStreams) {
         if (ep->matches(configuration)) {
+            mSharedFoundCount++;
             endpoint = ep;
             break;
         }
@@ -146,21 +162,21 @@
 
     // If we find an existing one then this one cannot be exclusive.
     if (endpoint.get() != nullptr) {
-        ALOGE("openExclusiveEndpoint() already in use");
+        ALOGW("openExclusiveEndpoint() already in use");
         // Already open so do not allow a second stream.
         return nullptr;
     } else {
         sp<AAudioServiceEndpointMMAP> endpointMMap = new AAudioServiceEndpointMMAP();
-        ALOGD("openExclusiveEndpoint(), no match so try to open MMAP %p for dev %d",
+        ALOGV("openExclusiveEndpoint(), no match so try to open MMAP %p for dev %d",
               endpointMMap.get(), configuration.getDeviceId());
         endpoint = endpointMMap;
 
         aaudio_result_t result = endpoint->open(request);
         if (result != AAUDIO_OK) {
-            ALOGE("openExclusiveEndpoint(), open failed");
             endpoint.clear();
         } else {
             mExclusiveStreams.push_back(endpointMMap);
+            mExclusiveOpenCount++;
         }
     }
 
@@ -201,13 +217,13 @@
         if (endpoint.get() != nullptr) {
             aaudio_result_t result = endpoint->open(request);
             if (result != AAUDIO_OK) {
-                ALOGE("%s(), open failed", __func__);
                 endpoint.clear();
             } else {
                 mSharedStreams.push_back(endpoint);
+                mSharedOpenCount++;
             }
         }
-        ALOGD("%s(), created endpoint %p, requested device = %d, dir = %d",
+        ALOGV("%s(), created endpoint %p, requested device = %d, dir = %d",
               __func__, endpoint.get(), configuration.getDeviceId(), (int)direction);
         IPCThreadState::self()->restoreCallingIdentity(token);
     }
@@ -244,7 +260,8 @@
                 mExclusiveStreams.end());
 
         serviceEndpoint->close();
-        ALOGD("%s() %p for device %d",
+        mExclusiveCloseCount++;
+        ALOGV("%s() %p for device %d",
               __func__, serviceEndpoint.get(), serviceEndpoint->getDeviceId());
     }
 }
@@ -266,7 +283,8 @@
                 mSharedStreams.end());
 
         serviceEndpoint->close();
-        ALOGD("%s() %p for device %d",
+        mSharedCloseCount++;
+        ALOGV("%s() %p for device %d",
               __func__, serviceEndpoint.get(), serviceEndpoint->getDeviceId());
     }
 }
diff --git a/services/oboeservice/AAudioEndpointManager.h b/services/oboeservice/AAudioEndpointManager.h
index f6aeb5a..193bdee 100644
--- a/services/oboeservice/AAudioEndpointManager.h
+++ b/services/oboeservice/AAudioEndpointManager.h
@@ -87,8 +87,17 @@
     mutable std::mutex                                     mExclusiveLock;
     std::vector<android::sp<AAudioServiceEndpointMMAP>>    mExclusiveStreams;
 
+    // Modified under a lock.
+    int32_t mExclusiveSearchCount = 0; // number of times we SEARCHED for an exclusive endpoint
+    int32_t mExclusiveFoundCount  = 0; // number of times we FOUND an exclusive endpoint
+    int32_t mExclusiveOpenCount   = 0; // number of times we OPENED an exclusive endpoint
+    int32_t mExclusiveCloseCount  = 0; // number of times we CLOSED an exclusive endpoint
+    // Same as above but for SHARED endpoints.
+    int32_t mSharedSearchCount    = 0;
+    int32_t mSharedFoundCount     = 0;
+    int32_t mSharedOpenCount      = 0;
+    int32_t mSharedCloseCount     = 0;
 };
-
 } /* namespace aaudio */
 
 #endif //AAUDIO_AAUDIO_ENDPOINT_MANAGER_H
diff --git a/services/oboeservice/AAudioServiceEndpointMMAP.cpp b/services/oboeservice/AAudioServiceEndpointMMAP.cpp
index 52990da..ea30c6d 100644
--- a/services/oboeservice/AAudioServiceEndpointMMAP.cpp
+++ b/services/oboeservice/AAudioServiceEndpointMMAP.cpp
@@ -84,6 +84,7 @@
 
     const audio_content_type_t contentType =
             AAudioConvert_contentTypeToInternal(getContentType());
+    // Usage only used for OUTPUT
     const audio_usage_t usage = (direction == AAUDIO_DIRECTION_OUTPUT)
             ? AAudioConvert_usageToInternal(getUsage())
             : AUDIO_USAGE_UNKNOWN;
diff --git a/services/oboeservice/AAudioServiceEndpointShared.cpp b/services/oboeservice/AAudioServiceEndpointShared.cpp
index f08a52f..63b9983 100644
--- a/services/oboeservice/AAudioServiceEndpointShared.cpp
+++ b/services/oboeservice/AAudioServiceEndpointShared.cpp
@@ -78,7 +78,6 @@
     setSamplesPerFrame(mStreamInternal->getSamplesPerFrame());
     setDeviceId(mStreamInternal->getDeviceId());
     setSessionId(mStreamInternal->getSessionId());
-    ALOGD("open() deviceId = %d, sessionId = %d", getDeviceId(), getSessionId());
     mFramesPerBurst = mStreamInternal->getFramesPerBurst();
 
