diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index bd6db55..a1e1702 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -55,6 +55,7 @@
 
 // for audio_track_cblk_t::mState, to match TrackBase.h
 static inline constexpr int CBLK_STATE_IDLE = 0;
+static inline constexpr int CBLK_STATE_ACTIVE = 6;
 static inline constexpr int CBLK_STATE_PAUSING = 7;
 
 /**
diff --git a/media/codec2/sfplugin/CCodecConfig.cpp b/media/codec2/sfplugin/CCodecConfig.cpp
index 242eeaf..c15b5ca 100644
--- a/media/codec2/sfplugin/CCodecConfig.cpp
+++ b/media/codec2/sfplugin/CCodecConfig.cpp
@@ -899,6 +899,9 @@
     add(ConfigMapper(KEY_AAC_MAX_OUTPUT_CHANNEL_COUNT, C2_PARAMKEY_MAX_CHANNEL_COUNT, "value")
         .limitTo(D::AUDIO & (D::CONFIG | D::PARAM | D::READ)));
 
+    add(ConfigMapper(KEY_MAX_OUTPUT_CHANNEL_COUNT, C2_PARAMKEY_MAX_CHANNEL_COUNT, "value")
+        .limitTo(D::AUDIO & (D::CONFIG | D::PARAM | D::READ)));
+
     add(ConfigMapper(KEY_AAC_SBR_MODE, C2_PARAMKEY_AAC_SBR_MODE, "value")
         .limitTo(D::AUDIO & D::ENCODER & (D::CONFIG | D::PARAM | D::READ))
         .withMapper([](C2Value v) -> C2Value {
diff --git a/media/libaaudio/src/core/AAudioAudio.cpp b/media/libaaudio/src/core/AAudioAudio.cpp
index f07e66e..90ff4a5 100644
--- a/media/libaaudio/src/core/AAudioAudio.cpp
+++ b/media/libaaudio/src/core/AAudioAudio.cpp
@@ -352,7 +352,8 @@
 {
 
     AudioStream *audioStream = convertAAudioStreamToAudioStream(stream);
-    return audioStream->waitForStateChange(inputState, nextState, timeoutNanoseconds);
+    android::sp<AudioStream> spAudioStream(audioStream);
+    return spAudioStream->waitForStateChange(inputState, nextState, timeoutNanoseconds);
 }
 
 // ============================================================
diff --git a/media/libaaudio/tests/test_various.cpp b/media/libaaudio/tests/test_various.cpp
index cbf863f..b68fc7b 100644
--- a/media/libaaudio/tests/test_various.cpp
+++ b/media/libaaudio/tests/test_various.cpp
@@ -25,6 +25,7 @@
 
 #include <gtest/gtest.h>
 #include <unistd.h>
+#include <thread>
 
 // Callback function that does nothing.
 aaudio_data_callback_result_t NoopDataCallbackProc(
@@ -51,6 +52,7 @@
 }
 
 constexpr int64_t NANOS_PER_MILLISECOND = 1000 * 1000;
+constexpr int64_t MICROS_PER_MILLISECOND = 1000;
 
 void checkReleaseThenClose(aaudio_performance_mode_t perfMode,
         aaudio_sharing_mode_t sharingMode,
@@ -762,6 +764,58 @@
     checkCallbackOnce(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY);
 }
 
