diff --git a/media/libaaudio/include/aaudio/AAudio.h b/media/libaaudio/include/aaudio/AAudio.h
index 4b08295..f97fe4d 100644
--- a/media/libaaudio/include/aaudio/AAudio.h
+++ b/media/libaaudio/include/aaudio/AAudio.h
@@ -565,6 +565,145 @@
 };
 typedef int32_t aaudio_session_id_t;
 
+/**
+ * Defines the audio channel mask.
+ * Channel masks are used to describe the samples and their
+ * arrangement in the audio frame. They are also used in the endpoint
+ * (e.g. a USB audio interface, a DAC connected to headphones) to
+ * specify allowable configurations of a particular device.
+ *
+ * Added in API level 32.
+ */
+enum {
+    /**
+     * Invalid channel mask
+     */
+    AAUDIO_CHANNEL_INVALID = -1,
+
+    /**
+     * Output audio channel mask
+     */
+    AAUDIO_CHANNEL_FRONT_LEFT = 1 << 0,
+    AAUDIO_CHANNEL_FRONT_RIGHT = 1 << 1,
+    AAUDIO_CHANNEL_FRONT_CENTER = 1 << 2,
+    AAUDIO_CHANNEL_LOW_FREQUENCY = 1 << 3,
+    AAUDIO_CHANNEL_BACK_LEFT = 1 << 4,
+    AAUDIO_CHANNEL_BACK_RIGHT = 1 << 5,
+    AAUDIO_CHANNEL_FRONT_LEFT_OF_CENTER = 1 << 6,
+    AAUDIO_CHANNEL_FRONT_RIGHT_OF_CENTER = 1 << 7,
+    AAUDIO_CHANNEL_BACK_CENTER = 1 << 8,
+    AAUDIO_CHANNEL_SIDE_LEFT = 1 << 9,
+    AAUDIO_CHANNEL_SIDE_RIGHT = 1 << 10,
+    AAUDIO_CHANNEL_TOP_CENTER = 1 << 11,
+    AAUDIO_CHANNEL_TOP_FRONT_LEFT = 1 << 12,
+    AAUDIO_CHANNEL_TOP_FRONT_CENTER = 1 << 13,
+    AAUDIO_CHANNEL_TOP_FRONT_RIGHT = 1 << 14,
+    AAUDIO_CHANNEL_TOP_BACK_LEFT = 1 << 15,
+    AAUDIO_CHANNEL_TOP_BACK_CENTER = 1 << 16,
+    AAUDIO_CHANNEL_TOP_BACK_RIGHT = 1 << 17,
+    AAUDIO_CHANNEL_TOP_SIDE_LEFT = 1 << 18,
+    AAUDIO_CHANNEL_TOP_SIDE_RIGHT = 1 << 19,
+    AAUDIO_CHANNEL_BOTTOM_FRONT_LEFT = 1 << 20,
+    AAUDIO_CHANNEL_BOTTOM_FRONT_CENTER = 1 << 21,
+    AAUDIO_CHANNEL_BOTTOM_FRONT_RIGHT = 1 << 22,
+    AAUDIO_CHANNEL_LOW_FREQUENCY_2 = 1 << 23,
+    AAUDIO_CHANNEL_FRONT_WIDE_LEFT = 1 << 24,
+    AAUDIO_CHANNEL_FRONT_WIDE_RIGHT = 1 << 25,
+
+    AAUDIO_CHANNEL_MONO = AAUDIO_CHANNEL_FRONT_LEFT,
+    AAUDIO_CHANNEL_STEREO = AAUDIO_CHANNEL_FRONT_LEFT |
+                            AAUDIO_CHANNEL_FRONT_RIGHT,
+    AAUDIO_CHANNEL_2POINT1 = AAUDIO_CHANNEL_FRONT_LEFT |
+                             AAUDIO_CHANNEL_FRONT_RIGHT |
+                             AAUDIO_CHANNEL_LOW_FREQUENCY,
+    AAUDIO_CHANNEL_TRI = AAUDIO_CHANNEL_FRONT_LEFT |
+                         AAUDIO_CHANNEL_FRONT_RIGHT |
+                         AAUDIO_CHANNEL_FRONT_CENTER,
+    AAUDIO_CHANNEL_TRI_BACK = AAUDIO_CHANNEL_FRONT_LEFT |
+                              AAUDIO_CHANNEL_FRONT_RIGHT |
+                              AAUDIO_CHANNEL_BACK_CENTER,
+    AAUDIO_CHANNEL_3POINT1 = AAUDIO_CHANNEL_FRONT_LEFT |
+                             AAUDIO_CHANNEL_FRONT_RIGHT |
+                             AAUDIO_CHANNEL_FRONT_CENTER |
+                             AAUDIO_CHANNEL_LOW_FREQUENCY,
+    AAUDIO_CHANNEL_2POINT0POINT2 = AAUDIO_CHANNEL_FRONT_LEFT |
+                                   AAUDIO_CHANNEL_FRONT_RIGHT |
+                                   AAUDIO_CHANNEL_TOP_SIDE_LEFT |
+                                   AAUDIO_CHANNEL_TOP_SIDE_RIGHT,
+    AAUDIO_CHANNEL_2POINT1POINT2 = AAUDIO_CHANNEL_2POINT0POINT2 |
+                                   AAUDIO_CHANNEL_LOW_FREQUENCY,
+    AAUDIO_CHANNEL_3POINT0POINT2 = AAUDIO_CHANNEL_FRONT_LEFT |
+                                   AAUDIO_CHANNEL_FRONT_RIGHT |
+                                   AAUDIO_CHANNEL_FRONT_CENTER |
+                                   AAUDIO_CHANNEL_TOP_SIDE_LEFT |
+                                   AAUDIO_CHANNEL_TOP_SIDE_RIGHT,
+    AAUDIO_CHANNEL_3POINT1POINT2 = AAUDIO_CHANNEL_3POINT0POINT2 |
+                                   AAUDIO_CHANNEL_LOW_FREQUENCY,
+    AAUDIO_CHANNEL_QUAD = AAUDIO_CHANNEL_FRONT_LEFT |
+                          AAUDIO_CHANNEL_FRONT_RIGHT |
+                          AAUDIO_CHANNEL_BACK_LEFT |
+                          AAUDIO_CHANNEL_BACK_RIGHT,
+    AAUDIO_CHANNEL_QUAD_SIDE = AAUDIO_CHANNEL_FRONT_LEFT |
+                               AAUDIO_CHANNEL_FRONT_RIGHT |
+                               AAUDIO_CHANNEL_SIDE_LEFT |
+                               AAUDIO_CHANNEL_SIDE_RIGHT,
+    AAUDIO_CHANNEL_SURROUND = AAUDIO_CHANNEL_FRONT_LEFT |
+                              AAUDIO_CHANNEL_FRONT_RIGHT |
+                              AAUDIO_CHANNEL_FRONT_CENTER |
+                              AAUDIO_CHANNEL_BACK_CENTER,
+    AAUDIO_CHANNEL_PENTA = AAUDIO_CHANNEL_QUAD |
+                           AAUDIO_CHANNEL_FRONT_CENTER,
+    // aka 5POINT1_BACK
+    AAUDIO_CHANNEL_5POINT1 = AAUDIO_CHANNEL_FRONT_LEFT |
+                             AAUDIO_CHANNEL_FRONT_RIGHT |
+                             AAUDIO_CHANNEL_FRONT_CENTER |
+                             AAUDIO_CHANNEL_LOW_FREQUENCY |
+                             AAUDIO_CHANNEL_BACK_LEFT |
+                             AAUDIO_CHANNEL_BACK_RIGHT,
+    AAUDIO_CHANNEL_5POINT1_SIDE = AAUDIO_CHANNEL_FRONT_LEFT |
+                                  AAUDIO_CHANNEL_FRONT_RIGHT |
+                                  AAUDIO_CHANNEL_FRONT_CENTER |
+                                  AAUDIO_CHANNEL_LOW_FREQUENCY |
+                                  AAUDIO_CHANNEL_SIDE_LEFT |
+                                  AAUDIO_CHANNEL_SIDE_RIGHT,
+    AAUDIO_CHANNEL_6POINT1 = AAUDIO_CHANNEL_FRONT_LEFT |
+                             AAUDIO_CHANNEL_FRONT_RIGHT |
+                             AAUDIO_CHANNEL_FRONT_CENTER |
+                             AAUDIO_CHANNEL_LOW_FREQUENCY |
+                             AAUDIO_CHANNEL_BACK_LEFT |
+                             AAUDIO_CHANNEL_BACK_RIGHT |
+                             AAUDIO_CHANNEL_BACK_CENTER,
+    AAUDIO_CHANNEL_7POINT1 = AAUDIO_CHANNEL_5POINT1 |
+                             AAUDIO_CHANNEL_SIDE_LEFT |
+                             AAUDIO_CHANNEL_SIDE_RIGHT,
+    AAUDIO_CHANNEL_5POINT1POINT2 = AAUDIO_CHANNEL_5POINT1 |
+                                   AAUDIO_CHANNEL_TOP_SIDE_LEFT |
+                                   AAUDIO_CHANNEL_TOP_SIDE_RIGHT,
+    AAUDIO_CHANNEL_5POINT1POINT4 = AAUDIO_CHANNEL_5POINT1 |
+                                   AAUDIO_CHANNEL_TOP_FRONT_LEFT |
+                                   AAUDIO_CHANNEL_TOP_FRONT_RIGHT |
+                                   AAUDIO_CHANNEL_TOP_BACK_LEFT |
+                                   AAUDIO_CHANNEL_TOP_BACK_RIGHT,
+    AAUDIO_CHANNEL_7POINT1POINT2 = AAUDIO_CHANNEL_7POINT1 |
+                                   AAUDIO_CHANNEL_TOP_SIDE_LEFT |
+                                   AAUDIO_CHANNEL_TOP_SIDE_RIGHT,
+    AAUDIO_CHANNEL_7POINT1POINT4 = AAUDIO_CHANNEL_7POINT1 |
+                                   AAUDIO_CHANNEL_TOP_FRONT_LEFT |
+                                   AAUDIO_CHANNEL_TOP_FRONT_RIGHT |
+                                   AAUDIO_CHANNEL_TOP_BACK_LEFT |
+                                   AAUDIO_CHANNEL_TOP_BACK_RIGHT,
+    AAUDIO_CHANNEL_9POINT1POINT4 = AAUDIO_CHANNEL_7POINT1POINT4 |
+                                   AAUDIO_CHANNEL_FRONT_WIDE_LEFT |
+                                   AAUDIO_CHANNEL_FRONT_WIDE_RIGHT,
+    AAUDIO_CHANNEL_9POINT1POINT6 = AAUDIO_CHANNEL_9POINT1POINT4 |
+                                   AAUDIO_CHANNEL_TOP_SIDE_LEFT |
+                                   AAUDIO_CHANNEL_TOP_SIDE_RIGHT,
+
+    AAUDIO_CHANNEL_FRONT_BACK = AAUDIO_CHANNEL_FRONT_CENTER |
+                                AAUDIO_CHANNEL_BACK_CENTER,
+};
+typedef uint32_t aaudio_channel_mask_t;
+
 typedef struct AAudioStreamStruct         AAudioStream;
 typedef struct AAudioStreamBuilderStruct  AAudioStreamBuilder;
 
@@ -699,6 +838,11 @@
  * If an exact value is specified then an opened stream will use that value.
  * If a stream cannot be opened with the specified value then the open will fail.
  *
+ * As the channel count provided here may be different from the corresponding channel count
+ * of channel mask used in {@link AAudioStreamBuilder_setChannelMask}, the last called function
+ * will be respected if both this function and {@link AAudioStreamBuilder_setChannelMask} are
+ * called.
+ *
  * Available since API level 26.
  *
  * @param builder reference provided by AAudio_createStreamBuilder()
@@ -714,6 +858,8 @@
  *
  * @param builder reference provided by AAudio_createStreamBuilder()
  * @param samplesPerFrame Number of samples in a frame.
+ *
+ * @deprecated use {@link AAudioStreamBuilder_setChannelCount}
  */
 AAUDIO_API void AAudioStreamBuilder_setSamplesPerFrame(AAudioStreamBuilder* builder,
                                                        int32_t samplesPerFrame) __INTRODUCED_IN(26);
@@ -1136,6 +1282,32 @@
 AAUDIO_API aaudio_result_t  AAudioStreamBuilder_delete(AAudioStreamBuilder* builder)
     __INTRODUCED_IN(26);
 
+/**
+ * Set audio channel mask for the stream.
+ *
+ * The default, if you do not call this function, is {@link #AAUDIO_UNSPECIFIED}.
+ * If both channel mask and count are not set, then stereo will then be chosen when the
+ * stream is opened.
+ * After opening a stream with an unspecified value, the application must query for the
+ * actual value, which may vary by device.
+ *
+ * If an exact value is specified then an opened stream will use that value.
+ * If a stream cannot be opened with the specified value then the open will fail.
+ *
+ * As the corresponding channel count of provided channel mask here may be different
+ * from the channel count used in {@link AAudioStreamBuilder_setChannelCount} or
+ * {@link AAudioStreamBuilder_setSamplesPerFrame}, the last called function will be
+ * respected if this function and {@link AAudioStreamBuilder_setChannelCount} or
+ * {@link AAudioStreamBuilder_setSamplesPerFrame} are called.
+ *
+ * Available since API level 32.
+ *
+ * @param builder reference provided by AAudio_createStreamBuilder()
+ * @param channelMask Audio channel mask desired.
+ */
+AAUDIO_API void AAudioStreamBuilder_setChannelMask(AAudioStreamBuilder* builder,
+        aaudio_channel_mask_t channelMask) __INTRODUCED_IN(32);
+
 // ============================================================
 // Stream Control
 // ============================================================
@@ -1652,6 +1824,18 @@
 AAUDIO_API bool AAudioStream_isPrivacySensitive(AAudioStream* stream)
         __INTRODUCED_IN(30);
 
+/**
+ * Return the channel mask for the stream. This will be the mask set using
+ * {@link #AAudioStreamBuilder_setChannelMask}, or {@link #AAUDIO_UNSPECIFIED} otherwise.
+ *
+ * Available since API level 32.
+ *
+ * @param stream reference provided by AAudioStreamBuilder_openStream()
+ * @return actual channel mask
+ */
+AAUDIO_API aaudio_channel_mask_t AAudioStream_getChannelMask(AAudioStream* stream)
+        __INTRODUCED_IN(32);
+
 #ifdef __cplusplus
 }
 #endif
diff --git a/media/libaaudio/src/binding/AAudioStreamConfiguration.cpp b/media/libaaudio/src/binding/AAudioStreamConfiguration.cpp
index 2d501ef..dc19bb3 100644
--- a/media/libaaudio/src/binding/AAudioStreamConfiguration.cpp
+++ b/media/libaaudio/src/binding/AAudioStreamConfiguration.cpp
@@ -30,7 +30,7 @@
 using android::media::audio::common::AudioFormat;
 
 AAudioStreamConfiguration::AAudioStreamConfiguration(const StreamParameters& parcelable) {
-    setSamplesPerFrame(parcelable.samplesPerFrame);
+    setChannelMask(parcelable.channelMask);
     setSampleRate(parcelable.sampleRate);
     setDeviceId(parcelable.deviceId);
     static_assert(sizeof(aaudio_sharing_mode_t) == sizeof(parcelable.sharingMode));
@@ -63,7 +63,7 @@
 
 StreamParameters AAudioStreamConfiguration::parcelable() const {
     StreamParameters result;
-    result.samplesPerFrame = getSamplesPerFrame();
+    result.channelMask = getChannelMask();
     result.sampleRate = getSampleRate();
     result.deviceId = getDeviceId();
     static_assert(sizeof(aaudio_sharing_mode_t) == sizeof(result.sharingMode));
diff --git a/media/libaaudio/src/binding/aidl/aaudio/StreamParameters.aidl b/media/libaaudio/src/binding/aidl/aaudio/StreamParameters.aidl
index b7c4f70..1a971b0 100644
--- a/media/libaaudio/src/binding/aidl/aaudio/StreamParameters.aidl
+++ b/media/libaaudio/src/binding/aidl/aaudio/StreamParameters.aidl
@@ -19,7 +19,7 @@
 import android.media.audio.common.AudioFormat;
 
 parcelable StreamParameters {
-    int                                       samplesPerFrame;  //      = AAUDIO_UNSPECIFIED;
+    int                                       channelMask;  //          = AAUDIO_UNSPECIFIED;
     int                                       sampleRate;  //           = AAUDIO_UNSPECIFIED;
     int                                       deviceId;  //             = AAUDIO_UNSPECIFIED;
     int /* aaudio_sharing_mode_t */           sharingMode;  //          = AAUDIO_SHARING_MODE_SHARED;
diff --git a/media/libaaudio/src/client/AudioStreamInternal.cpp b/media/libaaudio/src/client/AudioStreamInternal.cpp
index 6d2d464..3063a7b 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternal.cpp
@@ -123,9 +123,9 @@
 
     request.getConfiguration().setDeviceId(getDeviceId());
     request.getConfiguration().setSampleRate(getSampleRate());
-    request.getConfiguration().setSamplesPerFrame(getSamplesPerFrame());
     request.getConfiguration().setDirection(getDirection());
     request.getConfiguration().setSharingMode(getSharingMode());
+    request.getConfiguration().setChannelMask(getChannelMask());
 
     request.getConfiguration().setUsage(getUsage());
     request.getConfiguration().setContentType(getContentType());
@@ -138,7 +138,8 @@
 
     mServiceStreamHandle = mServiceInterface.openStream(request, configurationOutput);
     if (mServiceStreamHandle < 0
-            && request.getConfiguration().getSamplesPerFrame() == 1 // mono?
+            && (request.getConfiguration().getSamplesPerFrame() == 1
+                    || request.getConfiguration().getChannelMask() == AAUDIO_CHANNEL_MONO)
             && getDirection() == AAUDIO_DIRECTION_OUTPUT
             && !isInService()) {
         // if that failed then try switching from mono to stereo if OUTPUT.
@@ -146,7 +147,7 @@
         // that writes to a stereo MMAP stream.
         ALOGD("%s() - openStream() returned %d, try switching from MONO to STEREO",
               __func__, mServiceStreamHandle);
-        request.getConfiguration().setSamplesPerFrame(2); // stereo
+        request.getConfiguration().setChannelMask(AAUDIO_CHANNEL_STEREO);
         mServiceStreamHandle = mServiceInterface.openStream(request, configurationOutput);
     }
     if (mServiceStreamHandle < 0) {
@@ -174,9 +175,10 @@
         goto error;
     }
     // Save results of the open.
-    if (getSamplesPerFrame() == AAUDIO_UNSPECIFIED) {
-        setSamplesPerFrame(configurationOutput.getSamplesPerFrame());
+    if (getChannelMask() == AAUDIO_UNSPECIFIED) {
+        setChannelMask(configurationOutput.getChannelMask());
     }
+
     mDeviceChannelCount = configurationOutput.getSamplesPerFrame();
 
     setSampleRate(configurationOutput.getSampleRate());
diff --git a/media/libaaudio/src/core/AAudioAudio.cpp b/media/libaaudio/src/core/AAudioAudio.cpp
index d103aca..02e7f5f 100644
--- a/media/libaaudio/src/core/AAudioAudio.cpp
+++ b/media/libaaudio/src/core/AAudioAudio.cpp
@@ -128,7 +128,8 @@
                                                        int32_t samplesPerFrame)
 {
     AudioStreamBuilder *streamBuilder = convertAAudioBuilderToStreamBuilder(builder);
-    streamBuilder->setSamplesPerFrame(samplesPerFrame);
+    const aaudio_channel_mask_t channelMask = AAudioConvert_channelCountToMask(samplesPerFrame);
+    streamBuilder->setChannelMask(channelMask);
 }
 
 AAUDIO_API void AAudioStreamBuilder_setDirection(AAudioStreamBuilder* builder,
@@ -223,6 +224,13 @@
     streamBuilder->setFramesPerDataCallback(frames);
 }
 
+AAUDIO_API void AAudioStreamBuilder_setChannelMask(AAudioStreamBuilder* builder,
+                                                   aaudio_channel_mask_t channelMask)
+{
+    AudioStreamBuilder *streamBuilder = convertAAudioBuilderToStreamBuilder(builder);
+    streamBuilder->setChannelMask(channelMask);
+}
+
 AAUDIO_API aaudio_result_t  AAudioStreamBuilder_openStream(AAudioStreamBuilder* builder,
                                                      AAudioStream** streamPtr)
 {
@@ -562,3 +570,11 @@
     AudioStream *audioStream = convertAAudioStreamToAudioStream(stream);
     return audioStream->isPrivacySensitive();
 }
+
+AAUDIO_API aaudio_channel_mask_t AAudioStream_getChannelMask(AAudioStream* stream)
+{
+    AudioStream *audioStream = convertAAudioStreamToAudioStream(stream);
+    const aaudio_channel_mask_t channelMask = audioStream->getChannelMask();
+    // Do not return channel index masks as they are not public.
+    return AAudio_isChannelIndexMask(channelMask) ? AAUDIO_UNSPECIFIED : channelMask;
+}
diff --git a/media/libaaudio/src/core/AAudioStreamParameters.cpp b/media/libaaudio/src/core/AAudioStreamParameters.cpp
index acfac24..59d94eb 100644
--- a/media/libaaudio/src/core/AAudioStreamParameters.cpp
+++ b/media/libaaudio/src/core/AAudioStreamParameters.cpp
@@ -49,6 +49,7 @@
     mIsPrivacySensitive   = other.mIsPrivacySensitive;
     mOpPackageName        = other.mOpPackageName;
     mAttributionTag       = other.mAttributionTag;
+    mChannelMask          = other.mChannelMask;
 }
 
 static aaudio_result_t isFormatValid(audio_format_t format) {
@@ -187,7 +188,94 @@
             // break;
     }
 
-    return AAUDIO_OK;
+    return validateChannelMask();
+}
+
+bool AAudioStreamParameters::validateChannelMask() const {
+    if (mChannelMask == AAUDIO_UNSPECIFIED) {
+        return AAUDIO_OK;
+    }
+
+    if (mChannelMask & AAUDIO_CHANNEL_BIT_INDEX) {
+        switch (mChannelMask) {
+            case AAUDIO_CHANNEL_INDEX_MASK_1:
+            case AAUDIO_CHANNEL_INDEX_MASK_2:
+            case AAUDIO_CHANNEL_INDEX_MASK_3:
+            case AAUDIO_CHANNEL_INDEX_MASK_4:
+            case AAUDIO_CHANNEL_INDEX_MASK_5:
+            case AAUDIO_CHANNEL_INDEX_MASK_6:
+            case AAUDIO_CHANNEL_INDEX_MASK_7:
+            case AAUDIO_CHANNEL_INDEX_MASK_8:
+            case AAUDIO_CHANNEL_INDEX_MASK_9:
+            case AAUDIO_CHANNEL_INDEX_MASK_10:
+            case AAUDIO_CHANNEL_INDEX_MASK_11:
+            case AAUDIO_CHANNEL_INDEX_MASK_12:
+            case AAUDIO_CHANNEL_INDEX_MASK_13:
+            case AAUDIO_CHANNEL_INDEX_MASK_14:
+            case AAUDIO_CHANNEL_INDEX_MASK_15:
+            case AAUDIO_CHANNEL_INDEX_MASK_16:
+            case AAUDIO_CHANNEL_INDEX_MASK_17:
+            case AAUDIO_CHANNEL_INDEX_MASK_18:
+            case AAUDIO_CHANNEL_INDEX_MASK_19:
+            case AAUDIO_CHANNEL_INDEX_MASK_20:
+            case AAUDIO_CHANNEL_INDEX_MASK_21:
+            case AAUDIO_CHANNEL_INDEX_MASK_22:
+            case AAUDIO_CHANNEL_INDEX_MASK_23:
+            case AAUDIO_CHANNEL_INDEX_MASK_24:
+                return AAUDIO_OK;
+            default:
+                ALOGD("Invalid channel index mask %#x", mChannelMask);
+                return AAUDIO_ERROR_ILLEGAL_ARGUMENT;
+        }
+    }
+
+    if (getDirection() == AAUDIO_DIRECTION_INPUT) {
+        switch (mChannelMask) {
+            case AAUDIO_CHANNEL_MONO:
+            case AAUDIO_CHANNEL_STEREO:
+            case AAUDIO_CHANNEL_FRONT_BACK:
+            case AAUDIO_CHANNEL_2POINT0POINT2:
+            case AAUDIO_CHANNEL_2POINT1POINT2:
+            case AAUDIO_CHANNEL_3POINT0POINT2:
+            case AAUDIO_CHANNEL_3POINT1POINT2:
+            case AAUDIO_CHANNEL_5POINT1:
+                return AAUDIO_OK;
+            default:
+                ALOGD("Invalid channel mask %#x, IN", mChannelMask);
+                return AAUDIO_ERROR_ILLEGAL_ARGUMENT;
+        }
+    } else {
+        switch (mChannelMask) {
+            case AAUDIO_CHANNEL_MONO:
+            case AAUDIO_CHANNEL_STEREO:
+            case AAUDIO_CHANNEL_2POINT1:
+            case AAUDIO_CHANNEL_TRI:
+            case AAUDIO_CHANNEL_TRI_BACK:
+            case AAUDIO_CHANNEL_3POINT1:
+            case AAUDIO_CHANNEL_2POINT0POINT2:
+            case AAUDIO_CHANNEL_2POINT1POINT2:
+            case AAUDIO_CHANNEL_3POINT0POINT2:
+            case AAUDIO_CHANNEL_3POINT1POINT2:
+            case AAUDIO_CHANNEL_QUAD:
+            case AAUDIO_CHANNEL_QUAD_SIDE:
+            case AAUDIO_CHANNEL_SURROUND:
+            case AAUDIO_CHANNEL_PENTA:
+            case AAUDIO_CHANNEL_5POINT1:
+            case AAUDIO_CHANNEL_5POINT1_SIDE:
+            case AAUDIO_CHANNEL_5POINT1POINT2:
+            case AAUDIO_CHANNEL_5POINT1POINT4:
+            case AAUDIO_CHANNEL_6POINT1:
+            case AAUDIO_CHANNEL_7POINT1:
+            case AAUDIO_CHANNEL_7POINT1POINT2:
+            case AAUDIO_CHANNEL_7POINT1POINT4:
+            case AAUDIO_CHANNEL_9POINT1POINT4:
+            case AAUDIO_CHANNEL_9POINT1POINT6:
+                return AAUDIO_OK;
+            default:
+                ALOGD("Invalid channel mask %#x. OUT", mChannelMask);
+                return AAUDIO_ERROR_ILLEGAL_ARGUMENT;
+        }
+    }
 }
 
 void AAudioStreamParameters::dump() const {
@@ -195,6 +283,7 @@
     ALOGD("mSessionId            = %6d", mSessionId);
     ALOGD("mSampleRate           = %6d", mSampleRate);
     ALOGD("mSamplesPerFrame      = %6d", mSamplesPerFrame);
+    ALOGD("mChannelMask          = %#x", mChannelMask);
     ALOGD("mSharingMode          = %6d", (int)mSharingMode);
     ALOGD("mAudioFormat          = %6d", (int)mAudioFormat);
     ALOGD("mDirection            = %6d", mDirection);
diff --git a/media/libaaudio/src/core/AAudioStreamParameters.h b/media/libaaudio/src/core/AAudioStreamParameters.h
index 5737052..a5c8043 100644
--- a/media/libaaudio/src/core/AAudioStreamParameters.h
+++ b/media/libaaudio/src/core/AAudioStreamParameters.h
@@ -49,13 +49,6 @@
         return mSamplesPerFrame;
     }
 
-    /**
-     * This is also known as channelCount.
-     */
-    void setSamplesPerFrame(int32_t samplesPerFrame) {
-        mSamplesPerFrame = samplesPerFrame;
-    }
-
     audio_format_t getFormat() const {
         return mAudioFormat;
     }
@@ -153,6 +146,15 @@
         mAttributionTag = attributionTag;
     }
 
+    aaudio_channel_mask_t getChannelMask() const {
+        return mChannelMask;
+    }
+
+    void setChannelMask(aaudio_channel_mask_t channelMask) {
+        mChannelMask = channelMask;
+        mSamplesPerFrame = AAudioConvert_channelMaskToCount(channelMask);
+    }
+
     /**
      * @return bytes per frame of getFormat()
      */
@@ -171,6 +173,8 @@
     void dump() const;
 
 private:
+    bool validateChannelMask() const;
+
     int32_t                         mSamplesPerFrame      = AAUDIO_UNSPECIFIED;
     int32_t                         mSampleRate           = AAUDIO_UNSPECIFIED;
     int32_t                         mDeviceId             = AAUDIO_UNSPECIFIED;
@@ -186,6 +190,7 @@
     bool                            mIsPrivacySensitive   = false;
     std::optional<std::string>      mOpPackageName        = {};
     std::optional<std::string>      mAttributionTag       = {};
+    aaudio_channel_mask_t           mChannelMask          = AAUDIO_UNSPECIFIED;
 };
 
 } /* namespace aaudio */
diff --git a/media/libaaudio/src/core/AudioStream.cpp b/media/libaaudio/src/core/AudioStream.cpp
index 09d9535..ffc3b9d 100644
--- a/media/libaaudio/src/core/AudioStream.cpp
+++ b/media/libaaudio/src/core/AudioStream.cpp
@@ -76,6 +76,7 @@
     // Copy parameters from the Builder because the Builder may be deleted after this call.
     // TODO AudioStream should be a subclass of AudioStreamParameters
     mSamplesPerFrame = builder.getSamplesPerFrame();
+    mChannelMask = builder.getChannelMask();
     mSampleRate = builder.getSampleRate();
     mDeviceId = builder.getDeviceId();
     mFormat = builder.getFormat();
diff --git a/media/libaaudio/src/core/AudioStream.h b/media/libaaudio/src/core/AudioStream.h
index 9835c8c..47693f8 100644
--- a/media/libaaudio/src/core/AudioStream.h
+++ b/media/libaaudio/src/core/AudioStream.h
@@ -270,7 +270,8 @@
     }
 
     /**
-     * This is only valid after setSamplesPerFrame() and setFormat() have been called.
+     * This is only valid after setChannelMask() and setFormat()
+     * have been called.
      */
     int32_t getBytesPerFrame() const {
         return mSamplesPerFrame * getBytesPerSample();
@@ -284,7 +285,7 @@
     }
 
     /**
-     * This is only valid after setSamplesPerFrame() and setDeviceFormat() have been called.
+     * This is only valid after setChannelMask() and setDeviceFormat() have been called.
      */
     int32_t getBytesPerDeviceFrame() const {
         return getSamplesPerFrame() * audio_bytes_per_sample(getDeviceFormat());
@@ -318,6 +319,15 @@
         return mFramesPerDataCallback;
     }
 
+    aaudio_channel_mask_t getChannelMask() const {
+        return mChannelMask;
+    }
+
+    void setChannelMask(aaudio_channel_mask_t channelMask) {
+        mChannelMask = channelMask;
+        mSamplesPerFrame = AAudioConvert_channelMaskToCount(channelMask);
+    }
+
     /**
      * @return true if data callback has been specified
      */
@@ -495,11 +505,6 @@
     }
 
