diff --git a/include/media/AudioMixer.h b/include/media/AudioMixer.h
index de839c6..85ee950 120000
--- a/include/media/AudioMixer.h
+++ b/include/media/AudioMixer.h
@@ -1 +1 @@
-../../media/libaudioclient/include/media/AudioMixer.h
\ No newline at end of file
+../../media/libaudioprocessing/include/media/AudioMixer.h
\ No newline at end of file
diff --git a/include/media/BufferProviders.h b/include/media/BufferProviders.h
index 779bb15..778e1d8 120000
--- a/include/media/BufferProviders.h
+++ b/include/media/BufferProviders.h
@@ -1 +1 @@
-../../media/libmedia/include/media/BufferProviders.h
\ No newline at end of file
+../../media/libaudioprocessing/include/media/BufferProviders.h
\ No newline at end of file
diff --git a/include/media/ExtendedAudioBufferProvider.h b/include/media/ExtendedAudioBufferProvider.h
index d653cc3..99d3c13 120000
--- a/include/media/ExtendedAudioBufferProvider.h
+++ b/include/media/ExtendedAudioBufferProvider.h
@@ -1 +1 @@
-../../media/libmedia/include/media/ExtendedAudioBufferProvider.h
\ No newline at end of file
+../../media/libaudioclient/include/media/ExtendedAudioBufferProvider.h
\ No newline at end of file
diff --git a/include/media/SingleStateQueue.h b/include/media/SingleStateQueue.h
deleted file mode 120000
index 619f6ee..0000000
--- a/include/media/SingleStateQueue.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/SingleStateQueue.h
\ No newline at end of file
diff --git a/include/media/nbaio/SingleStateQueue.h b/include/media/nbaio/SingleStateQueue.h
new file mode 120000
index 0000000..d3e0553
--- /dev/null
+++ b/include/media/nbaio/SingleStateQueue.h
@@ -0,0 +1 @@
+../../../media/libnbaio/include_mono/media/nbaio/SingleStateQueue.h
\ No newline at end of file
diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index 5f19f74..1b1f149 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -28,7 +28,7 @@
 #include <media/AudioResamplerPublic.h>
 #include <media/AudioTimestamp.h>
 #include <media/Modulo.h>
-#include <media/SingleStateQueue.h>
+#include <media/nbaio/SingleStateQueue.h>
 
 namespace android {
 
diff --git a/media/audioserver/Android.mk b/media/audioserver/Android.mk
index 969f2ee..e11af12 100644
--- a/media/audioserver/Android.mk
+++ b/media/audioserver/Android.mk
@@ -9,6 +9,7 @@
 	libaaudioservice \
 	libaudioflinger \
 	libaudiopolicyservice \
+	libaudioprocessing \
 	libbinder \
 	libcutils \
 	liblog \
@@ -39,7 +40,6 @@
 	frameworks/av/media/libaaudio/src \
 	frameworks/av/media/libaaudio/src/binding \
 	frameworks/av/media/libmedia \
-	$(call include-path-for, audio-utils) \
 	external/sonic \
 
 # If AUDIOSERVER_MULTILIB in device.mk is non-empty then it is used to control
diff --git a/media/bufferpool/1.0/AccessorImpl.cpp b/media/bufferpool/1.0/AccessorImpl.cpp
index fa17f15..6b90088 100644
--- a/media/bufferpool/1.0/AccessorImpl.cpp
+++ b/media/bufferpool/1.0/AccessorImpl.cpp
@@ -247,7 +247,7 @@
     ALOGD("Destruction - bufferpool %p "
           "cached: %zu/%zuM, %zu/%d%% in use; "
           "allocs: %zu, %d%% recycled; "
-          "transfers: %zu, %d%% unfetced",
+          "transfers: %zu, %d%% unfetched",
           this, mStats.mBuffersCached, mStats.mSizeCached >> 20,
           mStats.mBuffersInUse, percentage(mStats.mBuffersInUse, mStats.mBuffersCached),
           mStats.mTotalAllocations, percentage(mStats.mTotalRecycles, mStats.mTotalAllocations),
diff --git a/media/bufferpool/2.0/AccessorImpl.cpp b/media/bufferpool/2.0/AccessorImpl.cpp
index 94cf006..32eaae9 100644
--- a/media/bufferpool/2.0/AccessorImpl.cpp
+++ b/media/bufferpool/2.0/AccessorImpl.cpp
@@ -303,7 +303,7 @@
     ALOGD("Destruction - bufferpool2 %p "
           "cached: %zu/%zuM, %zu/%d%% in use; "
           "allocs: %zu, %d%% recycled; "
-          "transfers: %zu, %d%% unfetced",
+          "transfers: %zu, %d%% unfetched",
           this, mStats.mBuffersCached, mStats.mSizeCached >> 20,
           mStats.mBuffersInUse, percentage(mStats.mBuffersInUse, mStats.mBuffersCached),
           mStats.mTotalAllocations, percentage(mStats.mTotalRecycles, mStats.mTotalAllocations),
diff --git a/media/bufferpool/2.0/ClientManager.cpp b/media/bufferpool/2.0/ClientManager.cpp
index c31d313..48c2da4 100644
--- a/media/bufferpool/2.0/ClientManager.cpp
+++ b/media/bufferpool/2.0/ClientManager.cpp
@@ -351,7 +351,17 @@
         }
         client = it->second;
     }
-    return client->allocate(params, handle, buffer);
+    native_handle_t *origHandle;
+    ResultStatus res = client->allocate(params, &origHandle, buffer);
+    if (res != ResultStatus::OK) {
+        return res;
+    }
+    *handle = native_handle_clone(origHandle);
+    if (handle == NULL) {
+        buffer->reset();
+        return ResultStatus::NO_MEMORY;
+    }
+    return ResultStatus::OK;
 }
 
 ResultStatus ClientManager::Impl::receive(
@@ -367,7 +377,18 @@
         }
         client = it->second;
     }
-    return client->receive(transactionId, bufferId, timestampUs, handle, buffer);
+    native_handle_t *origHandle;
+    ResultStatus res = client->receive(
+            transactionId, bufferId, timestampUs, &origHandle, buffer);
+    if (res != ResultStatus::OK) {
+        return res;
+    }
+    *handle = native_handle_clone(origHandle);
+    if (handle == NULL) {
+        buffer->reset();
+        return ResultStatus::NO_MEMORY;
+    }
+    return ResultStatus::OK;
 }
 
 ResultStatus ClientManager::Impl::postSend(
diff --git a/media/bufferpool/2.0/include/bufferpool/ClientManager.h b/media/bufferpool/2.0/include/bufferpool/ClientManager.h
index 953c304..24b61f4 100644
--- a/media/bufferpool/2.0/include/bufferpool/ClientManager.h
+++ b/media/bufferpool/2.0/include/bufferpool/ClientManager.h
@@ -104,7 +104,9 @@
     ResultStatus flush(ConnectionId connectionId);
 
     /**
-     * Allocates a buffer from the specified connection.
+     * Allocates a buffer from the specified connection. The output parameter
+     * handle is cloned from the internal handle. So it is safe to use directly,
+     * and it should be deleted and destroyed after use.
      *
      * @param connectionId  The id of the connection.
      * @param params        The allocation parameters.
@@ -123,7 +125,9 @@
                           std::shared_ptr<BufferPoolData> *buffer);
 
     /**
-     * Receives a buffer for the transaction.
+     * Receives a buffer for the transaction. The output parameter handle is
+     * cloned from the internal handle. So it is safe to use directly, and it
+     * should be deleted and destoyed after use.
      *
      * @param connectionId  The id of the receiving connection.
      * @param transactionId The id for the transaction.
diff --git a/media/codec2/components/aac/C2SoftAacEnc.cpp b/media/codec2/components/aac/C2SoftAacEnc.cpp
index 1dc676b..a8f39d5 100644
--- a/media/codec2/components/aac/C2SoftAacEnc.cpp
+++ b/media/codec2/components/aac/C2SoftAacEnc.cpp
@@ -159,7 +159,8 @@
       mInputSize(0),
       mNextFrameTimestampUs(0),
       mSignalledError(false),
-      mOutIndex(0u) {
+      mOutIndex(0u),
+      mRemainderLen(0u) {
 }
 
 C2SoftAacEnc::~C2SoftAacEnc() {
@@ -185,6 +186,7 @@
     mInputSize = 0u;
     mNextFrameTimestampUs = 0;
     mSignalledError = false;
+    mRemainderLen = 0;
     return C2_OK;
 }
 
@@ -369,18 +371,21 @@
         mInputTimeSet = true;
     }
 
-    size_t numFrames = (capacity + mInputSize + (eos ? mNumBytesPerInputFrame - 1 : 0))
-            / mNumBytesPerInputFrame;
+    size_t numFrames =
+        (mRemainderLen + capacity + mInputSize + (eos ? mNumBytesPerInputFrame - 1 : 0))
+        / mNumBytesPerInputFrame;
     ALOGV("capacity = %zu; mInputSize = %zu; numFrames = %zu "
-          "mNumBytesPerInputFrame = %u inputTS = %lld",
+          "mNumBytesPerInputFrame = %u inputTS = %lld remaining = %zu",
           capacity, mInputSize, numFrames,
-          mNumBytesPerInputFrame, work->input.ordinal.timestamp.peekll());
+          mNumBytesPerInputFrame, work->input.ordinal.timestamp.peekll(),
+          mRemainderLen);
 
     std::shared_ptr<C2LinearBlock> block;
     std::unique_ptr<C2WriteView> wView;
     uint8_t *outPtr = temp;
     size_t outAvailable = 0u;
     uint64_t inputIndex = work->input.ordinal.frameIndex.peeku();
+    size_t bytesPerSample = channelCount * sizeof(int16_t);
 
     AACENC_InArgs inargs;
     AACENC_OutArgs outargs;
@@ -449,7 +454,25 @@
     };
     std::list<OutputBuffer> outputBuffers;
 
-    while (encoderErr == AACENC_OK && inargs.numInSamples > 0) {
+    if (mRemainderLen > 0) {
+        size_t offset = 0;
+        for (; mRemainderLen < bytesPerSample && offset < capacity; ++offset) {
+            mRemainder[mRemainderLen++] = data[offset];
+        }
+        data += offset;
+        capacity -= offset;
+        if (mRemainderLen == bytesPerSample) {
+            inBuffer[0] = mRemainder;
+            inBufferSize[0] = bytesPerSample;
+            inargs.numInSamples = channelCount;
+            mRemainderLen = 0;
+            ALOGV("Processing remainder");
+        } else {
+            // We have exhausted the input already
+            inargs.numInSamples = 0;
+        }
+    }
+    while (encoderErr == AACENC_OK && inargs.numInSamples >= channelCount) {
         if (numFrames && !block) {
             C2MemoryUsage usage = { C2MemoryUsage::CPU_READ, C2MemoryUsage::CPU_WRITE };
             // TODO: error handling, proper usage, etc.
@@ -486,7 +509,7 @@
                 mNextFrameTimestampUs = work->input.ordinal.timestamp
                         + (consumed * 1000000ll / channelCount / sampleRate);
                 std::shared_ptr<C2Buffer> buffer = createLinearBuffer(block, 0, outargs.numOutBytes);
-#if defined(LOG_NDEBUG) && !LOG_NDEBUG
+#if 0
                 hexdump(outPtr, std::min(outargs.numOutBytes, 256));
 #endif
                 outPtr = temp;
@@ -503,12 +526,17 @@
                 inBufferSize[0] -= outargs.numInSamples * sizeof(int16_t);
                 inargs.numInSamples -= outargs.numInSamples;
             }
+
+            if (inBuffer[0] == mRemainder) {
+                inBuffer[0] = const_cast<uint8_t *>(data);
+                inBufferSize[0] = capacity;
+                inargs.numInSamples = capacity / sizeof(int16_t);
+            }
         }
         ALOGV("encoderErr = %d mInputSize = %zu "
               "inargs.numInSamples = %d, mNextFrameTimestampUs = %lld",
               encoderErr, mInputSize, inargs.numInSamples, mNextFrameTimestampUs.peekll());
     }
-
     if (eos && inBufferSize[0] > 0) {
         if (numFrames && !block) {
             C2MemoryUsage usage = { C2MemoryUsage::CPU_READ, C2MemoryUsage::CPU_WRITE };
@@ -539,6 +567,14 @@
                            &outBufDesc,
                            &inargs,
                            &outargs);
+        inBufferSize[0] = 0;
+    }
+
+    if (inBufferSize[0] > 0) {
+        for (size_t i = 0; i < inBufferSize[0]; ++i) {
+            mRemainder[i] = static_cast<uint8_t *>(inBuffer[0])[i];
+        }
+        mRemainderLen = inBufferSize[0];
     }
 
     while (outputBuffers.size() > 1) {
diff --git a/media/codec2/components/aac/C2SoftAacEnc.h b/media/codec2/components/aac/C2SoftAacEnc.h
index 2655039..6ecfbdd 100644
--- a/media/codec2/components/aac/C2SoftAacEnc.h
+++ b/media/codec2/components/aac/C2SoftAacEnc.h
@@ -61,6 +61,10 @@
     bool mSignalledError;
     std::atomic_uint64_t mOutIndex;
 
+    // We support max 6 channels
+    uint8_t mRemainder[6 * sizeof(int16_t)];
+    size_t mRemainderLen;
+
     status_t initEncoder();
 
     status_t setAudioParams();
diff --git a/media/codec2/components/mpeg2/C2SoftMpeg2Dec.cpp b/media/codec2/components/mpeg2/C2SoftMpeg2Dec.cpp
index df7b403..a1f8ff3 100644
--- a/media/codec2/components/mpeg2/C2SoftMpeg2Dec.cpp
+++ b/media/codec2/components/mpeg2/C2SoftMpeg2Dec.cpp
@@ -47,6 +47,12 @@
         noInputLatency();
         noTimeStretch();
 
+        // TODO: Proper support for reorder depth.
+        addParameter(
+                DefineParam(mActualOutputDelay, C2_PARAMKEY_OUTPUT_DELAY)
+                .withConstValue(new C2PortActualDelayTuning::output(3u))
+                .build());
+
         // TODO: output latency and reordering
 
         addParameter(
diff --git a/media/codec2/hidl/1.0/utils/InputBufferManager.cpp b/media/codec2/hidl/1.0/utils/InputBufferManager.cpp
index a023a05..8c0d0a4 100644
--- a/media/codec2/hidl/1.0/utils/InputBufferManager.cpp
+++ b/media/codec2/hidl/1.0/utils/InputBufferManager.cpp
@@ -70,7 +70,7 @@
                  << ".";
     std::lock_guard<std::mutex> lock(mMutex);
 
-    std::set<TrackedBuffer> &bufferIds =
+    std::set<TrackedBuffer*> &bufferIds =
             mTrackedBuffersMap[listener][frameIndex];
 
     for (size_t i = 0; i < input.buffers.size(); ++i) {
@@ -79,13 +79,14 @@
                          << "Input buffer at index " << i << " is null.";
             continue;
         }
-        const TrackedBuffer &bufferId =
-                *bufferIds.emplace(listener, frameIndex, i, input.buffers[i]).
-                first;
+        TrackedBuffer *bufferId =
+            new TrackedBuffer(listener, frameIndex, i, input.buffers[i]);
+        mTrackedBufferCache.emplace(bufferId);
+        bufferIds.emplace(bufferId);
 
         c2_status_t status = input.buffers[i]->registerOnDestroyNotify(
                 onBufferDestroyed,
-                const_cast<void*>(reinterpret_cast<const void*>(&bufferId)));
+                reinterpret_cast<void*>(bufferId));
         if (status != C2_OK) {
             LOG(DEBUG) << "InputBufferManager::_registerFrameData -- "
                        << "registerOnDestroyNotify() failed "
@@ -119,31 +120,32 @@
 
     auto findListener = mTrackedBuffersMap.find(listener);
     if (findListener != mTrackedBuffersMap.end()) {
-        std::map<uint64_t, std::set<TrackedBuffer>> &frameIndex2BufferIds
+        std::map<uint64_t, std::set<TrackedBuffer*>> &frameIndex2BufferIds
                 = findListener->second;
         auto findFrameIndex = frameIndex2BufferIds.find(frameIndex);
         if (findFrameIndex != frameIndex2BufferIds.end()) {
-            std::set<TrackedBuffer> &bufferIds = findFrameIndex->second;
-            for (const TrackedBuffer& bufferId : bufferIds) {
-                std::shared_ptr<C2Buffer> buffer = bufferId.buffer.lock();
+            std::set<TrackedBuffer*> &bufferIds = findFrameIndex->second;
+            for (TrackedBuffer* bufferId : bufferIds) {
+                std::shared_ptr<C2Buffer> buffer = bufferId->buffer.lock();
                 if (buffer) {
                     c2_status_t status = buffer->unregisterOnDestroyNotify(
                             onBufferDestroyed,
-                            const_cast<void*>(
-                            reinterpret_cast<const void*>(&bufferId)));
+                            reinterpret_cast<void*>(bufferId));
                     if (status != C2_OK) {
                         LOG(DEBUG) << "InputBufferManager::_unregisterFrameData "
                                    << "-- unregisterOnDestroyNotify() failed "
                                    << "(listener @ 0x"
                                         << std::hex
-                                        << bufferId.listener.unsafe_get()
+                                        << bufferId->listener.unsafe_get()
                                    << ", frameIndex = "
-                                        << std::dec << bufferId.frameIndex
-                                   << ", bufferIndex = " << bufferId.bufferIndex
+                                        << std::dec << bufferId->frameIndex
+                                   << ", bufferIndex = " << bufferId->bufferIndex
                                    << ") => status = " << status
                                    << ".";
                     }
                 }
+                mTrackedBufferCache.erase(bufferId);
+                delete bufferId;
             }
 
             frameIndex2BufferIds.erase(findFrameIndex);
@@ -179,31 +181,32 @@
 
     auto findListener = mTrackedBuffersMap.find(listener);
     if (findListener != mTrackedBuffersMap.end()) {
-        std::map<uint64_t, std::set<TrackedBuffer>> &frameIndex2BufferIds =
+        std::map<uint64_t, std::set<TrackedBuffer*>> &frameIndex2BufferIds =
                 findListener->second;
         for (auto findFrameIndex = frameIndex2BufferIds.begin();
                 findFrameIndex != frameIndex2BufferIds.end();
                 ++findFrameIndex) {
-            std::set<TrackedBuffer> &bufferIds = findFrameIndex->second;
-            for (const TrackedBuffer& bufferId : bufferIds) {
-                std::shared_ptr<C2Buffer> buffer = bufferId.buffer.lock();
+            std::set<TrackedBuffer*> &bufferIds = findFrameIndex->second;
+            for (TrackedBuffer* bufferId : bufferIds) {
+                std::shared_ptr<C2Buffer> buffer = bufferId->buffer.lock();
                 if (buffer) {
                     c2_status_t status = buffer->unregisterOnDestroyNotify(
                             onBufferDestroyed,
-                            const_cast<void*>(
-                            reinterpret_cast<const void*>(&bufferId)));
+                            reinterpret_cast<void*>(bufferId));
                     if (status != C2_OK) {
                         LOG(DEBUG) << "InputBufferManager::_unregisterFrameData "
                                    << "-- unregisterOnDestroyNotify() failed "
                                    << "(listener @ 0x"
                                         << std::hex
-                                        << bufferId.listener.unsafe_get()
+                                        << bufferId->listener.unsafe_get()
                                    << ", frameIndex = "
-                                        << std::dec << bufferId.frameIndex
-                                   << ", bufferIndex = " << bufferId.bufferIndex
+                                        << std::dec << bufferId->frameIndex
+                                   << ", bufferIndex = " << bufferId->bufferIndex
                                    << ") => status = " << status
                                    << ".";
                     }
+                    mTrackedBufferCache.erase(bufferId);
+                    delete bufferId;
                 }
             }
         }
@@ -236,50 +239,59 @@
                      << std::dec << ".";
         return;
     }
-    TrackedBuffer id(*reinterpret_cast<TrackedBuffer*>(arg));
+
+    std::lock_guard<std::mutex> lock(mMutex);
+    TrackedBuffer *bufferId = reinterpret_cast<TrackedBuffer*>(arg);
+
+    if (mTrackedBufferCache.find(bufferId) == mTrackedBufferCache.end()) {
+        LOG(VERBOSE) << "InputBufferManager::_onBufferDestroyed -- called with "
+                     << "unregistered buffer: "
+                     << "buf @ 0x" << std::hex << buf
+                     << ", arg @ 0x" << std::hex << arg
+                     << std::dec << ".";
+        return;
+    }
+
     LOG(VERBOSE) << "InputBufferManager::_onBufferDestroyed -- called with "
                  << "buf @ 0x" << std::hex << buf
                  << ", arg @ 0x" << std::hex << arg
                  << std::dec << " -- "
-                 << "listener @ 0x" << std::hex << id.listener.unsafe_get()
-                 << ", frameIndex = " << std::dec << id.frameIndex
-                 << ", bufferIndex = " << id.bufferIndex
+                 << "listener @ 0x" << std::hex << bufferId->listener.unsafe_get()
+                 << ", frameIndex = " << std::dec << bufferId->frameIndex
+                 << ", bufferIndex = " << bufferId->bufferIndex
                  << ".";
-
-    std::lock_guard<std::mutex> lock(mMutex);
-
-    auto findListener = mTrackedBuffersMap.find(id.listener);
+    auto findListener = mTrackedBuffersMap.find(bufferId->listener);
     if (findListener == mTrackedBuffersMap.end()) {
-        LOG(DEBUG) << "InputBufferManager::_onBufferDestroyed -- "
-                   << "received invalid listener: "
-                   << "listener @ 0x" << std::hex << id.listener.unsafe_get()
-                   << " (frameIndex = " << std::dec << id.frameIndex
-                   << ", bufferIndex = " << id.bufferIndex
-                   << ").";
+        LOG(VERBOSE) << "InputBufferManager::_onBufferDestroyed -- "
+                     << "received invalid listener: "
+                     << "listener @ 0x" << std::hex << bufferId->listener.unsafe_get()
+                     << " (frameIndex = " << std::dec << bufferId->frameIndex
+                     << ", bufferIndex = " << bufferId->bufferIndex
+                     << ").";
         return;
     }
 
-    std::map<uint64_t, std::set<TrackedBuffer>> &frameIndex2BufferIds
+    std::map<uint64_t, std::set<TrackedBuffer*>> &frameIndex2BufferIds
             = findListener->second;
-    auto findFrameIndex = frameIndex2BufferIds.find(id.frameIndex);
+    auto findFrameIndex = frameIndex2BufferIds.find(bufferId->frameIndex);
     if (findFrameIndex == frameIndex2BufferIds.end()) {
         LOG(DEBUG) << "InputBufferManager::_onBufferDestroyed -- "
                    << "received invalid frame index: "
-                   << "frameIndex = " << id.frameIndex
-                   << " (listener @ 0x" << std::hex << id.listener.unsafe_get()
-                   << ", bufferIndex = " << std::dec << id.bufferIndex
+                   << "frameIndex = " << bufferId->frameIndex
+                   << " (listener @ 0x" << std::hex << bufferId->listener.unsafe_get()
+                   << ", bufferIndex = " << std::dec << bufferId->bufferIndex
                    << ").";
         return;
     }
 
-    std::set<TrackedBuffer> &bufferIds = findFrameIndex->second;
-    auto findBufferId = bufferIds.find(id);
+    std::set<TrackedBuffer*> &bufferIds = findFrameIndex->second;
+    auto findBufferId = bufferIds.find(bufferId);
     if (findBufferId == bufferIds.end()) {
         LOG(DEBUG) << "InputBufferManager::_onBufferDestroyed -- "
                    << "received invalid buffer index: "
-                   << "bufferIndex = " << id.bufferIndex
-                   << " (frameIndex = " << id.frameIndex
-                   << ", listener @ 0x" << std::hex << id.listener.unsafe_get()
+                   << "bufferIndex = " << bufferId->bufferIndex
+                   << " (frameIndex = " << bufferId->frameIndex
+                   << ", listener @ 0x" << std::hex << bufferId->listener.unsafe_get()
                    << std::dec << ").";
         return;
     }
@@ -292,10 +304,13 @@
         }
     }
 
-    DeathNotifications &deathNotifications = mDeathNotifications[id.listener];
-    deathNotifications.indices[id.frameIndex].emplace_back(id.bufferIndex);
+    DeathNotifications &deathNotifications = mDeathNotifications[bufferId->listener];
+    deathNotifications.indices[bufferId->frameIndex].emplace_back(bufferId->bufferIndex);
     ++deathNotifications.count;
     mOnBufferDestroyed.notify_one();
+
+    mTrackedBufferCache.erase(bufferId);
+    delete bufferId;
 }
 
 // Notify the clients about buffer destructions.
diff --git a/media/codec2/hidl/1.0/utils/include/codec2/hidl/1.0/InputBufferManager.h b/media/codec2/hidl/1.0/utils/include/codec2/hidl/1.0/InputBufferManager.h
index b6857d5..42fa557 100644
--- a/media/codec2/hidl/1.0/utils/include/codec2/hidl/1.0/InputBufferManager.h
+++ b/media/codec2/hidl/1.0/utils/include/codec2/hidl/1.0/InputBufferManager.h
@@ -196,13 +196,9 @@
                 frameIndex(frameIndex),
                 bufferIndex(bufferIndex),
                 buffer(buffer) {}
-        TrackedBuffer(const TrackedBuffer&) = default;
-        bool operator<(const TrackedBuffer& other) const {
-            return bufferIndex < other.bufferIndex;
-        }
     };
 
-    // Map: listener -> frameIndex -> set<TrackedBuffer>.
+    // Map: listener -> frameIndex -> set<TrackedBuffer*>.
     // Essentially, this is used to store triples (listener, frameIndex,
     // bufferIndex) that's searchable by listener and (listener, frameIndex).
     // However, the value of the innermost map is TrackedBuffer, which also
@@ -210,7 +206,7 @@
     // because onBufferDestroyed() needs to know listener and frameIndex too.
     typedef std::map<wp<IComponentListener>,
                      std::map<uint64_t,
-                              std::set<TrackedBuffer>>> TrackedBuffersMap;
+                              std::set<TrackedBuffer*>>> TrackedBuffersMap;
 
     // Storage for pending (unsent) death notifications for one listener.
     // Each pair in member named "indices" are (frameIndex, bufferIndex) from
@@ -247,6 +243,16 @@
     // Mutex for the management of all input buffers.
     std::mutex mMutex;
 
+    // Cache for all TrackedBuffers.
+    //
+    // Whenever registerOnDestroyNotify() is called, an argument of type
+    // TrackedBuffer is created and stored into this cache.
+    // Whenever unregisterOnDestroyNotify() or onBufferDestroyed() is called,
+    // the TrackedBuffer is removed from this cache.
+    //
+    // mTrackedBuffersMap stores references to TrackedBuffers inside this cache.
+    std::set<TrackedBuffer*> mTrackedBufferCache;
+
     // Tracked input buffers.
     TrackedBuffersMap mTrackedBuffersMap;
 
diff --git a/media/codec2/hidl/1.0/utils/types.cpp b/media/codec2/hidl/1.0/utils/types.cpp
index 07dbf67..04fa59c 100644
--- a/media/codec2/hidl/1.0/utils/types.cpp
+++ b/media/codec2/hidl/1.0/utils/types.cpp
@@ -1434,6 +1434,11 @@
                 d->type = C2BaseBlock::GRAPHIC;
                 return true;
             }
+            if (cHandle) {
+                // Though we got cloned handle, creating block failed.
+                native_handle_close(cHandle);
+                native_handle_delete(cHandle);
+            }
 
             LOG(ERROR) << "Unknown handle type in BaseBlock::pooledBlock.";
             return false;
diff --git a/media/codec2/vndk/C2Buffer.cpp b/media/codec2/vndk/C2Buffer.cpp
index 710b536..2d99b53 100644
--- a/media/codec2/vndk/C2Buffer.cpp
+++ b/media/codec2/vndk/C2Buffer.cpp
@@ -413,17 +413,14 @@
 
     std::shared_ptr<C2LinearAllocation> alloc;
     if (C2AllocatorIon::isValid(cHandle)) {
-        native_handle_t *handle = native_handle_clone(cHandle);
-        if (handle) {
-            c2_status_t err = sAllocator->priorLinearAllocation(handle, &alloc);
-            const std::shared_ptr<C2PooledBlockPoolData> poolData =
-                    std::make_shared<C2PooledBlockPoolData>(data);
-            if (err == C2_OK && poolData) {
-                // TODO: config params?
-                std::shared_ptr<C2LinearBlock> block =
-                        _C2BlockFactory::CreateLinearBlock(alloc, poolData);
-                return block;
-            }
+        c2_status_t err = sAllocator->priorLinearAllocation(cHandle, &alloc);
+        const std::shared_ptr<C2PooledBlockPoolData> poolData =
+                std::make_shared<C2PooledBlockPoolData>(data);
+        if (err == C2_OK && poolData) {
+            // TODO: config params?
+            std::shared_ptr<C2LinearBlock> block =
+                    _C2BlockFactory::CreateLinearBlock(alloc, poolData);
+            return block;
         }
     }
     return nullptr;
@@ -674,17 +671,14 @@
         ResultStatus status = mBufferPoolManager->allocate(
                 mConnectionId, params, &cHandle, &bufferPoolData);
         if (status == ResultStatus::OK) {
-            native_handle_t *handle = native_handle_clone(cHandle);
-            if (handle) {
-                std::shared_ptr<C2LinearAllocation> alloc;
-                std::shared_ptr<C2PooledBlockPoolData> poolData =
-                        std::make_shared<C2PooledBlockPoolData>(bufferPoolData);
-                c2_status_t err = mAllocator->priorLinearAllocation(handle, &alloc);
-                if (err == C2_OK && poolData && alloc) {
-                    *block = _C2BlockFactory::CreateLinearBlock(alloc, poolData, 0, capacity);
-                    if (*block) {
-                        return C2_OK;
-                    }
+            std::shared_ptr<C2LinearAllocation> alloc;
+            std::shared_ptr<C2PooledBlockPoolData> poolData =
+                    std::make_shared<C2PooledBlockPoolData>(bufferPoolData);
+            c2_status_t err = mAllocator->priorLinearAllocation(cHandle, &alloc);
+            if (err == C2_OK && poolData && alloc) {
+                *block = _C2BlockFactory::CreateLinearBlock(alloc, poolData, 0, capacity);
+                if (*block) {
+                    return C2_OK;
                 }
             }
             return C2_NO_MEMORY;
@@ -710,19 +704,16 @@
         ResultStatus status = mBufferPoolManager->allocate(
                 mConnectionId, params, &cHandle, &bufferPoolData);
         if (status == ResultStatus::OK) {
-            native_handle_t *handle = native_handle_clone(cHandle);
-            if (handle) {
-                std::shared_ptr<C2GraphicAllocation> alloc;
-                std::shared_ptr<C2PooledBlockPoolData> poolData =
-                    std::make_shared<C2PooledBlockPoolData>(bufferPoolData);
-                c2_status_t err = mAllocator->priorGraphicAllocation(
-                        handle, &alloc);
-                if (err == C2_OK && poolData && alloc) {
-                    *block = _C2BlockFactory::CreateGraphicBlock(
-                            alloc, poolData, C2Rect(width, height));
-                    if (*block) {
-                        return C2_OK;
-                    }
+            std::shared_ptr<C2GraphicAllocation> alloc;
+            std::shared_ptr<C2PooledBlockPoolData> poolData =
+                std::make_shared<C2PooledBlockPoolData>(bufferPoolData);
+            c2_status_t err = mAllocator->priorGraphicAllocation(
+                    cHandle, &alloc);
+            if (err == C2_OK && poolData && alloc) {
+                *block = _C2BlockFactory::CreateGraphicBlock(
+                        alloc, poolData, C2Rect(width, height));
+                if (*block) {
+                    return C2_OK;
                 }
             }
             return C2_NO_MEMORY;
@@ -1117,17 +1108,14 @@
 
     std::shared_ptr<C2GraphicAllocation> alloc;
     if (C2AllocatorGralloc::isValid(cHandle)) {
-        native_handle_t *handle = native_handle_clone(cHandle);
-        if (handle) {
-            c2_status_t err = sAllocator->priorGraphicAllocation(handle, &alloc);
-            const std::shared_ptr<C2PooledBlockPoolData> poolData =
-                    std::make_shared<C2PooledBlockPoolData>(data);
-            if (err == C2_OK && poolData) {
-                // TODO: config setup?
-                std::shared_ptr<C2GraphicBlock> block =
-                        _C2BlockFactory::CreateGraphicBlock(alloc, poolData);
-                return block;
-            }
+        c2_status_t err = sAllocator->priorGraphicAllocation(cHandle, &alloc);
+        const std::shared_ptr<C2PooledBlockPoolData> poolData =
+                std::make_shared<C2PooledBlockPoolData>(data);
+        if (err == C2_OK && poolData) {
+            // TODO: config setup?
+            std::shared_ptr<C2GraphicBlock> block =
+                    _C2BlockFactory::CreateGraphicBlock(alloc, poolData);
+            return block;
         }
     }
     return nullptr;
diff --git a/media/libaudioclient/include/media/AudioMixer.h b/media/libaudioclient/include/media/AudioMixer.h
deleted file mode 100644
index 783eef3..0000000
--- a/media/libaudioclient/include/media/AudioMixer.h
+++ /dev/null
@@ -1,519 +0,0 @@
-/*
-**
-** Copyright 2007, 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.
-*/
-
-#ifndef ANDROID_AUDIO_MIXER_H
-#define ANDROID_AUDIO_MIXER_H
-
-#include <map>
-#include <pthread.h>
-#include <sstream>
-#include <stdint.h>
-#include <sys/types.h>
-#include <unordered_map>
-#include <vector>
-
-#include <android/os/IExternalVibratorService.h>
-#include <media/AudioBufferProvider.h>
-#include <media/AudioResampler.h>
-#include <media/AudioResamplerPublic.h>
-#include <media/BufferProviders.h>
-#include <system/audio.h>
-#include <utils/Compat.h>
-#include <utils/threads.h>
-
-// FIXME This is actually unity gain, which might not be max in future, expressed in U.12
-#define MAX_GAIN_INT AudioMixer::UNITY_GAIN_INT
-
-// This must match frameworks/av/services/audioflinger/Configuration.h
-#define FLOAT_AUX
-
-namespace android {
-
-namespace NBLog {
-class Writer;
-}   // namespace NBLog
-
-// ----------------------------------------------------------------------------
-
-class AudioMixer
-{
-public:
-    // Do not change these unless underlying code changes.
-    // This mixer has a hard-coded upper limit of 8 channels for output.
-    static constexpr uint32_t MAX_NUM_CHANNELS = FCC_8;
-    static constexpr uint32_t MAX_NUM_VOLUMES = FCC_2; // stereo volume only
-    // maximum number of channels supported for the content
-    static const uint32_t MAX_NUM_CHANNELS_TO_DOWNMIX = AUDIO_CHANNEL_COUNT_MAX;
-
-    static const uint16_t UNITY_GAIN_INT = 0x1000;
-    static const CONSTEXPR float UNITY_GAIN_FLOAT = 1.0f;
-
-    enum { // names
-        // setParameter targets
-        TRACK           = 0x3000,
-        RESAMPLE        = 0x3001,
-        RAMP_VOLUME     = 0x3002, // ramp to new volume
-        VOLUME          = 0x3003, // don't ramp
-        TIMESTRETCH     = 0x3004,
-
-        // set Parameter names
-        // for target TRACK
-        CHANNEL_MASK    = 0x4000,
-        FORMAT          = 0x4001,
-        MAIN_BUFFER     = 0x4002,
-        AUX_BUFFER      = 0x4003,
-        DOWNMIX_TYPE    = 0X4004,
-        MIXER_FORMAT    = 0x4005, // AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
-        MIXER_CHANNEL_MASK = 0x4006, // Channel mask for mixer output
-        // 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.
-        // for target RESAMPLE
-        SAMPLE_RATE     = 0x4100, // Configure sample rate conversion on this track name;
-                                  // parameter 'value' is the new sample rate in Hz.
-                                  // Only creates a sample rate converter the first time that
-                                  // the track sample rate is different from the mix sample rate.
-                                  // If the new sample rate is the same as the mix sample rate,
-                                  // and a sample rate converter already exists,
-                                  // then the sample rate converter remains present but is a no-op.
-        RESET           = 0x4101, // Reset sample rate converter without changing sample rate.
-                                  // This clears out the resampler's input buffer.
-        REMOVE          = 0x4102, // Remove the sample rate converter on this track name;
-                                  // the track is restored to the mix sample rate.
-        // for target RAMP_VOLUME and VOLUME (8 channels max)
-        // FIXME use float for these 3 to improve the dynamic range
-        VOLUME0         = 0x4200,
-        VOLUME1         = 0x4201,
-        AUXLEVEL        = 0x4210,
-        // for target TIMESTRETCH
-        PLAYBACK_RATE   = 0x4300, // Configure timestretch on this track name;
-                                  // parameter 'value' is a pointer to the new playback rate.
-    };
-
-    typedef enum { // Haptic intensity, should keep consistent with VibratorService
-        HAPTIC_SCALE_MUTE = os::IExternalVibratorService::SCALE_MUTE,
-        HAPTIC_SCALE_VERY_LOW = os::IExternalVibratorService::SCALE_VERY_LOW,
-        HAPTIC_SCALE_LOW = os::IExternalVibratorService::SCALE_LOW,
-        HAPTIC_SCALE_NONE = os::IExternalVibratorService::SCALE_NONE,
-        HAPTIC_SCALE_HIGH = os::IExternalVibratorService::SCALE_HIGH,
-        HAPTIC_SCALE_VERY_HIGH = os::IExternalVibratorService::SCALE_VERY_HIGH,
-    } haptic_intensity_t;
-    static constexpr float HAPTIC_SCALE_VERY_LOW_RATIO = 2.0f / 3.0f;
-    static constexpr float HAPTIC_SCALE_LOW_RATIO = 3.0f / 4.0f;
-    static const constexpr float HAPTIC_MAX_AMPLITUDE_FLOAT = 1.0f;
-
-    static inline bool isValidHapticIntensity(haptic_intensity_t hapticIntensity) {
-        switch (hapticIntensity) {
-        case HAPTIC_SCALE_MUTE:
-        case HAPTIC_SCALE_VERY_LOW:
-        case HAPTIC_SCALE_LOW:
-        case HAPTIC_SCALE_NONE:
-        case HAPTIC_SCALE_HIGH:
-        case HAPTIC_SCALE_VERY_HIGH:
-            return true;
-        default:
-            return false;
-        }
-    }
-
-    AudioMixer(size_t frameCount, uint32_t sampleRate)
-        : mSampleRate(sampleRate)
-        , mFrameCount(frameCount) {
-        pthread_once(&sOnceControl, &sInitRoutine);
-    }
-
-    // Create a new track in the mixer.
-    //
-    // \param name        a unique user-provided integer associated with the track.
-    //                    If name already exists, the function will abort.
-    // \param channelMask output channel mask.
-    // \param format      PCM format
-    // \param sessionId   Session id for the track. Tracks with the same
-    //                    session id will be submixed together.
-    //
-    // \return OK        on success.
-    //         BAD_VALUE if the format does not satisfy isValidFormat()
-    //                   or the channelMask does not satisfy isValidChannelMask().
-    status_t    create(
-            int name, audio_channel_mask_t channelMask, audio_format_t format, int sessionId);
-
-    bool        exists(int name) const {
-        return mTracks.count(name) > 0;
-    }
-
-    // Free an allocated track by name.
-    void        destroy(int name);
-
-    // Enable or disable an allocated track by name
-    void        enable(int name);
-    void        disable(int name);
-
-    void        setParameter(int name, int target, int param, void *value);
-
-    void        setBufferProvider(int name, AudioBufferProvider* bufferProvider);
-
-    void        process() {
-        for (const auto &pair : mTracks) {
-            // Clear contracted buffer before processing if contracted channels are saved
-            const std::shared_ptr<Track> &t = pair.second;
-            if (t->mKeepContractedChannels) {
-                t->clearContractedBuffer();
-            }
-        }
-        (this->*mHook)();
-        processHapticData();
-    }
-
-    size_t      getUnreleasedFrames(int name) const;
-
-    std::string trackNames() const {
-        std::stringstream ss;
-        for (const auto &pair : mTracks) {
-            ss << pair.first << " ";
-        }
-        return ss.str();
-    }
-
-    void        setNBLogWriter(NBLog::Writer *logWriter) {
-        mNBLogWriter = logWriter;
-    }
-
-    static inline bool isValidFormat(audio_format_t format) {
-        switch (format) {
-        case AUDIO_FORMAT_PCM_8_BIT:
-        case AUDIO_FORMAT_PCM_16_BIT:
-        case AUDIO_FORMAT_PCM_24_BIT_PACKED:
-        case AUDIO_FORMAT_PCM_32_BIT:
-        case AUDIO_FORMAT_PCM_FLOAT:
-            return true;
-        default:
-            return false;
-        }
-    }
-
-    static inline bool isValidChannelMask(audio_channel_mask_t channelMask) {
-        return audio_channel_mask_is_valid(channelMask); // the RemixBufferProvider is flexible.
-    }
-
-private:
-
-    /* For multi-format functions (calls template functions
-     * in AudioMixerOps.h).  The template parameters are as follows:
-     *
-     *   MIXTYPE     (see AudioMixerOps.h MIXTYPE_* enumeration)
-     *   USEFLOATVOL (set to true if float volume is used)
-     *   ADJUSTVOL   (set to true if volume ramp parameters needs adjustment afterwards)
-     *   TO: int32_t (Q4.27) or float
-     *   TI: int32_t (Q4.27) or int16_t (Q0.15) or float
-     *   TA: int32_t (Q4.27)
-     */
-
-    enum {
-        // FIXME this representation permits up to 8 channels
-        NEEDS_CHANNEL_COUNT__MASK   = 0x00000007,
-    };
-
-    enum {
-        NEEDS_CHANNEL_1             = 0x00000000,   // mono
-        NEEDS_CHANNEL_2             = 0x00000001,   // stereo
-
-        // sample format is not explicitly specified, and is assumed to be AUDIO_FORMAT_PCM_16_BIT
-
-        NEEDS_MUTE                  = 0x00000100,
-        NEEDS_RESAMPLE              = 0x00001000,
-        NEEDS_AUX                   = 0x00010000,
-    };
-
-    // hook types
-    enum {
-        PROCESSTYPE_NORESAMPLEONETRACK, // others set elsewhere
-    };
-
-    enum {
-        TRACKTYPE_NOP,
-        TRACKTYPE_RESAMPLE,
-        TRACKTYPE_NORESAMPLE,
-        TRACKTYPE_NORESAMPLEMONO,
-    };
-
-    // process hook functionality
-    using process_hook_t = void(AudioMixer::*)();
-
-    struct Track;
-    using hook_t = void(Track::*)(int32_t* output, size_t numOutFrames, int32_t* temp, int32_t* aux);
-
-    struct Track {
-        Track()
-            : bufferProvider(nullptr)
-        {
-            // TODO: move additional initialization here.
-        }
-
-        ~Track()
-        {
-            // bufferProvider, mInputBufferProvider need not be deleted.
-            mResampler.reset(nullptr);
-            // Ensure the order of destruction of buffer providers as they
-            // release the upstream provider in the destructor.
-            mTimestretchBufferProvider.reset(nullptr);
-            mPostDownmixReformatBufferProvider.reset(nullptr);
-            mDownmixerBufferProvider.reset(nullptr);
-            mReformatBufferProvider.reset(nullptr);
-            mContractChannelsNonDestructiveBufferProvider.reset(nullptr);
-            mAdjustChannelsBufferProvider.reset(nullptr);
-        }
-
-        bool        needsRamp() { return (volumeInc[0] | volumeInc[1] | auxInc) != 0; }
-        bool        setResampler(uint32_t trackSampleRate, uint32_t devSampleRate);
-        bool        doesResample() const { return mResampler.get() != nullptr; }
-        void        resetResampler() { if (mResampler.get() != nullptr) mResampler->reset(); }
-        void        adjustVolumeRamp(bool aux, bool useFloat = false);
-        size_t      getUnreleasedFrames() const { return mResampler.get() != nullptr ?
-                                                    mResampler->getUnreleasedFrames() : 0; };
-
-        status_t    prepareForDownmix();
-        void        unprepareForDownmix();
-        status_t    prepareForReformat();
-        void        unprepareForReformat();
-        status_t    prepareForAdjustChannels();
-        void        unprepareForAdjustChannels();
-        status_t    prepareForAdjustChannelsNonDestructive(size_t frames);
-        void        unprepareForAdjustChannelsNonDestructive();
-        void        clearContractedBuffer();
-        bool        setPlaybackRate(const AudioPlaybackRate &playbackRate);
-        void        reconfigureBufferProviders();
-
-        static hook_t getTrackHook(int trackType, uint32_t channelCount,
-                audio_format_t mixerInFormat, audio_format_t mixerOutFormat);
-
-        void track__nop(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
-
-        template <int MIXTYPE, bool USEFLOATVOL, bool ADJUSTVOL,
-            typename TO, typename TI, typename TA>
-        void volumeMix(TO *out, size_t outFrames, const TI *in, TA *aux, bool ramp);
-
-        uint32_t    needs;
-
-        // TODO: Eventually remove legacy integer volume settings
-        union {
-        int16_t     volume[MAX_NUM_VOLUMES]; // U4.12 fixed point (top bit should be zero)
-        int32_t     volumeRL;
-        };
-
-        int32_t     prevVolume[MAX_NUM_VOLUMES];
-        int32_t     volumeInc[MAX_NUM_VOLUMES];
-        int32_t     auxInc;
-        int32_t     prevAuxLevel;
-        int16_t     auxLevel;       // 0 <= auxLevel <= MAX_GAIN_INT, but signed for mul performance
-
-        uint16_t    frameCount;
-
-        uint8_t     channelCount;   // 1 or 2, redundant with (needs & NEEDS_CHANNEL_COUNT__MASK)
-        uint8_t     unused_padding; // formerly format, was always 16
-        uint16_t    enabled;        // actually bool
-        audio_channel_mask_t channelMask;
-
-        // actual buffer provider used by the track hooks, see DownmixerBufferProvider below
-        //  for how the Track buffer provider is wrapped by another one when dowmixing is required
-        AudioBufferProvider*                bufferProvider;
-
-        mutable AudioBufferProvider::Buffer buffer; // 8 bytes
-
-        hook_t      hook;
-        const void  *mIn;             // current location in buffer
-
-        std::unique_ptr<AudioResampler> mResampler;
-        uint32_t            sampleRate;
-        int32_t*           mainBuffer;
-        int32_t*           auxBuffer;
-
-        /* Buffer providers are constructed to translate the track input data as needed.
-         *
-         * TODO: perhaps make a single PlaybackConverterProvider class to move
-         * all pre-mixer track buffer conversions outside the AudioMixer class.
-         *
-         * 1) mInputBufferProvider: The AudioTrack buffer provider.
-         * 2) mAdjustChannelsBufferProvider: Expands or contracts sample data from one interleaved
-         *    channel format to another. Expanded channels are filled with zeros and put at the end
-         *    of each audio frame. Contracted channels are copied to the end of the buffer.
-         * 3) mContractChannelsNonDestructiveBufferProvider: Non-destructively contract sample data.
-         *    This is currently using at audio-haptic coupled playback to separate audio and haptic
-         *    data. Contracted channels could be written to given buffer.
-         * 4) mReformatBufferProvider: If not NULL, performs the audio reformat to
-         *    match either mMixerInFormat or mDownmixRequiresFormat, if the downmixer
-         *    requires reformat. For example, it may convert floating point input to
-         *    PCM_16_bit if that's required by the downmixer.
-         * 5) mDownmixerBufferProvider: If not NULL, performs the channel remixing to match
-         *    the number of channels required by the mixer sink.
-         * 6) mPostDownmixReformatBufferProvider: If not NULL, performs reformatting from
-         *    the downmixer requirements to the mixer engine input requirements.
-         * 7) mTimestretchBufferProvider: Adds timestretching for playback rate
-         */
-        AudioBufferProvider*     mInputBufferProvider;    // externally provided buffer provider.
-        // TODO: combine mAdjustChannelsBufferProvider and
-        // mContractChannelsNonDestructiveBufferProvider
-        std::unique_ptr<PassthruBufferProvider> mAdjustChannelsBufferProvider;
-        std::unique_ptr<PassthruBufferProvider> mContractChannelsNonDestructiveBufferProvider;
-        std::unique_ptr<PassthruBufferProvider> mReformatBufferProvider;
-        std::unique_ptr<PassthruBufferProvider> mDownmixerBufferProvider;
-        std::unique_ptr<PassthruBufferProvider> mPostDownmixReformatBufferProvider;
-        std::unique_ptr<PassthruBufferProvider> mTimestretchBufferProvider;
-
-        int32_t     sessionId;
-
-        audio_format_t mMixerFormat;     // output mix format: AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
-        audio_format_t mFormat;          // input track format
-        audio_format_t mMixerInFormat;   // mix internal format AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
-                                         // each track must be converted to this format.
-        audio_format_t mDownmixRequiresFormat;  // required downmixer format
-                                                // AUDIO_FORMAT_PCM_16_BIT if 16 bit necessary
-                                                // AUDIO_FORMAT_INVALID if no required format
-
-        float          mVolume[MAX_NUM_VOLUMES];     // floating point set volume
-        float          mPrevVolume[MAX_NUM_VOLUMES]; // floating point previous volume
-        float          mVolumeInc[MAX_NUM_VOLUMES];  // floating point volume increment
-
-        float          mAuxLevel;                     // floating point set aux level
-        float          mPrevAuxLevel;                 // floating point prev aux level
-        float          mAuxInc;                       // floating point aux increment
-
-        audio_channel_mask_t mMixerChannelMask;
-        uint32_t             mMixerChannelCount;
-
-        AudioPlaybackRate    mPlaybackRate;
-
-        // Haptic
-        bool                 mHapticPlaybackEnabled;
-        haptic_intensity_t   mHapticIntensity;
-        audio_channel_mask_t mHapticChannelMask;
-        uint32_t             mHapticChannelCount;
-        audio_channel_mask_t mMixerHapticChannelMask;
-        uint32_t             mMixerHapticChannelCount;
-        uint32_t             mAdjustInChannelCount;
-        uint32_t             mAdjustOutChannelCount;
-        uint32_t             mAdjustNonDestructiveInChannelCount;
-        uint32_t             mAdjustNonDestructiveOutChannelCount;
-        bool                 mKeepContractedChannels;
-
-        float getHapticScaleGamma() const {
-        // Need to keep consistent with the value in VibratorService.
-        switch (mHapticIntensity) {
-        case HAPTIC_SCALE_VERY_LOW:
-            return 2.0f;
-        case HAPTIC_SCALE_LOW:
-            return 1.5f;
-        case HAPTIC_SCALE_HIGH:
-            return 0.5f;
-        case HAPTIC_SCALE_VERY_HIGH:
-            return 0.25f;
-        default:
-            return 1.0f;
-        }
-        }
-
-        float getHapticMaxAmplitudeRatio() const {
-        // Need to keep consistent with the value in VibratorService.
-        switch (mHapticIntensity) {
-        case HAPTIC_SCALE_VERY_LOW:
-            return HAPTIC_SCALE_VERY_LOW_RATIO;
-        case HAPTIC_SCALE_LOW:
-            return HAPTIC_SCALE_LOW_RATIO;
-        case HAPTIC_SCALE_NONE:
-        case HAPTIC_SCALE_HIGH:
-        case HAPTIC_SCALE_VERY_HIGH:
-            return 1.0f;
-        default:
-            return 0.0f;
-        }
-        }
-
-    private:
-        // hooks
-        void track__genericResample(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
-        void track__16BitsStereo(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
-        void track__16BitsMono(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
-
-        void volumeRampStereo(int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
-        void volumeStereo(int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
-
-        // multi-format track hooks
-        template <int MIXTYPE, typename TO, typename TI, typename TA>
-        void track__Resample(TO* out, size_t frameCount, TO* temp __unused, TA* aux);
-        template <int MIXTYPE, typename TO, typename TI, typename TA>
-        void track__NoResample(TO* out, size_t frameCount, TO* temp __unused, TA* aux);
-    };
-
-    // TODO: remove BLOCKSIZE unit of processing - it isn't needed anymore.
-    static constexpr int BLOCKSIZE = 16;
-
-    bool setChannelMasks(int name,
-            audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask);
-
-    // Called when track info changes and a new process hook should be determined.
-    void invalidate() {
-        mHook = &AudioMixer::process__validate;
-    }
-
-    void process__validate();
-    void process__nop();
-    void process__genericNoResampling();
-    void process__genericResampling();
-    void process__oneTrack16BitsStereoNoResampling();
-
-    template <int MIXTYPE, typename TO, typename TI, typename TA>
-    void process__noResampleOneTrack();
-
-    void processHapticData();
-
-    static process_hook_t getProcessHook(int processType, uint32_t channelCount,
-            audio_format_t mixerInFormat, audio_format_t mixerOutFormat);
-
-    static void convertMixerFormat(void *out, audio_format_t mixerOutFormat,
-            void *in, audio_format_t mixerInFormat, size_t sampleCount);
-
-    static void sInitRoutine();
-
-    // initialization constants
-    const uint32_t mSampleRate;
-    const size_t mFrameCount;
-
-    NBLog::Writer *mNBLogWriter = nullptr;   // associated NBLog::Writer
-
-    process_hook_t mHook = &AudioMixer::process__nop;   // one of process__*, never nullptr
-
-    // the size of the type (int32_t) should be the largest of all types supported
-    // by the mixer.
-    std::unique_ptr<int32_t[]> mOutputTemp;
-    std::unique_ptr<int32_t[]> mResampleTemp;
-
-    // track names grouped by main buffer, in no particular order of main buffer.
-    // however names for a particular main buffer are in order (by construction).
-    std::unordered_map<void * /* mainBuffer */, std::vector<int /* name */>> mGroups;
-
-    // track names that are enabled, in increasing order (by construction).
-    std::vector<int /* name */> mEnabled;
-
-    // track smart pointers, by name, in increasing order of name.
-    std::map<int /* name */, std::shared_ptr<Track>> mTracks;
-
-    static pthread_once_t sOnceControl; // initialized in constructor by first new
-};
-
-// ----------------------------------------------------------------------------
-} // namespace android
-
-#endif // ANDROID_AUDIO_MIXER_H
diff --git a/media/libaudioclient/include/media/AudioParameter.h b/media/libaudioclient/include/media/AudioParameter.h
index 7469976..3c190f2 100644
--- a/media/libaudioclient/include/media/AudioParameter.h
+++ b/media/libaudioclient/include/media/AudioParameter.h
@@ -70,6 +70,9 @@
     //  keyDeviceConnect / Disconnect: value is an int in audio_devices_t
     static const char * const keyDeviceConnect;
     static const char * const keyDeviceDisconnect;
+    //  Need to be here because vendors still use them.
+    static const char * const keyStreamConnect;  // Deprecated: DO NOT USE.
+    static const char * const keyStreamDisconnect;  // Deprecated: DO NOT USE.
 