+void waitForStateChangeToClosingorClosed(AAudioStream **stream, std::atomic<bool>* isReady)
+{
+    *isReady = true;
+    aaudio_stream_state_t state = AAUDIO_STREAM_STATE_UNKNOWN;
+    EXPECT_EQ(AAUDIO_OK, AAudioStream_waitForStateChange(*stream,
+                                                         AAUDIO_STREAM_STATE_OPEN, &state,
+                                                         10000 * NANOS_PER_MILLISECOND));
+    if ((state != AAUDIO_STREAM_STATE_CLOSING) && (state != AAUDIO_STREAM_STATE_CLOSED)){
+        FAIL() << "ERROR - State not closing or closed. Current state: " <<
+                AAudio_convertStreamStateToText(state);
+    }
+}
+
+void testWaitForStateChangeClose(aaudio_performance_mode_t perfMode) {
+    AAudioStreamBuilder *aaudioBuilder = nullptr;
+    AAudioStream *aaudioStream = nullptr;
+
+    ASSERT_EQ(AAUDIO_OK, AAudio_createStreamBuilder(&aaudioBuilder));
+    AAudioStreamBuilder_setPerformanceMode(aaudioBuilder, perfMode);
+    ASSERT_EQ(AAUDIO_OK, AAudioStreamBuilder_openStream(aaudioBuilder, &aaudioStream));
+
+    // Verify Open State
+    aaudio_stream_state_t state = AAUDIO_STREAM_STATE_UNKNOWN;
+    EXPECT_EQ(AAUDIO_OK, AAudioStream_waitForStateChange(aaudioStream,
+                                                         AAUDIO_STREAM_STATE_UNKNOWN, &state,
+                                                         1000 * NANOS_PER_MILLISECOND));
+    EXPECT_EQ(AAUDIO_STREAM_STATE_OPEN, state);
+
+    std::atomic<bool> isWaitThreadReady{false};
+
+    // Spawn a new thread to wait for the state change
+    std::thread waitThread (waitForStateChangeToClosingorClosed, &aaudioStream,
+                            &isWaitThreadReady);
+
+    // Wait for worker thread to be ready
+    while (!isWaitThreadReady) {
+        usleep(MICROS_PER_MILLISECOND);
+    }
+    // Sleep an additional millisecond to make sure waitForAudioThread is called
+    usleep(MICROS_PER_MILLISECOND);
+    EXPECT_EQ(AAUDIO_OK, AAudioStream_close(aaudioStream));
+    waitThread.join();
+}
+
+TEST(test_various, wait_for_state_change_close_none) {
+    testWaitForStateChangeClose(AAUDIO_PERFORMANCE_MODE_NONE);
+}
+
+TEST(test_various, wait_for_state_change_close_lowlat) {
+    testWaitForStateChangeClose(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY);
+}
+
 // ************************************************************
 struct WakeUpCallbackData {
     void wakeOther() {
diff --git a/media/libaudioclient/AudioTrack.cpp b/media/libaudioclient/AudioTrack.cpp
index ad00bdb..e96cf87 100644
--- a/media/libaudioclient/AudioTrack.cpp
+++ b/media/libaudioclient/AudioTrack.cpp
@@ -973,8 +973,16 @@
 {
     using namespace std::chrono_literals;
 
+    // We use atomic access here for state variables - these are used as hints
+    // to ensure we have ramped down audio.
+    const int priorState = mProxy->getState();
+    const uint32_t priorPosition = mProxy->getPosition().unsignedValue();
+
     pause();
 
+    // Only if we were previously active, do we wait to ramp down the audio.
+    if (priorState != CBLK_STATE_ACTIVE) return true;
+
     AutoMutex lock(mLock);
     // offload and direct tracks do not wait because pause volume ramp is handled by hardware.
     if (isOffloadedOrDirect_l()) return true;
@@ -982,16 +990,25 @@
     // Wait for the track state to be anything besides pausing.
     // This ensures that the volume has ramped down.
     constexpr auto SLEEP_INTERVAL_MS = 10ms;
+    constexpr auto POSITION_TIMEOUT_MS = 40ms; // don't wait longer than this for position change.
     auto begin = std::chrono::steady_clock::now();
     while (true) {
-        // wait for state to change
+        // Wait for state and position to change.
+        // After pause() the server state should be PAUSING, but that may immediately
+        // convert to PAUSED by prepareTracks before data is read into the mixer.
+        // Hence we check that the state is not PAUSING and that the server position
+        // has advanced to be a more reliable estimate that the volume ramp has completed.
         const int state = mProxy->getState();
+        const uint32_t position = mProxy->getPosition().unsignedValue();
 
         mLock.unlock(); // only local variables accessed until lock.
         auto elapsed = std::chrono::duration_cast<std::chrono::milliseconds>(
                 std::chrono::steady_clock::now() - begin);
-        if (state != CBLK_STATE_PAUSING) {
-            ALOGV("%s: success state:%d after %lld ms", __func__, state, elapsed.count());
+        if (state != CBLK_STATE_PAUSING &&
+                (elapsed >= POSITION_TIMEOUT_MS || position != priorPosition)) {
+            ALOGV("%s: success state:%d, position:%u after %lld ms"
+                    " (prior state:%d  prior position:%u)",
+                    __func__, state, position, elapsed.count(), priorState, priorPosition);
             return true;
         }
         std::chrono::milliseconds remaining = timeout - elapsed;
diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index a4fbbbc..7917395 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -2195,7 +2195,10 @@
             }
 