     // This should not be called after the open() call.
-    void setSamplesPerFrame(int32_t samplesPerFrame) {
-        mSamplesPerFrame = samplesPerFrame;
-    }
-
-    // This should not be called after the open() call.
     void setFramesPerBurst(int32_t framesPerBurst) {
         mFramesPerBurst = framesPerBurst;
     }
@@ -633,6 +638,7 @@
 
     // These do not change after open().
     int32_t                     mSamplesPerFrame = AAUDIO_UNSPECIFIED;
+    aaudio_channel_mask_t       mChannelMask = AAUDIO_UNSPECIFIED;
     int32_t                     mSampleRate = AAUDIO_UNSPECIFIED;
     int32_t                     mDeviceId = AAUDIO_UNSPECIFIED;
     aaudio_sharing_mode_t       mSharingMode = AAUDIO_SHARING_MODE_SHARED;
diff --git a/media/libaaudio/src/core/AudioStreamBuilder.cpp b/media/libaaudio/src/core/AudioStreamBuilder.cpp
index e015592..5e1e007 100644
--- a/media/libaaudio/src/core/AudioStreamBuilder.cpp
+++ b/media/libaaudio/src/core/AudioStreamBuilder.cpp
@@ -268,8 +268,8 @@
 
 void AudioStreamBuilder::logParameters() const {
     // This is very helpful for debugging in the future. Please leave it in.
-    ALOGI("rate   = %6d, channels  = %d, format   = %d, sharing = %s, dir = %s",
-          getSampleRate(), getSamplesPerFrame(), getFormat(),
+    ALOGI("rate   = %6d, channels  = %d, channelMask = %#x, format   = %d, sharing = %s, dir = %s",
+          getSampleRate(), getSamplesPerFrame(), getChannelMask(), getFormat(),
           AAudio_convertSharingModeToShortText(getSharingMode()),
           AAudio_convertDirectionToText(getDirection()));
     ALOGI("device = %6d, sessionId = %d, perfMode = %d, callback: %s with frames = %d",
diff --git a/media/libaaudio/src/legacy/AudioStreamRecord.cpp b/media/libaaudio/src/legacy/AudioStreamRecord.cpp
index dc66742..fe8fb19 100644
--- a/media/libaaudio/src/legacy/AudioStreamRecord.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamRecord.cpp
@@ -65,11 +65,8 @@
     const audio_session_t sessionId = AAudioConvert_aaudioToAndroidSessionId(requestedSessionId);
 
     // TODO Support UNSPECIFIED in AudioRecord. For now, use stereo if unspecified.
-    int32_t samplesPerFrame = (getSamplesPerFrame() == AAUDIO_UNSPECIFIED)
-                              ? 2 : getSamplesPerFrame();
-    audio_channel_mask_t channelMask = samplesPerFrame <= 2 ?
-                               audio_channel_in_mask_from_count(samplesPerFrame) :
-                               audio_channel_mask_for_index_assignment_from_count(samplesPerFrame);
+    audio_channel_mask_t channelMask =
+            AAudio_getChannelMaskForOpen(getChannelMask(), getSamplesPerFrame(), true /*isInput*/);
 
     size_t frameCount = (builder.getBufferCapacity() == AAUDIO_UNSPECIFIED) ? 0
                         : builder.getBufferCapacity();
@@ -115,7 +112,7 @@
     constexpr int32_t kMostLikelySampleRateForFast = 48000;
     if (getFormat() == AUDIO_FORMAT_PCM_FLOAT
             && perfMode == AAUDIO_PERFORMANCE_MODE_LOW_LATENCY
-            && (samplesPerFrame <= 2) // FAST only for mono and stereo
+            && (audio_channel_count_from_in_mask(channelMask) <= 2) // FAST only for mono and stereo
             && (getSampleRate() == kMostLikelySampleRateForFast
                 || getSampleRate() == AAUDIO_UNSPECIFIED)) {
         setDeviceFormat(AUDIO_FORMAT_PCM_16_BIT);
@@ -228,7 +225,9 @@
             .set(AMEDIAMETRICS_PROP_ENCODINGCLIENT, toString(requestedFormat).c_str()).record();
 
     // Get the actual values from the AudioRecord.
-    setSamplesPerFrame(mAudioRecord->channelCount());
+    setChannelMask(AAudioConvert_androidToAAudioChannelMask(
+            mAudioRecord->channelMask(), true /*isInput*/,
+            AAudio_isChannelIndexMask(getChannelMask())));
     setSampleRate(mAudioRecord->getSampleRate());
     setBufferCapacity(getBufferCapacityFromDevice());
     setFramesPerBurst(getFramesPerBurstFromDevice());
diff --git a/media/libaaudio/src/legacy/AudioStreamTrack.cpp b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
index 1d412c0..2291ad7 100644
--- a/media/libaaudio/src/legacy/AudioStreamTrack.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
@@ -66,13 +66,8 @@
     const aaudio_session_id_t requestedSessionId = builder.getSessionId();
     const audio_session_t sessionId = AAudioConvert_aaudioToAndroidSessionId(requestedSessionId);
 
-    // Try to create an AudioTrack
-    // Use stereo if unspecified.
-    int32_t samplesPerFrame = (getSamplesPerFrame() == AAUDIO_UNSPECIFIED)
-                              ? 2 : getSamplesPerFrame();
-    audio_channel_mask_t channelMask = samplesPerFrame <= 2 ?
-                            audio_channel_out_mask_from_count(samplesPerFrame) :
-                            audio_channel_mask_for_index_assignment_from_count(samplesPerFrame);
+    audio_channel_mask_t channelMask =
+            AAudio_getChannelMaskForOpen(getChannelMask(), getSamplesPerFrame(), false /*isInput*/);
 
     audio_output_flags_t flags;
     aaudio_performance_mode_t perfMode = getPerformanceMode();
@@ -199,7 +194,9 @@
     doSetVolume();
 
     // Get the actual values from the AudioTrack.
-    setSamplesPerFrame(mAudioTrack->channelCount());
+    setChannelMask(AAudioConvert_androidToAAudioChannelMask(
+        mAudioTrack->channelMask(), false /*isInput*/,
+        AAudio_isChannelIndexMask(getChannelMask())));
     setFormat(mAudioTrack->format());
     setDeviceFormat(mAudioTrack->format());
     setSampleRate(mAudioTrack->getSampleRate());
diff --git a/media/libaaudio/src/libaaudio.map.txt b/media/libaaudio/src/libaaudio.map.txt
index 1dd44d1..8fa9e38 100644
--- a/media/libaaudio/src/libaaudio.map.txt
+++ b/media/libaaudio/src/libaaudio.map.txt
@@ -25,6 +25,7 @@
     AAudioStreamBuilder_setPrivacySensitive;   # introduced=30
     AAudioStreamBuilder_setPackageName;   # introduced=31
     AAudioStreamBuilder_setAttributionTag;   # introduced=31
+    AAudioStreamBuilder_setChannelMask;    # introduced=32
     AAudioStreamBuilder_openStream;
     AAudioStreamBuilder_delete;
     AAudioStream_close;
@@ -61,6 +62,7 @@
     AAudioStream_isMMapUsed;
     AAudioStream_isPrivacySensitive;   # introduced=30
     AAudioStream_release;        # introduced=30
+    AAudioStream_getChannelMask;  # introduced=32
   local:
     *;
 };
diff --git a/media/libaaudio/src/utility/AAudioUtilities.cpp b/media/libaaudio/src/utility/AAudioUtilities.cpp
index d795725..d829934 100644
--- a/media/libaaudio/src/utility/AAudioUtilities.cpp
+++ b/media/libaaudio/src/utility/AAudioUtilities.cpp
@@ -256,6 +256,248 @@
     return privacySensitive ? AUDIO_FLAG_CAPTURE_PRIVATE : AUDIO_FLAG_NONE;
 }
 
+audio_channel_mask_t AAudioConvert_aaudioToAndroidChannelLayoutMask(
+        aaudio_channel_mask_t channelMask, bool isInput) {
+    if (isInput) {
+        switch (channelMask) {
+            case AAUDIO_CHANNEL_MONO:
+                return AUDIO_CHANNEL_IN_MONO;
+            case AAUDIO_CHANNEL_STEREO:
+                return AUDIO_CHANNEL_IN_STEREO;
+            case AAUDIO_CHANNEL_FRONT_BACK:
+                return AUDIO_CHANNEL_IN_FRONT_BACK;
+            case AAUDIO_CHANNEL_2POINT0POINT2:
+                return AUDIO_CHANNEL_IN_2POINT0POINT2;
+            case AAUDIO_CHANNEL_2POINT1POINT2:
+                return AUDIO_CHANNEL_IN_2POINT1POINT2;
+            case AAUDIO_CHANNEL_3POINT0POINT2:
+                return AUDIO_CHANNEL_IN_3POINT0POINT2;
+            case AAUDIO_CHANNEL_3POINT1POINT2:
+                return AUDIO_CHANNEL_IN_3POINT1POINT2;
+            case AAUDIO_CHANNEL_5POINT1:
+                return AUDIO_CHANNEL_IN_5POINT1;
+            default:
+                ALOGE("%s() %#x unrecognized", __func__, channelMask);
+                return AUDIO_CHANNEL_INVALID;
+        }
+    } else {
+        switch (channelMask) {
+            case AAUDIO_CHANNEL_MONO:
+                return AUDIO_CHANNEL_OUT_MONO;
+            case AAUDIO_CHANNEL_STEREO:
+                return AUDIO_CHANNEL_OUT_STEREO;
+            case AAUDIO_CHANNEL_2POINT1:
+                return AUDIO_CHANNEL_OUT_2POINT1;
+            case AAUDIO_CHANNEL_TRI:
+                return AUDIO_CHANNEL_OUT_TRI;
+            case AAUDIO_CHANNEL_TRI_BACK:
+                return AUDIO_CHANNEL_OUT_TRI_BACK;
+            case AAUDIO_CHANNEL_3POINT1:
+                return AUDIO_CHANNEL_OUT_3POINT1;
+            case AAUDIO_CHANNEL_2POINT0POINT2:
+                return AUDIO_CHANNEL_OUT_2POINT0POINT2;
+            case AAUDIO_CHANNEL_2POINT1POINT2:
+                return AUDIO_CHANNEL_OUT_2POINT1POINT2;
+            case AAUDIO_CHANNEL_3POINT0POINT2:
+                return AUDIO_CHANNEL_OUT_3POINT0POINT2;
+            case AAUDIO_CHANNEL_3POINT1POINT2:
+                return AUDIO_CHANNEL_OUT_3POINT1POINT2;
+            case AAUDIO_CHANNEL_QUAD:
+                return AUDIO_CHANNEL_OUT_QUAD;
+            case AAUDIO_CHANNEL_QUAD_SIDE:
+                return AUDIO_CHANNEL_OUT_QUAD_SIDE;
+            case AAUDIO_CHANNEL_SURROUND:
+                return AUDIO_CHANNEL_OUT_SURROUND;
+            case AAUDIO_CHANNEL_PENTA:
+                return AUDIO_CHANNEL_OUT_PENTA;
+            case AAUDIO_CHANNEL_5POINT1:
+                return AUDIO_CHANNEL_OUT_5POINT1;
+            case AAUDIO_CHANNEL_5POINT1_SIDE:
+                return AUDIO_CHANNEL_OUT_5POINT1_SIDE;
+            case AAUDIO_CHANNEL_5POINT1POINT2:
+                return AUDIO_CHANNEL_OUT_5POINT1POINT2;
+            case AAUDIO_CHANNEL_5POINT1POINT4:
+                return AUDIO_CHANNEL_OUT_5POINT1POINT4;
+            case AAUDIO_CHANNEL_6POINT1:
+                return AUDIO_CHANNEL_OUT_6POINT1;
+            case AAUDIO_CHANNEL_7POINT1:
+                return AUDIO_CHANNEL_OUT_7POINT1;
+            case AAUDIO_CHANNEL_7POINT1POINT2:
+                return AUDIO_CHANNEL_OUT_7POINT1POINT2;
+            case AAUDIO_CHANNEL_7POINT1POINT4:
+                return AUDIO_CHANNEL_OUT_7POINT1POINT4;
+            // TODO: add 9point1point4 and 9point1point6 when they are added in audio-hal-enums.h
+            // case AAUDIO_CHANNEL_9POINT1POINT4:
+            //     return AUDIO_CHANNEL_OUT_9POINT1POINT4;
+            // case AAUDIO_CHANNEL_9POINT1POINT6:
+            //     return AUDIO_CHANNEL_OUT_9POINT1POINT6;
+            default:
+                ALOGE("%s() %#x unrecognized", __func__, channelMask);
+                return AUDIO_CHANNEL_INVALID;
+        }
+    }
+}
+
+aaudio_channel_mask_t AAudioConvert_androidToAAudioChannelLayoutMask(
+        audio_channel_mask_t channelMask, bool isInput) {
+    if (isInput) {
+        switch (channelMask) {
+            case AUDIO_CHANNEL_IN_MONO:
+                return AAUDIO_CHANNEL_MONO;
+            case AUDIO_CHANNEL_IN_STEREO:
+                return AAUDIO_CHANNEL_STEREO;
+            case AUDIO_CHANNEL_IN_FRONT_BACK:
+                return AAUDIO_CHANNEL_FRONT_BACK;
+            case AUDIO_CHANNEL_IN_2POINT0POINT2:
+                return AAUDIO_CHANNEL_2POINT0POINT2;
+            case AUDIO_CHANNEL_IN_2POINT1POINT2:
+                return AAUDIO_CHANNEL_2POINT1POINT2;
+            case AUDIO_CHANNEL_IN_3POINT0POINT2:
+                return AAUDIO_CHANNEL_3POINT0POINT2;
+            case AUDIO_CHANNEL_IN_3POINT1POINT2:
+                return AAUDIO_CHANNEL_3POINT1POINT2;
+            case AUDIO_CHANNEL_IN_5POINT1:
+                return AAUDIO_CHANNEL_5POINT1;
+            default:
+                ALOGE("%s() %#x unrecognized", __func__, channelMask);
+                return AAUDIO_CHANNEL_INVALID;
+        }
+    } else {
+        switch (channelMask) {
+            case AUDIO_CHANNEL_OUT_MONO:
+                return AAUDIO_CHANNEL_MONO;
+            case AUDIO_CHANNEL_OUT_STEREO:
+                return AAUDIO_CHANNEL_STEREO;
+            case AUDIO_CHANNEL_OUT_2POINT1:
+                return AAUDIO_CHANNEL_2POINT1;
+            case AUDIO_CHANNEL_OUT_TRI:
+                return AAUDIO_CHANNEL_TRI;
+            case AUDIO_CHANNEL_OUT_TRI_BACK:
+                return AAUDIO_CHANNEL_TRI_BACK;
+            case AUDIO_CHANNEL_OUT_3POINT1:
+                return AAUDIO_CHANNEL_3POINT1;
+            case AUDIO_CHANNEL_OUT_2POINT0POINT2:
+                return AAUDIO_CHANNEL_2POINT0POINT2;
+            case AUDIO_CHANNEL_OUT_2POINT1POINT2:
+                return AAUDIO_CHANNEL_2POINT1POINT2;
+            case AUDIO_CHANNEL_OUT_3POINT0POINT2:
+                return AAUDIO_CHANNEL_3POINT0POINT2;
+            case AUDIO_CHANNEL_OUT_3POINT1POINT2:
+                return AAUDIO_CHANNEL_3POINT1POINT2;
+            case AUDIO_CHANNEL_OUT_QUAD:
+                return AAUDIO_CHANNEL_QUAD;
+            case AUDIO_CHANNEL_OUT_QUAD_SIDE:
+                return AAUDIO_CHANNEL_QUAD_SIDE;
+            case AUDIO_CHANNEL_OUT_SURROUND:
+                return AAUDIO_CHANNEL_SURROUND;
+            case AUDIO_CHANNEL_OUT_PENTA:
+                return AAUDIO_CHANNEL_PENTA;
+            case AUDIO_CHANNEL_OUT_5POINT1:
+                return AAUDIO_CHANNEL_5POINT1;
+            case AUDIO_CHANNEL_OUT_5POINT1_SIDE:
+                return AAUDIO_CHANNEL_5POINT1_SIDE;
+            case AUDIO_CHANNEL_OUT_5POINT1POINT2:
+                return AAUDIO_CHANNEL_5POINT1POINT2;
+            case AUDIO_CHANNEL_OUT_5POINT1POINT4:
+                return AAUDIO_CHANNEL_5POINT1POINT4;
+            case AUDIO_CHANNEL_OUT_6POINT1:
+                return AAUDIO_CHANNEL_6POINT1;
+            case AUDIO_CHANNEL_OUT_7POINT1:
+                return AAUDIO_CHANNEL_7POINT1;
+            case AUDIO_CHANNEL_OUT_7POINT1POINT2:
+                return AAUDIO_CHANNEL_7POINT1POINT2;
+            case AUDIO_CHANNEL_OUT_7POINT1POINT4:
+                return AAUDIO_CHANNEL_7POINT1POINT4;
+            // TODO: add 9point1point4 and 9point1point6 when they are added in audio-hal-enums.h
+            // case AUDIO_CHANNEL_OUT_9POINT1POINT4:
+            //     return AAUDIO_CHANNEL_9POINT1POINT4;
+            // case AUDIO_CHANNEL_OUT_9POINT1POINT6:
+            //     return AAUDIO_CHANNEL_9POINT1POINT6;
+            default:
+                ALOGE("%s() %#x unrecognized", __func__, channelMask);
+                return AAUDIO_CHANNEL_INVALID;
+        }
+    }
+}
+
+int32_t AAudioConvert_channelMaskToCount(aaudio_channel_mask_t channelMask) {
+    return __builtin_popcount(channelMask & ~AAUDIO_CHANNEL_BIT_INDEX);
+}
+
+aaudio_channel_mask_t AAudioConvert_channelCountToMask(int32_t channelCount) {
+    if (channelCount < 0 || channelCount > AUDIO_CHANNEL_COUNT_MAX) {
+        return AAUDIO_CHANNEL_INVALID;
+    }
+
+    if (channelCount == 0) {
+        return AAUDIO_UNSPECIFIED;
+    }
+
+    // Return index mask if the channel count is greater than 2.
+    return AAUDIO_CHANNEL_BIT_INDEX | ((1 << channelCount) - 1);
+}
+
+aaudio_channel_mask_t AAudioConvert_androidToAAudioChannelIndexMask(
+        audio_channel_mask_t channelMask) {
+    if (audio_channel_mask_get_representation(channelMask) != AUDIO_CHANNEL_REPRESENTATION_INDEX) {
+        ALOGE("%s() %#x not an index mask", __func__, channelMask);
+        return AAUDIO_CHANNEL_INVALID;
+    }
+    return (channelMask & ~AUDIO_CHANNEL_INDEX_HDR) | AAUDIO_CHANNEL_BIT_INDEX;
+}
+
+audio_channel_mask_t AAudioConvert_aaudioToAndroidChannelIndexMask(
+        aaudio_channel_mask_t channelMask) {
+    if (!AAudio_isChannelIndexMask(channelMask)) {
+        ALOGE("%s() %#x not an index mask", __func__, channelMask);
+        return AUDIO_CHANNEL_INVALID;
+    }
+    return audio_channel_mask_for_index_assignment_from_count(
+            AAudioConvert_channelMaskToCount(channelMask));
+}
+
+aaudio_channel_mask_t AAudioConvert_androidToAAudioChannelMask(
+        audio_channel_mask_t channelMask, bool isInput, bool indexMaskRequired) {
+    if (audio_channel_mask_get_representation(channelMask) == AUDIO_CHANNEL_REPRESENTATION_INDEX) {
+        return AAudioConvert_androidToAAudioChannelIndexMask(channelMask);
+    }
+    if (indexMaskRequired) {
+        // Require index mask, `channelMask` here is a position mask.
+        const int channelCount = isInput ? audio_channel_count_from_in_mask(channelMask)
+                                         : audio_channel_count_from_out_mask(channelMask);
+        return AAudioConvert_channelCountToMask(channelCount);
+    }
+    return AAudioConvert_androidToAAudioChannelLayoutMask(channelMask, isInput);
+}
+
+audio_channel_mask_t AAudioConvert_aaudioToAndroidChannelMask(
+        aaudio_channel_mask_t channelMask, bool isInput) {
+    return AAudio_isChannelIndexMask(channelMask)
+            ? AAudioConvert_aaudioToAndroidChannelIndexMask(channelMask)
+            : AAudioConvert_aaudioToAndroidChannelLayoutMask(channelMask, isInput);
+}
+
+bool AAudio_isChannelIndexMask(aaudio_channel_mask_t channelMask) {
+    return (channelMask & AAUDIO_CHANNEL_BIT_INDEX) == AAUDIO_CHANNEL_BIT_INDEX;
+}
+
+audio_channel_mask_t AAudio_getChannelMaskForOpen(
+        aaudio_channel_mask_t channelMask, int32_t samplesPerFrame, bool isInput) {
+    if (channelMask != AAUDIO_UNSPECIFIED) {
+        if (AAudio_isChannelIndexMask(channelMask) && samplesPerFrame <= 2) {
+            // When it is index mask and the count is less than 3, use position mask
+            // instead of index mask for opening a stream. This may need to be revisited
+            // when making channel index mask public.
+            return isInput ? audio_channel_in_mask_from_count(samplesPerFrame)
+                           : audio_channel_out_mask_from_count(samplesPerFrame);
+        }
+        return AAudioConvert_aaudioToAndroidChannelMask(channelMask, isInput);
+    }
+
+    // Return stereo when unspecified.
+    return isInput ? AUDIO_CHANNEL_IN_STEREO : AUDIO_CHANNEL_OUT_STEREO;
+}
+
 int32_t AAudioConvert_framesToBytes(int32_t numFrames,
                                     int32_t bytesPerFrame,
                                     int32_t *sizeInBytes) {
diff --git a/media/libaaudio/src/utility/AAudioUtilities.h b/media/libaaudio/src/utility/AAudioUtilities.h
index 82eb77d..5eda30c 100644
--- a/media/libaaudio/src/utility/AAudioUtilities.h
+++ b/media/libaaudio/src/utility/AAudioUtilities.h
@@ -96,6 +96,33 @@
 audio_flags_mask_t AAudioConvert_privacySensitiveToAudioFlagsMask(
         bool privacySensitive);
 
+audio_channel_mask_t AAudioConvert_aaudioToAndroidChannelLayoutMask(
+        aaudio_channel_mask_t channelMask, bool isInput);
+
+aaudio_channel_mask_t AAudioConvert_androidToAAudioChannelLayoutMask(
+        audio_channel_mask_t channelMask, bool isInput);
+
+aaudio_channel_mask_t AAudioConvert_androidToAAudioChannelIndexMask(
+        audio_channel_mask_t channelMask);
+
+audio_channel_mask_t AAudioConvert_aaudioToAndroidChannelIndexMask(
+        aaudio_channel_mask_t channelMask);
+
+aaudio_channel_mask_t AAudioConvert_androidToAAudioChannelMask(
+        audio_channel_mask_t channelMask, bool isInput, bool indexMaskRequired);
+
+audio_channel_mask_t AAudioConvert_aaudioToAndroidChannelMask(
+        aaudio_channel_mask_t channelMask, bool isInput);
+
+bool AAudio_isChannelIndexMask(aaudio_channel_mask_t channelMask);
+
+int32_t AAudioConvert_channelMaskToCount(aaudio_channel_mask_t channelMask);
+
+aaudio_channel_mask_t AAudioConvert_channelCountToMask(int32_t channelCount);
+
+audio_channel_mask_t AAudio_getChannelMaskForOpen(
+        aaudio_channel_mask_t channelMask, int32_t samplesPerFrame, bool isInput);
+
 // Note that this code may be replaced by Settings or by some other system configuration tool.
 
 /**
@@ -318,4 +345,36 @@
     std::atomic<int> mRequested{0};
     std::atomic<int> mAcknowledged{0};
 };
+
+enum {
+    /**
+     * Audio channel index mask, only used internally.
+     */
+    AAUDIO_CHANNEL_BIT_INDEX = 0x80000000,
+    AAUDIO_CHANNEL_INDEX_MASK_1 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 1) - 1,
+    AAUDIO_CHANNEL_INDEX_MASK_2 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 2) - 1,
+    AAUDIO_CHANNEL_INDEX_MASK_3 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 3) - 1,
+    AAUDIO_CHANNEL_INDEX_MASK_4 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 4) - 1,
+    AAUDIO_CHANNEL_INDEX_MASK_5 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 5) - 1,
+    AAUDIO_CHANNEL_INDEX_MASK_6 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 6) - 1,
+    AAUDIO_CHANNEL_INDEX_MASK_7 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 7) - 1,
+    AAUDIO_CHANNEL_INDEX_MASK_8 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 8) - 1,
+    AAUDIO_CHANNEL_INDEX_MASK_9 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 9) - 1,
+    AAUDIO_CHANNEL_INDEX_MASK_10 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 10) - 1,
+    AAUDIO_CHANNEL_INDEX_MASK_11 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 11) - 1,
+    AAUDIO_CHANNEL_INDEX_MASK_12 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 12) - 1,
+    AAUDIO_CHANNEL_INDEX_MASK_13 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 13) - 1,
+    AAUDIO_CHANNEL_INDEX_MASK_14 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 14) - 1,
+    AAUDIO_CHANNEL_INDEX_MASK_15 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 15) - 1,
+    AAUDIO_CHANNEL_INDEX_MASK_16 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 16) - 1,
+    AAUDIO_CHANNEL_INDEX_MASK_17 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 17) - 1,
+    AAUDIO_CHANNEL_INDEX_MASK_18 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 18) - 1,
+    AAUDIO_CHANNEL_INDEX_MASK_19 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 19) - 1,
+    AAUDIO_CHANNEL_INDEX_MASK_20 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 20) - 1,
+    AAUDIO_CHANNEL_INDEX_MASK_21 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 21) - 1,
+    AAUDIO_CHANNEL_INDEX_MASK_22 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 22) - 1,
+    AAUDIO_CHANNEL_INDEX_MASK_23 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 23) - 1,
+    AAUDIO_CHANNEL_INDEX_MASK_24 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 24) - 1,
+};
+
 #endif //UTILITY_AAUDIO_UTILITIES_H
diff --git a/media/libaudioclient/AudioSystem.cpp b/media/libaudioclient/AudioSystem.cpp
index f1eeaa3..792be31 100644
--- a/media/libaudioclient/AudioSystem.cpp
+++ b/media/libaudioclient/AudioSystem.cpp
@@ -480,6 +480,12 @@
     return af->systemReady();
 }
 
+status_t AudioSystem::audioPolicyReady() {
+    const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
+    if (af == 0) return NO_INIT;
+    return af->audioPolicyReady();
+}
+
 status_t AudioSystem::getFrameCountHAL(audio_io_handle_t ioHandle,
                                        size_t* frameCount) {
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
diff --git a/media/libaudioclient/IAudioFlinger.cpp b/media/libaudioclient/IAudioFlinger.cpp
index cae81f0..2af1c50 100644
--- a/media/libaudioclient/IAudioFlinger.cpp
+++ b/media/libaudioclient/IAudioFlinger.cpp
@@ -715,6 +715,10 @@
     return statusTFromBinderStatus(mDelegate->systemReady());
 }
 
+status_t AudioFlingerClientAdapter::audioPolicyReady() {
+    return statusTFromBinderStatus(mDelegate->audioPolicyReady());
+}
+
 size_t AudioFlingerClientAdapter::frameCountHAL(audio_io_handle_t ioHandle) const {
     auto result = [&]() -> ConversionResult<size_t> {
         int32_t ioHandleAidl = VALUE_OR_RETURN(legacy2aidl_audio_io_handle_t_int32_t(ioHandle));
@@ -1189,6 +1193,11 @@
     return Status::fromStatusT(mDelegate->systemReady());
 }
 
+Status AudioFlingerServerAdapter::audioPolicyReady() {
+    mDelegate->audioPolicyReady();
+    return Status::ok();
+}
+
 Status AudioFlingerServerAdapter::frameCountHAL(int32_t ioHandle, int64_t* _aidl_return) {
     audio_io_handle_t ioHandleLegacy = VALUE_OR_RETURN_BINDER(
             aidl2legacy_int32_t_audio_io_handle_t(ioHandle));
diff --git a/media/libaudioclient/aidl/android/media/AudioVibratorInfo.aidl b/media/libaudioclient/aidl/android/media/AudioVibratorInfo.aidl
index f88fc3c..8538d8a 100644
--- a/media/libaudioclient/aidl/android/media/AudioVibratorInfo.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioVibratorInfo.aidl
@@ -24,4 +24,5 @@
     int id;
     float resonantFrequency;
     float qFactor;
+    float maxAmplitude;
 }
diff --git a/media/libaudioclient/aidl/android/media/IAudioFlingerService.aidl b/media/libaudioclient/aidl/android/media/IAudioFlingerService.aidl
index d2cae6d..7ffcc33 100644
--- a/media/libaudioclient/aidl/android/media/IAudioFlingerService.aidl
+++ b/media/libaudioclient/aidl/android/media/IAudioFlingerService.aidl
@@ -197,6 +197,9 @@
     /* Indicate JAVA services are ready (scheduling, power management ...) */
     oneway void systemReady();
 
+    /* Indicate audio policy service is ready */
+    oneway void audioPolicyReady();
+
     // Returns the number of frames per audio HAL buffer.
     long frameCountHAL(int /* audio_io_handle_t */ ioHandle);
 
diff --git a/media/libaudioclient/aidl/android/media/OpenOutputRequest.aidl b/media/libaudioclient/aidl/android/media/OpenOutputRequest.aidl
index 06b12e9..1541948 100644
--- a/media/libaudioclient/aidl/android/media/OpenOutputRequest.aidl
+++ b/media/libaudioclient/aidl/android/media/OpenOutputRequest.aidl
@@ -17,6 +17,7 @@
 package android.media;
 
 import android.media.AudioConfig;
+import android.media.AudioConfigBase;
 import android.media.AudioPort;
 
 /**
@@ -25,7 +26,8 @@
 parcelable OpenOutputRequest {
     /** Interpreted as audio_module_handle_t. */
     int module;
-    AudioConfig config;
+    AudioConfig halConfig;
+    AudioConfigBase mixerConfig;
     /** Type must be DEVICE. */
     AudioPort device;
     /** Bitmask, indexed by AudioOutputFlag. */
diff --git a/media/libaudioclient/fuzzer/audioflinger_fuzzer.cpp b/media/libaudioclient/fuzzer/audioflinger_fuzzer.cpp
index d03c6fa..12473fc 100644
--- a/media/libaudioclient/fuzzer/audioflinger_fuzzer.cpp
+++ b/media/libaudioclient/fuzzer/audioflinger_fuzzer.cpp
@@ -648,11 +648,15 @@
     sp<DeviceDescriptorBase> device = new DeviceDescriptorBase(getValue(&mFdp, kDevices));
     audio_output_flags_t flags = getValue(&mFdp, kOutputFlags);
 