     // For querying stream capabilities. All the returned values are lists.
     //   keyStreamSupportedFormats: audio_format_t
diff --git a/media/libmedia/include/media/ExtendedAudioBufferProvider.h b/media/libaudioclient/include/media/ExtendedAudioBufferProvider.h
similarity index 100%
rename from media/libmedia/include/media/ExtendedAudioBufferProvider.h
rename to media/libaudioclient/include/media/ExtendedAudioBufferProvider.h
diff --git a/media/libaudiofoundation/Android.bp b/media/libaudiofoundation/Android.bp
new file mode 100644
index 0000000..5045d87
--- /dev/null
+++ b/media/libaudiofoundation/Android.bp
@@ -0,0 +1,33 @@
+cc_library_headers {
+    name: "libaudiofoundation_headers",
+    vendor_available: true,
+    export_include_dirs: ["include"],
+}
+
+cc_library_shared {
+    name: "libaudiofoundation",
+    vendor_available: true,
+
+    srcs: [
+        "AudioGain.cpp",
+    ],
+
+    shared_libs: [
+        "libbase",
+        "libbinder",
+        "liblog",
+        "libutils",
+    ],
+
+    header_libs: [
+        "libaudio_system_headers",
+        "libaudiofoundation_headers",
+    ],
+
+    export_header_lib_headers: ["libaudiofoundation_headers"],
+
+    cflags: [
+        "-Werror",
+        "-Wall",
+    ],
+}
diff --git a/media/libaudiofoundation/AudioGain.cpp b/media/libaudiofoundation/AudioGain.cpp
new file mode 100644
index 0000000..9d1d6db
--- /dev/null
+++ b/media/libaudiofoundation/AudioGain.cpp
@@ -0,0 +1,174 @@
+/*
+ * Copyright (C) 2015 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_TAG "AudioGain"
+//#define LOG_NDEBUG 0
+
+//#define VERY_VERBOSE_LOGGING
+#ifdef VERY_VERBOSE_LOGGING
+#define ALOGVV ALOGV
+#else
+#define ALOGVV(a...) do { } while(0)
+#endif
+
+#include <android-base/stringprintf.h>
+#include <media/AudioGain.h>
+#include <utils/Log.h>
+
+#include <math.h>
+
+namespace android {
+
+AudioGain::AudioGain(int index, bool useInChannelMask)
+{
+    mIndex = index;
+    mUseInChannelMask = useInChannelMask;
+    memset(&mGain, 0, sizeof(struct audio_gain));
+}
+
+void AudioGain::getDefaultConfig(struct audio_gain_config *config)
+{
+    config->index = mIndex;
+    config->mode = mGain.mode;
+    config->channel_mask = mGain.channel_mask;
+    if ((mGain.mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) {
+        config->values[0] = mGain.default_value;
+    } else {
+        uint32_t numValues;
+        if (mUseInChannelMask) {
+            numValues = audio_channel_count_from_in_mask(mGain.channel_mask);
+        } else {
+            numValues = audio_channel_count_from_out_mask(mGain.channel_mask);
+        }
+        for (size_t i = 0; i < numValues; i++) {
+            config->values[i] = mGain.default_value;
+        }
+    }
+    if ((mGain.mode & AUDIO_GAIN_MODE_RAMP) == AUDIO_GAIN_MODE_RAMP) {
+        config->ramp_duration_ms = mGain.min_ramp_ms;
+    }
+}
+
+status_t AudioGain::checkConfig(const struct audio_gain_config *config)
+{
+    if ((config->mode & ~mGain.mode) != 0) {
+        return BAD_VALUE;
+    }
+    if ((config->mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) {
+        if ((config->values[0] < mGain.min_value) ||
+                    (config->values[0] > mGain.max_value)) {
+            return BAD_VALUE;
+        }
+    } else {
+        if ((config->channel_mask & ~mGain.channel_mask) != 0) {
+            return BAD_VALUE;
+        }
+        uint32_t numValues;
+        if (mUseInChannelMask) {
+            numValues = audio_channel_count_from_in_mask(config->channel_mask);
+        } else {
+            numValues = audio_channel_count_from_out_mask(config->channel_mask);
+        }
+        for (size_t i = 0; i < numValues; i++) {
+            if ((config->values[i] < mGain.min_value) ||
+                    (config->values[i] > mGain.max_value)) {
+                return BAD_VALUE;
+            }
+        }
+    }
+    if ((config->mode & AUDIO_GAIN_MODE_RAMP) == AUDIO_GAIN_MODE_RAMP) {
+        if ((config->ramp_duration_ms < mGain.min_ramp_ms) ||
+                    (config->ramp_duration_ms > mGain.max_ramp_ms)) {
+            return BAD_VALUE;
+        }
+    }
+    return NO_ERROR;
+}
+
+void AudioGain::dump(std::string *dst, int spaces, int index) const
+{
+    dst->append(base::StringPrintf("%*sGain %d:\n", spaces, "", index+1));
+    dst->append(base::StringPrintf("%*s- mode: %08x\n", spaces, "", mGain.mode));
+    dst->append(base::StringPrintf("%*s- channel_mask: %08x\n", spaces, "", mGain.channel_mask));
+    dst->append(base::StringPrintf("%*s- min_value: %d mB\n", spaces, "", mGain.min_value));
+    dst->append(base::StringPrintf("%*s- max_value: %d mB\n", spaces, "", mGain.max_value));
+    dst->append(base::StringPrintf("%*s- default_value: %d mB\n", spaces, "", mGain.default_value));
+    dst->append(base::StringPrintf("%*s- step_value: %d mB\n", spaces, "", mGain.step_value));
+    dst->append(base::StringPrintf("%*s- min_ramp_ms: %d ms\n", spaces, "", mGain.min_ramp_ms));
+    dst->append(base::StringPrintf("%*s- max_ramp_ms: %d ms\n", spaces, "", mGain.max_ramp_ms));
+}
+
+status_t AudioGain::writeToParcel(android::Parcel *parcel) const
+{
+    status_t status = NO_ERROR;
+    if ((status = parcel->writeInt32(mIndex)) != NO_ERROR) return status;
+    if ((status = parcel->writeBool(mUseInChannelMask)) != NO_ERROR) return status;
+    if ((status = parcel->writeBool(mUseForVolume)) != NO_ERROR) return status;
+    if ((status = parcel->writeUint32(mGain.mode)) != NO_ERROR) return status;
+    if ((status = parcel->writeUint32(mGain.channel_mask)) != NO_ERROR) return status;
+    if ((status = parcel->writeInt32(mGain.min_value)) != NO_ERROR) return status;
+    if ((status = parcel->writeInt32(mGain.max_value)) != NO_ERROR) return status;
+    if ((status = parcel->writeInt32(mGain.default_value)) != NO_ERROR) return status;
+    if ((status = parcel->writeUint32(mGain.step_value)) != NO_ERROR) return status;
+    if ((status = parcel->writeUint32(mGain.min_ramp_ms)) != NO_ERROR) return status;
+    status = parcel->writeUint32(mGain.max_ramp_ms);
+    return status;
+}
+
+status_t AudioGain::readFromParcel(const android::Parcel *parcel)
+{
+    status_t status = NO_ERROR;
+    if ((status = parcel->readInt32(&mIndex)) != NO_ERROR) return status;
+    if ((status = parcel->readBool(&mUseInChannelMask)) != NO_ERROR) return status;
+    if ((status = parcel->readBool(&mUseForVolume)) != NO_ERROR) return status;
+    if ((status = parcel->readUint32(&mGain.mode)) != NO_ERROR) return status;
+    if ((status = parcel->readUint32(&mGain.channel_mask)) != NO_ERROR) return status;
+    if ((status = parcel->readInt32(&mGain.min_value)) != NO_ERROR) return status;
+    if ((status = parcel->readInt32(&mGain.max_value)) != NO_ERROR) return status;
+    if ((status = parcel->readInt32(&mGain.default_value)) != NO_ERROR) return status;
+    if ((status = parcel->readUint32(&mGain.step_value)) != NO_ERROR) return status;
+    if ((status = parcel->readUint32(&mGain.min_ramp_ms)) != NO_ERROR) return status;
+    status = parcel->readUint32(&mGain.max_ramp_ms);
+    return status;
+}
+
+status_t AudioGains::writeToParcel(android::Parcel *parcel) const {
+    status_t status = NO_ERROR;
+    if ((status = parcel->writeUint64(this->size())) != NO_ERROR) return status;
+    for (const auto &audioGain : *this) {
+        if ((status = parcel->writeParcelable(*audioGain)) != NO_ERROR) {
+            break;
+        }
+    }
+    return status;
+}
+
+status_t AudioGains::readFromParcel(const android::Parcel *parcel) {
+    status_t status = NO_ERROR;
+    uint64_t count;
+    if ((status = parcel->readUint64(&count)) != NO_ERROR) return status;
+    for (uint64_t i = 0; i < count; i++) {
+        sp<AudioGain> audioGain = new AudioGain(0, false);
+        if ((status = parcel->readParcelable(audioGain.get())) != NO_ERROR) {
+            this->clear();
+            break;
+        }
+        this->push_back(audioGain);
+    }
+    return status;
+}
+
+} // namespace android
diff --git a/media/libaudiofoundation/include/media/AudioContainers.h b/media/libaudiofoundation/include/media/AudioContainers.h
new file mode 100644
index 0000000..3313224
--- /dev/null
+++ b/media/libaudiofoundation/include/media/AudioContainers.h
@@ -0,0 +1,52 @@
+/*
+ * Copyright (C) 2019 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 <set>
+#include <vector>
+
+#include <system/audio.h>
+
+namespace android {
+
+using ChannelMaskSet = std::set<audio_channel_mask_t>;
+using FormatSet = std::set<audio_format_t>;
+using SampleRateSet = std::set<uint32_t>;
+
+using FormatVector = std::vector<audio_format_t>;
+
+static inline ChannelMaskSet asInMask(const ChannelMaskSet& channelMasks) {
+    ChannelMaskSet inMaskSet;
+    for (const auto &channel : channelMasks) {
+        if (audio_channel_mask_out_to_in(channel) != AUDIO_CHANNEL_INVALID) {
+            inMaskSet.insert(audio_channel_mask_out_to_in(channel));
+        }
+    }
+    return inMaskSet;
+}
+
+static inline ChannelMaskSet asOutMask(const ChannelMaskSet& channelMasks) {
+    ChannelMaskSet outMaskSet;
+    for (const auto &channel : channelMasks) {
+        if (audio_channel_mask_in_to_out(channel) != AUDIO_CHANNEL_INVALID) {
+            outMaskSet.insert(audio_channel_mask_in_to_out(channel));
+        }
+    }
+    return outMaskSet;
+}
+
+} // namespace android
\ No newline at end of file
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioGain.h b/media/libaudiofoundation/include/media/AudioGain.h
similarity index 84%
rename from services/audiopolicy/common/managerdefinitions/include/AudioGain.h
rename to media/libaudiofoundation/include/media/AudioGain.h
index 4af93e1..6a7fb55 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioGain.h
+++ b/media/libaudiofoundation/include/media/AudioGain.h
@@ -16,15 +16,17 @@
 
 #pragma once
 
+#include <binder/Parcel.h>
+#include <binder/Parcelable.h>
 #include <utils/Errors.h>
 #include <utils/RefBase.h>
-#include <utils/String8.h>
 #include <system/audio.h>
+#include <string>
 #include <vector>
 
 namespace android {
 
-class AudioGain: public RefBase
+class AudioGain: public RefBase, public Parcelable
 {
 public:
     AudioGain(int index, bool useInChannelMask);
@@ -55,7 +57,7 @@
     int getMaxRampInMs() const { return mGain.max_ramp_ms; }
 
     // TODO: remove dump from here (split serialization)
-    void dump(String8 *dst, int spaces, int index) const;
+    void dump(std::string *dst, int spaces, int index) const;
 
     void getDefaultConfig(struct audio_gain_config *config);
     status_t checkConfig(const struct audio_gain_config *config);
@@ -65,6 +67,9 @@
 
     const struct audio_gain &getGain() const { return mGain; }
 
+    status_t writeToParcel(Parcel* parcel) const override;
+    status_t readFromParcel(const Parcel* parcel) override;
+
 private:
     int               mIndex;
     struct audio_gain mGain;
@@ -72,7 +77,7 @@
     bool              mUseForVolume = false;
 };
 
-class AudioGains : public std::vector<sp<AudioGain> >
+class AudioGains : public std::vector<sp<AudioGain> >, public Parcelable
 {
 public:
     bool canUseForVolume() const
@@ -90,6 +95,9 @@
         push_back(gain);
         return 0;
     }
+
+    status_t writeToParcel(Parcel* parcel) const override;
+    status_t readFromParcel(const Parcel* parcel) override;
 };
 
 } // namespace android
diff --git a/media/libaudiohal/Android.bp b/media/libaudiohal/Android.bp
index 9803473..5837fcf 100644
--- a/media/libaudiohal/Android.bp
+++ b/media/libaudiohal/Android.bp
@@ -52,4 +52,10 @@
     name: "libaudiohal_headers",
 
     export_include_dirs: ["include"],
+
+    // This is needed because the stream interface includes media/MicrophoneInfo.h
+    // which is not in any library but has a dependency on headers from libbinder.
+    header_libs: ["libbinder_headers"],
+
+    export_header_lib_headers: ["libbinder_headers"],
 }
diff --git a/media/libaudioprocessing/Android.bp b/media/libaudioprocessing/Android.bp
index cb78063..e8aa700 100644
--- a/media/libaudioprocessing/Android.bp
+++ b/media/libaudioprocessing/Android.bp
@@ -3,20 +3,13 @@
 
     export_include_dirs: ["include"],
 
+    header_libs: ["libaudioclient_headers"],
+
     shared_libs: [
-        "libaudiohal",
         "libaudioutils",
         "libcutils",
         "liblog",
-        "libnbaio",
-        "libnblog",
-        "libsonic",
         "libutils",
-        "libvibrator",
-    ],
-
-    header_libs: [
-        "libbase_headers",
     ],
 
     cflags: [
@@ -33,18 +26,31 @@
     defaults: ["libaudioprocessing_defaults"],
 
     srcs: [
+        "AudioMixer.cpp",
         "BufferProviders.cpp",
         "RecordBufferConverter.cpp",
     ],
-    whole_static_libs: ["libaudioprocessing_arm"],
+
+    header_libs: [
+        "libbase_headers",
+    ],
+
+    shared_libs: [
+        "libaudiohal",
+        "libsonic",
+        "libvibrator",
+    ],
+
+    whole_static_libs: ["libaudioprocessing_base"],
 }
 
 cc_library_static {
-    name: "libaudioprocessing_arm",
+    name: "libaudioprocessing_base",
     defaults: ["libaudioprocessing_defaults"],
+    vendor_available: true,
 
     srcs: [
-        "AudioMixer.cpp",
+        "AudioMixerBase.cpp",
         "AudioResampler.cpp",
         "AudioResamplerCubic.cpp",
         "AudioResamplerSinc.cpp",
diff --git a/media/libaudioprocessing/AudioMixer.cpp b/media/libaudioprocessing/AudioMixer.cpp
index f7cc096..c0b11a4 100644
--- a/media/libaudioprocessing/AudioMixer.cpp
+++ b/media/libaudioprocessing/AudioMixer.cpp
@@ -18,6 +18,7 @@
 #define LOG_TAG "AudioMixer"
 //#define LOG_NDEBUG 0
 
+#include <sstream>
 #include <stdint.h>
 #include <string.h>
 #include <stdlib.h>
@@ -27,9 +28,6 @@
 #include <utils/Errors.h>
 #include <utils/Log.h>
 
-#include <cutils/compiler.h>
-#include <utils/Debug.h>
-
 #include <system/audio.h>
 
 #include <audio_utils/primitives.h>
@@ -58,138 +56,15 @@
 #define ALOGVV(a...) do { } while (0)
 #endif
 
-#ifndef ARRAY_SIZE
-#define ARRAY_SIZE(x) (sizeof(x)/sizeof((x)[0]))
-#endif
-
-// Set kUseNewMixer to true to use the new mixer engine always. Otherwise the
-// original code will be used for stereo sinks, the new mixer for multichannel.
-static constexpr bool kUseNewMixer = true;
-
-// Set kUseFloat to true to allow floating input into the mixer engine.
-// If kUseNewMixer is false, this is ignored or may be overridden internally
-// because of downmix/upmix support.
-static constexpr bool kUseFloat = true;
-
-#ifdef FLOAT_AUX
-using TYPE_AUX = float;
-static_assert(kUseNewMixer && kUseFloat,
-        "kUseNewMixer and kUseFloat must be true for FLOAT_AUX option");
-#else
-using TYPE_AUX = int32_t; // q4.27
-#endif
-
 // Set to default copy buffer size in frames for input processing.
-static const size_t kCopyBufferFrameCount = 256;
+static constexpr size_t kCopyBufferFrameCount = 256;
 
 namespace android {
 
 // ----------------------------------------------------------------------------
 
-static inline audio_format_t selectMixerInFormat(audio_format_t inputFormat __unused) {
-    return kUseFloat && kUseNewMixer ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
-}
-
-status_t AudioMixer::create(
-        int name, audio_channel_mask_t channelMask, audio_format_t format, int sessionId)
-{
-    LOG_ALWAYS_FATAL_IF(exists(name), "name %d already exists", name);
-
-    if (!isValidChannelMask(channelMask)) {
-        ALOGE("%s invalid channelMask: %#x", __func__, channelMask);
-        return BAD_VALUE;
-    }
-    if (!isValidFormat(format)) {
-        ALOGE("%s invalid format: %#x", __func__, format);
-        return BAD_VALUE;
-    }
-
-    auto t = std::make_shared<Track>();
-    {
-        // TODO: move initialization to the Track constructor.
-        // assume default parameters for the track, except where noted below
-        t->needs = 0;
-
-        // Integer volume.
-        // Currently integer volume is kept for the legacy integer mixer.
-        // Will be removed when the legacy mixer path is removed.
-        t->volume[0] = 0;
-        t->volume[1] = 0;
-        t->prevVolume[0] = 0 << 16;
-        t->prevVolume[1] = 0 << 16;
-        t->volumeInc[0] = 0;
-        t->volumeInc[1] = 0;
-        t->auxLevel = 0;
-        t->auxInc = 0;
-        t->prevAuxLevel = 0;
-
-        // Floating point volume.
-        t->mVolume[0] = 0.f;
-        t->mVolume[1] = 0.f;
-        t->mPrevVolume[0] = 0.f;
-        t->mPrevVolume[1] = 0.f;
-        t->mVolumeInc[0] = 0.;
-        t->mVolumeInc[1] = 0.;
-        t->mAuxLevel = 0.;
-        t->mAuxInc = 0.;
-        t->mPrevAuxLevel = 0.;
-
-        // no initialization needed
-        // t->frameCount
-        t->mHapticChannelMask = channelMask & AUDIO_CHANNEL_HAPTIC_ALL;
-        t->mHapticChannelCount = audio_channel_count_from_out_mask(t->mHapticChannelMask);
-        channelMask &= ~AUDIO_CHANNEL_HAPTIC_ALL;
-        t->channelCount = audio_channel_count_from_out_mask(channelMask);
-        t->enabled = false;
-        ALOGV_IF(audio_channel_mask_get_bits(channelMask) != AUDIO_CHANNEL_OUT_STEREO,
-                "Non-stereo channel mask: %d\n", channelMask);
-        t->channelMask = channelMask;
-        t->sessionId = sessionId;
-        // setBufferProvider(name, AudioBufferProvider *) is required before enable(name)
-        t->bufferProvider = NULL;
-        t->buffer.raw = NULL;
-        // no initialization needed
-        // t->buffer.frameCount
-        t->hook = NULL;
-        t->mIn = NULL;
-        t->sampleRate = mSampleRate;
-        // setParameter(name, TRACK, MAIN_BUFFER, mixBuffer) is required before enable(name)
-        t->mainBuffer = NULL;
-        t->auxBuffer = NULL;
-        t->mInputBufferProvider = NULL;
-        t->mMixerFormat = AUDIO_FORMAT_PCM_16_BIT;
-        t->mFormat = format;
-        t->mMixerInFormat = selectMixerInFormat(format);
-        t->mDownmixRequiresFormat = AUDIO_FORMAT_INVALID; // no format required
-        t->mMixerChannelMask = audio_channel_mask_from_representation_and_bits(
-                AUDIO_CHANNEL_REPRESENTATION_POSITION, AUDIO_CHANNEL_OUT_STEREO);
-        t->mMixerChannelCount = audio_channel_count_from_out_mask(t->mMixerChannelMask);
-        t->mPlaybackRate = AUDIO_PLAYBACK_RATE_DEFAULT;
-        // haptic
-        t->mHapticPlaybackEnabled = false;
-        t->mHapticIntensity = HAPTIC_SCALE_NONE;
-        t->mMixerHapticChannelMask = AUDIO_CHANNEL_NONE;
-        t->mMixerHapticChannelCount = 0;
-        t->mAdjustInChannelCount = t->channelCount + t->mHapticChannelCount;
-        t->mAdjustOutChannelCount = t->channelCount + t->mMixerHapticChannelCount;
-        t->mAdjustNonDestructiveInChannelCount = t->mAdjustOutChannelCount;
-        t->mAdjustNonDestructiveOutChannelCount = t->channelCount;
-        t->mKeepContractedChannels = false;
-        // Check the downmixing (or upmixing) requirements.
-        status_t status = t->prepareForDownmix();
-        if (status != OK) {
-            ALOGE("AudioMixer::getTrackName invalid channelMask (%#x)", channelMask);
-            return BAD_VALUE;
-        }
-        // prepareForDownmix() may change mDownmixRequiresFormat
-        ALOGVV("mMixerFormat:%#x  mMixerInFormat:%#x\n", t->mMixerFormat, t->mMixerInFormat);
-        t->prepareForReformat();
-        t->prepareForAdjustChannelsNonDestructive(mFrameCount);
-        t->prepareForAdjustChannels();
-
-        mTracks[name] = t;
-        return OK;
-    }
+bool AudioMixer::isValidChannelMask(audio_channel_mask_t channelMask) const {
+    return audio_channel_mask_is_valid(channelMask); // the RemixBufferProvider is flexible.
 }
 
 // Called when channel masks have changed for a track name
@@ -198,7 +73,7 @@
 bool AudioMixer::setChannelMasks(int name,
         audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask) {
     LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
-    const std::shared_ptr<Track> &track = mTracks[name];
+    const std::shared_ptr<Track> &track = getTrack(name);
 
     if (trackChannelMask == (track->channelMask | track->mHapticChannelMask)
             && mixerChannelMask == (track->mMixerChannelMask | track->mMixerHapticChannelMask)) {
@@ -255,14 +130,8 @@
     track->prepareForAdjustChannelsNonDestructive(mFrameCount);
     track->prepareForAdjustChannels();
 
-    if (track->mResampler.get() != nullptr) {
-        // resampler channels may have changed.
-        const uint32_t resetToSampleRate = track->sampleRate;
-        track->mResampler.reset(nullptr);
-        track->sampleRate = mSampleRate; // without resampler, track rate is device sample rate.
-        // recreate the resampler with updated format, channels, saved sampleRate.
-        track->setResampler(resetToSampleRate /*trackSampleRate*/, mSampleRate /*devSampleRate*/);
-    }
+    // Resampler channels may have changed.
+    track->recreateResampler(mSampleRate);
     return true;
 }
 
@@ -477,171 +346,10 @@
     }
 }
 
-void AudioMixer::destroy(int name)
-{
-    LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
-    ALOGV("deleteTrackName(%d)", name);
-
-    if (mTracks[name]->enabled) {
-        invalidate();
-    }
-    mTracks.erase(name); // deallocate track
-}
-
-void AudioMixer::enable(int name)
-{
-    LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
-    const std::shared_ptr<Track> &track = mTracks[name];
-
-    if (!track->enabled) {
-        track->enabled = true;
-        ALOGV("enable(%d)", name);
-        invalidate();
-    }
-}
-
-void AudioMixer::disable(int name)
-{
-    LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
-    const std::shared_ptr<Track> &track = mTracks[name];
-
-    if (track->enabled) {
-        track->enabled = false;
-        ALOGV("disable(%d)", name);
-        invalidate();
-    }
-}
-
-/* Sets the volume ramp variables for the AudioMixer.
- *
- * The volume ramp variables are used to transition from the previous
- * volume to the set volume.  ramp controls the duration of the transition.
- * Its value is typically one state framecount period, but may also be 0,
- * meaning "immediate."
- *
- * FIXME: 1) Volume ramp is enabled only if there is a nonzero integer increment
- * even if there is a nonzero floating point increment (in that case, the volume
- * change is immediate).  This restriction should be changed when the legacy mixer
- * is removed (see #2).
- * FIXME: 2) Integer volume variables are used for Legacy mixing and should be removed
- * when no longer needed.
- *
- * @param newVolume set volume target in floating point [0.0, 1.0].
- * @param ramp number of frames to increment over. if ramp is 0, the volume
- * should be set immediately.  Currently ramp should not exceed 65535 (frames).
- * @param pIntSetVolume pointer to the U4.12 integer target volume, set on return.
- * @param pIntPrevVolume pointer to the U4.28 integer previous volume, set on return.
- * @param pIntVolumeInc pointer to the U4.28 increment per output audio frame, set on return.
- * @param pSetVolume pointer to the float target volume, set on return.
- * @param pPrevVolume pointer to the float previous volume, set on return.
- * @param pVolumeInc pointer to the float increment per output audio frame, set on return.
- * @return true if the volume has changed, false if volume is same.
- */
-static inline bool setVolumeRampVariables(float newVolume, int32_t ramp,
-        int16_t *pIntSetVolume, int32_t *pIntPrevVolume, int32_t *pIntVolumeInc,
-        float *pSetVolume, float *pPrevVolume, float *pVolumeInc) {
-    // check floating point volume to see if it is identical to the previously
-    // set volume.
-    // We do not use a tolerance here (and reject changes too small)
-    // as it may be confusing to use a different value than the one set.
-    // If the resulting volume is too small to ramp, it is a direct set of the volume.
-    if (newVolume == *pSetVolume) {
-        return false;
-    }
-    if (newVolume < 0) {
-        newVolume = 0; // should not have negative volumes
-    } else {
-        switch (fpclassify(newVolume)) {
-        case FP_SUBNORMAL:
-        case FP_NAN:
-            newVolume = 0;
-            break;
-        case FP_ZERO:
-            break; // zero volume is fine
-        case FP_INFINITE:
-            // Infinite volume could be handled consistently since
-            // floating point math saturates at infinities,
-            // but we limit volume to unity gain float.
-            // ramp = 0; break;
-            //
-            newVolume = AudioMixer::UNITY_GAIN_FLOAT;
-            break;
-        case FP_NORMAL:
-        default:
-            // Floating point does not have problems with overflow wrap
-            // that integer has.  However, we limit the volume to
-            // unity gain here.
-            // TODO: Revisit the volume limitation and perhaps parameterize.
-            if (newVolume > AudioMixer::UNITY_GAIN_FLOAT) {
-                newVolume = AudioMixer::UNITY_GAIN_FLOAT;
-            }
-            break;
-        }
-    }
-
-    // set floating point volume ramp
-    if (ramp != 0) {
-        // when the ramp completes, *pPrevVolume is set to *pSetVolume, so there
-        // is no computational mismatch; hence equality is checked here.
-        ALOGD_IF(*pPrevVolume != *pSetVolume, "previous float ramp hasn't finished,"
-                " prev:%f  set_to:%f", *pPrevVolume, *pSetVolume);
-        const float inc = (newVolume - *pPrevVolume) / ramp; // could be inf, nan, subnormal
-        // could be inf, cannot be nan, subnormal
-        const float maxv = std::max(newVolume, *pPrevVolume);
-
-        if (isnormal(inc) // inc must be a normal number (no subnormals, infinite, nan)
-                && maxv + inc != maxv) { // inc must make forward progress
-            *pVolumeInc = inc;
-            // ramp is set now.
-            // Note: if newVolume is 0, then near the end of the ramp,
-            // it may be possible that the ramped volume may be subnormal or
-            // temporarily negative by a small amount or subnormal due to floating
-            // point inaccuracies.
-        } else {
-            ramp = 0; // ramp not allowed
-        }
-    }
-
-    // compute and check integer volume, no need to check negative values
-    // The integer volume is limited to "unity_gain" to avoid wrapping and other
-    // audio artifacts, so it never reaches the range limit of U4.28.
-    // We safely use signed 16 and 32 bit integers here.
-    const float scaledVolume = newVolume * AudioMixer::UNITY_GAIN_INT; // not neg, subnormal, nan
-    const int32_t intVolume = (scaledVolume >= (float)AudioMixer::UNITY_GAIN_INT) ?
-            AudioMixer::UNITY_GAIN_INT : (int32_t)scaledVolume;
-
-    // set integer volume ramp
-    if (ramp != 0) {
-        // integer volume is U4.12 (to use 16 bit multiplies), but ramping uses U4.28.
-        // when the ramp completes, *pIntPrevVolume is set to *pIntSetVolume << 16, so there
-        // is no computational mismatch; hence equality is checked here.
-        ALOGD_IF(*pIntPrevVolume != *pIntSetVolume << 16, "previous int ramp hasn't finished,"
-                " prev:%d  set_to:%d", *pIntPrevVolume, *pIntSetVolume << 16);
-        const int32_t inc = ((intVolume << 16) - *pIntPrevVolume) / ramp;
-
-        if (inc != 0) { // inc must make forward progress
-            *pIntVolumeInc = inc;
-        } else {
-            ramp = 0; // ramp not allowed
-        }
-    }
-
-    // if no ramp, or ramp not allowed, then clear float and integer increments
-    if (ramp == 0) {
-        *pVolumeInc = 0;
-        *pPrevVolume = newVolume;
-        *pIntVolumeInc = 0;
-        *pIntPrevVolume = intVolume << 16;
-    }
-    *pSetVolume = newVolume;
-    *pIntSetVolume = intVolume;
-    return true;
-}
-
 void AudioMixer::setParameter(int name, int target, int param, void *value)
 {
     LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
-    const std::shared_ptr<Track> &track = mTracks[name];
+    const std::shared_ptr<Track> &track = getTrack(name);
 
     int valueInt = static_cast<int>(reinterpret_cast<uintptr_t>(value));
     int32_t *valueBuf = reinterpret_cast<int32_t*>(value);
@@ -670,11 +378,7 @@
             }
             break;
         case AUX_BUFFER:
-            if (track->auxBuffer != valueBuf) {
-                track->auxBuffer = valueBuf;
-                ALOGV("setParameter(TRACK, AUX_BUFFER, %p)", valueBuf);
-                invalidate();
-            }
+            AudioMixerBase::setParameter(name, target, param, value);
             break;
         case FORMAT: {
             audio_format_t format = static_cast<audio_format_t>(valueInt);
@@ -730,127 +434,38 @@
         break;
 
     case RESAMPLE:
-        switch (param) {
-        case SAMPLE_RATE:
-            ALOG_ASSERT(valueInt > 0, "bad sample rate %d", valueInt);
-            if (track->setResampler(uint32_t(valueInt), mSampleRate)) {
-                ALOGV("setParameter(RESAMPLE, SAMPLE_RATE, %u)",
-                        uint32_t(valueInt));
-                invalidate();
-            }
-            break;
-        case RESET:
-            track->resetResampler();
-            invalidate();
-            break;
-        case REMOVE:
-            track->mResampler.reset(nullptr);
-            track->sampleRate = mSampleRate;
-            invalidate();
-            break;
-        default:
-            LOG_ALWAYS_FATAL("setParameter resample: bad param %d", param);
-        }
-        break;
-
     case RAMP_VOLUME:
     case VOLUME:
+        AudioMixerBase::setParameter(name, target, param, value);
+        break;
+    case TIMESTRETCH:
         switch (param) {
-        case AUXLEVEL:
-            if (setVolumeRampVariables(*reinterpret_cast<float*>(value),
-                    target == RAMP_VOLUME ? mFrameCount : 0,
-                    &track->auxLevel, &track->prevAuxLevel, &track->auxInc,
-                    &track->mAuxLevel, &track->mPrevAuxLevel, &track->mAuxInc)) {
-                ALOGV("setParameter(%s, AUXLEVEL: %04x)",
-                        target == VOLUME ? "VOLUME" : "RAMP_VOLUME", track->auxLevel);
-                invalidate();
+        case PLAYBACK_RATE: {
+            const AudioPlaybackRate *playbackRate =
+                    reinterpret_cast<AudioPlaybackRate*>(value);
+            ALOGW_IF(!isAudioPlaybackRateValid(*playbackRate),
+                    "bad parameters speed %f, pitch %f",
+                    playbackRate->mSpeed, playbackRate->mPitch);
+            if (track->setPlaybackRate(*playbackRate)) {
+                ALOGV("setParameter(TIMESTRETCH, PLAYBACK_RATE, STRETCH_MODE, FALLBACK_MODE "
+                        "%f %f %d %d",
+                        playbackRate->mSpeed,
+                        playbackRate->mPitch,
+                        playbackRate->mStretchMode,
+                        playbackRate->mFallbackMode);
+                // invalidate();  (should not require reconfigure)
             }
-            break;
+        } break;
         default:
-            if ((unsigned)param >= VOLUME0 && (unsigned)param < VOLUME0 + MAX_NUM_VOLUMES) {
-                if (setVolumeRampVariables(*reinterpret_cast<float*>(value),
-                        target == RAMP_VOLUME ? mFrameCount : 0,
-                        &track->volume[param - VOLUME0],
-                        &track->prevVolume[param - VOLUME0],
-                        &track->volumeInc[param - VOLUME0],
-                        &track->mVolume[param - VOLUME0],
-                        &track->mPrevVolume[param - VOLUME0],
-                        &track->mVolumeInc[param - VOLUME0])) {
-                    ALOGV("setParameter(%s, VOLUME%d: %04x)",
-                            target == VOLUME ? "VOLUME" : "RAMP_VOLUME", param - VOLUME0,
-                                    track->volume[param - VOLUME0]);
-                    invalidate();
-                }
-            } else {
-                LOG_ALWAYS_FATAL("setParameter volume: bad param %d", param);
-            }
+            LOG_ALWAYS_FATAL("setParameter timestretch: bad param %d", param);
         }
         break;
-        case TIMESTRETCH:
-            switch (param) {
-            case PLAYBACK_RATE: {
-                const AudioPlaybackRate *playbackRate =
-                        reinterpret_cast<AudioPlaybackRate*>(value);
-                ALOGW_IF(!isAudioPlaybackRateValid(*playbackRate),
-                        "bad parameters speed %f, pitch %f",
-                        playbackRate->mSpeed, playbackRate->mPitch);
-                if (track->setPlaybackRate(*playbackRate)) {
-                    ALOGV("setParameter(TIMESTRETCH, PLAYBACK_RATE, STRETCH_MODE, FALLBACK_MODE "
-                            "%f %f %d %d",
-                            playbackRate->mSpeed,
-                            playbackRate->mPitch,
-                            playbackRate->mStretchMode,
-                            playbackRate->mFallbackMode);
-                    // invalidate();  (should not require reconfigure)
-                }
-            } break;
-            default:
-                LOG_ALWAYS_FATAL("setParameter timestretch: bad param %d", param);
-            }
-            break;
 
     default:
         LOG_ALWAYS_FATAL("setParameter: bad target %d", target);
     }
 }
 
-bool AudioMixer::Track::setResampler(uint32_t trackSampleRate, uint32_t devSampleRate)
-{
-    if (trackSampleRate != devSampleRate || mResampler.get() != nullptr) {
-        if (sampleRate != trackSampleRate) {
-            sampleRate = trackSampleRate;
-            if (mResampler.get() == nullptr) {
-                ALOGV("Creating resampler from track %d Hz to device %d Hz",
-                        trackSampleRate, devSampleRate);
-                AudioResampler::src_quality quality;
-                // force lowest quality level resampler if use case isn't music or video
-                // FIXME this is flawed for dynamic sample rates, as we choose the resampler
-                // quality level based on the initial ratio, but that could change later.
-                // Should have a way to distinguish tracks with static ratios vs. dynamic ratios.
-                if (isMusicRate(trackSampleRate)) {
-                    quality = AudioResampler::DEFAULT_QUALITY;
-                } else {
-                    quality = AudioResampler::DYN_LOW_QUALITY;
-                }
-
-                // TODO: Remove MONO_HACK. Resampler sees #channels after the downmixer
-                // but if none exists, it is the channel count (1 for mono).
-                const int resamplerChannelCount = mDownmixerBufferProvider.get() != nullptr
-                        ? mMixerChannelCount : channelCount;
-                ALOGVV("Creating resampler:"
-                        " format(%#x) channels(%d) devSampleRate(%u) quality(%d)\n",
-                        mMixerInFormat, resamplerChannelCount, devSampleRate, quality);
-                mResampler.reset(AudioResampler::create(
-                        mMixerInFormat,
-                        resamplerChannelCount,
-                        devSampleRate, quality));
-            }
-            return true;
-        }
-    }
-    return false;
-}
-
 bool AudioMixer::Track::setPlaybackRate(const AudioPlaybackRate &playbackRate)
 {
     if ((mTimestretchBufferProvider.get() == nullptr &&
@@ -863,8 +478,7 @@
     if (mTimestretchBufferProvider.get() == nullptr) {
         // TODO: Remove MONO_HACK. Resampler sees #channels after the downmixer
         // but if none exists, it is the channel count (1 for mono).
-        const int timestretchChannelCount = mDownmixerBufferProvider.get() != nullptr
-                ? mMixerChannelCount : channelCount;
+        const int timestretchChannelCount = getOutputChannelCount();
         mTimestretchBufferProvider.reset(new TimestretchBufferProvider(timestretchChannelCount,
                 mMixerInFormat, sampleRate, playbackRate));
         reconfigureBufferProviders();
@@ -875,84 +489,10 @@
     return true;
 }
 