             if (!msg->findInt32("aac-max-output-channel_count", &maxOutputChannelCount)) {
-                maxOutputChannelCount = -1;
+                // check non AAC-specific key
+                if (!msg->findInt32("max-output-channel-count", &maxOutputChannelCount)) {
+                    maxOutputChannelCount = -1;
+                }
             }
             if (!msg->findInt32("aac-pcm-limiter-enable", &pcmLimiterEnable)) {
                 // value is unknown
diff --git a/media/libstagefright/include/media/stagefright/MediaCodecConstants.h b/media/libstagefright/include/media/stagefright/MediaCodecConstants.h
index 9040e8b..78792c5 100644
--- a/media/libstagefright/include/media/stagefright/MediaCodecConstants.h
+++ b/media/libstagefright/include/media/stagefright/MediaCodecConstants.h
@@ -809,6 +809,7 @@
 constexpr char KEY_MAX_FPS_TO_ENCODER[] = "max-fps-to-encoder";
 constexpr char KEY_MAX_HEIGHT[] = "max-height";
 constexpr char KEY_MAX_INPUT_SIZE[] = "max-input-size";
+constexpr char KEY_MAX_OUTPUT_CHANNEL_COUNT[] = "max-output-channel-count";
 constexpr char KEY_MAX_PTS_GAP_TO_ENCODER[] = "max-pts-gap-to-encoder";
 constexpr char KEY_MAX_WIDTH[] = "max-width";
 constexpr char KEY_MIME[] = "mime";
diff --git a/media/libstagefright/rtsp/ARTPConnection.cpp b/media/libstagefright/rtsp/ARTPConnection.cpp
index 0bd342a..83291f3 100644
--- a/media/libstagefright/rtsp/ARTPConnection.cpp
+++ b/media/libstagefright/rtsp/ARTPConnection.cpp
@@ -558,6 +558,7 @@
                 if (n != (ssize_t)buffer->size()) {
                     ALOGW("failed to send RTCP receiver report (%s).",
                             n >= 0 ? "connection gone" : strerror(errno));
+                    ++it;
                     continue;
                 }
 
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
index 5c3bdb3..1cea42f 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
@@ -577,6 +577,11 @@
     audio_output_flags_t halFlags = mFlags;
     if ((mFlags & AUDIO_OUTPUT_FLAG_SPATIALIZER) != 0) {
         halFlags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_FAST | AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
+        // If no mixer config is specified for a spatializer output, default to 5.1 for proper
+        // configuration of the final downmixer or spatializer
+        if (mixerConfig == nullptr) {
+            lMixerConfig.channel_mask = AUDIO_CHANNEL_OUT_5POINT1;
+        }
     }
 
     ALOGV("opening output for device %s profile %p name %s",
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index 3cfb944..04798b4 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -1390,7 +1390,8 @@
     }
 
     if (mSpatializerOutput != nullptr
-            && canBeSpatialized(attr, config, devices.toTypeAddrVector())) {
+            && canBeSpatializedInt(attr, config,
+                    devices.toTypeAddrVector(), false /* allowCurrentOutputReconfig */)) {
         return mSpatializerOutput->mIoHandle;
     }
 
@@ -3039,9 +3040,10 @@
             }
         }
     }
-    return mEffects.registerEffect(desc, io, session, id,
-                                   (strategy == streamToStrategy(AUDIO_STREAM_MUSIC) ||
-                                   strategy == PRODUCT_STRATEGY_NONE));
+    bool isMusicEffect = (session != AUDIO_SESSION_OUTPUT_STAGE)
+                            && ((strategy == streamToStrategy(AUDIO_STREAM_MUSIC)
+                                    || strategy == PRODUCT_STRATEGY_NONE));
+    return mEffects.registerEffect(desc, io, session, id, isMusicEffect);
 }
 
 status_t AudioPolicyManager::unregisterEffect(int id)
@@ -4882,9 +4884,10 @@
     }
 }
 