+    audio_config_base_t mixerConfig = AUDIO_CONFIG_BASE_INITIALIZER;
+
     media::OpenOutputRequest request{};
     media::OpenOutputResponse response{};
 
     request.module = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_module_handle_t_int32_t(module));
-    request.config = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_config_t_AudioConfig(config));
+    request.halConfig = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_config_t_AudioConfig(config));
+    request.mixerConfig =
+            VALUE_OR_RETURN_STATUS(legacy2aidl_audio_config_base_t_AudioConfigBase(mixerConfig));
     request.device = VALUE_OR_RETURN_STATUS(legacy2aidl_DeviceDescriptorBase(device));
     request.flags = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_output_flags_t_int32_t_mask(flags));
 
diff --git a/media/libaudioclient/include/media/AudioCommonTypes.h b/media/libaudioclient/include/media/AudioCommonTypes.h
index 5dfe5fc..5f0c590 100644
--- a/media/libaudioclient/include/media/AudioCommonTypes.h
+++ b/media/libaudioclient/include/media/AudioCommonTypes.h
@@ -41,6 +41,42 @@
     return !(lhs==rhs);
 }
 
+constexpr bool operator==(const audio_offload_info_t &lhs, const audio_offload_info_t &rhs)
+{
+    return lhs.version == rhs.version && lhs.size == rhs.size &&
+           lhs.sample_rate == rhs.sample_rate && lhs.channel_mask == rhs.channel_mask &&
+           lhs.format == rhs.format && lhs.stream_type == rhs.stream_type &&
+           lhs.bit_rate == rhs.bit_rate && lhs.duration_us == rhs.duration_us &&
+           lhs.has_video == rhs.has_video && lhs.is_streaming == rhs.is_streaming &&
+           lhs.bit_width == rhs.bit_width && lhs.offload_buffer_size == rhs.offload_buffer_size &&
+           lhs.usage == rhs.usage && lhs.encapsulation_mode == rhs.encapsulation_mode &&
+           lhs.content_id == rhs.content_id && lhs.sync_id == rhs.sync_id;
+}
+constexpr bool operator!=(const audio_offload_info_t &lhs, const audio_offload_info_t &rhs)
+{
+    return !(lhs==rhs);
+}
+
+constexpr bool operator==(const audio_config_t &lhs, const audio_config_t &rhs)
+{
+    return lhs.sample_rate == rhs.sample_rate && lhs.channel_mask == rhs.channel_mask &&
+           lhs.format == rhs.format && lhs.offload_info == rhs.offload_info;
+}
+constexpr bool operator!=(const audio_config_t &lhs, const audio_config_t &rhs)
+{
+    return !(lhs==rhs);
+}
+
+constexpr bool operator==(const audio_config_base_t &lhs, const audio_config_base_t &rhs)
+{
+    return lhs.sample_rate == rhs.sample_rate && lhs.channel_mask == rhs.channel_mask &&
+           lhs.format == rhs.format;
+}
+constexpr bool operator!=(const audio_config_base_t &lhs, const audio_config_base_t &rhs)
+{
+    return !(lhs==rhs);
+}
+
 enum volume_group_t : uint32_t;
 static const volume_group_t VOLUME_GROUP_NONE = static_cast<volume_group_t>(-1);
 
diff --git a/media/libaudioclient/include/media/AudioRecord.h b/media/libaudioclient/include/media/AudioRecord.h
index 326919a..f17ee3a 100644
--- a/media/libaudioclient/include/media/AudioRecord.h
+++ b/media/libaudioclient/include/media/AudioRecord.h
@@ -264,6 +264,7 @@
             size_t      frameCount() const  { return mFrameCount; }
             size_t      frameSize() const   { return mFrameSize; }
             audio_source_t inputSource() const  { return mAttributes.source; }
+            audio_channel_mask_t channelMask() const { return mChannelMask; }
 
     /*
      * Return the period of the notification callback in frames.
diff --git a/media/libaudioclient/include/media/AudioSystem.h b/media/libaudioclient/include/media/AudioSystem.h
index 8ba23ad..9187de1 100644
--- a/media/libaudioclient/include/media/AudioSystem.h
+++ b/media/libaudioclient/include/media/AudioSystem.h
@@ -225,6 +225,9 @@
     // Indicate JAVA services are ready (scheduling, power management ...)
     static status_t systemReady();
 
+    // Indicate audio policy service is ready
+    static status_t audioPolicyReady();
+
     // Returns the number of frames per audio HAL buffer.
     // Corresponds to audio_stream->get_buffer_size()/audio_stream_in_frame_size() for input.
     // See also getFrameCount().
diff --git a/media/libaudioclient/include/media/AudioTrack.h b/media/libaudioclient/include/media/AudioTrack.h
index cb00990..5fea637 100644
--- a/media/libaudioclient/include/media/AudioTrack.h
+++ b/media/libaudioclient/include/media/AudioTrack.h
@@ -401,6 +401,7 @@
 
             uint32_t    channelCount() const { return mChannelCount; }
             size_t      frameCount() const  { return mFrameCount; }
+            audio_channel_mask_t channelMask() const { return mChannelMask; }
 
     /*
      * Return the period of the notification callback in frames.
diff --git a/media/libaudioclient/include/media/IAudioFlinger.h b/media/libaudioclient/include/media/IAudioFlinger.h
index 0e059f7..9e5019e 100644
--- a/media/libaudioclient/include/media/IAudioFlinger.h
+++ b/media/libaudioclient/include/media/IAudioFlinger.h
@@ -329,6 +329,9 @@
     /* Indicate JAVA services are ready (scheduling, power management ...) */
     virtual status_t systemReady() = 0;
 
+    // Indicate audio policy service is ready
+    virtual status_t audioPolicyReady() = 0;
+
     // Returns the number of frames per audio HAL buffer.
     virtual size_t frameCountHAL(audio_io_handle_t ioHandle) const = 0;
 
@@ -432,6 +435,8 @@
     status_t setAudioPortConfig(const struct audio_port_config* config) override;
     audio_hw_sync_t getAudioHwSyncForSession(audio_session_t sessionId) override;
     status_t systemReady() override;
+    status_t audioPolicyReady() override;
+
     size_t frameCountHAL(audio_io_handle_t ioHandle) const override;
     status_t getMicrophones(std::vector<media::MicrophoneInfo>* microphones) override;
     status_t setAudioHalPids(const std::vector<pid_t>& pids) override;
@@ -514,6 +519,7 @@
             SET_AUDIO_PORT_CONFIG = media::BnAudioFlingerService::TRANSACTION_setAudioPortConfig,
             GET_AUDIO_HW_SYNC_FOR_SESSION = media::BnAudioFlingerService::TRANSACTION_getAudioHwSyncForSession,
             SYSTEM_READY = media::BnAudioFlingerService::TRANSACTION_systemReady,
+            AUDIO_POLICY_READY = media::BnAudioFlingerService::TRANSACTION_audioPolicyReady,
             FRAME_COUNT_HAL = media::BnAudioFlingerService::TRANSACTION_frameCountHAL,
             GET_MICROPHONES = media::BnAudioFlingerService::TRANSACTION_getMicrophones,
             SET_MASTER_BALANCE = media::BnAudioFlingerService::TRANSACTION_setMasterBalance,
@@ -624,6 +630,7 @@
     Status setAudioPortConfig(const media::AudioPortConfig& config) override;
     Status getAudioHwSyncForSession(int32_t sessionId, int32_t* _aidl_return) override;
     Status systemReady() override;
+    Status audioPolicyReady() override;
     Status frameCountHAL(int32_t ioHandle, int64_t* _aidl_return) override;
     Status getMicrophones(std::vector<media::MicrophoneInfoData>* _aidl_return) override;
     Status setAudioHalPids(const std::vector<int32_t>& pids) override;
diff --git a/media/libaudiohal/impl/ConversionHelperHidl.cpp b/media/libaudiohal/impl/ConversionHelperHidl.cpp
index 32eaa31..0503698 100644
--- a/media/libaudiohal/impl/ConversionHelperHidl.cpp
+++ b/media/libaudiohal/impl/ConversionHelperHidl.cpp
@@ -105,6 +105,15 @@
 }
 
 // static
+void ConversionHelperHidl::argsFromHal(
+        const Vector<String16>& args, hidl_vec<hidl_string> *hidlArgs) {
+    hidlArgs->resize(args.size());
+    for (size_t i = 0; i < args.size(); ++i) {
+        (*hidlArgs)[i] = String8(args[i]).c_str();
+    }
+}
+
+// static
 status_t ConversionHelperHidl::analyzeResult(const Result& result) {
     switch (result) {
         case Result::OK: return OK;
diff --git a/media/libaudiohal/impl/ConversionHelperHidl.h b/media/libaudiohal/impl/ConversionHelperHidl.h
index 59122c7..2909013 100644
--- a/media/libaudiohal/impl/ConversionHelperHidl.h
+++ b/media/libaudiohal/impl/ConversionHelperHidl.h
@@ -21,6 +21,8 @@
 #include <hidl/HidlSupport.h>
 #include <system/audio.h>
 #include <utils/String8.h>
+#include <utils/String16.h>
+#include <utils/Vector.h>
 
 using ::android::hardware::audio::CPP_VERSION::ParameterValue;
 using CoreResult = ::android::hardware::audio::CPP_VERSION::Result;
@@ -37,6 +39,7 @@
     static status_t keysFromHal(const String8& keys, hidl_vec<hidl_string> *hidlKeys);
     static status_t parametersFromHal(const String8& kvPairs, hidl_vec<ParameterValue> *hidlParams);
     static void parametersToHal(const hidl_vec<ParameterValue>& parameters, String8 *values);
+    static void argsFromHal(const Vector<String16>& args, hidl_vec<hidl_string> *hidlArgs);
 
     ConversionHelperHidl(const char* className);
 
diff --git a/media/libaudiohal/impl/DeviceHalHidl.cpp b/media/libaudiohal/impl/DeviceHalHidl.cpp
index ca4f663..f86df1e 100644
--- a/media/libaudiohal/impl/DeviceHalHidl.cpp
+++ b/media/libaudiohal/impl/DeviceHalHidl.cpp
@@ -457,11 +457,13 @@
 }
 #endif
 
-status_t DeviceHalHidl::dump(int fd) {
+status_t DeviceHalHidl::dump(int fd, const Vector<String16>& args) {
     if (mDevice == 0) return NO_INIT;
     native_handle_t* hidlHandle = native_handle_create(1, 0);
     hidlHandle->data[0] = fd;
-    Return<void> ret = mDevice->debug(hidlHandle, {} /* options */);
+    hidl_vec<hidl_string> hidlArgs;
+    argsFromHal(args, &hidlArgs);
+    Return<void> ret = mDevice->debug(hidlHandle, hidlArgs);
     native_handle_delete(hidlHandle);
     return processReturn("dump", ret);
 }
diff --git a/media/libaudiohal/impl/DeviceHalHidl.h b/media/libaudiohal/impl/DeviceHalHidl.h
index 2c847cf..2694ab3 100644
--- a/media/libaudiohal/impl/DeviceHalHidl.h
+++ b/media/libaudiohal/impl/DeviceHalHidl.h
@@ -119,7 +119,7 @@
     status_t addDeviceEffect(audio_port_handle_t device, sp<EffectHalInterface> effect) override;
     status_t removeDeviceEffect(audio_port_handle_t device, sp<EffectHalInterface> effect) override;
 
-    virtual status_t dump(int fd);
+    status_t dump(int fd, const Vector<String16>& args) override;
 
   private:
     friend class DevicesFactoryHalHidl;
diff --git a/media/libaudiohal/impl/DeviceHalLocal.cpp b/media/libaudiohal/impl/DeviceHalLocal.cpp
index af7dc1a..e0304af 100644
--- a/media/libaudiohal/impl/DeviceHalLocal.cpp
+++ b/media/libaudiohal/impl/DeviceHalLocal.cpp
@@ -233,7 +233,7 @@
     return INVALID_OPERATION;
 }
 
-status_t DeviceHalLocal::dump(int fd) {
+status_t DeviceHalLocal::dump(int fd, const Vector<String16>& /* args */) {
     return mDev->dump(mDev, fd);
 }
 
diff --git a/media/libaudiohal/impl/DeviceHalLocal.h b/media/libaudiohal/impl/DeviceHalLocal.h
index 46b510b..2fde936 100644
--- a/media/libaudiohal/impl/DeviceHalLocal.h
+++ b/media/libaudiohal/impl/DeviceHalLocal.h
@@ -112,7 +112,7 @@
     status_t addDeviceEffect(audio_port_handle_t device, sp<EffectHalInterface> effect) override;
     status_t removeDeviceEffect(audio_port_handle_t device, sp<EffectHalInterface> effect) override;
 
-    virtual status_t dump(int fd);
+    status_t dump(int fd, const Vector<String16>& args) override;
 
     void closeOutputStream(struct audio_stream_out *stream_out);
     void closeInputStream(struct audio_stream_in *stream_in);
diff --git a/media/libaudiohal/impl/EffectsFactoryHalHidl.cpp b/media/libaudiohal/impl/EffectsFactoryHalHidl.cpp
index 9c4363c..f75bbf3 100644
--- a/media/libaudiohal/impl/EffectsFactoryHalHidl.cpp
+++ b/media/libaudiohal/impl/EffectsFactoryHalHidl.cpp
@@ -73,7 +73,9 @@
         uint32_t index, effect_descriptor_t *pDescriptor) {
     // TODO: We need somehow to track the changes on the server side
     // or figure out how to convert everybody to query all the descriptors at once.
-    // TODO: check for nullptr
+    if (pDescriptor == nullptr) {
+        return BAD_VALUE;
+    }
     if (mLastDescriptors.size() == 0) {
         status_t queryResult = queryAllDescriptors();
         if (queryResult != OK) return queryResult;
@@ -85,7 +87,9 @@
 
 status_t EffectsFactoryHalHidl::getDescriptor(
         const effect_uuid_t *pEffectUuid, effect_descriptor_t *pDescriptor) {
-    // TODO: check for nullptr
+    if (pDescriptor == nullptr || pEffectUuid == nullptr) {
+        return BAD_VALUE;
+    }
     if (mEffectsFactory == 0) return NO_INIT;
     Uuid hidlUuid;
     UuidUtils::uuidFromHal(*pEffectUuid, &hidlUuid);
@@ -105,6 +109,33 @@
     return processReturn(__FUNCTION__, ret);
 }
 
+status_t EffectsFactoryHalHidl::getDescriptors(const effect_uuid_t *pEffectType,
+                                               std::vector<effect_descriptor_t> *descriptors) {
+    if (pEffectType == nullptr || descriptors == nullptr) {
+        return BAD_VALUE;
+    }
+
+    uint32_t numEffects = 0;
+    status_t status = queryNumberEffects(&numEffects);
+    if (status != NO_ERROR) {
+        ALOGW("%s error %d from FactoryHal queryNumberEffects", __func__, status);
+        return status;
+    }
+
+    for (uint32_t i = 0; i < numEffects; i++) {
+        effect_descriptor_t descriptor;
+        status = getDescriptor(i, &descriptor);
+        if (status != NO_ERROR) {
+            ALOGW("%s error %d from FactoryHal getDescriptor", __func__, status);
+            continue;
+        }
+        if (memcmp(&descriptor.type, pEffectType, sizeof(effect_uuid_t)) == 0) {
+            descriptors->push_back(descriptor);
+        }
+    }
+    return descriptors->empty() ? NAME_NOT_FOUND : NO_ERROR;
+}
+
 status_t EffectsFactoryHalHidl::createEffect(
         const effect_uuid_t *pEffectUuid, int32_t sessionId, int32_t ioId,
         int32_t deviceId __unused, sp<EffectHalInterface> *effect) {
diff --git a/media/libaudiohal/impl/EffectsFactoryHalHidl.h b/media/libaudiohal/impl/EffectsFactoryHalHidl.h
index 5fa85e7..ff26d9f 100644
--- a/media/libaudiohal/impl/EffectsFactoryHalHidl.h
+++ b/media/libaudiohal/impl/EffectsFactoryHalHidl.h
@@ -45,6 +45,9 @@
     virtual status_t getDescriptor(const effect_uuid_t *pEffectUuid,
             effect_descriptor_t *pDescriptor);
 
+    virtual status_t getDescriptors(const effect_uuid_t *pEffectType,
+                                    std::vector<effect_descriptor_t> *descriptors);
+
     // Creates an effect engine of the specified type.
     // To release the effect engine, it is necessary to release references
     // to the returned effect object.
diff --git a/media/libaudiohal/impl/StreamHalHidl.cpp b/media/libaudiohal/impl/StreamHalHidl.cpp
index 539a149..452f84d 100644
--- a/media/libaudiohal/impl/StreamHalHidl.cpp
+++ b/media/libaudiohal/impl/StreamHalHidl.cpp
@@ -152,11 +152,13 @@
     return processReturn("standby", mStream->standby());
 }
 
-status_t StreamHalHidl::dump(int fd) {
+status_t StreamHalHidl::dump(int fd, const Vector<String16>& args) {
     if (!mStream) return NO_INIT;
     native_handle_t* hidlHandle = native_handle_create(1, 0);
     hidlHandle->data[0] = fd;
-    Return<void> ret = mStream->debug(hidlHandle, {} /* options */);
+    hidl_vec<hidl_string> hidlArgs;
+    argsFromHal(args, &hidlArgs);
+    Return<void> ret = mStream->debug(hidlHandle, hidlArgs);
     native_handle_delete(hidlHandle);
     mStreamPowerLog.dump(fd);
     return processReturn("dump", ret);
diff --git a/media/libaudiohal/impl/StreamHalHidl.h b/media/libaudiohal/impl/StreamHalHidl.h
index 970903b..6f5dd04 100644
--- a/media/libaudiohal/impl/StreamHalHidl.h
+++ b/media/libaudiohal/impl/StreamHalHidl.h
@@ -71,7 +71,7 @@
     // Put the audio hardware input/output into standby mode.
     virtual status_t standby();
 
-    virtual status_t dump(int fd);
+    virtual status_t dump(int fd, const Vector<String16>& args) override;
 
     // Start a stream operating in mmap mode.
     virtual status_t start();
diff --git a/media/libaudiohal/impl/StreamHalLocal.cpp b/media/libaudiohal/impl/StreamHalLocal.cpp
index 34bd5df..11fac61 100644
--- a/media/libaudiohal/impl/StreamHalLocal.cpp
+++ b/media/libaudiohal/impl/StreamHalLocal.cpp
@@ -87,7 +87,8 @@
     return mStream->standby(mStream);
 }
 
-status_t StreamHalLocal::dump(int fd) {
+status_t StreamHalLocal::dump(int fd, const Vector<String16>& args) {
+    (void) args;
     status_t status = mStream->dump(mStream, fd);
     mStreamPowerLog.dump(fd);
     return status;
diff --git a/media/libaudiohal/impl/StreamHalLocal.h b/media/libaudiohal/impl/StreamHalLocal.h
index b260495..493c521 100644
--- a/media/libaudiohal/impl/StreamHalLocal.h
+++ b/media/libaudiohal/impl/StreamHalLocal.h
@@ -50,7 +50,7 @@
     // Put the audio hardware input/output into standby mode.
     virtual status_t standby();
 
-    virtual status_t dump(int fd);
+    virtual status_t dump(int fd, const Vector<String16>& args) override;
 
     // Start a stream operating in mmap mode.
     virtual status_t start() = 0;
diff --git a/media/libaudiohal/include/media/audiohal/DeviceHalInterface.h b/media/libaudiohal/include/media/audiohal/DeviceHalInterface.h
index 29ef011..69cbcec 100644
--- a/media/libaudiohal/include/media/audiohal/DeviceHalInterface.h
+++ b/media/libaudiohal/include/media/audiohal/DeviceHalInterface.h
@@ -120,7 +120,7 @@
     virtual status_t removeDeviceEffect(
             audio_port_handle_t device, sp<EffectHalInterface> effect) = 0;
 
-    virtual status_t dump(int fd) = 0;
+    virtual status_t dump(int fd, const Vector<String16>& args) = 0;
 
   protected:
     // Subclasses can not be constructed directly by clients.
diff --git a/media/libaudiohal/include/media/audiohal/EffectsFactoryHalInterface.h b/media/libaudiohal/include/media/audiohal/EffectsFactoryHalInterface.h
index 9fb56ae..3e505bd 100644
--- a/media/libaudiohal/include/media/audiohal/EffectsFactoryHalInterface.h
+++ b/media/libaudiohal/include/media/audiohal/EffectsFactoryHalInterface.h
@@ -37,6 +37,9 @@
     virtual status_t getDescriptor(const effect_uuid_t *pEffectUuid,
             effect_descriptor_t *pDescriptor) = 0;
 
+    virtual status_t getDescriptors(const effect_uuid_t *pEffectType,
+                                    std::vector<effect_descriptor_t> *descriptors) = 0;
+
     // Creates an effect engine of the specified type.
     // To release the effect engine, it is necessary to release references
     // to the returned effect object.
diff --git a/media/libaudiohal/include/media/audiohal/StreamHalInterface.h b/media/libaudiohal/include/media/audiohal/StreamHalInterface.h
index 2be12fb..2b5b2db 100644
--- a/media/libaudiohal/include/media/audiohal/StreamHalInterface.h
+++ b/media/libaudiohal/include/media/audiohal/StreamHalInterface.h
@@ -25,6 +25,7 @@
 #include <utils/Errors.h>
 #include <utils/RefBase.h>
 #include <utils/String8.h>
+#include <utils/Vector.h>
 
 namespace android {
 
@@ -69,7 +70,7 @@
     // Put the audio hardware input/output into standby mode.
     virtual status_t standby() = 0;
 
-    virtual status_t dump(int fd) = 0;
+    virtual status_t dump(int fd, const Vector<String16>& args = {}) = 0;
 
     // Start a stream operating in mmap mode.
     virtual status_t start() = 0;
diff --git a/media/libaudioprocessing/AudioMixer.cpp b/media/libaudioprocessing/AudioMixer.cpp
index d85e2e9..e68c002 100644
--- a/media/libaudioprocessing/AudioMixer.cpp
+++ b/media/libaudioprocessing/AudioMixer.cpp
@@ -434,6 +434,12 @@
                 track->mHapticIntensity = hapticIntensity;
             }
             } break;
+        case HAPTIC_MAX_AMPLITUDE: {
+            const float hapticMaxAmplitude = *reinterpret_cast<float*>(value);
+            if (track->mHapticMaxAmplitude != hapticMaxAmplitude) {
+                track->mHapticMaxAmplitude = hapticMaxAmplitude;
+            }
+            } break;
         default:
             LOG_ALWAYS_FATAL("setParameter track: bad param %d", param);
         }
@@ -553,6 +559,7 @@
     // haptic
     t->mHapticPlaybackEnabled = false;
     t->mHapticIntensity = os::HapticScale::NONE;
+    t->mHapticMaxAmplitude = NAN;
     t->mMixerHapticChannelMask = AUDIO_CHANNEL_NONE;
     t->mMixerHapticChannelCount = 0;
     t->mAdjustInChannelCount = t->channelCount + t->mHapticChannelCount;
@@ -602,7 +609,8 @@
                 switch (t->mMixerFormat) {
                 // Mixer format should be AUDIO_FORMAT_PCM_FLOAT.
                 case AUDIO_FORMAT_PCM_FLOAT: {
-                    os::scaleHapticData((float*) buffer, sampleCount, t->mHapticIntensity);
+                    os::scaleHapticData((float*) buffer, sampleCount, t->mHapticIntensity,
+                                        t->mHapticMaxAmplitude);
                 } break;
                 default:
                     LOG_ALWAYS_FATAL("bad mMixerFormat: %#x", t->mMixerFormat);
diff --git a/media/libaudioprocessing/AudioMixerOps.h b/media/libaudioprocessing/AudioMixerOps.h
index cd47dc6..2988c67 100644
--- a/media/libaudioprocessing/AudioMixerOps.h
+++ b/media/libaudioprocessing/AudioMixerOps.h
@@ -17,6 +17,8 @@
 #ifndef ANDROID_AUDIO_MIXER_OPS_H
 #define ANDROID_AUDIO_MIXER_OPS_H
 
+#include <audio_utils/channels.h>
+#include <audio_utils/primitives.h>
 #include <system/audio.h>
 
 namespace android {
@@ -229,15 +231,26 @@
  * complexity of working on interleaved streams is now getting
  * too high, and likely limits compiler optimization.
  */
-template <int MIXTYPE, int NCHAN,
+
+// compile-time function.
+constexpr inline bool usesCenterChannel(audio_channel_mask_t mask) {
+    using namespace audio_utils::channels;
+    for (size_t i = 0; i < std::size(kSideFromChannelIdx); ++i) {
+        if ((mask & (1 << i)) != 0 && kSideFromChannelIdx[i] == AUDIO_GEOMETRY_SIDE_CENTER) {
+            return true;
+        }
+    }
+    return false;
+}
+
+/*
+ * Applies stereo volume to the audio data based on proper left right channel affinity
+ * (templated channel MASK parameter).
+ */
+template <int MIXTYPE, audio_channel_mask_t MASK,
         typename TO, typename TI, typename TV,
         typename F>
-void stereoVolumeHelper(TO*& out, const TI*& in, const TV *vol, F f) {
-    static_assert(NCHAN > 0 && NCHAN <= FCC_LIMIT);
-    static_assert(MIXTYPE == MIXTYPE_MULTI_STEREOVOL
-            || MIXTYPE == MIXTYPE_MULTI_SAVEONLY_STEREOVOL
-            || MIXTYPE == MIXTYPE_STEREOEXPAND
-            || MIXTYPE == MIXTYPE_MONOEXPAND);
+void stereoVolumeHelperWithChannelMask(TO*& out, const TI*& in, const TV *vol, F f) {
     auto proc = [](auto& a, const auto& b) {
         if constexpr (MIXTYPE == MIXTYPE_MULTI_STEREOVOL
                 || MIXTYPE == MIXTYPE_STEREOEXPAND
@@ -250,59 +263,109 @@
     auto inp = [&in]() -> const TI& {
         if constexpr (MIXTYPE == MIXTYPE_STEREOEXPAND
                 || MIXTYPE == MIXTYPE_MONOEXPAND) {
-            return *in;
+            return *in; // note STEREOEXPAND assumes replicated L/R channels (see doc below).
         } else {
             return *in++;
         }
     };
 
-    // HALs should only expose the canonical channel masks.
-    proc(*out++, f(inp(), vol[0])); // front left
-    if constexpr (NCHAN == 1) return;
-    proc(*out++, f(inp(), vol[1])); // front right
-    if constexpr (NCHAN == 2)  return;
-    if constexpr (NCHAN == 4) {
-        proc(*out++, f(inp(), vol[0])); // back left
-        proc(*out++, f(inp(), vol[1])); // back right
-        return;
-    }
-
-    // TODO: Precompute center volume if not ramping.
     std::decay_t<TV> center;
-    if constexpr (std::is_floating_point_v<TV>) {
-        center = (vol[0] + vol[1]) * 0.5;       // do not use divide
-    } else {
-        center = (vol[0] >> 1) + (vol[1] >> 1); // rounds to 0.
-    }
-    proc(*out++, f(inp(), center)); // center (or 2.1 LFE)
-    if constexpr (NCHAN == 3) return;
-    if constexpr (NCHAN == 5) {
-        proc(*out++, f(inp(), vol[0]));  // back left
-        proc(*out++, f(inp(), vol[1]));  // back right
-        return;
-    }
-
-    proc(*out++, f(inp(), center)); // lfe
-    proc(*out++, f(inp(), vol[0])); // back left
-    proc(*out++, f(inp(), vol[1])); // back right
-    if constexpr (NCHAN == 6) return;
-    if constexpr (NCHAN == 7) {
-        proc(*out++, f(inp(), center)); // back center
-        return;
-    }
-    // NCHAN == 8
-    proc(*out++, f(inp(), vol[0])); // side left
-    proc(*out++, f(inp(), vol[1])); // side right
-    if constexpr (NCHAN > FCC_8) {
-        // Mutes to zero extended surround channels.
-        // 7.1.4 has the correct behavior.
-        // 22.2 has the behavior that FLC and FRC will be mixed instead
-        // of SL and SR and LFE will be center, not left.
-        for (int i = 8; i < NCHAN; ++i) {
-            // TODO: Consider using android::audio_utils::channels::kSideFromChannelIdx
-            proc(*out++, f(inp(), 0.f));
+    constexpr bool USES_CENTER_CHANNEL = usesCenterChannel(MASK);
+    if constexpr (USES_CENTER_CHANNEL) {
+        if constexpr (std::is_floating_point_v<TV>) {
+            center = (vol[0] + vol[1]) * 0.5;       // do not use divide
+        } else {
+            center = (vol[0] >> 1) + (vol[1] >> 1); // rounds to 0.
         }
     }
+
+    using namespace audio_utils::channels;
+
+    // if LFE and LFE2 are both present, they take left and right volume respectively.
+    constexpr unsigned LFE_LFE2 = \
+             AUDIO_CHANNEL_OUT_LOW_FREQUENCY | AUDIO_CHANNEL_OUT_LOW_FREQUENCY_2;
+    constexpr bool has_LFE_LFE2 = (MASK & LFE_LFE2) == LFE_LFE2;
+
+#pragma push_macro("DO_CHANNEL_POSITION")
+#undef DO_CHANNEL_POSITION
+#define DO_CHANNEL_POSITION(BIT_INDEX) \
+    if constexpr ((MASK & (1 << BIT_INDEX)) != 0) { \
+        constexpr auto side = kSideFromChannelIdx[BIT_INDEX]; \
+        if constexpr (side == AUDIO_GEOMETRY_SIDE_LEFT || \
+               has_LFE_LFE2 && (1 << BIT_INDEX) == AUDIO_CHANNEL_OUT_LOW_FREQUENCY) { \
+            proc(*out++, f(inp(), vol[0])); \
+        } else if constexpr (side == AUDIO_GEOMETRY_SIDE_RIGHT || \
+               has_LFE_LFE2 && (1 << BIT_INDEX) == AUDIO_CHANNEL_OUT_LOW_FREQUENCY_2) { \
+            proc(*out++, f(inp(), vol[1])); \
+        } else /* constexpr */ { \
+            proc(*out++, f(inp(), center)); \
+        } \
+    }
+
+    DO_CHANNEL_POSITION(0);
+    DO_CHANNEL_POSITION(1);
+    DO_CHANNEL_POSITION(2);
+    DO_CHANNEL_POSITION(3);
+    DO_CHANNEL_POSITION(4);
+    DO_CHANNEL_POSITION(5);
+    DO_CHANNEL_POSITION(6);
+    DO_CHANNEL_POSITION(7);
+
+    DO_CHANNEL_POSITION(8);
+    DO_CHANNEL_POSITION(9);
+    DO_CHANNEL_POSITION(10);
+    DO_CHANNEL_POSITION(11);
+    DO_CHANNEL_POSITION(12);
+    DO_CHANNEL_POSITION(13);
+    DO_CHANNEL_POSITION(14);
+    DO_CHANNEL_POSITION(15);
+
+    DO_CHANNEL_POSITION(16);
+    DO_CHANNEL_POSITION(17);
+    DO_CHANNEL_POSITION(18);
+    DO_CHANNEL_POSITION(19);
+    DO_CHANNEL_POSITION(20);
+    DO_CHANNEL_POSITION(21);
+    DO_CHANNEL_POSITION(22);
+    DO_CHANNEL_POSITION(23);
+    static_assert(FCC_LIMIT <= FCC_24); // Note: this may need to change.
+#pragma pop_macro("DO_CHANNEL_POSITION")
+}
+
+// These are the channel position masks we expect from the HAL.
+// See audio_channel_out_mask_from_count() but this is constexpr
+constexpr inline audio_channel_mask_t canonicalChannelMaskFromCount(size_t channelCount) {
+    constexpr audio_channel_mask_t canonical[] = {
+        [0] = AUDIO_CHANNEL_NONE,
+        [1] = AUDIO_CHANNEL_OUT_MONO,
+        [2] = AUDIO_CHANNEL_OUT_STEREO,
+        [3] = AUDIO_CHANNEL_OUT_2POINT1,
+        [4] = AUDIO_CHANNEL_OUT_QUAD,
+        [5] = AUDIO_CHANNEL_OUT_PENTA,
+        [6] = AUDIO_CHANNEL_OUT_5POINT1,
+        [7] = AUDIO_CHANNEL_OUT_6POINT1,
+        [8] = AUDIO_CHANNEL_OUT_7POINT1,
+        [12] = AUDIO_CHANNEL_OUT_7POINT1POINT4,
+        [24] = AUDIO_CHANNEL_OUT_22POINT2,
+    };
+    return channelCount < std::size(canonical) ? canonical[channelCount] : AUDIO_CHANNEL_NONE;
+}
+
+template <int MIXTYPE, int NCHAN,
+        typename TO, typename TI, typename TV,
+        typename F>
+void stereoVolumeHelper(TO*& out, const TI*& in, const TV *vol, F f) {
+    static_assert(NCHAN > 0 && NCHAN <= FCC_LIMIT);
+    static_assert(MIXTYPE == MIXTYPE_MULTI_STEREOVOL
+            || MIXTYPE == MIXTYPE_MULTI_SAVEONLY_STEREOVOL
+            || MIXTYPE == MIXTYPE_STEREOEXPAND
+            || MIXTYPE == MIXTYPE_MONOEXPAND);
+    constexpr audio_channel_mask_t MASK{canonicalChannelMaskFromCount(NCHAN)};
+    if constexpr (MASK == AUDIO_CHANNEL_NONE) {
+        ALOGE("%s: Invalid position count %d", __func__, NCHAN);
+        return; // not a valid system mask, ignore.
+    }
+    stereoVolumeHelperWithChannelMask<MIXTYPE, MASK, TO, TI, TV, F>(out, in, vol, f);
 }
 