-/* Checks to see if the volume ramp has completed and clears the increment
- * variables appropriately.
- *
- * FIXME: There is code to handle int/float ramp variable switchover should it not
- * complete within a mixer buffer processing call, but it is preferred to avoid switchover
- * due to precision issues.  The switchover code is included for legacy code purposes
- * and can be removed once the integer volume is removed.
- *
- * It is not sufficient to clear only the volumeInc integer variable because
- * if one channel requires ramping, all channels are ramped.
- *
- * There is a bit of duplicated code here, but it keeps backward compatibility.
- */
-inline void AudioMixer::Track::adjustVolumeRamp(bool aux, bool useFloat)
-{
-    if (useFloat) {
-        for (uint32_t i = 0; i < MAX_NUM_VOLUMES; i++) {
-            if ((mVolumeInc[i] > 0 && mPrevVolume[i] + mVolumeInc[i] >= mVolume[i]) ||
-                     (mVolumeInc[i] < 0 && mPrevVolume[i] + mVolumeInc[i] <= mVolume[i])) {
-                volumeInc[i] = 0;
-                prevVolume[i] = volume[i] << 16;
-                mVolumeInc[i] = 0.;
-                mPrevVolume[i] = mVolume[i];
-            } else {
-                //ALOGV("ramp: %f %f %f", mVolume[i], mPrevVolume[i], mVolumeInc[i]);
-                prevVolume[i] = u4_28_from_float(mPrevVolume[i]);
-            }
-        }
-    } else {
-        for (uint32_t i = 0; i < MAX_NUM_VOLUMES; i++) {
-            if (((volumeInc[i]>0) && (((prevVolume[i]+volumeInc[i])>>16) >= volume[i])) ||
-                    ((volumeInc[i]<0) && (((prevVolume[i]+volumeInc[i])>>16) <= volume[i]))) {
-                volumeInc[i] = 0;
-                prevVolume[i] = volume[i] << 16;
-                mVolumeInc[i] = 0.;
-                mPrevVolume[i] = mVolume[i];
-            } else {
-                //ALOGV("ramp: %d %d %d", volume[i] << 16, prevVolume[i], volumeInc[i]);
-                mPrevVolume[i]  = float_from_u4_28(prevVolume[i]);
-            }
-        }
-    }
-
-    if (aux) {
-#ifdef FLOAT_AUX
-        if (useFloat) {
-            if ((mAuxInc > 0.f && mPrevAuxLevel + mAuxInc >= mAuxLevel) ||
-                    (mAuxInc < 0.f && mPrevAuxLevel + mAuxInc <= mAuxLevel)) {
-                auxInc = 0;
-                prevAuxLevel = auxLevel << 16;
-                mAuxInc = 0.f;
-                mPrevAuxLevel = mAuxLevel;
-            }
-        } else
-#endif
-        if ((auxInc > 0 && ((prevAuxLevel + auxInc) >> 16) >= auxLevel) ||
-                (auxInc < 0 && ((prevAuxLevel + auxInc) >> 16) <= auxLevel)) {
-            auxInc = 0;
-            prevAuxLevel = auxLevel << 16;
-            mAuxInc = 0.f;
-            mPrevAuxLevel = mAuxLevel;
-        }
-    }
-}
-
-size_t AudioMixer::getUnreleasedFrames(int name) const
-{
-    const auto it = mTracks.find(name);
-    if (it != mTracks.end()) {
-        return it->second->getUnreleasedFrames();
-    }
-    return 0;
-}
-
 void AudioMixer::setBufferProvider(int name, AudioBufferProvider* bufferProvider)
 {
     LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
-    const std::shared_ptr<Track> &track = mTracks[name];
+    const std::shared_ptr<Track> &track = getTrack(name);
 
     if (track->mInputBufferProvider == bufferProvider) {
         return; // don't reset any buffer providers if identical.
@@ -976,679 +516,6 @@
     track->reconfigureBufferProviders();
 }
 
-void AudioMixer::process__validate()
-{
-    // TODO: fix all16BitsStereNoResample logic to
-    // either properly handle muted tracks (it should ignore them)
-    // or remove altogether as an obsolete optimization.
-    bool all16BitsStereoNoResample = true;
-    bool resampling = false;
-    bool volumeRamp = false;
-
-    mEnabled.clear();
-    mGroups.clear();
-    for (const auto &pair : mTracks) {
-        const int name = pair.first;
-        const std::shared_ptr<Track> &t = pair.second;
-        if (!t->enabled) continue;
-
-        mEnabled.emplace_back(name);  // we add to mEnabled in order of name.
-        mGroups[t->mainBuffer].emplace_back(name); // mGroups also in order of name.
-
-        uint32_t n = 0;
-        // FIXME can overflow (mask is only 3 bits)
-        n |= NEEDS_CHANNEL_1 + t->channelCount - 1;
-        if (t->doesResample()) {
-            n |= NEEDS_RESAMPLE;
-        }
-        if (t->auxLevel != 0 && t->auxBuffer != NULL) {
-            n |= NEEDS_AUX;
-        }
-
-        if (t->volumeInc[0]|t->volumeInc[1]) {
-            volumeRamp = true;
-        } else if (!t->doesResample() && t->volumeRL == 0) {
-            n |= NEEDS_MUTE;
-        }
-        t->needs = n;
-
-        if (n & NEEDS_MUTE) {
-            t->hook = &Track::track__nop;
-        } else {
-            if (n & NEEDS_AUX) {
-                all16BitsStereoNoResample = false;
-            }
-            if (n & NEEDS_RESAMPLE) {
-                all16BitsStereoNoResample = false;
-                resampling = true;
-                t->hook = Track::getTrackHook(TRACKTYPE_RESAMPLE, t->mMixerChannelCount,
-                        t->mMixerInFormat, t->mMixerFormat);
-                ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2,
-                        "Track %d needs downmix + resample", name);
-            } else {
-                if ((n & NEEDS_CHANNEL_COUNT__MASK) == NEEDS_CHANNEL_1){
-                    t->hook = Track::getTrackHook(
-                            (t->mMixerChannelMask == AUDIO_CHANNEL_OUT_STEREO  // TODO: MONO_HACK
-                                    && t->channelMask == AUDIO_CHANNEL_OUT_MONO)
-                                ? TRACKTYPE_NORESAMPLEMONO : TRACKTYPE_NORESAMPLE,
-                            t->mMixerChannelCount,
-                            t->mMixerInFormat, t->mMixerFormat);
-                    all16BitsStereoNoResample = false;
-                }
-                if ((n & NEEDS_CHANNEL_COUNT__MASK) >= NEEDS_CHANNEL_2){
-                    t->hook = Track::getTrackHook(TRACKTYPE_NORESAMPLE, t->mMixerChannelCount,
-                            t->mMixerInFormat, t->mMixerFormat);
-                    ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2,
-                            "Track %d needs downmix", name);
-                }
-            }
-        }
-    }
-
-    // select the processing hooks
-    mHook = &AudioMixer::process__nop;
-    if (mEnabled.size() > 0) {
-        if (resampling) {
-            if (mOutputTemp.get() == nullptr) {
-                mOutputTemp.reset(new int32_t[MAX_NUM_CHANNELS * mFrameCount]);
-            }
-            if (mResampleTemp.get() == nullptr) {
-                mResampleTemp.reset(new int32_t[MAX_NUM_CHANNELS * mFrameCount]);
-            }
-            mHook = &AudioMixer::process__genericResampling;
-        } else {
-            // we keep temp arrays around.
-            mHook = &AudioMixer::process__genericNoResampling;
-            if (all16BitsStereoNoResample && !volumeRamp) {
-                if (mEnabled.size() == 1) {
-                    const std::shared_ptr<Track> &t = mTracks[mEnabled[0]];
-                    if ((t->needs & NEEDS_MUTE) == 0) {
-                        // The check prevents a muted track from acquiring a process hook.
-                        //
-                        // This is dangerous if the track is MONO as that requires
-                        // special case handling due to implicit channel duplication.
-                        // Stereo or Multichannel should actually be fine here.
-                        mHook = getProcessHook(PROCESSTYPE_NORESAMPLEONETRACK,
-                                t->mMixerChannelCount, t->mMixerInFormat, t->mMixerFormat);
-                    }
-                }
-            }
-        }
-    }
-
-    ALOGV("mixer configuration change: %zu "
-        "all16BitsStereoNoResample=%d, resampling=%d, volumeRamp=%d",
-        mEnabled.size(), all16BitsStereoNoResample, resampling, volumeRamp);
-
-   process();
-
-    // Now that the volume ramp has been done, set optimal state and
-    // track hooks for subsequent mixer process
-    if (mEnabled.size() > 0) {
-        bool allMuted = true;
-
-        for (const int name : mEnabled) {
-            const std::shared_ptr<Track> &t = mTracks[name];
-            if (!t->doesResample() && t->volumeRL == 0) {
-                t->needs |= NEEDS_MUTE;
-                t->hook = &Track::track__nop;
-            } else {
-                allMuted = false;
-            }
-        }
-        if (allMuted) {
-            mHook = &AudioMixer::process__nop;
-        } else if (all16BitsStereoNoResample) {
-            if (mEnabled.size() == 1) {
-                //const int i = 31 - __builtin_clz(enabledTracks);
-                const std::shared_ptr<Track> &t = mTracks[mEnabled[0]];
-                // Muted single tracks handled by allMuted above.
-                mHook = getProcessHook(PROCESSTYPE_NORESAMPLEONETRACK,
-                        t->mMixerChannelCount, t->mMixerInFormat, t->mMixerFormat);
-            }
-        }
-    }
-}
-
-void AudioMixer::Track::track__genericResample(
-        int32_t* out, size_t outFrameCount, int32_t* temp, int32_t* aux)
-{
-    ALOGVV("track__genericResample\n");
-    mResampler->setSampleRate(sampleRate);
-
-    // ramp gain - resample to temp buffer and scale/mix in 2nd step
-    if (aux != NULL) {
-        // always resample with unity gain when sending to auxiliary buffer to be able
-        // to apply send level after resampling
-        mResampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
-        memset(temp, 0, outFrameCount * mMixerChannelCount * sizeof(int32_t));
-        mResampler->resample(temp, outFrameCount, bufferProvider);
-        if (CC_UNLIKELY(volumeInc[0]|volumeInc[1]|auxInc)) {
-            volumeRampStereo(out, outFrameCount, temp, aux);
-        } else {
-            volumeStereo(out, outFrameCount, temp, aux);
-        }
-    } else {
-        if (CC_UNLIKELY(volumeInc[0]|volumeInc[1])) {
-            mResampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
-            memset(temp, 0, outFrameCount * MAX_NUM_CHANNELS * sizeof(int32_t));
-            mResampler->resample(temp, outFrameCount, bufferProvider);
-            volumeRampStereo(out, outFrameCount, temp, aux);
-        }
-
-        // constant gain
-        else {
-            mResampler->setVolume(mVolume[0], mVolume[1]);
-            mResampler->resample(out, outFrameCount, bufferProvider);
-        }
-    }
-}
-
-void AudioMixer::Track::track__nop(int32_t* out __unused,
-        size_t outFrameCount __unused, int32_t* temp __unused, int32_t* aux __unused)
-{
-}
-
-void AudioMixer::Track::volumeRampStereo(
-        int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
-{
-    int32_t vl = prevVolume[0];
-    int32_t vr = prevVolume[1];
-    const int32_t vlInc = volumeInc[0];
-    const int32_t vrInc = volumeInc[1];
-
-    //ALOGD("[0] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
-    //        t, vlInc/65536.0f, vl/65536.0f, volume[0],
-    //       (vl + vlInc*frameCount)/65536.0f, frameCount);
-
-    // ramp volume
-    if (CC_UNLIKELY(aux != NULL)) {
-        int32_t va = prevAuxLevel;
-        const int32_t vaInc = auxInc;
-        int32_t l;
-        int32_t r;
-
-        do {
-            l = (*temp++ >> 12);
-            r = (*temp++ >> 12);
-            *out++ += (vl >> 16) * l;
-            *out++ += (vr >> 16) * r;
-            *aux++ += (va >> 17) * (l + r);
-            vl += vlInc;
-            vr += vrInc;
-            va += vaInc;
-        } while (--frameCount);
-        prevAuxLevel = va;
-    } else {
-        do {
-            *out++ += (vl >> 16) * (*temp++ >> 12);
-            *out++ += (vr >> 16) * (*temp++ >> 12);
-            vl += vlInc;
-            vr += vrInc;
-        } while (--frameCount);
-    }
-    prevVolume[0] = vl;
-    prevVolume[1] = vr;
-    adjustVolumeRamp(aux != NULL);
-}
-
-void AudioMixer::Track::volumeStereo(
-        int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
-{
-    const int16_t vl = volume[0];
-    const int16_t vr = volume[1];
-
-    if (CC_UNLIKELY(aux != NULL)) {
-        const int16_t va = auxLevel;
-        do {
-            int16_t l = (int16_t)(*temp++ >> 12);
-            int16_t r = (int16_t)(*temp++ >> 12);
-            out[0] = mulAdd(l, vl, out[0]);
-            int16_t a = (int16_t)(((int32_t)l + r) >> 1);
-            out[1] = mulAdd(r, vr, out[1]);
-            out += 2;
-            aux[0] = mulAdd(a, va, aux[0]);
-            aux++;
-        } while (--frameCount);
-    } else {
-        do {
-            int16_t l = (int16_t)(*temp++ >> 12);
-            int16_t r = (int16_t)(*temp++ >> 12);
-            out[0] = mulAdd(l, vl, out[0]);
-            out[1] = mulAdd(r, vr, out[1]);
-            out += 2;
-        } while (--frameCount);
-    }
-}
-
-void AudioMixer::Track::track__16BitsStereo(
-        int32_t* out, size_t frameCount, int32_t* temp __unused, int32_t* aux)
-{
-    ALOGVV("track__16BitsStereo\n");
-    const int16_t *in = static_cast<const int16_t *>(mIn);
-
-    if (CC_UNLIKELY(aux != NULL)) {
-        int32_t l;
-        int32_t r;
-        // ramp gain
-        if (CC_UNLIKELY(volumeInc[0]|volumeInc[1]|auxInc)) {
-            int32_t vl = prevVolume[0];
-            int32_t vr = prevVolume[1];
-            int32_t va = prevAuxLevel;
-            const int32_t vlInc = volumeInc[0];
-            const int32_t vrInc = volumeInc[1];
-            const int32_t vaInc = auxInc;
-            // ALOGD("[1] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
-            //        t, vlInc/65536.0f, vl/65536.0f, volume[0],
-            //        (vl + vlInc*frameCount)/65536.0f, frameCount);
-
-            do {
-                l = (int32_t)*in++;
-                r = (int32_t)*in++;
-                *out++ += (vl >> 16) * l;
-                *out++ += (vr >> 16) * r;
-                *aux++ += (va >> 17) * (l + r);
-                vl += vlInc;
-                vr += vrInc;
-                va += vaInc;
-            } while (--frameCount);
-
-            prevVolume[0] = vl;
-            prevVolume[1] = vr;
-            prevAuxLevel = va;
-            adjustVolumeRamp(true);
-        }
-
-        // constant gain
-        else {
-            const uint32_t vrl = volumeRL;
-            const int16_t va = (int16_t)auxLevel;
-            do {
-                uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
-                int16_t a = (int16_t)(((int32_t)in[0] + in[1]) >> 1);
-                in += 2;
-                out[0] = mulAddRL(1, rl, vrl, out[0]);
-                out[1] = mulAddRL(0, rl, vrl, out[1]);
-                out += 2;
-                aux[0] = mulAdd(a, va, aux[0]);
-                aux++;
-            } while (--frameCount);
-        }
-    } else {
-        // ramp gain
-        if (CC_UNLIKELY(volumeInc[0]|volumeInc[1])) {
-            int32_t vl = prevVolume[0];
-            int32_t vr = prevVolume[1];
-            const int32_t vlInc = volumeInc[0];
-            const int32_t vrInc = volumeInc[1];
-
-            // ALOGD("[1] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
-            //        t, vlInc/65536.0f, vl/65536.0f, volume[0],
-            //        (vl + vlInc*frameCount)/65536.0f, frameCount);
-
-            do {
-                *out++ += (vl >> 16) * (int32_t) *in++;
-                *out++ += (vr >> 16) * (int32_t) *in++;
-                vl += vlInc;
-                vr += vrInc;
-            } while (--frameCount);
-
-            prevVolume[0] = vl;
-            prevVolume[1] = vr;
-            adjustVolumeRamp(false);
-        }
-
-        // constant gain
-        else {
-            const uint32_t vrl = volumeRL;
-            do {
-                uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
-                in += 2;
-                out[0] = mulAddRL(1, rl, vrl, out[0]);
-                out[1] = mulAddRL(0, rl, vrl, out[1]);
-                out += 2;
-            } while (--frameCount);
-        }
-    }
-    mIn = in;
-}
-
-void AudioMixer::Track::track__16BitsMono(
-        int32_t* out, size_t frameCount, int32_t* temp __unused, int32_t* aux)
-{
-    ALOGVV("track__16BitsMono\n");
-    const int16_t *in = static_cast<int16_t const *>(mIn);
-
-    if (CC_UNLIKELY(aux != NULL)) {
-        // ramp gain
-        if (CC_UNLIKELY(volumeInc[0]|volumeInc[1]|auxInc)) {
-            int32_t vl = prevVolume[0];
-            int32_t vr = prevVolume[1];
-            int32_t va = prevAuxLevel;
-            const int32_t vlInc = volumeInc[0];
-            const int32_t vrInc = volumeInc[1];
-            const int32_t vaInc = auxInc;
-
-            // ALOGD("[2] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
-            //         t, vlInc/65536.0f, vl/65536.0f, volume[0],
-            //         (vl + vlInc*frameCount)/65536.0f, frameCount);
-
-            do {
-                int32_t l = *in++;
-                *out++ += (vl >> 16) * l;
-                *out++ += (vr >> 16) * l;
-                *aux++ += (va >> 16) * l;
-                vl += vlInc;
-                vr += vrInc;
-                va += vaInc;
-            } while (--frameCount);
-
-            prevVolume[0] = vl;
-            prevVolume[1] = vr;
-            prevAuxLevel = va;
-            adjustVolumeRamp(true);
-        }
-        // constant gain
-        else {
-            const int16_t vl = volume[0];
-            const int16_t vr = volume[1];
-            const int16_t va = (int16_t)auxLevel;
-            do {
-                int16_t l = *in++;
-                out[0] = mulAdd(l, vl, out[0]);
-                out[1] = mulAdd(l, vr, out[1]);
-                out += 2;
-                aux[0] = mulAdd(l, va, aux[0]);
-                aux++;
-            } while (--frameCount);
-        }
-    } else {
-        // ramp gain
-        if (CC_UNLIKELY(volumeInc[0]|volumeInc[1])) {
-            int32_t vl = prevVolume[0];
-            int32_t vr = prevVolume[1];
-            const int32_t vlInc = volumeInc[0];
-            const int32_t vrInc = volumeInc[1];
-
-            // ALOGD("[2] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
-            //         t, vlInc/65536.0f, vl/65536.0f, volume[0],
-            //         (vl + vlInc*frameCount)/65536.0f, frameCount);
-
-            do {
-                int32_t l = *in++;
-                *out++ += (vl >> 16) * l;
-                *out++ += (vr >> 16) * l;
-                vl += vlInc;
-                vr += vrInc;
-            } while (--frameCount);
-
-            prevVolume[0] = vl;
-            prevVolume[1] = vr;
-            adjustVolumeRamp(false);
-        }
-        // constant gain
-        else {
-            const int16_t vl = volume[0];
-            const int16_t vr = volume[1];
-            do {
-                int16_t l = *in++;
-                out[0] = mulAdd(l, vl, out[0]);
-                out[1] = mulAdd(l, vr, out[1]);
-                out += 2;
-            } while (--frameCount);
-        }
-    }
-    mIn = in;
-}
-
-// no-op case
-void AudioMixer::process__nop()
-{
-    ALOGVV("process__nop\n");
-
-    for (const auto &pair : mGroups) {
-        // process by group of tracks with same output buffer to
-        // avoid multiple memset() on same buffer
-        const auto &group = pair.second;
-
-        const std::shared_ptr<Track> &t = mTracks[group[0]];
-        memset(t->mainBuffer, 0,
-                mFrameCount * audio_bytes_per_frame(
-                        t->mMixerChannelCount + t->mMixerHapticChannelCount, t->mMixerFormat));
-
-        // now consume data
-        for (const int name : group) {
-            const std::shared_ptr<Track> &t = mTracks[name];
-            size_t outFrames = mFrameCount;
-            while (outFrames) {
-                t->buffer.frameCount = outFrames;
-                t->bufferProvider->getNextBuffer(&t->buffer);
-                if (t->buffer.raw == NULL) break;
-                outFrames -= t->buffer.frameCount;
-                t->bufferProvider->releaseBuffer(&t->buffer);
-            }
-        }
-    }
-}
-
-// generic code without resampling
-void AudioMixer::process__genericNoResampling()
-{
-    ALOGVV("process__genericNoResampling\n");
-    int32_t outTemp[BLOCKSIZE * MAX_NUM_CHANNELS] __attribute__((aligned(32)));
-
-    for (const auto &pair : mGroups) {
-        // process by group of tracks with same output main buffer to
-        // avoid multiple memset() on same buffer
-        const auto &group = pair.second;
-
-        // acquire buffer
-        for (const int name : group) {
-            const std::shared_ptr<Track> &t = mTracks[name];
-            t->buffer.frameCount = mFrameCount;
-            t->bufferProvider->getNextBuffer(&t->buffer);
-            t->frameCount = t->buffer.frameCount;
-            t->mIn = t->buffer.raw;
-        }
-
-        int32_t *out = (int *)pair.first;
-        size_t numFrames = 0;
-        do {
-            const size_t frameCount = std::min((size_t)BLOCKSIZE, mFrameCount - numFrames);
-            memset(outTemp, 0, sizeof(outTemp));
-            for (const int name : group) {
-                const std::shared_ptr<Track> &t = mTracks[name];
-                int32_t *aux = NULL;
-                if (CC_UNLIKELY(t->needs & NEEDS_AUX)) {
-                    aux = t->auxBuffer + numFrames;
-                }
-                for (int outFrames = frameCount; outFrames > 0; ) {
-                    // t->in == nullptr can happen if the track was flushed just after having
-                    // been enabled for mixing.
-                    if (t->mIn == nullptr) {
-                        break;
-                    }
-                    size_t inFrames = (t->frameCount > outFrames)?outFrames:t->frameCount;
-                    if (inFrames > 0) {
-                        (t.get()->*t->hook)(
-                                outTemp + (frameCount - outFrames) * t->mMixerChannelCount,
-                                inFrames, mResampleTemp.get() /* naked ptr */, aux);
-                        t->frameCount -= inFrames;
-                        outFrames -= inFrames;
-                        if (CC_UNLIKELY(aux != NULL)) {
-                            aux += inFrames;
-                        }
-                    }
-                    if (t->frameCount == 0 && outFrames) {
-                        t->bufferProvider->releaseBuffer(&t->buffer);
-                        t->buffer.frameCount = (mFrameCount - numFrames) -
-                                (frameCount - outFrames);
-                        t->bufferProvider->getNextBuffer(&t->buffer);
-                        t->mIn = t->buffer.raw;
-                        if (t->mIn == nullptr) {
-                            break;
-                        }
-                        t->frameCount = t->buffer.frameCount;
-                    }
-                }
-            }
-
-            const std::shared_ptr<Track> &t1 = mTracks[group[0]];
-            convertMixerFormat(out, t1->mMixerFormat, outTemp, t1->mMixerInFormat,
-                    frameCount * t1->mMixerChannelCount);
-            // TODO: fix ugly casting due to choice of out pointer type
-            out = reinterpret_cast<int32_t*>((uint8_t*)out
-                    + frameCount * t1->mMixerChannelCount
-                    * audio_bytes_per_sample(t1->mMixerFormat));
-            numFrames += frameCount;
-        } while (numFrames < mFrameCount);
-
-        // release each track's buffer
-        for (const int name : group) {
-            const std::shared_ptr<Track> &t = mTracks[name];
-            t->bufferProvider->releaseBuffer(&t->buffer);
-        }
-    }
-}
-
-// generic code with resampling
-void AudioMixer::process__genericResampling()
-{
-    ALOGVV("process__genericResampling\n");
-    int32_t * const outTemp = mOutputTemp.get(); // naked ptr
-    size_t numFrames = mFrameCount;
-
-    for (const auto &pair : mGroups) {
-        const auto &group = pair.second;
-        const std::shared_ptr<Track> &t1 = mTracks[group[0]];
-
-        // clear temp buffer
-        memset(outTemp, 0, sizeof(*outTemp) * t1->mMixerChannelCount * mFrameCount);
-        for (const int name : group) {
-            const std::shared_ptr<Track> &t = mTracks[name];
-            int32_t *aux = NULL;
-            if (CC_UNLIKELY(t->needs & NEEDS_AUX)) {
-                aux = t->auxBuffer;
-            }
-
-            // this is a little goofy, on the resampling case we don't
-            // acquire/release the buffers because it's done by
-            // the resampler.
-            if (t->needs & NEEDS_RESAMPLE) {
-                (t.get()->*t->hook)(outTemp, numFrames, mResampleTemp.get() /* naked ptr */, aux);
-            } else {
-
-                size_t outFrames = 0;
-
-                while (outFrames < numFrames) {
-                    t->buffer.frameCount = numFrames - outFrames;
-                    t->bufferProvider->getNextBuffer(&t->buffer);
-                    t->mIn = t->buffer.raw;
-                    // t->mIn == nullptr can happen if the track was flushed just after having
-                    // been enabled for mixing.
-                    if (t->mIn == nullptr) break;
-
-                    (t.get()->*t->hook)(
-                            outTemp + outFrames * t->mMixerChannelCount, t->buffer.frameCount,
-                            mResampleTemp.get() /* naked ptr */,
-                            aux != nullptr ? aux + outFrames : nullptr);
-                    outFrames += t->buffer.frameCount;
-
-                    t->bufferProvider->releaseBuffer(&t->buffer);
-                }
-            }
-        }
-        convertMixerFormat(t1->mainBuffer, t1->mMixerFormat,
-                outTemp, t1->mMixerInFormat, numFrames * t1->mMixerChannelCount);
-    }
-}
-
-// one track, 16 bits stereo without resampling is the most common case
-void AudioMixer::process__oneTrack16BitsStereoNoResampling()
-{
-    ALOGVV("process__oneTrack16BitsStereoNoResampling\n");
-    LOG_ALWAYS_FATAL_IF(mEnabled.size() != 0,
-            "%zu != 1 tracks enabled", mEnabled.size());
-    const int name = mEnabled[0];
-    const std::shared_ptr<Track> &t = mTracks[name];
-
-    AudioBufferProvider::Buffer& b(t->buffer);
-
-    int32_t* out = t->mainBuffer;
-    float *fout = reinterpret_cast<float*>(out);
-    size_t numFrames = mFrameCount;
-
-    const int16_t vl = t->volume[0];
-    const int16_t vr = t->volume[1];
-    const uint32_t vrl = t->volumeRL;
-    while (numFrames) {
-        b.frameCount = numFrames;
-        t->bufferProvider->getNextBuffer(&b);
-        const int16_t *in = b.i16;
-
-        // in == NULL can happen if the track was flushed just after having
-        // been enabled for mixing.
-        if (in == NULL || (((uintptr_t)in) & 3)) {
-            if ( AUDIO_FORMAT_PCM_FLOAT == t->mMixerFormat ) {
-                 memset((char*)fout, 0, numFrames
-                         * t->mMixerChannelCount * audio_bytes_per_sample(t->mMixerFormat));
-            } else {
-                 memset((char*)out, 0, numFrames
-                         * t->mMixerChannelCount * audio_bytes_per_sample(t->mMixerFormat));
-            }
-            ALOGE_IF((((uintptr_t)in) & 3),
-                    "process__oneTrack16BitsStereoNoResampling: misaligned buffer"
-                    " %p track %d, channels %d, needs %08x, volume %08x vfl %f vfr %f",
-                    in, name, t->channelCount, t->needs, vrl, t->mVolume[0], t->mVolume[1]);
-            return;
-        }
-        size_t outFrames = b.frameCount;
-
-        switch (t->mMixerFormat) {
-        case AUDIO_FORMAT_PCM_FLOAT:
-            do {
-                uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
-                in += 2;
-                int32_t l = mulRL(1, rl, vrl);
-                int32_t r = mulRL(0, rl, vrl);
-                *fout++ = float_from_q4_27(l);
-                *fout++ = float_from_q4_27(r);
-                // Note: In case of later int16_t sink output,
-                // conversion and clamping is done by memcpy_to_i16_from_float().
-            } while (--outFrames);
-            break;
-        case AUDIO_FORMAT_PCM_16_BIT:
-            if (CC_UNLIKELY(uint32_t(vl) > UNITY_GAIN_INT || uint32_t(vr) > UNITY_GAIN_INT)) {
-                // volume is boosted, so we might need to clamp even though
-                // we process only one track.
-                do {
-                    uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
-                    in += 2;
-                    int32_t l = mulRL(1, rl, vrl) >> 12;
-                    int32_t r = mulRL(0, rl, vrl) >> 12;
-                    // clamping...
-                    l = clamp16(l);
-                    r = clamp16(r);
-                    *out++ = (r<<16) | (l & 0xFFFF);
-                } while (--outFrames);
-            } else {
-                do {
-                    uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
-                    in += 2;
-                    int32_t l = mulRL(1, rl, vrl) >> 12;
-                    int32_t r = mulRL(0, rl, vrl) >> 12;
-                    *out++ = (r<<16) | (l & 0xFFFF);
-                } while (--outFrames);
-            }
-            break;
-        default:
-            LOG_ALWAYS_FATAL("bad mixer format: %d", t->mMixerFormat);
-        }
-        numFrames -= b.frameCount;
-        t->bufferProvider->releaseBuffer(&b);
-    }
-}
-
 /*static*/ pthread_once_t AudioMixer::sOnceControl = PTHREAD_ONCE_INIT;
 
 /*static*/ void AudioMixer::sInitRoutine()
@@ -1656,211 +523,71 @@
     DownmixerBufferProvider::init(); // for the downmixer
 }
 
-/* TODO: consider whether this level of optimization is necessary.
- * Perhaps just stick with a single for loop.
- */
-
-// Needs to derive a compile time constant (constexpr).  Could be targeted to go
-// to a MONOVOL mixtype based on MAX_NUM_VOLUMES, but that's an unnecessary complication.
-#define MIXTYPE_MONOVOL(mixtype) ((mixtype) == MIXTYPE_MULTI ? MIXTYPE_MULTI_MONOVOL : \
-        (mixtype) == MIXTYPE_MULTI_SAVEONLY ? MIXTYPE_MULTI_SAVEONLY_MONOVOL : (mixtype))
-
-/* MIXTYPE     (see AudioMixerOps.h MIXTYPE_* enumeration)
- * TO: int32_t (Q4.27) or float
- * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
- * TA: int32_t (Q4.27) or float
- */
-template <int MIXTYPE,
-        typename TO, typename TI, typename TV, typename TA, typename TAV>
-static void volumeRampMulti(uint32_t channels, TO* out, size_t frameCount,
-        const TI* in, TA* aux, TV *vol, const TV *volinc, TAV *vola, TAV volainc)
+std::shared_ptr<AudioMixerBase::TrackBase> AudioMixer::preCreateTrack()
 {
-    switch (channels) {
-    case 1:
-        volumeRampMulti<MIXTYPE, 1>(out, frameCount, in, aux, vol, volinc, vola, volainc);
-        break;
-    case 2:
-        volumeRampMulti<MIXTYPE, 2>(out, frameCount, in, aux, vol, volinc, vola, volainc);
-        break;
-    case 3:
-        volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 3>(out,
-                frameCount, in, aux, vol, volinc, vola, volainc);
-        break;
-    case 4:
-        volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 4>(out,
-                frameCount, in, aux, vol, volinc, vola, volainc);
-        break;
-    case 5:
-        volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 5>(out,
-                frameCount, in, aux, vol, volinc, vola, volainc);
-        break;
-    case 6:
-        volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 6>(out,
-                frameCount, in, aux, vol, volinc, vola, volainc);
-        break;
-    case 7:
-        volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 7>(out,
-                frameCount, in, aux, vol, volinc, vola, volainc);
-        break;
-    case 8:
-        volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 8>(out,
-                frameCount, in, aux, vol, volinc, vola, volainc);
-        break;
-    }
+    return std::make_shared<Track>();
 }
 
-/* MIXTYPE     (see AudioMixerOps.h MIXTYPE_* enumeration)
- * TO: int32_t (Q4.27) or float
- * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
- * TA: int32_t (Q4.27) or float
- */
-template <int MIXTYPE,
-        typename TO, typename TI, typename TV, typename TA, typename TAV>
-static void volumeMulti(uint32_t channels, TO* out, size_t frameCount,
-        const TI* in, TA* aux, const TV *vol, TAV vola)
+status_t AudioMixer::postCreateTrack(TrackBase *track)
 {
-    switch (channels) {
-    case 1:
-        volumeMulti<MIXTYPE, 1>(out, frameCount, in, aux, vol, vola);
-        break;
-    case 2:
-        volumeMulti<MIXTYPE, 2>(out, frameCount, in, aux, vol, vola);
-        break;
-    case 3:
-        volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 3>(out, frameCount, in, aux, vol, vola);
-        break;
-    case 4:
-        volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 4>(out, frameCount, in, aux, vol, vola);
-        break;
-    case 5:
-        volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 5>(out, frameCount, in, aux, vol, vola);
-        break;
-    case 6:
-        volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 6>(out, frameCount, in, aux, vol, vola);
-        break;
-    case 7:
-        volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 7>(out, frameCount, in, aux, vol, vola);
-        break;
-    case 8:
-        volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 8>(out, frameCount, in, aux, vol, vola);
-        break;
+    Track* t = static_cast<Track*>(track);
+
+    audio_channel_mask_t channelMask = t->channelMask;
+    t->mHapticChannelMask = channelMask & AUDIO_CHANNEL_HAPTIC_ALL;
+    t->mHapticChannelCount = audio_channel_count_from_out_mask(t->mHapticChannelMask);
+    channelMask &= ~AUDIO_CHANNEL_HAPTIC_ALL;
+    t->channelCount = audio_channel_count_from_out_mask(channelMask);
+    ALOGV_IF(audio_channel_mask_get_bits(channelMask) != AUDIO_CHANNEL_OUT_STEREO,
+            "Non-stereo channel mask: %d\n", channelMask);
+    t->channelMask = channelMask;
+    t->mInputBufferProvider = NULL;
+    t->mDownmixRequiresFormat = AUDIO_FORMAT_INVALID; // no format required
+    t->mPlaybackRate = AUDIO_PLAYBACK_RATE_DEFAULT;
+    // haptic
+    t->mHapticPlaybackEnabled = false;
+    t->mHapticIntensity = HAPTIC_SCALE_NONE;
+    t->mMixerHapticChannelMask = AUDIO_CHANNEL_NONE;
+    t->mMixerHapticChannelCount = 0;
+    t->mAdjustInChannelCount = t->channelCount + t->mHapticChannelCount;
+    t->mAdjustOutChannelCount = t->channelCount + t->mMixerHapticChannelCount;
+    t->mAdjustNonDestructiveInChannelCount = t->mAdjustOutChannelCount;
+    t->mAdjustNonDestructiveOutChannelCount = t->channelCount;
+    t->mKeepContractedChannels = false;
+    // Check the downmixing (or upmixing) requirements.
+    status_t status = t->prepareForDownmix();
+    if (status != OK) {
+        ALOGE("AudioMixer::getTrackName invalid channelMask (%#x)", channelMask);
+        return BAD_VALUE;
     }
+    // prepareForDownmix() may change mDownmixRequiresFormat
+    ALOGVV("mMixerFormat:%#x  mMixerInFormat:%#x\n", t->mMixerFormat, t->mMixerInFormat);
+    t->prepareForReformat();
+    t->prepareForAdjustChannelsNonDestructive(mFrameCount);
+    t->prepareForAdjustChannels();
+    return OK;
 }
 
-/* MIXTYPE     (see AudioMixerOps.h MIXTYPE_* enumeration)
- * USEFLOATVOL (set to true if float volume is used)
- * ADJUSTVOL   (set to true if volume ramp parameters needs adjustment afterwards)
- * TO: int32_t (Q4.27) or float
- * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
- * TA: int32_t (Q4.27) or float
- */
-template <int MIXTYPE, bool USEFLOATVOL, bool ADJUSTVOL,
-    typename TO, typename TI, typename TA>
-void AudioMixer::Track::volumeMix(TO *out, size_t outFrames,
-        const TI *in, TA *aux, bool ramp)
+void AudioMixer::preProcess()
 {
-    if (USEFLOATVOL) {
-        if (ramp) {
-            volumeRampMulti<MIXTYPE>(mMixerChannelCount, out, outFrames, in, aux,
-                    mPrevVolume, mVolumeInc,
-#ifdef FLOAT_AUX
-                    &mPrevAuxLevel, mAuxInc
-#else
-                    &prevAuxLevel, auxInc
-#endif
-                );
-            if (ADJUSTVOL) {
-                adjustVolumeRamp(aux != NULL, true);
-            }
-        } else {
-            volumeMulti<MIXTYPE>(mMixerChannelCount, out, outFrames, in, aux,
-                    mVolume,
-#ifdef FLOAT_AUX
-                    mAuxLevel
-#else
-                    auxLevel
-#endif
-            );
-        }
-    } else {
-        if (ramp) {
-            volumeRampMulti<MIXTYPE>(mMixerChannelCount, out, outFrames, in, aux,
-                    prevVolume, volumeInc, &prevAuxLevel, auxInc);
-            if (ADJUSTVOL) {
-                adjustVolumeRamp(aux != NULL);
-            }
-        } else {
-            volumeMulti<MIXTYPE>(mMixerChannelCount, out, outFrames, in, aux,
-                    volume, auxLevel);
+    for (const auto &pair : mTracks) {
+        // Clear contracted buffer before processing if contracted channels are saved
+        const std::shared_ptr<TrackBase> &tb = pair.second;
+        Track *t = static_cast<Track*>(tb.get());
+        if (t->mKeepContractedChannels) {
+            t->clearContractedBuffer();
         }
     }
 }
 
-/* This process hook is called when there is a single track without
- * aux buffer, volume ramp, or resampling.
- * TODO: Update the hook selection: this can properly handle aux and ramp.
- *
- * MIXTYPE     (see AudioMixerOps.h MIXTYPE_* enumeration)
- * TO: int32_t (Q4.27) or float
- * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
- * TA: int32_t (Q4.27)
- */
-template <int MIXTYPE, typename TO, typename TI, typename TA>
-void AudioMixer::process__noResampleOneTrack()
+void AudioMixer::postProcess()
 {
-    ALOGVV("process__noResampleOneTrack\n");
-    LOG_ALWAYS_FATAL_IF(mEnabled.size() != 1,
-            "%zu != 1 tracks enabled", mEnabled.size());
-    const std::shared_ptr<Track> &t = mTracks[mEnabled[0]];
-    const uint32_t channels = t->mMixerChannelCount;
-    TO* out = reinterpret_cast<TO*>(t->mainBuffer);
-    TA* aux = reinterpret_cast<TA*>(t->auxBuffer);
-    const bool ramp = t->needsRamp();
-
-    for (size_t numFrames = mFrameCount; numFrames > 0; ) {
-        AudioBufferProvider::Buffer& b(t->buffer);
-        // get input buffer
-        b.frameCount = numFrames;
-        t->bufferProvider->getNextBuffer(&b);
-        const TI *in = reinterpret_cast<TI*>(b.raw);
-
-        // in == NULL can happen if the track was flushed just after having
-        // been enabled for mixing.
-        if (in == NULL || (((uintptr_t)in) & 3)) {
-            memset(out, 0, numFrames
-                    * channels * audio_bytes_per_sample(t->mMixerFormat));
-            ALOGE_IF((((uintptr_t)in) & 3), "process__noResampleOneTrack: bus error: "
-                    "buffer %p track %p, channels %d, needs %#x",
-                    in, &t, t->channelCount, t->needs);
-            return;
-        }
-
-        const size_t outFrames = b.frameCount;
-        t->volumeMix<MIXTYPE, is_same<TI, float>::value /* USEFLOATVOL */, false /* ADJUSTVOL */> (
-                out, outFrames, in, aux, ramp);
-
-        out += outFrames * channels;
-        if (aux != NULL) {
-            aux += outFrames;
-        }
-        numFrames -= b.frameCount;
-
-        // release buffer
-        t->bufferProvider->releaseBuffer(&b);
-    }
-    if (ramp) {
-        t->adjustVolumeRamp(aux != NULL, is_same<TI, float>::value);
-    }
-}
-
-void AudioMixer::processHapticData()
-{
+    // Process haptic data.
     // Need to keep consistent with VibrationEffect.scale(int, float, int)
     for (const auto &pair : mGroups) {
         // process by group of tracks with same output main buffer.
         const auto &group = pair.second;
         for (const int name : group) {
-            const std::shared_ptr<Track> &t = mTracks[name];
+            const std::shared_ptr<Track> &t = getTrack(name);
             if (t->mHapticPlaybackEnabled) {
                 size_t sampleCount = mFrameCount * t->mMixerHapticChannelCount;
                 float gamma = t->getHapticScaleGamma();
@@ -1887,225 +614,5 @@
     }
 }
 
-/* This track hook is called to do resampling then mixing,
- * pulling from the track's upstream AudioBufferProvider.
- *
- * MIXTYPE     (see AudioMixerOps.h MIXTYPE_* enumeration)
- * TO: int32_t (Q4.27) or float
- * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
- * TA: int32_t (Q4.27) or float
- */
-template <int MIXTYPE, typename TO, typename TI, typename TA>
-void AudioMixer::Track::track__Resample(TO* out, size_t outFrameCount, TO* temp, TA* aux)
-{
-    ALOGVV("track__Resample\n");
-    mResampler->setSampleRate(sampleRate);
-    const bool ramp = needsRamp();
-    if (ramp || aux != NULL) {
-        // if ramp:        resample with unity gain to temp buffer and scale/mix in 2nd step.
-        // if aux != NULL: resample with unity gain to temp buffer then apply send level.
-
-        mResampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
-        memset(temp, 0, outFrameCount * mMixerChannelCount * sizeof(TO));
-        mResampler->resample((int32_t*)temp, outFrameCount, bufferProvider);
-
-        volumeMix<MIXTYPE, is_same<TI, float>::value /* USEFLOATVOL */, true /* ADJUSTVOL */>(
-                out, outFrameCount, temp, aux, ramp);
-
-    } else { // constant volume gain
-        mResampler->setVolume(mVolume[0], mVolume[1]);
-        mResampler->resample((int32_t*)out, outFrameCount, bufferProvider);
-    }
-}
-
-/* This track hook is called to mix a track, when no resampling is required.
- * The input buffer should be present in in.
- *
- * MIXTYPE     (see AudioMixerOps.h MIXTYPE_* enumeration)
- * TO: int32_t (Q4.27) or float
- * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
- * TA: int32_t (Q4.27) or float
- */
-template <int MIXTYPE, typename TO, typename TI, typename TA>
-void AudioMixer::Track::track__NoResample(TO* out, size_t frameCount, TO* temp __unused, TA* aux)
-{
-    ALOGVV("track__NoResample\n");
-    const TI *in = static_cast<const TI *>(mIn);
-
-    volumeMix<MIXTYPE, is_same<TI, float>::value /* USEFLOATVOL */, true /* ADJUSTVOL */>(
-            out, frameCount, in, aux, needsRamp());
-
-    // MIXTYPE_MONOEXPAND reads a single input channel and expands to NCHAN output channels.
-    // MIXTYPE_MULTI reads NCHAN input channels and places to NCHAN output channels.
-    in += (MIXTYPE == MIXTYPE_MONOEXPAND) ? frameCount : frameCount * mMixerChannelCount;
-    mIn = in;
-}
-
-/* The Mixer engine generates either int32_t (Q4_27) or float data.
- * We use this function to convert the engine buffers
- * to the desired mixer output format, either int16_t (Q.15) or float.
- */
-/* static */
-void AudioMixer::convertMixerFormat(void *out, audio_format_t mixerOutFormat,
-        void *in, audio_format_t mixerInFormat, size_t sampleCount)
-{
-    switch (mixerInFormat) {
-    case AUDIO_FORMAT_PCM_FLOAT:
-        switch (mixerOutFormat) {
-        case AUDIO_FORMAT_PCM_FLOAT:
-            memcpy(out, in, sampleCount * sizeof(float)); // MEMCPY. TODO optimize out
-            break;
-        case AUDIO_FORMAT_PCM_16_BIT:
-            memcpy_to_i16_from_float((int16_t*)out, (float*)in, sampleCount);
-            break;
-        default:
-            LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
-            break;
-        }
-        break;
-    case AUDIO_FORMAT_PCM_16_BIT:
-        switch (mixerOutFormat) {
-        case AUDIO_FORMAT_PCM_FLOAT:
-            memcpy_to_float_from_q4_27((float*)out, (const int32_t*)in, sampleCount);
-            break;
-        case AUDIO_FORMAT_PCM_16_BIT:
-            memcpy_to_i16_from_q4_27((int16_t*)out, (const int32_t*)in, sampleCount);
-            break;
-        default:
-            LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
-            break;
-        }
-        break;
-    default:
-        LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
-        break;
-    }
-}
-
-/* Returns the proper track hook to use for mixing the track into the output buffer.
- */
-/* static */
-AudioMixer::hook_t AudioMixer::Track::getTrackHook(int trackType, uint32_t channelCount,
-        audio_format_t mixerInFormat, audio_format_t mixerOutFormat __unused)
-{
-    if (!kUseNewMixer && channelCount == FCC_2 && mixerInFormat == AUDIO_FORMAT_PCM_16_BIT) {
-        switch (trackType) {
-        case TRACKTYPE_NOP:
-            return &Track::track__nop;
-        case TRACKTYPE_RESAMPLE:
-            return &Track::track__genericResample;
-        case TRACKTYPE_NORESAMPLEMONO:
-            return &Track::track__16BitsMono;
-        case TRACKTYPE_NORESAMPLE:
-            return &Track::track__16BitsStereo;
-        default:
-            LOG_ALWAYS_FATAL("bad trackType: %d", trackType);
-            break;
-        }
-    }
-    LOG_ALWAYS_FATAL_IF(channelCount > MAX_NUM_CHANNELS);
-    switch (trackType) {
-    case TRACKTYPE_NOP:
-        return &Track::track__nop;
-    case TRACKTYPE_RESAMPLE:
-        switch (mixerInFormat) {
-        case AUDIO_FORMAT_PCM_FLOAT:
-            return (AudioMixer::hook_t) &Track::track__Resample<
-                    MIXTYPE_MULTI, float /*TO*/, float /*TI*/, TYPE_AUX>;
-        case AUDIO_FORMAT_PCM_16_BIT:
-            return (AudioMixer::hook_t) &Track::track__Resample<
-                    MIXTYPE_MULTI, int32_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;
-        default:
-            LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
-            break;
-        }
-        break;
-    case TRACKTYPE_NORESAMPLEMONO:
-        switch (mixerInFormat) {
-        case AUDIO_FORMAT_PCM_FLOAT:
-            return (AudioMixer::hook_t) &Track::track__NoResample<
-                            MIXTYPE_MONOEXPAND, float /*TO*/, float /*TI*/, TYPE_AUX>;
-        case AUDIO_FORMAT_PCM_16_BIT:
-            return (AudioMixer::hook_t) &Track::track__NoResample<
-                            MIXTYPE_MONOEXPAND, int32_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;
-        default:
-            LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
-            break;
-        }
-        break;
-    case TRACKTYPE_NORESAMPLE:
-        switch (mixerInFormat) {
-        case AUDIO_FORMAT_PCM_FLOAT:
-            return (AudioMixer::hook_t) &Track::track__NoResample<
-                    MIXTYPE_MULTI, float /*TO*/, float /*TI*/, TYPE_AUX>;
-        case AUDIO_FORMAT_PCM_16_BIT:
-            return (AudioMixer::hook_t) &Track::track__NoResample<
-                    MIXTYPE_MULTI, int32_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;
-        default:
-            LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
-            break;
-        }
-        break;
-    default:
-        LOG_ALWAYS_FATAL("bad trackType: %d", trackType);
-        break;
-    }
-    return NULL;
-}
-
-/* Returns the proper process hook for mixing tracks. Currently works only for
- * PROCESSTYPE_NORESAMPLEONETRACK, a mix involving one track, no resampling.
- *
- * TODO: Due to the special mixing considerations of duplicating to
- * a stereo output track, the input track cannot be MONO.  This should be
- * prevented by the caller.
- */
-/* static */
-AudioMixer::process_hook_t AudioMixer::getProcessHook(
-        int processType, uint32_t channelCount,
-        audio_format_t mixerInFormat, audio_format_t mixerOutFormat)
-{
-    if (processType != PROCESSTYPE_NORESAMPLEONETRACK) { // Only NORESAMPLEONETRACK
-        LOG_ALWAYS_FATAL("bad processType: %d", processType);
-        return NULL;
-    }
-    if (!kUseNewMixer && channelCount == FCC_2 && mixerInFormat == AUDIO_FORMAT_PCM_16_BIT) {
-        return &AudioMixer::process__oneTrack16BitsStereoNoResampling;
-    }
-    LOG_ALWAYS_FATAL_IF(channelCount > MAX_NUM_CHANNELS);
-    switch (mixerInFormat) {
-    case AUDIO_FORMAT_PCM_FLOAT:
-        switch (mixerOutFormat) {
-        case AUDIO_FORMAT_PCM_FLOAT:
-            return &AudioMixer::process__noResampleOneTrack<
-                    MIXTYPE_MULTI_SAVEONLY, float /*TO*/, float /*TI*/, TYPE_AUX>;
-        case AUDIO_FORMAT_PCM_16_BIT:
-            return &AudioMixer::process__noResampleOneTrack<
-                    MIXTYPE_MULTI_SAVEONLY, int16_t /*TO*/, float /*TI*/, TYPE_AUX>;
-        default:
-            LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
-            break;
-        }
-        break;
-    case AUDIO_FORMAT_PCM_16_BIT:
-        switch (mixerOutFormat) {
-        case AUDIO_FORMAT_PCM_FLOAT:
-            return &AudioMixer::process__noResampleOneTrack<
-                    MIXTYPE_MULTI_SAVEONLY, float /*TO*/, int16_t /*TI*/, TYPE_AUX>;
-        case AUDIO_FORMAT_PCM_16_BIT:
-            return &AudioMixer::process__noResampleOneTrack<
-                    MIXTYPE_MULTI_SAVEONLY, int16_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;
-        default:
-            LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
-            break;
-        }
-        break;
-    default:
-        LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
-        break;
-    }
-    return NULL;
-}
-
 // ----------------------------------------------------------------------------
 } // namespace android
diff --git a/media/libaudioprocessing/AudioMixerBase.cpp b/media/libaudioprocessing/AudioMixerBase.cpp
new file mode 100644
index 0000000..75c077d
--- /dev/null
+++ b/media/libaudioprocessing/AudioMixerBase.cpp
@@ -0,0 +1,1692 @@
+/*
+**
+** Copyright 2019, 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_TAG "AudioMixer"
+//#define LOG_NDEBUG 0
+
+#include <sstream>
+#include <string.h>
+
+#include <audio_utils/primitives.h>
+#include <cutils/compiler.h>
+#include <media/AudioMixerBase.h>
+#include <utils/Log.h>
+
+#include "AudioMixerOps.h"
+
+// The FCC_2 macro refers to the Fixed Channel Count of 2 for the legacy integer mixer.
+#ifndef FCC_2
+#define FCC_2 2
+#endif
+
+// Look for MONO_HACK for any Mono hack involving legacy mono channel to
+// stereo channel conversion.
+
+/* VERY_VERY_VERBOSE_LOGGING will show exactly which process hook and track hook is
+ * being used. This is a considerable amount of log spam, so don't enable unless you
+ * are verifying the hook based code.
+ */
+//#define VERY_VERY_VERBOSE_LOGGING
+#ifdef VERY_VERY_VERBOSE_LOGGING
+#define ALOGVV ALOGV
+//define ALOGVV printf  // for test-mixer.cpp
+#else
+#define ALOGVV(a...) do { } while (0)
+#endif
+
+// TODO: remove BLOCKSIZE unit of processing - it isn't needed anymore.
+static constexpr int BLOCKSIZE = 16;
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+bool AudioMixerBase::isValidFormat(audio_format_t format) const
+{
+    switch (format) {
+    case AUDIO_FORMAT_PCM_8_BIT:
+    case AUDIO_FORMAT_PCM_16_BIT:
+    case AUDIO_FORMAT_PCM_24_BIT_PACKED:
+    case AUDIO_FORMAT_PCM_32_BIT:
+    case AUDIO_FORMAT_PCM_FLOAT:
+        return true;
+    default:
+        return false;
+    }
+}
+
+bool AudioMixerBase::isValidChannelMask(audio_channel_mask_t channelMask) const
+{
+    return audio_channel_count_from_out_mask(channelMask) <= MAX_NUM_CHANNELS;
+}
+
+std::shared_ptr<AudioMixerBase::TrackBase> AudioMixerBase::preCreateTrack()
+{
+    return std::make_shared<TrackBase>();
+}
+
+status_t AudioMixerBase::create(
+        int name, audio_channel_mask_t channelMask, audio_format_t format, int sessionId)
+{
+    LOG_ALWAYS_FATAL_IF(exists(name), "name %d already exists", name);
+
+    if (!isValidChannelMask(channelMask)) {
+        ALOGE("%s invalid channelMask: %#x", __func__, channelMask);
+        return BAD_VALUE;
+    }
+    if (!isValidFormat(format)) {
+        ALOGE("%s invalid format: %#x", __func__, format);
+        return BAD_VALUE;
+    }
+
+    auto t = preCreateTrack();
+    {
+        // TODO: move initialization to the Track constructor.
+        // assume default parameters for the track, except where noted below
+        t->needs = 0;
+
+        // Integer volume.
+        // Currently integer volume is kept for the legacy integer mixer.
+        // Will be removed when the legacy mixer path is removed.
+        t->volume[0] = 0;
+        t->volume[1] = 0;
+        t->prevVolume[0] = 0 << 16;
+        t->prevVolume[1] = 0 << 16;
+        t->volumeInc[0] = 0;
+        t->volumeInc[1] = 0;
+        t->auxLevel = 0;
+        t->auxInc = 0;
+        t->prevAuxLevel = 0;
+
+        // Floating point volume.
+        t->mVolume[0] = 0.f;
+        t->mVolume[1] = 0.f;
+        t->mPrevVolume[0] = 0.f;
+        t->mPrevVolume[1] = 0.f;
+        t->mVolumeInc[0] = 0.;
+        t->mVolumeInc[1] = 0.;
+        t->mAuxLevel = 0.;
+        t->mAuxInc = 0.;
+        t->mPrevAuxLevel = 0.;
+
+        // no initialization needed
+        // t->frameCount
+        t->channelCount = audio_channel_count_from_out_mask(channelMask);
+        t->enabled = false;
+        ALOGV_IF(audio_channel_mask_get_bits(channelMask) != AUDIO_CHANNEL_OUT_STEREO,
+                "Non-stereo channel mask: %d\n", channelMask);
+        t->channelMask = channelMask;
+        t->sessionId = sessionId;
+        // setBufferProvider(name, AudioBufferProvider *) is required before enable(name)
+        t->bufferProvider = NULL;
+        t->buffer.raw = NULL;
+        // no initialization needed
+        // t->buffer.frameCount
+        t->hook = NULL;
+        t->mIn = NULL;
+        t->sampleRate = mSampleRate;
+        // setParameter(name, TRACK, MAIN_BUFFER, mixBuffer) is required before enable(name)
+        t->mainBuffer = NULL;
+        t->auxBuffer = NULL;
+        t->mMixerFormat = AUDIO_FORMAT_PCM_16_BIT;
+        t->mFormat = format;
+        t->mMixerInFormat = kUseFloat && kUseNewMixer ?
+                AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
+        t->mMixerChannelMask = audio_channel_mask_from_representation_and_bits(
+                AUDIO_CHANNEL_REPRESENTATION_POSITION, AUDIO_CHANNEL_OUT_STEREO);
+        t->mMixerChannelCount = audio_channel_count_from_out_mask(t->mMixerChannelMask);
+        status_t status = postCreateTrack(t.get());
+        if (status != OK) return status;
+        mTracks[name] = t;
+        return OK;
+    }
+}
+
+// Called when channel masks have changed for a track name
+bool AudioMixerBase::setChannelMasks(int name,
+        audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask)
+{
+    LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
+    const std::shared_ptr<TrackBase> &track = mTracks[name];
+
+    if (trackChannelMask == track->channelMask && mixerChannelMask == track->mMixerChannelMask) {
+        return false;  // no need to change
+    }
+    // always recompute for both channel masks even if only one has changed.
+    const uint32_t trackChannelCount = audio_channel_count_from_out_mask(trackChannelMask);
+    const uint32_t mixerChannelCount = audio_channel_count_from_out_mask(mixerChannelMask);
+
+    ALOG_ASSERT(trackChannelCount && mixerChannelCount);
+    track->channelMask = trackChannelMask;
+    track->channelCount = trackChannelCount;
+    track->mMixerChannelMask = mixerChannelMask;
+    track->mMixerChannelCount = mixerChannelCount;
+
+    // Resampler channels may have changed.
+    track->recreateResampler(mSampleRate);
+    return true;
+}
+
+void AudioMixerBase::destroy(int name)
+{
+    LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
+    ALOGV("deleteTrackName(%d)", name);
+
+    if (mTracks[name]->enabled) {
+        invalidate();
+    }
+    mTracks.erase(name); // deallocate track
+}
+
+void AudioMixerBase::enable(int name)
+{
+    LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
+    const std::shared_ptr<TrackBase> &track = mTracks[name];
+
+    if (!track->enabled) {
+        track->enabled = true;
+        ALOGV("enable(%d)", name);
+        invalidate();
+    }
+}
+
+void AudioMixerBase::disable(int name)
+{
+    LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
+    const std::shared_ptr<TrackBase> &track = mTracks[name];
+
+    if (track->enabled) {
+        track->enabled = false;
+        ALOGV("disable(%d)", name);
+        invalidate();
+    }
+}
+
+/* Sets the volume ramp variables for the AudioMixer.
+ *
+ * The volume ramp variables are used to transition from the previous
+ * volume to the set volume.  ramp controls the duration of the transition.
+ * Its value is typically one state framecount period, but may also be 0,
+ * meaning "immediate."
+ *
+ * FIXME: 1) Volume ramp is enabled only if there is a nonzero integer increment
+ * even if there is a nonzero floating point increment (in that case, the volume
+ * change is immediate).  This restriction should be changed when the legacy mixer
+ * is removed (see #2).
+ * FIXME: 2) Integer volume variables are used for Legacy mixing and should be removed
+ * when no longer needed.
+ *
+ * @param newVolume set volume target in floating point [0.0, 1.0].
+ * @param ramp number of frames to increment over. if ramp is 0, the volume
+ * should be set immediately.  Currently ramp should not exceed 65535 (frames).
+ * @param pIntSetVolume pointer to the U4.12 integer target volume, set on return.
+ * @param pIntPrevVolume pointer to the U4.28 integer previous volume, set on return.
+ * @param pIntVolumeInc pointer to the U4.28 increment per output audio frame, set on return.
+ * @param pSetVolume pointer to the float target volume, set on return.
+ * @param pPrevVolume pointer to the float previous volume, set on return.
+ * @param pVolumeInc pointer to the float increment per output audio frame, set on return.
+ * @return true if the volume has changed, false if volume is same.
+ */
+static inline bool setVolumeRampVariables(float newVolume, int32_t ramp,
+        int16_t *pIntSetVolume, int32_t *pIntPrevVolume, int32_t *pIntVolumeInc,
+        float *pSetVolume, float *pPrevVolume, float *pVolumeInc) {
+    // check floating point volume to see if it is identical to the previously
+    // set volume.
+    // We do not use a tolerance here (and reject changes too small)
+    // as it may be confusing to use a different value than the one set.
+    // If the resulting volume is too small to ramp, it is a direct set of the volume.
+    if (newVolume == *pSetVolume) {
+        return false;
+    }
+    if (newVolume < 0) {
+        newVolume = 0; // should not have negative volumes
+    } else {
+        switch (fpclassify(newVolume)) {
+        case FP_SUBNORMAL:
+        case FP_NAN:
+            newVolume = 0;
+            break;
+        case FP_ZERO:
+            break; // zero volume is fine
+        case FP_INFINITE:
+            // Infinite volume could be handled consistently since
+            // floating point math saturates at infinities,
+            // but we limit volume to unity gain float.
+            // ramp = 0; break;
+            //
+            newVolume = AudioMixerBase::UNITY_GAIN_FLOAT;
+            break;
+        case FP_NORMAL:
+        default:
+            // Floating point does not have problems with overflow wrap
+            // that integer has.  However, we limit the volume to
+            // unity gain here.
+            // TODO: Revisit the volume limitation and perhaps parameterize.
+            if (newVolume > AudioMixerBase::UNITY_GAIN_FLOAT) {
+                newVolume = AudioMixerBase::UNITY_GAIN_FLOAT;
+            }
+            break;
+        }
+    }
+
+    // set floating point volume ramp
+    if (ramp != 0) {
+        // when the ramp completes, *pPrevVolume is set to *pSetVolume, so there
+        // is no computational mismatch; hence equality is checked here.
+        ALOGD_IF(*pPrevVolume != *pSetVolume, "previous float ramp hasn't finished,"
+                " prev:%f  set_to:%f", *pPrevVolume, *pSetVolume);
+        const float inc = (newVolume - *pPrevVolume) / ramp; // could be inf, nan, subnormal
+        // could be inf, cannot be nan, subnormal
+        const float maxv = std::max(newVolume, *pPrevVolume);
+
+        if (isnormal(inc) // inc must be a normal number (no subnormals, infinite, nan)
+                && maxv + inc != maxv) { // inc must make forward progress
+            *pVolumeInc = inc;
+            // ramp is set now.
+            // Note: if newVolume is 0, then near the end of the ramp,
+            // it may be possible that the ramped volume may be subnormal or
+            // temporarily negative by a small amount or subnormal due to floating
+            // point inaccuracies.
+        } else {
+            ramp = 0; // ramp not allowed
+        }
+    }
+
+    // compute and check integer volume, no need to check negative values
+    // The integer volume is limited to "unity_gain" to avoid wrapping and other
+    // audio artifacts, so it never reaches the range limit of U4.28.
+    // We safely use signed 16 and 32 bit integers here.
+    const float scaledVolume = newVolume * AudioMixerBase::UNITY_GAIN_INT; // not neg, subnormal, nan
+    const int32_t intVolume = (scaledVolume >= (float)AudioMixerBase::UNITY_GAIN_INT) ?
+            AudioMixerBase::UNITY_GAIN_INT : (int32_t)scaledVolume;
+
+    // set integer volume ramp
+    if (ramp != 0) {
+        // integer volume is U4.12 (to use 16 bit multiplies), but ramping uses U4.28.
+        // when the ramp completes, *pIntPrevVolume is set to *pIntSetVolume << 16, so there
+        // is no computational mismatch; hence equality is checked here.
+        ALOGD_IF(*pIntPrevVolume != *pIntSetVolume << 16, "previous int ramp hasn't finished,"
+                " prev:%d  set_to:%d", *pIntPrevVolume, *pIntSetVolume << 16);
+        const int32_t inc = ((intVolume << 16) - *pIntPrevVolume) / ramp;
+
+        if (inc != 0) { // inc must make forward progress
+            *pIntVolumeInc = inc;
+        } else {
+            ramp = 0; // ramp not allowed
+        }
+    }
+
+    // if no ramp, or ramp not allowed, then clear float and integer increments
+    if (ramp == 0) {
+        *pVolumeInc = 0;
+        *pPrevVolume = newVolume;
+        *pIntVolumeInc = 0;
+        *pIntPrevVolume = intVolume << 16;
+    }
+    *pSetVolume = newVolume;
+    *pIntSetVolume = intVolume;
+    return true;
+}
+
+void AudioMixerBase::setParameter(int name, int target, int param, void *value)
+{
+    LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
+    const std::shared_ptr<TrackBase> &track = mTracks[name];
+
+    int valueInt = static_cast<int>(reinterpret_cast<uintptr_t>(value));
+    int32_t *valueBuf = reinterpret_cast<int32_t*>(value);
+
+    switch (target) {
+
+    case TRACK:
+        switch (param) {
+        case CHANNEL_MASK: {
+            const audio_channel_mask_t trackChannelMask =
+                static_cast<audio_channel_mask_t>(valueInt);
+            if (setChannelMasks(name, trackChannelMask, track->mMixerChannelMask)) {
+                ALOGV("setParameter(TRACK, CHANNEL_MASK, %x)", trackChannelMask);
+                invalidate();
+            }
+            } break;
+        case MAIN_BUFFER:
+            if (track->mainBuffer != valueBuf) {
+                track->mainBuffer = valueBuf;
+                ALOGV("setParameter(TRACK, MAIN_BUFFER, %p)", valueBuf);
+                invalidate();
+            }
+            break;
+        case AUX_BUFFER:
+            if (track->auxBuffer != valueBuf) {
+                track->auxBuffer = valueBuf;
+                ALOGV("setParameter(TRACK, AUX_BUFFER, %p)", valueBuf);
+                invalidate();
+            }
+            break;
+        case FORMAT: {
+            audio_format_t format = static_cast<audio_format_t>(valueInt);
+            if (track->mFormat != format) {
+                ALOG_ASSERT(audio_is_linear_pcm(format), "Invalid format %#x", format);
+                track->mFormat = format;
+                ALOGV("setParameter(TRACK, FORMAT, %#x)", format);
+                invalidate();
+            }
+            } break;
+        case MIXER_FORMAT: {
+            audio_format_t format = static_cast<audio_format_t>(valueInt);
+            if (track->mMixerFormat != format) {
+                track->mMixerFormat = format;
+                ALOGV("setParameter(TRACK, MIXER_FORMAT, %#x)", format);
+            }
+            } break;
+        case MIXER_CHANNEL_MASK: {
+            const audio_channel_mask_t mixerChannelMask =
+                    static_cast<audio_channel_mask_t>(valueInt);
+            if (setChannelMasks(name, track->channelMask, mixerChannelMask)) {
+                ALOGV("setParameter(TRACK, MIXER_CHANNEL_MASK, %#x)", mixerChannelMask);
+                invalidate();
+            }
+            } break;
+        default:
+            LOG_ALWAYS_FATAL("setParameter track: bad param %d", param);
+        }
+        break;
+
+    case RESAMPLE:
+        switch (param) {
+        case SAMPLE_RATE:
+            ALOG_ASSERT(valueInt > 0, "bad sample rate %d", valueInt);
+            if (track->setResampler(uint32_t(valueInt), mSampleRate)) {
+                ALOGV("setParameter(RESAMPLE, SAMPLE_RATE, %u)",
+                        uint32_t(valueInt));
+                invalidate();
+            }
+            break;
+        case RESET:
+            track->resetResampler();
+            invalidate();
+            break;
+        case REMOVE:
+            track->mResampler.reset(nullptr);
+            track->sampleRate = mSampleRate;
+            invalidate();
+            break;
+        default:
+            LOG_ALWAYS_FATAL("setParameter resample: bad param %d", param);
+        }
+        break;
+
+    case RAMP_VOLUME:
+    case VOLUME:
+        switch (param) {
+        case AUXLEVEL:
+            if (setVolumeRampVariables(*reinterpret_cast<float*>(value),
+                    target == RAMP_VOLUME ? mFrameCount : 0,
+                    &track->auxLevel, &track->prevAuxLevel, &track->auxInc,
+                    &track->mAuxLevel, &track->mPrevAuxLevel, &track->mAuxInc)) {
+                ALOGV("setParameter(%s, AUXLEVEL: %04x)",
+                        target == VOLUME ? "VOLUME" : "RAMP_VOLUME", track->auxLevel);
+                invalidate();
+            }
+            break;
+        default:
+            if ((unsigned)param >= VOLUME0 && (unsigned)param < VOLUME0 + MAX_NUM_VOLUMES) {
+                if (setVolumeRampVariables(*reinterpret_cast<float*>(value),
+                        target == RAMP_VOLUME ? mFrameCount : 0,
+                        &track->volume[param - VOLUME0],
+                        &track->prevVolume[param - VOLUME0],
+                        &track->volumeInc[param - VOLUME0],
+                        &track->mVolume[param - VOLUME0],
+                        &track->mPrevVolume[param - VOLUME0],
+                        &track->mVolumeInc[param - VOLUME0])) {
+                    ALOGV("setParameter(%s, VOLUME%d: %04x)",
+                            target == VOLUME ? "VOLUME" : "RAMP_VOLUME", param - VOLUME0,
+                                    track->volume[param - VOLUME0]);
+                    invalidate();
+                }
+            } else {
+                LOG_ALWAYS_FATAL("setParameter volume: bad param %d", param);
+            }
+        }
+        break;
+
+    default:
+        LOG_ALWAYS_FATAL("setParameter: bad target %d", target);
+    }
+}
+
+bool AudioMixerBase::TrackBase::setResampler(uint32_t trackSampleRate, uint32_t devSampleRate)
+{
+    if (trackSampleRate != devSampleRate || mResampler.get() != nullptr) {
+        if (sampleRate != trackSampleRate) {
+            sampleRate = trackSampleRate;
+            if (mResampler.get() == nullptr) {
+                ALOGV("Creating resampler from track %d Hz to device %d Hz",
+                        trackSampleRate, devSampleRate);
+                AudioResampler::src_quality quality;
+                // force lowest quality level resampler if use case isn't music or video
+                // FIXME this is flawed for dynamic sample rates, as we choose the resampler
+                // quality level based on the initial ratio, but that could change later.
+                // Should have a way to distinguish tracks with static ratios vs. dynamic ratios.
+                if (isMusicRate(trackSampleRate)) {
+                    quality = AudioResampler::DEFAULT_QUALITY;
+                } else {
+                    quality = AudioResampler::DYN_LOW_QUALITY;
+                }
+
+                // TODO: Remove MONO_HACK. Resampler sees #channels after the downmixer
+                // but if none exists, it is the channel count (1 for mono).
+                const int resamplerChannelCount = getOutputChannelCount();
+                ALOGVV("Creating resampler:"
+                        " format(%#x) channels(%d) devSampleRate(%u) quality(%d)\n",
+                        mMixerInFormat, resamplerChannelCount, devSampleRate, quality);
+                mResampler.reset(AudioResampler::create(
+                        mMixerInFormat,
+                        resamplerChannelCount,
+                        devSampleRate, quality));
+            }
+            return true;
+        }
+    }
+    return false;
+}
+
+/* Checks to see if the volume ramp has completed and clears the increment
+ * variables appropriately.
+ *
+ * FIXME: There is code to handle int/float ramp variable switchover should it not
+ * complete within a mixer buffer processing call, but it is preferred to avoid switchover
+ * due to precision issues.  The switchover code is included for legacy code purposes
+ * and can be removed once the integer volume is removed.
+ *
+ * It is not sufficient to clear only the volumeInc integer variable because
+ * if one channel requires ramping, all channels are ramped.
+ *
+ * There is a bit of duplicated code here, but it keeps backward compatibility.
+ */
+void AudioMixerBase::TrackBase::adjustVolumeRamp(bool aux, bool useFloat)
+{
+    if (useFloat) {
+        for (uint32_t i = 0; i < MAX_NUM_VOLUMES; i++) {
+            if ((mVolumeInc[i] > 0 && mPrevVolume[i] + mVolumeInc[i] >= mVolume[i]) ||
+                     (mVolumeInc[i] < 0 && mPrevVolume[i] + mVolumeInc[i] <= mVolume[i])) {
+                volumeInc[i] = 0;
+                prevVolume[i] = volume[i] << 16;
+                mVolumeInc[i] = 0.;
+                mPrevVolume[i] = mVolume[i];
+            } else {
+                //ALOGV("ramp: %f %f %f", mVolume[i], mPrevVolume[i], mVolumeInc[i]);
+                prevVolume[i] = u4_28_from_float(mPrevVolume[i]);
+            }
+        }
+    } else {
+        for (uint32_t i = 0; i < MAX_NUM_VOLUMES; i++) {
+            if (((volumeInc[i]>0) && (((prevVolume[i]+volumeInc[i])>>16) >= volume[i])) ||
+                    ((volumeInc[i]<0) && (((prevVolume[i]+volumeInc[i])>>16) <= volume[i]))) {
+                volumeInc[i] = 0;
+                prevVolume[i] = volume[i] << 16;
+                mVolumeInc[i] = 0.;
+                mPrevVolume[i] = mVolume[i];
+            } else {
+                //ALOGV("ramp: %d %d %d", volume[i] << 16, prevVolume[i], volumeInc[i]);
+                mPrevVolume[i]  = float_from_u4_28(prevVolume[i]);
+            }
+        }
+    }
+
+    if (aux) {
+#ifdef FLOAT_AUX
+        if (useFloat) {
+            if ((mAuxInc > 0.f && mPrevAuxLevel + mAuxInc >= mAuxLevel) ||
+                    (mAuxInc < 0.f && mPrevAuxLevel + mAuxInc <= mAuxLevel)) {
+                auxInc = 0;
+                prevAuxLevel = auxLevel << 16;
+                mAuxInc = 0.f;
+                mPrevAuxLevel = mAuxLevel;
+            }
+        } else
+#endif
+        if ((auxInc > 0 && ((prevAuxLevel + auxInc) >> 16) >= auxLevel) ||
+                (auxInc < 0 && ((prevAuxLevel + auxInc) >> 16) <= auxLevel)) {
+            auxInc = 0;
+            prevAuxLevel = auxLevel << 16;
+            mAuxInc = 0.f;
+            mPrevAuxLevel = mAuxLevel;
+        }
+    }
+}
+
+void AudioMixerBase::TrackBase::recreateResampler(uint32_t devSampleRate)
+{
+    if (mResampler.get() != nullptr) {
+        const uint32_t resetToSampleRate = sampleRate;
+        mResampler.reset(nullptr);
+        sampleRate = devSampleRate; // without resampler, track rate is device sample rate.
+        // recreate the resampler with updated format, channels, saved sampleRate.
+        setResampler(resetToSampleRate /*trackSampleRate*/, devSampleRate);
+    }
+}
+
+size_t AudioMixerBase::getUnreleasedFrames(int name) const
+{
+    const auto it = mTracks.find(name);
+    if (it != mTracks.end()) {
+        return it->second->getUnreleasedFrames();
+    }
+    return 0;
+}
+
+std::string AudioMixerBase::trackNames() const
+{
+    std::stringstream ss;
+    for (const auto &pair : mTracks) {
+        ss << pair.first << " ";
+    }
+    return ss.str();
+}
+
+void AudioMixerBase::process__validate()
+{
+    // TODO: fix all16BitsStereNoResample logic to
+    // either properly handle muted tracks (it should ignore them)
+    // or remove altogether as an obsolete optimization.
+    bool all16BitsStereoNoResample = true;
+    bool resampling = false;
+    bool volumeRamp = false;
+
+    mEnabled.clear();
+    mGroups.clear();
+    for (const auto &pair : mTracks) {
+        const int name = pair.first;
+        const std::shared_ptr<TrackBase> &t = pair.second;
+        if (!t->enabled) continue;
+
+        mEnabled.emplace_back(name);  // we add to mEnabled in order of name.
+        mGroups[t->mainBuffer].emplace_back(name); // mGroups also in order of name.
+
+        uint32_t n = 0;
+        // FIXME can overflow (mask is only 3 bits)
+        n |= NEEDS_CHANNEL_1 + t->channelCount - 1;
+        if (t->doesResample()) {
+            n |= NEEDS_RESAMPLE;
+        }
+        if (t->auxLevel != 0 && t->auxBuffer != NULL) {
+            n |= NEEDS_AUX;
+        }
+
+        if (t->volumeInc[0]|t->volumeInc[1]) {
+            volumeRamp = true;
+        } else if (!t->doesResample() && t->volumeRL == 0) {
+            n |= NEEDS_MUTE;
+        }
+        t->needs = n;
+
+        if (n & NEEDS_MUTE) {
+            t->hook = &TrackBase::track__nop;
+        } else {
+            if (n & NEEDS_AUX) {
+                all16BitsStereoNoResample = false;
+            }
+            if (n & NEEDS_RESAMPLE) {
+                all16BitsStereoNoResample = false;
+                resampling = true;
+                t->hook = TrackBase::getTrackHook(TRACKTYPE_RESAMPLE, t->mMixerChannelCount,
+                        t->mMixerInFormat, t->mMixerFormat);
+                ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2,
+                        "Track %d needs downmix + resample", name);
+            } else {
+                if ((n & NEEDS_CHANNEL_COUNT__MASK) == NEEDS_CHANNEL_1){
+                    t->hook = TrackBase::getTrackHook(
+                            (t->mMixerChannelMask == AUDIO_CHANNEL_OUT_STEREO  // TODO: MONO_HACK
+                                    && t->channelMask == AUDIO_CHANNEL_OUT_MONO)
+                                ? TRACKTYPE_NORESAMPLEMONO : TRACKTYPE_NORESAMPLE,
+                            t->mMixerChannelCount,
+                            t->mMixerInFormat, t->mMixerFormat);
+                    all16BitsStereoNoResample = false;
+                }
+                if ((n & NEEDS_CHANNEL_COUNT__MASK) >= NEEDS_CHANNEL_2){
+                    t->hook = TrackBase::getTrackHook(TRACKTYPE_NORESAMPLE, t->mMixerChannelCount,
+                            t->mMixerInFormat, t->mMixerFormat);
+                    ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2,
+                            "Track %d needs downmix", name);
+                }
+            }
+        }
+    }
+
+    // select the processing hooks
+    mHook = &AudioMixerBase::process__nop;
+    if (mEnabled.size() > 0) {
+        if (resampling) {
+            if (mOutputTemp.get() == nullptr) {
+                mOutputTemp.reset(new int32_t[MAX_NUM_CHANNELS * mFrameCount]);
+            }
+            if (mResampleTemp.get() == nullptr) {
+                mResampleTemp.reset(new int32_t[MAX_NUM_CHANNELS * mFrameCount]);
+            }
+            mHook = &AudioMixerBase::process__genericResampling;
+        } else {
+            // we keep temp arrays around.
+            mHook = &AudioMixerBase::process__genericNoResampling;
+            if (all16BitsStereoNoResample && !volumeRamp) {
+                if (mEnabled.size() == 1) {
+                    const std::shared_ptr<TrackBase> &t = mTracks[mEnabled[0]];
+                    if ((t->needs & NEEDS_MUTE) == 0) {
+                        // The check prevents a muted track from acquiring a process hook.
+                        //
+                        // This is dangerous if the track is MONO as that requires
+                        // special case handling due to implicit channel duplication.
+                        // Stereo or Multichannel should actually be fine here.
+                        mHook = getProcessHook(PROCESSTYPE_NORESAMPLEONETRACK,
+                                t->mMixerChannelCount, t->mMixerInFormat, t->mMixerFormat);
+                    }
+                }
+            }
+        }
+    }
+
+    ALOGV("mixer configuration change: %zu "
+        "all16BitsStereoNoResample=%d, resampling=%d, volumeRamp=%d",
+        mEnabled.size(), all16BitsStereoNoResample, resampling, volumeRamp);
+
+    process();
+
+    // Now that the volume ramp has been done, set optimal state and
+    // track hooks for subsequent mixer process
+    if (mEnabled.size() > 0) {
+        bool allMuted = true;
+
+        for (const int name : mEnabled) {
+            const std::shared_ptr<TrackBase> &t = mTracks[name];
+            if (!t->doesResample() && t->volumeRL == 0) {
+                t->needs |= NEEDS_MUTE;
+                t->hook = &TrackBase::track__nop;
+            } else {
+                allMuted = false;
+            }
+        }
+        if (allMuted) {
+            mHook = &AudioMixerBase::process__nop;
+        } else if (all16BitsStereoNoResample) {
+            if (mEnabled.size() == 1) {
+                //const int i = 31 - __builtin_clz(enabledTracks);
+                const std::shared_ptr<TrackBase> &t = mTracks[mEnabled[0]];
+                // Muted single tracks handled by allMuted above.
+                mHook = getProcessHook(PROCESSTYPE_NORESAMPLEONETRACK,
+                        t->mMixerChannelCount, t->mMixerInFormat, t->mMixerFormat);
+            }
+        }
+    }
+}
+
+void AudioMixerBase::TrackBase::track__genericResample(
+        int32_t* out, size_t outFrameCount, int32_t* temp, int32_t* aux)
+{
+    ALOGVV("track__genericResample\n");
+    mResampler->setSampleRate(sampleRate);
+
+    // ramp gain - resample to temp buffer and scale/mix in 2nd step
+    if (aux != NULL) {
+        // always resample with unity gain when sending to auxiliary buffer to be able
+        // to apply send level after resampling
+        mResampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
+        memset(temp, 0, outFrameCount * mMixerChannelCount * sizeof(int32_t));
+        mResampler->resample(temp, outFrameCount, bufferProvider);
+        if (CC_UNLIKELY(volumeInc[0]|volumeInc[1]|auxInc)) {
+            volumeRampStereo(out, outFrameCount, temp, aux);
+        } else {
+            volumeStereo(out, outFrameCount, temp, aux);
+        }
+    } else {
+        if (CC_UNLIKELY(volumeInc[0]|volumeInc[1])) {
+            mResampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
+            memset(temp, 0, outFrameCount * MAX_NUM_CHANNELS * sizeof(int32_t));
+            mResampler->resample(temp, outFrameCount, bufferProvider);
+            volumeRampStereo(out, outFrameCount, temp, aux);
+        }
+
+        // constant gain
+        else {
+            mResampler->setVolume(mVolume[0], mVolume[1]);
+            mResampler->resample(out, outFrameCount, bufferProvider);
+        }
+    }
+}
+
+void AudioMixerBase::TrackBase::track__nop(int32_t* out __unused,
+        size_t outFrameCount __unused, int32_t* temp __unused, int32_t* aux __unused)
+{
+}
+
+void AudioMixerBase::TrackBase::volumeRampStereo(
+        int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
+{
+    int32_t vl = prevVolume[0];
+    int32_t vr = prevVolume[1];
+    const int32_t vlInc = volumeInc[0];
+    const int32_t vrInc = volumeInc[1];
+
+    //ALOGD("[0] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
+    //        t, vlInc/65536.0f, vl/65536.0f, volume[0],
+    //       (vl + vlInc*frameCount)/65536.0f, frameCount);
+
+    // ramp volume
+    if (CC_UNLIKELY(aux != NULL)) {
+        int32_t va = prevAuxLevel;
+        const int32_t vaInc = auxInc;
+        int32_t l;
+        int32_t r;
+
+        do {
+            l = (*temp++ >> 12);
+            r = (*temp++ >> 12);
+            *out++ += (vl >> 16) * l;
+            *out++ += (vr >> 16) * r;
+            *aux++ += (va >> 17) * (l + r);
+            vl += vlInc;
+            vr += vrInc;
+            va += vaInc;
+        } while (--frameCount);
+        prevAuxLevel = va;
+    } else {
+        do {
+            *out++ += (vl >> 16) * (*temp++ >> 12);
+            *out++ += (vr >> 16) * (*temp++ >> 12);
+            vl += vlInc;
+            vr += vrInc;
+        } while (--frameCount);
+    }
+    prevVolume[0] = vl;
+    prevVolume[1] = vr;
+    adjustVolumeRamp(aux != NULL);
+}
+
+void AudioMixerBase::TrackBase::volumeStereo(
+        int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
+{
+    const int16_t vl = volume[0];
+    const int16_t vr = volume[1];
+
+    if (CC_UNLIKELY(aux != NULL)) {
+        const int16_t va = auxLevel;
+        do {
+            int16_t l = (int16_t)(*temp++ >> 12);
+            int16_t r = (int16_t)(*temp++ >> 12);
+            out[0] = mulAdd(l, vl, out[0]);
+            int16_t a = (int16_t)(((int32_t)l + r) >> 1);
+            out[1] = mulAdd(r, vr, out[1]);
+            out += 2;
+            aux[0] = mulAdd(a, va, aux[0]);
+            aux++;
+        } while (--frameCount);
+    } else {
+        do {
+            int16_t l = (int16_t)(*temp++ >> 12);
+            int16_t r = (int16_t)(*temp++ >> 12);
+            out[0] = mulAdd(l, vl, out[0]);
+            out[1] = mulAdd(r, vr, out[1]);
+            out += 2;
+        } while (--frameCount);
+    }
+}
+
+void AudioMixerBase::TrackBase::track__16BitsStereo(
+        int32_t* out, size_t frameCount, int32_t* temp __unused, int32_t* aux)
+{
+    ALOGVV("track__16BitsStereo\n");
+    const int16_t *in = static_cast<const int16_t *>(mIn);
+
+    if (CC_UNLIKELY(aux != NULL)) {
+        int32_t l;
+        int32_t r;
+        // ramp gain
+        if (CC_UNLIKELY(volumeInc[0]|volumeInc[1]|auxInc)) {
+            int32_t vl = prevVolume[0];
+            int32_t vr = prevVolume[1];
+            int32_t va = prevAuxLevel;
+            const int32_t vlInc = volumeInc[0];
+            const int32_t vrInc = volumeInc[1];
+            const int32_t vaInc = auxInc;
+            // ALOGD("[1] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
+            //        t, vlInc/65536.0f, vl/65536.0f, volume[0],
+            //        (vl + vlInc*frameCount)/65536.0f, frameCount);
+
+            do {
+                l = (int32_t)*in++;
+                r = (int32_t)*in++;
+                *out++ += (vl >> 16) * l;
+                *out++ += (vr >> 16) * r;
+                *aux++ += (va >> 17) * (l + r);
+                vl += vlInc;
+                vr += vrInc;
+                va += vaInc;
+            } while (--frameCount);
+
+            prevVolume[0] = vl;
+            prevVolume[1] = vr;
+            prevAuxLevel = va;
+            adjustVolumeRamp(true);
+        }
+
+        // constant gain
+        else {
+            const uint32_t vrl = volumeRL;
+            const int16_t va = (int16_t)auxLevel;
+            do {
+                uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
+                int16_t a = (int16_t)(((int32_t)in[0] + in[1]) >> 1);
+                in += 2;
+                out[0] = mulAddRL(1, rl, vrl, out[0]);
+                out[1] = mulAddRL(0, rl, vrl, out[1]);
+                out += 2;
+                aux[0] = mulAdd(a, va, aux[0]);
+                aux++;
+            } while (--frameCount);
+        }
+    } else {
+        // ramp gain
+        if (CC_UNLIKELY(volumeInc[0]|volumeInc[1])) {
+            int32_t vl = prevVolume[0];
+            int32_t vr = prevVolume[1];
+            const int32_t vlInc = volumeInc[0];
+            const int32_t vrInc = volumeInc[1];
+
+            // ALOGD("[1] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
+            //        t, vlInc/65536.0f, vl/65536.0f, volume[0],
+            //        (vl + vlInc*frameCount)/65536.0f, frameCount);
+
+            do {
+                *out++ += (vl >> 16) * (int32_t) *in++;
+                *out++ += (vr >> 16) * (int32_t) *in++;
+                vl += vlInc;
+                vr += vrInc;
+            } while (--frameCount);
+
+            prevVolume[0] = vl;
+            prevVolume[1] = vr;
+            adjustVolumeRamp(false);
+        }
+
+        // constant gain
+        else {
+            const uint32_t vrl = volumeRL;
+            do {
+                uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
+                in += 2;
+                out[0] = mulAddRL(1, rl, vrl, out[0]);
+                out[1] = mulAddRL(0, rl, vrl, out[1]);
+                out += 2;
+            } while (--frameCount);
+        }
+    }
+    mIn = in;
+}
+
+void AudioMixerBase::TrackBase::track__16BitsMono(
+        int32_t* out, size_t frameCount, int32_t* temp __unused, int32_t* aux)
+{
+    ALOGVV("track__16BitsMono\n");
+    const int16_t *in = static_cast<int16_t const *>(mIn);
+
+    if (CC_UNLIKELY(aux != NULL)) {
+        // ramp gain
+        if (CC_UNLIKELY(volumeInc[0]|volumeInc[1]|auxInc)) {
+            int32_t vl = prevVolume[0];
+            int32_t vr = prevVolume[1];
+            int32_t va = prevAuxLevel;
+            const int32_t vlInc = volumeInc[0];
+            const int32_t vrInc = volumeInc[1];
+            const int32_t vaInc = auxInc;
+
+            // ALOGD("[2] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
+            //         t, vlInc/65536.0f, vl/65536.0f, volume[0],
+            //         (vl + vlInc*frameCount)/65536.0f, frameCount);
+
+            do {
+                int32_t l = *in++;
+                *out++ += (vl >> 16) * l;
+                *out++ += (vr >> 16) * l;
+                *aux++ += (va >> 16) * l;
+                vl += vlInc;
+                vr += vrInc;
+                va += vaInc;
+            } while (--frameCount);
+
+            prevVolume[0] = vl;
+            prevVolume[1] = vr;
+            prevAuxLevel = va;
+            adjustVolumeRamp(true);
+        }
+        // constant gain
+        else {
+            const int16_t vl = volume[0];
+            const int16_t vr = volume[1];
+            const int16_t va = (int16_t)auxLevel;
+            do {
+                int16_t l = *in++;
+                out[0] = mulAdd(l, vl, out[0]);
+                out[1] = mulAdd(l, vr, out[1]);
+                out += 2;
+                aux[0] = mulAdd(l, va, aux[0]);
+                aux++;
+            } while (--frameCount);
+        }
+    } else {
+        // ramp gain
+        if (CC_UNLIKELY(volumeInc[0]|volumeInc[1])) {
+            int32_t vl = prevVolume[0];
+            int32_t vr = prevVolume[1];
+            const int32_t vlInc = volumeInc[0];
+            const int32_t vrInc = volumeInc[1];
+
+            // ALOGD("[2] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
+            //         t, vlInc/65536.0f, vl/65536.0f, volume[0],
+            //         (vl + vlInc*frameCount)/65536.0f, frameCount);
+
+            do {
+                int32_t l = *in++;
+                *out++ += (vl >> 16) * l;
+                *out++ += (vr >> 16) * l;
+                vl += vlInc;
+                vr += vrInc;
+            } while (--frameCount);
+
+            prevVolume[0] = vl;
+            prevVolume[1] = vr;
+            adjustVolumeRamp(false);
+        }
+        // constant gain
+        else {
+            const int16_t vl = volume[0];
+            const int16_t vr = volume[1];
+            do {
+                int16_t l = *in++;
+                out[0] = mulAdd(l, vl, out[0]);
+                out[1] = mulAdd(l, vr, out[1]);
+                out += 2;
+            } while (--frameCount);
+        }
+    }
+    mIn = in;
+}
+
+// no-op case
+void AudioMixerBase::process__nop()
+{
+    ALOGVV("process__nop\n");
+
+    for (const auto &pair : mGroups) {
+        // process by group of tracks with same output buffer to
+        // avoid multiple memset() on same buffer
+        const auto &group = pair.second;
+
+        const std::shared_ptr<TrackBase> &t = mTracks[group[0]];
+        memset(t->mainBuffer, 0,
+                mFrameCount * audio_bytes_per_frame(t->getMixerChannelCount(), t->mMixerFormat));
+
+        // now consume data
+        for (const int name : group) {
+            const std::shared_ptr<TrackBase> &t = mTracks[name];
+            size_t outFrames = mFrameCount;
+            while (outFrames) {
+                t->buffer.frameCount = outFrames;
+                t->bufferProvider->getNextBuffer(&t->buffer);
+                if (t->buffer.raw == NULL) break;
+                outFrames -= t->buffer.frameCount;
+                t->bufferProvider->releaseBuffer(&t->buffer);
+            }
+        }
+    }
+}
+
+// generic code without resampling
+void AudioMixerBase::process__genericNoResampling()
+{
+    ALOGVV("process__genericNoResampling\n");
+    int32_t outTemp[BLOCKSIZE * MAX_NUM_CHANNELS] __attribute__((aligned(32)));
+
+    for (const auto &pair : mGroups) {
+        // process by group of tracks with same output main buffer to
+        // avoid multiple memset() on same buffer
+        const auto &group = pair.second;
+
+        // acquire buffer
+        for (const int name : group) {
+            const std::shared_ptr<TrackBase> &t = mTracks[name];
+            t->buffer.frameCount = mFrameCount;
+            t->bufferProvider->getNextBuffer(&t->buffer);
+            t->frameCount = t->buffer.frameCount;
+            t->mIn = t->buffer.raw;
+        }
+
+        int32_t *out = (int *)pair.first;
+        size_t numFrames = 0;
+        do {
+            const size_t frameCount = std::min((size_t)BLOCKSIZE, mFrameCount - numFrames);
+            memset(outTemp, 0, sizeof(outTemp));
+            for (const int name : group) {
+                const std::shared_ptr<TrackBase> &t = mTracks[name];
+                int32_t *aux = NULL;
+                if (CC_UNLIKELY(t->needs & NEEDS_AUX)) {
+                    aux = t->auxBuffer + numFrames;
+                }
+                for (int outFrames = frameCount; outFrames > 0; ) {
+                    // t->in == nullptr can happen if the track was flushed just after having
+                    // been enabled for mixing.
+                    if (t->mIn == nullptr) {
+                        break;
+                    }
+                    size_t inFrames = (t->frameCount > outFrames)?outFrames:t->frameCount;
+                    if (inFrames > 0) {
+                        (t.get()->*t->hook)(
+                                outTemp + (frameCount - outFrames) * t->mMixerChannelCount,
+                                inFrames, mResampleTemp.get() /* naked ptr */, aux);
+                        t->frameCount -= inFrames;
+                        outFrames -= inFrames;
+                        if (CC_UNLIKELY(aux != NULL)) {
+                            aux += inFrames;
+                        }
+                    }
+                    if (t->frameCount == 0 && outFrames) {
+                        t->bufferProvider->releaseBuffer(&t->buffer);
+                        t->buffer.frameCount = (mFrameCount - numFrames) -
+                                (frameCount - outFrames);
+                        t->bufferProvider->getNextBuffer(&t->buffer);
+                        t->mIn = t->buffer.raw;
+                        if (t->mIn == nullptr) {
+                            break;
+                        }
+                        t->frameCount = t->buffer.frameCount;
+                    }
+                }
+            }
+
+            const std::shared_ptr<TrackBase> &t1 = mTracks[group[0]];
+            convertMixerFormat(out, t1->mMixerFormat, outTemp, t1->mMixerInFormat,
+                    frameCount * t1->mMixerChannelCount);
+            // TODO: fix ugly casting due to choice of out pointer type
+            out = reinterpret_cast<int32_t*>((uint8_t*)out
+                    + frameCount * t1->mMixerChannelCount
+                    * audio_bytes_per_sample(t1->mMixerFormat));
+            numFrames += frameCount;
+        } while (numFrames < mFrameCount);
+
+        // release each track's buffer
+        for (const int name : group) {
+            const std::shared_ptr<TrackBase> &t = mTracks[name];
+            t->bufferProvider->releaseBuffer(&t->buffer);
+        }
+    }
+}
+
+// generic code with resampling
+void AudioMixerBase::process__genericResampling()
+{
+    ALOGVV("process__genericResampling\n");
+    int32_t * const outTemp = mOutputTemp.get(); // naked ptr
+    size_t numFrames = mFrameCount;
+
+    for (const auto &pair : mGroups) {
+        const auto &group = pair.second;
+        const std::shared_ptr<TrackBase> &t1 = mTracks[group[0]];
+
+        // clear temp buffer
+        memset(outTemp, 0, sizeof(*outTemp) * t1->mMixerChannelCount * mFrameCount);
+        for (const int name : group) {
+            const std::shared_ptr<TrackBase> &t = mTracks[name];
+            int32_t *aux = NULL;
+            if (CC_UNLIKELY(t->needs & NEEDS_AUX)) {
+                aux = t->auxBuffer;
+            }
+
+            // this is a little goofy, on the resampling case we don't
+            // acquire/release the buffers because it's done by
+            // the resampler.
+            if (t->needs & NEEDS_RESAMPLE) {
+                (t.get()->*t->hook)(outTemp, numFrames, mResampleTemp.get() /* naked ptr */, aux);
+            } else {
+
+                size_t outFrames = 0;
+
+                while (outFrames < numFrames) {
+                    t->buffer.frameCount = numFrames - outFrames;
+                    t->bufferProvider->getNextBuffer(&t->buffer);
+                    t->mIn = t->buffer.raw;
+                    // t->mIn == nullptr can happen if the track was flushed just after having
+                    // been enabled for mixing.
+                    if (t->mIn == nullptr) break;
+
+                    (t.get()->*t->hook)(
+                            outTemp + outFrames * t->mMixerChannelCount, t->buffer.frameCount,
+                            mResampleTemp.get() /* naked ptr */,
+                            aux != nullptr ? aux + outFrames : nullptr);
+                    outFrames += t->buffer.frameCount;
+
+                    t->bufferProvider->releaseBuffer(&t->buffer);
+                }
+            }
+        }
+        convertMixerFormat(t1->mainBuffer, t1->mMixerFormat,
+                outTemp, t1->mMixerInFormat, numFrames * t1->mMixerChannelCount);
+    }
+}
+
+// one track, 16 bits stereo without resampling is the most common case
+void AudioMixerBase::process__oneTrack16BitsStereoNoResampling()
+{
+    ALOGVV("process__oneTrack16BitsStereoNoResampling\n");
+    LOG_ALWAYS_FATAL_IF(mEnabled.size() != 0,
+            "%zu != 1 tracks enabled", mEnabled.size());
+    const int name = mEnabled[0];
+    const std::shared_ptr<TrackBase> &t = mTracks[name];
+
+    AudioBufferProvider::Buffer& b(t->buffer);
+
+    int32_t* out = t->mainBuffer;
+    float *fout = reinterpret_cast<float*>(out);
+    size_t numFrames = mFrameCount;
+
+    const int16_t vl = t->volume[0];
+    const int16_t vr = t->volume[1];
+    const uint32_t vrl = t->volumeRL;
+    while (numFrames) {
+        b.frameCount = numFrames;
+        t->bufferProvider->getNextBuffer(&b);
+        const int16_t *in = b.i16;
+
+        // in == NULL can happen if the track was flushed just after having
+        // been enabled for mixing.
+        if (in == NULL || (((uintptr_t)in) & 3)) {
+            if ( AUDIO_FORMAT_PCM_FLOAT == t->mMixerFormat ) {
+                 memset((char*)fout, 0, numFrames
+                         * t->mMixerChannelCount * audio_bytes_per_sample(t->mMixerFormat));
+            } else {
+                 memset((char*)out, 0, numFrames
+                         * t->mMixerChannelCount * audio_bytes_per_sample(t->mMixerFormat));
+            }
+            ALOGE_IF((((uintptr_t)in) & 3),
+                    "process__oneTrack16BitsStereoNoResampling: misaligned buffer"
+                    " %p track %d, channels %d, needs %08x, volume %08x vfl %f vfr %f",
+                    in, name, t->channelCount, t->needs, vrl, t->mVolume[0], t->mVolume[1]);
+            return;
+        }
+        size_t outFrames = b.frameCount;
+
+        switch (t->mMixerFormat) {
+        case AUDIO_FORMAT_PCM_FLOAT:
+            do {
+                uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
+                in += 2;
+                int32_t l = mulRL(1, rl, vrl);
+                int32_t r = mulRL(0, rl, vrl);
+                *fout++ = float_from_q4_27(l);
+                *fout++ = float_from_q4_27(r);
+                // Note: In case of later int16_t sink output,
+                // conversion and clamping is done by memcpy_to_i16_from_float().
+            } while (--outFrames);
+            break;
+        case AUDIO_FORMAT_PCM_16_BIT:
+            if (CC_UNLIKELY(uint32_t(vl) > UNITY_GAIN_INT || uint32_t(vr) > UNITY_GAIN_INT)) {
+                // volume is boosted, so we might need to clamp even though
+                // we process only one track.
+                do {
+                    uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
+                    in += 2;
+                    int32_t l = mulRL(1, rl, vrl) >> 12;
+                    int32_t r = mulRL(0, rl, vrl) >> 12;
+                    // clamping...
+                    l = clamp16(l);
+                    r = clamp16(r);
+                    *out++ = (r<<16) | (l & 0xFFFF);
+                } while (--outFrames);
+            } else {
+                do {
+                    uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
+                    in += 2;
+                    int32_t l = mulRL(1, rl, vrl) >> 12;
+                    int32_t r = mulRL(0, rl, vrl) >> 12;
+                    *out++ = (r<<16) | (l & 0xFFFF);
+                } while (--outFrames);
+            }
+            break;
+        default:
+            LOG_ALWAYS_FATAL("bad mixer format: %d", t->mMixerFormat);
+        }
+        numFrames -= b.frameCount;
+        t->bufferProvider->releaseBuffer(&b);
+    }
+}
+
+/* TODO: consider whether this level of optimization is necessary.
+ * Perhaps just stick with a single for loop.
+ */
+
+// Needs to derive a compile time constant (constexpr).  Could be targeted to go
+// to a MONOVOL mixtype based on MAX_NUM_VOLUMES, but that's an unnecessary complication.
+#define MIXTYPE_MONOVOL(mixtype) ((mixtype) == MIXTYPE_MULTI ? MIXTYPE_MULTI_MONOVOL : \
+        (mixtype) == MIXTYPE_MULTI_SAVEONLY ? MIXTYPE_MULTI_SAVEONLY_MONOVOL : (mixtype))
+
+/* MIXTYPE     (see AudioMixerOps.h MIXTYPE_* enumeration)
+ * TO: int32_t (Q4.27) or float
+ * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
+ * TA: int32_t (Q4.27) or float
+ */
+template <int MIXTYPE,
+        typename TO, typename TI, typename TV, typename TA, typename TAV>
+static void volumeRampMulti(uint32_t channels, TO* out, size_t frameCount,
+        const TI* in, TA* aux, TV *vol, const TV *volinc, TAV *vola, TAV volainc)
+{
+    switch (channels) {
+    case 1:
+        volumeRampMulti<MIXTYPE, 1>(out, frameCount, in, aux, vol, volinc, vola, volainc);
+        break;
+    case 2:
+        volumeRampMulti<MIXTYPE, 2>(out, frameCount, in, aux, vol, volinc, vola, volainc);
+        break;
+    case 3:
+        volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 3>(out,
+                frameCount, in, aux, vol, volinc, vola, volainc);
+        break;
+    case 4:
+        volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 4>(out,
+                frameCount, in, aux, vol, volinc, vola, volainc);
+        break;
+    case 5:
+        volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 5>(out,
+                frameCount, in, aux, vol, volinc, vola, volainc);
+        break;
+    case 6:
+        volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 6>(out,
+                frameCount, in, aux, vol, volinc, vola, volainc);
+        break;
+    case 7:
+        volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 7>(out,
+                frameCount, in, aux, vol, volinc, vola, volainc);
+        break;
+    case 8:
+        volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 8>(out,
+                frameCount, in, aux, vol, volinc, vola, volainc);
+        break;
+    }
+}
+
+/* MIXTYPE     (see AudioMixerOps.h MIXTYPE_* enumeration)
+ * TO: int32_t (Q4.27) or float
+ * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
+ * TA: int32_t (Q4.27) or float
+ */
+template <int MIXTYPE,
+        typename TO, typename TI, typename TV, typename TA, typename TAV>
+static void volumeMulti(uint32_t channels, TO* out, size_t frameCount,
+        const TI* in, TA* aux, const TV *vol, TAV vola)
+{
+    switch (channels) {
+    case 1:
+        volumeMulti<MIXTYPE, 1>(out, frameCount, in, aux, vol, vola);
+        break;
+    case 2:
+        volumeMulti<MIXTYPE, 2>(out, frameCount, in, aux, vol, vola);
+        break;
+    case 3:
+        volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 3>(out, frameCount, in, aux, vol, vola);
+        break;
+    case 4:
+        volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 4>(out, frameCount, in, aux, vol, vola);
+        break;
+    case 5:
+        volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 5>(out, frameCount, in, aux, vol, vola);
+        break;
+    case 6:
+        volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 6>(out, frameCount, in, aux, vol, vola);
+        break;
+    case 7:
+        volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 7>(out, frameCount, in, aux, vol, vola);
+        break;
+    case 8:
+        volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 8>(out, frameCount, in, aux, vol, vola);
+        break;
+    }
+}
+
+/* MIXTYPE     (see AudioMixerOps.h MIXTYPE_* enumeration)
+ * USEFLOATVOL (set to true if float volume is used)
+ * ADJUSTVOL   (set to true if volume ramp parameters needs adjustment afterwards)
+ * TO: int32_t (Q4.27) or float
+ * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
+ * TA: int32_t (Q4.27) or float
+ */
+template <int MIXTYPE, bool USEFLOATVOL, bool ADJUSTVOL,
+    typename TO, typename TI, typename TA>
+void AudioMixerBase::TrackBase::volumeMix(TO *out, size_t outFrames,
+        const TI *in, TA *aux, bool ramp)
+{
+    if (USEFLOATVOL) {
+        if (ramp) {
+            volumeRampMulti<MIXTYPE>(mMixerChannelCount, out, outFrames, in, aux,
+                    mPrevVolume, mVolumeInc,
+#ifdef FLOAT_AUX
+                    &mPrevAuxLevel, mAuxInc
+#else
+                    &prevAuxLevel, auxInc
+#endif
+                );
+            if (ADJUSTVOL) {
+                adjustVolumeRamp(aux != NULL, true);
+            }
+        } else {
+            volumeMulti<MIXTYPE>(mMixerChannelCount, out, outFrames, in, aux,
+                    mVolume,
+#ifdef FLOAT_AUX
+                    mAuxLevel
+#else
+                    auxLevel
+#endif
+            );
+        }
+    } else {
+        if (ramp) {
+            volumeRampMulti<MIXTYPE>(mMixerChannelCount, out, outFrames, in, aux,
+                    prevVolume, volumeInc, &prevAuxLevel, auxInc);
+            if (ADJUSTVOL) {
+                adjustVolumeRamp(aux != NULL);
+            }
+        } else {
+            volumeMulti<MIXTYPE>(mMixerChannelCount, out, outFrames, in, aux,
+                    volume, auxLevel);
+        }
+    }
+}
+
+/* This process hook is called when there is a single track without
+ * aux buffer, volume ramp, or resampling.
+ * TODO: Update the hook selection: this can properly handle aux and ramp.
+ *
+ * MIXTYPE     (see AudioMixerOps.h MIXTYPE_* enumeration)
+ * TO: int32_t (Q4.27) or float
+ * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
+ * TA: int32_t (Q4.27)
+ */
+template <int MIXTYPE, typename TO, typename TI, typename TA>
+void AudioMixerBase::process__noResampleOneTrack()
+{
+    ALOGVV("process__noResampleOneTrack\n");
+    LOG_ALWAYS_FATAL_IF(mEnabled.size() != 1,
+            "%zu != 1 tracks enabled", mEnabled.size());
+    const std::shared_ptr<TrackBase> &t = mTracks[mEnabled[0]];
+    const uint32_t channels = t->mMixerChannelCount;
+    TO* out = reinterpret_cast<TO*>(t->mainBuffer);
+    TA* aux = reinterpret_cast<TA*>(t->auxBuffer);
+    const bool ramp = t->needsRamp();
+
+    for (size_t numFrames = mFrameCount; numFrames > 0; ) {
+        AudioBufferProvider::Buffer& b(t->buffer);
+        // get input buffer
+        b.frameCount = numFrames;
+        t->bufferProvider->getNextBuffer(&b);
+        const TI *in = reinterpret_cast<TI*>(b.raw);
+
+        // in == NULL can happen if the track was flushed just after having
+        // been enabled for mixing.
+        if (in == NULL || (((uintptr_t)in) & 3)) {
+            memset(out, 0, numFrames
+                    * channels * audio_bytes_per_sample(t->mMixerFormat));
+            ALOGE_IF((((uintptr_t)in) & 3), "process__noResampleOneTrack: bus error: "
+                    "buffer %p track %p, channels %d, needs %#x",
+                    in, &t, t->channelCount, t->needs);
+            return;
+        }
+
+        const size_t outFrames = b.frameCount;
+        t->volumeMix<MIXTYPE, is_same<TI, float>::value /* USEFLOATVOL */, false /* ADJUSTVOL */> (
+                out, outFrames, in, aux, ramp);
+
+        out += outFrames * channels;
+        if (aux != NULL) {
+            aux += outFrames;
+        }
+        numFrames -= b.frameCount;
+
+        // release buffer
+        t->bufferProvider->releaseBuffer(&b);
+    }
+    if (ramp) {
+        t->adjustVolumeRamp(aux != NULL, is_same<TI, float>::value);
+    }
+}
+
+/* This track hook is called to do resampling then mixing,
+ * pulling from the track's upstream AudioBufferProvider.
+ *
+ * MIXTYPE     (see AudioMixerOps.h MIXTYPE_* enumeration)
+ * TO: int32_t (Q4.27) or float
+ * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
+ * TA: int32_t (Q4.27) or float
+ */
+template <int MIXTYPE, typename TO, typename TI, typename TA>
+void AudioMixerBase::TrackBase::track__Resample(TO* out, size_t outFrameCount, TO* temp, TA* aux)
+{
+    ALOGVV("track__Resample\n");
+    mResampler->setSampleRate(sampleRate);
+    const bool ramp = needsRamp();
+    if (ramp || aux != NULL) {
+        // if ramp:        resample with unity gain to temp buffer and scale/mix in 2nd step.
+        // if aux != NULL: resample with unity gain to temp buffer then apply send level.
+
+        mResampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
+        memset(temp, 0, outFrameCount * mMixerChannelCount * sizeof(TO));
+        mResampler->resample((int32_t*)temp, outFrameCount, bufferProvider);
+
+        volumeMix<MIXTYPE, is_same<TI, float>::value /* USEFLOATVOL */, true /* ADJUSTVOL */>(
+                out, outFrameCount, temp, aux, ramp);
+
+    } else { // constant volume gain
+        mResampler->setVolume(mVolume[0], mVolume[1]);
+        mResampler->resample((int32_t*)out, outFrameCount, bufferProvider);
+    }
+}
+
+/* This track hook is called to mix a track, when no resampling is required.
+ * The input buffer should be present in in.
+ *
+ * MIXTYPE     (see AudioMixerOps.h MIXTYPE_* enumeration)
+ * TO: int32_t (Q4.27) or float
+ * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
+ * TA: int32_t (Q4.27) or float
+ */
+template <int MIXTYPE, typename TO, typename TI, typename TA>
+void AudioMixerBase::TrackBase::track__NoResample(
+        TO* out, size_t frameCount, TO* temp __unused, TA* aux)
+{
+    ALOGVV("track__NoResample\n");
+    const TI *in = static_cast<const TI *>(mIn);
+
+    volumeMix<MIXTYPE, is_same<TI, float>::value /* USEFLOATVOL */, true /* ADJUSTVOL */>(
+            out, frameCount, in, aux, needsRamp());
+
+    // MIXTYPE_MONOEXPAND reads a single input channel and expands to NCHAN output channels.
+    // MIXTYPE_MULTI reads NCHAN input channels and places to NCHAN output channels.
+    in += (MIXTYPE == MIXTYPE_MONOEXPAND) ? frameCount : frameCount * mMixerChannelCount;
+    mIn = in;
+}
+
+/* The Mixer engine generates either int32_t (Q4_27) or float data.
+ * We use this function to convert the engine buffers
+ * to the desired mixer output format, either int16_t (Q.15) or float.
+ */
+/* static */
+void AudioMixerBase::convertMixerFormat(void *out, audio_format_t mixerOutFormat,
+        void *in, audio_format_t mixerInFormat, size_t sampleCount)
+{
+    switch (mixerInFormat) {
+    case AUDIO_FORMAT_PCM_FLOAT:
+        switch (mixerOutFormat) {
+        case AUDIO_FORMAT_PCM_FLOAT:
+            memcpy(out, in, sampleCount * sizeof(float)); // MEMCPY. TODO optimize out
+            break;
+        case AUDIO_FORMAT_PCM_16_BIT:
+            memcpy_to_i16_from_float((int16_t*)out, (float*)in, sampleCount);
+            break;
+        default:
+            LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
+            break;
+        }
+        break;
+    case AUDIO_FORMAT_PCM_16_BIT:
+        switch (mixerOutFormat) {
+        case AUDIO_FORMAT_PCM_FLOAT:
+            memcpy_to_float_from_q4_27((float*)out, (const int32_t*)in, sampleCount);
+            break;
+        case AUDIO_FORMAT_PCM_16_BIT:
+            memcpy_to_i16_from_q4_27((int16_t*)out, (const int32_t*)in, sampleCount);
+            break;
+        default:
+            LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
+            break;
+        }
+        break;
+    default:
+        LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
+        break;
+    }
+}
+
+/* Returns the proper track hook to use for mixing the track into the output buffer.
+ */
+/* static */
+AudioMixerBase::hook_t AudioMixerBase::TrackBase::getTrackHook(int trackType, uint32_t channelCount,
+        audio_format_t mixerInFormat, audio_format_t mixerOutFormat __unused)
+{
+    if (!kUseNewMixer && channelCount == FCC_2 && mixerInFormat == AUDIO_FORMAT_PCM_16_BIT) {
+        switch (trackType) {
+        case TRACKTYPE_NOP:
+            return &TrackBase::track__nop;
+        case TRACKTYPE_RESAMPLE:
+            return &TrackBase::track__genericResample;
+        case TRACKTYPE_NORESAMPLEMONO:
+            return &TrackBase::track__16BitsMono;
+        case TRACKTYPE_NORESAMPLE:
+            return &TrackBase::track__16BitsStereo;
+        default:
+            LOG_ALWAYS_FATAL("bad trackType: %d", trackType);
+            break;
+        }
+    }
+    LOG_ALWAYS_FATAL_IF(channelCount > MAX_NUM_CHANNELS);
+    switch (trackType) {
+    case TRACKTYPE_NOP:
+        return &TrackBase::track__nop;
+    case TRACKTYPE_RESAMPLE:
+        switch (mixerInFormat) {
+        case AUDIO_FORMAT_PCM_FLOAT:
+            return (AudioMixerBase::hook_t) &TrackBase::track__Resample<
+                    MIXTYPE_MULTI, float /*TO*/, float /*TI*/, TYPE_AUX>;
+        case AUDIO_FORMAT_PCM_16_BIT:
+            return (AudioMixerBase::hook_t) &TrackBase::track__Resample<
+                    MIXTYPE_MULTI, int32_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;
+        default:
+            LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
+            break;
+        }
+        break;
+    case TRACKTYPE_NORESAMPLEMONO:
+        switch (mixerInFormat) {
+        case AUDIO_FORMAT_PCM_FLOAT:
+            return (AudioMixerBase::hook_t) &TrackBase::track__NoResample<
+                            MIXTYPE_MONOEXPAND, float /*TO*/, float /*TI*/, TYPE_AUX>;
+        case AUDIO_FORMAT_PCM_16_BIT:
+            return (AudioMixerBase::hook_t) &TrackBase::track__NoResample<
+                            MIXTYPE_MONOEXPAND, int32_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;
+        default:
+            LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
+            break;
+        }
+        break;
+    case TRACKTYPE_NORESAMPLE:
+        switch (mixerInFormat) {
+        case AUDIO_FORMAT_PCM_FLOAT:
+            return (AudioMixerBase::hook_t) &TrackBase::track__NoResample<
+                    MIXTYPE_MULTI, float /*TO*/, float /*TI*/, TYPE_AUX>;
+        case AUDIO_FORMAT_PCM_16_BIT:
+            return (AudioMixerBase::hook_t) &TrackBase::track__NoResample<
+                    MIXTYPE_MULTI, int32_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;
+        default:
+            LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
+            break;
+        }
+        break;
+    default:
+        LOG_ALWAYS_FATAL("bad trackType: %d", trackType);
+        break;
+    }
+    return NULL;
+}
+
+/* Returns the proper process hook for mixing tracks. Currently works only for
+ * PROCESSTYPE_NORESAMPLEONETRACK, a mix involving one track, no resampling.
+ *
+ * TODO: Due to the special mixing considerations of duplicating to
+ * a stereo output track, the input track cannot be MONO.  This should be
+ * prevented by the caller.
+ */
+/* static */
+AudioMixerBase::process_hook_t AudioMixerBase::getProcessHook(
+        int processType, uint32_t channelCount,
+        audio_format_t mixerInFormat, audio_format_t mixerOutFormat)
+{
+    if (processType != PROCESSTYPE_NORESAMPLEONETRACK) { // Only NORESAMPLEONETRACK
+        LOG_ALWAYS_FATAL("bad processType: %d", processType);
+        return NULL;
+    }
+    if (!kUseNewMixer && channelCount == FCC_2 && mixerInFormat == AUDIO_FORMAT_PCM_16_BIT) {
+        return &AudioMixerBase::process__oneTrack16BitsStereoNoResampling;
+    }
+    LOG_ALWAYS_FATAL_IF(channelCount > MAX_NUM_CHANNELS);
+    switch (mixerInFormat) {
+    case AUDIO_FORMAT_PCM_FLOAT:
+        switch (mixerOutFormat) {
+        case AUDIO_FORMAT_PCM_FLOAT:
+            return &AudioMixerBase::process__noResampleOneTrack<
+                    MIXTYPE_MULTI_SAVEONLY, float /*TO*/, float /*TI*/, TYPE_AUX>;
+        case AUDIO_FORMAT_PCM_16_BIT:
+            return &AudioMixerBase::process__noResampleOneTrack<
+                    MIXTYPE_MULTI_SAVEONLY, int16_t /*TO*/, float /*TI*/, TYPE_AUX>;
+        default:
+            LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
+            break;
+        }
+        break;
+    case AUDIO_FORMAT_PCM_16_BIT:
+        switch (mixerOutFormat) {
+        case AUDIO_FORMAT_PCM_FLOAT:
+            return &AudioMixerBase::process__noResampleOneTrack<
+                    MIXTYPE_MULTI_SAVEONLY, float /*TO*/, int16_t /*TI*/, TYPE_AUX>;
+        case AUDIO_FORMAT_PCM_16_BIT:
+            return &AudioMixerBase::process__noResampleOneTrack<
+                    MIXTYPE_MULTI_SAVEONLY, int16_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;
+        default:
+            LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
+            break;
+        }
+        break;
+    default:
+        LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
+        break;
+    }
+    return NULL;
+}
+
+// ----------------------------------------------------------------------------
+} // namespace android
diff --git a/media/libaudioprocessing/include/media/AudioMixer.h b/media/libaudioprocessing/include/media/AudioMixer.h
new file mode 100644
index 0000000..3f7cd48
--- /dev/null
+++ b/media/libaudioprocessing/include/media/AudioMixer.h
@@ -0,0 +1,238 @@
+/*
+**
+** Copyright 2007, 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.
+*/
+
+#ifndef ANDROID_AUDIO_MIXER_H
+#define ANDROID_AUDIO_MIXER_H
+
+#include <pthread.h>
+#include <stdint.h>
+#include <sys/types.h>
+
+#include <android/os/IExternalVibratorService.h>
+#include <media/AudioMixerBase.h>
+#include <media/BufferProviders.h>
+#include <utils/threads.h>
+
+// FIXME This is actually unity gain, which might not be max in future, expressed in U.12
+#define MAX_GAIN_INT AudioMixerBase::UNITY_GAIN_INT
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+// AudioMixer extends AudioMixerBase by adding support for down- and up-mixing
+// and time stretch that are implemented via Effects HAL, and adding support
+// for haptic channels which depends on Vibrator service. This is the version
+// that is used by Audioflinger.
+
+class AudioMixer : public AudioMixerBase
+{
+public:
+    // maximum number of channels supported for the content
+    static const uint32_t MAX_NUM_CHANNELS_TO_DOWNMIX = AUDIO_CHANNEL_COUNT_MAX;
+
+    enum { // extension of AudioMixerBase parameters
+        DOWNMIX_TYPE    = 0x4004,
+        // 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.
+        // for target TIMESTRETCH
+        PLAYBACK_RATE   = 0x4300, // Configure timestretch on this track name;
+                                  // parameter 'value' is a pointer to the new playback rate.
+    };
+
+    typedef enum { // Haptic intensity, should keep consistent with VibratorService
+        HAPTIC_SCALE_MUTE = os::IExternalVibratorService::SCALE_MUTE,
+        HAPTIC_SCALE_VERY_LOW = os::IExternalVibratorService::SCALE_VERY_LOW,
+        HAPTIC_SCALE_LOW = os::IExternalVibratorService::SCALE_LOW,
+        HAPTIC_SCALE_NONE = os::IExternalVibratorService::SCALE_NONE,
+        HAPTIC_SCALE_HIGH = os::IExternalVibratorService::SCALE_HIGH,
+        HAPTIC_SCALE_VERY_HIGH = os::IExternalVibratorService::SCALE_VERY_HIGH,
+    } haptic_intensity_t;
+    static constexpr float HAPTIC_SCALE_VERY_LOW_RATIO = 2.0f / 3.0f;
+    static constexpr float HAPTIC_SCALE_LOW_RATIO = 3.0f / 4.0f;
+    static const constexpr float HAPTIC_MAX_AMPLITUDE_FLOAT = 1.0f;
+
+    static inline bool isValidHapticIntensity(haptic_intensity_t hapticIntensity) {
+        switch (hapticIntensity) {
+        case HAPTIC_SCALE_MUTE:
+        case HAPTIC_SCALE_VERY_LOW:
+        case HAPTIC_SCALE_LOW:
+        case HAPTIC_SCALE_NONE:
+        case HAPTIC_SCALE_HIGH:
+        case HAPTIC_SCALE_VERY_HIGH:
+            return true;
+        default:
+            return false;
+        }
+    }
+
+    AudioMixer(size_t frameCount, uint32_t sampleRate)
+            : AudioMixerBase(frameCount, sampleRate) {
+        pthread_once(&sOnceControl, &sInitRoutine);
+    }
+
+    bool isValidChannelMask(audio_channel_mask_t channelMask) const override;
+
+    void setParameter(int name, int target, int param, void *value) override;
+    void setBufferProvider(int name, AudioBufferProvider* bufferProvider);
+
+private:
+
+    struct Track : public TrackBase {
+        Track() : TrackBase() {}
+
+        ~Track()
+        {
+            // mInputBufferProvider need not be deleted.
+            // Ensure the order of destruction of buffer providers as they
+            // release the upstream provider in the destructor.
+            mTimestretchBufferProvider.reset(nullptr);
+            mPostDownmixReformatBufferProvider.reset(nullptr);
+            mDownmixerBufferProvider.reset(nullptr);
+            mReformatBufferProvider.reset(nullptr);
+            mContractChannelsNonDestructiveBufferProvider.reset(nullptr);
+            mAdjustChannelsBufferProvider.reset(nullptr);
+        }
+
+        uint32_t getOutputChannelCount() override {
+            return mDownmixerBufferProvider.get() != nullptr ? mMixerChannelCount : channelCount;
+        }
+        uint32_t getMixerChannelCount() override {
+            return mMixerChannelCount + mMixerHapticChannelCount;
+        }
+
+        status_t    prepareForDownmix();
+        void        unprepareForDownmix();
+        status_t    prepareForReformat();
+        void        unprepareForReformat();
+        status_t    prepareForAdjustChannels();
+        void        unprepareForAdjustChannels();
+        status_t    prepareForAdjustChannelsNonDestructive(size_t frames);
+        void        unprepareForAdjustChannelsNonDestructive();
+        void        clearContractedBuffer();
+        bool        setPlaybackRate(const AudioPlaybackRate &playbackRate);
+        void        reconfigureBufferProviders();
+
+        /* Buffer providers are constructed to translate the track input data as needed.
+         * See DownmixerBufferProvider below for how the Track buffer provider
+         * is wrapped by another one when dowmixing is required.
+         *
+         * TODO: perhaps make a single PlaybackConverterProvider class to move
+         * all pre-mixer track buffer conversions outside the AudioMixer class.
+         *
+         * 1) mInputBufferProvider: The AudioTrack buffer provider.
+         * 2) mAdjustChannelsBufferProvider: Expands or contracts sample data from one interleaved
+         *    channel format to another. Expanded channels are filled with zeros and put at the end
+         *    of each audio frame. Contracted channels are copied to the end of the buffer.
+         * 3) mContractChannelsNonDestructiveBufferProvider: Non-destructively contract sample data.
+         *    This is currently using at audio-haptic coupled playback to separate audio and haptic
+         *    data. Contracted channels could be written to given buffer.
+         * 4) mReformatBufferProvider: If not NULL, performs the audio reformat to
+         *    match either mMixerInFormat or mDownmixRequiresFormat, if the downmixer
+         *    requires reformat. For example, it may convert floating point input to
+         *    PCM_16_bit if that's required by the downmixer.
+         * 5) mDownmixerBufferProvider: If not NULL, performs the channel remixing to match
+         *    the number of channels required by the mixer sink.
+         * 6) mPostDownmixReformatBufferProvider: If not NULL, performs reformatting from
+         *    the downmixer requirements to the mixer engine input requirements.
+         * 7) mTimestretchBufferProvider: Adds timestretching for playback rate
+         */
+        AudioBufferProvider* mInputBufferProvider;    // externally provided buffer provider.
+        // TODO: combine mAdjustChannelsBufferProvider and
+        // mContractChannelsNonDestructiveBufferProvider
+        std::unique_ptr<PassthruBufferProvider> mAdjustChannelsBufferProvider;
+        std::unique_ptr<PassthruBufferProvider> mContractChannelsNonDestructiveBufferProvider;
+        std::unique_ptr<PassthruBufferProvider> mReformatBufferProvider;
+        std::unique_ptr<PassthruBufferProvider> mDownmixerBufferProvider;
+        std::unique_ptr<PassthruBufferProvider> mPostDownmixReformatBufferProvider;
+        std::unique_ptr<PassthruBufferProvider> mTimestretchBufferProvider;
+
+        audio_format_t mDownmixRequiresFormat;  // required downmixer format
+                                                // AUDIO_FORMAT_PCM_16_BIT if 16 bit necessary
+                                                // AUDIO_FORMAT_INVALID if no required format
+
+        AudioPlaybackRate    mPlaybackRate;
+
+        // Haptic
+        bool                 mHapticPlaybackEnabled;
+        haptic_intensity_t   mHapticIntensity;
+        audio_channel_mask_t mHapticChannelMask;
+        uint32_t             mHapticChannelCount;
+        audio_channel_mask_t mMixerHapticChannelMask;
+        uint32_t             mMixerHapticChannelCount;
+        uint32_t             mAdjustInChannelCount;
+        uint32_t             mAdjustOutChannelCount;
+        uint32_t             mAdjustNonDestructiveInChannelCount;
+        uint32_t             mAdjustNonDestructiveOutChannelCount;
+        bool                 mKeepContractedChannels;
+
+        float getHapticScaleGamma() const {
+        // Need to keep consistent with the value in VibratorService.
+        switch (mHapticIntensity) {
+        case HAPTIC_SCALE_VERY_LOW:
+            return 2.0f;
+        case HAPTIC_SCALE_LOW:
+            return 1.5f;
+        case HAPTIC_SCALE_HIGH:
+            return 0.5f;
+        case HAPTIC_SCALE_VERY_HIGH:
+            return 0.25f;
+        default:
+            return 1.0f;
+        }
+        }
+
+        float getHapticMaxAmplitudeRatio() const {
+        // Need to keep consistent with the value in VibratorService.
+        switch (mHapticIntensity) {
+        case HAPTIC_SCALE_VERY_LOW:
+            return HAPTIC_SCALE_VERY_LOW_RATIO;
+        case HAPTIC_SCALE_LOW:
+            return HAPTIC_SCALE_LOW_RATIO;
+        case HAPTIC_SCALE_NONE:
+        case HAPTIC_SCALE_HIGH:
+        case HAPTIC_SCALE_VERY_HIGH:
+            return 1.0f;
+        default:
+            return 0.0f;
+        }
+        }
+    };
+
+    inline std::shared_ptr<Track> getTrack(int name) {
+        return std::static_pointer_cast<Track>(mTracks[name]);
+    }
+
+    std::shared_ptr<TrackBase> preCreateTrack() override;
+    status_t postCreateTrack(TrackBase *track) override;
+
+    void preProcess() override;
+    void postProcess() override;
+
+    bool setChannelMasks(int name,
+            audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask) override;
+
+    static void sInitRoutine();
+
+    static pthread_once_t sOnceControl; // initialized in constructor by first new
+};
+
+// ----------------------------------------------------------------------------
+} // namespace android
+
+#endif // ANDROID_AUDIO_MIXER_H
diff --git a/media/libaudioprocessing/include/media/AudioMixerBase.h b/media/libaudioprocessing/include/media/AudioMixerBase.h
new file mode 100644
index 0000000..805b6d0
--- /dev/null
+++ b/media/libaudioprocessing/include/media/AudioMixerBase.h
@@ -0,0 +1,359 @@
+/*
+**
+** Copyright 2019, 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.
+*/
+
+#ifndef ANDROID_AUDIO_MIXER_BASE_H
+#define ANDROID_AUDIO_MIXER_BASE_H
+
+#include <map>
+#include <memory>
+#include <string>
+#include <unordered_map>
+#include <vector>
+
+#include <media/AudioBufferProvider.h>
+#include <media/AudioResampler.h>
+#include <media/AudioResamplerPublic.h>
+#include <system/audio.h>
+#include <utils/Compat.h>
+
+// This must match frameworks/av/services/audioflinger/Configuration.h
+// when used with the Audio Framework.
+#define FLOAT_AUX
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+// AudioMixerBase is functional on its own if only mixing and resampling
+// is needed.
+
+class AudioMixerBase
+{
+public:
+    // Do not change these unless underlying code changes.
+    // This mixer has a hard-coded upper limit of 8 channels for output.
+    static constexpr uint32_t MAX_NUM_CHANNELS = FCC_8;
+    static constexpr uint32_t MAX_NUM_VOLUMES = FCC_2; // stereo volume only
+
+    static const uint16_t UNITY_GAIN_INT = 0x1000;
+    static const CONSTEXPR float UNITY_GAIN_FLOAT = 1.0f;
+
+    enum { // names
+        // setParameter targets
+        TRACK           = 0x3000,
+        RESAMPLE        = 0x3001,
+        RAMP_VOLUME     = 0x3002, // ramp to new volume
+        VOLUME          = 0x3003, // don't ramp
+        TIMESTRETCH     = 0x3004,
+
+        // set Parameter names
+        // for target TRACK
+        CHANNEL_MASK    = 0x4000,
+        FORMAT          = 0x4001,
+        MAIN_BUFFER     = 0x4002,
+        AUX_BUFFER      = 0x4003,
+        // 0x4004 reserved
+        MIXER_FORMAT    = 0x4005, // AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
+        MIXER_CHANNEL_MASK = 0x4006, // Channel mask for mixer output
+        // for target RESAMPLE
+        SAMPLE_RATE     = 0x4100, // Configure sample rate conversion on this track name;
+                                  // parameter 'value' is the new sample rate in Hz.
+                                  // Only creates a sample rate converter the first time that
+                                  // the track sample rate is different from the mix sample rate.
+                                  // If the new sample rate is the same as the mix sample rate,
+                                  // and a sample rate converter already exists,
+                                  // then the sample rate converter remains present but is a no-op.
+        RESET           = 0x4101, // Reset sample rate converter without changing sample rate.
+                                  // This clears out the resampler's input buffer.
+        REMOVE          = 0x4102, // Remove the sample rate converter on this track name;
+                                  // the track is restored to the mix sample rate.
+        // for target RAMP_VOLUME and VOLUME (8 channels max)
+        // FIXME use float for these 3 to improve the dynamic range
+        VOLUME0         = 0x4200,
+        VOLUME1         = 0x4201,
+        AUXLEVEL        = 0x4210,
+    };
+
+    AudioMixerBase(size_t frameCount, uint32_t sampleRate)
+        : mSampleRate(sampleRate)
+        , mFrameCount(frameCount) {
+    }
+
+    virtual ~AudioMixerBase() {}
+
+    virtual bool isValidFormat(audio_format_t format) const;
+    virtual bool isValidChannelMask(audio_channel_mask_t channelMask) const;
+
+    // Create a new track in the mixer.
+    //
+    // \param name        a unique user-provided integer associated with the track.
+    //                    If name already exists, the function will abort.
+    // \param channelMask output channel mask.
+    // \param format      PCM format
+    // \param sessionId   Session id for the track. Tracks with the same
+    //                    session id will be submixed together.
+    //
+    // \return OK        on success.
+    //         BAD_VALUE if the format does not satisfy isValidFormat()
+    //                   or the channelMask does not satisfy isValidChannelMask().
+    status_t    create(
+            int name, audio_channel_mask_t channelMask, audio_format_t format, int sessionId);
+
+    bool        exists(int name) const {
+        return mTracks.count(name) > 0;
+    }
+
+    // Free an allocated track by name.
+    void        destroy(int name);
+
+    // Enable or disable an allocated track by name
+    void        enable(int name);
+    void        disable(int name);
+
+    virtual void setParameter(int name, int target, int param, void *value);
+
+    void        process() {
+        preProcess();
+        (this->*mHook)();
+        postProcess();
+    }
+
+    size_t      getUnreleasedFrames(int name) const;
+
+    std::string trackNames() const;
+
+  protected:
+    // Set kUseNewMixer to true to use the new mixer engine always. Otherwise the
+    // original code will be used for stereo sinks, the new mixer for everything else.
+    static constexpr bool kUseNewMixer = true;
+
+    // Set kUseFloat to true to allow floating input into the mixer engine.
+    // If kUseNewMixer is false, this is ignored or may be overridden internally
+    static constexpr bool kUseFloat = true;
+
+#ifdef FLOAT_AUX
+    using TYPE_AUX = float;
+    static_assert(kUseNewMixer && kUseFloat,
+            "kUseNewMixer and kUseFloat must be true for FLOAT_AUX option");
+#else
+    using TYPE_AUX = int32_t; // q4.27
+#endif
+
+    /* For multi-format functions (calls template functions
+     * in AudioMixerOps.h).  The template parameters are as follows:
+     *
+     *   MIXTYPE     (see AudioMixerOps.h MIXTYPE_* enumeration)
+     *   USEFLOATVOL (set to true if float volume is used)
+     *   ADJUSTVOL   (set to true if volume ramp parameters needs adjustment afterwards)
+     *   TO: int32_t (Q4.27) or float
+     *   TI: int32_t (Q4.27) or int16_t (Q0.15) or float
+     *   TA: int32_t (Q4.27)
+     */
+
+    enum {
+        // FIXME this representation permits up to 8 channels
+        NEEDS_CHANNEL_COUNT__MASK   = 0x00000007,
+    };
+
+    enum {
+        NEEDS_CHANNEL_1             = 0x00000000,   // mono
+        NEEDS_CHANNEL_2             = 0x00000001,   // stereo
+
+        // sample format is not explicitly specified, and is assumed to be AUDIO_FORMAT_PCM_16_BIT
+
+        NEEDS_MUTE                  = 0x00000100,
+        NEEDS_RESAMPLE              = 0x00001000,
+        NEEDS_AUX                   = 0x00010000,
+    };
+
+    // hook types
+    enum {
+        PROCESSTYPE_NORESAMPLEONETRACK, // others set elsewhere
+    };
+
+    enum {
+        TRACKTYPE_NOP,
+        TRACKTYPE_RESAMPLE,
+        TRACKTYPE_NORESAMPLE,
+        TRACKTYPE_NORESAMPLEMONO,
+    };
+
+    // process hook functionality
+    using process_hook_t = void(AudioMixerBase::*)();
+
+    struct TrackBase;
+    using hook_t = void(TrackBase::*)(
+            int32_t* output, size_t numOutFrames, int32_t* temp, int32_t* aux);
+
+    struct TrackBase {
+        TrackBase()
+            : bufferProvider(nullptr)
+        {
+            // TODO: move additional initialization here.
+        }
+        virtual ~TrackBase() {}
+
+        virtual uint32_t getOutputChannelCount() { return channelCount; }
+        virtual uint32_t getMixerChannelCount() { return mMixerChannelCount; }
+
+        bool        needsRamp() { return (volumeInc[0] | volumeInc[1] | auxInc) != 0; }
+        bool        setResampler(uint32_t trackSampleRate, uint32_t devSampleRate);
+        bool        doesResample() const { return mResampler.get() != nullptr; }
+        void        recreateResampler(uint32_t devSampleRate);
+        void        resetResampler() { if (mResampler.get() != nullptr) mResampler->reset(); }
+        void        adjustVolumeRamp(bool aux, bool useFloat = false);
+        size_t      getUnreleasedFrames() const { return mResampler.get() != nullptr ?
+                                                    mResampler->getUnreleasedFrames() : 0; };
+
+        static hook_t getTrackHook(int trackType, uint32_t channelCount,
+                audio_format_t mixerInFormat, audio_format_t mixerOutFormat);
+
+        void track__nop(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
+
+        template <int MIXTYPE, bool USEFLOATVOL, bool ADJUSTVOL,
+            typename TO, typename TI, typename TA>
+        void volumeMix(TO *out, size_t outFrames, const TI *in, TA *aux, bool ramp);
+
+        uint32_t    needs;
+
+        // TODO: Eventually remove legacy integer volume settings
+        union {
+        int16_t     volume[MAX_NUM_VOLUMES]; // U4.12 fixed point (top bit should be zero)
+        int32_t     volumeRL;
+        };
+
+        int32_t     prevVolume[MAX_NUM_VOLUMES];
+        int32_t     volumeInc[MAX_NUM_VOLUMES];
+        int32_t     auxInc;
+        int32_t     prevAuxLevel;
+        int16_t     auxLevel;       // 0 <= auxLevel <= MAX_GAIN_INT, but signed for mul performance
+
+        uint16_t    frameCount;
+
+        uint8_t     channelCount;   // 1 or 2, redundant with (needs & NEEDS_CHANNEL_COUNT__MASK)
+        uint8_t     unused_padding; // formerly format, was always 16
+        uint16_t    enabled;        // actually bool
+        audio_channel_mask_t channelMask;
+
+        // actual buffer provider used by the track hooks
+        AudioBufferProvider*                bufferProvider;
+
+        mutable AudioBufferProvider::Buffer buffer; // 8 bytes
+
+        hook_t      hook;
+        const void  *mIn;             // current location in buffer
+
+        std::unique_ptr<AudioResampler> mResampler;
+        uint32_t    sampleRate;
+        int32_t*    mainBuffer;
+        int32_t*    auxBuffer;
+
+        int32_t     sessionId;
+
+        audio_format_t mMixerFormat;     // output mix format: AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
+        audio_format_t mFormat;          // input track format
+        audio_format_t mMixerInFormat;   // mix internal format AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
+                                         // each track must be converted to this format.
+
+        float          mVolume[MAX_NUM_VOLUMES];     // floating point set volume
+        float          mPrevVolume[MAX_NUM_VOLUMES]; // floating point previous volume
+        float          mVolumeInc[MAX_NUM_VOLUMES];  // floating point volume increment
+
+        float          mAuxLevel;                     // floating point set aux level
+        float          mPrevAuxLevel;                 // floating point prev aux level
+        float          mAuxInc;                       // floating point aux increment
+
+        audio_channel_mask_t mMixerChannelMask;
+        uint32_t             mMixerChannelCount;
+
+      protected:
+
+        // hooks
+        void track__genericResample(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
+        void track__16BitsStereo(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
+        void track__16BitsMono(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
+
+        void volumeRampStereo(int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
+        void volumeStereo(int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
+
+        // multi-format track hooks
+        template <int MIXTYPE, typename TO, typename TI, typename TA>
+        void track__Resample(TO* out, size_t frameCount, TO* temp __unused, TA* aux);
+        template <int MIXTYPE, typename TO, typename TI, typename TA>
+        void track__NoResample(TO* out, size_t frameCount, TO* temp __unused, TA* aux);
+    };
+
+    // preCreateTrack must create an instance of a proper TrackBase descendant.
+    // postCreateTrack is called after filling out fields of TrackBase. It can
+    // abort track creation by returning non-OK status. See the implementation
+    // of create() for details.
+    virtual std::shared_ptr<TrackBase> preCreateTrack();
+    virtual status_t postCreateTrack(TrackBase *track __unused) { return OK; }
+
+    // preProcess is called before the process hook, postProcess after,
+    // see the implementation of process() method.
+    virtual void preProcess() {}
+    virtual void postProcess() {}
+
+    virtual bool setChannelMasks(int name,
+            audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask);
+
+    // Called when track info changes and a new process hook should be determined.
+    void invalidate() {
+        mHook = &AudioMixerBase::process__validate;
+    }
+
+    void process__validate();
+    void process__nop();
+    void process__genericNoResampling();
+    void process__genericResampling();
+    void process__oneTrack16BitsStereoNoResampling();
+
+    template <int MIXTYPE, typename TO, typename TI, typename TA>
+    void process__noResampleOneTrack();
+
+    static process_hook_t getProcessHook(int processType, uint32_t channelCount,
+            audio_format_t mixerInFormat, audio_format_t mixerOutFormat);
+
+    static void convertMixerFormat(void *out, audio_format_t mixerOutFormat,
+            void *in, audio_format_t mixerInFormat, size_t sampleCount);
+
+    // initialization constants
+    const uint32_t mSampleRate;
+    const size_t mFrameCount;
+
+    process_hook_t mHook = &AudioMixerBase::process__nop;   // one of process__*, never nullptr
+
+    // the size of the type (int32_t) should be the largest of all types supported
+    // by the mixer.
+    std::unique_ptr<int32_t[]> mOutputTemp;
+    std::unique_ptr<int32_t[]> mResampleTemp;
+
+    // track names grouped by main buffer, in no particular order of main buffer.
+    // however names for a particular main buffer are in order (by construction).
+    std::unordered_map<void * /* mainBuffer */, std::vector<int /* name */>> mGroups;
+
+    // track names that are enabled, in increasing order (by construction).
+    std::vector<int /* name */> mEnabled;
+
+    // track smart pointers, by name, in increasing order of name.
+    std::map<int /* name */, std::shared_ptr<TrackBase>> mTracks;
+};
+
+}  // namespace android
+
+#endif  // ANDROID_AUDIO_MIXER_BASE_H
diff --git a/media/libmedia/include/media/BufferProviders.h b/media/libaudioprocessing/include/media/BufferProviders.h
similarity index 100%
rename from media/libmedia/include/media/BufferProviders.h
rename to media/libaudioprocessing/include/media/BufferProviders.h
diff --git a/media/libmedia/AudioParameter.cpp b/media/libmedia/AudioParameter.cpp
index 060b92b..9f34035 100644
--- a/media/libmedia/AudioParameter.cpp
+++ b/media/libmedia/AudioParameter.cpp
@@ -42,6 +42,8 @@
 const char * const AudioParameter::keyStreamHwAvSync = AUDIO_PARAMETER_STREAM_HW_AV_SYNC;
 const char * const AudioParameter::keyDeviceConnect = AUDIO_PARAMETER_DEVICE_CONNECT;
 const char * const AudioParameter::keyDeviceDisconnect = AUDIO_PARAMETER_DEVICE_DISCONNECT;
+const char * const AudioParameter::keyStreamConnect = AUDIO_PARAMETER_DEVICE_CONNECT;
+const char * const AudioParameter::keyStreamDisconnect = AUDIO_PARAMETER_DEVICE_DISCONNECT;
 const char * const AudioParameter::keyStreamSupportedFormats = AUDIO_PARAMETER_STREAM_SUP_FORMATS;
 const char * const AudioParameter::keyStreamSupportedChannels = AUDIO_PARAMETER_STREAM_SUP_CHANNELS;
 const char * const AudioParameter::keyStreamSupportedSamplingRates =
diff --git a/media/libmedia/IResourceManagerService.cpp b/media/libmedia/IResourceManagerService.cpp
index 9724fc1..00f9b88 100644
--- a/media/libmedia/IResourceManagerService.cpp
+++ b/media/libmedia/IResourceManagerService.cpp
@@ -72,12 +72,14 @@
 
     virtual void addResource(
             int pid,
+            int uid,
             int64_t clientId,
             const sp<IResourceManagerClient> client,
             const Vector<MediaResource> &resources) {
         Parcel data, reply;
         data.writeInterfaceToken(IResourceManagerService::getInterfaceDescriptor());
         data.writeInt32(pid);
+        data.writeInt32(uid);
         data.writeInt64(clientId);
         data.writeStrongBinder(IInterface::asBinder(client));
         writeToParcel(&data, resources);
@@ -129,6 +131,7 @@
         case ADD_RESOURCE: {
             CHECK_INTERFACE(IResourceManagerService, data, reply);
             int pid = data.readInt32();
+            int uid = data.readInt32();
             int64_t clientId = data.readInt64();
             sp<IResourceManagerClient> client(
                     interface_cast<IResourceManagerClient>(data.readStrongBinder()));
@@ -137,7 +140,7 @@
             }
             Vector<MediaResource> resources;
             readFromParcel(data, &resources);
-            addResource(pid, clientId, client, resources);
+            addResource(pid, uid, clientId, client, resources);
             return NO_ERROR;
         } break;
 
diff --git a/media/libmedia/include/media/IResourceManagerService.h b/media/libmedia/include/media/IResourceManagerService.h
index 1e4f6de..404519b 100644
--- a/media/libmedia/include/media/IResourceManagerService.h
+++ b/media/libmedia/include/media/IResourceManagerService.h
@@ -39,6 +39,7 @@
 
     virtual void addResource(
             int pid,
+            int uid,
             int64_t clientId,
             const sp<IResourceManagerClient> client,
             const Vector<MediaResource> &resources) = 0;
diff --git a/media/libmedia/include/media/MediaResource.h b/media/libmedia/include/media/MediaResource.h
index e1fdb9b..10d0e3b 100644
--- a/media/libmedia/include/media/MediaResource.h
+++ b/media/libmedia/include/media/MediaResource.h
@@ -31,12 +31,13 @@
         kNonSecureCodec,
         kGraphicMemory,
         kCpuBoost,
+        kBattery,
     };
 
     enum SubType {
         kUnspecifiedSubType = 0,
         kAudioCodec,
-        kVideoCodec
+        kVideoCodec,
     };
 
     MediaResource();
diff --git a/media/libmedia/include/media/TypeConverter.h b/media/libmedia/include/media/TypeConverter.h
index 2f8c209..011498a 100644
--- a/media/libmedia/include/media/TypeConverter.h
+++ b/media/libmedia/include/media/TypeConverter.h
@@ -17,10 +17,11 @@
 #ifndef ANDROID_TYPE_CONVERTER_H_
 #define ANDROID_TYPE_CONVERTER_H_
 
+#include <set>
 #include <string>
 #include <string.h>
-
 #include <vector>
+
 #include <system/audio.h>
 #include <utils/Log.h>
 #include <utils/Vector.h>
@@ -42,16 +43,6 @@
     }
 };
 template <typename T>
-struct VectorTraits
-{
-    typedef T Type;
-    typedef Vector<Type> Collection;
-    static void add(Collection &collection, Type value)
-    {
-        collection.add(value);
-    }
-};
-template <typename T>
 struct SortedVectorTraits
 {
     typedef T Type;
@@ -61,18 +52,28 @@
         collection.add(value);
     }
 };
+template <typename T>
+struct SetTraits
+{
+    typedef T Type;
+    typedef std::set<Type> Collection;
+    static void add(Collection &collection, Type value)
+    {
+        collection.insert(value);
+    }
+};
 
-using SampleRateTraits = SortedVectorTraits<uint32_t>;
+using SampleRateTraits = SetTraits<uint32_t>;
 using DeviceTraits = DefaultTraits<audio_devices_t>;
 struct OutputDeviceTraits : public DeviceTraits {};
 struct InputDeviceTraits : public DeviceTraits {};
-using ChannelTraits = SortedVectorTraits<audio_channel_mask_t>;
+using ChannelTraits = SetTraits<audio_channel_mask_t>;
 struct OutputChannelTraits : public ChannelTraits {};
 struct InputChannelTraits : public ChannelTraits {};
 struct ChannelIndexTraits : public ChannelTraits {};
 using InputFlagTraits = DefaultTraits<audio_input_flags_t>;
 using OutputFlagTraits = DefaultTraits<audio_output_flags_t>;
-using FormatTraits = VectorTraits<audio_format_t>;
+using FormatTraits = DefaultTraits<audio_format_t>;
 using GainModeTraits = DefaultTraits<audio_gain_mode_t>;
 using StreamTraits = DefaultTraits<audio_stream_type_t>;
 using AudioModeTraits = DefaultTraits<audio_mode_t>;
@@ -259,6 +260,7 @@
                                     || std::is_same<T, audio_source_t>::value
                                     || std::is_same<T, audio_stream_type_t>::value
                                     || std::is_same<T, audio_usage_t>::value
+                                    || std::is_same<T, audio_format_t>::value
                                     , int> = 0>
 static inline std::string toString(const T& value)
 {
@@ -291,14 +293,6 @@
     return result;
 }
 
-// TODO: Remove when FormatTraits uses DefaultTraits.
-static inline std::string toString(const audio_format_t& format)
-{
-    std::string result;
-    return TypeConverter<VectorTraits<audio_format_t>>::toString(format, result)
-            ? result : std::to_string(static_cast<int>(format));
-}
-
 static inline std::string toString(const audio_attributes_t& attributes)
 {
     std::ostringstream result;
diff --git a/media/libmedia/xsd/vts/Android.bp b/media/libmedia/xsd/vts/Android.bp
index b590a12..4739504 100644
--- a/media/libmedia/xsd/vts/Android.bp
+++ b/media/libmedia/xsd/vts/Android.bp
@@ -24,6 +24,7 @@
         "libxml2",
     ],
     shared_libs: [
+        "libbase",
         "liblog",
     ],
     cflags: [
diff --git a/media/libmedia/xsd/vts/ValidateMediaProfiles.cpp b/media/libmedia/xsd/vts/ValidateMediaProfiles.cpp
index ff9b060..7729d52 100644
--- a/media/libmedia/xsd/vts/ValidateMediaProfiles.cpp
+++ b/media/libmedia/xsd/vts/ValidateMediaProfiles.cpp
@@ -14,6 +14,10 @@
  * limitations under the License.
  */
 
+#include <string>
+
+#include <android-base/file.h>
+#include <android-base/properties.h>
 #include "utility/ValidateXml.h"
 
 TEST(CheckConfig, mediaProfilesValidation) {
@@ -21,8 +25,12 @@
                    "Verify that the media profiles file "
                    "is valid according to the schema");
 
-    const char* location = "/vendor/etc";
+    std::string mediaSettingsPath = android::base::GetProperty("media.settings.xml", "");
+    if (mediaSettingsPath.empty()) {
+        mediaSettingsPath.assign("/vendor/etc/media_profiles_V1_0.xml");
+    }
 
-    EXPECT_ONE_VALID_XML_MULTIPLE_LOCATIONS("media_profiles_V1_0.xml", {location},
+    EXPECT_ONE_VALID_XML_MULTIPLE_LOCATIONS(android::base::Basename(mediaSettingsPath).c_str(),
+                                            {android::base::Dirname(mediaSettingsPath).c_str()},
                                             "/data/local/tmp/media_profiles.xsd");
 }
diff --git a/media/libmediametrics/Android.bp b/media/libmediametrics/Android.bp
index 33368be..9d348ec 100644
--- a/media/libmediametrics/Android.bp
+++ b/media/libmediametrics/Android.bp
@@ -40,7 +40,12 @@
 
     header_abi_checker: {
         enabled: true,
+        symbol_file: "libmediametrics.map.txt",
     },
+
+    visibility: [
+        "//frameworks/av:__subpackages__",
+        "//frameworks/base/core/jni",
+        "//frameworks/base/media/jni",
+    ],
 }
-
-
diff --git a/media/libmediaplayerservice/nuplayer/RTSPSource.cpp b/media/libmediaplayerservice/nuplayer/RTSPSource.cpp
index bf14ec2..83da092 100644
--- a/media/libmediaplayerservice/nuplayer/RTSPSource.cpp
+++ b/media/libmediaplayerservice/nuplayer/RTSPSource.cpp
@@ -144,6 +144,10 @@
     if (mLooper == NULL) {
         return;
     }
+
+    // Close socket before posting message to RTSPSource message handler.
+    close(mHandler->getARTSPConnection()->getSocket());
+
     sp<AMessage> msg = new AMessage(kWhatDisconnect, this);
 
     sp<AMessage> dummy;
diff --git a/media/libnbaio/Android.bp b/media/libnbaio/Android.bp
index a4df38d..04ddcff 100644
--- a/media/libnbaio/Android.bp
+++ b/media/libnbaio/Android.bp
@@ -1,4 +1,3 @@
-
 cc_defaults {
     name: "libnbaio_mono_defaults",
     srcs: [
@@ -9,20 +8,27 @@
     header_libs: [
         "libaudioclient_headers",
         "libaudio_system_headers",
-        "libmedia_headers",
     ],
     export_header_lib_headers: [
         "libaudioclient_headers",
-        "libmedia_headers",
     ],
 
     shared_libs: [
         "libaudioutils",
+        "libcutils",
         "liblog",
         "libutils",
     ],
+    export_shared_lib_headers: [
+        "libaudioutils",
+    ],
 
     export_include_dirs: ["include_mono"],
+
+    cflags: [
+        "-Werror",
+        "-Wall",
+    ],
 }
 
 // libnbaio_mono is the part of libnbaio that is available for vendors to use. Vendor modules can't
@@ -53,20 +59,7 @@
     // ],
     // static_libs: ["libsndfile"],
 
-    shared_libs: [
-        "libaudioutils",
-        "libbinder",
-        "libcutils",
-        "liblog",
-        "libutils",
-    ],
-
-    cflags: [
-        "-Werror",
-        "-Wall",
-    ],
-
-    include_dirs: ["system/media/audio_utils/include"],
+    header_libs: ["libaudiohal_headers"],
 
     export_include_dirs: ["include"],
 }
diff --git a/media/libnbaio/include_mono/media/nbaio/MonoPipe.h b/media/libnbaio/include_mono/media/nbaio/MonoPipe.h
index c51d0fe..926d84a 100644
--- a/media/libnbaio/include_mono/media/nbaio/MonoPipe.h
+++ b/media/libnbaio/include_mono/media/nbaio/MonoPipe.h
@@ -19,7 +19,7 @@
 
 #include <time.h>
 #include <audio_utils/fifo.h>
-#include <media/SingleStateQueue.h>
+#include <media/nbaio/SingleStateQueue.h>
 #include <media/nbaio/NBAIO.h>
 
 namespace android {
diff --git a/media/libmedia/include/media/SingleStateQueue.h b/media/libnbaio/include_mono/media/nbaio/SingleStateQueue.h
similarity index 100%
rename from media/libmedia/include/media/SingleStateQueue.h
rename to media/libnbaio/include_mono/media/nbaio/SingleStateQueue.h
diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index 3d67c91..d198d39 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -4505,22 +4505,38 @@
 status_t ACodec::configureImageGrid(
         const sp<AMessage> &msg, sp<AMessage> &outputFormat) {
     int32_t tileWidth, tileHeight, gridRows, gridCols;
-    if (!msg->findInt32("tile-width", &tileWidth) ||
-        !msg->findInt32("tile-height", &tileHeight) ||
-        !msg->findInt32("grid-rows", &gridRows) ||
-        !msg->findInt32("grid-cols", &gridCols)) {
+    OMX_BOOL useGrid = OMX_FALSE;
+    if (msg->findInt32("tile-width", &tileWidth) &&
+        msg->findInt32("tile-height", &tileHeight) &&
+        msg->findInt32("grid-rows", &gridRows) &&
+        msg->findInt32("grid-cols", &gridCols)) {
+        useGrid = OMX_TRUE;
+    } else {
+        // when bEnabled is false, the tile info is not used,
+        // but clear out these too.
+        tileWidth = tileHeight = gridRows = gridCols = 0;
+    }
+
+    if (!mIsImage && !useGrid) {
         return OK;
     }
 
     OMX_VIDEO_PARAM_ANDROID_IMAGEGRIDTYPE gridType;
     InitOMXParams(&gridType);
     gridType.nPortIndex = kPortIndexOutput;
-    gridType.bEnabled = OMX_TRUE;
+    gridType.bEnabled = useGrid;
     gridType.nTileWidth = tileWidth;
     gridType.nTileHeight = tileHeight;
     gridType.nGridRows = gridRows;
     gridType.nGridCols = gridCols;
 
+    ALOGV("sending image grid info to component: bEnabled %d, tile %dx%d, grid %dx%d",
+            gridType.bEnabled,
+            gridType.nTileWidth,
+            gridType.nTileHeight,
+            gridType.nGridRows,
+            gridType.nGridCols);
+
     status_t err = mOMXNode->setParameter(
             (OMX_INDEXTYPE)OMX_IndexParamVideoAndroidImageGrid,
             &gridType, sizeof(gridType));
@@ -4541,6 +4557,13 @@
             (OMX_INDEXTYPE)OMX_IndexParamVideoAndroidImageGrid,
             &gridType, sizeof(gridType));
 
+    ALOGV("received image grid info from component: bEnabled %d, tile %dx%d, grid %dx%d",
+            gridType.bEnabled,
+            gridType.nTileWidth,
+            gridType.nTileHeight,
+            gridType.nGridRows,
+            gridType.nGridCols);
+
     if (err == OK && gridType.bEnabled) {
         outputFormat->setInt32("tile-width", gridType.nTileWidth);
         outputFormat->setInt32("tile-height", gridType.nTileHeight);
diff --git a/media/libstagefright/MPEG4Writer.cpp b/media/libstagefright/MPEG4Writer.cpp
index 2f13dc9..f130c9b 100644
--- a/media/libstagefright/MPEG4Writer.cpp
+++ b/media/libstagefright/MPEG4Writer.cpp
@@ -1635,8 +1635,13 @@
         return BAD_VALUE;
     }
 
+    // Increase moovExtraSize once only irrespective of how many times
+    // setCaptureRate is called.
+    bool containsCaptureFps = mMetaKeys->contains(kMetaKey_CaptureFps);
     mMetaKeys->setFloat(kMetaKey_CaptureFps, captureFps);
-    mMoovExtraSize += sizeof(kMetaKey_CaptureFps) + 4 + 32;
+    if (!containsCaptureFps) {
+        mMoovExtraSize += sizeof(kMetaKey_CaptureFps) + 4 + 32;
+    }
 
     return OK;
 }
diff --git a/media/libstagefright/MediaCodec.cpp b/media/libstagefright/MediaCodec.cpp
index f579e9d..a6a856c 100644
--- a/media/libstagefright/MediaCodec.cpp
+++ b/media/libstagefright/MediaCodec.cpp
@@ -57,7 +57,6 @@
 #include <media/stagefright/OMXClient.h>
 #include <media/stagefright/PersistentSurface.h>
 #include <media/stagefright/SurfaceUtils.h>
-#include <mediautils/BatteryNotifier.h>
 #include <private/android_filesystem_config.h>
 #include <utils/Singleton.h>
 
@@ -166,8 +165,9 @@
     DISALLOW_EVIL_CONSTRUCTORS(ResourceManagerClient);
 };
 
-MediaCodec::ResourceManagerServiceProxy::ResourceManagerServiceProxy(pid_t pid)
-        : mPid(pid) {
+MediaCodec::ResourceManagerServiceProxy::ResourceManagerServiceProxy(
+        pid_t pid, uid_t uid)
+        : mPid(pid), mUid(uid) {
     if (mPid == MediaCodec::kNoPid) {
         mPid = IPCThreadState::self()->getCallingPid();
     }
@@ -204,7 +204,7 @@
     if (mService == NULL) {
         return;
     }
-    mService->addResource(mPid, clientId, client, resources);
+    mService->addResource(mPid, mUid, clientId, client, resources);
 }
 
 void MediaCodec::ResourceManagerServiceProxy::removeResource(int64_t clientId) {
@@ -517,8 +517,6 @@
       mStickyError(OK),
       mSoftRenderer(NULL),
       mAnalyticsItem(NULL),
-      mResourceManagerClient(new ResourceManagerClient(this)),
-      mResourceManagerService(new ResourceManagerServiceProxy(pid)),
       mBatteryStatNotified(false),
       mIsVideo(false),
       mVideoWidth(0),
@@ -537,6 +535,8 @@
     } else {
         mUid = uid;
     }
+    mResourceManagerClient = new ResourceManagerClient(this);
+    mResourceManagerService = new ResourceManagerServiceProxy(pid, mUid);
 
     initAnalyticsItem();
 }
@@ -1977,6 +1977,11 @@
                     if (mIsVideo) {
                         // audio codec is currently ignored.
                         addResource(resourceType, MediaResource::kVideoCodec, 1);
+                        // TODO: track battery on/off by actual queueing/dequeueing
+                        // For now, keep existing behavior and request battery on/off
+                        // together with codec init/uninit. We'll improve the tracking
+                        // later by adding/removing this based on queue/dequeue timing.
+                        addResource(MediaResource::kBattery, MediaResource::kVideoCodec, 1);
                     }
 
                     (new AMessage)->postReply(mReplyID);
@@ -3126,8 +3131,6 @@
     mState = newState;
 
     cancelPendingDequeueOperations();
-
-    updateBatteryStat();
 }
 
 void MediaCodec::returnBuffersToCodec(bool isReclaim) {
@@ -3631,20 +3634,6 @@
     return OK;
 }
 
-void MediaCodec::updateBatteryStat() {
-    if (!mIsVideo) {
-        return;
-    }
-
-    if (mState == CONFIGURED && !mBatteryStatNotified) {
-        BatteryNotifier::getInstance().noteStartVideo(mUid);
-        mBatteryStatNotified = true;
-    } else if (mState == UNINITIALIZED && mBatteryStatNotified) {
-        BatteryNotifier::getInstance().noteStopVideo(mUid);
-        mBatteryStatNotified = false;
-    }
-}
-
 std::string MediaCodec::stateString(State state) {
     const char *rval = NULL;
     char rawbuffer[16]; // room for "%d"
diff --git a/media/libstagefright/MediaMuxer.cpp b/media/libstagefright/MediaMuxer.cpp
index 9ba2add..7ebdb1a 100644
--- a/media/libstagefright/MediaMuxer.cpp
+++ b/media/libstagefright/MediaMuxer.cpp
@@ -96,10 +96,18 @@
 
     sp<MediaAdapter> newTrack = new MediaAdapter(trackMeta);
     status_t result = mWriter->addSource(newTrack);
-    if (result == OK) {
-        return mTrackList.add(newTrack);
+    if (result != OK) {
+        return -1;
     }
-    return -1;
+    float captureFps = -1.0;
+    if (format->findAsFloat("time-lapse-fps", &captureFps)) {
+        ALOGV("addTrack() time-lapse-fps: %f", captureFps);
+        result = mWriter->setCaptureRate(captureFps);
+        if (result != OK) {
+            ALOGW("addTrack() setCaptureRate failed :%d", result);
+        }
+    }
+    return mTrackList.add(newTrack);
 }
 
 status_t MediaMuxer::setOrientationHint(int degrees) {
diff --git a/media/libstagefright/codecs/m4v_h263/dec/src/vop.cpp b/media/libstagefright/codecs/m4v_h263/dec/src/vop.cpp
index f18f789..679b091 100644
--- a/media/libstagefright/codecs/m4v_h263/dec/src/vop.cpp
+++ b/media/libstagefright/codecs/m4v_h263/dec/src/vop.cpp
@@ -1355,6 +1355,14 @@
             int tmpHeight = (tmpDisplayHeight + 15) & -16;
             int tmpWidth = (tmpDisplayWidth + 15) & -16;
 
+            if (tmpWidth > video->width)
+            {
+                // while allowed by the spec, this decoder does not actually
+                // support an increase in size.
+                ALOGE("width increase not supported");
+                status = PV_FAIL;
+                goto return_point;
+            }
             if (tmpHeight * tmpWidth > video->size)
             {
                 // This is just possibly "b/37079296".
diff --git a/media/libstagefright/codecs/vorbis/dec/SoftVorbis.cpp b/media/libstagefright/codecs/vorbis/dec/SoftVorbis.cpp
index 8d0ea3a..08e20cc 100644
--- a/media/libstagefright/codecs/vorbis/dec/SoftVorbis.cpp
+++ b/media/libstagefright/codecs/vorbis/dec/SoftVorbis.cpp
@@ -290,7 +290,7 @@
 }
 
 bool SoftVorbis::isConfigured() const {
-    return mInputBufferCount >= 2;
+    return (mState != NULL && mVi != NULL);
 }
 
 static void makeBitReader(
diff --git a/media/libstagefright/include/media/stagefright/MediaCodec.h b/media/libstagefright/include/media/stagefright/MediaCodec.h
index 89cca63..0218a88 100644
--- a/media/libstagefright/include/media/stagefright/MediaCodec.h
+++ b/media/libstagefright/include/media/stagefright/MediaCodec.h
@@ -283,7 +283,7 @@
     };
 
     struct ResourceManagerServiceProxy : public IBinder::DeathRecipient {
-        ResourceManagerServiceProxy(pid_t pid);
+        ResourceManagerServiceProxy(pid_t pid, uid_t uid);
         ~ResourceManagerServiceProxy();
 
         void init();
@@ -304,6 +304,7 @@
         Mutex mLock;
         sp<IResourceManagerService> mService;
         pid_t mPid;
+        uid_t mUid;
     };
 
     State mState;
@@ -425,7 +426,6 @@
     status_t onSetParameters(const sp<AMessage> &params);
 
     status_t amendOutputFormatWithCodecSpecificData(const sp<MediaCodecBuffer> &buffer);
-    void updateBatteryStat();
     bool isExecuting() const;
 
     uint64_t getGraphicBufferSize();
diff --git a/media/libstagefright/include/media/stagefright/MediaWriter.h b/media/libstagefright/include/media/stagefright/MediaWriter.h
index 2c12a87..972ae1d 100644
--- a/media/libstagefright/include/media/stagefright/MediaWriter.h
+++ b/media/libstagefright/include/media/stagefright/MediaWriter.h
@@ -35,6 +35,10 @@
     virtual status_t start(MetaData *params = NULL) = 0;
     virtual status_t stop() = 0;
     virtual status_t pause() = 0;
+    virtual status_t setCaptureRate(float /* captureFps */) {
+        ALOGW("setCaptureRate unsupported");
+        return ERROR_UNSUPPORTED;
+    }
 
     virtual void setMaxFileSize(int64_t bytes) { mMaxFileSizeLimitBytes = bytes; }
     virtual void setMaxFileDuration(int64_t durationUs) { mMaxFileDurationLimitUs = durationUs; }
diff --git a/media/libstagefright/rtsp/ARTSPConnection.h b/media/libstagefright/rtsp/ARTSPConnection.h
index 8df2676..56b604d 100644
--- a/media/libstagefright/rtsp/ARTSPConnection.h
+++ b/media/libstagefright/rtsp/ARTSPConnection.h
@@ -46,6 +46,8 @@
             const char *url, AString *host, unsigned *port, AString *path,
             AString *user, AString *pass);
 
+    int getSocket() { return mSocket; }
+
 protected:
     virtual ~ARTSPConnection();
     virtual void onMessageReceived(const sp<AMessage> &msg);
diff --git a/media/libstagefright/rtsp/MyHandler.h b/media/libstagefright/rtsp/MyHandler.h
index 48bc8ce..85ffba2 100644
--- a/media/libstagefright/rtsp/MyHandler.h
+++ b/media/libstagefright/rtsp/MyHandler.h
@@ -257,6 +257,10 @@
         msg->post();
     }
 
+    sp<ARTSPConnection> getARTSPConnection() {
+      return mConn;
+    }
+
     static void addRR(const sp<ABuffer> &buf) {
         uint8_t *ptr = buf->data() + buf->size();
         ptr[0] = 0x80 | 0;
diff --git a/services/audioflinger/FastMixer.cpp b/services/audioflinger/FastMixer.cpp
index c5b9953..3eacc8c 100644
--- a/services/audioflinger/FastMixer.cpp
+++ b/services/audioflinger/FastMixer.cpp
@@ -105,13 +105,8 @@
     return mSQ.poll();
 }
 
-void FastMixer::setNBLogWriter(NBLog::Writer *logWriter)
+void FastMixer::setNBLogWriter(NBLog::Writer *logWriter __unused)
 {
-    // FIXME If mMixer is set or changed prior to this, we don't inform correctly.
-    //       Should cache logWriter and re-apply it at the assignment to mMixer.
-    if (mMixer != NULL) {
-        mMixer->setNBLogWriter(logWriter);
-    }
 }
 
 void FastMixer::onIdle()
diff --git a/services/audioflinger/FastThread.cpp b/services/audioflinger/FastThread.cpp
index 04b32c2..8b7a124 100644
--- a/services/audioflinger/FastThread.cpp
+++ b/services/audioflinger/FastThread.cpp
@@ -124,7 +124,7 @@
             mDumpState = next->mDumpState != NULL ? next->mDumpState : mDummyDumpState;
             tlNBLogWriter = next->mNBLogWriter != NULL ?
                     next->mNBLogWriter : mDummyNBLogWriter.get();
-            setNBLogWriter(tlNBLogWriter); // FastMixer informs its AudioMixer, FastCapture ignores
+            setNBLogWriter(tlNBLogWriter); // This is used for debugging only
 
             // We want to always have a valid reference to the previous (non-idle) state.
             // However, the state queue only guarantees access to current and previous states.
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index f81dacf..cf15045 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -5301,11 +5301,11 @@
         return false;
     }
     // Check validity as we don't call AudioMixer::create() here.
-    if (!AudioMixer::isValidFormat(format)) {
+    if (!mAudioMixer->isValidFormat(format)) {
         ALOGW("%s: invalid format: %#x", __func__, format);
         return false;
     }
-    if (!AudioMixer::isValidChannelMask(channelMask)) {
+    if (!mAudioMixer->isValidChannelMask(channelMask)) {
         ALOGW("%s: invalid channelMask: %#x", __func__, channelMask);
         return false;
     }
diff --git a/services/audiopolicy/common/managerdefinitions/Android.bp b/services/audiopolicy/common/managerdefinitions/Android.bp
index f02f3cf..ebfba83 100644
--- a/services/audiopolicy/common/managerdefinitions/Android.bp
+++ b/services/audiopolicy/common/managerdefinitions/Android.bp
@@ -3,7 +3,6 @@
 
     srcs: [
         "src/AudioCollections.cpp",
-        "src/AudioGain.cpp",
         "src/AudioInputDescriptor.cpp",
         "src/AudioOutputDescriptor.cpp",
         "src/AudioPatch.cpp",
@@ -21,6 +20,7 @@
         "src/TypeConverter.cpp",
     ],
     shared_libs: [
+        "libaudiofoundation",
         "libcutils",
         "libhidlbase",
         "liblog",
@@ -28,7 +28,10 @@
         "libutils",
         "libxml2",
     ],
-    export_shared_lib_headers: ["libmedia"],
+    export_shared_lib_headers: [
+        "libaudiofoundation",
+        "libmedia",
+    ],
     static_libs: [
         "libaudioutils",
     ],
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioPolicyConfig.h b/services/audiopolicy/common/managerdefinitions/include/AudioPolicyConfig.h
index 0776a8d..31c5041 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioPolicyConfig.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioPolicyConfig.h
@@ -19,7 +19,6 @@
 #include <unordered_map>
 #include <unordered_set>
 
-#include <AudioGain.h>
 #include <AudioPort.h>
 #include <AudioPatch.h>
 #include <DeviceDescriptor.h>
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioPort.h b/services/audiopolicy/common/managerdefinitions/include/AudioPort.h
index d906f11..641cebf 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioPort.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioPort.h
@@ -18,8 +18,8 @@
 
 #include "AudioCollections.h"
 #include "AudioProfile.h"
-#include "AudioGain.h"
 #include "HandleGenerator.h"
+#include <media/AudioGain.h>
 #include <utils/String8.h>
 #include <utils/Vector.h>
 #include <utils/RefBase.h>
@@ -110,8 +110,8 @@
     // Used to select an audio HAL output stream with a sample format providing the
     // less degradation for a given AudioTrack sample format.
     static bool isBetterFormatMatch(audio_format_t newFormat,
-                                        audio_format_t currentFormat,
-                                        audio_format_t targetFormat);
+                                    audio_format_t currentFormat,
+                                    audio_format_t targetFormat);
     static uint32_t formatDistance(audio_format_t format1,
                                    audio_format_t format2);
     static const uint32_t kFormatDistanceMax = 4;
@@ -143,8 +143,9 @@
     AudioGains mGains; // gain controllers
 
 private:
-    void pickChannelMask(audio_channel_mask_t &channelMask, const ChannelsVector &channelMasks) const;
-    void pickSamplingRate(uint32_t &rate,const SampleRateVector &samplingRates) const;
+    void pickChannelMask(audio_channel_mask_t &channelMask,
+                         const ChannelMaskSet &channelMasks) const;
+    void pickSamplingRate(uint32_t &rate, const SampleRateSet &samplingRates) const;
 
     sp<HwModule> mModule;                 // audio HW module exposing this I/O stream
     String8  mName;
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioProfile.h b/services/audiopolicy/common/managerdefinitions/include/AudioProfile.h
index b588d57..ea56729 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioProfile.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioProfile.h
@@ -16,55 +16,15 @@
 
 #pragma once
 
-#include <vector>
-
+#include <media/AudioContainers.h>
 #include <system/audio.h>
 #include <utils/RefBase.h>
-#include <utils/SortedVector.h>
 #include <utils/String8.h>
 
 #include "policy.h"
 
 namespace android {
 
-typedef SortedVector<uint32_t> SampleRateVector;
-typedef Vector<audio_format_t> FormatVector;
-
-template <typename T>
-bool operator== (const SortedVector<T> &left, const SortedVector<T> &right)
-{
-    if (left.size() != right.size()) {
-        return false;
-    }
-    for (size_t index = 0; index < right.size(); index++) {
-        if (left[index] != right[index]) {
-            return false;
-        }
-    }
-    return true;
-}
-
-template <typename T>
-bool operator!= (const SortedVector<T> &left, const SortedVector<T> &right)
-{
-    return !(left == right);
-}
-
-class ChannelsVector : public SortedVector<audio_channel_mask_t>
-{
-public:
-    ChannelsVector() = default;
-    ChannelsVector(const ChannelsVector&) = default;
-    ChannelsVector(const SortedVector<audio_channel_mask_t>& sv) :
-            SortedVector<audio_channel_mask_t>(sv) {}
-    ChannelsVector& operator=(const ChannelsVector&) = default;
-
-    // Applies audio_channel_mask_out_to_in to all elements and returns the result.
-    ChannelsVector asInMask() const;
-    // Applies audio_channel_mask_in_to_out to all elements and returns the result.
-    ChannelsVector asOutMask() const;
-};
-
 class AudioProfile : public virtual RefBase
 {
 public:
@@ -72,22 +32,22 @@
 
     AudioProfile(audio_format_t format, audio_channel_mask_t channelMasks, uint32_t samplingRate);
     AudioProfile(audio_format_t format,
-                 const ChannelsVector &channelMasks,
-                 const SampleRateVector &samplingRateCollection);
+                 const ChannelMaskSet &channelMasks,
+                 const SampleRateSet &samplingRateCollection);
 
     audio_format_t getFormat() const { return mFormat; }
-    const ChannelsVector &getChannels() const { return mChannelMasks; }
-    const SampleRateVector &getSampleRates() const { return mSamplingRates; }
-    void setChannels(const ChannelsVector &channelMasks);
-    void setSampleRates(const SampleRateVector &sampleRates);
+    const ChannelMaskSet &getChannels() const { return mChannelMasks; }
+    const SampleRateSet &getSampleRates() const { return mSamplingRates; }
+    void setChannels(const ChannelMaskSet &channelMasks);
+    void setSampleRates(const SampleRateSet &sampleRates);
 
     void clear();
     bool isValid() const { return hasValidFormat() && hasValidRates() && hasValidChannels(); }
     bool supportsChannels(audio_channel_mask_t channels) const
     {
-        return mChannelMasks.indexOf(channels) >= 0;
+        return mChannelMasks.count(channels) != 0;
     }
-    bool supportsRate(uint32_t rate) const { return mSamplingRates.indexOf(rate) >= 0; }
+    bool supportsRate(uint32_t rate) const { return mSamplingRates.count(rate) != 0; }
 
     status_t checkExact(uint32_t rate, audio_channel_mask_t channels, audio_format_t format) const;
     status_t checkCompatibleChannelMask(audio_channel_mask_t channelMask,
@@ -98,8 +58,8 @@
                                          uint32_t &updatedSamplingRate) const;
 
     bool hasValidFormat() const { return mFormat != AUDIO_FORMAT_DEFAULT; }
-    bool hasValidRates() const { return !mSamplingRates.isEmpty(); }
-    bool hasValidChannels() const { return !mChannelMasks.isEmpty(); }
+    bool hasValidRates() const { return !mSamplingRates.empty(); }
+    bool hasValidChannels() const { return !mChannelMasks.empty(); }
 
     void setDynamicChannels(bool dynamic) { mIsDynamicChannels = dynamic; }
     bool isDynamicChannels() const { return mIsDynamicChannels; }
@@ -117,8 +77,8 @@
 private:
     String8  mName;
     audio_format_t mFormat;
-    ChannelsVector mChannelMasks;
-    SampleRateVector mSamplingRates;
+    ChannelMaskSet mChannelMasks;
+    SampleRateSet mSamplingRates;
 
     bool mIsDynamicFormat = false;
     bool mIsDynamicChannels = false;
@@ -126,13 +86,16 @@
 };
 
 
-class AudioProfileVector : public Vector<sp<AudioProfile> >
+class AudioProfileVector : public std::vector<sp<AudioProfile> >
 {
 public:
     ssize_t add(const sp<AudioProfile> &profile);
     // This API is intended to be used by the policy manager once retrieving capabilities
     // for a profile with dynamic format, rate and channels attributes
     ssize_t addProfileFromHal(const sp<AudioProfile> &profileToAdd);
+    void appendProfiles(const AudioProfileVector& audioProfiles) {
+        insert(end(), audioProfiles.begin(), audioProfiles.end());
+    }
 
     status_t checkExactProfile(uint32_t samplingRate, audio_channel_mask_t channelMask,
                                audio_format_t format) const;
@@ -169,10 +132,8 @@
 
 private:
     sp<AudioProfile> getProfileFor(audio_format_t format) const;
-    void setSampleRatesFor(const SampleRateVector &sampleRates, audio_format_t format);
-    void setChannelsFor(const ChannelsVector &channelMasks, audio_format_t format);
-
-    static int compareFormats(const sp<AudioProfile> *profile1, const sp<AudioProfile> *profile2);
+    void setSampleRatesFor(const SampleRateSet &sampleRates, audio_format_t format);
+    void setChannelsFor(const ChannelMaskSet &channelMasks, audio_format_t format);
 };
 
 bool operator == (const AudioProfile &left, const AudioProfile &right);
diff --git a/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
index 33e506f..c7c1fee 100644
--- a/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
@@ -116,6 +116,13 @@
     DeviceVector getDevicesFromHwModule(audio_module_handle_t moduleHandle) const;
     audio_devices_t getDeviceTypesFromHwModule(audio_module_handle_t moduleHandle) const;
 
+    DeviceVector getFirstDevicesFromTypes(std::vector<audio_devices_t> orderedTypes) const;
+    sp<DeviceDescriptor> getFirstExistingDevice(std::vector<audio_devices_t> orderedTypes) 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);
+
     bool contains(const sp<DeviceDescriptor>& item) const { return indexOf(item) >= 0; }
 
     /**
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioCollections.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioCollections.cpp
index c90a582..e8cf485 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioCollections.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioCollections.cpp
@@ -21,7 +21,6 @@
 #include "AudioPort.h"
 #include "AudioRoute.h"
 #include "HwModule.h"
-#include "AudioGain.h"
 
 namespace android {
 
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioGain.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioGain.cpp
deleted file mode 100644
index 2725870..0000000
--- a/services/audiopolicy/common/managerdefinitions/src/AudioGain.cpp
+++ /dev/null
@@ -1,114 +0,0 @@
-/*
- * Copyright (C) 2015 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_TAG "APM::AudioGain"
-//#define LOG_NDEBUG 0
-
-//#define VERY_VERBOSE_LOGGING
-#ifdef VERY_VERBOSE_LOGGING
-#define ALOGVV ALOGV
-#else
-#define ALOGVV(a...) do { } while(0)
-#endif
-
-#include "AudioGain.h"
-#include <utils/Log.h>
-#include <utils/String8.h>
-
-#include <math.h>
-
-namespace android {
-
-AudioGain::AudioGain(int index, bool useInChannelMask)
-{
-    mIndex = index;
-    mUseInChannelMask = useInChannelMask;
-    memset(&mGain, 0, sizeof(struct audio_gain));
-}
-
-void AudioGain::getDefaultConfig(struct audio_gain_config *config)
-{
-    config->index = mIndex;
-    config->mode = mGain.mode;
-    config->channel_mask = mGain.channel_mask;
-    if ((mGain.mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) {
-        config->values[0] = mGain.default_value;
-    } else {
-        uint32_t numValues;
-        if (mUseInChannelMask) {
-            numValues = audio_channel_count_from_in_mask(mGain.channel_mask);
-        } else {
-            numValues = audio_channel_count_from_out_mask(mGain.channel_mask);
-        }
-        for (size_t i = 0; i < numValues; i++) {
-            config->values[i] = mGain.default_value;
-        }
-    }
-    if ((mGain.mode & AUDIO_GAIN_MODE_RAMP) == AUDIO_GAIN_MODE_RAMP) {
-        config->ramp_duration_ms = mGain.min_ramp_ms;
-    }
-}
-
-status_t AudioGain::checkConfig(const struct audio_gain_config *config)
-{
-    if ((config->mode & ~mGain.mode) != 0) {
-        return BAD_VALUE;
-    }
-    if ((config->mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) {
-        if ((config->values[0] < mGain.min_value) ||
-                    (config->values[0] > mGain.max_value)) {
-            return BAD_VALUE;
-        }
-    } else {
-        if ((config->channel_mask & ~mGain.channel_mask) != 0) {
-            return BAD_VALUE;
-        }
-        uint32_t numValues;
-        if (mUseInChannelMask) {
-            numValues = audio_channel_count_from_in_mask(config->channel_mask);
-        } else {
-            numValues = audio_channel_count_from_out_mask(config->channel_mask);
-        }
-        for (size_t i = 0; i < numValues; i++) {
-            if ((config->values[i] < mGain.min_value) ||
-                    (config->values[i] > mGain.max_value)) {
-                return BAD_VALUE;
-            }
-        }
-    }
-    if ((config->mode & AUDIO_GAIN_MODE_RAMP) == AUDIO_GAIN_MODE_RAMP) {
-        if ((config->ramp_duration_ms < mGain.min_ramp_ms) ||
-                    (config->ramp_duration_ms > mGain.max_ramp_ms)) {
-            return BAD_VALUE;
-        }
-    }
-    return NO_ERROR;
-}
-
-void AudioGain::dump(String8 *dst, int spaces, int index) const
-{
-    dst->appendFormat("%*sGain %d:\n", spaces, "", index+1);
-    dst->appendFormat("%*s- mode: %08x\n", spaces, "", mGain.mode);
-    dst->appendFormat("%*s- channel_mask: %08x\n", spaces, "", mGain.channel_mask);
-    dst->appendFormat("%*s- min_value: %d mB\n", spaces, "", mGain.min_value);
-    dst->appendFormat("%*s- max_value: %d mB\n", spaces, "", mGain.max_value);
-    dst->appendFormat("%*s- default_value: %d mB\n", spaces, "", mGain.default_value);
-    dst->appendFormat("%*s- step_value: %d mB\n", spaces, "", mGain.step_value);
-    dst->appendFormat("%*s- min_ramp_ms: %d ms\n", spaces, "", mGain.min_ramp_ms);
-    dst->appendFormat("%*s- max_ramp_ms: %d ms\n", spaces, "", mGain.max_ramp_ms);
-}
-
-} // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp
index a096e8f..a9b87e3 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp
@@ -22,7 +22,6 @@
 #include <policy.h>
 #include <AudioPolicyInterface.h>
 #include "AudioInputDescriptor.h"
-#include "AudioGain.h"
 #include "AudioPolicyMix.h"
 #include "HwModule.h"
 
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
index 8a60cf2..49524b0 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
@@ -21,10 +21,10 @@
 #include "AudioOutputDescriptor.h"
 #include "AudioPolicyMix.h"
 #include "IOProfile.h"
-#include "AudioGain.h"
 #include "Volume.h"
 #include "HwModule.h"
 #include "TypeConverter.h"
+#include <media/AudioGain.h>
 #include <media/AudioParameter.h>
 #include <media/AudioPolicy.h>
 
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioPatch.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioPatch.cpp
index 3a4db90..bf0cc94 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioPatch.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioPatch.cpp
@@ -18,7 +18,6 @@
 //#define LOG_NDEBUG 0
 
 #include "AudioPatch.h"
-#include "AudioGain.h"
 #include "TypeConverter.h"
 
 #include <log/log.h>
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp
index c42923a..0221348 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp
@@ -22,7 +22,6 @@
 #include "HwModule.h"
 #include "AudioPort.h"
 #include "IOProfile.h"
-#include "AudioGain.h"
 #include <AudioOutputDescriptor.h>
 
 namespace android {
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp
index c11490a..ff32284 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp
@@ -19,7 +19,6 @@
 #include "TypeConverter.h"
 #include "AudioPort.h"
 #include "HwModule.h"
-#include "AudioGain.h"
 #include <policy.h>
 
 #ifndef ARRAY_SIZE
@@ -65,30 +64,19 @@
 {
     // TODO: update this function once audio_port structure reflects the new profile definition.
     // For compatibility reason: flatening the AudioProfile into audio_port structure.
-    SortedVector<audio_format_t> flatenedFormats;
-    SampleRateVector flatenedRates;
-    ChannelsVector flatenedChannels;
+    FormatSet flatenedFormats;
+    SampleRateSet flatenedRates;
+    ChannelMaskSet flatenedChannels;
     for (const auto& profile : mProfiles) {
         if (profile->isValid()) {
             audio_format_t formatToExport = profile->getFormat();
-            const SampleRateVector &ratesToExport = profile->getSampleRates();
-            const ChannelsVector &channelsToExport = profile->getChannels();
+            const SampleRateSet &ratesToExport = profile->getSampleRates();
+            const ChannelMaskSet &channelsToExport = profile->getChannels();
 
-            if (flatenedFormats.indexOf(formatToExport) < 0) {
-                flatenedFormats.add(formatToExport);
-            }
-            for (size_t rateIndex = 0; rateIndex < ratesToExport.size(); rateIndex++) {
-                uint32_t rate = ratesToExport[rateIndex];
-                if (flatenedRates.indexOf(rate) < 0) {
-                    flatenedRates.add(rate);
-                }
-            }
-            for (size_t chanIndex = 0; chanIndex < channelsToExport.size(); chanIndex++) {
-                audio_channel_mask_t channels = channelsToExport[chanIndex];
-                if (flatenedChannels.indexOf(channels) < 0) {
-                    flatenedChannels.add(channels);
-                }
-            }
+            flatenedFormats.insert(formatToExport);
+            flatenedRates.insert(ratesToExport.begin(), ratesToExport.end());
+            flatenedChannels.insert(channelsToExport.begin(), channelsToExport.end());
+
             if (flatenedRates.size() > AUDIO_PORT_MAX_SAMPLING_RATES ||
                     flatenedChannels.size() > AUDIO_PORT_MAX_CHANNEL_MASKS ||
                     flatenedFormats.size() > AUDIO_PORT_MAX_FORMATS) {
@@ -103,23 +91,16 @@
     port->num_sample_rates = flatenedRates.size();
     port->num_channel_masks = flatenedChannels.size();
     port->num_formats = flatenedFormats.size();
-    for (size_t i = 0; i < flatenedRates.size(); i++) {
-        port->sample_rates[i] = flatenedRates[i];
-    }
-    for (size_t i = 0; i < flatenedChannels.size(); i++) {
-        port->channel_masks[i] = flatenedChannels[i];
-    }
-    for (size_t i = 0; i < flatenedFormats.size(); i++) {
-        port->formats[i] = flatenedFormats[i];
-    }
+    std::copy(flatenedRates.begin(), flatenedRates.end(), port->sample_rates);
+    std::copy(flatenedChannels.begin(), flatenedChannels.end(), port->channel_masks);
+    std::copy(flatenedFormats.begin(), flatenedFormats.end(), port->formats);
 
     ALOGV("AudioPort::toAudioPort() num gains %zu", mGains.size());
 
-    uint32_t i;
-    for (i = 0; i < mGains.size() && i < AUDIO_PORT_MAX_GAINS; i++) {
+    port->num_gains = std::min(mGains.size(), (size_t) AUDIO_PORT_MAX_GAINS);
+    for (size_t i = 0; i < port->num_gains; i++) {
         port->gains[i] = mGains[i]->getGain();
     }
-    port->num_gains = i;
 }
 
 void AudioPort::importAudioPort(const sp<AudioPort>& port, bool force __unused)
@@ -163,7 +144,7 @@
     return status;
 }
 
-void AudioPort::pickSamplingRate(uint32_t &pickedRate,const SampleRateVector &samplingRates) const
+void AudioPort::pickSamplingRate(uint32_t &pickedRate,const SampleRateSet &samplingRates) const
 {
     pickedRate = 0;
     // For direct outputs, pick minimum sampling rate: this helps ensuring that the
@@ -171,9 +152,9 @@
     // sink
     if (isDirectOutput()) {
         uint32_t samplingRate = UINT_MAX;
-        for (size_t i = 0; i < samplingRates.size(); i ++) {
-            if ((samplingRates[i] < samplingRate) && (samplingRates[i] > 0)) {
-                samplingRate = samplingRates[i];
+        for (const auto rate : samplingRates) {
+            if ((rate < samplingRate) && (rate > 0)) {
+                samplingRate = rate;
             }
         }
         pickedRate = (samplingRate == UINT_MAX) ? 0 : samplingRate;
@@ -189,16 +170,16 @@
         // TODO: should mSamplingRates[] be ordered in terms of our preference
         // and we return the first (and hence most preferred) match?  This is of concern if
         // we want to choose 96kHz over 192kHz for USB driver stability or resource constraints.
-        for (size_t i = 0; i < samplingRates.size(); i ++) {
-            if ((samplingRates[i] > pickedRate) && (samplingRates[i] <= maxRate)) {
-                pickedRate = samplingRates[i];
+        for (const auto rate : samplingRates) {
+            if ((rate > pickedRate) && (rate <= maxRate)) {
+                pickedRate = rate;
             }
         }
     }
 }
 
 void AudioPort::pickChannelMask(audio_channel_mask_t &pickedChannelMask,
-                                const ChannelsVector &channelMasks) const
+                                const ChannelMaskSet &channelMasks) const
 {
     pickedChannelMask = AUDIO_CHANNEL_NONE;
     // For direct outputs, pick minimum channel count: this helps ensuring that the
@@ -206,15 +187,15 @@
     // sink
     if (isDirectOutput()) {
         uint32_t channelCount = UINT_MAX;
-        for (size_t i = 0; i < channelMasks.size(); i ++) {
+        for (const auto channelMask : channelMasks) {
             uint32_t cnlCount;
             if (useInputChannelMask()) {
-                cnlCount = audio_channel_count_from_in_mask(channelMasks[i]);
+                cnlCount = audio_channel_count_from_in_mask(channelMask);
             } else {
-                cnlCount = audio_channel_count_from_out_mask(channelMasks[i]);
+                cnlCount = audio_channel_count_from_out_mask(channelMask);
             }
             if ((cnlCount < channelCount) && (cnlCount > 0)) {
-                pickedChannelMask = channelMasks[i];
+                pickedChannelMask = channelMask;
                 channelCount = cnlCount;
             }
         }
@@ -227,15 +208,15 @@
         if (mType != AUDIO_PORT_TYPE_MIX) {
             maxCount = UINT_MAX;
         }
-        for (size_t i = 0; i < channelMasks.size(); i ++) {
+        for (const auto channelMask : channelMasks) {
             uint32_t cnlCount;
             if (useInputChannelMask()) {
-                cnlCount = audio_channel_count_from_in_mask(channelMasks[i]);
+                cnlCount = audio_channel_count_from_in_mask(channelMask);
             } else {
-                cnlCount = audio_channel_count_from_out_mask(channelMasks[i]);
+                cnlCount = audio_channel_count_from_out_mask(channelMask);
             }
             if ((cnlCount > channelCount) && (cnlCount <= maxCount)) {
-                pickedChannelMask = channelMasks[i];
+                pickedChannelMask = channelMask;
                 channelCount = cnlCount;
             }
         }
@@ -366,7 +347,9 @@
         if (mGains.size() != 0) {
             dst->appendFormat("%*s- gains:\n", spaces, "");
             for (size_t i = 0; i < mGains.size(); i++) {
-                mGains[i]->dump(dst, spaces + 2, i);
+                std::string gainStr;
+                mGains[i]->dump(&gainStr, spaces + 2, i);
+                dst->append(gainStr.c_str());
             }
         }
     }
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioProfile.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioProfile.cpp
index 69d6b0c..d1082e8 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioProfile.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioProfile.cpp
@@ -21,10 +21,10 @@
 #define LOG_TAG "APM::AudioProfile"
 //#define LOG_NDEBUG 0
 
+#include <media/AudioContainers.h>
 #include <media/AudioResamplerPublic.h>
 #include <utils/Errors.h>
 
-#include "AudioGain.h"
 #include "AudioPort.h"
 #include "AudioProfile.h"
 #include "HwModule.h"
@@ -32,28 +32,6 @@
 
 namespace android {
 
-ChannelsVector ChannelsVector::asInMask() const
-{
-    ChannelsVector inMaskVector;
-    for (const auto& channel : *this) {
-        if (audio_channel_mask_out_to_in(channel) != AUDIO_CHANNEL_INVALID) {
-            inMaskVector.add(audio_channel_mask_out_to_in(channel));
-        }
-    }
-    return inMaskVector;
-}
-
-ChannelsVector ChannelsVector::asOutMask() const
-{
-    ChannelsVector outMaskVector;
-    for (const auto& channel : *this) {
-        if (audio_channel_mask_in_to_out(channel) != AUDIO_CHANNEL_INVALID) {
-            outMaskVector.add(audio_channel_mask_in_to_out(channel));
-        }
-    }
-    return outMaskVector;
-}
-
 bool operator == (const AudioProfile &left, const AudioProfile &compareTo)
 {
     return (left.getFormat() == compareTo.getFormat()) &&
@@ -64,7 +42,7 @@
 static AudioProfile* createFullDynamicImpl()
 {
     AudioProfile* dynamicProfile = new AudioProfile(gDynamicFormat,
-            ChannelsVector(), SampleRateVector());
+            ChannelMaskSet(), SampleRateSet());
     dynamicProfile->setDynamicFormat(true);
     dynamicProfile->setDynamicChannels(true);
     dynamicProfile->setDynamicRate(true);
@@ -84,26 +62,26 @@
         mName(String8("")),
         mFormat(format)
 {
-    mChannelMasks.add(channelMasks);
-    mSamplingRates.add(samplingRate);
+    mChannelMasks.insert(channelMasks);
+    mSamplingRates.insert(samplingRate);
 }
 
 AudioProfile::AudioProfile(audio_format_t format,
-                           const ChannelsVector &channelMasks,
-                           const SampleRateVector &samplingRateCollection) :
+                           const ChannelMaskSet &channelMasks,
+                           const SampleRateSet &samplingRateCollection) :
         mName(String8("")),
         mFormat(format),
         mChannelMasks(channelMasks),
         mSamplingRates(samplingRateCollection) {}
 
-void AudioProfile::setChannels(const ChannelsVector &channelMasks)
+void AudioProfile::setChannels(const ChannelMaskSet &channelMasks)
 {
     if (mIsDynamicChannels) {
         mChannelMasks = channelMasks;
     }
 }
 
-void AudioProfile::setSampleRates(const SampleRateVector &sampleRates)
+void AudioProfile::setSampleRates(const SampleRateSet &sampleRates)
 {
     if (mIsDynamicRate) {
         mSamplingRates = sampleRates;
@@ -136,7 +114,7 @@
 {
     ALOG_ASSERT(samplingRate > 0);
 
-    if (mSamplingRates.isEmpty()) {
+    if (mSamplingRates.empty()) {
         updatedSamplingRate = samplingRate;
         return NO_ERROR;
     }
@@ -144,19 +122,18 @@
     // Search for the closest supported sampling rate that is above (preferred)
     // or below (acceptable) the desired sampling rate, within a permitted ratio.
     // The sampling rates are sorted in ascending order.
-    size_t orderOfDesiredRate = mSamplingRates.orderOf(samplingRate);
+    auto desiredRate = mSamplingRates.lower_bound(samplingRate);
 
     // Prefer to down-sample from a higher sampling rate, as we get the desired frequency spectrum.
-    if (orderOfDesiredRate < mSamplingRates.size()) {
-        uint32_t candidate = mSamplingRates[orderOfDesiredRate];
-        if (candidate / AUDIO_RESAMPLER_DOWN_RATIO_MAX <= samplingRate) {
-            updatedSamplingRate = candidate;
+    if (desiredRate != mSamplingRates.end()) {
+        if (*desiredRate / AUDIO_RESAMPLER_DOWN_RATIO_MAX <= samplingRate) {
+            updatedSamplingRate = *desiredRate;
             return NO_ERROR;
         }
     }
     // But if we have to up-sample from a lower sampling rate, that's OK.
-    if (orderOfDesiredRate != 0) {
-        uint32_t candidate = mSamplingRates[orderOfDesiredRate - 1];
+    if (desiredRate != mSamplingRates.begin()) {
+        uint32_t candidate = *(--desiredRate);
         if (candidate * AUDIO_RESAMPLER_UP_RATIO_MAX >= samplingRate) {
             updatedSamplingRate = candidate;
             return NO_ERROR;
@@ -171,7 +148,7 @@
                                                   audio_port_type_t portType,
                                                   audio_port_role_t portRole) const
 {
-    if (mChannelMasks.isEmpty()) {
+    if (mChannelMasks.empty()) {
         updatedChannelMask = channelMask;
         return NO_ERROR;
     }
@@ -180,8 +157,7 @@
             == AUDIO_CHANNEL_REPRESENTATION_INDEX;
     const uint32_t channelCount = audio_channel_count_from_in_mask(channelMask);
     int bestMatch = 0;
-    for (size_t i = 0; i < mChannelMasks.size(); i ++) {
-        audio_channel_mask_t supported = mChannelMasks[i];
+    for (const auto &supported : mChannelMasks) {
         if (supported == channelMask) {
             // Exact matches always taken.
             updatedChannelMask = channelMask;
@@ -271,20 +247,20 @@
     if (FormatConverter::toString(mFormat, formatLiteral)) {
         dst->appendFormat("%*s- format: %s\n", spaces, "", formatLiteral.c_str());
     }
-    if (!mSamplingRates.isEmpty()) {
+    if (!mSamplingRates.empty()) {
         dst->appendFormat("%*s- sampling rates:", spaces, "");
-        for (size_t i = 0; i < mSamplingRates.size(); i++) {
-            dst->appendFormat("%d", mSamplingRates[i]);
-            dst->append(i == (mSamplingRates.size() - 1) ? "" : ", ");
+        for (auto it = mSamplingRates.begin(); it != mSamplingRates.end();) {
+            dst->appendFormat("%d", *it);
+            dst->append(++it == mSamplingRates.end() ? "" : ", ");
         }
         dst->append("\n");
     }
 
-    if (!mChannelMasks.isEmpty()) {
+    if (!mChannelMasks.empty()) {
         dst->appendFormat("%*s- channel masks:", spaces, "");
-        for (size_t i = 0; i < mChannelMasks.size(); i++) {
-            dst->appendFormat("0x%04x", mChannelMasks[i]);
-            dst->append(i == (mChannelMasks.size() - 1) ? "" : ", ");
+        for (auto it = mChannelMasks.begin(); it != mChannelMasks.end();) {
+            dst->appendFormat("0x%04x", *it);
+            dst->append(++it == mChannelMasks.end() ? "" : ", ");
         }
         dst->append("\n");
     }
@@ -292,13 +268,14 @@
 
 ssize_t AudioProfileVector::add(const sp<AudioProfile> &profile)
 {
-    ssize_t index = Vector::add(profile);
+    ssize_t index = size();
+    push_back(profile);
     // we sort from worst to best, so that AUDIO_FORMAT_DEFAULT is always the first entry.
-    // TODO: compareFormats could be a lambda to convert between pointer-to-format to format:
-    // [](const audio_format_t *format1, const audio_format_t *format2) {
-    //     return compareFormats(*format1, *format2);
-    // }
-    sort(compareFormats);
+    std::sort(begin(), end(),
+            [](const sp<AudioProfile> & a, const sp<AudioProfile> & b)
+            {
+                return AudioPort::compareFormats(a->getFormat(), b->getFormat()) < 0;
+            });
     return index;
 }
 
@@ -310,7 +287,7 @@
     }
     if (!profileToAdd->hasValidChannels() && !profileToAdd->hasValidRates()) {
         FormatVector formats;
-        formats.add(profileToAdd->getFormat());
+        formats.push_back(profileToAdd->getFormat());
         setFormats(FormatVector(formats));
         return 0;
     }
@@ -324,7 +301,7 @@
     }
     // Go through the list of profile to avoid duplicates
     for (size_t profileIndex = 0; profileIndex < size(); profileIndex++) {
-        const sp<AudioProfile> &profile = itemAt(profileIndex);
+        const sp<AudioProfile> &profile = at(profileIndex);
         if (profile->isValid() && profile == profileToAdd) {
             // Nothing to do
             return profileIndex;
@@ -338,7 +315,7 @@
                                                audio_channel_mask_t channelMask,
                                                audio_format_t format) const
 {
-    if (isEmpty()) {
+    if (empty()) {
         return NO_ERROR;
     }
 
@@ -356,7 +333,7 @@
                                                     audio_port_type_t portType,
                                                     audio_port_role_t portRole) const
 {
-    if (isEmpty()) {
+    if (empty()) {
         return NO_ERROR;
     }
 
@@ -366,7 +343,7 @@
 
     // iterate from best format to worst format (reverse order)
     for (ssize_t i = size() - 1; i >= 0 ; --i) {
-        const sp<AudioProfile> profile = itemAt(i);
+        const sp<AudioProfile> profile = at(i);
         audio_format_t formatToCompare = profile->getFormat();
         if (formatToCompare == format ||
                 (checkInexact
@@ -392,13 +369,13 @@
 
 void AudioProfileVector::clearProfiles()
 {
-    for (size_t i = size(); i != 0; ) {
-        sp<AudioProfile> profile = itemAt(--i);
-        if (profile->isDynamicFormat() && profile->hasValidFormat()) {
-            removeAt(i);
-            continue;
+    for (auto it = begin(); it != end();) {
+        if ((*it)->isDynamicFormat() && (*it)->hasValidFormat()) {
+            it = erase(it);
+        } else {
+            (*it)->clear();
+            ++it;
         }
-        profile->clear();
     }
 }
 
@@ -449,16 +426,16 @@
         if (inputProfile == nullptr || outputProfile == nullptr) {
             continue;
         }
-        auto channels = intersectFilterAndOrder(inputProfile->getChannels().asOutMask(),
+        auto channels = intersectFilterAndOrder(asOutMask(inputProfile->getChannels()),
                 outputProfile->getChannels(), preferredOutputChannels);
         if (channels.empty()) {
             continue;
         }
         auto sampleRates = preferHigherSamplingRates ?
                 intersectAndOrder(inputProfile->getSampleRates(), outputProfile->getSampleRates(),
-                        std::greater<typename SampleRateVector::value_type>()) :
+                        std::greater<typename SampleRateSet::value_type>()) :
                 intersectAndOrder(inputProfile->getSampleRates(), outputProfile->getSampleRates(),
-                        std::less<typename SampleRateVector::value_type>());
+                        std::less<typename SampleRateSet::value_type>());
         if (sampleRates.empty()) {
             continue;
         }
@@ -474,30 +451,30 @@
 
 sp<AudioProfile> AudioProfileVector::getFirstValidProfile() const
 {
-    for (size_t i = 0; i < size(); i++) {
-        if (itemAt(i)->isValid()) {
-            return itemAt(i);
+    for (const auto &profile : *this) {
+        if (profile->isValid()) {
+            return profile;
         }
     }
-    return 0;
+    return nullptr;
 }
 
 sp<AudioProfile> AudioProfileVector::getFirstValidProfileFor(audio_format_t format) const
 {
-    for (size_t i = 0; i < size(); i++) {
-        if (itemAt(i)->isValid() && itemAt(i)->getFormat() == format) {
-            return itemAt(i);
+    for (const auto &profile : *this) {
+        if (profile->isValid() && profile->getFormat() == format) {
+            return profile;
         }
     }
-    return 0;
+    return nullptr;
 }
 
 FormatVector AudioProfileVector::getSupportedFormats() const
 {
     FormatVector supportedFormats;
-    for (size_t i = 0; i < size(); i++) {
-        if (itemAt(i)->hasValidFormat()) {
-            supportedFormats.add(itemAt(i)->getFormat());
+    for (const auto &profile : *this) {
+        if (profile->hasValidFormat()) {
+            supportedFormats.push_back(profile->getFormat());
         }
     }
     return supportedFormats;
@@ -505,8 +482,7 @@
 
 bool AudioProfileVector::hasDynamicChannelsFor(audio_format_t format) const
 {
-    for (size_t i = 0; i < size(); i++) {
-        sp<AudioProfile> profile = itemAt(i);
+    for (const auto &profile : *this) {
         if (profile->getFormat() == format && profile->isDynamicChannels()) {
             return true;
         }
@@ -516,8 +492,8 @@
 
 bool AudioProfileVector::hasDynamicProfile() const
 {
-    for (size_t i = 0; i < size(); i++) {
-        if (itemAt(i)->isDynamic()) {
+    for (const auto &profile : *this) {
+        if (profile->isDynamic()) {
             return true;
         }
     }
@@ -526,8 +502,7 @@
 
 bool AudioProfileVector::hasDynamicRateFor(audio_format_t format) const
 {
-    for (size_t i = 0; i < size(); i++) {
-        sp<AudioProfile> profile = itemAt(i);
+    for (const auto &profile : *this) {
         if (profile->getFormat() == format && profile->isDynamicRate()) {
             return true;
         }
@@ -542,8 +517,8 @@
     if (dynamicFormatProfile == 0) {
         return;
     }
-    for (size_t i = 0; i < formats.size(); i++) {
-        sp<AudioProfile> profile = new AudioProfile(formats[i],
+    for (const auto &format : formats) {
+        sp<AudioProfile> profile = new AudioProfile(format,
                 dynamicFormatProfile->getChannels(),
                 dynamicFormatProfile->getSampleRates());
         profile->setDynamicFormat(true);
@@ -558,25 +533,24 @@
     dst->appendFormat("%*s- Profiles:\n", spaces, "");
     for (size_t i = 0; i < size(); i++) {
         dst->appendFormat("%*sProfile %zu:", spaces + 4, "", i);
-        itemAt(i)->dump(dst, spaces + 8);
+        at(i)->dump(dst, spaces + 8);
     }
 }
 
 sp<AudioProfile> AudioProfileVector::getProfileFor(audio_format_t format) const
 {
-    for (size_t i = 0; i < size(); i++) {
-        if (itemAt(i)->getFormat() == format) {
-            return itemAt(i);
+    for (const auto &profile : *this) {
+        if (profile->getFormat() == format) {
+            return profile;
         }
     }
-    return 0;
+    return nullptr;
 }
 
 void AudioProfileVector::setSampleRatesFor(
-        const SampleRateVector &sampleRates, audio_format_t format)
+        const SampleRateSet &sampleRates, audio_format_t format)
 {
-    for (size_t i = 0; i < size(); i++) {
-        sp<AudioProfile> profile = itemAt(i);
+    for (const auto &profile : *this) {
         if (profile->getFormat() == format && profile->isDynamicRate()) {
             if (profile->hasValidRates()) {
                 // Need to create a new profile with same format
@@ -592,10 +566,9 @@
     }
 }
 
-void AudioProfileVector::setChannelsFor(const ChannelsVector &channelMasks, audio_format_t format)
+void AudioProfileVector::setChannelsFor(const ChannelMaskSet &channelMasks, audio_format_t format)
 {
-    for (size_t i = 0; i < size(); i++) {
-        sp<AudioProfile> profile = itemAt(i);
+    for (const auto &profile : *this) {
         if (profile->getFormat() == format && profile->isDynamicChannels()) {
             if (profile->hasValidChannels()) {
                 // Need to create a new profile with same format
@@ -611,11 +584,4 @@
     }
 }
 
-// static
-int AudioProfileVector::compareFormats(const sp<AudioProfile> *profile1,
-                                       const sp<AudioProfile> *profile2)
-{
-    return AudioPort::compareFormats((*profile1)->getFormat(), (*profile2)->getFormat());
-}
-
 } // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioRoute.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioRoute.cpp
index 79f0919..92cbe4e 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioRoute.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioRoute.cpp
@@ -19,7 +19,6 @@
 
 #include "AudioRoute.h"
 #include "HwModule.h"
-#include "AudioGain.h"
 
 namespace android
 {
diff --git a/services/audiopolicy/common/managerdefinitions/src/ClientDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/ClientDescriptor.cpp
index ad07ab1..1dc7020 100644
--- a/services/audiopolicy/common/managerdefinitions/src/ClientDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/ClientDescriptor.cpp
@@ -21,7 +21,6 @@
 #include <utils/Log.h>
 #include <utils/String8.h>
 #include <TypeConverter.h>
-#include "AudioGain.h"
 #include "AudioOutputDescriptor.h"
 #include "AudioPatch.h"
 #include "ClientDescriptor.h"
diff --git a/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
index ecd5b34..e395caa 100644
--- a/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
@@ -22,7 +22,6 @@
 #include <set>
 #include "DeviceDescriptor.h"
 #include "TypeConverter.h"
-#include "AudioGain.h"
 #include "HwModule.h"
 
 namespace android {
@@ -48,9 +47,9 @@
      * For now, the workaround to remove AC3 and IEC61937 support on HDMI is to declare
      * something like 'encodedFormats="AUDIO_FORMAT_PCM_16_BIT"' on the HDMI devicePort.
      */
-    if (type == AUDIO_DEVICE_OUT_HDMI && mEncodedFormats.isEmpty()) {
-        mEncodedFormats.add(AUDIO_FORMAT_AC3);
-        mEncodedFormats.add(AUDIO_FORMAT_IEC61937);
+    if (type == AUDIO_DEVICE_OUT_HDMI && mEncodedFormats.empty()) {
+        mEncodedFormats.push_back(AUDIO_FORMAT_AC3);
+        mEncodedFormats.push_back(AUDIO_FORMAT_IEC61937);
     }
 }
 
@@ -97,7 +96,7 @@
     if (!device_has_encoding_capability(type())) {
         return true;
     }
-    if (mEncodedFormats.isEmpty()) {
+    if (mEncodedFormats.empty()) {
         return true;
     }
 
@@ -106,7 +105,7 @@
 
 bool DeviceDescriptor::supportsFormat(audio_format_t format)
 {
-    if (mEncodedFormats.isEmpty()) {
+    if (mEncodedFormats.empty()) {
         return true;
     }
 
@@ -256,7 +255,6 @@
         audio_devices_t curType = itemAt(i)->type() & ~AUDIO_DEVICE_BIT_IN;
         if ((isOutput == curIsOutput) && ((type & curType) != 0)) {
             devices.add(itemAt(i));
-            type &= ~curType;
             ALOGV("DeviceVector::%s() for type %08x found %p",
                     __func__, itemAt(i)->type(), itemAt(i).get());
         }
@@ -274,6 +272,38 @@
     return nullptr;
 }
 
+DeviceVector DeviceVector::getFirstDevicesFromTypes(
+        std::vector<audio_devices_t> orderedTypes) const
+{
+    DeviceVector devices;
+    for (auto deviceType : orderedTypes) {
+        if (!(devices = getDevicesFromTypeMask(deviceType)).isEmpty()) {
+            break;
+        }
+    }
+    return devices;
+}
+
+sp<DeviceDescriptor> DeviceVector::getFirstExistingDevice(
+        std::vector<audio_devices_t> orderedTypes) const {
+    sp<DeviceDescriptor> device;
+    for (auto deviceType : orderedTypes) {
+        if ((device = getDevice(deviceType, String8(""), AUDIO_FORMAT_DEFAULT)) != nullptr) {
+            break;
+        }
+    }
+    return device;
+}
+
+void DeviceVector::replaceDevicesByType(
+        audio_devices_t typeToRemove, const DeviceVector &devicesToAdd) {
+    DeviceVector devicesToRemove = getDevicesFromTypeMask(typeToRemove);
+    if (!devicesToRemove.isEmpty() && !devicesToAdd.isEmpty()) {
+        remove(devicesToRemove);
+        add(devicesToAdd);
+    }
+}
+
 void DeviceVector::dump(String8 *dst, const String8 &tag, int spaces, bool verbose) const
 {
     if (isEmpty()) {
diff --git a/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp b/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp
index 1f9b725..99e282e 100644
--- a/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp
@@ -19,7 +19,6 @@
 
 #include "HwModule.h"
 #include "IOProfile.h"
-#include "AudioGain.h"
 #include <policy.h>
 #include <system/audio.h>
 
diff --git a/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp b/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp
index fe2eaee..5662dcf 100644
--- a/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp
@@ -20,7 +20,6 @@
 #include <system/audio-base.h>
 #include "IOProfile.h"
 #include "HwModule.h"
-#include "AudioGain.h"
 #include "TypeConverter.h"
 
 namespace android {
diff --git a/services/audiopolicy/common/managerdefinitions/src/Serializer.cpp b/services/audiopolicy/common/managerdefinitions/src/Serializer.cpp
index 5f820c2..c699aa7 100644
--- a/services/audiopolicy/common/managerdefinitions/src/Serializer.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/Serializer.cpp
@@ -406,8 +406,8 @@
             samplingRatesFromString(samplingRates, ","));
 
     profile->setDynamicFormat(profile->getFormat() == gDynamicFormat);
-    profile->setDynamicChannels(profile->getChannels().isEmpty());
-    profile->setDynamicRate(profile->getSampleRates().isEmpty());
+    profile->setDynamicChannels(profile->getChannels().empty());
+    profile->setDynamicRate(profile->getSampleRates().empty());
 
     return profile;
 }
@@ -437,7 +437,7 @@
     if (status != NO_ERROR) {
         return Status::fromStatusT(status);
     }
-    if (profiles.isEmpty()) {
+    if (profiles.empty()) {
         profiles.add(AudioProfile::createFullDynamic());
     }
     mixPort->setAudioProfiles(profiles);
@@ -521,7 +521,7 @@
     if (status != NO_ERROR) {
         return Status::fromStatusT(status);
     }
-    if (profiles.isEmpty()) {
+    if (profiles.empty()) {
         profiles.add(AudioProfile::createFullDynamic());
     }
     deviceDesc->setAudioProfiles(profiles);
diff --git a/services/audiopolicy/engine/interface/AudioPolicyManagerObserver.h b/services/audiopolicy/engine/interface/AudioPolicyManagerObserver.h
index ebd82a7..ae3fc79 100644
--- a/services/audiopolicy/engine/interface/AudioPolicyManagerObserver.h
+++ b/services/audiopolicy/engine/interface/AudioPolicyManagerObserver.h
@@ -16,7 +16,6 @@
 
 #pragma once
 
-#include <AudioGain.h>
 #include <AudioPort.h>
 #include <AudioPatch.h>
 #include <IOProfile.h>
diff --git a/services/audiopolicy/engineconfigurable/Android.bp b/services/audiopolicy/engineconfigurable/Android.bp
index c27dc88..8f522f0 100644
--- a/services/audiopolicy/engineconfigurable/Android.bp
+++ b/services/audiopolicy/engineconfigurable/Android.bp
@@ -33,6 +33,7 @@
 
     ],
     shared_libs: [
+        "libaudiofoundation",
         "liblog",
         "libcutils",
         "libutils",
diff --git a/services/audiopolicy/engineconfigurable/wrapper/include/ParameterManagerWrapper.h b/services/audiopolicy/engineconfigurable/wrapper/include/ParameterManagerWrapper.h
index 5bfad29..72c8de1 100644
--- a/services/audiopolicy/engineconfigurable/wrapper/include/ParameterManagerWrapper.h
+++ b/services/audiopolicy/engineconfigurable/wrapper/include/ParameterManagerWrapper.h
@@ -16,7 +16,6 @@
 
 #pragma once
 
-#include <AudioGain.h>
 #include <AudioPort.h>
 #include <HwModule.h>
 #include <DeviceDescriptor.h>
diff --git a/services/audiopolicy/enginedefault/Android.bp b/services/audiopolicy/enginedefault/Android.bp
index 2b9cf09..aaf4158 100644
--- a/services/audiopolicy/enginedefault/Android.bp
+++ b/services/audiopolicy/enginedefault/Android.bp
@@ -21,6 +21,7 @@
         "libaudiopolicyengine_config",
     ],
     shared_libs: [
+        "libaudiofoundation",
         "liblog",
         "libcutils",
         "libutils",
diff --git a/services/audiopolicy/enginedefault/src/Engine.cpp b/services/audiopolicy/enginedefault/src/Engine.cpp
index c602f3a..cbc46d5 100644
--- a/services/audiopolicy/enginedefault/src/Engine.cpp
+++ b/services/audiopolicy/enginedefault/src/Engine.cpp
@@ -136,27 +136,23 @@
     return EngineBase::setForceUse(usage, config);
 }
 
-audio_devices_t Engine::getDeviceForStrategyInt(legacy_strategy strategy,
-                                                DeviceVector availableOutputDevices,
-                                                DeviceVector availableInputDevices,
-                                                const SwAudioOutputCollection &outputs,
-                                                uint32_t outputDeviceTypesToIgnore) const
+DeviceVector Engine::getDevicesForStrategyInt(legacy_strategy strategy,
+                                              DeviceVector availableOutputDevices,
+                                              DeviceVector availableInputDevices,
+                                              const SwAudioOutputCollection &outputs) const
 {
-    uint32_t device = AUDIO_DEVICE_NONE;
-    uint32_t availableOutputDevicesType =
-            availableOutputDevices.types() & ~outputDeviceTypesToIgnore;
+    DeviceVector devices;
 
     switch (strategy) {
 
     case STRATEGY_TRANSMITTED_THROUGH_SPEAKER:
-        device = availableOutputDevicesType & AUDIO_DEVICE_OUT_SPEAKER;
+        devices = availableOutputDevices.getDevicesFromTypeMask(AUDIO_DEVICE_OUT_SPEAKER);
         break;
 
     case STRATEGY_SONIFICATION_RESPECTFUL:
         if (isInCall() || outputs.isActiveLocally(toVolumeSource(AUDIO_STREAM_VOICE_CALL))) {
-            device = getDeviceForStrategyInt(
-                    STRATEGY_SONIFICATION, availableOutputDevices, availableInputDevices, outputs,
-                    outputDeviceTypesToIgnore);
+            devices = getDevicesForStrategyInt(
+                    STRATEGY_SONIFICATION, availableOutputDevices, availableInputDevices, outputs);
         } else {
             bool media_active_locally =
                     outputs.isActiveLocally(toVolumeSource(AUDIO_STREAM_MUSIC),
@@ -165,17 +161,18 @@
                         toVolumeSource(AUDIO_STREAM_ACCESSIBILITY),
                         SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY);
             // routing is same as media without the "remote" device
-            device = getDeviceForStrategyInt(STRATEGY_MEDIA,
+            availableOutputDevices.remove(availableOutputDevices.getDevicesFromTypeMask(
+                    AUDIO_DEVICE_OUT_REMOTE_SUBMIX));
+            devices = getDevicesForStrategyInt(STRATEGY_MEDIA,
                     availableOutputDevices,
-                    availableInputDevices, outputs,
-                    AUDIO_DEVICE_OUT_REMOTE_SUBMIX | outputDeviceTypesToIgnore);
+                    availableInputDevices, outputs);
             // if no media is playing on the device, check for mandatory use of "safe" speaker
             // when media would have played on speaker, and the safe speaker path is available
-            if (!media_active_locally
-                    && (device & AUDIO_DEVICE_OUT_SPEAKER)
-                    && (availableOutputDevicesType & AUDIO_DEVICE_OUT_SPEAKER_SAFE)) {
-                device |= AUDIO_DEVICE_OUT_SPEAKER_SAFE;
-                device &= ~AUDIO_DEVICE_OUT_SPEAKER;
+            if (!media_active_locally) {
+                devices.replaceDevicesByType(
+                        AUDIO_DEVICE_OUT_SPEAKER,
+                        availableOutputDevices.getDevicesFromTypeMask(
+                                AUDIO_DEVICE_OUT_SPEAKER_SAFE));
             }
         }
         break;
@@ -183,9 +180,8 @@
     case STRATEGY_DTMF:
         if (!isInCall()) {
             // when off call, DTMF strategy follows the same rules as MEDIA strategy
-            device = getDeviceForStrategyInt(
-                    STRATEGY_MEDIA, availableOutputDevices, availableInputDevices, outputs,
-                    outputDeviceTypesToIgnore);
+            devices = getDevicesForStrategyInt(
+                    STRATEGY_MEDIA, availableOutputDevices, availableInputDevices, outputs);
             break;
         }
         // when in call, DTMF and PHONE strategies follow the same rules
@@ -197,24 +193,27 @@
         //   - cannot route from voice call RX OR
         //   - audio HAL version is < 3.0 and TX device is on the primary HW module
         if (getPhoneState() == AUDIO_MODE_IN_CALL) {
-            audio_devices_t txDevice = getDeviceForInputSource(AUDIO_SOURCE_VOICE_COMMUNICATION);
+            audio_devices_t txDevice = getDeviceForInputSource(
+                    AUDIO_SOURCE_VOICE_COMMUNICATION)->type();
             sp<AudioOutputDescriptor> primaryOutput = outputs.getPrimaryOutput();
-            audio_devices_t availPrimaryInputDevices =
-                 availableInputDevices.getDeviceTypesFromHwModule(primaryOutput->getModuleHandle());
+            DeviceVector availPrimaryInputDevices =
+                    availableInputDevices.getDevicesFromHwModule(primaryOutput->getModuleHandle());
 
             // TODO: getPrimaryOutput return only devices from first module in
             // audio_policy_configuration.xml, hearing aid is not there, but it's
             // a primary device
             // FIXME: this is not the right way of solving this problem
-            audio_devices_t availPrimaryOutputDevices =
-                (primaryOutput->supportedDevices().types() | AUDIO_DEVICE_OUT_HEARING_AID) &
-                availableOutputDevices.types();
+            DeviceVector availPrimaryOutputDevices = availableOutputDevices.getDevicesFromTypeMask(
+                    primaryOutput->supportedDevices().types());
+            availPrimaryOutputDevices.add(
+                    availableOutputDevices.getDevicesFromTypeMask(AUDIO_DEVICE_OUT_HEARING_AID));
 
-            if (((availableInputDevices.types() &
-                    AUDIO_DEVICE_IN_TELEPHONY_RX & ~AUDIO_DEVICE_BIT_IN) == 0) ||
-                    (((txDevice & availPrimaryInputDevices & ~AUDIO_DEVICE_BIT_IN) != 0) &&
-                         (primaryOutput->getAudioPort()->getModuleVersionMajor() < 3))) {
-                availableOutputDevicesType = availPrimaryOutputDevices;
+            if ((availableInputDevices.getDevice(AUDIO_DEVICE_IN_TELEPHONY_RX,
+                    String8(""), AUDIO_FORMAT_DEFAULT) == nullptr) ||
+                    ((availPrimaryInputDevices.getDevice(
+                            txDevice, String8(""), AUDIO_FORMAT_DEFAULT) != nullptr) &&
+                            (primaryOutput->getAudioPort()->getModuleVersionMajor() < 3))) {
+                availableOutputDevices = availPrimaryOutputDevices;
             }
         }
         // for phone strategy, we first consider the forced use and then the available devices by
@@ -222,49 +221,40 @@
         switch (getForceUse(AUDIO_POLICY_FORCE_FOR_COMMUNICATION)) {
         case AUDIO_POLICY_FORCE_BT_SCO:
             if (!isInCall() || strategy != STRATEGY_DTMF) {
-                device = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT;
-                if (device) break;
+                devices = availableOutputDevices.getDevicesFromTypeMask(
+                        AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT);
+                if (!devices.isEmpty()) break;
             }
-            device = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET;
-            if (device) break;
-            device = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_SCO;
-            if (device) break;
+            devices = availableOutputDevices.getFirstDevicesFromTypes({
+                    AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET, AUDIO_DEVICE_OUT_BLUETOOTH_SCO});
+            if (!devices.isEmpty()) break;
             // if SCO device is requested but no SCO device is available, fall back to default case
             FALLTHROUGH_INTENDED;
 
         default:    // FORCE_NONE
-            device = availableOutputDevicesType & AUDIO_DEVICE_OUT_HEARING_AID;
-            if (device) break;
+            devices = availableOutputDevices.getDevicesFromTypeMask(AUDIO_DEVICE_OUT_HEARING_AID);
+            if (!devices.isEmpty()) break;
             // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to A2DP
             if (!isInCall() &&
                     (getForceUse(AUDIO_POLICY_FORCE_FOR_MEDIA) != AUDIO_POLICY_FORCE_NO_BT_A2DP) &&
                      outputs.isA2dpSupported()) {
-                device = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP;
-                if (device) break;
-                device = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
-                if (device) break;
+                devices = availableOutputDevices.getFirstDevicesFromTypes({
+                        AUDIO_DEVICE_OUT_BLUETOOTH_A2DP,
+                        AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES});
+                if (!devices.isEmpty()) break;
             }
-            device = availableOutputDevicesType & AUDIO_DEVICE_OUT_WIRED_HEADPHONE;
-            if (device) break;
-            device = availableOutputDevicesType & AUDIO_DEVICE_OUT_WIRED_HEADSET;
-            if (device) break;
-            device = availableOutputDevicesType & AUDIO_DEVICE_OUT_LINE;
-            if (device) break;
-            device = availableOutputDevicesType & AUDIO_DEVICE_OUT_USB_HEADSET;
-            if (device) break;
-            device = availableOutputDevicesType & AUDIO_DEVICE_OUT_USB_DEVICE;
-            if (device) break;
+            devices = availableOutputDevices.getFirstDevicesFromTypes({
+                    AUDIO_DEVICE_OUT_WIRED_HEADPHONE, AUDIO_DEVICE_OUT_WIRED_HEADSET,
+                    AUDIO_DEVICE_OUT_LINE, AUDIO_DEVICE_OUT_USB_HEADSET,
+                    AUDIO_DEVICE_OUT_USB_DEVICE});
+            if (!devices.isEmpty()) break;
             if (!isInCall()) {
-                device = availableOutputDevicesType & AUDIO_DEVICE_OUT_USB_ACCESSORY;
-                if (device) break;
-                device = availableOutputDevicesType & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
-                if (device) break;
-                device = availableOutputDevicesType & AUDIO_DEVICE_OUT_AUX_DIGITAL;
-                if (device) break;
-                device = availableOutputDevicesType & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
-                if (device) break;
+                devices = availableOutputDevices.getFirstDevicesFromTypes({
+                        AUDIO_DEVICE_OUT_USB_ACCESSORY, AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET,
+                        AUDIO_DEVICE_OUT_AUX_DIGITAL, AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET});
+                if (!devices.isEmpty()) break;
             }
-            device = availableOutputDevicesType & AUDIO_DEVICE_OUT_EARPIECE;
+            devices = availableOutputDevices.getDevicesFromTypeMask(AUDIO_DEVICE_OUT_EARPIECE);
             break;
 
         case AUDIO_POLICY_FORCE_SPEAKER:
@@ -273,22 +263,18 @@
             if (!isInCall() &&
                     (getForceUse(AUDIO_POLICY_FORCE_FOR_MEDIA) != AUDIO_POLICY_FORCE_NO_BT_A2DP) &&
                      outputs.isA2dpSupported()) {
-                device = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
-                if (device) break;
+                devices = availableOutputDevices.getDevicesFromTypeMask(
+                        AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER);
+                if (!devices.isEmpty()) break;
             }
             if (!isInCall()) {
-                device = availableOutputDevicesType & AUDIO_DEVICE_OUT_USB_ACCESSORY;
-                if (device) break;
-                device = availableOutputDevicesType & AUDIO_DEVICE_OUT_USB_DEVICE;
-                if (device) break;
-                device = availableOutputDevicesType & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
-                if (device) break;
-                device = availableOutputDevicesType & AUDIO_DEVICE_OUT_AUX_DIGITAL;
-                if (device) break;
-                device = availableOutputDevicesType & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
-                if (device) break;
+                devices = availableOutputDevices.getFirstDevicesFromTypes({
+                        AUDIO_DEVICE_OUT_USB_ACCESSORY, AUDIO_DEVICE_OUT_USB_DEVICE,
+                        AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET, AUDIO_DEVICE_OUT_AUX_DIGITAL,
+                        AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET});
+                if (!devices.isEmpty()) break;
             }
-            device = availableOutputDevicesType & AUDIO_DEVICE_OUT_SPEAKER;
+            devices = availableOutputDevices.getDevicesFromTypeMask(AUDIO_DEVICE_OUT_SPEAKER);
             break;
         }
     break;
@@ -298,9 +284,8 @@
         // If incall, just select the STRATEGY_PHONE device
         if (isInCall() ||
                 outputs.isActiveLocally(toVolumeSource(AUDIO_STREAM_VOICE_CALL))) {
-            device = getDeviceForStrategyInt(
-                    STRATEGY_PHONE, availableOutputDevices, availableInputDevices, outputs,
-                    outputDeviceTypesToIgnore);
+            devices = getDevicesForStrategyInt(
+                    STRATEGY_PHONE, availableOutputDevices, availableInputDevices, outputs);
             break;
         }
         FALLTHROUGH_INTENDED;
@@ -313,41 +298,37 @@
 
         if ((strategy == STRATEGY_SONIFICATION) ||
                 (getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED)) {
-            device = availableOutputDevicesType & AUDIO_DEVICE_OUT_SPEAKER;
+            devices = availableOutputDevices.getDevicesFromTypeMask(AUDIO_DEVICE_OUT_SPEAKER);
         }
 
         // if SCO headset is connected and we are told to use it, play ringtone over
         // speaker and BT SCO
-        if ((availableOutputDevicesType & AUDIO_DEVICE_OUT_ALL_SCO) != 0) {
-            uint32_t device2 = AUDIO_DEVICE_NONE;
-            device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT;
-            if (device2 == AUDIO_DEVICE_NONE) {
-                device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET;
-            }
-            if (device2 == AUDIO_DEVICE_NONE) {
-                device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_SCO;
-            }
+        if (!availableOutputDevices.getDevicesFromTypeMask(AUDIO_DEVICE_OUT_ALL_SCO).isEmpty()) {
+            DeviceVector devices2;
+            devices2 = availableOutputDevices.getFirstDevicesFromTypes({
+                    AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT, AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET,
+                    AUDIO_DEVICE_OUT_BLUETOOTH_SCO});
             // Use ONLY Bluetooth SCO output when ringing in vibration mode
             if (!((getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED)
                     && (strategy == STRATEGY_ENFORCED_AUDIBLE))) {
                 if (getForceUse(AUDIO_POLICY_FORCE_FOR_VIBRATE_RINGING)
                         == AUDIO_POLICY_FORCE_BT_SCO) {
-                    if (device2 != AUDIO_DEVICE_NONE) {
-                        device = device2;
+                    if (!devices2.isEmpty()) {
+                        devices = devices2;
                         break;
                     }
                 }
             }
             // Use both Bluetooth SCO and phone default output when ringing in normal mode
             if (getForceUse(AUDIO_POLICY_FORCE_FOR_COMMUNICATION) == AUDIO_POLICY_FORCE_BT_SCO) {
-                if ((strategy == STRATEGY_SONIFICATION) &&
-                        (device & AUDIO_DEVICE_OUT_SPEAKER) &&
-                        (availableOutputDevicesType & AUDIO_DEVICE_OUT_SPEAKER_SAFE)) {
-                    device |= AUDIO_DEVICE_OUT_SPEAKER_SAFE;
-                    device &= ~AUDIO_DEVICE_OUT_SPEAKER;
+                if (strategy == STRATEGY_SONIFICATION) {
+                    devices.replaceDevicesByType(
+                            AUDIO_DEVICE_OUT_SPEAKER,
+                            availableOutputDevices.getDevicesFromTypeMask(
+                                    AUDIO_DEVICE_OUT_SPEAKER_SAFE));
                 }
-                if (device2 != AUDIO_DEVICE_NONE) {
-                    device |= device2;
+                if (!devices2.isEmpty()) {
+                    devices.add(devices2);
                     break;
                 }
             }
@@ -361,25 +342,20 @@
             // compressed format as they would likely not be mixed and dropped.
             for (size_t i = 0; i < outputs.size(); i++) {
                 sp<AudioOutputDescriptor> desc = outputs.valueAt(i);
-                audio_devices_t devices = desc->devices().types() &
-                    (AUDIO_DEVICE_OUT_HDMI | AUDIO_DEVICE_OUT_SPDIF | AUDIO_DEVICE_OUT_HDMI_ARC);
-                if (desc->isActive() && !audio_is_linear_pcm(desc->mFormat) &&
-                        devices != AUDIO_DEVICE_NONE) {
-                    availableOutputDevicesType = availableOutputDevices.types() & ~devices;
+                if (desc->isActive() && !audio_is_linear_pcm(desc->mFormat)) {
+                    availableOutputDevices.remove(desc->devices().getDevicesFromTypeMask(
+                            AUDIO_DEVICE_OUT_HDMI | AUDIO_DEVICE_OUT_SPDIF
+                            | AUDIO_DEVICE_OUT_HDMI_ARC));
                 }
             }
-            availableOutputDevices =
-                    availableOutputDevices.getDevicesFromTypeMask(availableOutputDevicesType);
             if (outputs.isActive(toVolumeSource(AUDIO_STREAM_RING)) ||
                     outputs.isActive(toVolumeSource(AUDIO_STREAM_ALARM))) {
-                return getDeviceForStrategyInt(
-                    STRATEGY_SONIFICATION, availableOutputDevices, availableInputDevices, outputs,
-                    outputDeviceTypesToIgnore);
+                return getDevicesForStrategyInt(
+                    STRATEGY_SONIFICATION, availableOutputDevices, availableInputDevices, outputs);
             }
             if (isInCall()) {
-                return getDeviceForStrategyInt(
-                        STRATEGY_PHONE, availableOutputDevices, availableInputDevices, outputs,
-                        outputDeviceTypesToIgnore);
+                return getDevicesForStrategyInt(
+                        STRATEGY_PHONE, availableOutputDevices, availableInputDevices, outputs);
             }
         }
         // For other cases, STRATEGY_ACCESSIBILITY behaves like STRATEGY_MEDIA
@@ -388,128 +364,116 @@
     // FIXME: STRATEGY_REROUTING follow STRATEGY_MEDIA for now
     case STRATEGY_REROUTING:
     case STRATEGY_MEDIA: {
-        uint32_t device2 = AUDIO_DEVICE_NONE;
+        DeviceVector devices2;
         if (strategy != STRATEGY_SONIFICATION) {
             // no sonification on remote submix (e.g. WFD)
-            if (availableOutputDevices.getDevice(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
-                                                 String8("0"), AUDIO_FORMAT_DEFAULT) != 0) {
-                device2 = availableOutputDevices.types() & AUDIO_DEVICE_OUT_REMOTE_SUBMIX;
+            sp<DeviceDescriptor> remoteSubmix;
+            if ((remoteSubmix = availableOutputDevices.getDevice(
+                    AUDIO_DEVICE_OUT_REMOTE_SUBMIX, String8("0"),
+                    AUDIO_FORMAT_DEFAULT)) != nullptr) {
+                devices2.add(remoteSubmix);
             }
         }
         if (isInCall() && (strategy == STRATEGY_MEDIA)) {
-            device = getDeviceForStrategyInt(
-                    STRATEGY_PHONE, availableOutputDevices, availableInputDevices, outputs,
-                    outputDeviceTypesToIgnore);
+            devices = getDevicesForStrategyInt(
+                    STRATEGY_PHONE, availableOutputDevices, availableInputDevices, outputs);
             break;
         }
         // FIXME: Find a better solution to prevent routing to BT hearing aid(b/122931261).
-        if ((device2 == AUDIO_DEVICE_NONE) &&
+        if ((devices2.isEmpty()) &&
                 (getForceUse(AUDIO_POLICY_FORCE_FOR_MEDIA) != AUDIO_POLICY_FORCE_NO_BT_A2DP)) {
-            device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_HEARING_AID;
+            devices2 = availableOutputDevices.getDevicesFromTypeMask(AUDIO_DEVICE_OUT_HEARING_AID);
         }
-        if ((device2 == AUDIO_DEVICE_NONE) &&
+        if ((devices2.isEmpty()) &&
                 (getForceUse(AUDIO_POLICY_FORCE_FOR_MEDIA) != AUDIO_POLICY_FORCE_NO_BT_A2DP) &&
                  outputs.isA2dpSupported()) {
-            device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP;
-            if (device2 == AUDIO_DEVICE_NONE) {
-                device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
-            }
-            if (device2 == AUDIO_DEVICE_NONE) {
-                device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
-            }
+            devices2 = availableOutputDevices.getFirstDevicesFromTypes({
+                    AUDIO_DEVICE_OUT_BLUETOOTH_A2DP, AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES,
+                    AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER});
         }
-        if ((device2 == AUDIO_DEVICE_NONE) &&
+        if ((devices2.isEmpty()) &&
             (getForceUse(AUDIO_POLICY_FORCE_FOR_MEDIA) == AUDIO_POLICY_FORCE_SPEAKER)) {
-            device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_SPEAKER;
+            devices2 = availableOutputDevices.getDevicesFromTypeMask(AUDIO_DEVICE_OUT_SPEAKER);
         }
-        if (device2 == AUDIO_DEVICE_NONE) {
-            device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_WIRED_HEADPHONE;
+        if (devices2.isEmpty()) {
+            devices2 = availableOutputDevices.getFirstDevicesFromTypes({
+                    AUDIO_DEVICE_OUT_WIRED_HEADPHONE, AUDIO_DEVICE_OUT_LINE,
+                    AUDIO_DEVICE_OUT_WIRED_HEADSET, AUDIO_DEVICE_OUT_USB_HEADSET,
+                    AUDIO_DEVICE_OUT_USB_ACCESSORY, AUDIO_DEVICE_OUT_USB_DEVICE,
+                    AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET});
         }
-        if (device2 == AUDIO_DEVICE_NONE) {
-            device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_LINE;
-        }
-        if (device2 == AUDIO_DEVICE_NONE) {
-            device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_WIRED_HEADSET;
-        }
-        if (device2 == AUDIO_DEVICE_NONE) {
-            device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_USB_HEADSET;
-        }
-        if (device2 == AUDIO_DEVICE_NONE) {
-            device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_USB_ACCESSORY;
-        }
-        if (device2 == AUDIO_DEVICE_NONE) {
-            device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_USB_DEVICE;
-        }
-        if (device2 == AUDIO_DEVICE_NONE) {
-            device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
-        }
-        if ((device2 == AUDIO_DEVICE_NONE) && (strategy != STRATEGY_SONIFICATION)) {
+        if ((devices2.isEmpty()) && (strategy != STRATEGY_SONIFICATION)) {
             // no sonification on aux digital (e.g. HDMI)
-            device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_AUX_DIGITAL;
+            devices2 = availableOutputDevices.getDevicesFromTypeMask(AUDIO_DEVICE_OUT_AUX_DIGITAL);
         }
-        if ((device2 == AUDIO_DEVICE_NONE) &&
+        if ((devices2.isEmpty()) &&
                 (getForceUse(AUDIO_POLICY_FORCE_FOR_DOCK) == AUDIO_POLICY_FORCE_ANALOG_DOCK)) {
-            device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
+            devices2 = availableOutputDevices.getDevicesFromTypeMask(
+                    AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET);
         }
-        if (device2 == AUDIO_DEVICE_NONE) {
-            device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_SPEAKER;
+        if (devices2.isEmpty()) {
+            devices2 = availableOutputDevices.getDevicesFromTypeMask(AUDIO_DEVICE_OUT_SPEAKER);
         }
-        int device3 = AUDIO_DEVICE_NONE;
+        DeviceVector devices3;
         if (strategy == STRATEGY_MEDIA) {
             // ARC, SPDIF and AUX_LINE can co-exist with others.
-            device3 = availableOutputDevicesType & AUDIO_DEVICE_OUT_HDMI_ARC;
-            device3 |= (availableOutputDevicesType & AUDIO_DEVICE_OUT_SPDIF);
-            device3 |= (availableOutputDevicesType & AUDIO_DEVICE_OUT_AUX_LINE);
+            devices3 = availableOutputDevices.getDevicesFromTypeMask(
+                    AUDIO_DEVICE_OUT_HDMI_ARC | AUDIO_DEVICE_OUT_SPDIF | AUDIO_DEVICE_OUT_AUX_LINE);
         }
 
-        device2 |= device3;
+        devices2.add(devices3);
         // device is DEVICE_OUT_SPEAKER if we come from case STRATEGY_SONIFICATION or
         // STRATEGY_ENFORCED_AUDIBLE, AUDIO_DEVICE_NONE otherwise
-        device |= device2;
+        devices.add(devices2);
 
         // If hdmi system audio mode is on, remove speaker out of output list.
         if ((strategy == STRATEGY_MEDIA) &&
             (getForceUse(AUDIO_POLICY_FORCE_FOR_HDMI_SYSTEM_AUDIO) ==
                 AUDIO_POLICY_FORCE_HDMI_SYSTEM_AUDIO_ENFORCED)) {
-            device &= ~AUDIO_DEVICE_OUT_SPEAKER;
+            devices.remove(devices.getDevicesFromTypeMask(AUDIO_DEVICE_OUT_SPEAKER));
         }
 
         // for STRATEGY_SONIFICATION:
         // if SPEAKER was selected, and SPEAKER_SAFE is available, use SPEAKER_SAFE instead
-        if ((strategy == STRATEGY_SONIFICATION) &&
-                (device & AUDIO_DEVICE_OUT_SPEAKER) &&
-                (availableOutputDevicesType & AUDIO_DEVICE_OUT_SPEAKER_SAFE)) {
-            device |= AUDIO_DEVICE_OUT_SPEAKER_SAFE;
-            device &= ~AUDIO_DEVICE_OUT_SPEAKER;
+        if (strategy == STRATEGY_SONIFICATION) {
+            devices.replaceDevicesByType(
+                    AUDIO_DEVICE_OUT_SPEAKER,
+                    availableOutputDevices.getDevicesFromTypeMask(
+                            AUDIO_DEVICE_OUT_SPEAKER_SAFE));
         }
         } break;
 
     default:
-        ALOGW("getDeviceForStrategy() unknown strategy: %d", strategy);
+        ALOGW("getDevicesForStrategy() unknown strategy: %d", strategy);
         break;
     }
 
-    if (device == AUDIO_DEVICE_NONE) {
-        ALOGV("getDeviceForStrategy() no device found for strategy %d", strategy);
-        device = getApmObserver()->getDefaultOutputDevice()->type();
-        ALOGE_IF(device == AUDIO_DEVICE_NONE,
-                 "getDeviceForStrategy() no default device defined");
+    if (devices.isEmpty()) {
+        ALOGV("getDevicesForStrategy() no device found for strategy %d", strategy);
+        sp<DeviceDescriptor> defaultOutputDevice = getApmObserver()->getDefaultOutputDevice();
+        if (defaultOutputDevice != nullptr) {
+            devices.add(defaultOutputDevice);
+        }
+        ALOGE_IF(devices.isEmpty(),
+                 "getDevicesForStrategy() no default device defined");
     }
-    ALOGVV("getDeviceForStrategy() strategy %d, device %x", strategy, device);
-    return device;
+
+    ALOGVV("getDevices"
+           "ForStrategy() strategy %d, device %x", strategy, devices.types());
+    return devices;
 }
 
 
-audio_devices_t Engine::getDeviceForInputSource(audio_source_t inputSource) const
+sp<DeviceDescriptor> Engine::getDeviceForInputSource(audio_source_t inputSource) const
 {
     const DeviceVector availableOutputDevices = getApmObserver()->getAvailableOutputDevices();
     const DeviceVector availableInputDevices = getApmObserver()->getAvailableInputDevices();
     const SwAudioOutputCollection &outputs = getApmObserver()->getOutputs();
-    audio_devices_t availableDeviceTypes = availableInputDevices.types() & ~AUDIO_DEVICE_BIT_IN;
+    DeviceVector availableDevices = availableInputDevices;
     sp<AudioOutputDescriptor> primaryOutput = outputs.getPrimaryOutput();
-    audio_devices_t availablePrimaryDeviceTypes = availableInputDevices.getDeviceTypesFromHwModule(
-        primaryOutput->getModuleHandle()) & ~AUDIO_DEVICE_BIT_IN;
-    uint32_t device = AUDIO_DEVICE_NONE;
+    DeviceVector availablePrimaryDevices = availableInputDevices.getDevicesFromHwModule(
+            primaryOutput->getModuleHandle());
+    sp<DeviceDescriptor> device;
 
     // when a call is active, force device selection to match source VOICE_COMMUNICATION
     // for most other input sources to avoid rerouting call TX audio
@@ -532,57 +496,47 @@
     switch (inputSource) {
     case AUDIO_SOURCE_DEFAULT:
     case AUDIO_SOURCE_MIC:
-    if (availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_A2DP) {
-        device = AUDIO_DEVICE_IN_BLUETOOTH_A2DP;
-    } else if ((getForceUse(AUDIO_POLICY_FORCE_FOR_RECORD) == AUDIO_POLICY_FORCE_BT_SCO) &&
-        (availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET)) {
-        device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET;
-    } else if (availableDeviceTypes & AUDIO_DEVICE_IN_WIRED_HEADSET) {
-        device = AUDIO_DEVICE_IN_WIRED_HEADSET;
-    } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_HEADSET) {
-        device = AUDIO_DEVICE_IN_USB_HEADSET;
-    } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) {
-        device = AUDIO_DEVICE_IN_USB_DEVICE;
-    } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) {
-        device = AUDIO_DEVICE_IN_BUILTIN_MIC;
-    }
-    break;
+        device = availableDevices.getDevice(
+                AUDIO_DEVICE_IN_BLUETOOTH_A2DP, String8(""), AUDIO_FORMAT_DEFAULT);
+        if (device != nullptr) break;
+        if (getForceUse(AUDIO_POLICY_FORCE_FOR_RECORD) == AUDIO_POLICY_FORCE_BT_SCO) {
+            device = availableDevices.getDevice(
+                    AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET, String8(""), AUDIO_FORMAT_DEFAULT);
+            if (device != nullptr) break;
+        }
+        device = availableDevices.getFirstExistingDevice({
+                AUDIO_DEVICE_IN_WIRED_HEADSET, AUDIO_DEVICE_IN_USB_HEADSET,
+                AUDIO_DEVICE_IN_USB_DEVICE, AUDIO_DEVICE_IN_BUILTIN_MIC});
+        break;
 
     case AUDIO_SOURCE_VOICE_COMMUNICATION:
         // Allow only use of devices on primary input if in call and HAL does not support routing
         // to voice call path.
         if ((getPhoneState() == AUDIO_MODE_IN_CALL) &&
-                (availableOutputDevices.types() & AUDIO_DEVICE_OUT_TELEPHONY_TX) == 0) {
-            availableDeviceTypes = availablePrimaryDeviceTypes;
+                (availableOutputDevices.getDevice(AUDIO_DEVICE_OUT_TELEPHONY_TX,
+                        String8(""), AUDIO_FORMAT_DEFAULT)) == nullptr) {
+            availableDevices = availablePrimaryDevices;
         }
 
         switch (getForceUse(AUDIO_POLICY_FORCE_FOR_COMMUNICATION)) {
         case AUDIO_POLICY_FORCE_BT_SCO:
             // if SCO device is requested but no SCO device is available, fall back to default case
-            if (availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) {
-                device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET;
+            device = availableDevices.getDevice(
+                    AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET, String8(""), AUDIO_FORMAT_DEFAULT);
+            if (device != nullptr) {
                 break;
             }
             FALLTHROUGH_INTENDED;
 
         default:    // FORCE_NONE
-            if (availableDeviceTypes & AUDIO_DEVICE_IN_WIRED_HEADSET) {
-                device = AUDIO_DEVICE_IN_WIRED_HEADSET;
-            } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_HEADSET) {
-                device = AUDIO_DEVICE_IN_USB_HEADSET;
-            } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) {
-                device = AUDIO_DEVICE_IN_USB_DEVICE;
-            } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) {
-                device = AUDIO_DEVICE_IN_BUILTIN_MIC;
-            }
+            device = availableDevices.getFirstExistingDevice({
+                    AUDIO_DEVICE_IN_WIRED_HEADSET, AUDIO_DEVICE_IN_USB_HEADSET,
+                    AUDIO_DEVICE_IN_USB_DEVICE, AUDIO_DEVICE_IN_BUILTIN_MIC});
             break;
 
         case AUDIO_POLICY_FORCE_SPEAKER:
-            if (availableDeviceTypes & AUDIO_DEVICE_IN_BACK_MIC) {
-                device = AUDIO_DEVICE_IN_BACK_MIC;
-            } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) {
-                device = AUDIO_DEVICE_IN_BUILTIN_MIC;
-            }
+            device = availableDevices.getFirstExistingDevice({
+                    AUDIO_DEVICE_IN_BACK_MIC, AUDIO_DEVICE_IN_BUILTIN_MIC});
             break;
         }
         break;
@@ -591,77 +545,60 @@
     case AUDIO_SOURCE_UNPROCESSED:
     case AUDIO_SOURCE_HOTWORD:
         if (inputSource == AUDIO_SOURCE_HOTWORD) {
-            availableDeviceTypes = availablePrimaryDeviceTypes;
+            availableDevices = availablePrimaryDevices;
         }
-        if (getForceUse(AUDIO_POLICY_FORCE_FOR_RECORD) == AUDIO_POLICY_FORCE_BT_SCO &&
-                availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) {
-            device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET;
-        } else if (availableDeviceTypes & AUDIO_DEVICE_IN_WIRED_HEADSET) {
-            device = AUDIO_DEVICE_IN_WIRED_HEADSET;
-        } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_HEADSET) {
-            device = AUDIO_DEVICE_IN_USB_HEADSET;
-        } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) {
-            device = AUDIO_DEVICE_IN_USB_DEVICE;
-        } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) {
-            device = AUDIO_DEVICE_IN_BUILTIN_MIC;
+        if (getForceUse(AUDIO_POLICY_FORCE_FOR_RECORD) == AUDIO_POLICY_FORCE_BT_SCO) {
+            device = availableDevices.getDevice(
+                    AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET, String8(""), AUDIO_FORMAT_DEFAULT);
+            if (device != nullptr) break;
         }
+        device = availableDevices.getFirstExistingDevice({
+                AUDIO_DEVICE_IN_WIRED_HEADSET, AUDIO_DEVICE_IN_USB_HEADSET,
+                AUDIO_DEVICE_IN_USB_DEVICE, AUDIO_DEVICE_IN_BUILTIN_MIC});
         break;
     case AUDIO_SOURCE_CAMCORDER:
-        if (availableDeviceTypes & AUDIO_DEVICE_IN_BACK_MIC) {
-            device = AUDIO_DEVICE_IN_BACK_MIC;
-        } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) {
-            device = AUDIO_DEVICE_IN_BUILTIN_MIC;
-        } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) {
-            // This is specifically for a device without built-in mic
-            device = AUDIO_DEVICE_IN_USB_DEVICE;
-        }
+        // For a device without built-in mic, adding usb device
+        device = availableDevices.getFirstExistingDevice({
+                AUDIO_DEVICE_IN_BACK_MIC, AUDIO_DEVICE_IN_BUILTIN_MIC,
+                AUDIO_DEVICE_IN_USB_DEVICE});
         break;
     case AUDIO_SOURCE_VOICE_DOWNLINK:
     case AUDIO_SOURCE_VOICE_CALL:
     case AUDIO_SOURCE_VOICE_UPLINK:
-        if (availableDeviceTypes & AUDIO_DEVICE_IN_VOICE_CALL) {
-            device = AUDIO_DEVICE_IN_VOICE_CALL;
-        }
+        device = availableDevices.getDevice(
+                AUDIO_DEVICE_IN_VOICE_CALL, String8(""), AUDIO_FORMAT_DEFAULT);
         break;
     case AUDIO_SOURCE_VOICE_PERFORMANCE:
-        if (availableDeviceTypes & AUDIO_DEVICE_IN_WIRED_HEADSET) {
-            device = AUDIO_DEVICE_IN_WIRED_HEADSET;
-        } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_HEADSET) {
-            device = AUDIO_DEVICE_IN_USB_HEADSET;
-        } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) {
-            device = AUDIO_DEVICE_IN_USB_DEVICE;
-        } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) {
-            device = AUDIO_DEVICE_IN_BUILTIN_MIC;
-        }
+        device = availableDevices.getFirstExistingDevice({
+                AUDIO_DEVICE_IN_WIRED_HEADSET, AUDIO_DEVICE_IN_USB_HEADSET,
+                AUDIO_DEVICE_IN_USB_DEVICE, AUDIO_DEVICE_IN_BUILTIN_MIC});
         break;
     case AUDIO_SOURCE_REMOTE_SUBMIX:
-        if (availableDeviceTypes & AUDIO_DEVICE_IN_REMOTE_SUBMIX) {
-            device = AUDIO_DEVICE_IN_REMOTE_SUBMIX;
-        }
+        device = availableDevices.getDevice(
+                AUDIO_DEVICE_IN_REMOTE_SUBMIX, String8(""), AUDIO_FORMAT_DEFAULT);
         break;
     case AUDIO_SOURCE_FM_TUNER:
-        if (availableDeviceTypes & AUDIO_DEVICE_IN_FM_TUNER) {
-            device = AUDIO_DEVICE_IN_FM_TUNER;
-        }
+        device = availableDevices.getDevice(
+                AUDIO_DEVICE_IN_FM_TUNER, String8(""), AUDIO_FORMAT_DEFAULT);
         break;
     case AUDIO_SOURCE_ECHO_REFERENCE:
-        if (availableDeviceTypes & AUDIO_DEVICE_IN_ECHO_REFERENCE) {
-            device = AUDIO_DEVICE_IN_ECHO_REFERENCE;
-        }
+        device = availableDevices.getDevice(
+                AUDIO_DEVICE_IN_ECHO_REFERENCE, String8(""), AUDIO_FORMAT_DEFAULT);
         break;
     default:
         ALOGW("getDeviceForInputSource() invalid input source %d", inputSource);
         break;
     }
-    if (device == AUDIO_DEVICE_NONE) {
+    if (device == nullptr) {
         ALOGV("getDeviceForInputSource() no device found for source %d", inputSource);
-        if (availableDeviceTypes & AUDIO_DEVICE_IN_STUB) {
-            device = AUDIO_DEVICE_IN_STUB;
-        }
-        ALOGE_IF(device == AUDIO_DEVICE_NONE,
+        device = availableDevices.getDevice(
+                AUDIO_DEVICE_IN_STUB, String8(""), AUDIO_FORMAT_DEFAULT);
+        ALOGE_IF(device == nullptr,
                  "getDeviceForInputSource() no default device defined");
     }
-    ALOGV("getDeviceForInputSource()input source %d, device %08x", inputSource, device);
+    ALOGV_IF(device != nullptr,
+             "getDeviceForInputSource()input source %d, device %08x",
+             inputSource, device->type());
     return device;
 }
 
@@ -684,11 +621,9 @@
 
     auto legacyStrategy = mLegacyStrategyMap.find(strategy) != end(mLegacyStrategyMap) ?
                 mLegacyStrategyMap.at(strategy) : STRATEGY_NONE;
-    audio_devices_t devices = getDeviceForStrategyInt(legacyStrategy,
-                                                      availableOutputDevices,
-                                                      availableInputDevices, outputs,
-                                                      (uint32_t)AUDIO_DEVICE_NONE);
-    return availableOutputDevices.getDevicesFromTypeMask(devices);
+    return getDevicesForStrategyInt(legacyStrategy,
+                                    availableOutputDevices,
+                                    availableInputDevices, outputs);
 }
 
 DeviceVector Engine::getOutputDevicesForAttributes(const audio_attributes_t &attributes,
@@ -747,17 +682,21 @@
     if (device != nullptr) {
         return device;
     }
-    audio_devices_t deviceType = getDeviceForInputSource(attr.source);
 
-    if (audio_is_remote_submix_device(deviceType)) {
-        address = "0";
-        std::size_t pos;
-        std::string tags { attr.tags };
-        if ((pos = tags.find("addr=")) != std::string::npos) {
-            address = tags.substr(pos + std::strlen("addr="));
-        }
+    device = getDeviceForInputSource(attr.source);
+    if (device == nullptr || !audio_is_remote_submix_device(device->type())) {
+        // Return immediately if the device is null or it is not a remote submix device.
+        return device;
     }
-    return availableInputDevices.getDevice(deviceType,
+
+    // For remote submix device, try to find the device by address.
+    address = "0";
+    std::size_t pos;
+    std::string tags { attr.tags };
+    if ((pos = tags.find("addr=")) != std::string::npos) {
+        address = tags.substr(pos + std::strlen("addr="));
+    }
+    return availableInputDevices.getDevice(device->type(),
                                            String8(address.c_str()),
                                            AUDIO_FORMAT_DEFAULT);
 }
diff --git a/services/audiopolicy/enginedefault/src/Engine.h b/services/audiopolicy/enginedefault/src/Engine.h
index 62938cf..4360c6f 100644
--- a/services/audiopolicy/enginedefault/src/Engine.h
+++ b/services/audiopolicy/enginedefault/src/Engine.h
@@ -18,7 +18,6 @@
 
 #include "EngineBase.h"
 #include "EngineInterface.h"
-#include <AudioGain.h>
 #include <policy.h>
 
 namespace android
@@ -74,15 +73,14 @@
 
     status_t setDefaultDevice(audio_devices_t device);
 
-    audio_devices_t getDeviceForStrategyInt(legacy_strategy strategy,
-                                            DeviceVector availableOutputDevices,
-                                            DeviceVector availableInputDevices,
-                                            const SwAudioOutputCollection &outputs,
-                                            uint32_t outputDeviceTypesToIgnore) const;
+    DeviceVector getDevicesForStrategyInt(legacy_strategy strategy,
+                                          DeviceVector availableOutputDevices,
+                                          DeviceVector availableInputDevices,
+                                          const SwAudioOutputCollection &outputs) const;
 
     DeviceVector getDevicesForProductStrategy(product_strategy_t strategy) const;
 
-    audio_devices_t getDeviceForInputSource(audio_source_t inputSource) const;
+    sp<DeviceDescriptor> getDeviceForInputSource(audio_source_t inputSource) const;
 
     DeviceStrategyMap mDevicesForStrategies;
 
diff --git a/services/audiopolicy/managerdefault/Android.bp b/services/audiopolicy/managerdefault/Android.bp
index 8fbeff9..1fa0d19 100644
--- a/services/audiopolicy/managerdefault/Android.bp
+++ b/services/audiopolicy/managerdefault/Android.bp
@@ -9,6 +9,7 @@
     export_include_dirs: ["."],
 
     shared_libs: [
+        "libaudiofoundation",
         "libcutils",
         "libdl",
         "libutils",
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index 83ae35e..499fc8a 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -73,6 +73,26 @@
         AUDIO_CHANNEL_OUT_2POINT1POINT2, AUDIO_CHANNEL_OUT_2POINT0POINT2,
         AUDIO_CHANNEL_OUT_5POINT1, AUDIO_CHANNEL_OUT_STEREO }};
 
+template <typename T>
+bool operator== (const SortedVector<T> &left, const SortedVector<T> &right)
+{
+    if (left.size() != right.size()) {
+        return false;
+    }
+    for (size_t index = 0; index < right.size(); index++) {
+        if (left[index] != right[index]) {
+            return false;
+        }
+    }
+    return true;
+}
+
+template <typename T>
+bool operator!= (const SortedVector<T> &left, const SortedVector<T> &right)
+{
+    return !(left == right);
+}
+
 // ----------------------------------------------------------------------------
 // AudioPolicyInterface implementation
 // ----------------------------------------------------------------------------
@@ -1339,13 +1359,13 @@
     // Each IOProfile represents a MixPort from audio_policy_configuration.xml
     for (const auto &inProfile : inputProfiles) {
         if (hwAvSync == ((inProfile->getFlags() & AUDIO_INPUT_FLAG_HW_AV_SYNC) != 0)) {
-            msdProfiles.appendVector(inProfile->getAudioProfiles());
+            msdProfiles.appendProfiles(inProfile->getAudioProfiles());
         }
     }
     AudioProfileVector deviceProfiles;
     for (const auto &outProfile : outputProfiles) {
         if (hwAvSync == ((outProfile->getFlags() & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0)) {
-            deviceProfiles.appendVector(outProfile->getAudioProfiles());
+            deviceProfiles.appendProfiles(outProfile->getAudioProfiles());
         }
     }
     struct audio_config_base bestSinkConfig;
@@ -6084,24 +6104,24 @@
         formatSet.insert(enforcedSurround.begin(), enforcedSurround.end());
     }
     for (const auto& format : formatSet) {
-        formatsPtr->push(format);
+        formatsPtr->push_back(format);
     }
 }
 
-void AudioPolicyManager::modifySurroundChannelMasks(ChannelsVector *channelMasksPtr) {
-    ChannelsVector &channelMasks = *channelMasksPtr;
+void AudioPolicyManager::modifySurroundChannelMasks(ChannelMaskSet *channelMasksPtr) {
+    ChannelMaskSet &channelMasks = *channelMasksPtr;
     audio_policy_forced_cfg_t forceUse = mEngine->getForceUse(
             AUDIO_POLICY_FORCE_FOR_ENCODED_SURROUND);
 
     // If NEVER, then remove support for channelMasks > stereo.
     if (forceUse == AUDIO_POLICY_FORCE_ENCODED_SURROUND_NEVER) {
-        for (size_t maskIndex = 0; maskIndex < channelMasks.size(); ) {
-            audio_channel_mask_t channelMask = channelMasks[maskIndex];
+        for (auto it = channelMasks.begin(); it != channelMasks.end();) {
+            audio_channel_mask_t channelMask = *it;
             if (channelMask & ~AUDIO_CHANNEL_OUT_STEREO) {
                 ALOGI("%s: force NEVER, so remove channelMask 0x%08x", __FUNCTION__, channelMask);
-                channelMasks.removeAt(maskIndex);
+                it = channelMasks.erase(it);
             } else {
-                maskIndex++;
+                ++it;
             }
         }
     // If ALWAYS or MANUAL, then make sure we at least support 5.1
@@ -6117,7 +6137,7 @@
         }
         // If not then add 5.1 support.
         if (!supports5dot1) {
-            channelMasks.add(AUDIO_CHANNEL_OUT_5POINT1);
+            channelMasks.insert(AUDIO_CHANNEL_OUT_5POINT1);
             ALOGI("%s: force MANUAL or ALWAYS, so adding channelMask for 5.1 surround", __func__);
         }
     }
@@ -6150,8 +6170,8 @@
     }
 
     for (audio_format_t format : profiles.getSupportedFormats()) {
-        ChannelsVector channelMasks;
-        SampleRateVector samplingRates;
+        ChannelMaskSet channelMasks;
+        SampleRateSet samplingRates;
         AudioParameter requestedParameters;
         requestedParameters.addInt(String8(AudioParameter::keyFormat), format);
 
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h
index 5f651cc..02c6171 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.h
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h
@@ -35,7 +35,6 @@
 #include "AudioPolicyInterface.h"
 
 #include <AudioPolicyManagerObserver.h>
-#include <AudioGain.h>
 #include <AudioPolicyConfig.h>
 #include <AudioPort.h>
 #include <AudioPatch.h>
@@ -763,7 +762,7 @@
 private:
         // Add or remove AC3 DTS encodings based on user preferences.
         void modifySurroundFormats(const sp<DeviceDescriptor>& devDesc, FormatVector *formatsPtr);
-        void modifySurroundChannelMasks(ChannelsVector *channelMasksPtr);
+        void modifySurroundChannelMasks(ChannelMaskSet *channelMasksPtr);
 
         // Support for Multi-Stream Decoder (MSD) module
         sp<DeviceDescriptor> getMsdAudioInDevice() const;
diff --git a/services/audiopolicy/service/AudioPolicyService.cpp b/services/audiopolicy/service/AudioPolicyService.cpp
index 62010e1..d1b59c1 100644
--- a/services/audiopolicy/service/AudioPolicyService.cpp
+++ b/services/audiopolicy/service/AudioPolicyService.cpp
@@ -409,12 +409,17 @@
 //    Another client in the same UID has already been allowed to capture
 //    OR The client is the assistant
 //        AND an accessibility service is on TOP or a RTT call is active
-//               AND the source is VOICE_RECOGNITION or HOTWORD
-//        OR uses VOICE_RECOGNITION AND is on TOP
-//               OR uses HOTWORD
+//                AND the source is VOICE_RECOGNITION or HOTWORD
+//            OR uses VOICE_RECOGNITION AND is on TOP
+//                OR uses HOTWORD
 //            AND there is no active privacy sensitive capture or call
 //                OR client has CAPTURE_AUDIO_OUTPUT privileged permission
 //    OR The client is an accessibility service
+//        AND Is on TOP
+//                AND the source is VOICE_RECOGNITION or HOTWORD
+//            OR The assistant is not on TOP
+//                AND there is no active privacy sensitive capture or call
+//                    OR client has CAPTURE_AUDIO_OUTPUT privileged permission
 //        AND is on TOP
 //        AND the source is VOICE_RECOGNITION or HOTWORD
 //    OR the client source is virtual (remote submix, call audio TX or RX...)
@@ -422,7 +427,7 @@
 //        AND The assistant is not on TOP
 //        AND is on TOP or latest started
 //        AND there is no active privacy sensitive capture or call
-//                OR client has CAPTURE_AUDIO_OUTPUT privileged permission
+//            OR client has CAPTURE_AUDIO_OUTPUT privileged permission
 
     sp<AudioRecordClient> topActive;
     sp<AudioRecordClient> latestActive;
@@ -459,7 +464,8 @@
         }
 
         bool isAssistant = mUidPolicy->isAssistantUid(current->uid);
-        if (appState == APP_STATE_TOP) {
+        bool isAccessibility = mUidPolicy->isA11yUid(current->uid);
+        if (appState == APP_STATE_TOP && !isAccessibility) {
             if (current->startTimeNs > topStartNs) {
                 topActive = current;
                 topStartNs = current->startTimeNs;
@@ -468,10 +474,13 @@
                 isAssistantOnTop = true;
             }
         }
-        // Assistant capturing for HOTWORD not considered for latest active to avoid
-        // masking regular clients started before
-        if (current->startTimeNs > latestStartNs &&
-            !(current->attributes.source == AUDIO_SOURCE_HOTWORD && isAssistant)) {
+        // Assistant capturing for HOTWORD or Accessibility services not considered
+        // for latest active to avoid masking regular clients started before
+        if (current->startTimeNs > latestStartNs
+                && !((current->attributes.source == AUDIO_SOURCE_HOTWORD
+                        || isA11yOnTop || rttCallActive)
+                    && isAssistant)
+                && !isAccessibility) {
             latestActive = current;
             latestStartNs = current->startTimeNs;
         }
@@ -544,10 +553,20 @@
         } else if (mUidPolicy->isA11yUid(current->uid)) {
             // For accessibility service allow capture if:
             //     Is on TOP
-            //     AND the source is VOICE_RECOGNITION or HOTWORD
-            if (isA11yOnTop &&
-                    (source == AUDIO_SOURCE_VOICE_RECOGNITION || source == AUDIO_SOURCE_HOTWORD)) {
-                allowCapture = true;
+            //          AND the source is VOICE_RECOGNITION or HOTWORD
+            //     Or
+            //          The assistant is not on TOP
+            //          AND there is no active privacy sensitive capture or call
+            //             OR client has CAPTURE_AUDIO_OUTPUT privileged permission
+            if (isA11yOnTop) {
+                if (source == AUDIO_SOURCE_VOICE_RECOGNITION || source == AUDIO_SOURCE_HOTWORD) {
+                    allowCapture = true;
+                }
+            } else {
+                if (!isAssistantOnTop
+                        && (!(isSensitiveActive || isInCall) || current->canCaptureOutput)) {
+                    allowCapture = true;
+                }
             }
         }
         setAppState_l(current->uid,
diff --git a/services/audiopolicy/tests/Android.mk b/services/audiopolicy/tests/Android.mk
index ab9f78b..c8d1459 100644
--- a/services/audiopolicy/tests/Android.mk
+++ b/services/audiopolicy/tests/Android.mk
@@ -7,6 +7,7 @@
   $(call include-path-for, audio-utils) \
 
 LOCAL_SHARED_LIBRARIES := \
+  libaudiofoundation \
   libaudiopolicymanagerdefault \
   libbase \
   liblog \
@@ -41,6 +42,7 @@
 include $(CLEAR_VARS)
 
 LOCAL_SHARED_LIBRARIES := \
+  libaudiofoundation \
   libbase \
   liblog \
   libmedia_helper \
diff --git a/services/camera/libcameraservice/common/DepthPhotoProcessor.cpp b/services/camera/libcameraservice/common/DepthPhotoProcessor.cpp
index 3c90de0..94541d8 100644
--- a/services/camera/libcameraservice/common/DepthPhotoProcessor.cpp
+++ b/services/camera/libcameraservice/common/DepthPhotoProcessor.cpp
@@ -419,7 +419,7 @@
 
     std::vector<std::unique_ptr<Item>> items;
     std::vector<std::unique_ptr<Camera>> cameraList;
-    auto image = Image::FromDataForPrimaryImage("android/mainimage", &items);
+    auto image = Image::FromDataForPrimaryImage("image/jpeg", &items);
     std::unique_ptr<CameraParams> cameraParams(new CameraParams(std::move(image)));
     if (cameraParams == nullptr) {
         ALOGE("%s: Failed to initialize camera parameters", __FUNCTION__);
diff --git a/services/mediaresourcemanager/ResourceManagerService.cpp b/services/mediaresourcemanager/ResourceManagerService.cpp
index 28bfd3f..117a211 100644
--- a/services/mediaresourcemanager/ResourceManagerService.cpp
+++ b/services/mediaresourcemanager/ResourceManagerService.cpp
@@ -21,8 +21,11 @@
 
 #include <binder/IMediaResourceMonitor.h>
 #include <binder/IServiceManager.h>
+#include <cutils/sched_policy.h>
 #include <dirent.h>
 #include <media/stagefright/ProcessInfo.h>
+#include <mediautils/BatteryNotifier.h>
+#include <mediautils/SchedulingPolicyService.h>
 #include <string.h>
 #include <sys/types.h>
 #include <sys/stat.h>
@@ -31,8 +34,7 @@
 
 #include "ResourceManagerService.h"
 #include "ServiceLog.h"
-#include "mediautils/SchedulingPolicyService.h"
-#include <cutils/sched_policy.h>
+
 namespace android {
 
 namespace {
@@ -101,6 +103,7 @@
 }
 
 static ResourceInfo& getResourceInfoForEdit(
+        uid_t uid,
         int64_t clientId,
         const sp<IResourceManagerClient>& client,
         ResourceInfos& infos) {
@@ -110,9 +113,11 @@
         }
     }
     ResourceInfo info;
+    info.uid = uid;
     info.clientId = clientId;
     info.client = client;
     info.cpuBoost = false;
+    info.batteryNoted = false;
     infos.push_back(info);
     return infos.editItemAt(infos.size() - 1);
 }
@@ -204,7 +209,9 @@
       mServiceLog(new ServiceLog()),
       mSupportsMultipleSecureCodecs(true),
       mSupportsSecureWithNonSecureCodec(true),
-      mCpuBoostCount(0) {}
+      mCpuBoostCount(0) {
+    BatteryNotifier::getInstance().noteResetVideo();
+}
 
 ResourceManagerService::~ResourceManagerService() {}
 
@@ -226,6 +233,7 @@
 
 void ResourceManagerService::addResource(
         int pid,
+        int uid,
         int64_t clientId,
         const sp<IResourceManagerClient> client,
         const Vector<MediaResource> &resources) {
@@ -239,7 +247,7 @@
         return;
     }
     ResourceInfos& infos = getResourceInfosForEdit(pid, mMap);
-    ResourceInfo& info = getResourceInfoForEdit(clientId, client, infos);
+    ResourceInfo& info = getResourceInfoForEdit(uid, clientId, client, infos);
     // TODO: do the merge instead of append.
     info.resources.appendVector(resources);
 
@@ -253,6 +261,11 @@
                 ALOGW("couldn't request cpuset boost");
             }
             mCpuBoostCount++;
+        } else if (resources[i].mType == MediaResource::kBattery
+                && resources[i].mSubType == MediaResource::kVideoCodec
+                && !info.batteryNoted) {
+            info.batteryNoted = true;
+            BatteryNotifier::getInstance().noteStartVideo(info.uid);
         }
     }
     if (info.deathNotifier == nullptr) {
@@ -291,6 +304,9 @@
                     requestCpusetBoost(false, this);
                 }
             }
+            if (infos[j].batteryNoted) {
+                BatteryNotifier::getInstance().noteStopVideo(infos[j].uid);
+            }
             IInterface::asBinder(infos[j].client)->unlinkToDeath(infos[j].deathNotifier);
             j = infos.removeAt(j);
             found = true;
diff --git a/services/mediaresourcemanager/ResourceManagerService.h b/services/mediaresourcemanager/ResourceManagerService.h
index 82d2a0b..741ef73 100644
--- a/services/mediaresourcemanager/ResourceManagerService.h
+++ b/services/mediaresourcemanager/ResourceManagerService.h
@@ -35,10 +35,12 @@
 
 struct ResourceInfo {
     int64_t clientId;
+    uid_t uid;
     sp<IResourceManagerClient> client;
     sp<IBinder::DeathRecipient> deathNotifier;
     Vector<MediaResource> resources;
     bool cpuBoost;
+    bool batteryNoted;
 };
 
 typedef Vector<ResourceInfo> ResourceInfos;
@@ -61,6 +63,7 @@
 
     virtual void addResource(
             int pid,
+            int uid,
             int64_t clientId,
             const sp<IResourceManagerClient> client,
             const Vector<MediaResource> &resources);
diff --git a/services/mediaresourcemanager/test/ResourceManagerService_test.cpp b/services/mediaresourcemanager/test/ResourceManagerService_test.cpp
index ed0b0ef..8a3987a 100644
--- a/services/mediaresourcemanager/test/ResourceManagerService_test.cpp
+++ b/services/mediaresourcemanager/test/ResourceManagerService_test.cpp
@@ -86,7 +86,10 @@
 };
 
 static const int kTestPid1 = 30;
+static const int kTestUid1 = 1010;
+
 static const int kTestPid2 = 20;
+static const int kTestUid2 = 1011;
 
 static const int kLowPriorityPid = 40;
 static const int kMidPriorityPid = 25;
@@ -115,8 +118,11 @@
         return true;
     }
 
-    static void expectEqResourceInfo(const ResourceInfo &info, sp<IResourceManagerClient> client,
+    static void expectEqResourceInfo(const ResourceInfo &info,
+            int uid,
+            sp<IResourceManagerClient> client,
             const Vector<MediaResource> &resources) {
+        EXPECT_EQ(uid, info.uid);
         EXPECT_EQ(client, info.client);
         EXPECT_TRUE(isEqualResources(resources, info.resources));
     }
@@ -153,24 +159,24 @@
         // kTestPid1 mTestClient1
         Vector<MediaResource> resources1;
         resources1.push_back(MediaResource(MediaResource::kSecureCodec, 1));
-        mService->addResource(kTestPid1, getId(mTestClient1), mTestClient1, resources1);
+        mService->addResource(kTestPid1, kTestUid1, getId(mTestClient1), mTestClient1, resources1);
         resources1.push_back(MediaResource(MediaResource::kGraphicMemory, 200));
         Vector<MediaResource> resources11;
         resources11.push_back(MediaResource(MediaResource::kGraphicMemory, 200));
-        mService->addResource(kTestPid1, getId(mTestClient1), mTestClient1, resources11);
+        mService->addResource(kTestPid1, kTestUid1, getId(mTestClient1), mTestClient1, resources11);
 
         // kTestPid2 mTestClient2
         Vector<MediaResource> resources2;
         resources2.push_back(MediaResource(MediaResource::kNonSecureCodec, 1));
         resources2.push_back(MediaResource(MediaResource::kGraphicMemory, 300));
-        mService->addResource(kTestPid2, getId(mTestClient2), mTestClient2, resources2);
+        mService->addResource(kTestPid2, kTestUid2, getId(mTestClient2), mTestClient2, resources2);
 
         // kTestPid2 mTestClient3
         Vector<MediaResource> resources3;
-        mService->addResource(kTestPid2, getId(mTestClient3), mTestClient3, resources3);
+        mService->addResource(kTestPid2, kTestUid2, getId(mTestClient3), mTestClient3, resources3);
         resources3.push_back(MediaResource(MediaResource::kSecureCodec, 1));
         resources3.push_back(MediaResource(MediaResource::kGraphicMemory, 100));
-        mService->addResource(kTestPid2, getId(mTestClient3), mTestClient3, resources3);
+        mService->addResource(kTestPid2, kTestUid2, getId(mTestClient3), mTestClient3, resources3);
 
         const PidResourceInfosMap &map = mService->mMap;
         EXPECT_EQ(2u, map.size());
@@ -178,14 +184,14 @@
         ASSERT_GE(index1, 0);
         const ResourceInfos &infos1 = map[index1];
         EXPECT_EQ(1u, infos1.size());
-        expectEqResourceInfo(infos1[0], mTestClient1, resources1);
+        expectEqResourceInfo(infos1[0], kTestUid1, mTestClient1, resources1);
 
         ssize_t index2 = map.indexOfKey(kTestPid2);
         ASSERT_GE(index2, 0);
         const ResourceInfos &infos2 = map[index2];
         EXPECT_EQ(2u, infos2.size());
-        expectEqResourceInfo(infos2[0], mTestClient2, resources2);
-        expectEqResourceInfo(infos2[1], mTestClient3, resources3);
+        expectEqResourceInfo(infos2[0], kTestUid2, mTestClient2, resources2);
+        expectEqResourceInfo(infos2[1], kTestUid2, mTestClient3, resources3);
     }
 
     void testConfig() {