-bool AudioPolicyManager::canBeSpatialized(const audio_attributes_t *attr,
+bool AudioPolicyManager::canBeSpatializedInt(const audio_attributes_t *attr,
                                       const audio_config_t *config,
-                                      const AudioDeviceTypeAddrVector &devices)  const
+                                      const AudioDeviceTypeAddrVector &devices,
+                                      bool allowCurrentOutputReconfig)  const
 {
     // The caller can have the audio attributes criteria ignored by either passing a null ptr or
     // the AUDIO_ATTRIBUTES_INITIALIZER value.
@@ -4920,7 +4923,8 @@
         if (!isChannelMaskSpatialized(config->channel_mask)) {
             return false;
         }
-        if (mSpatializerOutput != nullptr && mSpatializerOutput->mProfile == profile) {
+        if (!allowCurrentOutputReconfig && mSpatializerOutput != nullptr
+                && mSpatializerOutput->mProfile == profile) {
             if ((config->channel_mask & mSpatializerOutput->mMixerChannelMask)
                     != config->channel_mask) {
                 return false;
@@ -4941,7 +4945,8 @@
             audio_config_base_t clientConfig = client->config();
             audio_config_t config = audio_config_initializer(&clientConfig);
             if (desc != mSpatializerOutput
-                    && canBeSpatialized(&attr, &config, devicesTypeAddress)) {
+                    && canBeSpatializedInt(&attr, &config,
+                            devicesTypeAddress, false /* allowCurrentOutputReconfig */)) {
                 streamsToInvalidate.insert(client->stream());
             }
         }
@@ -4965,7 +4970,8 @@
         config = audio_config_initializer(mixerConfig);
         configPtr = &config;
     }
-    if (!canBeSpatialized(attr, configPtr, devicesTypeAddress)) {
+    if (!canBeSpatializedInt(
+            attr, configPtr, devicesTypeAddress)) {
         ALOGW("%s provided attributes or mixer config cannot be spatialized", __func__);
         return BAD_VALUE;
     }
@@ -4988,6 +4994,7 @@
     for (size_t i = 0; i < mOutputs.size(); i++) {
         sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
         if (!desc->isDuplicated() && desc->mProfile == profile) {
+            ALOGV("%s found output %d for spatializer profile", __func__, desc->mIoHandle);
             mSpatializerOutput = desc;
             break;
         }
@@ -5007,39 +5014,29 @@
         };
         DeviceVector savedDevices = mSpatializerOutput->devices();
 
-        closeOutput(mSpatializerOutput->mIoHandle);
-        mSpatializerOutput.clear();
+        ALOGV("%s reopening spatializer output to match channel mask %#x (current mask %#x)",
+            __func__, configPtr->channel_mask, mSpatializerOutput->mMixerChannelMask);
 
-        const sp<SwAudioOutputDescriptor> desc =
-                new SwAudioOutputDescriptor(profile, mpClientInterface);
-        status_t status = desc->open(nullptr, mixerConfig, devices,
-                                                    mEngine->getStreamTypeForAttributes(*attr),
-                                                    AUDIO_OUTPUT_FLAG_SPATIALIZER, output);
-        if (status != NO_ERROR) {
-            ALOGW("%s failed opening output: status %d, output %d", __func__, status, *output);
-            if (*output != AUDIO_IO_HANDLE_NONE) {
-                desc->close();
-            }
+        closeOutput(mSpatializerOutput->mIoHandle);
+        //from now on mSpatializerOutput is null
+
+        sp<SwAudioOutputDescriptor> desc =
+                openOutputWithProfileAndDevice(profile, devices, mixerConfig);
+        if (desc == nullptr) {
             // re open the spatializer output with previous channel mask
-            status_t newStatus = desc->open(nullptr, &savedMixerConfig, savedDevices,
-                                mEngine->getStreamTypeForAttributes(*attr),
-                                AUDIO_OUTPUT_FLAG_SPATIALIZER, output);
-            if (newStatus != NO_ERROR) {
-                if (*output != AUDIO_IO_HANDLE_NONE) {
-                    desc->close();
-                }
-                ALOGE("%s failed to re-open mSpatializerOutput, status %d", __func__, newStatus);
+            desc = openOutputWithProfileAndDevice(profile, savedDevices, &savedMixerConfig);
+            if (desc == nullptr) {
+                ALOGE("%s failed to restore mSpatializerOutput with previous config", __func__);
             } else {
                 mSpatializerOutput = desc;
-                addOutput(*output, desc);
             }
             mPreviousOutputs = mOutputs;
             mpClientInterface->onAudioPortListUpdate();
             *output = AUDIO_IO_HANDLE_NONE;
-            return status;
+            ALOGW("%s could not open spatializer output with requested config", __func__);
+            return BAD_VALUE;
         }
         mSpatializerOutput = desc;
-        addOutput(*output, desc);
         mPreviousOutputs = mOutputs;
         mpClientInterface->onAudioPortListUpdate();
     }
@@ -5711,6 +5708,9 @@
 
     removeOutput(output);
     mPreviousOutputs = mOutputs;
+    if (closingOutput == mSpatializerOutput) {
+        mSpatializerOutput.clear();
+    }
 