 /*
diff --git a/media/libaudioprocessing/include/media/AudioMixer.h b/media/libaudioprocessing/include/media/AudioMixer.h
index 70eafe3..5a9fa07 100644
--- a/media/libaudioprocessing/include/media/AudioMixer.h
+++ b/media/libaudioprocessing/include/media/AudioMixer.h
@@ -50,6 +50,7 @@
         // for haptic
         HAPTIC_ENABLED  = 0x4007, // Set haptic data from this track should be played or not.
         HAPTIC_INTENSITY = 0x4008, // Set the intensity to play haptic data.
+        HAPTIC_MAX_AMPLITUDE = 0x4009, // Set the max amplitude allowed for haptic data.
         // for target TIMESTRETCH
         PLAYBACK_RATE   = 0x4300, // Configure timestretch on this track name;
                                   // parameter 'value' is a pointer to the new playback rate.
@@ -145,6 +146,7 @@
         // Haptic
         bool                 mHapticPlaybackEnabled;
         os::HapticScale      mHapticIntensity;
+        float                mHapticMaxAmplitude;
         audio_channel_mask_t mHapticChannelMask;
         uint32_t             mHapticChannelCount;
         audio_channel_mask_t mMixerHapticChannelMask;
diff --git a/media/libaudioprocessing/tests/Android.bp b/media/libaudioprocessing/tests/Android.bp
index 3856817..ad402db 100644
--- a/media/libaudioprocessing/tests/Android.bp
+++ b/media/libaudioprocessing/tests/Android.bp
@@ -76,6 +76,7 @@
 //
 cc_binary {
     name: "mixerops_objdump",
+    header_libs: ["libaudioutils_headers"],
     srcs: ["mixerops_objdump.cpp"],
 }
 
@@ -84,6 +85,16 @@
 //
 cc_benchmark {
     name: "mixerops_benchmark",
+    header_libs: ["libaudioutils_headers"],
     srcs: ["mixerops_benchmark.cpp"],
     static_libs: ["libgoogle-benchmark"],
 }
+
+//
+// mixerops unit test
+//
+cc_test {
+    name: "mixerops_tests",
+    defaults: ["libaudioprocessing_test_defaults"],
+    srcs: ["mixerops_tests.cpp"],
+}
diff --git a/media/libaudioprocessing/tests/mixerops_benchmark.cpp b/media/libaudioprocessing/tests/mixerops_benchmark.cpp
index 7a4c5c7..f866b1a 100644
--- a/media/libaudioprocessing/tests/mixerops_benchmark.cpp
+++ b/media/libaudioprocessing/tests/mixerops_benchmark.cpp
@@ -16,11 +16,9 @@
 
 #include <inttypes.h>
 #include <type_traits>
-#include "../../../../system/media/audio_utils/include/audio_utils/primitives.h"
 #define LOG_ALWAYS_FATAL(...)
 
 #include <../AudioMixerOps.h>
-
 #include <benchmark/benchmark.h>
 
 using namespace android;
diff --git a/media/libaudioprocessing/tests/mixerops_tests.cpp b/media/libaudioprocessing/tests/mixerops_tests.cpp
new file mode 100644
index 0000000..2500ba9
--- /dev/null
+++ b/media/libaudioprocessing/tests/mixerops_tests.cpp
@@ -0,0 +1,175 @@
+/*
+ * Copyright (C) 2021 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "mixerop_tests"
+#include <log/log.h>
+
+#include <inttypes.h>
+#include <type_traits>
+
+#include <../AudioMixerOps.h>
+#include <gtest/gtest.h>
+
+using namespace android;
+
+// Note: gtest templated tests require typenames, not integers.
+template <int MIXTYPE, int NCHAN>
+class MixerOpsBasicTest {
+public:
+    static void testStereoVolume() {
+        using namespace android::audio_utils::channels;
+
+        constexpr size_t FRAME_COUNT = 1000;
+        constexpr size_t SAMPLE_COUNT = FRAME_COUNT * NCHAN;
+
+        const float in[SAMPLE_COUNT] = {[0 ... (SAMPLE_COUNT - 1)] = 1.f};
+
+        AUDIO_GEOMETRY_SIDE sides[NCHAN];
+        size_t i = 0;
+        unsigned channel = canonicalChannelMaskFromCount(NCHAN);
+        constexpr unsigned LFE_LFE2 =
+                AUDIO_CHANNEL_OUT_LOW_FREQUENCY | AUDIO_CHANNEL_OUT_LOW_FREQUENCY_2;
+        bool has_LFE_LFE2 = (channel & LFE_LFE2) == LFE_LFE2;
+        while (channel != 0) {
+            const int index = __builtin_ctz(channel);
+            if (has_LFE_LFE2 && (1 << index) == AUDIO_CHANNEL_OUT_LOW_FREQUENCY) {
+                sides[i++] = AUDIO_GEOMETRY_SIDE_LEFT; // special case
+            } else if (has_LFE_LFE2 && (1 << index) == AUDIO_CHANNEL_OUT_LOW_FREQUENCY_2) {
+                sides[i++] = AUDIO_GEOMETRY_SIDE_RIGHT; // special case
+            } else {
+                sides[i++] = sideFromChannelIdx(index);
+            }
+            channel &= ~(1 << index);
+        }
+
+        float vola[2] = {1.f, 0.f}; // left volume at max.
+        float out[SAMPLE_COUNT]{};
+        float aux[FRAME_COUNT]{};
+        float volaux = 0.5;
+        {
+            volumeMulti<MIXTYPE, NCHAN>(out, FRAME_COUNT, in, aux, vola, volaux);
+            const float *outp = out;
+            const float *auxp = aux;
+            const float left = vola[0];
+            const float center = (vola[0] + vola[1]) * 0.5;
+            const float right = vola[1];
+            for (size_t i = 0; i < FRAME_COUNT; ++i) {
+                for (size_t j = 0; j < NCHAN; ++j) {
+                    const float audio = *outp++;
+                    if (sides[j] == AUDIO_GEOMETRY_SIDE_LEFT) {
+                        EXPECT_EQ(left, audio);
+                    } else if (sides[j] == AUDIO_GEOMETRY_SIDE_CENTER) {
+                        EXPECT_EQ(center, audio);
+                    } else {
+                        EXPECT_EQ(right, audio);
+                    }
+                }
+                EXPECT_EQ(volaux, *auxp++);  // works if all channels contain 1.f
+            }
+        }
+        float volb[2] = {0.f, 0.5f}; // right volume at half max.
+        {
+            // this accumulates into out, aux.
+            // float out[SAMPLE_COUNT]{};
+            // float aux[FRAME_COUNT]{};
+            volumeMulti<MIXTYPE, NCHAN>(out, FRAME_COUNT, in, aux, volb, volaux);
+            const float *outp = out;
+            const float *auxp = aux;
+            const float left = vola[0] + volb[0];
+            const float center = (vola[0] + vola[1] + volb[0] + volb[1]) * 0.5;
+            const float right = vola[1] + volb[1];
+            for (size_t i = 0; i < FRAME_COUNT; ++i) {
+                for (size_t j = 0; j < NCHAN; ++j) {
+                    const float audio = *outp++;
+                    if (sides[j] == AUDIO_GEOMETRY_SIDE_LEFT) {
+                        EXPECT_EQ(left, audio);
+                    } else if (sides[j] == AUDIO_GEOMETRY_SIDE_CENTER) {
+                        EXPECT_EQ(center, audio);
+                    } else {
+                        EXPECT_EQ(right, audio);
+                    }
+                }
+                // aux is accumulated so 2x the amplitude
+                EXPECT_EQ(volaux * 2.f, *auxp++);  // works if all channels contain 1.f
+            }
+        }
+
+        { // test aux as derived from out.
+            // AUX channel is the weighted sum of all of the output channels prior to volume
+            // adjustment.  We must set L and R to the same volume to allow computation
+            // of AUX from the output values.
+            const float volmono = 0.25f;
+            const float vollr[2] = {volmono, volmono}; // all the same.
+            float out[SAMPLE_COUNT]{};
+            float aux[FRAME_COUNT]{};
+            volumeMulti<MIXTYPE, NCHAN>(out, FRAME_COUNT, in, aux, vollr, volaux);
+            const float *outp = out;
+            const float *auxp = aux;
+            for (size_t i = 0; i < FRAME_COUNT; ++i) {
+                float accum = 0.f;
+                for (size_t j = 0; j < NCHAN; ++j) {
+                    accum += *outp++;
+                }
+                EXPECT_EQ(accum / NCHAN * volaux / volmono, *auxp++);
+            }
+        }
+    }
+};
+
+TEST(mixerops, stereovolume_1) { // Note: mono not used for output sinks yet.
+    MixerOpsBasicTest<MIXTYPE_MULTI_STEREOVOL, 1>::testStereoVolume();
+}
+TEST(mixerops, stereovolume_2) {
+    MixerOpsBasicTest<MIXTYPE_MULTI_STEREOVOL, 2>::testStereoVolume();
+}
+TEST(mixerops, stereovolume_3) {
+    MixerOpsBasicTest<MIXTYPE_MULTI_STEREOVOL, 3>::testStereoVolume();
+}
+TEST(mixerops, stereovolume_4) {
+    MixerOpsBasicTest<MIXTYPE_MULTI_STEREOVOL, 4>::testStereoVolume();
+}
+TEST(mixerops, stereovolume_5) {
+    MixerOpsBasicTest<MIXTYPE_MULTI_STEREOVOL, 5>::testStereoVolume();
+}
+TEST(mixerops, stereovolume_6) {
+    MixerOpsBasicTest<MIXTYPE_MULTI_STEREOVOL, 6>::testStereoVolume();
+}
+TEST(mixerops, stereovolume_7) {
+    MixerOpsBasicTest<MIXTYPE_MULTI_STEREOVOL, 7>::testStereoVolume();
+}
+TEST(mixerops, stereovolume_8) {
+    MixerOpsBasicTest<MIXTYPE_MULTI_STEREOVOL, 8>::testStereoVolume();
+}
+TEST(mixerops, stereovolume_12) {
+    if constexpr (FCC_LIMIT >= 12) { // NOTE: FCC_LIMIT is an enum, so can't #if
+        MixerOpsBasicTest<MIXTYPE_MULTI_STEREOVOL, 12>::testStereoVolume();
+    }
+}
+TEST(mixerops, stereovolume_24) {
+    if constexpr (FCC_LIMIT >= 24) {
+        MixerOpsBasicTest<MIXTYPE_MULTI_STEREOVOL, 24>::testStereoVolume();
+    }
+}
+TEST(mixerops, channel_equivalence) {
+    // we must match the constexpr function with the system determined channel mask from count.
+    for (size_t i = 0; i < FCC_LIMIT; ++i) {
+        const audio_channel_mask_t actual = canonicalChannelMaskFromCount(i);
+        const audio_channel_mask_t system = audio_channel_out_mask_from_count(i);
+        if (system == AUDIO_CHANNEL_INVALID) continue;
+        EXPECT_EQ(system, actual);
+    }
+}
diff --git a/media/libeffects/hapticgenerator/Android.bp b/media/libeffects/hapticgenerator/Android.bp
index a660957..03ce329 100644
--- a/media/libeffects/hapticgenerator/Android.bp
+++ b/media/libeffects/hapticgenerator/Android.bp
@@ -45,6 +45,7 @@
 
     shared_libs: [
         "libaudioutils",
+        "libbase",
         "libbinder",
         "liblog",
         "libutils",
diff --git a/media/libeffects/hapticgenerator/EffectHapticGenerator.cpp b/media/libeffects/hapticgenerator/EffectHapticGenerator.cpp
index 65a20a7..3137e13 100644
--- a/media/libeffects/hapticgenerator/EffectHapticGenerator.cpp
+++ b/media/libeffects/hapticgenerator/EffectHapticGenerator.cpp
@@ -22,12 +22,15 @@
 
 #include <algorithm>
 #include <memory>
+#include <string>
 #include <utility>
 
 #include <errno.h>
 #include <inttypes.h>
 #include <math.h>
 
+#include <android-base/parsedouble.h>
+#include <android-base/properties.h>
 #include <audio_effects/effect_hapticgenerator.h>
 #include <audio_utils/format.h>
 #include <system/audio.h>
@@ -35,6 +38,7 @@
 static constexpr float DEFAULT_RESONANT_FREQUENCY = 150.0f;
 static constexpr float DEFAULT_BSF_ZERO_Q = 8.0f;
 static constexpr float DEFAULT_BSF_POLE_Q = 4.0f;
+static constexpr float DEFAULT_DISTORTION_OUTPUT_GAIN = 1.5f;
 
 // This is the only symbol that needs to be exported
 __attribute__ ((visibility ("default")))
@@ -81,6 +85,15 @@
 
 namespace {
 
+float getFloatProperty(const std::string& key, float defaultValue) {
+    float result;
+    std::string value = android::base::GetProperty(key, "");
+    if (!value.empty() && android::base::ParseFloat(value, &result)) {
+        return result;
+    }
+    return defaultValue;
+}
+
 int HapticGenerator_Init(struct HapticGeneratorContext *context) {
     context->itfe = &gHapticGeneratorInterface;
 
@@ -114,7 +127,9 @@
     context->param.distortionCornerFrequency = 300.0f;
     context->param.distortionInputGain = 0.3f;
     context->param.distortionCubeThreshold = 0.1f;
-    context->param.distortionOutputGain = 1.5f;
+    context->param.distortionOutputGain = getFloatProperty(
+            "vendor.audio.hapticgenerator.distortion.output.gain", DEFAULT_DISTORTION_OUTPUT_GAIN);
+    ALOGD("Using distortion output gain as %f", context->param.distortionOutputGain);
 
     context->state = HAPTICGENERATOR_STATE_INITIALIZED;
     return 0;
@@ -287,15 +302,17 @@
         break;
     }
     case HG_PARAM_VIBRATOR_INFO: {
-        if (value == nullptr || size != 2 * sizeof(float)) {
+        if (value == nullptr || size != 3 * sizeof(float)) {
             return -EINVAL;
         }
         const float resonantFrequency = *(float*) value;
         const float qFactor = *((float *) value + 1);
+        const float maxAmplitude = *((float *) value + 2);
         context->param.resonantFrequency =
                 isnan(resonantFrequency) ? DEFAULT_RESONANT_FREQUENCY : resonantFrequency;
         context->param.bsfZeroQ = isnan(qFactor) ? DEFAULT_BSF_POLE_Q : qFactor;
         context->param.bsfPoleQ = context->param.bsfZeroQ / 2.0f;
+        context->param.maxHapticAmplitude = maxAmplitude;
 
         if (context->processorsRecord.bpf != nullptr) {
             context->processorsRecord.bpf->setCoefficients(
@@ -448,7 +465,8 @@
     float* hapticOutBuffer = HapticGenerator_runProcessingChain(
             context->processingChain, context->inputBuffer.data(),
             context->outputBuffer.data(), inBuffer->frameCount);
-    os::scaleHapticData(hapticOutBuffer, hapticSampleCount, context->param.maxHapticIntensity);
+    os::scaleHapticData(hapticOutBuffer, hapticSampleCount, context->param.maxHapticIntensity,
+                        context->param.maxHapticAmplitude);
 
     // For haptic data, the haptic playback thread will copy the data from effect input buffer,
     // which contains haptic data at the end of the buffer, directly to sink buffer.
diff --git a/media/libeffects/hapticgenerator/EffectHapticGenerator.h b/media/libeffects/hapticgenerator/EffectHapticGenerator.h
index 96b744a..85e961f 100644
--- a/media/libeffects/hapticgenerator/EffectHapticGenerator.h
+++ b/media/libeffects/hapticgenerator/EffectHapticGenerator.h
@@ -51,6 +51,7 @@
     // A map from track id to haptic intensity.
     std::map<int, os::HapticScale> id2Intensity;
     os::HapticScale maxHapticIntensity; // max intensity will be used to scale haptic data.
+    float maxHapticAmplitude; // max amplitude will be used to limit haptic data absolute values.
 
     float resonantFrequency;
     float bpfQ;
diff --git a/media/ndk/NdkMediaFormat.cpp b/media/ndk/NdkMediaFormat.cpp
index c1793ce..b035e5a 100644
--- a/media/ndk/NdkMediaFormat.cpp
+++ b/media/ndk/NdkMediaFormat.cpp
@@ -351,6 +351,11 @@
 EXPORT const char* AMEDIAFORMAT_KEY_MIME = "mime";
 EXPORT const char* AMEDIAFORMAT_KEY_MPEG_USER_DATA = "mpeg-user-data";
 EXPORT const char* AMEDIAFORMAT_KEY_MPEG2_STREAM_HEADER = "mpeg2-stream-header";
+EXPORT const char* AMEDIAFORMAT_KEY_MPEGH_COMPATIBLE_SETS = "mpegh-compatible-sets";
+EXPORT const char* AMEDIAFORMAT_KEY_MPEGH_PROFILE_LEVEL_INDICATION =
+        "mpegh-profile-level-indication";
+EXPORT const char* AMEDIAFORMAT_KEY_MPEGH_REFERENCE_CHANNEL_LAYOUT =
+        "mpegh-reference-channel-layout";
 EXPORT const char* AMEDIAFORMAT_KEY_OPERATING_RATE = "operating-rate";
 EXPORT const char* AMEDIAFORMAT_KEY_PCM_ENCODING = "pcm-encoding";
 EXPORT const char* AMEDIAFORMAT_KEY_PRIORITY = "priority";
diff --git a/media/ndk/include/media/NdkMediaFormat.h b/media/ndk/include/media/NdkMediaFormat.h
index fbd855d..2d2fcc0 100644
--- a/media/ndk/include/media/NdkMediaFormat.h
+++ b/media/ndk/include/media/NdkMediaFormat.h
@@ -320,6 +320,34 @@
 extern const char* AMEDIAFORMAT_VIDEO_QP_P_MAX __INTRODUCED_IN(31);
 extern const char* AMEDIAFORMAT_VIDEO_QP_P_MIN __INTRODUCED_IN(31);
 
+/**
+ * MPEG-H audio profile and level compatibility.
+ *
+ * See FDAmd_2 of ISO_IEC_23008-3;2019 MHAProfileAndLevelCompatibilitySetBox.
+ *
+ * Available since API level 32.
+ */
+extern const char* AMEDIAFORMAT_KEY_MPEGH_COMPATIBLE_SETS __INTRODUCED_IN(32);
+
+/**
+ * MPEG-H audio profile level indication.
+ *
+ * See ISO_IEC_23008-3;2019 MHADecoderConfigurationRecord mpegh3daProfileLevelIndication.
+ *
+ * Available since API level 32.
+ */
+extern const char* AMEDIAFORMAT_KEY_MPEGH_PROFILE_LEVEL_INDICATION __INTRODUCED_IN(32);
+
+/**
+ * MPEG-H audio reference channel layout.
+ *
+ * See ISO_IEC_23008-3;2019 MHADecoderConfigurationRecord referenceChannelLayout
+ * and ISO_IEC_23001‐8 ChannelConfiguration value.
+ *
+ * Available since API level 32.
+ */
+extern const char* AMEDIAFORMAT_KEY_MPEGH_REFERENCE_CHANNEL_LAYOUT __INTRODUCED_IN(32);
+
 __END_DECLS
 
 #endif // _NDK_MEDIA_FORMAT_H
diff --git a/media/ndk/libmediandk.map.txt b/media/ndk/libmediandk.map.txt
index 7e9e57e..6f275c7 100644
--- a/media/ndk/libmediandk.map.txt
+++ b/media/ndk/libmediandk.map.txt
@@ -126,6 +126,9 @@
     AMEDIAFORMAT_KEY_MIME; # var introduced=21
     AMEDIAFORMAT_KEY_MPEG_USER_DATA; # var introduced=28
     AMEDIAFORMAT_KEY_MPEG2_STREAM_HEADER; # var introduced=29
+    AMEDIAFORMAT_KEY_MPEGH_COMPATIBLE_SETS; # var introduced=32
+    AMEDIAFORMAT_KEY_MPEGH_PROFILE_LEVEL_INDICATION; # var introduced=32
+    AMEDIAFORMAT_KEY_MPEGH_REFERENCE_CHANNEL_LAYOUT; # var introduced=32
     AMEDIAFORMAT_KEY_OPERATING_RATE; # var introduced=28
     AMEDIAFORMAT_KEY_PCM_BIG_ENDIAN; # var introduced=29
     AMEDIAFORMAT_KEY_PCM_ENCODING; # var introduced=28
diff --git a/media/utils/Android.bp b/media/utils/Android.bp
index bfe73d5..73c4e3b 100644
--- a/media/utils/Android.bp
+++ b/media/utils/Android.bp
@@ -81,6 +81,36 @@
     export_include_dirs: ["include"],
 }
 
+cc_library {
+    name: "libmediautils_vendor",
+    vendor_available: true,  // required for platform/hardware/interfaces
+    srcs: [
+        "MemoryLeakTrackUtil.cpp",
+    ],
+
+    cflags: [
+        "-Wall",
+        "-Wextra",
+        "-Werror",
+    ],
+    shared_libs: [
+        "liblog",
+        "libutils",
+    ],
+
+    static_libs: [
+        "libc_malloc_debug_backtrace",
+    ],
+
+    header_libs: [
+        "bionic_libc_platform_headers",
+    ],
+
+    local_include_dirs: ["include"],
+    export_include_dirs: ["include"],
+}
+
+
 cc_library_headers {
     name: "libmediautils_headers",
     vendor_available: true,  // required for platform/hardware/interfaces
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 6cdb3cd..211d860 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -336,11 +336,11 @@
 }
 
 // getDefaultVibratorInfo_l must be called with AudioFlinger lock held.
-const media::AudioVibratorInfo* AudioFlinger::getDefaultVibratorInfo_l() {
+std::optional<media::AudioVibratorInfo> AudioFlinger::getDefaultVibratorInfo_l() {
     if (mAudioVibratorInfos.empty()) {
-        return nullptr;
+        return {};
     }
-    return &mAudioVibratorInfos.front();
+    return mAudioVibratorInfos.front();
 }
 
 AudioFlinger::~AudioFlinger()
@@ -695,7 +695,7 @@
         // dump all hardware devs
         for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
             sp<DeviceHalInterface> dev = mAudioHwDevs.valueAt(i)->hwDevice();
-            dev->dump(fd);
+            dev->dump(fd, args);
         }
 
         mPatchPanel.dump(fd);
@@ -2456,6 +2456,10 @@
         ThreadBase *thread = (ThreadBase *)mRecordThreads.valueAt(i).get();
         thread->systemReady();
     }
+    for (size_t i = 0; i < mMmapThreads.size(); i++) {
+        ThreadBase *thread = (ThreadBase *)mMmapThreads.valueAt(i).get();
+        thread->systemReady();
+    }
     return NO_ERROR;
 }
 
@@ -2502,7 +2506,8 @@
 
 sp<AudioFlinger::ThreadBase> AudioFlinger::openOutput_l(audio_module_handle_t module,
                                                         audio_io_handle_t *output,
-                                                        audio_config_t *config,
+                                                        audio_config_t *halConfig,
+                                                        audio_config_base_t *mixerConfig __unused,
                                                         audio_devices_t deviceType,
                                                         const String8& address,
                                                         audio_output_flags_t flags)
@@ -2530,16 +2535,16 @@
         // Check only for Normal Mixing mode
         if (kEnableExtendedPrecision) {
             // Specify format (uncomment one below to choose)
-            //config->format = AUDIO_FORMAT_PCM_FLOAT;
-            //config->format = AUDIO_FORMAT_PCM_24_BIT_PACKED;
-            //config->format = AUDIO_FORMAT_PCM_32_BIT;
-            //config->format = AUDIO_FORMAT_PCM_8_24_BIT;
-            // ALOGV("openOutput_l() upgrading format to %#08x", config->format);
+            //halConfig->format = AUDIO_FORMAT_PCM_FLOAT;
+            //halConfig->format = AUDIO_FORMAT_PCM_24_BIT_PACKED;
+            //halConfig->format = AUDIO_FORMAT_PCM_32_BIT;
+            //halConfig->format = AUDIO_FORMAT_PCM_8_24_BIT;
+            // ALOGV("openOutput_l() upgrading format to %#08x", halConfig->format);
         }
         if (kEnableExtendedChannels) {
             // Specify channel mask (uncomment one below to choose)
-            //config->channel_mask = audio_channel_out_mask_from_count(4);  // for USB 4ch
-            //config->channel_mask = audio_channel_mask_from_representation_and_bits(
+            //halConfig->channel_mask = audio_channel_out_mask_from_count(4);  // for USB 4ch
+            //halConfig->channel_mask = audio_channel_mask_from_representation_and_bits(
             //        AUDIO_CHANNEL_REPRESENTATION_INDEX, (1 << 4) - 1);  // another 4ch example
         }
     }
@@ -2550,7 +2555,7 @@
             *output,
             deviceType,
             flags,
-            config,
+            halConfig,
             address.string());
 
     mHardwareStatus = AUDIO_HW_IDLE;
@@ -2565,13 +2570,20 @@
             return thread;
         } else {
             sp<PlaybackThread> thread;
-            if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
+            //TODO: b/193496180 use virtualizer stage flag at audio HAL when available
+            if (flags == (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_FAST
+                                                    | AUDIO_OUTPUT_FLAG_DEEP_BUFFER)) {
+                thread = new VirtualizerStageThread(this, outputStream, *output,
+                                                    mSystemReady, mixerConfig);
+                ALOGD("openOutput_l() created virtualizer output: ID %d thread %p",
+                      *output, thread.get());
+            } else if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
                 thread = new OffloadThread(this, outputStream, *output, mSystemReady);
                 ALOGV("openOutput_l() created offload output: ID %d thread %p",
                       *output, thread.get());
             } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT)
-                    || !isValidPcmSinkFormat(config->format)
-                    || !isValidPcmSinkChannelMask(config->channel_mask)) {
+                    || !isValidPcmSinkFormat(halConfig->format)
+                    || !isValidPcmSinkChannelMask(halConfig->channel_mask)) {
                 thread = new DirectOutputThread(this, outputStream, *output, mSystemReady);
                 ALOGV("openOutput_l() created direct output: ID %d thread %p",
                       *output, thread.get());
@@ -2598,8 +2610,10 @@
 {
     audio_module_handle_t module = VALUE_OR_RETURN_STATUS(
             aidl2legacy_int32_t_audio_module_handle_t(request.module));
-    audio_config_t config = VALUE_OR_RETURN_STATUS(
-            aidl2legacy_AudioConfig_audio_config_t(request.config));
+    audio_config_t halConfig = VALUE_OR_RETURN_STATUS(
+            aidl2legacy_AudioConfig_audio_config_t(request.halConfig));
+    audio_config_base_t mixerConfig = VALUE_OR_RETURN_STATUS(
+            aidl2legacy_AudioConfigBase_audio_config_base_t(request.mixerConfig));
     sp<DeviceDescriptorBase> device = VALUE_OR_RETURN_STATUS(
             aidl2legacy_DeviceDescriptorBase(request.device));
     audio_output_flags_t flags = VALUE_OR_RETURN_STATUS(
@@ -2612,9 +2626,9 @@
               "Channels %#x, flags %#x",
               this, module,
               device->toString().c_str(),
-              config.sample_rate,
-              config.format,
-              config.channel_mask,
+              halConfig.sample_rate,
+              halConfig.format,
+              halConfig.channel_mask,
               flags);
 
     audio_devices_t deviceType = device->type();
@@ -2626,7 +2640,8 @@
 
     Mutex::Autolock _l(mLock);
 
-    sp<ThreadBase> thread = openOutput_l(module, &output, &config, deviceType, address, flags);
+    sp<ThreadBase> thread = openOutput_l(module, &output, &halConfig,
+            &mixerConfig, deviceType, address, flags);
     if (thread != 0) {
         if ((flags & AUDIO_OUTPUT_FLAG_MMAP_NOIRQ) == 0) {
             PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
@@ -2651,7 +2666,8 @@
             mmapThread->ioConfigChanged(AUDIO_OUTPUT_OPENED);
         }
         response->output = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_io_handle_t_int32_t(output));
-        response->config = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_config_t_AudioConfig(config));
+        response->config =
+                VALUE_OR_RETURN_STATUS(legacy2aidl_audio_config_t_AudioConfig(halConfig));
         response->latencyMs = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(latencyMs));
         response->flags = VALUE_OR_RETURN_STATUS(
                 legacy2aidl_audio_output_flags_t_int32_t_mask(flags));
@@ -3798,7 +3814,8 @@
                 io = mPlaybackThreads.keyAt(0);
             }
             ALOGV("createEffect() got io %d for effect %s", io, descOut.name);
-        } else if (checkPlaybackThread_l(io) != nullptr) {
+        } else if (checkPlaybackThread_l(io) != nullptr
+                        && sessionId != AUDIO_SESSION_OUTPUT_STAGE) {
             // allow only one effect chain per sessionId on mPlaybackThreads.
             for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
                 const audio_io_handle_t checkIo = mPlaybackThreads.keyAt(i);
@@ -4179,6 +4196,7 @@
         case TransactionCode::LIST_AUDIO_PATCHES:
         case TransactionCode::SET_AUDIO_PORT_CONFIG:
         case TransactionCode::SET_RECORD_SILENCED:
+        case TransactionCode::AUDIO_POLICY_READY:
             ALOGW("%s: transaction %d received from PID %d",
                   __func__, code, IPCThreadState::self()->getCallingPid());
             // return status only for non void methods
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index fff61f8..8fcd6e4 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -269,6 +269,9 @@
 