     return result;
diff --git a/services/oboeservice/AAudioServiceStreamShared.cpp b/services/oboeservice/AAudioServiceStreamShared.cpp
index 75d88cf..864a008 100644
--- a/services/oboeservice/AAudioServiceStreamShared.cpp
+++ b/services/oboeservice/AAudioServiceStreamShared.cpp
@@ -122,19 +122,18 @@
 
     aaudio_result_t result = AAudioServiceStreamBase::open(request, AAUDIO_SHARING_MODE_SHARED);
     if (result != AAUDIO_OK) {
-        ALOGE("open() returned %d", result);
+        ALOGE("%s() returned %d", __func__, result);
         return result;
     }
 
     const AAudioStreamConfiguration &configurationInput = request.getConstantConfiguration();
 
-
     // Is the request compatible with the shared endpoint?
     setFormat(configurationInput.getFormat());
     if (getFormat() == AAUDIO_FORMAT_UNSPECIFIED) {
         setFormat(AAUDIO_FORMAT_PCM_FLOAT);
     } else if (getFormat() != AAUDIO_FORMAT_PCM_FLOAT) {
-        ALOGE("open() mAudioFormat = %d, need FLOAT", getFormat());
+        ALOGD("%s() mAudioFormat = %d, need FLOAT", __func__, getFormat());
         result = AAUDIO_ERROR_INVALID_FORMAT;
         goto error;
     }
@@ -143,8 +142,8 @@
     if (getSampleRate() == AAUDIO_UNSPECIFIED) {
         setSampleRate(mServiceEndpoint->getSampleRate());
     } else if (getSampleRate() != mServiceEndpoint->getSampleRate()) {
-        ALOGE("open() mSampleRate = %d, need %d",
-              getSampleRate(), mServiceEndpoint->getSampleRate());
+        ALOGD("%s() mSampleRate = %d, need %d",
+              __func__, getSampleRate(), mServiceEndpoint->getSampleRate());
         result = AAUDIO_ERROR_INVALID_RATE;
         goto error;
     }
@@ -153,8 +152,8 @@
     if (getSamplesPerFrame() == AAUDIO_UNSPECIFIED) {
         setSamplesPerFrame(mServiceEndpoint->getSamplesPerFrame());
     } else if (getSamplesPerFrame() != mServiceEndpoint->getSamplesPerFrame()) {
-        ALOGE("open() mSamplesPerFrame = %d, need %d",
-              getSamplesPerFrame(), mServiceEndpoint->getSamplesPerFrame());
+        ALOGD("%s() mSamplesPerFrame = %d, need %d",
+              __func__, getSamplesPerFrame(), mServiceEndpoint->getSamplesPerFrame());
         result = AAUDIO_ERROR_OUT_OF_RANGE;
         goto error;
     }
@@ -173,16 +172,13 @@
         mAudioDataQueue = new SharedRingBuffer();
         result = mAudioDataQueue->allocate(calculateBytesPerFrame(), getBufferCapacity());
         if (result != AAUDIO_OK) {
-            ALOGE("open() could not allocate FIFO with %d frames",
-                  getBufferCapacity());
+            ALOGE("%s() could not allocate FIFO with %d frames",
+                  __func__, getBufferCapacity());
             result = AAUDIO_ERROR_NO_MEMORY;
             goto error;
         }
     }
 
-    ALOGD("open() actual rate = %d, channels = %d, deviceId = %d",
-          getSampleRate(), getSamplesPerFrame(), mServiceEndpoint->getDeviceId());
-
     result = mServiceEndpoint->registerStream(keep);
     if (result != AAUDIO_OK) {
         goto error;
@@ -217,7 +213,7 @@
 {
     std::lock_guard<std::mutex> lock(mAudioDataQueueLock);
     if (mAudioDataQueue == nullptr) {
-        ALOGE("getAudioDataDescription(): mUpMessageQueue null! - stream not open");
+        ALOGE("%s(): mUpMessageQueue null! - stream not open", __func__);
         return AAUDIO_ERROR_NULL;
     }
     // Gather information on the data queue.
@@ -255,8 +251,8 @@
         int64_t offset = mTimestampPositionOffset.load();
         // TODO, do not go below starting value
         position -= offset; // Offset from shared MMAP stream
-        ALOGV("getHardwareTimestamp() %8lld = %8lld - %8lld",
-              (long long) position, (long long) (position + offset), (long long) offset);
+        ALOGV("%s() %8lld = %8lld - %8lld",
+              __func__, (long long) position, (long long) (position + offset), (long long) offset);
     }
     *positionFrames = position;
     return result;