     // MSD patches may have been released to support a non-MSD direct output. Reset MSD patch if
     // no direct outputs are open.
@@ -7269,7 +7269,8 @@
 }
 
 sp<SwAudioOutputDescriptor> AudioPolicyManager::openOutputWithProfileAndDevice(
-        const sp<IOProfile>& profile, const DeviceVector& devices)
+        const sp<IOProfile>& profile, const DeviceVector& devices,
+        const audio_config_base_t *mixerConfig)
 {
     for (const auto& device : devices) {
         // TODO: This should be checking if the profile supports the device combo.
@@ -7279,7 +7280,7 @@
     }
     sp<SwAudioOutputDescriptor> desc = new SwAudioOutputDescriptor(profile, mpClientInterface);
     audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
-    status_t status = desc->open(nullptr /* halConfig */, nullptr /* mixerConfig */, devices,
+    status_t status = desc->open(nullptr /* halConfig */, mixerConfig, devices,
             AUDIO_STREAM_DEFAULT, AUDIO_OUTPUT_FLAG_NONE, &output);
     if (status != NO_ERROR) {
         return nullptr;
@@ -7309,7 +7310,7 @@
         config.offload_info.channel_mask = config.channel_mask;
         config.offload_info.format = config.format;
 
-        status = desc->open(&config, nullptr /* mixerConfig */, devices,
+        status = desc->open(&config, mixerConfig, devices,
                             AUDIO_STREAM_DEFAULT, AUDIO_OUTPUT_FLAG_NONE, &output);
         if (status != NO_ERROR) {
             return nullptr;
@@ -7317,6 +7318,7 @@
     }
 
     addOutput(output, desc);
+
     if (audio_is_remote_submix_device(deviceType) && address != "0") {
         sp<AudioPolicyMix> policyMix;
         if (mPolicyMixes.getAudioPolicyMix(deviceType, address, policyMix) == NO_ERROR) {
@@ -7327,9 +7329,13 @@
                     address.string());
         }
 
-    } else if (((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) == 0) && hasPrimaryOutput()) {
-        // no duplicated output for direct outputs and
-        // outputs used by dynamic policy mixes
+    } else if (hasPrimaryOutput() && profile->getModule()
+                != mHwModules.getModuleFromName(AUDIO_HARDWARE_MODULE_ID_PRIMARY)
+            && ((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) == 0)) {
+        // no duplicated output for:
+        // - direct outputs
+        // - outputs used by dynamic policy mixes
+        // - outputs opened on the primary HW module
         audio_io_handle_t duplicatedOutput = AUDIO_IO_HANDLE_NONE;
 
         //TODO: configure audio effect output stage here
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h
index dcd12cd..317aa1c 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.h
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h
@@ -358,7 +358,9 @@
 
         virtual bool canBeSpatialized(const audio_attributes_t *attr,
                                       const audio_config_t *config,
-                                      const AudioDeviceTypeAddrVector &devices) const;
+                                      const AudioDeviceTypeAddrVector &devices) const {
+            return canBeSpatializedInt(attr, config, devices);
+        }
 
         virtual status_t getSpatializerOutput(const audio_config_base_t *config,
                                                 const audio_attributes_t *attr,
@@ -991,6 +993,30 @@
                 const DeviceVector &devices,
                 audio_io_handle_t *output);
 
+        /**
+         * @brief Queries if some kind of spatialization will be performed if the audio playback
+         * context described by the provided arguments is present.
+         * The context is made of:
+         * - The audio attributes describing the playback use case.
+         * - The audio configuration describing the audio format, channels, sampling rate ...
+         * - The devices describing the sink audio device selected for playback.
+         * All arguments are optional and only the specified arguments are used to match against
+         * supported criteria. For instance, supplying no argument will tell if spatialization is
+         * supported or not in general.
+         * @param attr audio attributes describing the playback use case
+         * @param config audio configuration describing the audio format, channels, sample rate...
+         * @param devices the sink audio device selected for playback
+         * @param allowCurrentOutputReconfig if true, the result will be considering it is possible
+         *      to close and reopen an existing spatializer output stream to match the requested
+         *      criteria. If false, the criteria must be compatible with the opened sptializer
+         *      output.
+         * @return true if spatialization is possible for this context, false otherwise.
+         */
+        virtual bool canBeSpatializedInt(const audio_attributes_t *attr,
+                                      const audio_config_t *config,
+                                      const AudioDeviceTypeAddrVector &devices,
+                                      bool allowCurrentOutputReconfig = true) const;
+
         sp<IOProfile> getSpatializerOutputProfile(const audio_config_t *config,
                                                   const AudioDeviceTypeAddrVector &devices) const;
 