     /* Indicate JAVA services are ready (scheduling, power management ...) */
     virtual status_t systemReady();
+    virtual status_t audioPolicyReady() { mAudioPolicyReady.store(true); return NO_ERROR; }
+            bool isAudioPolicyReady() const { return mAudioPolicyReady.load(); }
+
 
     virtual status_t getMicrophones(std::vector<media::MicrophoneInfo> *microphones);
 
@@ -309,7 +312,7 @@
     void updateDownStreamPatches_l(const struct audio_patch *patch,
                                    const std::set<audio_io_handle_t> streams);
 
-    const media::AudioVibratorInfo* getDefaultVibratorInfo_l();
+    std::optional<media::AudioVibratorInfo> getDefaultVibratorInfo_l();
 
 private:
     // FIXME The 400 is temporarily too high until a leak of writers in media.log is fixed.
@@ -735,7 +738,8 @@
                                            const String8& outputDeviceAddress);
               sp<ThreadBase> openOutput_l(audio_module_handle_t module,
                                           audio_io_handle_t *output,
-                                          audio_config_t *config,
+                                          audio_config_t *halConfig,
+                                          audio_config_base_t *mixerConfig,
                                           audio_devices_t deviceType,
                                           const String8& address,
                                           audio_output_flags_t flags);
@@ -986,6 +990,7 @@
     DeviceEffectManager mDeviceEffectManager;
 
     bool       mSystemReady;
+    std::atomic_bool mAudioPolicyReady{};
 
     mediautils::UidInfo mUidInfo;
 
diff --git a/services/audioflinger/DeviceEffectManager.h b/services/audioflinger/DeviceEffectManager.h
index a05f5fe..c222de8 100644
--- a/services/audioflinger/DeviceEffectManager.h
+++ b/services/audioflinger/DeviceEffectManager.h
@@ -163,8 +163,13 @@
     bool isOffloadOrMmap() const override { return false; }
 
     uint32_t  sampleRate() const override { return 0; }
-    audio_channel_mask_t channelMask() const override { return AUDIO_CHANNEL_NONE; }
-    uint32_t channelCount() const override { return 0; }
+    audio_channel_mask_t inChannelMask(int id __unused) const override {
+        return AUDIO_CHANNEL_NONE;
+    }
+    uint32_t inChannelCount(int id __unused) const override { return 0; }
+    audio_channel_mask_t outChannelMask() const override { return AUDIO_CHANNEL_NONE; }
+    uint32_t outChannelCount() const override { return 0; }
+
     audio_channel_mask_t hapticChannelMask() const override { return AUDIO_CHANNEL_NONE; }
     size_t    frameCount() const override  { return 0; }
     uint32_t  latency() const override  { return 0; }
@@ -190,6 +195,10 @@
 
     wp<EffectChain> chain() const override { return nullptr; }
 
+    bool isAudioPolicyReady() const override {
+        return mManager.audioFlinger().isAudioPolicyReady();
+    }
+
     int newEffectId() { return mManager.audioFlinger().nextUniqueId(AUDIO_UNIQUE_ID_USE_EFFECT); }
 
     status_t addEffectToHal(audio_port_handle_t deviceId,
diff --git a/services/audioflinger/Effects.cpp b/services/audioflinger/Effects.cpp
index b267d88..bd661f9 100644
--- a/services/audioflinger/Effects.cpp
+++ b/services/audioflinger/Effects.cpp
@@ -242,6 +242,12 @@
 
     {
         Mutex::Autolock _l(mLock);
+
+        if ((isInternal_l() && !mPolicyRegistered)
+                || !getCallback()->isAudioPolicyReady()) {
+            return NO_ERROR;
+        }
+
         // register effect when first handle is attached and unregister when last handle is removed
         if (mPolicyRegistered != mHandles.size() > 0) {
             doRegister = true;
@@ -875,9 +881,9 @@
     // similar to output EFFECT_FLAG_TYPE_INSERT/REPLACE,
     // in which case input channel masks should be used here.
     callback = getCallback();
-    channelMask = callback->channelMask();
+    channelMask = callback->inChannelMask(mId);
     mConfig.inputCfg.channels = channelMask;
-    mConfig.outputCfg.channels = channelMask;
+    mConfig.outputCfg.channels = callback->outChannelMask();
 
     if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
         if (mConfig.inputCfg.channels != AUDIO_CHANNEL_OUT_MONO) {
@@ -1600,7 +1606,7 @@
     return status;
 }
 
-status_t AudioFlinger::EffectModule::setVibratorInfo(const media::AudioVibratorInfo* vibratorInfo)
+status_t AudioFlinger::EffectModule::setVibratorInfo(const media::AudioVibratorInfo& vibratorInfo)
 {
     if (mStatus != NO_ERROR) {
         return mStatus;
@@ -1610,15 +1616,17 @@
         return INVALID_OPERATION;
     }
 
+    const size_t paramCount = 3;
     std::vector<uint8_t> request(
-            sizeof(effect_param_t) + sizeof(int32_t) + 2 * sizeof(float));
+            sizeof(effect_param_t) + sizeof(int32_t) + paramCount * sizeof(float));
     effect_param_t *param = (effect_param_t*) request.data();
     param->psize = sizeof(int32_t);
-    param->vsize = 2 * sizeof(float);
+    param->vsize = paramCount * sizeof(float);
     *(int32_t*)param->data = HG_PARAM_VIBRATOR_INFO;
     float* vibratorInfoPtr = reinterpret_cast<float*>(param->data + sizeof(int32_t));
-    vibratorInfoPtr[0] = vibratorInfo->resonantFrequency;
-    vibratorInfoPtr[1] = vibratorInfo->qFactor;
+    vibratorInfoPtr[0] = vibratorInfo.resonantFrequency;
+    vibratorInfoPtr[1] = vibratorInfo.qFactor;
+    vibratorInfoPtr[2] = vibratorInfo.maxAmplitude;
     std::vector<uint8_t> response;
     status_t status = command(EFFECT_CMD_SET_PARAM, request, sizeof(int32_t), &response);
     if (status == NO_ERROR) {
@@ -2048,11 +2056,11 @@
       mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX),
       mEffectCallback(new EffectCallback(wp<EffectChain>(this), thread))
 {
-    mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
     sp<ThreadBase> p = thread.promote();
     if (p == nullptr) {
         return;
     }
+    mStrategy = p->getStrategyForStream(AUDIO_STREAM_MUSIC);
     mMaxTailBuffers = ((kProcessTailDurationMs * p->sampleRate()) / 1000) /
                                     p->frameCount();
 }
@@ -2125,8 +2133,8 @@
     if (mInBuffer == NULL) {
         return;
     }
-    const size_t frameSize =
-            audio_bytes_per_sample(EFFECT_BUFFER_FORMAT) * mEffectCallback->channelCount();
+    const size_t frameSize = audio_bytes_per_sample(EFFECT_BUFFER_FORMAT)
+            * mEffectCallback->inChannelCount(mEffects[0]->id());
 
     memset(mInBuffer->audioBuffer()->raw, 0, mEffectCallback->frameCount() * frameSize);
     mInBuffer->commit();
@@ -2236,6 +2244,9 @@
                 numSamples * sizeof(int32_t), &halBuffer);
 #endif
         if (result != OK) return result;
+
+        effect->configure();
+
         effect->setInBuffer(halBuffer);
         // auxiliary effects output samples to chain input buffer for further processing
         // by insert effects
@@ -2303,6 +2314,10 @@
             }
         }
 
+        mEffects.insertAt(effect, idx_insert);
+
+        effect->configure();
+
         // always read samples from chain input buffer
         effect->setInBuffer(mInBuffer);
 
@@ -2310,14 +2325,13 @@
         // output buffer, otherwise to chain input buffer
         if (idx_insert == size) {
             if (idx_insert != 0) {
-                mEffects[idx_insert-1]->setOutBuffer(mInBuffer);
                 mEffects[idx_insert-1]->configure();
+                mEffects[idx_insert-1]->setOutBuffer(mInBuffer);
             }
             effect->setOutBuffer(mOutBuffer);
         } else {
             effect->setOutBuffer(mInBuffer);
         }
-        mEffects.insertAt(effect, idx_insert);
 
         ALOGV("addEffect_l() effect %p, added in chain %p at rank %zu", effect.get(), this,
                 idx_insert);
@@ -2350,14 +2364,21 @@
 
             if (type != EFFECT_FLAG_TYPE_AUXILIARY) {
                 if (i == size - 1 && i != 0) {
-                    mEffects[i - 1]->setOutBuffer(mOutBuffer);
                     mEffects[i - 1]->configure();
+                    mEffects[i - 1]->setOutBuffer(mOutBuffer);
                 }
             }
             mEffects.removeAt(i);
+
+            // make sure the input buffer configuration for the new first effect in the chain
+            // is updated if needed (can switch from HAL channel mask to mixer channel mask)
+            if (i == 0 && size > 1) {
+                mEffects[0]->configure();
+                mEffects[0]->setInBuffer(mInBuffer);
+            }
+
             ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %zu", effect.get(),
                     this, i);
-
             break;
         }
     }
@@ -2932,7 +2953,43 @@
     return t->sampleRate();
 }
 
-audio_channel_mask_t AudioFlinger::EffectChain::EffectCallback::channelMask() const {
+audio_channel_mask_t AudioFlinger::EffectChain::EffectCallback::inChannelMask(int id) const {
+    sp<ThreadBase> t = thread().promote();
+    if (t == nullptr) {
+        return AUDIO_CHANNEL_NONE;
+    }
+    sp<EffectChain> c = chain().promote();
+    if (c == nullptr) {
+        return AUDIO_CHANNEL_NONE;
+    }
+
+    if (c->sessionId() != AUDIO_SESSION_OUTPUT_STAGE
+            || c->isFirstEffect(id)) {
+        return t->mixerChannelMask();
+    } else {
+        return t->channelMask();
+    }
+}
+
+uint32_t AudioFlinger::EffectChain::EffectCallback::inChannelCount(int id) const {
+    sp<ThreadBase> t = thread().promote();
+    if (t == nullptr) {
+        return 0;
+    }
+    sp<EffectChain> c = chain().promote();
+    if (c == nullptr) {
+        return 0;
+    }
+
+    if (c->sessionId() != AUDIO_SESSION_OUTPUT_STAGE
+            || c->isFirstEffect(id)) {
+        return audio_channel_count_from_out_mask(t->mixerChannelMask());
+    } else {
+        return t->channelCount();
+    }
+}
+
+audio_channel_mask_t AudioFlinger::EffectChain::EffectCallback::outChannelMask() const {
     sp<ThreadBase> t = thread().promote();
     if (t == nullptr) {
         return AUDIO_CHANNEL_NONE;
@@ -2940,7 +2997,7 @@
     return t->channelMask();
 }
 
-uint32_t AudioFlinger::EffectChain::EffectCallback::channelCount() const {
+uint32_t AudioFlinger::EffectChain::EffectCallback::outChannelCount() const {
     sp<ThreadBase> t = thread().promote();
     if (t == nullptr) {
         return 0;
@@ -3364,7 +3421,8 @@
     return proxy->sampleRate();
 }
 
-audio_channel_mask_t AudioFlinger::DeviceEffectProxy::ProxyCallback::channelMask() const {
+audio_channel_mask_t AudioFlinger::DeviceEffectProxy::ProxyCallback::inChannelMask(
+        int id __unused) const {
     sp<DeviceEffectProxy> proxy = mProxy.promote();
     if (proxy == nullptr) {
         return AUDIO_CHANNEL_OUT_STEREO;
@@ -3372,7 +3430,23 @@
     return proxy->channelMask();
 }
 
-uint32_t AudioFlinger::DeviceEffectProxy::ProxyCallback::channelCount() const {
+uint32_t AudioFlinger::DeviceEffectProxy::ProxyCallback::inChannelCount(int id __unused) const {
+    sp<DeviceEffectProxy> proxy = mProxy.promote();
+    if (proxy == nullptr) {
+        return 2;
+    }
+    return proxy->channelCount();
+}
+
+audio_channel_mask_t AudioFlinger::DeviceEffectProxy::ProxyCallback::outChannelMask() const {
+    sp<DeviceEffectProxy> proxy = mProxy.promote();
+    if (proxy == nullptr) {
+        return AUDIO_CHANNEL_OUT_STEREO;
+    }
+    return proxy->channelMask();
+}
+
+uint32_t AudioFlinger::DeviceEffectProxy::ProxyCallback::outChannelCount() const {
     sp<DeviceEffectProxy> proxy = mProxy.promote();
     if (proxy == nullptr) {
         return 2;
diff --git a/services/audioflinger/Effects.h b/services/audioflinger/Effects.h
index a727e04..1d0d00d 100644
--- a/services/audioflinger/Effects.h
+++ b/services/audioflinger/Effects.h
@@ -34,8 +34,10 @@
     virtual bool isOffloadOrDirect() const = 0;
     virtual bool isOffloadOrMmap() const = 0;
     virtual uint32_t sampleRate() const = 0;
-    virtual audio_channel_mask_t channelMask() const = 0;
-    virtual uint32_t channelCount() const = 0;
+    virtual audio_channel_mask_t inChannelMask(int id) const = 0;
+    virtual uint32_t inChannelCount(int id) const = 0;
+    virtual audio_channel_mask_t outChannelMask() const = 0;
+    virtual uint32_t outChannelCount() const = 0;
     virtual audio_channel_mask_t hapticChannelMask() const = 0;
     virtual size_t frameCount() const = 0;
 
@@ -64,6 +66,8 @@
     virtual void resetVolume() = 0;
 
     virtual wp<EffectChain> chain() const = 0;
+
+    virtual bool isAudioPolicyReady() const = 0;
 };
 
 // EffectBase(EffectModule) and EffectChain classes both have their own mutex to protect
@@ -164,6 +168,16 @@
 
     void             dump(int fd, const Vector<String16>& args);
 
+protected:
+    bool             isInternal_l() const {
+                         for (auto handle : mHandles) {
+                            if (handle->client() != nullptr) {
+                                return false;
+                            }
+                         }
+                         return true;
+                     }
+
 private:
     friend class AudioFlinger;      // for mHandles
     bool             mPinned = false;
@@ -259,7 +273,7 @@
     bool             isHapticGenerator() const;
 
     status_t         setHapticIntensity(int id, int intensity);
-    status_t         setVibratorInfo(const media::AudioVibratorInfo* vibratorInfo);
+    status_t         setVibratorInfo(const media::AudioVibratorInfo& vibratorInfo);
 
     void             dump(int fd, const Vector<String16>& args);
 
@@ -342,6 +356,8 @@
     android::binder::Status disconnect() override;
     android::binder::Status getCblk(media::SharedFileRegion* _aidl_return) override;
 
+    sp<Client> client() const { return mClient; }
+
 private:
     void disconnect(bool unpinIfLast);
 
@@ -511,6 +527,8 @@
     sp<EffectCallbackInterface> effectCallback() const { return mEffectCallback; }
     wp<ThreadBase> thread() const { return mEffectCallback->thread(); }
 
+    bool isFirstEffect(int id) const { return !mEffects.isEmpty() && id == mEffects[0]->id(); }
+
     void dump(int fd, const Vector<String16>& args);
 
 private:
@@ -544,8 +562,10 @@
         bool isOffloadOrMmap() const override;
 
         uint32_t sampleRate() const override;
-        audio_channel_mask_t channelMask() const override;
-        uint32_t channelCount() const override;
+        audio_channel_mask_t inChannelMask(int id) const override;
+        uint32_t inChannelCount(int id) const override;
+        audio_channel_mask_t outChannelMask() const override;
+        uint32_t outChannelCount() const override;
         audio_channel_mask_t hapticChannelMask() const override;
         size_t frameCount() const override;
         uint32_t latency() const override;
@@ -566,6 +586,10 @@
 
         wp<EffectChain> chain() const override { return mChain; }
 
+        bool isAudioPolicyReady() const override {
+            return mAudioFlinger.isAudioPolicyReady();
+        }
+
         wp<ThreadBase> thread() const { return mThread.load(); }
 
         void setThread(const wp<ThreadBase>& thread) {
@@ -694,8 +718,10 @@
         bool isOffloadOrMmap() const override { return false; }
 
         uint32_t sampleRate() const override;
-        audio_channel_mask_t channelMask() const override;
-        uint32_t channelCount() const override;
+        audio_channel_mask_t inChannelMask(int id) const override;
+        uint32_t inChannelCount(int id) const override;
+        audio_channel_mask_t outChannelMask() const override;
+        uint32_t outChannelCount() const override;
         audio_channel_mask_t hapticChannelMask() const override { return AUDIO_CHANNEL_NONE; }
         size_t frameCount() const override  { return 0; }
         uint32_t latency() const override  { return 0; }
@@ -716,6 +742,10 @@
 
         wp<EffectChain> chain() const override { return nullptr; }
 
+        bool isAudioPolicyReady() const override {
+            return mManagerCallback->isAudioPolicyReady();
+        }
+
         int newEffectId();
 
     private:
diff --git a/services/audioflinger/FastMixer.cpp b/services/audioflinger/FastMixer.cpp
index 88d4eaf..fc34d95 100644
--- a/services/audioflinger/FastMixer.cpp
+++ b/services/audioflinger/FastMixer.cpp
@@ -204,6 +204,8 @@
                 (void *)(uintptr_t)fastTrack->mHapticPlaybackEnabled);
         mMixer->setParameter(index, AudioMixer::TRACK, AudioMixer::HAPTIC_INTENSITY,
                 (void *)(uintptr_t)fastTrack->mHapticIntensity);
+        mMixer->setParameter(index, AudioMixer::TRACK, AudioMixer::HAPTIC_MAX_AMPLITUDE,
+                (void *)(&(fastTrack->mHapticMaxAmplitude)));
 
         mMixer->enable(index);
         break;
diff --git a/services/audioflinger/FastMixerState.h b/services/audioflinger/FastMixerState.h
index 857d3de..ce3cc14 100644
--- a/services/audioflinger/FastMixerState.h
+++ b/services/audioflinger/FastMixerState.h
@@ -17,6 +17,8 @@
 #ifndef ANDROID_AUDIO_FAST_MIXER_STATE_H
 #define ANDROID_AUDIO_FAST_MIXER_STATE_H
 
+#include <math.h>
+
 #include <audio_utils/minifloat.h>
 #include <system/audio.h>
 #include <media/AudioMixer.h>
@@ -51,6 +53,7 @@
     int                     mGeneration;     // increment when any field is assigned
     bool                    mHapticPlaybackEnabled = false; // haptic playback is enabled or not
     os::HapticScale         mHapticIntensity = os::HapticScale::MUTE; // intensity of haptic data
+    float                   mHapticMaxAmplitude = NAN; // max amplitude allowed for haptic data
 };
 
 // Represents a single state of the fast mixer
diff --git a/services/audioflinger/PatchPanel.cpp b/services/audioflinger/PatchPanel.cpp
index a381c7d..93118b8 100644
--- a/services/audioflinger/PatchPanel.cpp
+++ b/services/audioflinger/PatchPanel.cpp
@@ -258,6 +258,7 @@
                             reinterpret_cast<PlaybackThread*>(thread.get()), false /*closeThread*/);
                 } else {
                     audio_config_t config = AUDIO_CONFIG_INITIALIZER;
+                    audio_config_base_t mixerConfig = AUDIO_CONFIG_BASE_INITIALIZER;
                     audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
                     audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE;
                     if (patch->sinks[0].config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
@@ -276,6 +277,7 @@
                                                             patch->sinks[0].ext.device.hw_module,
                                                             &output,
                                                             &config,
+                                                            &mixerConfig,
                                                             outputDevice,
                                                             outputDeviceAddress,
                                                             flags);
diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h
index 0929055..e9e98ca 100644
--- a/services/audioflinger/PlaybackTracks.h
+++ b/services/audioflinger/PlaybackTracks.h
@@ -19,6 +19,8 @@
     #error This header file should only be included from AudioFlinger.h
 #endif
 
+#include <math.h>
+
 // Checks and monitors OP_PLAY_AUDIO
 class OpPlayAudioMonitor : public RefBase {
 public:
@@ -161,6 +163,8 @@
             }
             /** Return at what intensity to play haptics, used in mixer. */
             os::HapticScale getHapticIntensity() const { return mHapticIntensity; }
+            /** Return the maximum amplitude allowed for haptics data, used in mixer. */
+            float getHapticMaxAmplitude() const { return mHapticMaxAmplitude; }
             /** Set intensity of haptic playback, should be set after querying vibrator service. */
             void    setHapticIntensity(os::HapticScale hapticIntensity) {
                 if (os::isValidHapticScale(hapticIntensity)) {
@@ -168,6 +172,12 @@
                     setHapticPlaybackEnabled(mHapticIntensity != os::HapticScale::MUTE);
                 }
             }
+            /** Set maximum amplitude allowed for haptic data, should be set after querying
+             *  vibrator service.
+             */
+            void    setHapticMaxAmplitude(float maxAmplitude) {
+                mHapticMaxAmplitude = maxAmplitude;
+            }
             sp<os::ExternalVibration> getExternalVibration() const { return mExternalVibration; }
 
             void    setTeePatches(TeePatches teePatches);
@@ -282,6 +292,8 @@
     bool                mHapticPlaybackEnabled = false; // indicates haptic playback enabled or not
     // intensity to play haptic data
     os::HapticScale mHapticIntensity = os::HapticScale::MUTE;
+    // max amplitude allowed for haptic data
+    float mHapticMaxAmplitude = NAN;
     class AudioVibrationController : public os::BnExternalVibrationController {
     public:
         explicit AudioVibrationController(Track* track) : mTrack(track) {}
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index b9cdab8..7292527 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -50,8 +50,10 @@
 #include <audio_utils/format.h>
 #include <audio_utils/minifloat.h>
 #include <audio_utils/safe_math.h>
-#include <system/audio_effects/effect_ns.h>
 #include <system/audio_effects/effect_aec.h>
+#include <system/audio_effects/effect_downmix.h>
+#include <system/audio_effects/effect_ns.h>
+#include <system/audio_effects/effect_virtualizer_stage.h>
 #include <system/audio.h>
 
 // NBAIO implementations
@@ -507,6 +509,8 @@
         return "MMAP_PLAYBACK";
     case MMAP_CAPTURE:
         return "MMAP_CAPTURE";
+    case VIRTUALIZER_STAGE:
+        return "VIRTUALIZER_STAGE";
     default:
         return "unknown";
     }
@@ -722,6 +726,19 @@
     sendConfigEvent_l(configEvent);
 }
 
+void AudioFlinger::ThreadBase::sendCheckOutputStageEffectsEvent()
+{
+    Mutex::Autolock _l(mLock);
+    sendCheckOutputStageEffectsEvent_l();
+}
+
+void AudioFlinger::ThreadBase::sendCheckOutputStageEffectsEvent_l()
+{
+    sp<ConfigEvent> configEvent =
+            (ConfigEvent *)new CheckOutputStageEffectsEvent();
+    sendConfigEvent_l(configEvent);
+}
+
 // post condition: mConfigEvents.isEmpty()
 void AudioFlinger::ThreadBase::processConfigEvents_l()
 {
@@ -784,6 +801,11 @@
                     (ResizeBufferConfigEventData *)event->mData.get();
             resizeInputBuffer_l(data->mMaxSharedAudioHistoryMs);
         } break;
+
+        case CFG_EVENT_CHECK_OUTPUT_STAGE_EFFECTS: {
+            setCheckOutputStageEffects();
+        } break;
+
         default:
             ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType);
             break;
@@ -1008,6 +1030,8 @@
         return String16("MmapPlayback");
     case MMAP_CAPTURE:
         return String16("MmapCapture");
+    case VIRTUALIZER_STAGE:
+        return String16("AudioVirt");
     default:
         ALOG_ASSERT(false);
         return String16("AudioUnknown");
@@ -1401,6 +1425,13 @@
             return BAD_VALUE;
         }
         break;
+    case VIRTUALIZER_STAGE:
+        if (!audio_is_global_session(sessionId)) {
+            ALOGW("checkEffectCompatibility_l(): non global effect %s on VIRTUALIZER_STAGE"
+                    " thread %s", desc->name, mThreadName);
+            return BAD_VALUE;
+        }
+        break;
     default:
         LOG_ALWAYS_FATAL("checkEffectCompatibility_l(): wrong thread type %d", mType);
     }
@@ -1477,11 +1508,11 @@
         if (effect->isHapticGenerator()) {
             // TODO(b/184194057): Use the vibrator information from the vibrator that will be used
             // for the HapticGenerator.
-            const media::AudioVibratorInfo* defaultVibratorInfo =
-                    mAudioFlinger->getDefaultVibratorInfo_l();
-            if (defaultVibratorInfo != nullptr) {
+            const std::optional<media::AudioVibratorInfo> defaultVibratorInfo =
+                    std::move(mAudioFlinger->getDefaultVibratorInfo_l());
+            if (defaultVibratorInfo) {
                 // Only set the vibrator info when it is a valid one.
-                effect->setVibratorInfo(defaultVibratorInfo);
+                effect->setVibratorInfo(*defaultVibratorInfo);
             }
         }
         // create effect handle and connect it to effect module
@@ -1489,6 +1520,7 @@
         lStatus = handle->initCheck();
         if (lStatus == OK) {
             lStatus = effect->addHandle(handle.get());
+            sendCheckOutputStageEffectsEvent_l();
         }
         if (enabled != NULL) {
             *enabled = (int)effect->isEnabled();
@@ -1531,6 +1563,7 @@
         if (remove) {
             removeEffect_l(effect, true);
         }
+        sendCheckOutputStageEffectsEvent_l();
     }
     if (remove) {
         mAudioFlinger->updateOrphanEffectChains(effect);
@@ -1888,6 +1921,14 @@
     item->selfrecord();
 }
 
+product_strategy_t AudioFlinger::ThreadBase::getStrategyForStream(audio_stream_type_t stream) const
+{
+    if (!mAudioFlinger->isAudioPolicyReady()) {
+        return PRODUCT_STRATEGY_NONE;
+    }
+    return AudioSystem::getStrategyForStream(stream);
+}
+
 // ----------------------------------------------------------------------------
 //      Playback
 // ----------------------------------------------------------------------------
@@ -1896,15 +1937,16 @@
                                              AudioStreamOut* output,
                                              audio_io_handle_t id,
                                              type_t type,
-                                             bool systemReady)
+                                             bool systemReady,
+                                             audio_config_base_t *mixerConfig)
     :   ThreadBase(audioFlinger, id, type, systemReady, true /* isOut */),
         mNormalFrameCount(0), mSinkBuffer(NULL),
-        mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision),
+        mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision || type == VIRTUALIZER_STAGE),
         mMixerBuffer(NULL),
         mMixerBufferSize(0),
         mMixerBufferFormat(AUDIO_FORMAT_INVALID),
         mMixerBufferValid(false),
-        mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision),
+        mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision || type == VIRTUALIZER_STAGE),
         mEffectBuffer(NULL),
         mEffectBufferSize(0),
         mEffectBufferFormat(AUDIO_FORMAT_INVALID),
@@ -1956,8 +1998,18 @@
                 mOutput->audioHwDev->moduleName(), AUDIO_HARDWARE_MODULE_ID_MSD) == 0;
     }
 
+    if (mixerConfig != nullptr && mixerConfig->channel_mask != AUDIO_CHANNEL_NONE) {
+        mMixerChannelMask = mixerConfig->channel_mask;
+    }
+
     readOutputParameters_l();
 
+    if (mType != VIRTUALIZER_STAGE
+            && mMixerChannelMask != mChannelMask) {
+        LOG_ALWAYS_FATAL("HAL channel mask %#x does not match mixer channel mask %#x",
+                mChannelMask, mMixerChannelMask);
+    }
+
     // TODO: We may also match on address as well as device type for
     // AUDIO_DEVICE_OUT_BUS, AUDIO_DEVICE_OUT_ALL_A2DP, AUDIO_DEVICE_OUT_REMOTE_SUBMIX
     if (type == MIXER || type == DIRECT || type == OFFLOAD) {
@@ -2080,10 +2132,12 @@
     write(fd, result.string(), result.size());
 }
 
-void AudioFlinger::PlaybackThread::dumpInternals_l(int fd, const Vector<String16>& args __unused)
+void AudioFlinger::PlaybackThread::dumpInternals_l(int fd, const Vector<String16>& args)
 {
     dprintf(fd, "  Master volume: %f\n", mMasterVolume);
     dprintf(fd, "  Master mute: %s\n", mMasterMute ? "on" : "off");
+    dprintf(fd, "  Mixer channel Mask: %#x (%s)\n",
+            mMixerChannelMask, channelMaskToString(mMixerChannelMask, true /* output */).c_str());
     if (mHapticChannelMask != AUDIO_CHANNEL_NONE) {
         dprintf(fd, "  Haptic channel mask: %#x (%s)\n", mHapticChannelMask,
                 channelMaskToString(mHapticChannelMask, true /* output */).c_str());
@@ -2109,7 +2163,7 @@
     }
     if (output != nullptr) {
         dprintf(fd, "  Hal stream dump:\n");
-        (void)output->stream->dump(fd);
+        (void)output->stream->dump(fd, args);
     }
 }
 
@@ -2397,11 +2451,11 @@
         // all tracks in same audio session must share the same routing strategy otherwise
         // conflicts will happen when tracks are moved from one output to another by audio policy
         // manager
-        product_strategy_t strategy = AudioSystem::getStrategyForStream(streamType);
+        product_strategy_t strategy = getStrategyForStream(streamType);
         for (size_t i = 0; i < mTracks.size(); ++i) {
             sp<Track> t = mTracks[i];
             if (t != 0 && t->isExternalTrack()) {
-                product_strategy_t actual = AudioSystem::getStrategyForStream(t->streamType());
+                product_strategy_t actual = getStrategyForStream(t->streamType());
                 if (sessionId == t->sessionId() && strategy != actual) {
                     ALOGE("createTrack_l() mismatched strategy; expected %u but found %u",
                             strategy, actual);
@@ -2445,7 +2499,7 @@
         if (chain != 0) {
             ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer());
             track->setMainBuffer(chain->inBuffer());
-            chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType()));
+            chain->setStrategy(getStrategyForStream(track->streamType()));
             chain->incTrackCnt();
         }
 
@@ -2613,8 +2667,19 @@
             mLock.unlock();
             const int intensity = AudioFlinger::onExternalVibrationStart(
                     track->getExternalVibration());
+            std::optional<media::AudioVibratorInfo> vibratorInfo;
+            {
+                // TODO(b/184194780): Use the vibrator information from the vibrator that will be
+                // used to play this track.
+                Mutex::Autolock _l(mAudioFlinger->mLock);
+                vibratorInfo = std::move(mAudioFlinger->getDefaultVibratorInfo_l());
+            }
             mLock.lock();
             track->setHapticIntensity(static_cast<os::HapticScale>(intensity));
+            if (vibratorInfo) {
+                track->setHapticMaxAmplitude(vibratorInfo->maxAmplitude);
+            }
+
             // Haptic playback should be enabled by vibrator service.
             if (track->getHapticPlaybackEnabled()) {
                 // Disable haptic playback of all active track to ensure only
@@ -2814,14 +2879,20 @@
     if (!audio_is_output_channel(mChannelMask)) {
         LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask);
     }
-    if ((mType == MIXER || mType == DUPLICATING)
-            && !isValidPcmSinkChannelMask(mChannelMask)) {
+    if (hasMixer() && !isValidPcmSinkChannelMask(mChannelMask)) {
         LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output",
                 mChannelMask);
     }
+
+    if (mMixerChannelMask == AUDIO_CHANNEL_NONE) {
+        mMixerChannelMask = mChannelMask;
+    }
+
     mChannelCount = audio_channel_count_from_out_mask(mChannelMask);
     mBalance.setChannelMask(mChannelMask);
 
+    uint32_t mixerChannelCount = audio_channel_count_from_out_mask(mMixerChannelMask);
+
     // Get actual HAL format.
     status_t result = mOutput->stream->getAudioProperties(nullptr, nullptr, &mHALFormat);
     LOG_ALWAYS_FATAL_IF(result != OK, "Error when retrieving output stream format: %d", result);
@@ -2831,8 +2902,7 @@
     if (!audio_is_valid_format(mFormat)) {
         LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat);
     }
-    if ((mType == MIXER || mType == DUPLICATING)
-            && !isValidPcmSinkFormat(mFormat)) {
+    if (hasMixer() && !isValidPcmSinkFormat(mFormat)) {
         LOG_FATAL("HAL format %#x not supported for mixed output",
                 mFormat);
     }
@@ -2841,7 +2911,7 @@
     LOG_ALWAYS_FATAL_IF(result != OK,
             "Error when retrieving output stream buffer size: %d", result);
     mFrameCount = mBufferSize / mFrameSize;
-    if ((mType == MIXER || mType == DUPLICATING) && (mFrameCount & 15)) {
+    if (hasMixer() && (mFrameCount & 15)) {
         ALOGW("HAL output buffer size is %zu frames but AudioMixer requires multiples of 16 frames",
                 mFrameCount);
     }
@@ -2914,7 +2984,7 @@
     }
     mNormalFrameCount = multiplier * mFrameCount;
     // round up to nearest 16 frames to satisfy AudioMixer
-    if (mType == MIXER || mType == DUPLICATING) {
+    if (hasMixer()) {
         mNormalFrameCount = (mNormalFrameCount + 15) & ~15;
     }
     ALOGI("HAL output buffer size %zu frames, normal sink buffer size %zu frames", mFrameCount,
@@ -2941,7 +3011,7 @@
     mMixerBuffer = NULL;
     if (mMixerBufferEnabled) {
         mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // no longer valid: AUDIO_FORMAT_PCM_16_BIT.
-        mMixerBufferSize = mNormalFrameCount * mChannelCount
+        mMixerBufferSize = mNormalFrameCount * mixerChannelCount
                 * audio_bytes_per_sample(mMixerBufferFormat);
         (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize);
     }
@@ -2949,7 +3019,7 @@
     mEffectBuffer = NULL;
     if (mEffectBufferEnabled) {
         mEffectBufferFormat = EFFECT_BUFFER_FORMAT;
-        mEffectBufferSize = mNormalFrameCount * mChannelCount
+        mEffectBufferSize = mNormalFrameCount * mixerChannelCount
                 * audio_bytes_per_sample(mEffectBufferFormat);
         (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize);
     }
@@ -2958,6 +3028,7 @@
     mChannelMask = static_cast<audio_channel_mask_t>(mChannelMask & ~mHapticChannelMask);
     mHapticChannelCount = audio_channel_count_from_out_mask(mHapticChannelMask);
     mChannelCount -= mHapticChannelCount;
+    mMixerChannelMask = static_cast<audio_channel_mask_t>(mMixerChannelMask & ~mHapticChannelMask);
 
     // force reconfiguration of effect chains and engines to take new buffer size and audio
     // parameters into account
@@ -3051,15 +3122,15 @@
     // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that
     // it is moved to correct output by audio policy manager when A2DP is connected or disconnected
     if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
-        return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
+        return getStrategyForStream(AUDIO_STREAM_MUSIC);
     }
     for (size_t i = 0; i < mTracks.size(); i++) {
         sp<Track> track = mTracks[i];
         if (sessionId == track->sessionId() && !track->isInvalid()) {
-            return AudioSystem::getStrategyForStream(track->streamType());
+            return getStrategyForStream(track->streamType());
         }
     }
-    return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
+    return getStrategyForStream(AUDIO_STREAM_MUSIC);
 }
 
 
@@ -3348,7 +3419,8 @@
         // Only one effect chain can be present in direct output thread and it uses
         // the sink buffer as input
         if (mType != DIRECT) {
-            size_t numSamples = mNormalFrameCount * (mChannelCount + mHapticChannelCount);
+            size_t numSamples = mNormalFrameCount
+                    * (audio_channel_count_from_out_mask(mMixerChannelMask) + mHapticChannelCount);
             status_t result = mAudioFlinger->mEffectsFactoryHal->allocateBuffer(
                     numSamples * sizeof(effect_buffer_t),
                     &halInBuffer);
@@ -3531,6 +3603,8 @@
 
     audio_patch_handle_t lastDownstreamPatchHandle = AUDIO_PATCH_HANDLE_NONE;
 
+    sendCheckOutputStageEffectsEvent();
+
     // loopCount is used for statistics and diagnostics.
     for (int64_t loopCount = 0; !exitPending(); ++loopCount)
     {
@@ -3587,11 +3661,18 @@
             }
         }
 
+        if (mCheckOutputStageEffects.exchange(false)) {
+            checkOutputStageEffects();
+        }
+
         { // scope for mLock
 
             Mutex::Autolock _l(mLock);
 
             processConfigEvents_l();
+            if (mCheckOutputStageEffects.load()) {
+                continue;
+            }
 
             // See comment at declaration of logString for why this is done under mLock
             if (logString != NULL) {
@@ -3757,6 +3838,8 @@
             if (mMixerBufferValid) {
                 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer;
                 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat;
+                uint32_t channelCount = mEffectBufferValid ?
+                            audio_channel_count_from_out_mask(mMixerChannelMask) : mChannelCount;
 
                 // mono blend occurs for mixer threads only (not direct or offloaded)
                 // and is handled here if we're going directly to the sink.
@@ -3774,7 +3857,7 @@
                 }
 
                 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat,
-                        mNormalFrameCount * (mChannelCount + mHapticChannelCount));
+                        mNormalFrameCount * (channelCount + mHapticChannelCount));
 
                 // If we're going directly to the sink and there are haptic channels,
                 // we should adjust channels as the sample data is partially interleaved
@@ -4448,8 +4531,8 @@
 // ----------------------------------------------------------------------------
 
 AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
-        audio_io_handle_t id, bool systemReady, type_t type)
-    :   PlaybackThread(audioFlinger, output, id, type, systemReady),
+        audio_io_handle_t id, bool systemReady, type_t type, audio_config_base_t *mixerConfig)
+    :   PlaybackThread(audioFlinger, output, id, type, systemReady, mixerConfig),
         // mAudioMixer below
         // mFastMixer below
         mFastMixerFutex(0),
@@ -4566,6 +4649,7 @@
         fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer
         fastTrack->mHapticPlaybackEnabled = mHapticChannelMask != AUDIO_CHANNEL_NONE;
         fastTrack->mHapticIntensity = os::HapticScale::NONE;
+        fastTrack->mHapticMaxAmplitude = NAN;
         fastTrack->mGeneration++;
         state->mFastTracksGen++;
         state->mTrackMask = 1;
@@ -5103,6 +5187,7 @@
                     fastTrack->mFormat = track->mFormat;
                     fastTrack->mHapticPlaybackEnabled = track->getHapticPlaybackEnabled();
                     fastTrack->mHapticIntensity = track->getHapticIntensity();
+                    fastTrack->mHapticMaxAmplitude = track->getHapticMaxAmplitude();
                     fastTrack->mGeneration++;
                     state->mTrackMask |= 1 << j;
                     didModify = true;
@@ -5356,7 +5441,7 @@
                 trackId,
                 AudioMixer::TRACK,
                 AudioMixer::MIXER_CHANNEL_MASK,
-                (void *)(uintptr_t)(mChannelMask | mHapticChannelMask));
+                (void *)(uintptr_t)(mMixerChannelMask | mHapticChannelMask));
             // limit track sample rate to 2 x output sample rate, which changes at re-configuration
             uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX;
             uint32_t reqSampleRate = proxy->getSampleRate();
@@ -5425,6 +5510,10 @@
                 trackId,
                 AudioMixer::TRACK,
                 AudioMixer::HAPTIC_INTENSITY, (void *)(uintptr_t)track->getHapticIntensity());
+            mAudioMixer->setParameter(
+                trackId,
+                AudioMixer::TRACK,
+                AudioMixer::HAPTIC_MAX_AMPLITUDE, (void *)(&(track->mHapticMaxAmplitude)));
 
             // reset retry count
             track->mRetryCount = kMaxTrackRetries;
@@ -5575,7 +5664,8 @@
     // remove all the tracks that need to be...
     removeTracks_l(*tracksToRemove);
 
-    if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) {
+    if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0 ||
+            getEffectChain_l(AUDIO_SESSION_OUTPUT_STAGE) != 0) {
         mEffectBufferValid = true;
     }
 
@@ -6971,6 +7061,69 @@
     MixerThread::cacheParameters_l();
 }
 
+// ----------------------------------------------------------------------------
+
+AudioFlinger::VirtualizerStageThread::VirtualizerStageThread(const sp<AudioFlinger>& audioFlinger,
+                                                             AudioStreamOut* output,
+                                                             audio_io_handle_t id,
+                                                             bool systemReady,
+                                                             audio_config_base_t *mixerConfig)
+    : MixerThread(audioFlinger, output, id, systemReady, VIRTUALIZER_STAGE, mixerConfig)
+{
+}
+
+void AudioFlinger::VirtualizerStageThread::checkOutputStageEffects()
+{
+    bool hasVirtualizer = false;
+    bool hasDownMixer = false;
+    sp<EffectHandle> finalDownMixer;
+    {
+        Mutex::Autolock _l(mLock);
+        sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_STAGE);
+        if (chain != 0) {
+            hasVirtualizer = chain->getEffectFromType_l(FX_IID_VIRTUALIZER_STAGE) != nullptr;
+            hasDownMixer = chain->getEffectFromType_l(EFFECT_UIID_DOWNMIX) != nullptr;
+        }
+
+        finalDownMixer = mFinalDownMixer;
+        mFinalDownMixer.clear();
+    }
+
+    if (hasVirtualizer) {
+        if (finalDownMixer != nullptr) {
+            int32_t ret;
+            finalDownMixer->disable(&ret);
+        }
+        finalDownMixer.clear();
+    } else if (!hasDownMixer) {
+        std::vector<effect_descriptor_t> descriptors;
+        status_t status = mAudioFlinger->mEffectsFactoryHal->getDescriptors(
+                                                        EFFECT_UIID_DOWNMIX, &descriptors);
+        if (status != NO_ERROR) {
+            return;
+        }
+        ALOG_ASSERT(!descriptors.empty(),
+                "%s getDescriptors() returned no error but empty list", __func__);
+
+        finalDownMixer = createEffect_l(nullptr /*client*/, nullptr /*effectClient*/,
+                0 /*priority*/, AUDIO_SESSION_OUTPUT_STAGE, &descriptors[0], nullptr /*enabled*/,
+                &status, false /*pinned*/, false /*probe*/);
+
+        if (finalDownMixer == nullptr || (status != NO_ERROR && status != ALREADY_EXISTS)) {
+            ALOGW("%s error creating downmixer %d", __func__, status);
+            finalDownMixer.clear();
+        } else {
+            int32_t ret;
+            finalDownMixer->enable(&ret);
+        }
+    }
+
+    {
+        Mutex::Autolock _l(mLock);
+        mFinalDownMixer = finalDownMixer;
+    }
+}
+
 
 // ----------------------------------------------------------------------------
 //      Record
@@ -9278,7 +9431,7 @@
     mActiveTracks.add(track);
     sp<EffectChain> chain = getEffectChain_l(mSessionId);
     if (chain != 0) {
-        chain->setStrategy(AudioSystem::getStrategyForStream(streamType()));
+        chain->setStrategy(getStrategyForStream(streamType()));
         chain->incTrackCnt();
         chain->incActiveTrackCnt();
     }
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 16082a9..3001863 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -32,6 +32,7 @@
         OFFLOAD,            // Thread class is OffloadThread
         MMAP_PLAYBACK,      // Thread class for MMAP playback stream
         MMAP_CAPTURE,       // Thread class for MMAP capture stream
+        VIRTUALIZER_STAGE,  //
         // If you add any values here, also update ThreadBase::threadTypeToString()
     };
 
@@ -53,7 +54,8 @@
         CFG_EVENT_CREATE_AUDIO_PATCH,
         CFG_EVENT_RELEASE_AUDIO_PATCH,
         CFG_EVENT_UPDATE_OUT_DEVICE,
-        CFG_EVENT_RESIZE_BUFFER
+        CFG_EVENT_RESIZE_BUFFER,
+        CFG_EVENT_CHECK_OUTPUT_STAGE_EFFECTS
     };
 
     class ConfigEventData: public RefBase {
@@ -87,7 +89,13 @@
     public:
         virtual ~ConfigEvent() {}
 
-        void dump(char *buffer, size_t size) { mData->dump(buffer, size); }
+        void dump(char *buffer, size_t size) {
+            snprintf(buffer, size, "Event type: %d\n", mType);
+            if (mData != nullptr) {
+                snprintf(buffer, size, "Data:\n");
+                mData->dump(buffer, size);
+            }
+        }
 
         const int mType; // event type e.g. CFG_EVENT_IO
         Mutex mLock;     // mutex associated with mCond
@@ -110,7 +118,7 @@
             mEvent(event), mPid(pid), mPortId(portId) {}
 
         virtual  void dump(char *buffer, size_t size) {
-            snprintf(buffer, size, "IO event: event %d\n", mEvent);
+            snprintf(buffer, size, "- IO event: event %d\n", mEvent);
         }
 
         const audio_io_config_event mEvent;
@@ -133,7 +141,7 @@
             mPid(pid), mTid(tid), mPrio(prio), mForApp(forApp) {}
 
         virtual  void dump(char *buffer, size_t size) {
-            snprintf(buffer, size, "Prio event: pid %d, tid %d, prio %d, for app? %d\n",
+            snprintf(buffer, size, "- Prio event: pid %d, tid %d, prio %d, for app? %d\n",
                     mPid, mTid, mPrio, mForApp);
         }
 
@@ -158,7 +166,7 @@
             mKeyValuePairs(keyValuePairs) {}
 
         virtual  void dump(char *buffer, size_t size) {
-            snprintf(buffer, size, "KeyValue: %s\n", mKeyValuePairs.string());
+            snprintf(buffer, size, "- KeyValue: %s\n", mKeyValuePairs.string());
         }
 
         const String8 mKeyValuePairs;
@@ -181,7 +189,7 @@
             mPatch(patch), mHandle(handle) {}
 
         virtual  void dump(char *buffer, size_t size) {
-            snprintf(buffer, size, "Patch handle: %u\n", mHandle);
+            snprintf(buffer, size, "- Patch handle: %u\n", mHandle);
         }
 
         const struct audio_patch mPatch;
@@ -205,7 +213,7 @@
             mHandle(handle) {}
 
         virtual  void dump(char *buffer, size_t size) {
-            snprintf(buffer, size, "Patch handle: %u\n", mHandle);
+            snprintf(buffer, size, "- Patch handle: %u\n", mHandle);
         }
 
         audio_patch_handle_t mHandle;
@@ -227,7 +235,7 @@
             mOutDevices(outDevices) {}
 
         virtual void dump(char *buffer, size_t size) {
-            snprintf(buffer, size, "Devices: %s", android::toString(mOutDevices).c_str());
+            snprintf(buffer, size, "- Devices: %s", android::toString(mOutDevices).c_str());
         }
 
         DeviceDescriptorBaseVector mOutDevices;
@@ -249,7 +257,7 @@
             mMaxSharedAudioHistoryMs(maxSharedAudioHistoryMs) {}
 
         virtual void dump(char *buffer, size_t size) {
-            snprintf(buffer, size, "mMaxSharedAudioHistoryMs: %d", mMaxSharedAudioHistoryMs);
+            snprintf(buffer, size, "- mMaxSharedAudioHistoryMs: %d", mMaxSharedAudioHistoryMs);
         }
 
         int32_t mMaxSharedAudioHistoryMs;
@@ -265,6 +273,16 @@
         virtual ~ResizeBufferConfigEvent() {}
     };
 
+    class CheckOutputStageEffectsEvent : public ConfigEvent {
+    public:
+        CheckOutputStageEffectsEvent() :
+            ConfigEvent(CFG_EVENT_CHECK_OUTPUT_STAGE_EFFECTS) {
+        }
+
+        virtual ~CheckOutputStageEffectsEvent() {}
+    };
+
+
     class PMDeathRecipient : public IBinder::DeathRecipient {
     public:
         explicit    PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {}
@@ -290,8 +308,11 @@
                 // dynamic externally-visible
                 uint32_t    sampleRate() const { return mSampleRate; }
                 audio_channel_mask_t channelMask() const { return mChannelMask; }
+    virtual     audio_channel_mask_t mixerChannelMask() const { return mChannelMask; }
+
                 audio_format_t format() const { return mHALFormat; }
                 uint32_t channelCount() const { return mChannelCount; }
+
                 // Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects,
                 // and returns the [normal mix] buffer's frame count.
     virtual     size_t      frameCount() const = 0;
@@ -330,7 +351,11 @@
                 status_t    sendUpdateOutDeviceConfigEvent(
                                     const DeviceDescriptorBaseVector& outDevices);
                 void        sendResizeBufferConfigEvent_l(int32_t maxSharedAudioHistoryMs);
+                void        sendCheckOutputStageEffectsEvent();
+                void        sendCheckOutputStageEffectsEvent_l();
+
                 void        processConfigEvents_l();
+    virtual     void        setCheckOutputStageEffects() {}
     virtual     void        cacheParameters_l() = 0;
     virtual     status_t    createAudioPatch_l(const struct audio_patch *patch,
                                                audio_patch_handle_t *handle) = 0;
@@ -574,6 +599,8 @@
                                 return INVALID_OPERATION;
                             }
 
+                product_strategy_t getStrategyForStream(audio_stream_type_t stream) const;
+
     virtual     void        dumpInternals_l(int fd __unused, const Vector<String16>& args __unused)
                             { }
     virtual     void        dumpTracks_l(int fd __unused, const Vector<String16>& args __unused) { }
@@ -824,7 +851,8 @@
     static const nsecs_t kMaxNextBufferDelayNs = 100000000;
 
     PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
-                   audio_io_handle_t id, type_t type, bool systemReady);
+                   audio_io_handle_t id, type_t type, bool systemReady,
+                   audio_config_base_t *mixerConfig = nullptr);
     virtual             ~PlaybackThread();
 
     // Thread virtuals
@@ -881,6 +909,8 @@
                                 mActiveTracks.updatePowerState(this, true /* force */);
                             }
 
+    virtual     void        checkOutputStageEffects() {}
+
                 void        dumpInternals_l(int fd, const Vector<String16>& args) override;
                 void        dumpTracks_l(int fd, const Vector<String16>& args) override;
 
@@ -973,6 +1003,10 @@
 
     virtual     size_t      frameCount() const { return mNormalFrameCount; }
 
+                audio_channel_mask_t mixerChannelMask() const override {
+                    return mMixerChannelMask;
+                }
+
                 status_t    getTimestamp_l(AudioTimestamp& timestamp);
 
                 void        addPatchTrack(const sp<PatchTrack>& track);
@@ -1015,6 +1049,9 @@
 
                 PlaybackThread::Track* getTrackById_l(audio_port_handle_t trackId);
 
+                bool hasMixer() const {
+                    return mType == MIXER || mType == DUPLICATING || mType == VIRTUALIZER_STAGE;
+                }
 protected:
     // updated by readOutputParameters_l()
     size_t                          mNormalFrameCount;  // normal mixer and effects
@@ -1101,6 +1138,9 @@
     // haptic playback.
     audio_channel_mask_t            mHapticChannelMask = AUDIO_CHANNEL_NONE;
     uint32_t                        mHapticChannelCount = 0;
+
+    audio_channel_mask_t            mMixerChannelMask = AUDIO_CHANNEL_NONE;
+
 private:
     // mMasterMute is in both PlaybackThread and in AudioFlinger.  When a
     // PlaybackThread needs to find out if master-muted, it checks it's local
@@ -1134,6 +1174,9 @@
 
     // Cache various calculated values, at threadLoop() entry and after a parameter change
     virtual     void        cacheParameters_l();
+                void        setCheckOutputStageEffects() override {
+                                mCheckOutputStageEffects.store(true);
+                            }
 
     virtual     uint32_t    correctLatency_l(uint32_t latency) const;
 
@@ -1314,6 +1357,8 @@
                 // audio patch used by the downstream software patch.
                 // Only used if ThreadBase::mIsMsdDevice is true.
                 struct audio_patch mDownStreamPatch;
+
+                std::atomic_bool mCheckOutputStageEffects{};
 };
 
 class MixerThread : public PlaybackThread {
@@ -1322,7 +1367,8 @@
                 AudioStreamOut* output,
                 audio_io_handle_t id,
                 bool systemReady,
-                type_t type = MIXER);
+                type_t type = MIXER,
+                audio_config_base_t *mixerConfig = nullptr);
     virtual             ~MixerThread();
 
     // Thread virtuals
@@ -1611,6 +1657,24 @@
     }
 };
 
+class VirtualizerStageThread : public MixerThread {
+public:
+    VirtualizerStageThread(const sp<AudioFlinger>& audioFlinger,
+                           AudioStreamOut* output,
+                           audio_io_handle_t id,
+                           bool systemReady,
+                           audio_config_base_t *mixerConfig);
+            ~VirtualizerStageThread() override {}
+
+            bool hasFastMixer() const override { return false; }
+
+protected:
+            void checkOutputStageEffects() override;
+
+private:
+            sp<EffectHandle> mFinalDownMixer;
+};
+
 // record thread
 class RecordThread : public ThreadBase
 {
diff --git a/services/audiopolicy/AudioPolicyInterface.h b/services/audiopolicy/AudioPolicyInterface.h
index 2e49e71..6b11d9a 100644
--- a/services/audiopolicy/AudioPolicyInterface.h
+++ b/services/audiopolicy/AudioPolicyInterface.h
@@ -359,7 +359,8 @@
     // The audio policy manager can check if the proposed parameters are suitable or not and act accordingly.
     virtual status_t openOutput(audio_module_handle_t module,
                                 audio_io_handle_t *output,
-                                audio_config_t *config,
+                                audio_config_t *halConfig,
+                                audio_config_base_t *mixerConfig,
                                 const sp<DeviceDescriptorBase>& device,
                                 uint32_t *latencyMs,
                                 audio_output_flags_t flags) = 0;
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
index 1f9b535..7c7f02d 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
@@ -362,7 +362,8 @@
                            const struct audio_port_config *srcConfig = NULL) const;
     virtual void toAudioPort(struct audio_port_v7 *port) const;
 
-        status_t open(const audio_config_t *config,
+        status_t open(const audio_config_t *halConfig,
+                      const audio_config_base_t *mixerConfig,
                       const DeviceVector &devices,
                       audio_stream_type_t stream,
                       audio_output_flags_t flags,
@@ -423,6 +424,7 @@
     uint32_t mDirectOpenCount; // number of clients using this output (direct outputs only)
     audio_session_t mDirectClientSession; // session id of the direct output client
     bool mPendingReopenToQueryProfiles = false;
+    audio_channel_mask_t mMixerChannelMask = AUDIO_CHANNEL_NONE;
 };
 
 // Audio output driven by an input device directly.
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioPolicyConfig.h b/services/audiopolicy/common/managerdefinitions/include/AudioPolicyConfig.h
index cf1f64c..9837336 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioPolicyConfig.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioPolicyConfig.h
@@ -202,6 +202,20 @@
             {AUDIO_FORMAT_AC4, {}}};
     }
 
+    //TODO: b/193496180 use virtualizer stage flag at audio HAL when available
+    // until then, use DEEP_BUFFER+FAST flag combo to indicate the virtualizer stage output profile
+    void convertVirtualizerStageFlag()
+    {
+        for (const auto& hwModule : mHwModules) {
+            for (const auto& curProfile : hwModule->getOutputProfiles()) {
+                if (curProfile->getFlags()
+                        == (AUDIO_OUTPUT_FLAG_FAST | AUDIO_OUTPUT_FLAG_DEEP_BUFFER)) {
+                    curProfile->setFlags(AUDIO_OUTPUT_FLAG_VIRTUALIZER_STAGE);
+                }
+            }
+        }
+    }
+
 private:
     static const constexpr char* const kDefaultEngineLibraryNameSuffix = "default";
 
diff --git a/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
index 20b4044..58d05c6 100644
--- a/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
@@ -168,6 +168,10 @@
     DeviceVector getDevicesFromDeviceTypeAddrVec(
             const AudioDeviceTypeAddrVector& deviceTypeAddrVector) const;
 
+    // Return the device vector that contains device descriptor whose AudioDeviceTypeAddr appears
+    // in the given AudioDeviceTypeAddrVector
+    AudioDeviceTypeAddrVector toTypeAddrVector() const;
+
     // If there are devices with the given type and the devices to add is not empty,
     // remove all the devices with the given type and add all the devices to add.
     void replaceDevicesByType(audio_devices_t typeToRemove, const DeviceVector &devicesToAdd);
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
index 6b08f7c..6c3386f 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
@@ -491,7 +491,8 @@
     return true;
 }
 
-status_t SwAudioOutputDescriptor::open(const audio_config_t *config,
+status_t SwAudioOutputDescriptor::open(const audio_config_t *halConfig,
+                                       const audio_config_base_t *mixerConfig,
                                        const DeviceVector &devices,
                                        audio_stream_type_t stream,
                                        audio_output_flags_t flags,
@@ -504,45 +505,62 @@
                         "with the requested devices, all device types: %s",
                         __func__, dumpDeviceTypes(devices.types()).c_str());
 
-    audio_config_t lConfig;
-    if (config == nullptr) {
-        lConfig = AUDIO_CONFIG_INITIALIZER;
-        lConfig.sample_rate = mSamplingRate;
-        lConfig.channel_mask = mChannelMask;
-        lConfig.format = mFormat;
+    audio_config_t lHalConfig;
+    if (halConfig == nullptr) {
+        lHalConfig = AUDIO_CONFIG_INITIALIZER;
+        lHalConfig.sample_rate = mSamplingRate;
+        lHalConfig.channel_mask = mChannelMask;
+        lHalConfig.format = mFormat;
     } else {
-        lConfig = *config;
+        lHalConfig = *halConfig;
     }
 
     // if the selected profile is offloaded and no offload info was specified,
     // create a default one
     if ((mProfile->getFlags() & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) &&
-            lConfig.offload_info.format == AUDIO_FORMAT_DEFAULT) {
+            lHalConfig.offload_info.format == AUDIO_FORMAT_DEFAULT) {
         flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
-        lConfig.offload_info = AUDIO_INFO_INITIALIZER;
-        lConfig.offload_info.sample_rate = lConfig.sample_rate;
-        lConfig.offload_info.channel_mask = lConfig.channel_mask;
-        lConfig.offload_info.format = lConfig.format;
-        lConfig.offload_info.stream_type = stream;
-        lConfig.offload_info.duration_us = -1;
-        lConfig.offload_info.has_video = true; // conservative
-        lConfig.offload_info.is_streaming = true; // likely
-        lConfig.offload_info.encapsulation_mode = lConfig.offload_info.encapsulation_mode;
-        lConfig.offload_info.content_id = lConfig.offload_info.content_id;
-        lConfig.offload_info.sync_id = lConfig.offload_info.sync_id;
+        lHalConfig.offload_info = AUDIO_INFO_INITIALIZER;
+        lHalConfig.offload_info.sample_rate = lHalConfig.sample_rate;
+        lHalConfig.offload_info.channel_mask = lHalConfig.channel_mask;
+        lHalConfig.offload_info.format = lHalConfig.format;
+        lHalConfig.offload_info.stream_type = stream;
+        lHalConfig.offload_info.duration_us = -1;
+        lHalConfig.offload_info.has_video = true; // conservative
+        lHalConfig.offload_info.is_streaming = true; // likely
+        lHalConfig.offload_info.encapsulation_mode = lHalConfig.offload_info.encapsulation_mode;
+        lHalConfig.offload_info.content_id = lHalConfig.offload_info.content_id;
+        lHalConfig.offload_info.sync_id = lHalConfig.offload_info.sync_id;
+    }
+
+    audio_config_base_t lMixerConfig;
+    if (mixerConfig == nullptr) {
+        lMixerConfig = AUDIO_CONFIG_BASE_INITIALIZER;
+        lMixerConfig.sample_rate = lHalConfig.sample_rate;
+        lMixerConfig.channel_mask = lHalConfig.channel_mask;
+        lMixerConfig.format = lHalConfig.format;
+    } else {
+        lMixerConfig = *mixerConfig;
     }
 
     mFlags = (audio_output_flags_t)(mFlags | flags);
 
+    //TODO: b/193496180 use virtualizer stage flag at audio HAL when available
+    audio_output_flags_t halFlags = mFlags;
+    if ((mFlags & AUDIO_OUTPUT_FLAG_VIRTUALIZER_STAGE) != 0) {
+        halFlags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_FAST | AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
+    }
+
     ALOGV("opening output for device %s profile %p name %s",
           mDevices.toString().c_str(), mProfile.get(), mProfile->getName().c_str());
 
     status_t status = mClientInterface->openOutput(mProfile->getModuleHandle(),
                                                    output,
-                                                   &lConfig,
+                                                   &lHalConfig,
+                                                   &lMixerConfig,
                                                    device,
                                                    &mLatency,
-                                                   mFlags);
+                                                   halFlags);
 
     if (status == NO_ERROR) {
         LOG_ALWAYS_FATAL_IF(*output == AUDIO_IO_HANDLE_NONE,
@@ -550,9 +568,10 @@
                             "selected device %s for opening",
                             __FUNCTION__, *output, devices.toString().c_str(),
                             device->toString().c_str());
-        mSamplingRate = lConfig.sample_rate;
-        mChannelMask = lConfig.channel_mask;
-        mFormat = lConfig.format;
+        mSamplingRate = lHalConfig.sample_rate;
+        mChannelMask = lHalConfig.channel_mask;
+        mFormat = lHalConfig.format;
+        mMixerChannelMask = lMixerConfig.channel_mask;
         mId = PolicyAudioPort::getNextUniqueId();
         mIoHandle = *output;
         mProfile->curOpenCount++;
diff --git a/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
index a92d31e..1722032 100644
--- a/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
@@ -451,6 +451,14 @@
     return devices;
 }
 
+AudioDeviceTypeAddrVector DeviceVector::toTypeAddrVector() const {
+    AudioDeviceTypeAddrVector result;
+    for (const auto& device : *this) {
+        result.push_back(AudioDeviceTypeAddr(device->type(), device->address()));
+    }
+    return result;
+}
+
 void DeviceVector::replaceDevicesByType(
         audio_devices_t typeToRemove, const DeviceVector &devicesToAdd) {
     DeviceVector devicesToRemove = getDevicesFromType(typeToRemove);
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index cc2d8e8..a20612e 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -246,9 +246,11 @@
                     sp<SwAudioOutputDescriptor> desc = mOutputs.valueFor(output);
                     // close unused outputs after device disconnection or direct outputs that have
                     // been opened by checkOutputsForDevice() to query dynamic parameters
-                    if ((state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) ||
-                            (((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) &&
-                                (desc->mDirectOpenCount == 0))) {
+                    if ((state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE)
+                            || (((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) &&
+                                (desc->mDirectOpenCount == 0))
+                            || (((desc->mFlags & AUDIO_OUTPUT_FLAG_VIRTUALIZER_STAGE) != 0) &&
+                                (desc != mVirtualizerStageOutput))) {
                         clearAudioSourcesForOutput(output);
                         closeOutput(output);
                     }
@@ -925,6 +927,36 @@
     return profile;
 }
 
+sp<IOProfile> AudioPolicyManager::getVirtualizerStageOutputProfile(
+        const audio_config_t *config __unused, const AudioDeviceTypeAddrVector &devices,
+        bool forOpening) const
+{
+    for (const auto& hwModule : mHwModules) {
+        for (const auto& curProfile : hwModule->getOutputProfiles()) {
+            if (curProfile->getFlags() != AUDIO_OUTPUT_FLAG_VIRTUALIZER_STAGE) {
+                continue;
+            }
+            // reject profiles not corresponding to a device currently available
+            DeviceVector supportedDevices = curProfile->getSupportedDevices();
+            if (!mAvailableOutputDevices.containsAtLeastOne(supportedDevices)) {
+                continue;
+            }
+            if (!devices.empty()) {
+                if (supportedDevices.getDevicesFromDeviceTypeAddrVec(devices).size()
+                        != devices.size()) {
+                    continue;
+                }
+            }
+            if (forOpening && !curProfile->canOpenNewIo()) {
+                continue;
+            }
+            ALOGV("%s found profile %s", __func__, curProfile->getName().c_str());
+            return curProfile;
+        }
+    }
+    return nullptr;
+}
+
 audio_io_handle_t AudioPolicyManager::getOutput(audio_stream_type_t stream)
 {
     DeviceVector devices = mEngine->getOutputDevicesForStream(stream, false /*fromCache*/);
@@ -1094,7 +1126,7 @@
 
     *output = AUDIO_IO_HANDLE_NONE;
     if (!msdDevices.isEmpty()) {
-        *output = getOutputForDevices(msdDevices, session, *stream, config, flags);
+        *output = getOutputForDevices(msdDevices, session, resultAttr, config, flags);
         if (*output != AUDIO_IO_HANDLE_NONE && setMsdOutputPatches(&outputDevices) == NO_ERROR) {
             ALOGV("%s() Using MSD devices %s instead of devices %s",
                   __func__, msdDevices.toString().c_str(), outputDevices.toString().c_str());
@@ -1103,7 +1135,7 @@
         }
     }
     if (*output == AUDIO_IO_HANDLE_NONE) {
-        *output = getOutputForDevices(outputDevices, session, *stream, config,
+        *output = getOutputForDevices(outputDevices, session, resultAttr, config,
                 flags, resultAttr->flags & AUDIO_FLAG_MUTE_HAPTIC);
     }
     if (*output == AUDIO_IO_HANDLE_NONE) {
@@ -1265,7 +1297,8 @@
     // all MSD patches to prioritize this request over any active output on MSD.
     releaseMsdOutputPatches(devices);
 
-    status_t status = outputDesc->open(config, devices, stream, flags, output);
+    status_t status =
+            outputDesc->open(config, nullptr /* mixerConfig */, devices, stream, flags, output);
 
     // only accept an output with the requested parameters
     if (status != NO_ERROR ||
@@ -1300,7 +1333,7 @@
 audio_io_handle_t AudioPolicyManager::getOutputForDevices(
         const DeviceVector &devices,
         audio_session_t session,
-        audio_stream_type_t stream,
+        const audio_attributes_t *attr,
         const audio_config_t *config,
         audio_output_flags_t *flags,
         bool forceMutingHaptic)
@@ -1322,6 +1355,9 @@
     if ((*flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) {
         *flags = (audio_output_flags_t)(*flags | AUDIO_OUTPUT_FLAG_DIRECT);
     }
+
+    audio_stream_type_t stream = mEngine->getStreamTypeForAttributes(*attr);
+
     // only allow deep buffering for music stream type
     if (stream != AUDIO_STREAM_MUSIC) {
         *flags = (audio_output_flags_t)(*flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
@@ -1341,6 +1377,11 @@
         ALOGV("Set VoIP and Direct output flags for PCM format");
     }
 
+    if (mVirtualizerStageOutput != nullptr
+            && canBeVirtualized(attr, config, devices.toTypeAddrVector())) {
+        return mVirtualizerStageOutput->mIoHandle;
+    }
+
     audio_config_t directConfig = *config;
     directConfig.channel_mask = channelMask;
     status_t status = openDirectOutput(stream, session, &directConfig, *flags, devices, &output);
@@ -4802,6 +4843,136 @@
     return source;
 }
 
+bool AudioPolicyManager::canBeVirtualized(const audio_attributes_t *attr,
+                                      const audio_config_t *config,
+                                      const AudioDeviceTypeAddrVector &devices)  const
+{
+    // The caller can have the audio attributes criteria ignored by either passing a null ptr or
+    // the AUDIO_ATTRIBUTES_INITIALIZER value.
+    // If attributes are specified, current policy is to only allow virtualization for media
+    // and game usages.
+    if (attr != nullptr && *attr != AUDIO_ATTRIBUTES_INITIALIZER &&
+            attr->usage != AUDIO_USAGE_MEDIA && attr->usage != AUDIO_USAGE_GAME) {
+        return false;
+    }
+
+    // The caller can have the devices criteria ignored by passing and empty vector, and
+    // getVirtualizerStageOutputProfile() will ignore the devices when looking for a match.
+    // Otherwise an output profile supporting a virtualizer stage effect that can be routed
+    // to the specified devices must exist.
+    sp<IOProfile> profile =
+            getVirtualizerStageOutputProfile(config, devices, false /*forOpening*/);
+    if (profile == nullptr) {
+        return false;
+    }
+
+    // The caller can have the audio config criteria ignored by either passing a null ptr or
+    // the AUDIO_CONFIG_INITIALIZER value.
+    // If an audio config is specified, current policy is to only allow virtualization for
+    // 5.1, 7.1and 7.1.4 audio.
+    // If the virtualizer stage output is already opened, only channel masks included in the
+    // virtualizer stage output mixer channel mask are allowed.
+    if (config != nullptr && *config != AUDIO_CONFIG_INITIALIZER) {
+        if (config->channel_mask != AUDIO_CHANNEL_OUT_5POINT1
+                && config->channel_mask != AUDIO_CHANNEL_OUT_7POINT1
+                && config->channel_mask != AUDIO_CHANNEL_OUT_7POINT1POINT4) {
+            return false;
+        }
+        if (mVirtualizerStageOutput != nullptr) {
+            if ((config->channel_mask & mVirtualizerStageOutput->mMixerChannelMask)
+                    != config->channel_mask) {
+                return false;
+            }
+        }
+    }
+
+    return true;
+}
+
+void AudioPolicyManager::checkVirtualizerClientRoutes() {
+    std::set<audio_stream_type_t> streamsToInvalidate;
+    for (size_t i = 0; i < mOutputs.size(); i++) {
+        const sp<SwAudioOutputDescriptor>& outputDescriptor = mOutputs[i];
+        for (const sp<TrackClientDescriptor>& client : outputDescriptor->getClientIterable()) {
+            audio_attributes_t attr = client->attributes();
+            DeviceVector devices = mEngine->getOutputDevicesForAttributes(attr, nullptr, false);
+            AudioDeviceTypeAddrVector devicesTypeAddress = devices.toTypeAddrVector();
+            audio_config_base_t clientConfig = client->config();
+            audio_config_t config = audio_config_initializer(&clientConfig);
+            if (canBeVirtualized(&attr, &config, devicesTypeAddress)) {
+                streamsToInvalidate.insert(client->stream());
+            }
+        }
+    }
+
+    for (audio_stream_type_t stream : streamsToInvalidate) {
+        mpClientInterface->invalidateStream(stream);
+    }
+}
+
+status_t AudioPolicyManager::getVirtualizerStageOutput(const audio_config_base_t *mixerConfig,
+                                                        const audio_attributes_t *attr,
+                                                        audio_io_handle_t *output) {
+    *output = AUDIO_IO_HANDLE_NONE;
+
+    if (mVirtualizerStageOutput != nullptr) {
+        return INVALID_OPERATION;
+    }
+
+    DeviceVector devices = mEngine->getOutputDevicesForAttributes(*attr, nullptr, false);
+    AudioDeviceTypeAddrVector devicesTypeAddress = devices.toTypeAddrVector();
+    audio_config_t *configPtr = nullptr;
+    audio_config_t config;
+    if (mixerConfig != nullptr) {
+        config = audio_config_initializer(mixerConfig);
+        configPtr = &config;
+    }
+    if (!canBeVirtualized(attr, configPtr, devicesTypeAddress)) {
+        return BAD_VALUE;
+    }
+
+    sp<IOProfile> profile =
+            getVirtualizerStageOutputProfile(configPtr, devicesTypeAddress, true /*forOpening*/);
+    if (profile == nullptr) {
+        return BAD_VALUE;
+    }
+
+    mVirtualizerStageOutput = new SwAudioOutputDescriptor(profile, mpClientInterface);
+    status_t status = mVirtualizerStageOutput->open(nullptr, mixerConfig, devices,
+                                                    mEngine->getStreamTypeForAttributes(*attr),
+                                                    AUDIO_OUTPUT_FLAG_VIRTUALIZER_STAGE, output);
+    if (status != NO_ERROR) {
+        ALOGV("%s failed opening output: status %d, output %d", __func__, status, *output);
+        if (*output != AUDIO_IO_HANDLE_NONE) {
+            mVirtualizerStageOutput->close();
+        }
+        mVirtualizerStageOutput.clear();
+        *output = AUDIO_IO_HANDLE_NONE;
+        return status;
+    }
+
+    checkVirtualizerClientRoutes();
+
+    addOutput(*output, mVirtualizerStageOutput);
+    mPreviousOutputs = mOutputs;
+    mpClientInterface->onAudioPortListUpdate();
+
+    ALOGV("%s returns new virtualizer stage output %d", __func__, *output);
+    return NO_ERROR;
+}
+
+status_t AudioPolicyManager::releaseVirtualizerStageOutput(audio_io_handle_t output) {
+    if (mVirtualizerStageOutput == nullptr) {
+        return INVALID_OPERATION;
+    }
+    if (mVirtualizerStageOutput->mIoHandle != output) {
+        return BAD_VALUE;
+    }
+    closeOutput(output);
+    mVirtualizerStageOutput.clear();
+    return NO_ERROR;
+}
+
 // ----------------------------------------------------------------------------
 // AudioPolicyManager
 // ----------------------------------------------------------------------------
@@ -4851,6 +5022,8 @@
         ALOGE("could not load audio policy configuration file, setting defaults");
         getConfig().setDefault();
     }
+    //TODO: b/193496180 use virtualizer stage flag at audio HAL when available
+    getConfig().convertVirtualizerStageFlag();
 }
 
 status_t AudioPolicyManager::initialize() {
@@ -4990,7 +5163,8 @@
             sp<SwAudioOutputDescriptor> outputDesc = new SwAudioOutputDescriptor(outProfile,
                                                                                  mpClientInterface);
             audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
-            status_t status = outputDesc->open(nullptr, DeviceVector(supportedDevice),
+            status_t status = outputDesc->open(nullptr /* halConfig */, nullptr /* mixerConfig */,
+                                               DeviceVector(supportedDevice),
                                                AUDIO_STREAM_DEFAULT,
                                                AUDIO_OUTPUT_FLAG_NONE, &output);
             if (status != NO_ERROR) {
@@ -5012,7 +5186,8 @@
                     outProfile->getFlags() & AUDIO_OUTPUT_FLAG_PRIMARY) {
                 mPrimaryOutput = outputDesc;
             }
-            if ((outProfile->getFlags() & AUDIO_OUTPUT_FLAG_DIRECT) != 0) {
+            if ((outProfile->getFlags() & AUDIO_OUTPUT_FLAG_DIRECT) != 0
+                || (outProfile->getFlags() & AUDIO_OUTPUT_FLAG_VIRTUALIZER_STAGE) != 0 ) {
                 outputDesc->close();
             } else {
                 addOutput(output, outputDesc);
@@ -6995,7 +7170,7 @@
     }
     sp<SwAudioOutputDescriptor> desc = new SwAudioOutputDescriptor(profile, mpClientInterface);
     audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
-    status_t status = desc->open(nullptr, devices,
+    status_t status = desc->open(nullptr /* halConfig */, nullptr /* mixerConfig */, devices,
             AUDIO_STREAM_DEFAULT, AUDIO_OUTPUT_FLAG_NONE, &output);
     if (status != NO_ERROR) {
         return nullptr;
@@ -7025,7 +7200,7 @@
         config.offload_info.channel_mask = config.channel_mask;
         config.offload_info.format = config.format;
 
-        status = desc->open(&config, devices,
+        status = desc->open(&config, nullptr /* mixerConfig */, devices,
                             AUDIO_STREAM_DEFAULT, AUDIO_OUTPUT_FLAG_NONE, &output);
         if (status != NO_ERROR) {
             return nullptr;
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h
index 98f96d1..8668f5e 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.h
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h
@@ -356,6 +356,16 @@
                     BAD_VALUE : NO_ERROR;
         }
 
+        virtual bool canBeVirtualized(const audio_attributes_t *attr,
+                                      const audio_config_t *config,
+                                      const AudioDeviceTypeAddrVector &devices) const;
+
+        virtual status_t getVirtualizerStageOutput(const audio_config_base_t *config,
+                                                const audio_attributes_t *attr,
+                                                audio_io_handle_t *output);
+
+        virtual status_t releaseVirtualizerStageOutput(audio_io_handle_t output);
+
         bool isCallScreenModeSupported() override;
 
         void onNewAudioModulesAvailable() override;
@@ -797,6 +807,8 @@
         sp<SwAudioOutputDescriptor> mPrimaryOutput;     // primary output descriptor
         // list of descriptors for outputs currently opened
 
+        sp<SwAudioOutputDescriptor> mVirtualizerStageOutput;
+
         SwAudioOutputCollection mOutputs;
         // copy of mOutputs before setDeviceConnectionState() opens new outputs
         // reset to mOutputs when updateDevicesAndOutputs() is called.
@@ -933,7 +945,7 @@
         audio_io_handle_t getOutputForDevices(
                 const DeviceVector &devices,
                 audio_session_t session,
-                audio_stream_type_t stream,
+                const audio_attributes_t *attr,
                 const audio_config_t *config,
                 audio_output_flags_t *flags,
                 bool forceMutingHaptic = false);
@@ -948,6 +960,13 @@
                 audio_output_flags_t flags,
                 const DeviceVector &devices,
                 audio_io_handle_t *output);
+
+        sp<IOProfile> getVirtualizerStageOutputProfile(const audio_config_t *config,
+                                                       const AudioDeviceTypeAddrVector &devices,
+                                                       bool forOpening) const;
+
+        void checkVirtualizerClientRoutes();
+
         /**
          * @brief getInputForDevice selects an input handle for a given input device and
          * requester context
diff --git a/services/audiopolicy/service/AudioPolicyClientImpl.cpp b/services/audiopolicy/service/AudioPolicyClientImpl.cpp
index cd53073..79252d4 100644
--- a/services/audiopolicy/service/AudioPolicyClientImpl.cpp
+++ b/services/audiopolicy/service/AudioPolicyClientImpl.cpp
@@ -40,7 +40,8 @@
 
 status_t AudioPolicyService::AudioPolicyClient::openOutput(audio_module_handle_t module,
                                                            audio_io_handle_t *output,
-                                                           audio_config_t *config,
+                                                           audio_config_t *halConfig,
+                                                           audio_config_base_t *mixerConfig,
                                                            const sp<DeviceDescriptorBase>& device,
                                                            uint32_t *latencyMs,
                                                            audio_output_flags_t flags)
@@ -55,14 +56,17 @@
     media::OpenOutputResponse response;
 
     request.module = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_module_handle_t_int32_t(module));
-    request.config = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_config_t_AudioConfig(*config));
+    request.halConfig = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_config_t_AudioConfig(*halConfig));
+    request.mixerConfig =
+            VALUE_OR_RETURN_STATUS(legacy2aidl_audio_config_base_t_AudioConfigBase(*mixerConfig));
     request.device = VALUE_OR_RETURN_STATUS(legacy2aidl_DeviceDescriptorBase(device));
     request.flags = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_output_flags_t_int32_t_mask(flags));
 
     status_t status = af->openOutput(request, &response);
     if (status == OK) {
         *output = VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_audio_io_handle_t(response.output));
-        *config = VALUE_OR_RETURN_STATUS(aidl2legacy_AudioConfig_audio_config_t(response.config));
+        *halConfig =
+                VALUE_OR_RETURN_STATUS(aidl2legacy_AudioConfig_audio_config_t(response.config));
         *latencyMs = VALUE_OR_RETURN_STATUS(convertIntegral<uint32_t>(response.latencyMs));
     }
     return status;
diff --git a/services/audiopolicy/service/AudioPolicyService.cpp b/services/audiopolicy/service/AudioPolicyService.cpp
index 4d0e1f1..3c757b3 100644
--- a/services/audiopolicy/service/AudioPolicyService.cpp
+++ b/services/audiopolicy/service/AudioPolicyService.cpp
@@ -127,6 +127,7 @@
         loadAudioPolicyManager();
         mAudioPolicyManager = mCreateAudioPolicyManager(mAudioPolicyClient);
     }
+
     // load audio processing modules
     sp<AudioPolicyEffects> audioPolicyEffects = new AudioPolicyEffects();
     sp<UidPolicy> uidPolicy = new UidPolicy(this);
@@ -139,6 +140,8 @@
     }
     uidPolicy->registerSelf();
     sensorPrivacyPolicy->registerSelf();
+
+    AudioSystem::audioPolicyReady();
 }
 
 void AudioPolicyService::unloadAudioPolicyManager()
diff --git a/services/audiopolicy/service/AudioPolicyService.h b/services/audiopolicy/service/AudioPolicyService.h
index 7ed829c..b03e35e 100644
--- a/services/audiopolicy/service/AudioPolicyService.h
+++ b/services/audiopolicy/service/AudioPolicyService.h
@@ -667,7 +667,8 @@
         // The audio policy manager can check if the proposed parameters are suitable or not and act accordingly.
         virtual status_t openOutput(audio_module_handle_t module,
                                     audio_io_handle_t *output,
-                                    audio_config_t *config,
+                                    audio_config_t *halConfig,
+                                    audio_config_base_t *mixerConfig,
                                     const sp<DeviceDescriptorBase>& device,
                                     uint32_t *latencyMs,
                                     audio_output_flags_t flags);
diff --git a/services/audiopolicy/tests/AudioPolicyManagerTestClient.h b/services/audiopolicy/tests/AudioPolicyManagerTestClient.h
index f7b0565..84b40d2 100644
--- a/services/audiopolicy/tests/AudioPolicyManagerTestClient.h
+++ b/services/audiopolicy/tests/AudioPolicyManagerTestClient.h
@@ -37,7 +37,8 @@
 
     status_t openOutput(audio_module_handle_t module,
                         audio_io_handle_t *output,
-                        audio_config_t * /*config*/,
+                        audio_config_t * /*halConfig*/,
+                        audio_config_base_t * /*mixerConfig*/,
                         const sp<DeviceDescriptorBase>& /*device*/,
                         uint32_t * /*latencyMs*/,
                         audio_output_flags_t /*flags*/) override {
diff --git a/services/audiopolicy/tests/AudioPolicyTestClient.h b/services/audiopolicy/tests/AudioPolicyTestClient.h
index 1384864..4e0735b 100644
--- a/services/audiopolicy/tests/AudioPolicyTestClient.h
+++ b/services/audiopolicy/tests/AudioPolicyTestClient.h
@@ -30,7 +30,8 @@
     }
     status_t openOutput(audio_module_handle_t /*module*/,
                         audio_io_handle_t* /*output*/,
-                        audio_config_t* /*config*/,
+                        audio_config_t* /*halConfig*/,
+                        audio_config_base_t* /*mixerConfig*/,
                         const sp<DeviceDescriptorBase>& /*device*/,
                         uint32_t* /*latencyMs*/,
                         audio_output_flags_t /*flags*/) override { return NO_INIT; }
diff --git a/services/mediametrics/Android.bp b/services/mediametrics/Android.bp
index 5989181..0351d2d 100644
--- a/services/mediametrics/Android.bp
+++ b/services/mediametrics/Android.bp
@@ -148,7 +148,8 @@
         "statsd_mediaparser.cpp",
         "statsd_nuplayer.cpp",
         "statsd_recorder.cpp",
-        "StringUtils.cpp"
+        "StringUtils.cpp",
+        "ValidateId.cpp",
     ],
 
     proto: {
diff --git a/services/mediametrics/AudioAnalytics.cpp b/services/mediametrics/AudioAnalytics.cpp
index 45c9f56..270fe2f 100644
--- a/services/mediametrics/AudioAnalytics.cpp
+++ b/services/mediametrics/AudioAnalytics.cpp
@@ -29,6 +29,7 @@
 #include "AudioTypes.h"           // string to int conversions
 #include "MediaMetricsService.h"  // package info
 #include "StringUtils.h"
+#include "ValidateId.h"
 
 #define PROP_AUDIO_ANALYTICS_CLOUD_ENABLED "persist.audio.analytics.cloud.enabled"
 
@@ -563,7 +564,7 @@
         const auto flagsForStats = types::lookup<types::INPUT_FLAG, short_enum_type_t>(flags);
         const auto sourceForStats = types::lookup<types::SOURCE_TYPE, short_enum_type_t>(source);
         // Android S
-        const auto logSessionIdForStats = stringutils::sanitizeLogSessionId(logSessionId);
+        const auto logSessionIdForStats = ValidateId::get()->validateId(logSessionId);
 
         LOG(LOG_LEVEL) << "key:" << key
               << " id:" << id
@@ -718,7 +719,7 @@
                  types::lookup<types::TRACK_TRAITS, short_enum_type_t>(traits);
         const auto usageForStats = types::lookup<types::USAGE, short_enum_type_t>(usage);
         // Android S
-        const auto logSessionIdForStats = stringutils::sanitizeLogSessionId(logSessionId);
+        const auto logSessionIdForStats = ValidateId::get()->validateId(logSessionId);
 
         LOG(LOG_LEVEL) << "key:" << key
               << " id:" << id
diff --git a/services/mediametrics/LruSet.h b/services/mediametrics/LruSet.h
new file mode 100644
index 0000000..1f0ab60
--- /dev/null
+++ b/services/mediametrics/LruSet.h
@@ -0,0 +1,123 @@
+/*
+ * Copyright (C) 2021 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#pragma once
+
+#include <list>
+#include <sstream>
+#include <unordered_map>
+
+namespace android::mediametrics {
+
+/**
+ * LruSet keeps a set of the last "Size" elements added or accessed.
+ *
+ * (Lru stands for least-recently-used eviction policy).
+ *
+ * Runs in O(1) time for add, remove, and check.  Internally implemented
+ * with an unordered_map and a list.  In order to remove elements,
+ * a list iterator is stored in the unordered_map
+ * (noting that std::list::erase() contractually
+ * does not affect iterators other than the one erased).
+ */
+
+template <typename T>
+class LruSet {
+    const size_t mMaxSize;
+    std::list<T> mAccessOrder;                 // front is the most recent, back is the oldest.
+    // item T with its access order iterator.
+    std::unordered_map<T, typename std::list<T>::iterator> mMap;
+
+public:
+    /**
+     * Constructs a LruSet which checks whether the element was
+     * accessed or added recently.
+     *
+     * The parameter maxSize is used to cap growth of LruSet;
+     * eviction is based on least recently used LRU.
+     * If maxSize is zero, the LruSet contains no elements
+     * and check() always returns false.
+     *
+     * \param maxSize the maximum number of elements that are tracked.
+     */
+    explicit LruSet(size_t maxSize) : mMaxSize(maxSize) {}
+
+    /**
+     * Returns the number of entries in the LruSet.
+     *
+     * This is a number between 0 and maxSize.
+     */
+    size_t size() const {
+        return mMap.size();
+    }
+
+    /** Clears the container contents. */
+    void clear() {
+        mMap.clear();
+        mAccessOrder.clear();
+    }
+
+    /** Returns a string dump of the last n entries. */
+    std::string dump(size_t n) const {
+        std::stringstream ss;
+        auto it = mAccessOrder.cbegin();
+        for (size_t i = 0; i < n && it != mAccessOrder.cend(); ++i) {
+            ss << *it++ << "\n";
+        }
+        return ss.str();
+    }
+
+    /** Adds a new item to the set. */
+    void add(const T& t) {
+        if (mMaxSize == 0) return;
+        auto it = mMap.find(t);
+        if (it != mMap.end()) { // already exists.
+            mAccessOrder.erase(it->second);  // move-to-front on the chronologically ordered list.
+        } else if (mAccessOrder.size() >= mMaxSize) {
+            const T last = mAccessOrder.back();
+            mAccessOrder.pop_back();
+            mMap.erase(last);
+        }
+        mAccessOrder.push_front(t);
+        mMap[t] = mAccessOrder.begin();
+    }
+
+    /**
+     * Removes an item from the set.
+     *
+     * \param t item to be removed.
+     * \return false if the item doesn't exist.
+     */
+    bool remove(const T& t) {
+        auto it = mMap.find(t);
+        if (it == mMap.end()) return false;
+        mAccessOrder.erase(it->second);
+        mMap.erase(it);
+        return true;
+    }
+
+    /** Returns true if t is present (and moves the access order of t to the front). */
+    bool check(const T& t) { // not const, as it adjusts the least-recently-used order.
+        auto it = mMap.find(t);
+        if (it == mMap.end()) return false;
+        mAccessOrder.erase(it->second);
+        mAccessOrder.push_front(it->first);
+        it->second = mAccessOrder.begin();
+        return true;
+    }
+};
+
+} // namespace android::mediametrics
diff --git a/services/mediametrics/MediaMetricsService.cpp b/services/mediametrics/MediaMetricsService.cpp
index 1d64878..35e0ae4 100644
--- a/services/mediametrics/MediaMetricsService.cpp
+++ b/services/mediametrics/MediaMetricsService.cpp
@@ -19,6 +19,7 @@
 #include <utils/Log.h>
 
 #include "MediaMetricsService.h"
+#include "ValidateId.h"
 #include "iface_statsd.h"
 
 #include <pwd.h> //getpwuid
@@ -204,6 +205,15 @@
     // now attach either the item or its dup to a const shared pointer
     std::shared_ptr<const mediametrics::Item> sitem(release ? item : item->dup());
 
+    // register log session ids with singleton.
+    if (startsWith(item->getKey(), "metrics.manager")) {
+        std::string logSessionId;
+        if (item->get("logSessionId", &logSessionId)
+                && mediametrics::stringutils::isLogSessionId(logSessionId.c_str())) {
+            mediametrics::ValidateId::get()->registerId(logSessionId);
+        }
+    }
+
     (void)mAudioAnalytics.submit(sitem, isTrusted);
 
     (void)dump2Statsd(sitem, mStatsdLog);  // failure should be logged in function.
@@ -309,6 +319,9 @@
                 result << "-- some lines may be truncated --\n";
             }
 
+            result << "LogSessionId:\n"
+                   << mediametrics::ValidateId::get()->dump();
+
             // Dump the statsd atoms we sent out.
             result << "Statsd atoms:\n"
                    << mStatsdLog->dumpToString("  " /* prefix */,
diff --git a/services/mediametrics/ValidateId.cpp b/services/mediametrics/ValidateId.cpp
new file mode 100644
index 0000000..0cc8593
--- /dev/null
+++ b/services/mediametrics/ValidateId.cpp
@@ -0,0 +1,65 @@
+/*
+ * Copyright (C) 2021 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "MediaMetricsService"  // not ValidateId
+#include <utils/Log.h>
+
+#include "ValidateId.h"
+
+namespace android::mediametrics {
+
+std::string ValidateId::dump() const
+{
+    std::stringstream ss;
+    ss << "Entries:" << mIdSet.size() << "  InvalidIds:" << mInvalidIds << "\n";
+    ss << mIdSet.dump(10);
+    return ss.str();
+}
+
+void ValidateId::registerId(const std::string& id)
+{
+    if (id.empty()) return;
+    if (!mediametrics::stringutils::isLogSessionId(id.c_str())) {
+        ALOGW("%s: rejecting malformed id %s", __func__, id.c_str());
+        return;
+    }
+    ALOGV("%s: registering %s", __func__, id.c_str());
+    mIdSet.add(id);
+}
+
+const std::string& ValidateId::validateId(const std::string& id)
+{
+    static const std::string empty{};
+    if (id.empty()) return empty;
+
+    // reject because the id is malformed
+    if (!mediametrics::stringutils::isLogSessionId(id.c_str())) {
+        ALOGW("%s: rejecting malformed id %s", __func__, id.c_str());
+        ++mInvalidIds;
+        return empty;
+    }
+
+    // reject because the id is unregistered
+    if (!mIdSet.check(id)) {
+        ALOGW("%s: rejecting unregistered id %s", __func__, id.c_str());
+        ++mInvalidIds;
+        return empty;
+    }
+    return id;
+}
+
+} // namespace android::mediametrics
diff --git a/services/mediametrics/ValidateId.h b/services/mediametrics/ValidateId.h
new file mode 100644
index 0000000..166b39a
--- /dev/null
+++ b/services/mediametrics/ValidateId.h
@@ -0,0 +1,80 @@
+/*
+ * Copyright (C) 2021 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#pragma once
+
+#include "LruSet.h"
+#include "StringUtils.h"
+#include "Wrap.h"
+
+namespace android::mediametrics {
+
+/*
+ * ValidateId is used to check whether the log session id is properly formed
+ * and has been registered (i.e. from the Java MediaMetricsManagerService).
+ *
+ * The default memory window to track registered ids is set to SINGLETON_LRU_SET_SIZE.
+ *
+ * This class is not thread-safe, but the singleton returned by get() uses LockWrap<>
+ * to ensure thread-safety.
+ */
+class ValidateId {
+    mediametrics::LruSet<std::string> mIdSet;
+    size_t mInvalidIds = 0;  // count invalid ids encountered.
+public:
+    /** Creates a ValidateId object with size memory window. */
+    explicit ValidateId(size_t size) : mIdSet{size} {}
+
+    /** Returns a string dump of recent contents and stats. */
+    std::string dump() const;
+
+    /**
+     * Registers the id string.
+     *
+     * If id string is malformed (not 16 Base64Url chars), it is ignored.
+     * Once registered, calling validateId() will return id (instead of the empty string).
+     * ValidateId may "forget" the id after not encountering it within the past N ids,
+     * where N is the size set in the constructor.
+     *
+     * param id string (from MediaMetricsManagerService).
+     */
+    void registerId(const std::string& id);
+
+    /**
+     * Returns the empty string if id string is malformed (not 16 Base64Url chars)
+     * or if id string has not been seen (in the recent size ids);
+     * otherwise it returns the same id parameter.
+     *
+     * \param id string (to be sent to statsd).
+     */
+    const std::string& validateId(const std::string& id);
+
+    /** Singleton set size */
+    static inline constexpr size_t SINGLETON_LRU_SET_SIZE = 2000;
+
+    using LockedValidateId = mediametrics::LockWrap<ValidateId>;
+    /**
+     * Returns a singleton locked ValidateId object that is thread-safe using LockWrap<>.
+     *
+     * The Singleton ValidateId object is created with size LRU_SET_SIZE (during first call).
+     */
+    static inline LockedValidateId& get() {
+        static LockedValidateId privateSet{SINGLETON_LRU_SET_SIZE};
+        return privateSet;
+    }
+};
+
+} // namespace android::mediametrics
diff --git a/services/mediametrics/statsd_audiorecord.cpp b/services/mediametrics/statsd_audiorecord.cpp
index 41efcaa..c53b6f3 100644
--- a/services/mediametrics/statsd_audiorecord.cpp
+++ b/services/mediametrics/statsd_audiorecord.cpp
@@ -32,7 +32,7 @@
 #include <statslog.h>
 
 #include "MediaMetricsService.h"
-#include "StringUtils.h"
+#include "ValidateId.h"
 #include "frameworks/proto_logging/stats/message/mediametrics_message.pb.h"
 #include "iface_statsd.h"
 
@@ -143,8 +143,7 @@
     // log_session_id (string)
     std::string logSessionId;
     (void)item->getString("android.media.audiorecord.logSessionId", &logSessionId);
-    const auto log_session_id =
-            mediametrics::stringutils::sanitizeLogSessionId(logSessionId);
+    const auto log_session_id = mediametrics::ValidateId::get()->validateId(logSessionId);
 
     android::util::BytesField bf_serialized( serialized.c_str(), serialized.size());
     int result = android::util::stats_write(android::util::MEDIAMETRICS_AUDIORECORD_REPORTED,
diff --git a/services/mediametrics/statsd_audiotrack.cpp b/services/mediametrics/statsd_audiotrack.cpp
index 59627ae..707effd 100644
--- a/services/mediametrics/statsd_audiotrack.cpp
+++ b/services/mediametrics/statsd_audiotrack.cpp
@@ -32,7 +32,7 @@
 #include <statslog.h>
 
 #include "MediaMetricsService.h"
-#include "StringUtils.h"
+#include "ValidateId.h"
 #include "frameworks/proto_logging/stats/message/mediametrics_message.pb.h"
 #include "iface_statsd.h"
 
@@ -137,8 +137,7 @@
     // log_session_id (string)
     std::string logSessionId;
     (void)item->getString("android.media.audiotrack.logSessionId", &logSessionId);
-    const auto log_session_id =
-            mediametrics::stringutils::sanitizeLogSessionId(logSessionId);
+    const auto log_session_id = mediametrics::ValidateId::get()->validateId(logSessionId);
 
     android::util::BytesField bf_serialized( serialized.c_str(), serialized.size());
     int result = android::util::stats_write(android::util::MEDIAMETRICS_AUDIOTRACK_REPORTED,
diff --git a/services/mediametrics/statsd_codec.cpp b/services/mediametrics/statsd_codec.cpp
index 46cbdc8..8581437 100644
--- a/services/mediametrics/statsd_codec.cpp
+++ b/services/mediametrics/statsd_codec.cpp
@@ -34,7 +34,7 @@
 
 #include "cleaner.h"
 #include "MediaMetricsService.h"
-#include "StringUtils.h"
+#include "ValidateId.h"
 #include "frameworks/proto_logging/stats/message/mediametrics_message.pb.h"
 #include "iface_statsd.h"
 
@@ -228,7 +228,7 @@
 
     std::string sessionId;
     if (item->getString("android.media.mediacodec.log-session-id", &sessionId)) {
-        sessionId = mediametrics::stringutils::sanitizeLogSessionId(sessionId);
+        sessionId = mediametrics::ValidateId::get()->validateId(sessionId);
         metrics_proto.set_log_session_id(sessionId);
     }
     AStatsEvent_writeString(event, codec.c_str());
diff --git a/services/mediametrics/statsd_extractor.cpp b/services/mediametrics/statsd_extractor.cpp
index bcf2e0a..a8bfeaa 100644
--- a/services/mediametrics/statsd_extractor.cpp
+++ b/services/mediametrics/statsd_extractor.cpp
@@ -32,7 +32,7 @@
 #include <statslog.h>
 
 #include "MediaMetricsService.h"
-#include "StringUtils.h"
+#include "ValidateId.h"
 #include "frameworks/proto_logging/stats/message/mediametrics_message.pb.h"
 #include "iface_statsd.h"
 
@@ -86,7 +86,7 @@
 
     std::string log_session_id;
     if (item->getString("android.media.mediaextractor.logSessionId", &log_session_id)) {
-        log_session_id = mediametrics::stringutils::sanitizeLogSessionId(log_session_id);
+        log_session_id = mediametrics::ValidateId::get()->validateId(log_session_id);
         metrics_proto.set_log_session_id(log_session_id);
     }
 
diff --git a/services/mediametrics/statsd_mediaparser.cpp b/services/mediametrics/statsd_mediaparser.cpp
index 921b320..67ca874b 100644
--- a/services/mediametrics/statsd_mediaparser.cpp
+++ b/services/mediametrics/statsd_mediaparser.cpp
@@ -31,7 +31,7 @@
 #include <statslog.h>
 
 #include "MediaMetricsService.h"
-#include "StringUtils.h"
+#include "ValidateId.h"
 #include "frameworks/proto_logging/stats/enums/stats/mediametrics/mediametrics.pb.h"
 #include "iface_statsd.h"
 
@@ -81,7 +81,7 @@
 
     std::string logSessionId;
     item->getString("android.media.mediaparser.logSessionId", &logSessionId);
-    logSessionId = mediametrics::stringutils::sanitizeLogSessionId(logSessionId);
+    logSessionId = mediametrics::ValidateId::get()->validateId(logSessionId);
 
     int result = android::util::stats_write(android::util::MEDIAMETRICS_MEDIAPARSER_REPORTED,
                                timestamp_nanos,
diff --git a/services/mediametrics/statsd_recorder.cpp b/services/mediametrics/statsd_recorder.cpp
index b29ad73..5f54a68 100644
--- a/services/mediametrics/statsd_recorder.cpp
+++ b/services/mediametrics/statsd_recorder.cpp
@@ -32,7 +32,7 @@
 #include <statslog.h>
 
 #include "MediaMetricsService.h"
-#include "StringUtils.h"
+#include "ValidateId.h"
 #include "frameworks/proto_logging/stats/message/mediametrics_message.pb.h"
 #include "iface_statsd.h"
 
@@ -59,7 +59,7 @@
     // string kRecorderLogSessionId = "android.media.mediarecorder.log-session-id";
     std::string log_session_id;
     if (item->getString("android.media.mediarecorder.log-session-id", &log_session_id)) {
-        log_session_id = mediametrics::stringutils::sanitizeLogSessionId(log_session_id);
+        log_session_id = mediametrics::ValidateId::get()->validateId(log_session_id);
         metrics_proto.set_log_session_id(log_session_id);
     }
     // string kRecorderAudioMime = "android.media.mediarecorder.audio.mime";
diff --git a/services/mediametrics/tests/mediametrics_tests.cpp b/services/mediametrics/tests/mediametrics_tests.cpp
index 2336d6f..69ec947 100644
--- a/services/mediametrics/tests/mediametrics_tests.cpp
+++ b/services/mediametrics/tests/mediametrics_tests.cpp
@@ -28,6 +28,7 @@
 
 #include "AudioTypes.h"
 #include "StringUtils.h"
+#include "ValidateId.h"
 
 using namespace android;
 
@@ -1127,3 +1128,100 @@
    validId2[3] = '!';
    ASSERT_EQ("", mediametrics::stringutils::sanitizeLogSessionId(validId2));
 }
+
+TEST(mediametrics_tests, LruSet) {
+    constexpr size_t LRU_SET_SIZE = 2;
+    mediametrics::LruSet<std::string> lruSet(LRU_SET_SIZE);
+
+    // test adding a couple strings.
+    lruSet.add("abc");
+    ASSERT_EQ(1u, lruSet.size());
+    ASSERT_TRUE(lruSet.check("abc"));
+    lruSet.add("def");
+    ASSERT_EQ(2u, lruSet.size());
+
+    // now adding the third string causes eviction of the oldest.
+    lruSet.add("ghi");
+    ASSERT_FALSE(lruSet.check("abc"));
+    ASSERT_TRUE(lruSet.check("ghi"));
+    ASSERT_TRUE(lruSet.check("def"));  // "def" is most recent.
+    ASSERT_EQ(2u, lruSet.size());      // "abc" is correctly discarded.
+
+    // adding another string will evict the oldest.
+    lruSet.add("foo");
+    ASSERT_FALSE(lruSet.check("ghi")); // note: "ghi" discarded when "foo" added.
+    ASSERT_TRUE(lruSet.check("foo"));
+    ASSERT_TRUE(lruSet.check("def"));
+
+    // manual removing of a string works, too.
+    ASSERT_TRUE(lruSet.remove("def"));
+    ASSERT_FALSE(lruSet.check("def")); // we manually removed "def".
+    ASSERT_TRUE(lruSet.check("foo"));  // "foo" is still there.
+    ASSERT_EQ(1u, lruSet.size());
+
+    // you can't remove a string that has not been added.
+    ASSERT_FALSE(lruSet.remove("bar")); // Note: "bar" doesn't exist, so remove returns false.
+    ASSERT_EQ(1u, lruSet.size());
+
+    lruSet.add("foo");   // adding "foo" (which already exists) doesn't change size.
+    ASSERT_EQ(1u, lruSet.size());
+    lruSet.add("bar");   // add "bar"
+    ASSERT_EQ(2u, lruSet.size());
+    lruSet.add("glorp"); // add "glorp" evicts "foo".
+    ASSERT_EQ(2u, lruSet.size());
+    ASSERT_TRUE(lruSet.check("bar"));
+    ASSERT_TRUE(lruSet.check("glorp"));
+    ASSERT_FALSE(lruSet.check("foo"));
+}
+
+TEST(mediametrics_tests, LruSet0) {
+    constexpr size_t LRU_SET_SIZE = 0;
+    mediametrics::LruSet<std::string> lruSet(LRU_SET_SIZE);
+
+    lruSet.add("a");
+    ASSERT_EQ(0u, lruSet.size());
+    ASSERT_FALSE(lruSet.check("a"));
+    ASSERT_FALSE(lruSet.remove("a")); // never added.
+    ASSERT_EQ(0u, lruSet.size());
+}
+
+// Returns a 16 Base64Url string representing the decimal representation of value
+// (with leading 0s) e.g. 0000000000000000, 0000000000000001, 0000000000000002, ...
+static std::string generateId(size_t value)
+{
+    char id[16 + 1]; // to be filled with 16 Base64Url chars (and zero termination)
+    char *sptr = id + 16; // start at the end.
+    *sptr-- = 0; // zero terminate.
+    // output the digits from least significant to most significant.
+    while (value) {
+        *sptr-- = value % 10;
+        value /= 10;
+    }
+    // add leading 0's
+    while (sptr > id) {
+        *sptr-- = '0';
+    }
+    return std::string(id);
+}
+
+TEST(mediametrics_tests, ValidateId) {
+    constexpr size_t LRU_SET_SIZE = 3;
+    constexpr size_t IDS = 10;
+    static_assert(IDS > LRU_SET_SIZE);  // IDS must be greater than LRU_SET_SIZE.
+    mediametrics::ValidateId validateId(LRU_SET_SIZE);
+
+
+    // register IDs as integer strings counting from 0.
+    for (size_t i = 0; i < IDS; ++i) {
+        validateId.registerId(generateId(i));
+    }
+
+    // only the last LRU_SET_SIZE exist.
+    for (size_t i = 0; i < IDS - LRU_SET_SIZE; ++i) {
+        ASSERT_EQ("", validateId.validateId(generateId(i)));
+    }
+    for (size_t i = IDS - LRU_SET_SIZE; i < IDS; ++i) {
+        const std::string id = generateId(i);
+        ASSERT_EQ(id, validateId.validateId(id));
+    }
+}
diff --git a/services/oboeservice/AAudioServiceEndpoint.cpp b/services/oboeservice/AAudioServiceEndpoint.cpp
index 13dd3d3..340076e 100644
--- a/services/oboeservice/AAudioServiceEndpoint.cpp
+++ b/services/oboeservice/AAudioServiceEndpoint.cpp
@@ -59,6 +59,7 @@
     result << "    Device Id:            " << getDeviceId() << "\n";
     result << "    Sample Rate:          " << getSampleRate() << "\n";
     result << "    Channel Count:        " << getSamplesPerFrame() << "\n";
+    result << "    Channel Mask:         0x" << std::hex << getChannelMask() << std::dec << "\n";
     result << "    Format:               " << getFormat() << "\n";
     result << "    Frames Per Burst:     " << mFramesPerBurst << "\n";
     result << "    Usage:                " << getUsage() << "\n";
@@ -164,6 +165,10 @@
         configuration.getSamplesPerFrame() != getSamplesPerFrame()) {
         return false;
     }
+    if (configuration.getChannelMask() != AAUDIO_UNSPECIFIED &&
+        configuration.getChannelMask() != getChannelMask()) {
+        return false;
+    }
     return true;
 }
 
diff --git a/services/oboeservice/AAudioServiceEndpointMMAP.cpp b/services/oboeservice/AAudioServiceEndpointMMAP.cpp
index a08098c..35a0890 100644
--- a/services/oboeservice/AAudioServiceEndpointMMAP.cpp
+++ b/services/oboeservice/AAudioServiceEndpointMMAP.cpp
@@ -126,20 +126,15 @@
     }
     config.sample_rate = aaudioSampleRate;
 
-    int32_t aaudioSamplesPerFrame = getSamplesPerFrame();
-
     const aaudio_direction_t direction = getDirection();
 
+    config.channel_mask = AAudio_getChannelMaskForOpen(
+            getChannelMask(), getSamplesPerFrame(), direction == AAUDIO_DIRECTION_INPUT);
+
     if (direction == AAUDIO_DIRECTION_OUTPUT) {
-        config.channel_mask = (aaudioSamplesPerFrame == AAUDIO_UNSPECIFIED)
-                              ? AUDIO_CHANNEL_OUT_STEREO
-                              : audio_channel_out_mask_from_count(aaudioSamplesPerFrame);
         mHardwareTimeOffsetNanos = OUTPUT_ESTIMATED_HARDWARE_OFFSET_NANOS; // frames at DAC later
 
     } else if (direction == AAUDIO_DIRECTION_INPUT) {
-        config.channel_mask =  (aaudioSamplesPerFrame == AAUDIO_UNSPECIFIED)
-                               ? AUDIO_CHANNEL_IN_STEREO
-                               : audio_channel_in_mask_from_count(aaudioSamplesPerFrame);
         mHardwareTimeOffsetNanos = INPUT_ESTIMATED_HARDWARE_OFFSET_NANOS; // frames at ADC earlier
 
     } else {
@@ -225,9 +220,9 @@
     }
 
     // Get information about the stream and pass it back to the caller.
-    setSamplesPerFrame((direction == AAUDIO_DIRECTION_OUTPUT)
-                       ? audio_channel_count_from_out_mask(config.channel_mask)
-                       : audio_channel_count_from_in_mask(config.channel_mask));
+    setChannelMask(AAudioConvert_androidToAAudioChannelMask(
+            config.channel_mask, getDirection() == AAUDIO_DIRECTION_INPUT,
+            AAudio_isChannelIndexMask(config.channel_mask)));
 
     // AAudio creates a copy of this FD and retains ownership of the copy.
     // Assume that AudioFlinger will close the original shared_memory_fd.
@@ -247,9 +242,9 @@
     setFormat(config.format);
     setSampleRate(config.sample_rate);
 
-    ALOGD("%s() actual rate = %d, channels = %d"
-          ", deviceId = %d, capacity = %d\n",
-          __func__, getSampleRate(), getSamplesPerFrame(), deviceId, getBufferCapacity());
+    ALOGD("%s() actual rate = %d, channels = %d channelMask = %#x, deviceId = %d, capacity = %d\n",
+          __func__, getSampleRate(), getSamplesPerFrame(), getChannelMask(),
+          deviceId, getBufferCapacity());
 
     ALOGD("%s() format = 0x%08x, frame size = %d, burst size = %d",
           __func__, getFormat(), calculateBytesPerFrame(), mFramesPerBurst);
diff --git a/services/oboeservice/AAudioServiceEndpointShared.cpp b/services/oboeservice/AAudioServiceEndpointShared.cpp
index 5fbcadb..5af0a91 100644
--- a/services/oboeservice/AAudioServiceEndpointShared.cpp
+++ b/services/oboeservice/AAudioServiceEndpointShared.cpp
@@ -78,7 +78,7 @@
     result = mStreamInternal->open(builder);
 
     setSampleRate(mStreamInternal->getSampleRate());
-    setSamplesPerFrame(mStreamInternal->getSamplesPerFrame());
+    setChannelMask(mStreamInternal->getChannelMask());
     setDeviceId(mStreamInternal->getDeviceId());
     setSessionId(mStreamInternal->getSessionId());
     setFormat(AUDIO_FORMAT_PCM_FLOAT); // force for mixer
diff --git a/services/oboeservice/AAudioServiceStreamBase.cpp b/services/oboeservice/AAudioServiceStreamBase.cpp
index 34ddd4d..4ffc127 100644
--- a/services/oboeservice/AAudioServiceStreamBase.cpp
+++ b/services/oboeservice/AAudioServiceStreamBase.cpp
@@ -73,7 +73,8 @@
 }
 
 std::string AAudioServiceStreamBase::dumpHeader() {
-    return std::string("    T   Handle   UId   Port Run State Format Burst Chan Capacity");
+    return std::string(
+            "    T   Handle   UId   Port Run State Format Burst Chan Mask     Capacity");
 }
 
 std::string AAudioServiceStreamBase::dump() const {
@@ -88,6 +89,7 @@
     result << std::setw(7) << getFormat();
     result << std::setw(6) << mFramesPerBurst;
     result << std::setw(5) << getSamplesPerFrame();
+    result << std::setw(8) << std::hex << getChannelMask() << std::dec;
     result << std::setw(9) << getBufferCapacity();
 
     return result.str();
diff --git a/services/oboeservice/AAudioServiceStreamShared.cpp b/services/oboeservice/AAudioServiceStreamShared.cpp
index c665cda..ad06d97 100644
--- a/services/oboeservice/AAudioServiceStreamShared.cpp
+++ b/services/oboeservice/AAudioServiceStreamShared.cpp
@@ -164,11 +164,11 @@
         goto error;
     }
 
-    setSamplesPerFrame(configurationInput.getSamplesPerFrame());
-    if (getSamplesPerFrame() == AAUDIO_UNSPECIFIED) {
-        setSamplesPerFrame(endpoint->getSamplesPerFrame());
+    setChannelMask(configurationInput.getChannelMask());
+    if (getChannelMask() == AAUDIO_UNSPECIFIED) {
+        setChannelMask(endpoint->getChannelMask());
     } else if (getSamplesPerFrame() != endpoint->getSamplesPerFrame()) {
-        ALOGD("%s() mSamplesPerFrame = %d, need %d",
+        ALOGD("%s() mSamplesPerFrame = %#x, need %#x",
               __func__, getSamplesPerFrame(), endpoint->getSamplesPerFrame());
         result = AAUDIO_ERROR_OUT_OF_RANGE;
         goto error;
diff --git a/services/oboeservice/fuzzer/README.md b/services/oboeservice/fuzzer/README.md
index 00b85df..ae7af3eb 100644
--- a/services/oboeservice/fuzzer/README.md
+++ b/services/oboeservice/fuzzer/README.md
@@ -15,7 +15,7 @@
 4. InService
 5. DeviceId
 6. SampleRate
-7. SamplesPerFrame
+7. ChannelMask
 8. Direction
 9. SharingMode
 10. Usage
@@ -31,7 +31,7 @@
 | `InService`   | `bool` | Value obtained from FuzzedDataProvider |
 | `DeviceId`   | `INT32_MIN` to `INT32_MAX` | Value obtained from FuzzedDataProvider |
 | `SampleRate`   | `INT32_MIN` to `INT32_MAX` | Value obtained from FuzzedDataProvider |
-| `SamplesPerFrame` | `INT32_MIN` to `INT32_MAX` | Value obtained from FuzzedDataProvider |
+| `ChannelMask` | `AAUDIO_UNSPECIFIED`, `AAUDIO_CHANNEL_INDEX_MASK_1`, `AAUDIO_CHANNEL_INDEX_MASK_2`, `AAUDIO_CHANNEL_INDEX_MASK_3`, `AAUDIO_CHANNEL_INDEX_MASK_4`, `AAUDIO_CHANNEL_INDEX_MASK_5`, `AAUDIO_CHANNEL_INDEX_MASK_6`, `AAUDIO_CHANNEL_INDEX_MASK_7`, `AAUDIO_CHANNEL_INDEX_MASK_8`, `AAUDIO_CHANNEL_INDEX_MASK_9`, `AAUDIO_CHANNEL_INDEX_MASK_10`, `AAUDIO_CHANNEL_INDEX_MASK_11`, `AAUDIO_CHANNEL_INDEX_MASK_12`, `AAUDIO_CHANNEL_INDEX_MASK_13`, `AAUDIO_CHANNEL_INDEX_MASK_14`, `AAUDIO_CHANNEL_INDEX_MASK_15`, `AAUDIO_CHANNEL_INDEX_MASK_16`, `AAUDIO_CHANNEL_INDEX_MASK_17`, `AAUDIO_CHANNEL_INDEX_MASK_18`, `AAUDIO_CHANNEL_INDEX_MASK_19`, `AAUDIO_CHANNEL_INDEX_MASK_20`, `AAUDIO_CHANNEL_INDEX_MASK_21`, `AAUDIO_CHANNEL_INDEX_MASK_22`, `AAUDIO_CHANNEL_INDEX_MASK_23`, `AAUDIO_CHANNEL_INDEX_MASK_24`, `AAUDIO_CHANNEL_MONO`, `AAUDIO_CHANNEL_STEREO`, `AAUDIO_CHANNEL_FRONT_BACK`, `AAUDIO_CHANNEL_2POINT0POINT2`, `AAUDIO_CHANNEL_2POINT1POINT2`, `AAUDIO_CHANNEL_3POINT0POINT2`, `AAUDIO_CHANNEL_3POINT1POINT2`, `AAUDIO_CHANNEL_5POINT1`, `AAUDIO_CHANNEL_MONO`, `AAUDIO_CHANNEL_STEREO`, `AAUDIO_CHANNEL_2POINT1`, `AAUDIO_CHANNEL_TRI`, `AAUDIO_CHANNEL_TRI_BACK`, `AAUDIO_CHANNEL_3POINT1`, `AAUDIO_CHANNEL_2POINT0POINT2`, `AAUDIO_CHANNEL_2POINT1POINT2`, `AAUDIO_CHANNEL_3POINT0POINT2`, `AAUDIO_CHANNEL_3POINT1POINT2`, `AAUDIO_CHANNEL_QUAD`, `AAUDIO_CHANNEL_QUAD_SIDE`, `AAUDIO_CHANNEL_SURROUND`, `AAUDIO_CHANNEL_PENTA`, `AAUDIO_CHANNEL_5POINT1`, `AAUDIO_CHANNEL_5POINT1_SIDE`, `AAUDIO_CHANNEL_5POINT1POINT2`, `AAUDIO_CHANNEL_5POINT1POINT4`, `AAUDIO_CHANNEL_6POINT1`, `AAUDIO_CHANNEL_7POINT1`, `AAUDIO_CHANNEL_7POINT1POINT2`, `AAUDIO_CHANNEL_7POINT1POINT4`, `AAUDIO_CHANNEL_9POINT1POINT4`, `AAUDIO_CHANNEL_9POINT1POINT6` | Value obtained from FuzzedDataProvider |
 | `Direction` | `AAUDIO_DIRECTION_OUTPUT`, `AAUDIO_DIRECTION_INPUT` | Value chosen from valid values by obtaining index from FuzzedDataProvider |
 | `SharingMode` | `AAUDIO_SHARING_MODE_EXCLUSIVE`, `AAUDIO_SHARING_MODE_SHARED` | Value chosen from valid values by obtaining index from FuzzedDataProvider |
 | `Usage` | `AAUDIO_USAGE_MEDIA`, `AAUDIO_USAGE_VOICE_COMMUNICATION`, `AAUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING`, `AAUDIO_USAGE_ALARM`, `AAUDIO_USAGE_NOTIFICATION`, `AAUDIO_USAGE_NOTIFICATION_RINGTONE`, `AAUDIO_USAGE_NOTIFICATION_EVENT`, `AAUDIO_USAGE_ASSISTANCE_ACCESSIBILITY`, `AAUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE`, `AAUDIO_USAGE_ASSISTANCE_SONIFICATION`, `AAUDIO_USAGE_GAME`, `AAUDIO_USAGE_ASSISTANT`, `AAUDIO_SYSTEM_USAGE_EMERGENCY`, `AAUDIO_SYSTEM_USAGE_SAFETY`, `AAUDIO_SYSTEM_USAGE_VEHICLE_STATUS`, `AAUDIO_SYSTEM_USAGE_ANNOUNCEMENT` | Value chosen from valid values by obtaining index from FuzzedDataProvider |
diff --git a/services/oboeservice/fuzzer/oboeservice_fuzzer.cpp b/services/oboeservice/fuzzer/oboeservice_fuzzer.cpp
index 4bc661c..17e8d36 100644
--- a/services/oboeservice/fuzzer/oboeservice_fuzzer.cpp
+++ b/services/oboeservice/fuzzer/oboeservice_fuzzer.cpp
@@ -68,10 +68,71 @@
     AAUDIO_INPUT_PRESET_UNPROCESSED,       AAUDIO_INPUT_PRESET_VOICE_PERFORMANCE,
 };
 
+aaudio_channel_mask_t kAAudioChannelMasks[] = {
+    AAUDIO_UNSPECIFIED,
+    AAUDIO_CHANNEL_INDEX_MASK_1,
+    AAUDIO_CHANNEL_INDEX_MASK_2,
+    AAUDIO_CHANNEL_INDEX_MASK_3,
+    AAUDIO_CHANNEL_INDEX_MASK_4,
+    AAUDIO_CHANNEL_INDEX_MASK_5,
+    AAUDIO_CHANNEL_INDEX_MASK_6,
+    AAUDIO_CHANNEL_INDEX_MASK_7,
+    AAUDIO_CHANNEL_INDEX_MASK_8,
+    AAUDIO_CHANNEL_INDEX_MASK_9,
+    AAUDIO_CHANNEL_INDEX_MASK_10,
+    AAUDIO_CHANNEL_INDEX_MASK_11,
+    AAUDIO_CHANNEL_INDEX_MASK_12,
+    AAUDIO_CHANNEL_INDEX_MASK_13,
+    AAUDIO_CHANNEL_INDEX_MASK_14,
+    AAUDIO_CHANNEL_INDEX_MASK_15,
+    AAUDIO_CHANNEL_INDEX_MASK_16,
+    AAUDIO_CHANNEL_INDEX_MASK_17,
+    AAUDIO_CHANNEL_INDEX_MASK_18,
+    AAUDIO_CHANNEL_INDEX_MASK_19,
+    AAUDIO_CHANNEL_INDEX_MASK_20,
+    AAUDIO_CHANNEL_INDEX_MASK_21,
+    AAUDIO_CHANNEL_INDEX_MASK_22,
+    AAUDIO_CHANNEL_INDEX_MASK_23,
+    AAUDIO_CHANNEL_INDEX_MASK_24,
+    AAUDIO_CHANNEL_MONO,
+    AAUDIO_CHANNEL_STEREO,
+    AAUDIO_CHANNEL_FRONT_BACK,
+    AAUDIO_CHANNEL_2POINT0POINT2,
+    AAUDIO_CHANNEL_2POINT1POINT2,
+    AAUDIO_CHANNEL_3POINT0POINT2,
+    AAUDIO_CHANNEL_3POINT1POINT2,
+    AAUDIO_CHANNEL_5POINT1,
+    AAUDIO_CHANNEL_MONO,
+    AAUDIO_CHANNEL_STEREO,
+    AAUDIO_CHANNEL_2POINT1,
+    AAUDIO_CHANNEL_TRI,
+    AAUDIO_CHANNEL_TRI_BACK,
+    AAUDIO_CHANNEL_3POINT1,
+    AAUDIO_CHANNEL_2POINT0POINT2,
+    AAUDIO_CHANNEL_2POINT1POINT2,
+    AAUDIO_CHANNEL_3POINT0POINT2,
+    AAUDIO_CHANNEL_3POINT1POINT2,
+    AAUDIO_CHANNEL_QUAD,
+    AAUDIO_CHANNEL_QUAD_SIDE,
+    AAUDIO_CHANNEL_SURROUND,
+    AAUDIO_CHANNEL_PENTA,
+    AAUDIO_CHANNEL_5POINT1,
+    AAUDIO_CHANNEL_5POINT1_SIDE,
+    AAUDIO_CHANNEL_5POINT1POINT2,
+    AAUDIO_CHANNEL_5POINT1POINT4,
+    AAUDIO_CHANNEL_6POINT1,
+    AAUDIO_CHANNEL_7POINT1,
+    AAUDIO_CHANNEL_7POINT1POINT2,
+    AAUDIO_CHANNEL_7POINT1POINT4,
+    AAUDIO_CHANNEL_9POINT1POINT4,
+    AAUDIO_CHANNEL_9POINT1POINT6,
+};
+
 const size_t kNumAAudioFormats = std::size(kAAudioFormats);
 const size_t kNumAAudioUsages = std::size(kAAudioUsages);
 const size_t kNumAAudioContentTypes = std::size(kAAudioContentTypes);
 const size_t kNumAAudioInputPresets = std::size(kAAudioInputPresets);
+const size_t kNumAAudioChannelMasks = std::size(kAAudioChannelMasks);
 
 class FuzzAAudioClient : public virtual RefBase, public AAudioServiceInterface {
    public:
@@ -305,7 +366,11 @@
 
     request.getConfiguration().setDeviceId(fdp.ConsumeIntegral<int32_t>());
     request.getConfiguration().setSampleRate(fdp.ConsumeIntegral<int32_t>());
-    request.getConfiguration().setSamplesPerFrame(fdp.ConsumeIntegral<int32_t>());
+    request.getConfiguration().setChannelMask((aaudio_channel_mask_t)(
+        fdp.ConsumeBool()
+            ? fdp.ConsumeIntegral<int32_t>()
+            : kAAudioChannelMasks[fdp.ConsumeIntegralInRange<int32_t>(
+                    0, kNumAAudioChannelMasks - 1)]));
     request.getConfiguration().setDirection(
         fdp.ConsumeBool() ? fdp.ConsumeIntegral<int32_t>()
                           : (fdp.ConsumeBool() ? AAUDIO_DIRECTION_OUTPUT : AAUDIO_DIRECTION_INPUT));