@@ -1086,8 +1112,20 @@
 
         bool areAllActiveTracksRerouted(const sp<SwAudioOutputDescriptor>& output);
 
-        sp<SwAudioOutputDescriptor> openOutputWithProfileAndDevice(const sp<IOProfile>& profile,
-                                                                   const DeviceVector& devices);
+        /**
+         * @brief Opens an output stream from the supplied IOProfile and route it to the
+         * supplied audio devices. If a mixer config is specified, it is forwarded to audio
+         * flinger. If not, a default config is derived from the output stream config.
+         * Also opens a duplicating output if needed and queries the audio HAL for supported
+         * audio profiles if the IOProfile is dynamic.
+         * @param[in] profile IOProfile to use as template
+         * @param[in] devices initial route to apply to this output stream
+         * @param[in] mixerConfig if not null, use this to configure the mixer
+         * @return an output descriptor for the newly opened stream or null in case of error.
+         */
+        sp<SwAudioOutputDescriptor> openOutputWithProfileAndDevice(
+                const sp<IOProfile>& profile, const DeviceVector& devices,
+                const audio_config_base_t *mixerConfig = nullptr);
 
 };
 
diff --git a/services/audiopolicy/service/AudioPolicyService.cpp b/services/audiopolicy/service/AudioPolicyService.cpp
index cd83900..1109793 100644
--- a/services/audiopolicy/service/AudioPolicyService.cpp
+++ b/services/audiopolicy/service/AudioPolicyService.cpp
@@ -392,7 +392,8 @@
             audio_config_base_t config = mSpatializer->getAudioInConfig();
             status_t status =
                     mAudioPolicyManager->getSpatializerOutput(&config, &attr, &newOutput);
-
+            ALOGV("%s currentOutput %d newOutput %d channel_mask %#x",
+                    __func__, currentOutput, newOutput, config.channel_mask);
             if (status == NO_ERROR && currentOutput == newOutput) {
                 return;
             }
diff --git a/services/audiopolicy/service/Spatializer.cpp b/services/audiopolicy/service/Spatializer.cpp
index 0fdbe20..6745005 100644
--- a/services/audiopolicy/service/Spatializer.cpp
+++ b/services/audiopolicy/service/Spatializer.cpp
@@ -216,7 +216,9 @@
     if (status != NO_ERROR) {
         return status;
     }
-    mSupportsHeadTracking = supportsHeadTracking[0];
+// Disable head tracking until head sensor activity is properly controlled.
+//    mSupportsHeadTracking = supportsHeadTracking[0];
+    mSupportsHeadTracking = false;
 
     status = getHalParameter<true>(effect, SPATIALIZER_PARAM_SUPPORTED_LEVELS, &mLevels);
     if (status != NO_ERROR) {
@@ -227,12 +229,8 @@
     if (status != NO_ERROR) {
         return status;
     }
-    status = getHalParameter<true>(effect, SPATIALIZER_PARAM_SUPPORTED_CHANNEL_MASKS,
+    return getHalParameter<true>(effect, SPATIALIZER_PARAM_SUPPORTED_CHANNEL_MASKS,
                                  &mChannelMasks);
-    if (status != NO_ERROR) {
-        return status;
-    }
-    return NO_ERROR;
 }
 
 /** Gets the channel mask, sampling rate and format set for the spatializer input. */
@@ -727,7 +725,7 @@
     switch (event) {
         case AudioEffect::EVENT_FRAMES_PROCESSED: {
             int frames = info == nullptr ? 0 : *(int*)info;
-            ALOGD("%s frames processed %d for me %p", __func__, frames, me);
+            // ALOGD("%s frames processed %d for me %p", __func__, frames, me);
             me->postFramesProcessedMsg(frames);
         } break;
         default: