Merge "Create host specific lib variant" into main
diff --git a/media/codec2/hal/aidl/ComponentStore.cpp b/media/codec2/hal/aidl/ComponentStore.cpp
index 356bf72..de9332b 100644
--- a/media/codec2/hal/aidl/ComponentStore.cpp
+++ b/media/codec2/hal/aidl/ComponentStore.cpp
@@ -153,6 +153,13 @@
mParamReflectors.push_back(paramReflector);
}
#endif
+ // MultiAccessUnit reflector helper is allocated once per store.
+ // All components in this store can reuse this reflector helper.
+ if (MultiAccessUnitHelper::isEnabledOnPlatform()) {
+ std::shared_ptr<C2ReflectorHelper> helper = std::make_shared<C2ReflectorHelper>();
+ mParamReflectors.push_back(helper);
+ mMultiAccessUnitReflector = helper;
+ }
// Retrieve supported parameters from store
using namespace std::placeholders;
@@ -240,11 +247,9 @@
// param reflectors. Currently filters work on video domain only,
// and the MultiAccessUnitHelper is only enabled on audio domain;
// thus we pass the component's param reflector, which is mParamReflectors[0].
- std::shared_ptr<C2ReflectorHelper> multiAccessReflector(new C2ReflectorHelper());
multiAccessUnitIntf = std::make_shared<MultiAccessUnitInterface>(
c2interface,
- multiAccessReflector);
- mParamReflectors.push_back(multiAccessReflector);
+ mMultiAccessUnitReflector);
}
}
}
diff --git a/media/codec2/hal/aidl/include/codec2/aidl/ComponentStore.h b/media/codec2/hal/aidl/include/codec2/aidl/ComponentStore.h
index b2158a6..de0f566 100644
--- a/media/codec2/hal/aidl/include/codec2/aidl/ComponentStore.h
+++ b/media/codec2/hal/aidl/include/codec2/aidl/ComponentStore.h
@@ -120,6 +120,9 @@
std::shared_ptr<C2ComponentStore> mStore;
std::vector<std::shared_ptr<C2ParamReflector>> mParamReflectors;
+ // Reflector helper for MultiAccessUnitHelper
+ std::shared_ptr<C2ReflectorHelper> mMultiAccessUnitReflector;
+
std::map<C2Param::CoreIndex, std::shared_ptr<C2StructDescriptor>> mStructDescriptors;
std::set<C2Param::CoreIndex> mUnsupportedStructDescriptors;
std::set<C2String> mLoadedInterfaces;
diff --git a/media/codec2/hal/hidl/1.0/utils/ComponentStore.cpp b/media/codec2/hal/hidl/1.0/utils/ComponentStore.cpp
index f09e232..108ba06 100644
--- a/media/codec2/hal/hidl/1.0/utils/ComponentStore.cpp
+++ b/media/codec2/hal/hidl/1.0/utils/ComponentStore.cpp
@@ -149,6 +149,14 @@
}
#endif
+ // MultiAccessUnit reflector helper is allocated once per store.
+ // All components in this store can reuse this reflector helper.
+ if (MultiAccessUnitHelper::isEnabledOnPlatform()) {
+ std::shared_ptr<C2ReflectorHelper> helper = std::make_shared<C2ReflectorHelper>();
+ mParamReflectors.push_back(helper);
+ mMultiAccessUnitReflector = helper;
+ }
+
// Retrieve supported parameters from store
using namespace std::placeholders;
mInit = mConfigurable->init(mParameterCache);
@@ -231,12 +239,9 @@
}
}
if (!isComponentSupportsLargeAudioFrame) {
- std::shared_ptr<C2ReflectorHelper> multiAccessReflector(new C2ReflectorHelper());
multiAccessUnitIntf = std::make_shared<MultiAccessUnitInterface>(
c2interface,
- multiAccessReflector);
- mParamReflectors.push_back(multiAccessReflector);
-
+ mMultiAccessUnitReflector);
}
}
}
diff --git a/media/codec2/hal/hidl/1.0/utils/include/codec2/hidl/1.0/ComponentStore.h b/media/codec2/hal/hidl/1.0/utils/include/codec2/hidl/1.0/ComponentStore.h
index 44b8ec1..847c90c 100644
--- a/media/codec2/hal/hidl/1.0/utils/include/codec2/hidl/1.0/ComponentStore.h
+++ b/media/codec2/hal/hidl/1.0/utils/include/codec2/hidl/1.0/ComponentStore.h
@@ -124,6 +124,9 @@
std::shared_ptr<C2ComponentStore> mStore;
std::vector<std::shared_ptr<C2ParamReflector>> mParamReflectors;
+ // Reflector helper for MultiAccessUnitHelper
+ std::shared_ptr<C2ReflectorHelper> mMultiAccessUnitReflector;
+
std::map<C2Param::CoreIndex, std::shared_ptr<C2StructDescriptor>> mStructDescriptors;
std::set<C2Param::CoreIndex> mUnsupportedStructDescriptors;
std::set<C2String> mLoadedInterfaces;
diff --git a/media/codec2/hal/hidl/1.1/utils/ComponentStore.cpp b/media/codec2/hal/hidl/1.1/utils/ComponentStore.cpp
index 009a326..84f5d26 100644
--- a/media/codec2/hal/hidl/1.1/utils/ComponentStore.cpp
+++ b/media/codec2/hal/hidl/1.1/utils/ComponentStore.cpp
@@ -149,6 +149,14 @@
}
#endif
+ // MultiAccessUnit reflector helper is allocated once per store.
+ // All components in this store can reuse this reflector helper.
+ if (MultiAccessUnitHelper::isEnabledOnPlatform()) {
+ std::shared_ptr<C2ReflectorHelper> helper = std::make_shared<C2ReflectorHelper>();
+ mParamReflectors.push_back(helper);
+ mMultiAccessUnitReflector = helper;
+ }
+
// Retrieve supported parameters from store
using namespace std::placeholders;
mInit = mConfigurable->init(mParameterCache);
@@ -230,13 +238,10 @@
break;
}
}
-
if (!isComponentSupportsLargeAudioFrame) {
- std::shared_ptr<C2ReflectorHelper> multiAccessReflector(new C2ReflectorHelper());
multiAccessUnitIntf = std::make_shared<MultiAccessUnitInterface>(
c2interface,
- multiAccessReflector);
- mParamReflectors.push_back(multiAccessReflector);
+ mMultiAccessUnitReflector);
}
}
}
diff --git a/media/codec2/hal/hidl/1.1/utils/include/codec2/hidl/1.1/ComponentStore.h b/media/codec2/hal/hidl/1.1/utils/include/codec2/hidl/1.1/ComponentStore.h
index 52d2945..9028149 100644
--- a/media/codec2/hal/hidl/1.1/utils/include/codec2/hidl/1.1/ComponentStore.h
+++ b/media/codec2/hal/hidl/1.1/utils/include/codec2/hidl/1.1/ComponentStore.h
@@ -132,6 +132,9 @@
std::shared_ptr<C2ComponentStore> mStore;
std::vector<std::shared_ptr<C2ParamReflector>> mParamReflectors;
+ // Reflector helper for MultiAccessUnitHelper
+ std::shared_ptr<C2ReflectorHelper> mMultiAccessUnitReflector;
+
std::map<C2Param::CoreIndex, std::shared_ptr<C2StructDescriptor>> mStructDescriptors;
std::set<C2Param::CoreIndex> mUnsupportedStructDescriptors;
std::set<C2String> mLoadedInterfaces;
diff --git a/media/codec2/hal/hidl/1.2/utils/ComponentStore.cpp b/media/codec2/hal/hidl/1.2/utils/ComponentStore.cpp
index 89f71a9..5585be8 100644
--- a/media/codec2/hal/hidl/1.2/utils/ComponentStore.cpp
+++ b/media/codec2/hal/hidl/1.2/utils/ComponentStore.cpp
@@ -149,6 +149,14 @@
}
#endif
+ // MultiAccessUnit reflector helper is allocated once per store.
+ // All components in this store can reuse this reflector helper.
+ if (MultiAccessUnitHelper::isEnabledOnPlatform()) {
+ std::shared_ptr<C2ReflectorHelper> helper = std::make_shared<C2ReflectorHelper>();
+ mParamReflectors.push_back(helper);
+ mMultiAccessUnitReflector = helper;
+ }
+
// Retrieve supported parameters from store
using namespace std::placeholders;
mInit = mConfigurable->init(mParameterCache);
@@ -231,11 +239,9 @@
}
}
if (!isComponentSupportsLargeAudioFrame) {
- std::shared_ptr<C2ReflectorHelper> multiAccessReflector(new C2ReflectorHelper());
multiAccessUnitIntf = std::make_shared<MultiAccessUnitInterface>(
c2interface,
- multiAccessReflector);
- mParamReflectors.push_back(multiAccessReflector);
+ mMultiAccessUnitReflector);
}
}
}
diff --git a/media/codec2/hal/hidl/1.2/utils/include/codec2/hidl/1.2/ComponentStore.h b/media/codec2/hal/hidl/1.2/utils/include/codec2/hidl/1.2/ComponentStore.h
index 1b209e2..4fd260b 100644
--- a/media/codec2/hal/hidl/1.2/utils/include/codec2/hidl/1.2/ComponentStore.h
+++ b/media/codec2/hal/hidl/1.2/utils/include/codec2/hidl/1.2/ComponentStore.h
@@ -139,6 +139,9 @@
std::shared_ptr<C2ComponentStore> mStore;
std::vector<std::shared_ptr<C2ParamReflector>> mParamReflectors;
+ // Reflector helper for MultiAccessUnitHelper
+ std::shared_ptr<C2ReflectorHelper> mMultiAccessUnitReflector;
+
std::map<C2Param::CoreIndex, std::shared_ptr<C2StructDescriptor>> mStructDescriptors;
std::set<C2Param::CoreIndex> mUnsupportedStructDescriptors;
std::set<C2String> mLoadedInterfaces;
diff --git a/media/codec2/hal/plugin/samples/SampleFilterPlugin.cpp b/media/codec2/hal/plugin/samples/SampleFilterPlugin.cpp
index 47412b7..34872f0 100644
--- a/media/codec2/hal/plugin/samples/SampleFilterPlugin.cpp
+++ b/media/codec2/hal/plugin/samples/SampleFilterPlugin.cpp
@@ -856,21 +856,31 @@
C2String getName() const override { return "android.sample.filter-plugin-store"; }
c2_status_t createComponent(
C2String name, std::shared_ptr<C2Component>* const component) override {
- if (mFactories.count(name) == 0) {
+ auto it = std::find_if(
+ mFactories.begin(), mFactories.end(),
+ [&name](const std::unique_ptr<ComponentFactory> &factory) {
+ return name == factory->getTraits()->name;
+ });
+ if (it == mFactories.end()) {
return C2_BAD_VALUE;
}
- return mFactories.at(name)->createComponent(++mNodeId, component);
+ return (*it)->createComponent(++mNodeId, component);
}
c2_status_t createInterface(
C2String name, std::shared_ptr<C2ComponentInterface>* const interface) override {
- if (mFactories.count(name) == 0) {
+ auto it = std::find_if(
+ mFactories.begin(), mFactories.end(),
+ [&name](const std::unique_ptr<ComponentFactory> &factory) {
+ return name == factory->getTraits()->name;
+ });
+ if (it == mFactories.end()) {
return C2_BAD_VALUE;
}
- return mFactories.at(name)->createInterface(++mNodeId, interface);
+ return (*it)->createInterface(++mNodeId, interface);
}
std::vector<std::shared_ptr<const C2Component::Traits>> listComponents() override {
std::vector<std::shared_ptr<const C2Component::Traits>> ret;
- for (const auto &[name, factory] : mFactories) {
+ for (const auto &factory : mFactories) {
ret.push_back(factory->getTraits());
}
return ret;
@@ -951,20 +961,18 @@
template <class T>
static void AddFactory(
- std::map<C2String, std::unique_ptr<ComponentFactory>> *factories,
+ std::vector<std::unique_ptr<ComponentFactory>> *factories,
const std::shared_ptr<C2ReflectorHelper> &reflector) {
std::shared_ptr<C2ComponentInterface> intf{new typename T::Interface(0, reflector)};
std::shared_ptr<C2Component::Traits> traits(new (std::nothrow) C2Component::Traits);
CHECK(C2InterfaceUtils::FillTraitsFromInterface(traits.get(), intf))
<< "Failed to fill traits from interface";
- factories->emplace(
- traits->name,
- new ComponentFactoryImpl<T>(traits, reflector));
+ factories->emplace_back(new ComponentFactoryImpl<T>(traits, reflector));
}
- static std::map<C2String, std::unique_ptr<ComponentFactory>> CreateFactories(
+ static std::vector<std::unique_ptr<ComponentFactory>> CreateFactories(
const std::shared_ptr<C2ReflectorHelper> &reflector) {
- std::map<C2String, std::unique_ptr<ComponentFactory>> factories;
+ std::vector<std::unique_ptr<ComponentFactory>> factories;
AddFactory<SampleToneMappingFilter>(&factories, reflector);
return factories;
}
@@ -977,7 +985,7 @@
}
} mIntf;
- const std::map<C2String, std::unique_ptr<ComponentFactory>> mFactories;
+ const std::vector<std::unique_ptr<ComponentFactory>> mFactories;
std::atomic_int32_t mNodeId{0};
};
diff --git a/media/codec2/sfplugin/CCodecBufferChannel.cpp b/media/codec2/sfplugin/CCodecBufferChannel.cpp
index 3e74d8f..1cbfc75 100644
--- a/media/codec2/sfplugin/CCodecBufferChannel.cpp
+++ b/media/codec2/sfplugin/CCodecBufferChannel.cpp
@@ -228,17 +228,23 @@
status_t CCodecBufferChannel::setInputSurface(
const std::shared_ptr<InputSurfaceWrapper> &surface) {
ALOGV("[%s] setInputSurface", mName);
- Mutexed<std::shared_ptr<InputSurfaceWrapper>>::Locked inputSurface(mInputSurface);
- *inputSurface = surface;
- return (*inputSurface)->connect(mComponent);
+ if (!surface) {
+ ALOGE("[%s] setInputSurface: surface must not be null", mName);
+ return BAD_VALUE;
+ }
+ Mutexed<InputSurface>::Locked inputSurface(mInputSurface);
+ inputSurface->numProcessingBuffersBalance = 0;
+ inputSurface->surface = surface;
+ mHasInputSurface = true;
+ return inputSurface->surface->connect(mComponent);
}
status_t CCodecBufferChannel::signalEndOfInputStream() {
- Mutexed<std::shared_ptr<InputSurfaceWrapper>>::Locked inputSurface(mInputSurface);
- if ((*inputSurface) == nullptr) {
+ Mutexed<InputSurface>::Locked inputSurface(mInputSurface);
+ if (inputSurface->surface == nullptr) {
return INVALID_OPERATION;
}
- return (*inputSurface)->signalEndOfInputStream();
+ return inputSurface->surface->signalEndOfInputStream();
}
status_t CCodecBufferChannel::queueInputBufferInternal(
@@ -1063,19 +1069,36 @@
if (mInputMetEos) {
return;
}
- {
+ int64_t numOutputSlots = 0;
+ bool outputFull = [this, &numOutputSlots]() {
Mutexed<Output>::Locked output(mOutput);
- if (!output->buffers ||
- output->buffers->hasPending() ||
+ if (!output->buffers) {
+ ALOGV("[%s] feedInputBufferIfAvailableInternal: "
+ "return because output buffers are null", mName);
+ return true;
+ }
+ numOutputSlots = int64_t(output->numSlots);
+ if (output->buffers->hasPending() ||
(!output->bounded && output->buffers->numActiveSlots() >= output->numSlots)) {
- return;
+ ALOGV("[%s] feedInputBufferIfAvailableInternal: "
+ "return because there are no room for more output buffers", mName);
+ return true;
+ }
+ return false;
+ }();
+ if (android::media::codec::provider_->input_surface_throttle()) {
+ Mutexed<InputSurface>::Locked inputSurface(mInputSurface);
+ if (inputSurface->surface) {
+ if (inputSurface->numProcessingBuffersBalance <= numOutputSlots) {
+ ++inputSurface->numProcessingBuffersBalance;
+ ALOGV("[%s] feedInputBufferIfAvailableInternal: numProcessingBuffersBalance = %lld",
+ mName, static_cast<long long>(inputSurface->numProcessingBuffersBalance));
+ inputSurface->surface->onInputBufferEmptied();
+ }
}
}
- if (android::media::codec::provider_->input_surface_throttle()) {
- Mutexed<std::shared_ptr<InputSurfaceWrapper>>::Locked inputSurface(mInputSurface);
- if ((*inputSurface) != nullptr) {
- (*inputSurface)->onInputBufferEmptied();
- }
+ if (outputFull) {
+ return;
}
size_t numActiveSlots = 0;
while (!mPipelineWatcher.lock()->pipelineFull()) {
@@ -1704,7 +1727,7 @@
&& (hasCryptoOrDescrambler() || conforming)) {
input->buffers.reset(new SlotInputBuffers(mName));
} else if (graphic) {
- if (mInputSurface.lock()->get()) {
+ if (mHasInputSurface) {
input->buffers.reset(new DummyInputBuffers(mName));
} else if (mMetaMode == MODE_ANW) {
input->buffers.reset(new GraphicMetadataInputBuffers(mName));
@@ -1987,7 +2010,7 @@
status_t CCodecBufferChannel::prepareInitialInputBuffers(
std::map<size_t, sp<MediaCodecBuffer>> *clientInputBuffers, bool retry) {
- if (mInputSurface.lock()->get()) {
+ if (mHasInputSurface) {
return OK;
}
@@ -2113,9 +2136,13 @@
void CCodecBufferChannel::reset() {
stop();
- mInputSurface.lock()->reset();
mPipelineWatcher.lock()->flush();
{
+ mHasInputSurface = false;
+ Mutexed<InputSurface>::Locked inputSurface(mInputSurface);
+ inputSurface->surface.reset();
+ }
+ {
Mutexed<Input>::Locked input(mInput);
input->buffers.reset(new DummyInputBuffers(""));
input->extraBuffers.flush();
@@ -2208,9 +2235,6 @@
void CCodecBufferChannel::onInputBufferDone(
uint64_t frameIndex, size_t arrayIndex) {
- if (mInputSurface.lock()->get()) {
- return;
- }
std::shared_ptr<C2Buffer> buffer =
mPipelineWatcher.lock()->onInputBufferReleased(frameIndex, arrayIndex);
bool newInputSlotAvailable = false;
@@ -2265,8 +2289,7 @@
notifyClient = false;
}
- bool hasInputSurface = (mInputSurface.lock()->get() != nullptr);
- if (!hasInputSurface && (work->worklets.size() != 1u
+ if (!mHasInputSurface && (work->worklets.size() != 1u
|| !work->worklets.front()
|| !(work->worklets.front()->output.flags &
C2FrameData::FLAG_INCOMPLETE))) {
@@ -2475,7 +2498,7 @@
c2_cntr64_t timestamp =
worklet->output.ordinal.timestamp + work->input.ordinal.customOrdinal
- work->input.ordinal.timestamp;
- if (hasInputSurface) {
+ if (mHasInputSurface) {
// When using input surface we need to restore the original input timestamp.
timestamp = work->input.ordinal.customOrdinal;
}
@@ -2633,6 +2656,12 @@
outBuffer->meta()->setObject("accessUnitInfo", obj);
}
}
+ if (mHasInputSurface && android::media::codec::provider_->input_surface_throttle()) {
+ Mutexed<InputSurface>::Locked inputSurface(mInputSurface);
+ --inputSurface->numProcessingBuffersBalance;
+ ALOGV("[%s] onOutputBufferAvailable: numProcessingBuffersBalance = %lld",
+ mName, static_cast<long long>(inputSurface->numProcessingBuffersBalance));
+ }
mCallback->onOutputBufferAvailable(index, outBuffer);
break;
}
@@ -2802,7 +2831,7 @@
}
void CCodecBufferChannel::setInfoBuffer(const std::shared_ptr<C2InfoBuffer> &buffer) {
- if (mInputSurface.lock()->get() == nullptr) {
+ if (!mHasInputSurface) {
mInfoBuffers.push_back(buffer);
} else {
std::list<std::unique_ptr<C2Work>> items;
diff --git a/media/codec2/sfplugin/CCodecBufferChannel.h b/media/codec2/sfplugin/CCodecBufferChannel.h
index f5e268a..4d296fd 100644
--- a/media/codec2/sfplugin/CCodecBufferChannel.h
+++ b/media/codec2/sfplugin/CCodecBufferChannel.h
@@ -391,7 +391,21 @@
};
Mutexed<BlockPools> mBlockPools;
- Mutexed<std::shared_ptr<InputSurfaceWrapper>> mInputSurface;
+ std::atomic_bool mHasInputSurface;
+ struct InputSurface {
+ std::shared_ptr<InputSurfaceWrapper> surface;
+ // This variable tracks the number of buffers processing
+ // in the input surface and codec by counting the # of buffers to
+ // be filled in and queued from the input surface and the # of
+ // buffers generated from the codec.
+ //
+ // Note that this variable can go below 0, because it does not take
+ // account the number of buffers initially in the buffer queue at
+ // start. This is okay, as we only track how many more we allow
+ // from the initial state.
+ int64_t numProcessingBuffersBalance;
+ };
+ Mutexed<InputSurface> mInputSurface;
MetaMode mMetaMode;
diff --git a/media/libaudioclient/Android.bp b/media/libaudioclient/Android.bp
index b1517bb..61204ae 100644
--- a/media/libaudioclient/Android.bp
+++ b/media/libaudioclient/Android.bp
@@ -158,6 +158,7 @@
"framework-permission-aidl-cpp",
"libbinder",
"libmediametrics",
+ "libmediautils",
"spatializer-aidl-cpp",
],
diff --git a/media/libaudioclient/TrackPlayerBase.cpp b/media/libaudioclient/TrackPlayerBase.cpp
index 4fc1c44..bc38251 100644
--- a/media/libaudioclient/TrackPlayerBase.cpp
+++ b/media/libaudioclient/TrackPlayerBase.cpp
@@ -38,12 +38,12 @@
player_type_t playerType, audio_usage_t usage,
audio_session_t sessionId) {
PlayerBase::init(playerType, usage, sessionId);
- mAudioTrack = pat;
- if (mAudioTrack != 0) {
+ mAudioTrack.store(pat);
+ if (pat != 0) {
mCallbackHandle = callback;
mSelfAudioDeviceCallback = new SelfAudioDeviceCallback(*this);
- mAudioTrack->addAudioDeviceCallback(mSelfAudioDeviceCallback);
- mAudioTrack->setPlayerIId(mPIId); // set in PlayerBase::init().
+ pat->addAudioDeviceCallback(mSelfAudioDeviceCallback);
+ pat->setPlayerIId(mPIId); // set in PlayerBase::init().
}
}
@@ -65,12 +65,15 @@
}
void TrackPlayerBase::doDestroy() {
- if (mAudioTrack != 0) {
- mAudioTrack->stop();
- mAudioTrack->removeAudioDeviceCallback(mSelfAudioDeviceCallback);
+ sp<AudioTrack> audioTrack = getAudioTrack();
+
+ // Note that there may still be another reference in post-unlock phase of SetPlayState
+ clearAudioTrack();
+
+ if (audioTrack != 0) {
+ audioTrack->stop();
+ audioTrack->removeAudioDeviceCallback(mSelfAudioDeviceCallback);
mSelfAudioDeviceCallback.clear();
- // Note that there may still be another reference in post-unlock phase of SetPlayState
- mAudioTrack.clear();
}
}
@@ -87,16 +90,16 @@
// Implementation of IPlayer
status_t TrackPlayerBase::playerStart() {
status_t status = NO_INIT;
- if (mAudioTrack != 0) {
- status = mAudioTrack->start();
+ if (sp<AudioTrack> audioTrack = getAudioTrack(); audioTrack != 0) {
+ status = audioTrack->start();
}
return status;
}
status_t TrackPlayerBase::playerPause() {
status_t status = NO_INIT;
- if (mAudioTrack != 0) {
- mAudioTrack->pause();
+ if (sp<AudioTrack> audioTrack = getAudioTrack(); audioTrack != 0) {
+ audioTrack->pause();
status = NO_ERROR;
}
return status;
@@ -105,8 +108,8 @@
status_t TrackPlayerBase::playerStop() {
status_t status = NO_INIT;
- if (mAudioTrack != 0) {
- mAudioTrack->stop();
+ if (sp<AudioTrack> audioTrack = getAudioTrack(); audioTrack != 0) {
+ audioTrack->stop();
status = NO_ERROR;
}
return status;
@@ -118,10 +121,10 @@
status_t TrackPlayerBase::doSetVolume() {
status_t status = NO_INIT;
- if (mAudioTrack != 0) {
+ if (sp<AudioTrack> audioTrack = getAudioTrack(); audioTrack != 0) {
float tl = mPlayerVolumeL * mPanMultiplierL * mVolumeMultiplierL;
float tr = mPlayerVolumeR * mPanMultiplierR * mVolumeMultiplierR;
- mAudioTrack->setVolume(tl, tr);
+ audioTrack->setVolume(tl, tr);
status = NO_ERROR;
}
return status;
@@ -140,10 +143,9 @@
if (s != OK) {
return binderStatusFromStatusT(s);
}
-
- if (mAudioTrack != 0) {
+ if (sp<AudioTrack> audioTrack = getAudioTrack(); audioTrack != 0) {
ALOGD("TrackPlayerBase::applyVolumeShaper() from IPlayer");
- VolumeShaper::Status status = mAudioTrack->applyVolumeShaper(spConfiguration, spOperation);
+ VolumeShaper::Status status = audioTrack->applyVolumeShaper(spConfiguration, spOperation);
if (status < 0) { // a non-negative value is the volume shaper id.
ALOGE("TrackPlayerBase::applyVolumeShaper() failed with status %d", status);
}
diff --git a/media/libaudioclient/include/media/TrackPlayerBase.h b/media/libaudioclient/include/media/TrackPlayerBase.h
index fe88116..8df9ff8 100644
--- a/media/libaudioclient/include/media/TrackPlayerBase.h
+++ b/media/libaudioclient/include/media/TrackPlayerBase.h
@@ -19,6 +19,7 @@
#include <media/AudioTrack.h>
#include <media/PlayerBase.h>
+#include <mediautils/Synchronization.h>
namespace android {
@@ -37,10 +38,11 @@
const media::VolumeShaperConfiguration& configuration,
const media::VolumeShaperOperation& operation);
- //FIXME move to protected field, so far made public to minimize changes to AudioTrack logic
- sp<AudioTrack> mAudioTrack;
+ sp<AudioTrack> getAudioTrack() { return mAudioTrack.load(); }
- void setPlayerVolume(float vl, float vr);
+ void clearAudioTrack() { mAudioTrack.store(nullptr); }
+
+ void setPlayerVolume(float vl, float vr);
protected:
@@ -68,6 +70,7 @@
float mPlayerVolumeL, mPlayerVolumeR;
sp<AudioTrack::IAudioTrackCallback> mCallbackHandle;
sp<SelfAudioDeviceCallback> mSelfAudioDeviceCallback;
+ mediautils::atomic_sp<AudioTrack> mAudioTrack;
};
} // namespace android
diff --git a/media/libaudioclient/tests/trackplayerbase_tests.cpp b/media/libaudioclient/tests/trackplayerbase_tests.cpp
index 7317bf0..a4dba9b 100644
--- a/media/libaudioclient/tests/trackplayerbase_tests.cpp
+++ b/media/libaudioclient/tests/trackplayerbase_tests.cpp
@@ -54,7 +54,7 @@
mPlayer = new TrackPlayer();
mPlayer->init(track.get(), mPlayer, PLAYER_TYPE_AAUDIO, AUDIO_USAGE_MEDIA,
AUDIO_SESSION_NONE);
- sp<AudioTrack> playerTrack = mPlayer->mAudioTrack;
+ sp<AudioTrack> playerTrack = mPlayer->getAudioTrack();
ASSERT_EQ(playerTrack->initCheck(), NO_ERROR);
mBufferSize = mFrameCount * playerTrack->frameSize();
@@ -74,7 +74,7 @@
void playBuffer() {
bool blocking = true;
- ssize_t nbytes = mPlayer->mAudioTrack->write(mBuffer.data(), mBufferSize, blocking);
+ ssize_t nbytes = mPlayer->getAudioTrack()->write(mBuffer.data(), mBufferSize, blocking);
EXPECT_EQ(nbytes, mBufferSize) << "Did not write all data in blocking mode";
}
diff --git a/media/libaudiohal/impl/EffectConversionHelperAidl.cpp b/media/libaudiohal/impl/EffectConversionHelperAidl.cpp
index a13903b..f719d97 100644
--- a/media/libaudiohal/impl/EffectConversionHelperAidl.cpp
+++ b/media/libaudiohal/impl/EffectConversionHelperAidl.cpp
@@ -532,5 +532,13 @@
AudioChannelLayout::LAYOUT_HAPTIC_AB /* mask */);
}
+size_t EffectConversionHelperAidl::getInputChannelCount() const {
+ return getChannelCount(mCommon.input.base.channelMask);
+}
+
+size_t EffectConversionHelperAidl::getOutputChannelCount() const {
+ return getChannelCount(mCommon.output.base.channelMask);
+}
+
} // namespace effect
} // namespace android
diff --git a/media/libaudiohal/impl/EffectConversionHelperAidl.h b/media/libaudiohal/impl/EffectConversionHelperAidl.h
index 50b47a9..e9e9fc2 100644
--- a/media/libaudiohal/impl/EffectConversionHelperAidl.h
+++ b/media/libaudiohal/impl/EffectConversionHelperAidl.h
@@ -51,6 +51,8 @@
size_t getAudioChannelCount() const;
size_t getHapticChannelCount() const;
+ size_t getInputChannelCount() const;
+ size_t getOutputChannelCount() const;
uint8_t mOutputAccessMode = EFFECT_BUFFER_ACCESS_WRITE;
diff --git a/media/libaudiohal/impl/EffectHalAidl.cpp b/media/libaudiohal/impl/EffectHalAidl.cpp
index ea4dbf6..f68dd8a 100644
--- a/media/libaudiohal/impl/EffectHalAidl.cpp
+++ b/media/libaudiohal/impl/EffectHalAidl.cpp
@@ -75,7 +75,12 @@
mEffect(effect),
mSessionId(sessionId),
mIoId(ioId),
- mIsProxyEffect(isProxyEffect) {
+ mIsProxyEffect(isProxyEffect),
+ mHalVersion([factory]() {
+ int version = 0;
+ // use factory HAL version because effect can be an EffectProxy instance
+ return factory->getInterfaceVersion(&version).isOk() ? version : 0;
+ }()) {
assert(mFactory != nullptr);
assert(mEffect != nullptr);
createAidlConversion(effect, sessionId, ioId, desc);
@@ -159,6 +164,7 @@
mConversion = std::make_unique<android::effect::AidlConversionVendorExtension>(
effect, sessionId, ioId, desc, mIsProxyEffect);
}
+ mEffectName = mConversion->getDescriptor().common.name;
return OK;
}
@@ -174,100 +180,155 @@
// write to input FMQ here, wait for statusMQ STATUS_OK, and read from output FMQ
status_t EffectHalAidl::process() {
- const std::string effectName = mConversion->getDescriptor().common.name;
State state = State::INIT;
if (mConversion->isBypassing() || !mEffect->getState(&state).isOk() ||
state != State::PROCESSING) {
- ALOGI("%s skipping %s process because it's %s", __func__, effectName.c_str(),
+ ALOGI("%s skipping process because it's %s", mEffectName.c_str(),
mConversion->isBypassing()
? "bypassing"
: aidl::android::hardware::audio::effect::toString(state).c_str());
return -ENODATA;
}
- // check if the DataMq needs any update, timeout at 1ns to avoid being blocked
- auto efGroup = mConversion->getEventFlagGroup();
+ const std::shared_ptr<android::hardware::EventFlag> efGroup = mConversion->getEventFlagGroup();
if (!efGroup) {
- ALOGE("%s invalid efGroup", __func__);
+ ALOGE("%s invalid efGroup", mEffectName.c_str());
return INVALID_OPERATION;
}
- // use IFactory HAL version because IEffect can be an EffectProxy instance
- static const int halVersion = [&]() {
- int version = 0;
- return mFactory->getInterfaceVersion(&version).isOk() ? version : 0;
- }();
+ // reopen if halVersion >= kReopenSupportedVersion and receive kEventFlagDataMqUpdate
+ RETURN_STATUS_IF_ERROR(maybeReopen(efGroup));
+ const size_t samplesWritten = writeToHalInputFmqAndSignal(efGroup);
+ if (0 == samplesWritten) {
+ return INVALID_OPERATION;
+ }
- if (uint32_t efState = 0; halVersion >= kReopenSupportedVersion &&
- ::android::OK == efGroup->wait(kEventFlagDataMqUpdate, &efState,
+ RETURN_STATUS_IF_ERROR(waitHalStatusFmq(samplesWritten));
+ RETURN_STATUS_IF_ERROR(readFromHalOutputFmq(samplesWritten));
+ return OK;
+}
+
+status_t EffectHalAidl::maybeReopen(
+ const std::shared_ptr<android::hardware::EventFlag>& efGroup) const {
+ if (mHalVersion < kReopenSupportedVersion) {
+ return OK;
+ }
+
+ // check if the DataMq needs any update, timeout at 1ns to avoid being blocked
+ if (uint32_t efState = 0; ::android::OK == efGroup->wait(kEventFlagDataMqUpdate, &efState,
1 /* ns */, true /* retry */) &&
efState & kEventFlagDataMqUpdate) {
- ALOGD("%s %s V%d receive dataMQUpdate eventFlag from HAL", __func__, effectName.c_str(),
- halVersion);
-
- mConversion->reopen();
+ ALOGD("%s V%d receive dataMQUpdate eventFlag from HAL", mEffectName.c_str(), mHalVersion);
+ return mConversion->reopen();
}
- auto statusQ = mConversion->getStatusMQ();
- auto inputQ = mConversion->getInputMQ();
- auto outputQ = mConversion->getOutputMQ();
- if (!statusQ || !statusQ->isValid() || !inputQ || !inputQ->isValid() || !outputQ ||
- !outputQ->isValid()) {
- ALOGE("%s invalid FMQ [Status %d I %d O %d]", __func__, statusQ ? statusQ->isValid() : 0,
- inputQ ? inputQ->isValid() : 0, outputQ ? outputQ->isValid() : 0);
- return INVALID_OPERATION;
+ return OK;
+}
+
+size_t EffectHalAidl::writeToHalInputFmqAndSignal(
+ const std::shared_ptr<android::hardware::EventFlag>& efGroup) const {
+ const auto inputQ = mConversion->getInputMQ();
+ if (!inputQ || !inputQ->isValid()) {
+ ALOGE("%s invalid input FMQ", mEffectName.c_str());
+ return 0;
}
- size_t available = inputQ->availableToWrite();
- const size_t floatsToWrite = std::min(available, mInBuffer->getSize() / sizeof(float));
- if (floatsToWrite == 0) {
- ALOGE("%s not able to write, floats in buffer %zu, space in FMQ %zu", __func__,
- mInBuffer->getSize() / sizeof(float), available);
- return INVALID_OPERATION;
+ const size_t fmqSpaceSamples = inputQ->availableToWrite();
+ const size_t samplesInBuffer =
+ mInBuffer->audioBuffer()->frameCount * mConversion->getInputChannelCount();
+ const size_t samplesToWrite = std::min(fmqSpaceSamples, samplesInBuffer);
+ if (samplesToWrite == 0) {
+ ALOGE("%s not able to write, samplesInBuffer %zu, fmqSpaceSamples %zu", mEffectName.c_str(),
+ samplesInBuffer, fmqSpaceSamples);
+ return 0;
}
- if (!mInBuffer->audioBuffer() ||
- !inputQ->write((float*)mInBuffer->audioBuffer()->f32, floatsToWrite)) {
- ALOGE("%s failed to write %zu floats from audiobuffer %p to inputQ [avail %zu]", __func__,
- floatsToWrite, mInBuffer->audioBuffer(), inputQ->availableToWrite());
- return INVALID_OPERATION;
+
+ const float* const inputRawBuffer = static_cast<const float*>(mInBuffer->audioBuffer()->f32);
+ if (!inputQ->write(inputRawBuffer, samplesToWrite)) {
+ ALOGE("%s failed to write %zu samples to inputQ [avail %zu]", mEffectName.c_str(),
+ samplesToWrite, inputQ->availableToWrite());
+ return 0;
}
// for V2 audio effect HAL, expect different EventFlag to avoid bit conflict with FMQ_NOT_EMPTY
- efGroup->wake(halVersion >= kReopenSupportedVersion ? kEventFlagDataMqNotEmpty
- : kEventFlagNotEmpty);
+ efGroup->wake(mHalVersion >= kReopenSupportedVersion ? kEventFlagDataMqNotEmpty
+ : kEventFlagNotEmpty);
+ return samplesToWrite;
+}
+
+void EffectHalAidl::writeHapticGeneratorData(size_t totalSamples, float* const outputRawBuffer,
+ float* const fmqOutputBuffer) const {
+ const auto audioChNum = mConversion->getAudioChannelCount();
+ const auto audioSamples =
+ totalSamples * audioChNum / (audioChNum + mConversion->getHapticChannelCount());
+
+ static constexpr float kHalFloatSampleLimit = 2.0f;
+ // for HapticGenerator, the input data buffer will be updated
+ float* const inputRawBuffer = static_cast<float*>(mInBuffer->audioBuffer()->f32);
+ // accumulate or copy input to output, haptic samples remains all zero
+ if (mConversion->mOutputAccessMode == EFFECT_BUFFER_ACCESS_ACCUMULATE) {
+ accumulate_float(outputRawBuffer, inputRawBuffer, audioSamples);
+ } else {
+ memcpy_to_float_from_float_with_clamping(outputRawBuffer, inputRawBuffer, audioSamples,
+ kHalFloatSampleLimit);
+ }
+ // append the haptic sample at the end of input audio samples
+ memcpy_to_float_from_float_with_clamping(inputRawBuffer + audioSamples,
+ fmqOutputBuffer + audioSamples,
+ totalSamples - audioSamples, kHalFloatSampleLimit);
+}
+
+status_t EffectHalAidl::waitHalStatusFmq(size_t samplesWritten) const {
+ const auto statusQ = mConversion->getStatusMQ();
+ if (const bool statusValid = statusQ && statusQ->isValid(); !statusValid) {
+ ALOGE("%s statusFMQ %s", mEffectName.c_str(), statusValid ? "valid" : "invalid");
+ return INVALID_OPERATION;
+ }
IEffect::Status retStatus{};
if (!statusQ->readBlocking(&retStatus, 1)) {
- ALOGE("%s %s V%d read status from status FMQ failed", __func__, effectName.c_str(),
- halVersion);
+ ALOGE("%s V%d read status from status FMQ failed", mEffectName.c_str(), mHalVersion);
return INVALID_OPERATION;
}
- if (retStatus.status != OK || (size_t)retStatus.fmqConsumed != floatsToWrite ||
+ if (retStatus.status != OK || (size_t)retStatus.fmqConsumed != samplesWritten ||
retStatus.fmqProduced == 0) {
- ALOGE("%s read status failed: %s, consumed %d (of %zu) produced %d", __func__,
- retStatus.toString().c_str(), retStatus.fmqConsumed, floatsToWrite,
- retStatus.fmqProduced);
+ ALOGE("%s read status failed: %s, FMQ consumed %d (of %zu) produced %d",
+ mEffectName.c_str(), retStatus.toString().c_str(), retStatus.fmqConsumed,
+ samplesWritten, retStatus.fmqProduced);
return INVALID_OPERATION;
}
- available = outputQ->availableToRead();
- const size_t floatsToRead = std::min(available, mOutBuffer->getSize() / sizeof(float));
- if (floatsToRead == 0) {
- ALOGE("%s not able to read, buffer space %zu, floats in FMQ %zu", __func__,
- mOutBuffer->getSize() / sizeof(float), available);
+ return OK;
+}
+
+status_t EffectHalAidl::readFromHalOutputFmq(size_t samplesWritten) const {
+ const auto outputQ = mConversion->getOutputMQ();
+ if (const bool outputValid = outputQ && outputQ->isValid(); !outputValid) {
+ ALOGE("%s outputFMQ %s", mEffectName.c_str(), outputValid ? "valid" : "invalid");
return INVALID_OPERATION;
}
- float *outputRawBuffer = mOutBuffer->audioBuffer()->f32;
+ const size_t fmqProducedSamples = outputQ->availableToRead();
+ const size_t bufferSpaceSamples =
+ mOutBuffer->audioBuffer()->frameCount * mConversion->getOutputChannelCount();
+ const size_t samplesToRead = std::min(fmqProducedSamples, bufferSpaceSamples);
+ if (samplesToRead == 0) {
+ ALOGE("%s unable to read, bufferSpace %zu, fmqProduced %zu samplesWritten %zu",
+ mEffectName.c_str(), bufferSpaceSamples, fmqProducedSamples, samplesWritten);
+ return INVALID_OPERATION;
+ }
+
+ float* const outputRawBuffer = static_cast<float*>(mOutBuffer->audioBuffer()->f32);
+ float* fmqOutputBuffer = outputRawBuffer;
std::vector<float> tempBuffer;
// keep original data in the output buffer for accumulate mode or HapticGenerator effect
if (mConversion->mOutputAccessMode == EFFECT_BUFFER_ACCESS_ACCUMULATE || mIsHapticGenerator) {
- tempBuffer.resize(floatsToRead);
- outputRawBuffer = tempBuffer.data();
+ tempBuffer.resize(samplesToRead, 0);
+ fmqOutputBuffer = tempBuffer.data();
}
// always read floating point data for AIDL
- if (!outputQ->read(outputRawBuffer, floatsToRead)) {
- ALOGE("%s failed to read %zu from outputQ to audioBuffer %p", __func__, floatsToRead,
- mOutBuffer->audioBuffer());
+ if (!outputQ->read(fmqOutputBuffer, samplesToRead)) {
+ ALOGE("%s failed to read %zu from outputQ to audioBuffer %p", mEffectName.c_str(),
+ samplesToRead, fmqOutputBuffer);
return INVALID_OPERATION;
}
@@ -276,26 +337,10 @@
// offset as input buffer, here we skip the audio samples in output FMQ and append haptic
// samples to the end of input buffer
if (mIsHapticGenerator) {
- static constexpr float kHalFloatSampleLimit = 2.0f;
- assert(floatsToRead == floatsToWrite);
- const auto audioChNum = mConversion->getAudioChannelCount();
- const auto audioSamples =
- floatsToWrite * audioChNum / (audioChNum + mConversion->getHapticChannelCount());
- // accumulate or copy input to output, haptic samples remains all zero
- if (mConversion->mOutputAccessMode == EFFECT_BUFFER_ACCESS_ACCUMULATE) {
- accumulate_float(mOutBuffer->audioBuffer()->f32, mInBuffer->audioBuffer()->f32,
- audioSamples);
- } else {
- memcpy_to_float_from_float_with_clamping(mOutBuffer->audioBuffer()->f32,
- mInBuffer->audioBuffer()->f32, audioSamples,
- kHalFloatSampleLimit);
- }
- // append the haptic sample at the end of input audio samples
- memcpy_to_float_from_float_with_clamping(mInBuffer->audioBuffer()->f32 + audioSamples,
- outputRawBuffer + audioSamples,
- floatsToRead - audioSamples, kHalFloatSampleLimit);
+ assert(samplesRead == samplesWritten);
+ writeHapticGeneratorData(samplesToRead, outputRawBuffer, fmqOutputBuffer);
} else if (mConversion->mOutputAccessMode == EFFECT_BUFFER_ACCESS_ACCUMULATE) {
- accumulate_float(mOutBuffer->audioBuffer()->f32, outputRawBuffer, floatsToRead);
+ accumulate_float(outputRawBuffer, fmqOutputBuffer, samplesToRead);
}
return OK;
diff --git a/media/libaudiohal/impl/EffectHalAidl.h b/media/libaudiohal/impl/EffectHalAidl.h
index 4f7de7c..c3982a7 100644
--- a/media/libaudiohal/impl/EffectHalAidl.h
+++ b/media/libaudiohal/impl/EffectHalAidl.h
@@ -73,7 +73,9 @@
const int32_t mSessionId;
const int32_t mIoId;
const bool mIsProxyEffect;
+ const int mHalVersion;
bool mIsHapticGenerator = false;
+ std::string mEffectName;
std::unique_ptr<EffectConversionHelperAidl> mConversion;
@@ -93,6 +95,14 @@
bool setEffectReverse(bool reverse);
bool needUpdateReturnParam(uint32_t cmdCode);
+ status_t maybeReopen(const std::shared_ptr<android::hardware::EventFlag>& efGroup) const;
+ void writeHapticGeneratorData(size_t totalSamples, float* const outputRawBuffer,
+ float* const fmqOutputBuffer) const;
+ size_t writeToHalInputFmqAndSignal(
+ const std::shared_ptr<android::hardware::EventFlag>& efGroup) const;
+ status_t waitHalStatusFmq(size_t samplesWritten) const;
+ status_t readFromHalOutputFmq(size_t samplesWritten) const;
+
// The destructor automatically releases the effect.
virtual ~EffectHalAidl();
};
diff --git a/media/libaudioprocessing/AudioMixerOps.h b/media/libaudioprocessing/AudioMixerOps.h
index ab6a8b6..8f60d29 100644
--- a/media/libaudioprocessing/AudioMixerOps.h
+++ b/media/libaudioprocessing/AudioMixerOps.h
@@ -347,6 +347,7 @@
[6] = AUDIO_CHANNEL_OUT_5POINT1,
[7] = AUDIO_CHANNEL_OUT_6POINT1,
[8] = AUDIO_CHANNEL_OUT_7POINT1,
+ [10] = AUDIO_CHANNEL_OUT_5POINT1POINT4,
[12] = AUDIO_CHANNEL_OUT_7POINT1POINT4,
[14] = AUDIO_CHANNEL_OUT_9POINT1POINT4,
[16] = AUDIO_CHANNEL_OUT_9POINT1POINT6,
diff --git a/media/libaudioprocessing/tests/mixerops_tests.cpp b/media/libaudioprocessing/tests/mixerops_tests.cpp
index 2500ba9..235129f 100644
--- a/media/libaudioprocessing/tests/mixerops_tests.cpp
+++ b/media/libaudioprocessing/tests/mixerops_tests.cpp
@@ -154,6 +154,9 @@
TEST(mixerops, stereovolume_8) {
MixerOpsBasicTest<MIXTYPE_MULTI_STEREOVOL, 8>::testStereoVolume();
}
+TEST(mixerops, stereovolume_10) {
+ MixerOpsBasicTest<MIXTYPE_MULTI_STEREOVOL, 10>::testStereoVolume();
+}
TEST(mixerops, stereovolume_12) {
if constexpr (FCC_LIMIT >= 12) { // NOTE: FCC_LIMIT is an enum, so can't #if
MixerOpsBasicTest<MIXTYPE_MULTI_STEREOVOL, 12>::testStereoVolume();
diff --git a/media/module/codecserviceregistrant/CodecServiceRegistrant.cpp b/media/module/codecserviceregistrant/CodecServiceRegistrant.cpp
index 42fd94e..433b5e9 100644
--- a/media/module/codecserviceregistrant/CodecServiceRegistrant.cpp
+++ b/media/module/codecserviceregistrant/CodecServiceRegistrant.cpp
@@ -790,47 +790,6 @@
}
using namespace ::android::hardware::media::c2;
-
- if (!ionPropertiesDefined()) {
- using IComponentStore =
- ::android::hardware::media::c2::V1_0::IComponentStore;
- std::string const preferredStoreName = "default";
- if (aidlSelected) {
- std::shared_ptr<c2_aidl::IComponentStore> preferredStore;
- if (__builtin_available(android __ANDROID_API_S__, *)) {
- std::string instanceName = ::android::base::StringPrintf(
- "%s/%s", c2_aidl::IComponentStore::descriptor, preferredStoreName.c_str());
- if (AServiceManager_isDeclared(instanceName.c_str())) {
- preferredStore = c2_aidl::IComponentStore::fromBinder(::ndk::SpAIBinder(
- AServiceManager_waitForService(instanceName.c_str())));
- }
- }
- if (preferredStore) {
- ::android::SetPreferredCodec2ComponentStore(
- std::make_shared<H2C2ComponentStore>(preferredStore));
- LOG(INFO) <<
- "Preferred Codec2 AIDL store is set to \"" <<
- preferredStoreName << "\".";
- } else {
- LOG(INFO) <<
- "Preferred Codec2 AIDL store is defaulted to \"software\".";
- }
- } else {
- sp<IComponentStore> preferredStore =
- IComponentStore::getService(preferredStoreName.c_str());
- if (preferredStore) {
- ::android::SetPreferredCodec2ComponentStore(
- std::make_shared<H2C2ComponentStore>(preferredStore));
- LOG(INFO) <<
- "Preferred Codec2 HIDL store is set to \"" <<
- preferredStoreName << "\".";
- } else {
- LOG(INFO) <<
- "Preferred Codec2 HIDL store is defaulted to \"software\".";
- }
- }
- }
-
bool registered = false;
const std::string aidlServiceName =
std::string(c2_aidl::IComponentStore::descriptor) + "/software";
@@ -876,6 +835,48 @@
" so it is not being registered with hwservicemanager.";
}
+ // Preferred store must be set after the store is registered to ensure that
+ // the correct preferred store is set.
+ if (!ionPropertiesDefined()) {
+ using IComponentStore =
+ ::android::hardware::media::c2::V1_0::IComponentStore;
+ std::string const preferredStoreName = "default";
+ if (aidlSelected) {
+ std::shared_ptr<c2_aidl::IComponentStore> preferredStore;
+ if (__builtin_available(android __ANDROID_API_S__, *)) {
+ std::string instanceName = ::android::base::StringPrintf(
+ "%s/%s", c2_aidl::IComponentStore::descriptor, preferredStoreName.c_str());
+ if (AServiceManager_isDeclared(instanceName.c_str())) {
+ preferredStore = c2_aidl::IComponentStore::fromBinder(::ndk::SpAIBinder(
+ AServiceManager_waitForService(instanceName.c_str())));
+ }
+ }
+ if (preferredStore) {
+ ::android::SetPreferredCodec2ComponentStore(
+ std::make_shared<H2C2ComponentStore>(preferredStore));
+ LOG(INFO) <<
+ "Preferred Codec2 AIDL store is set to \"" <<
+ preferredStoreName << "\".";
+ } else {
+ LOG(INFO) <<
+ "Preferred Codec2 AIDL store is defaulted to \"software\".";
+ }
+ } else {
+ sp<IComponentStore> preferredStore =
+ IComponentStore::getService(preferredStoreName.c_str());
+ if (preferredStore) {
+ ::android::SetPreferredCodec2ComponentStore(
+ std::make_shared<H2C2ComponentStore>(preferredStore));
+ LOG(INFO) <<
+ "Preferred Codec2 HIDL store is set to \"" <<
+ preferredStoreName << "\".";
+ } else {
+ LOG(INFO) <<
+ "Preferred Codec2 HIDL store is defaulted to \"software\".";
+ }
+ }
+ }
+
if (registered) {
LOG(INFO) << "Software Codec2 service created and registered.";
}
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 2c1976b..537a097 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -3948,8 +3948,12 @@
// TODO: We could check compatibility of the secondaryThread with the PatchTrack
// for fast usage: thread has fast mixer, sample rate matches, etc.;
// for now, we exclude fast tracks by removing the Fast flag.
+ constexpr audio_output_flags_t kIncompatiblePatchTrackFlags =
+ static_cast<audio_output_flags_t>(AUDIO_OUTPUT_FLAG_FAST
+ | AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
+
const audio_output_flags_t outputFlags =
- (audio_output_flags_t)(track->getOutputFlags() & ~AUDIO_OUTPUT_FLAG_FAST);
+ (audio_output_flags_t)(track->getOutputFlags() & ~kIncompatiblePatchTrackFlags);
sp<IAfPatchTrack> patchTrack = IAfPatchTrack::create(secondaryThread,
track->streamType(),
track->sampleRate(),
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioPolicyConfig.h b/services/audiopolicy/common/managerdefinitions/include/AudioPolicyConfig.h
index 794c7c0..b193cb8 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioPolicyConfig.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioPolicyConfig.h
@@ -16,10 +16,10 @@
#pragma once
+#include <optional>
#include <string>
#include <unordered_map>
#include <unordered_set>
-#include <vector>
#include <DeviceDescriptor.h>
#include <HwModule.h>
@@ -141,6 +141,12 @@
void setDefault();
+ void setUseDeepBufferForMediaOverrideForTests(bool useDeepBufferForMedia)
+ {
+ mUseDeepBufferForMediaOverride = useDeepBufferForMedia;
+ }
+ bool useDeepBufferForMedia() const;
+
private:
friend class sp<AudioPolicyConfig>;
@@ -158,6 +164,7 @@
sp<DeviceDescriptor> mDefaultOutputDevice;
bool mIsCallScreenModeSupported = false;
SurroundFormats mSurroundFormats;
+ std::optional<bool> mUseDeepBufferForMediaOverride;
};
} // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioPolicyConfig.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioPolicyConfig.cpp
index abeaaf8..f5e135e 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioPolicyConfig.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioPolicyConfig.cpp
@@ -16,6 +16,7 @@
#define LOG_TAG "APM_Config"
+#include <android-base/properties.h>
#include <AudioPolicyConfig.h>
#include <IOProfile.h>
#include <Serializer.h>
@@ -344,4 +345,9 @@
{AUDIO_FORMAT_AC4, {}}};
}
+bool AudioPolicyConfig::useDeepBufferForMedia() const {
+ if (mUseDeepBufferForMediaOverride.has_value()) return *mUseDeepBufferForMediaOverride;
+ return property_get_bool("audio.deep_buffer.media", false /* default_value */);
+}
+
} // namespace android
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index f36d8d5..2ff2907 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -1153,8 +1153,7 @@
SortedVector<audio_io_handle_t> outputs = getOutputsForDevices(devices, mOutputs);
audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE;
- if (stream == AUDIO_STREAM_MUSIC &&
- property_get_bool("audio.deep_buffer.media", false /* default_value */)) {
+ if (stream == AUDIO_STREAM_MUSIC && mConfig->useDeepBufferForMedia()) {
flags = AUDIO_OUTPUT_FLAG_DEEP_BUFFER;
}
const audio_io_handle_t output = selectOutput(outputs, flags);
@@ -1251,7 +1250,8 @@
// FIXME: in case of RENDER policy, the output capabilities should be checked
if ((secondaryMixes != nullptr && !secondaryMixes->empty())
- && !audio_is_linear_pcm(config->format)) {
+ && (!audio_is_linear_pcm(config->format) ||
+ *flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) {
ALOGD("%s: rejecting request as secondary mixes only support pcm", __func__);
return BAD_VALUE;
}
@@ -1527,6 +1527,13 @@
return NAME_NOT_FOUND;
}
+ // Reject flag combinations that do not make sense. Note that the requested flags might not
+ // have the 'DIRECT' flag set, however once a direct-capable profile is found, it will
+ // combine the requested flags with its own flags, yielding an unsupported combination.
+ if ((flags & AUDIO_OUTPUT_FLAG_DEEP_BUFFER) != 0) {
+ return NAME_NOT_FOUND;
+ }
+
// Do not allow offloading if one non offloadable effect is enabled or MasterMono is enabled.
// This prevents creating an offloaded track and tearing it down immediately after start
// when audioflinger detects there is an active non offloadable effect.
@@ -1675,8 +1682,7 @@
if (stream != AUDIO_STREAM_MUSIC) {
*flags = (audio_output_flags_t)(*flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
} else if (/* stream == AUDIO_STREAM_MUSIC && */
- *flags == AUDIO_OUTPUT_FLAG_NONE &&
- property_get_bool("audio.deep_buffer.media", false /* default_value */)) {
+ *flags == AUDIO_OUTPUT_FLAG_NONE && mConfig->useDeepBufferForMedia()) {
// use DEEP_BUFFER as default output for music stream type
*flags = (audio_output_flags_t)AUDIO_OUTPUT_FLAG_DEEP_BUFFER;
}
@@ -7415,7 +7421,8 @@
client->getSecondaryOutputs().begin(),
client->getSecondaryOutputs().end(),
secondaryDescs.begin(), secondaryDescs.end())) {
- if (!audio_is_linear_pcm(client->config().format)) {
+ if (client->flags() & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD
+ || !audio_is_linear_pcm(client->config().format)) {
// If the format is not PCM, the tracks should be invalidated to get correct
// behavior when the secondary output is changed.
clientsToInvalidate.push_back(client->portId());
diff --git a/services/audiopolicy/tests/AudioPolicyManagerTestClient.h b/services/audiopolicy/tests/AudioPolicyManagerTestClient.h
index 5a25a77..ca7ad40 100644
--- a/services/audiopolicy/tests/AudioPolicyManagerTestClient.h
+++ b/services/audiopolicy/tests/AudioPolicyManagerTestClient.h
@@ -17,6 +17,7 @@
#include <map>
#include <set>
+#include <media/TypeConverter.h>
#include <system/audio.h>
#include <utils/Log.h>
#include <utils/String8.h>
@@ -37,11 +38,11 @@
status_t openOutput(audio_module_handle_t module,
audio_io_handle_t *output,
- audio_config_t * /*halConfig*/,
- audio_config_base_t * /*mixerConfig*/,
+ audio_config_t *halConfig,
+ audio_config_base_t *mixerConfig,
const sp<DeviceDescriptorBase>& /*device*/,
uint32_t * /*latencyMs*/,
- audio_output_flags_t /*flags*/,
+ audio_output_flags_t flags,
audio_attributes_t /*attributes*/) override {
if (module >= mNextModuleHandle) {
ALOGE("%s: Module handle %d has not been allocated yet (next is %d)",
@@ -49,6 +50,13 @@
return BAD_VALUE;
}
*output = mNextIoHandle++;
+ mOpenedOutputs[*output] = flags;
+ ALOGD("%s: opened output %d: HAL(%s %s %d) Mixer(%s %s %d) %s", __func__, *output,
+ audio_channel_out_mask_to_string(halConfig->channel_mask),
+ audio_format_to_string(halConfig->format), halConfig->sample_rate,
+ audio_channel_out_mask_to_string(mixerConfig->channel_mask),
+ audio_format_to_string(mixerConfig->format), mixerConfig->sample_rate,
+ android::toString(flags).c_str());
return NO_ERROR;
}
@@ -58,6 +66,16 @@
return id;
}
+ status_t closeOutput(audio_io_handle_t output) override {
+ if (auto iter = mOpenedOutputs.find(output); iter != mOpenedOutputs.end()) {
+ mOpenedOutputs.erase(iter);
+ return NO_ERROR;
+ } else {
+ ALOGE("%s: Unknown output %d", __func__, output);
+ return BAD_VALUE;
+ }
+ }
+
status_t openInput(audio_module_handle_t module,
audio_io_handle_t *input,
audio_config_t * /*config*/,
@@ -262,6 +280,13 @@
return it == mTracksInternalMute.end() ? false : it->second;
}
+ std::optional<audio_output_flags_t> getOpenOutputFlags(audio_io_handle_t output) const {
+ if (auto iter = mOpenedOutputs.find(output); iter != mOpenedOutputs.end()) {
+ return iter->second;
+ }
+ return std::nullopt;
+ }
+
private:
audio_module_handle_t mNextModuleHandle = AUDIO_MODULE_HANDLE_NONE + 1;
audio_io_handle_t mNextIoHandle = AUDIO_IO_HANDLE_NONE + 1;
@@ -276,6 +301,7 @@
std::set<audio_channel_mask_t> mSupportedChannelMasks;
std::map<audio_port_handle_t, bool> mTracksInternalMute;
std::set<audio_io_handle_t> mOpenedInputs;
+ std::map<audio_io_handle_t, audio_output_flags_t> mOpenedOutputs;
};
} // namespace android
diff --git a/services/audiopolicy/tests/audiopolicymanager_tests.cpp b/services/audiopolicy/tests/audiopolicymanager_tests.cpp
index 9445af1..08855c9 100644
--- a/services/audiopolicy/tests/audiopolicymanager_tests.cpp
+++ b/services/audiopolicy/tests/audiopolicymanager_tests.cpp
@@ -35,6 +35,7 @@
#include <media/AudioPolicy.h>
#include <media/PatchBuilder.h>
#include <media/RecordingActivityTracker.h>
+#include <media/TypeConverter.h>
#include <utils/Log.h>
#include <utils/Vector.h>
#include <cutils/multiuser.h>
@@ -175,6 +176,11 @@
};
class AudioPolicyManagerTest : public testing::Test {
+ public:
+ constexpr static uint32_t k384000SamplingRate = 384000;
+ constexpr static uint32_t k48000SamplingRate = 48000;
+ constexpr static uint32_t k96000SamplingRate = 96000;
+
protected:
void SetUp() override;
void TearDown() override;
@@ -191,7 +197,7 @@
audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
audio_io_handle_t *output = nullptr,
audio_port_handle_t *portId = nullptr,
- audio_attributes_t attr = {},
+ audio_attributes_t attr = AUDIO_ATTRIBUTES_INITIALIZER,
audio_session_t session = AUDIO_SESSION_NONE,
int uid = 0,
bool* isBitPerfect = nullptr);
@@ -222,8 +228,6 @@
std::unique_ptr<AudioPolicyManagerTestClient> mClient;
std::unique_ptr<AudioPolicyTestManager> mManager;
- constexpr static const uint32_t k48000SamplingRate = 48000;
-
static const std::string sTestEngineConfig;
};
@@ -787,27 +791,27 @@
audio_config_base_t directConfig = AUDIO_CONFIG_BASE_INITIALIZER;
directConfig.format = AUDIO_FORMAT_DTS;
- directConfig.sample_rate = 48000;
+ directConfig.sample_rate = k48000SamplingRate;
directConfig.channel_mask = AUDIO_CHANNEL_OUT_5POINT1;
audio_config_base_t nonDirectConfig = AUDIO_CONFIG_BASE_INITIALIZER;
nonDirectConfig.format = AUDIO_FORMAT_PCM_16_BIT;
- nonDirectConfig.sample_rate = 48000;
+ nonDirectConfig.sample_rate = k48000SamplingRate;
nonDirectConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
audio_config_base_t nonExistentConfig = AUDIO_CONFIG_BASE_INITIALIZER;
nonExistentConfig.format = AUDIO_FORMAT_E_AC3;
- nonExistentConfig.sample_rate = 48000;
+ nonExistentConfig.sample_rate = k48000SamplingRate;
nonExistentConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
audio_config_base_t msdDirectConfig1 = AUDIO_CONFIG_BASE_INITIALIZER;
msdDirectConfig1.format = AUDIO_FORMAT_AC3;
- msdDirectConfig1.sample_rate = 48000;
+ msdDirectConfig1.sample_rate = k48000SamplingRate;
msdDirectConfig1.channel_mask = AUDIO_CHANNEL_OUT_5POINT1;
audio_config_base_t msdDirectConfig2 = AUDIO_CONFIG_BASE_INITIALIZER;
msdDirectConfig2.format = AUDIO_FORMAT_IEC60958;
- msdDirectConfig2.sample_rate = 48000;
+ msdDirectConfig2.sample_rate = k48000SamplingRate;
msdDirectConfig2.channel_mask = AUDIO_CHANNEL_INDEX_MASK_24;
audio_config_base_t msdNonDirectConfig = AUDIO_CONFIG_BASE_INITIALIZER;
@@ -854,27 +858,27 @@
audio_config_t directConfig = AUDIO_CONFIG_INITIALIZER;
directConfig.format = AUDIO_FORMAT_DTS;
- directConfig.sample_rate = 48000;
+ directConfig.sample_rate = k48000SamplingRate;
directConfig.channel_mask = AUDIO_CHANNEL_OUT_5POINT1;
audio_config_t nonDirectConfig = AUDIO_CONFIG_INITIALIZER;
nonDirectConfig.format = AUDIO_FORMAT_PCM_16_BIT;
- nonDirectConfig.sample_rate = 48000;
+ nonDirectConfig.sample_rate = k48000SamplingRate;
nonDirectConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
audio_config_t nonExistentConfig = AUDIO_CONFIG_INITIALIZER;
nonExistentConfig.format = AUDIO_FORMAT_E_AC3;
- nonExistentConfig.sample_rate = 48000;
+ nonExistentConfig.sample_rate = k48000SamplingRate;
nonExistentConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
audio_config_t msdDirectConfig1 = AUDIO_CONFIG_INITIALIZER;
msdDirectConfig1.format = AUDIO_FORMAT_AC3;
- msdDirectConfig1.sample_rate = 48000;
+ msdDirectConfig1.sample_rate = k48000SamplingRate;
msdDirectConfig1.channel_mask = AUDIO_CHANNEL_OUT_5POINT1;
audio_config_t msdDirectConfig2 = AUDIO_CONFIG_INITIALIZER;
msdDirectConfig2.format = AUDIO_FORMAT_IEC60958;
- msdDirectConfig2.sample_rate = 48000;
+ msdDirectConfig2.sample_rate = k48000SamplingRate;
msdDirectConfig2.channel_mask = AUDIO_CHANNEL_INDEX_MASK_24;
audio_config_t msdNonDirectConfig = AUDIO_CONFIG_INITIALIZER;
@@ -1130,12 +1134,12 @@
audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE;
getOutputForAttr(&selectedDeviceId, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO,
- 48000, AUDIO_OUTPUT_FLAG_NONE, &output, &portId, mediaAttr,
+ k48000SamplingRate, AUDIO_OUTPUT_FLAG_NONE, &output, &portId, mediaAttr,
AUDIO_SESSION_NONE, uid);
status_t status = mManager->startOutput(portId);
if (status == DEAD_OBJECT) {
getOutputForAttr(&selectedDeviceId, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO,
- 48000, AUDIO_OUTPUT_FLAG_NONE, &output, &portId, mediaAttr,
+ k48000SamplingRate, AUDIO_OUTPUT_FLAG_NONE, &output, &portId, mediaAttr,
AUDIO_SESSION_NONE, uid);
status = mManager->startOutput(portId);
}
@@ -1176,9 +1180,9 @@
audio_io_handle_t input = AUDIO_PORT_HANDLE_NONE;
AttributionSourceState attributionSource = createAttributionSourceState(/*uid=*/ 0);
audio_config_base_t requestedConfig = {
+ .sample_rate = k48000SamplingRate,
.channel_mask = AUDIO_CHANNEL_IN_STEREO,
.format = AUDIO_FORMAT_PCM_16_BIT,
- .sample_rate = 48000
};
audio_config_base_t config = requestedConfig;
audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE;
@@ -3184,6 +3188,259 @@
"low latency");
}
+class AudioPolicyManagerPhoneTest : public AudioPolicyManagerTestWithConfigurationFile {
+protected:
+ std::string getConfigFile() override { return sPhoneConfig; }
+ void testOutputMixPortSelectionForAttr(audio_output_flags_t flags, audio_format_t format,
+ int samplingRate, bool isMusic, const char* expectedMixPortName);
+ void testOutputMixPortSelectionForStream(
+ audio_stream_type_t stream, const char* expectedMixPortName);
+ void verifyMixPortNameAndFlags(audio_io_handle_t output, const char* expectedMixPortName);
+
+ static const std::string sPhoneConfig;
+ static const std::map<std::string, audio_output_flags_t> sMixPortFlags;
+};
+
+const std::string AudioPolicyManagerPhoneTest::sPhoneConfig =
+ AudioPolicyManagerPhoneTest::sExecutableDir + "test_phone_apm_configuration.xml";
+
+// Must be in sync with the contents of the sPhoneConfig file.
+const std::map<std::string, audio_output_flags_t> AudioPolicyManagerPhoneTest::sMixPortFlags = {
+ {"primary output",
+ (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_PRIMARY | AUDIO_OUTPUT_FLAG_FAST)},
+ {"direct", AUDIO_OUTPUT_FLAG_DIRECT},
+ {"deep buffer", AUDIO_OUTPUT_FLAG_DEEP_BUFFER},
+ {"compressed_offload",
+ (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD |
+ AUDIO_OUTPUT_FLAG_NON_BLOCKING |
+ AUDIO_OUTPUT_FLAG_GAPLESS_OFFLOAD)},
+ {"raw", (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_RAW | AUDIO_OUTPUT_FLAG_FAST)},
+ {"mmap_no_irq_out",
+ (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_MMAP_NOIRQ)},
+ {"voip_rx", AUDIO_OUTPUT_FLAG_VOIP_RX},
+};
+
+void AudioPolicyManagerPhoneTest::testOutputMixPortSelectionForAttr(
+ audio_output_flags_t flags, audio_format_t format, int samplingRate, bool isMusic,
+ const char* expectedMixPortName) {
+ audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
+ audio_io_handle_t output;
+ audio_port_handle_t portId;
+ audio_attributes_t attr = AUDIO_ATTRIBUTES_INITIALIZER;
+ if (isMusic) {
+ attr.content_type = AUDIO_CONTENT_TYPE_MUSIC;
+ attr.usage = AUDIO_USAGE_MEDIA;
+ }
+ getOutputForAttr(&selectedDeviceId, format, AUDIO_CHANNEL_OUT_STEREO, samplingRate, flags,
+ &output, &portId, attr);
+ EXPECT_NO_FATAL_FAILURE(verifyMixPortNameAndFlags(output, expectedMixPortName));
+ mManager->releaseOutput(portId);
+}
+
+void AudioPolicyManagerPhoneTest::testOutputMixPortSelectionForStream(
+ audio_stream_type_t stream, const char* expectedMixPortName) {
+ audio_io_handle_t output = mManager->getOutput(stream);
+ EXPECT_NO_FATAL_FAILURE(verifyMixPortNameAndFlags(output, expectedMixPortName));
+}
+
+void AudioPolicyManagerPhoneTest::verifyMixPortNameAndFlags(audio_io_handle_t output,
+ const char* expectedMixPortName) {
+ ALOGI("%s: checking output %d", __func__, output);
+ sp<SwAudioOutputDescriptor> outDesc = mManager->getOutputs().valueFor(output);
+ ASSERT_NE(nullptr, outDesc.get());
+ audio_port_v7 port = {};
+ outDesc->toAudioPort(&port);
+ EXPECT_EQ(AUDIO_PORT_TYPE_MIX, port.type);
+ EXPECT_EQ(AUDIO_PORT_ROLE_SOURCE, port.role);
+ ASSERT_STREQ(expectedMixPortName, port.name);
+
+ auto iter = sMixPortFlags.find(port.name);
+ ASSERT_NE(iter, sMixPortFlags.end()) << "\"" << port.name << "\" is not in sMixPortFlags";
+ auto actualFlags = mClient->getOpenOutputFlags(output);
+ ASSERT_TRUE(actualFlags.has_value()) << "\"" << port.name << "\" was not opened via client";
+ EXPECT_EQ(*actualFlags, iter->second);
+}
+
+TEST_F(AudioPolicyManagerPhoneTest, InitSuccess) {
+ // SetUp must finish with no assertions.
+}
+
+enum {
+ MIX_PORT_ATTR_EXPECTED_NAME_PARAMETER,
+ MIX_PORT_ATTR_EXPECTED_NAME_WITH_DBFM_PARAMETER,
+ MIX_PORT_ATTR_FLAGS_PARAMETER,
+ MIX_PORT_ATTR_FORMAT_PARAMETER,
+ MIX_PORT_ATTR_SAMPLING_RATE_PARAMETER,
+};
+using MixPortSelectionForAttr =
+ std::tuple<const char*, const char*, audio_output_flags_t, audio_format_t, int>;
+
+class AudioPolicyManagerOutputMixPortForAttrSelectionTest
+ : public AudioPolicyManagerPhoneTest,
+ public testing::WithParamInterface<MixPortSelectionForAttr> {
+};
+
+// There is no easy way to create a flat tuple from tuples via ::testing::Combine.
+// Instead, just run the same selection twice while altering the deep buffer for media setting.
+TEST_P(AudioPolicyManagerOutputMixPortForAttrSelectionTest, SelectPortByFlags) {
+ mConfig->setUseDeepBufferForMediaOverrideForTests(false);
+ ASSERT_NO_FATAL_FAILURE(testOutputMixPortSelectionForAttr(
+ std::get<MIX_PORT_ATTR_FLAGS_PARAMETER>(GetParam()),
+ std::get<MIX_PORT_ATTR_FORMAT_PARAMETER>(GetParam()),
+ std::get<MIX_PORT_ATTR_SAMPLING_RATE_PARAMETER>(GetParam()),
+ false /*isMusic*/,
+ std::get<MIX_PORT_ATTR_EXPECTED_NAME_PARAMETER>(GetParam())));
+}
+TEST_P(AudioPolicyManagerOutputMixPortForAttrSelectionTest, SelectPortByFlags_Music) {
+ mConfig->setUseDeepBufferForMediaOverrideForTests(false);
+ ASSERT_NO_FATAL_FAILURE(testOutputMixPortSelectionForAttr(
+ std::get<MIX_PORT_ATTR_FLAGS_PARAMETER>(GetParam()),
+ std::get<MIX_PORT_ATTR_FORMAT_PARAMETER>(GetParam()),
+ std::get<MIX_PORT_ATTR_SAMPLING_RATE_PARAMETER>(GetParam()),
+ true /*isMusic*/,
+ std::get<MIX_PORT_ATTR_EXPECTED_NAME_PARAMETER>(GetParam())));
+}
+TEST_P(AudioPolicyManagerOutputMixPortForAttrSelectionTest, SelectPortByFlags_DeepMedia) {
+ mConfig->setUseDeepBufferForMediaOverrideForTests(true);
+ const char* fallbackName = std::get<MIX_PORT_ATTR_EXPECTED_NAME_PARAMETER>(GetParam());
+ ASSERT_NO_FATAL_FAILURE(
+ testOutputMixPortSelectionForAttr(std::get<MIX_PORT_ATTR_FLAGS_PARAMETER>(GetParam()),
+ std::get<MIX_PORT_ATTR_FORMAT_PARAMETER>(GetParam()),
+ std::get<MIX_PORT_ATTR_SAMPLING_RATE_PARAMETER>(GetParam()),
+ false /*isMusic*/,
+ std::get<MIX_PORT_ATTR_EXPECTED_NAME_WITH_DBFM_PARAMETER>(
+ GetParam()) ?: fallbackName));
+}
+TEST_P(AudioPolicyManagerOutputMixPortForAttrSelectionTest, SelectPortByFlags_DeepMedia_Music) {
+ mConfig->setUseDeepBufferForMediaOverrideForTests(true);
+ const char* fallbackName = std::get<MIX_PORT_ATTR_EXPECTED_NAME_PARAMETER>(GetParam());
+ ASSERT_NO_FATAL_FAILURE(
+ testOutputMixPortSelectionForAttr(std::get<MIX_PORT_ATTR_FLAGS_PARAMETER>(GetParam()),
+ std::get<MIX_PORT_ATTR_FORMAT_PARAMETER>(GetParam()),
+ std::get<MIX_PORT_ATTR_SAMPLING_RATE_PARAMETER>(GetParam()),
+ true /*isMusic*/,
+ std::get<MIX_PORT_ATTR_EXPECTED_NAME_WITH_DBFM_PARAMETER>(
+ GetParam()) ?: fallbackName));
+}
+
+INSTANTIATE_TEST_CASE_P(AudioPolicyManagerOutputMixPortForAttrSelection,
+ AudioPolicyManagerOutputMixPortForAttrSelectionTest,
+ ::testing::Values(
+ std::make_tuple("primary output", "deep buffer", AUDIO_OUTPUT_FLAG_NONE,
+ AUDIO_FORMAT_PCM_16_BIT, AudioPolicyManagerTest::k48000SamplingRate),
+ std::make_tuple("primary output", "deep buffer", AUDIO_OUTPUT_FLAG_NONE,
+ AUDIO_FORMAT_PCM_FLOAT, AudioPolicyManagerTest::k48000SamplingRate),
+ // Note: this goes to "direct" because 384000 > SAMPLE_RATE_HZ_MAX (192000)
+ std::make_tuple("direct", "deep buffer", AUDIO_OUTPUT_FLAG_NONE,
+ AUDIO_FORMAT_PCM_FLOAT, AudioPolicyManagerTest::k384000SamplingRate),
+ std::make_tuple("primary output", nullptr, AUDIO_OUTPUT_FLAG_FAST,
+ AUDIO_FORMAT_PCM_16_BIT, AudioPolicyManagerTest::k48000SamplingRate),
+ std::make_tuple("direct", nullptr, AUDIO_OUTPUT_FLAG_DIRECT,
+ AUDIO_FORMAT_PCM_FLOAT, AudioPolicyManagerTest::k96000SamplingRate),
+ std::make_tuple("direct", nullptr, AUDIO_OUTPUT_FLAG_DIRECT,
+ AUDIO_FORMAT_PCM_FLOAT, AudioPolicyManagerTest::k384000SamplingRate),
+ std::make_tuple("deep buffer", nullptr, AUDIO_OUTPUT_FLAG_DEEP_BUFFER,
+ AUDIO_FORMAT_PCM_16_BIT, AudioPolicyManagerTest::k48000SamplingRate),
+ std::make_tuple("deep buffer", nullptr, AUDIO_OUTPUT_FLAG_DEEP_BUFFER,
+ AUDIO_FORMAT_PCM_FLOAT, AudioPolicyManagerTest::k384000SamplingRate),
+ std::make_tuple("compressed_offload", nullptr,
+ (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD |
+ AUDIO_OUTPUT_FLAG_NON_BLOCKING),
+ AUDIO_FORMAT_MP3, AudioPolicyManagerTest::k48000SamplingRate),
+ std::make_tuple("raw", nullptr,
+ AUDIO_OUTPUT_FLAG_RAW, AUDIO_FORMAT_PCM_32_BIT,
+ AudioPolicyManagerTest::k48000SamplingRate),
+ std::make_tuple("mmap_no_irq_out", nullptr,
+ (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_DIRECT |
+ AUDIO_OUTPUT_FLAG_MMAP_NOIRQ),
+ AUDIO_FORMAT_PCM_FLOAT, AudioPolicyManagerTest::k48000SamplingRate),
+ std::make_tuple("mmap_no_irq_out", nullptr,
+ (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_DIRECT |
+ AUDIO_OUTPUT_FLAG_MMAP_NOIRQ),
+ AUDIO_FORMAT_PCM_FLOAT, AudioPolicyManagerTest::k384000SamplingRate),
+ std::make_tuple("voip_rx", nullptr, AUDIO_OUTPUT_FLAG_VOIP_RX,
+ AUDIO_FORMAT_PCM_16_BIT, AudioPolicyManagerTest::k48000SamplingRate)),
+ [](const ::testing::TestParamInfo<MixPortSelectionForAttr>& info) {
+ static const std::string flagPrefix = "AUDIO_OUTPUT_FLAG_";
+ static const std::string formatPrefix = "AUDIO_FORMAT_";
+ std::string flags;
+ TypeConverter<OutputFlagTraits>::maskToString(
+ std::get<MIX_PORT_ATTR_FLAGS_PARAMETER>(info.param), flags, "__");
+ size_t index = 0;
+ while (true) {
+ index = flags.rfind(flagPrefix);
+ if (index == std::string::npos) break;
+ flags.erase(index, flagPrefix.length());
+ }
+ std::string format;
+ TypeConverter<FormatTraits>::toString(
+ std::get<MIX_PORT_ATTR_FORMAT_PARAMETER>(info.param), format);
+ if (size_t index = format.find(formatPrefix); index != std::string::npos) {
+ format.erase(index, formatPrefix.length());
+ }
+ return flags + "__" + format + "__" +
+ std::to_string(std::get<MIX_PORT_ATTR_SAMPLING_RATE_PARAMETER>(info.param));
+ }
+);
+
+
+enum {
+ MIX_PORT_STRM_EXPECTED_NAME_PARAMETER,
+ MIX_PORT_STRM_EXPECTED_NAME_WITH_DBFM_PARAMETER,
+ MIX_PORT_STRM_STREAM_PARAMETER,
+};
+using MixPortSelectionForStream =
+ std::tuple<const char*, const char*, audio_stream_type_t>;
+
+class AudioPolicyManagerOutputMixPortForStreamSelectionTest
+ : public AudioPolicyManagerPhoneTest,
+ public testing::WithParamInterface<MixPortSelectionForStream> {
+};
+
+// There is no easy way to create a flat tuple from tuples via ::testing::Combine.
+// Instead, just run the same selection twice while altering the deep buffer for media setting.
+TEST_P(AudioPolicyManagerOutputMixPortForStreamSelectionTest, SelectPort_NoDBFM) {
+ mConfig->setUseDeepBufferForMediaOverrideForTests(false);
+ ASSERT_NO_FATAL_FAILURE(testOutputMixPortSelectionForStream(
+ std::get<MIX_PORT_STRM_STREAM_PARAMETER>(GetParam()),
+ std::get<MIX_PORT_STRM_EXPECTED_NAME_PARAMETER>(GetParam())));
+}
+TEST_P(AudioPolicyManagerOutputMixPortForStreamSelectionTest, SelectPort_WithDBFM) {
+ mConfig->setUseDeepBufferForMediaOverrideForTests(true);
+ const char* fallbackName = std::get<MIX_PORT_STRM_EXPECTED_NAME_PARAMETER>(GetParam());
+ ASSERT_NO_FATAL_FAILURE(testOutputMixPortSelectionForStream(
+ std::get<MIX_PORT_STRM_STREAM_PARAMETER>(GetParam()),
+ std::get<MIX_PORT_STRM_EXPECTED_NAME_WITH_DBFM_PARAMETER>(
+ GetParam()) ?: fallbackName));
+}
+
+INSTANTIATE_TEST_CASE_P(
+ AudioPolicyManagerOutputMixPortForStreamSelection,
+ AudioPolicyManagerOutputMixPortForStreamSelectionTest,
+ ::testing::Values(std::make_tuple("primary output", nullptr, AUDIO_STREAM_DEFAULT),
+ std::make_tuple("primary output", nullptr, AUDIO_STREAM_SYSTEM),
+ std::make_tuple("primary output", nullptr, AUDIO_STREAM_RING),
+ std::make_tuple("primary output", "deep buffer", AUDIO_STREAM_MUSIC),
+ std::make_tuple("primary output", nullptr, AUDIO_STREAM_ALARM),
+ std::make_tuple("primary output", nullptr, AUDIO_STREAM_NOTIFICATION),
+ std::make_tuple("primary output", nullptr, AUDIO_STREAM_BLUETOOTH_SCO),
+ std::make_tuple("primary output", nullptr, AUDIO_STREAM_ENFORCED_AUDIBLE),
+ std::make_tuple("primary output", nullptr, AUDIO_STREAM_DTMF),
+ std::make_tuple("primary output", nullptr, AUDIO_STREAM_TTS),
+ std::make_tuple("primary output", nullptr, AUDIO_STREAM_ACCESSIBILITY),
+ std::make_tuple("primary output", nullptr, AUDIO_STREAM_ASSISTANT)),
+ [](const ::testing::TestParamInfo<MixPortSelectionForStream>& info) {
+ static const std::string streamPrefix = "AUDIO_STREAM_";
+ std::string stream;
+ TypeConverter<StreamTraits>::toString(
+ std::get<MIX_PORT_STRM_STREAM_PARAMETER>(info.param), stream);
+ if (size_t index = stream.find(streamPrefix); index != std::string::npos) {
+ stream.erase(index, streamPrefix.length());
+ }
+ return stream;
+ }
+);
+
class AudioPolicyManagerDynamicHwModulesTest : public AudioPolicyManagerTestWithConfigurationFile {
protected:
void SetUpManagerConfig() override;
@@ -3336,7 +3593,7 @@
audio_io_handle_t input = AUDIO_PORT_HANDLE_NONE;
ASSERT_NO_FATAL_FAILURE(getInputForAttr(attr, &input, AUDIO_SESSION_NONE, 1, &selectedDeviceId,
AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_IN_STEREO,
- 48000));
+ k48000SamplingRate));
auto selectedDevice = availableDevices.getDeviceFromId(selectedDeviceId);
ASSERT_NE(nullptr, selectedDevice);
@@ -3357,7 +3614,7 @@
input = AUDIO_PORT_HANDLE_NONE;
ASSERT_NO_FATAL_FAILURE(getInputForAttr(attr, &input, AUDIO_SESSION_NONE, 1, &selectedDeviceId,
AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_IN_STEREO,
- 48000));
+ k48000SamplingRate));
ASSERT_EQ(preferredDevice, availableDevices.getDeviceFromId(selectedDeviceId));
// After clearing preferred device for capture preset, the selected device for input should be
@@ -3368,7 +3625,7 @@
input = AUDIO_PORT_HANDLE_NONE;
ASSERT_NO_FATAL_FAILURE(getInputForAttr(attr, &input, AUDIO_SESSION_NONE, 1, &selectedDeviceId,
AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_IN_STEREO,
- 48000));
+ k48000SamplingRate));
ASSERT_EQ(selectedDevice, availableDevices.getDeviceFromId(selectedDeviceId));
ASSERT_EQ(NO_ERROR, mManager->setDeviceConnectionState(
@@ -3394,7 +3651,7 @@
audio_io_handle_t input = AUDIO_PORT_HANDLE_NONE;
ASSERT_NO_FATAL_FAILURE(getInputForAttr(attr, &input, AUDIO_SESSION_NONE, 1, &selectedDeviceId,
AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_IN_STEREO,
- 48000));
+ k48000SamplingRate));
auto selectedDevice = availableDevices.getDeviceFromId(selectedDeviceId);
ASSERT_NE(nullptr, selectedDevice);
@@ -3407,7 +3664,7 @@
input = AUDIO_PORT_HANDLE_NONE;
ASSERT_NO_FATAL_FAILURE(getInputForAttr(attr, &input, AUDIO_SESSION_NONE, 1,
&selectedDeviceId, AUDIO_FORMAT_PCM_16_BIT,
- AUDIO_CHANNEL_IN_STEREO, 48000));
+ AUDIO_CHANNEL_IN_STEREO, k48000SamplingRate));
ASSERT_NE(selectedDevice, availableDevices.getDeviceFromId(selectedDeviceId));
// After clearing disabled device for capture preset, the selected device for input should be
@@ -3418,7 +3675,7 @@
input = AUDIO_PORT_HANDLE_NONE;
ASSERT_NO_FATAL_FAILURE(getInputForAttr(attr, &input, AUDIO_SESSION_NONE, 1, &selectedDeviceId,
AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_IN_STEREO,
- 48000));
+ k48000SamplingRate));
ASSERT_EQ(selectedDevice, availableDevices.getDeviceFromId(selectedDeviceId));
ASSERT_EQ(NO_ERROR, mManager->setDeviceConnectionState(
@@ -3492,7 +3749,7 @@
audio_port_handle_t routedPortId = devicePort.id;
ASSERT_NO_FATAL_FAILURE(getInputForAttr(attr, &inputClientHandle, session, 1, &routedPortId,
AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_IN_STEREO,
- 48000, AUDIO_INPUT_FLAG_NONE, &portId));
+ k48000SamplingRate, AUDIO_INPUT_FLAG_NONE, &portId));
ASSERT_EQ(devicePort.id, routedPortId);
auto selectedDevice = availableDevices.getDeviceFromId(routedPortId);
ASSERT_NE(nullptr, selectedDevice);
@@ -3521,7 +3778,7 @@
// effect attached again
ASSERT_NO_FATAL_FAILURE(getInputForAttr(attr, &inputClientHandle, session, 1, &routedPortId,
AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_IN_STEREO,
- 48000));
+ k48000SamplingRate));
// unregister effect should succeed since effect shall have been restore on the client session
ASSERT_EQ(NO_ERROR, mManager->unregisterEffect(effectId));
diff --git a/services/audiopolicy/tests/resources/Android.bp b/services/audiopolicy/tests/resources/Android.bp
index 15e51b0..8e7a697 100644
--- a/services/audiopolicy/tests/resources/Android.bp
+++ b/services/audiopolicy/tests/resources/Android.bp
@@ -20,6 +20,7 @@
"test_audio_policy_primary_only_configuration.xml",
"test_car_ap_atmos_offload_configuration.xml",
"test_invalid_audio_policy_configuration.xml",
+ "test_phone_apm_configuration.xml",
"test_settop_box_surround_configuration.xml",
"test_tv_apm_configuration.xml",
],
diff --git a/services/audiopolicy/tests/resources/test_phone_apm_configuration.xml b/services/audiopolicy/tests/resources/test_phone_apm_configuration.xml
new file mode 100644
index 0000000..efe1400
--- /dev/null
+++ b/services/audiopolicy/tests/resources/test_phone_apm_configuration.xml
@@ -0,0 +1,279 @@
+<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
+<!-- Copyright (C) 2024 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.
+-->
+<audioPolicyConfiguration version="7.0" xmlns:xi="http://www.w3.org/2001/XInclude">
+ <globalConfiguration speaker_drc_enabled="false" call_screen_mode_supported="true" />
+ <modules>
+ <!-- Primary Audio HAL -->
+ <module name="primary" halVersion="2.0">
+ <attachedDevices>
+ <item>Speaker</item>
+ <item>Speaker Safe</item>
+ <item>Earpiece</item>
+ <item>Built-In Mic</item>
+ <item>Built-In Back Mic</item>
+ <item>Telephony Tx</item>
+ <item>Voice Call And Telephony Rx</item>
+ <item>Echo Ref In</item>
+ </attachedDevices>
+ <defaultOutputDevice>Speaker</defaultOutputDevice>
+ <mixPorts>
+ <mixPort name="primary output" role="source" flags="AUDIO_OUTPUT_FLAG_PRIMARY AUDIO_OUTPUT_FLAG_FAST"
+ recommendedMuteDurationMs="40">
+ <profile name="" format="AUDIO_FORMAT_PCM_FLOAT"
+ samplingRates="48000 96000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ <mixPort name="direct" role="source" flags="AUDIO_OUTPUT_FLAG_DIRECT"
+ recommendedMuteDurationMs="40">
+ <profile name="" format="AUDIO_FORMAT_PCM_FLOAT"
+ samplingRates="48000 96000 384000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ <mixPort name="deep buffer" role="source" flags="AUDIO_OUTPUT_FLAG_DEEP_BUFFER">
+ <profile name="" format="AUDIO_FORMAT_PCM_FLOAT"
+ samplingRates="48000 96000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ <mixPort name="compressed_offload" role="source"
+ flags="AUDIO_OUTPUT_FLAG_DIRECT AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD AUDIO_OUTPUT_FLAG_NON_BLOCKING AUDIO_OUTPUT_FLAG_GAPLESS_OFFLOAD">
+ <profile name="" format="AUDIO_FORMAT_MP3"
+ samplingRates="8000 16000 24000 32000 44100 48000 96000"
+ channelMasks="AUDIO_CHANNEL_OUT_STEREO AUDIO_CHANNEL_OUT_MONO"/>
+ <profile name="" format="AUDIO_FORMAT_AAC_LC"
+ samplingRates="8000 16000 24000 32000 44100 48000 96000"
+ channelMasks="AUDIO_CHANNEL_OUT_STEREO AUDIO_CHANNEL_OUT_MONO"/>
+ <profile name="" format="AUDIO_FORMAT_AAC_HE_V1"
+ samplingRates="8000 16000 24000 32000 44100 48000 96000"
+ channelMasks="AUDIO_CHANNEL_OUT_STEREO AUDIO_CHANNEL_OUT_MONO"/>
+ <profile name="" format="AUDIO_FORMAT_AAC_HE_V2"
+ samplingRates="8000 16000 24000 32000 44100 48000 96000"
+ channelMasks="AUDIO_CHANNEL_OUT_STEREO AUDIO_CHANNEL_OUT_MONO"/>
+ <profile name="" format="AUDIO_FORMAT_OPUS"
+ samplingRates="48000 96000"
+ channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ <mixPort name="haptic" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000 96000"
+ channelMasks="AUDIO_CHANNEL_OUT_STEREO_HAPTIC_AB" />
+ </mixPort>
+ <mixPort name="raw" role="source" flags="AUDIO_OUTPUT_FLAG_RAW AUDIO_OUTPUT_FLAG_FAST">
+ <profile name="" format="AUDIO_FORMAT_PCM_32_BIT"
+ samplingRates="48000 96000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ <mixPort name="mmap_no_irq_out" role="source" flags="AUDIO_OUTPUT_FLAG_DIRECT AUDIO_OUTPUT_FLAG_MMAP_NOIRQ">
+ <profile name="" format="AUDIO_FORMAT_PCM_FLOAT"
+ samplingRates="48000 96000 384000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ <mixPort name="immersive_out" role="source" flags="AUDIO_OUTPUT_FLAG_SPATIALIZER">
+ <profile name="" format="AUDIO_FORMAT_PCM_32_BIT"
+ samplingRates="48000 96000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ <mixPort name="incall playback" role="source"
+ flags="AUDIO_OUTPUT_FLAG_INCALL_MUSIC">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000 96000"
+ channelMasks="AUDIO_CHANNEL_OUT_STEREO" />
+ </mixPort>
+ <mixPort name="voice call tx" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000 96000"
+ channelMasks="AUDIO_CHANNEL_OUT_STEREO" />
+ </mixPort>
+ <mixPort name="voip_rx" role="source"
+ flags="AUDIO_OUTPUT_FLAG_VOIP_RX">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000 96000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ <mixPort name="primary input" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000 96000"
+ channelMasks="AUDIO_CHANNEL_IN_MONO AUDIO_CHANNEL_IN_STEREO AUDIO_CHANNEL_INDEX_MASK_3"/>
+ </mixPort>
+ <mixPort name="hotword input" role="sink" flags="AUDIO_INPUT_FLAG_HW_HOTWORD" maxActiveCount="0" >
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="8000 11025 12000 16000 22050 24000 32000 44100 48000 96000"
+ channelMasks="AUDIO_CHANNEL_IN_MONO AUDIO_CHANNEL_IN_STEREO"/>
+ </mixPort>
+ <mixPort name="incall capture" role="sink" maxActiveCount="2" maxOpenCount="2">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000 96000"
+ channelMasks="AUDIO_CHANNEL_IN_MONO"/>
+ </mixPort>
+ <mixPort name="voice call rx" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000 96000"
+ channelMasks="AUDIO_CHANNEL_IN_MONO"/>
+ </mixPort>
+ <mixPort name="voip_tx" role="sink"
+ flags="AUDIO_INPUT_FLAG_VOIP_TX">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000 96000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
+ </mixPort>
+ <mixPort name="fast input" role="sink" flags="AUDIO_INPUT_FLAG_RAW AUDIO_INPUT_FLAG_FAST">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000 96000"
+ channelMasks="AUDIO_CHANNEL_IN_MONO AUDIO_CHANNEL_IN_STEREO"/>
+ </mixPort>
+ <mixPort name="mmap_no_irq_in" role="sink" flags="AUDIO_INPUT_FLAG_MMAP_NOIRQ">
+ <profile name="" format="AUDIO_FORMAT_PCM_32_BIT"
+ samplingRates="48000 96000"
+ channelMasks="AUDIO_CHANNEL_IN_MONO AUDIO_CHANNEL_IN_STEREO"/>
+ </mixPort>
+ <mixPort name="hifi_playback" role="source" />
+ <mixPort name="hifi_input" role="sink" />
+ <mixPort name="echo_ref_input" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_32_BIT"
+ samplingRates="48000 96000"
+ channelMasks="AUDIO_CHANNEL_IN_STEREO"/>
+ </mixPort>
+ </mixPorts>
+ <devicePorts>
+ <!-- Output devices declaration, i.e. Sink DEVICE PORT -->
+ <devicePort tagName="Earpiece" type="AUDIO_DEVICE_OUT_EARPIECE" role="sink">
+ </devicePort>
+ <devicePort tagName="Speaker" type="AUDIO_DEVICE_OUT_SPEAKER" role="sink">
+ </devicePort>
+ <devicePort tagName="Speaker Safe" type="AUDIO_DEVICE_OUT_SPEAKER_SAFE" role="sink">
+ </devicePort>
+ <devicePort tagName="BT SCO" type="AUDIO_DEVICE_OUT_BLUETOOTH_SCO" role="sink">
+ </devicePort>
+ <devicePort tagName="BT SCO Headset" type="AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET" role="sink">
+ </devicePort>
+ <devicePort tagName="BT SCO Car Kit" type="AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT" role="sink">
+ </devicePort>
+ <devicePort tagName="USB Device Out" type="AUDIO_DEVICE_OUT_USB_DEVICE" role="sink">
+ </devicePort>
+ <devicePort tagName="USB Headset Out" type="AUDIO_DEVICE_OUT_USB_HEADSET" role="sink">
+ </devicePort>
+ <devicePort tagName="HDMI Out" type="AUDIO_DEVICE_OUT_HDMI" role="sink">
+ </devicePort>
+ <devicePort tagName="Telephony Tx" type="AUDIO_DEVICE_OUT_TELEPHONY_TX" role="sink">
+ </devicePort>
+ <!-- Input devices declaration, i.e. Source DEVICE PORT -->
+ <devicePort tagName="Built-In Mic" type="AUDIO_DEVICE_IN_BUILTIN_MIC" role="source">
+ </devicePort>
+ <devicePort tagName="Built-In Back Mic" type="AUDIO_DEVICE_IN_BACK_MIC" role="source">
+ </devicePort>
+ <devicePort tagName="BT SCO Headset Mic" type="AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET" role="source">
+ </devicePort>
+ <devicePort tagName="BT A2DP Out" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP" role="sink"
+ encodedFormats="AUDIO_FORMAT_OPUS AUDIO_FORMAT_AAC AUDIO_FORMAT_SBC">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="44100 48000 96000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </devicePort>
+ <devicePort tagName="BT A2DP Headphones" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES" role="sink"
+ encodedFormats="AUDIO_FORMAT_OPUS AUDIO_FORMAT_AAC AUDIO_FORMAT_SBC">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="44100 48000 96000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </devicePort>
+ <devicePort tagName="BT A2DP Speaker" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER" role="sink"
+ encodedFormats="AUDIO_FORMAT_OPUS AUDIO_FORMAT_AAC AUDIO_FORMAT_SBC">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="44100 48000 96000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </devicePort>
+ <devicePort tagName="BT BLE Headset" type="AUDIO_DEVICE_OUT_BLE_HEADSET" role="sink"
+ encodedFormats="AUDIO_FORMAT_LC3">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000 96000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </devicePort>
+ <devicePort tagName="BT BLE Speaker" type="AUDIO_DEVICE_OUT_BLE_SPEAKER" role="sink"
+ encodedFormats="AUDIO_FORMAT_LC3">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000 96000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </devicePort>
+ <devicePort tagName="BT BLE Broadcast" type="AUDIO_DEVICE_OUT_BLE_BROADCAST" role="sink"
+ encodedFormats="AUDIO_FORMAT_LC3">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000 96000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </devicePort>
+ <devicePort tagName="BLE Headset Mic" type="AUDIO_DEVICE_IN_BLE_HEADSET" role="source">
+ </devicePort>
+ <devicePort tagName="USB Device In" type="AUDIO_DEVICE_IN_USB_DEVICE" role="source">
+ </devicePort>
+ <devicePort tagName="USB Headset In" type="AUDIO_DEVICE_IN_USB_HEADSET" role="source">
+ </devicePort>
+ <!-- AUDIO_DEVICE_IN_VOICE_CALL and AUDIO_DEVICE_IN_TELEPHONY_RX are in the same value -->
+ <devicePort tagName="Voice Call And Telephony Rx" type="AUDIO_DEVICE_IN_VOICE_CALL" role="source">
+ </devicePort>
+ <devicePort tagName="Echo Ref In" type="AUDIO_DEVICE_IN_ECHO_REFERENCE" role="source">
+ </devicePort>
+ </devicePorts>
+ <!-- route declaration, i.e. list all available sources for a given sink -->
+ <routes>
+ <route type="mix" sink="Speaker"
+ sources="primary output,direct,deep buffer,haptic,raw,mmap_no_irq_out,voip_rx,compressed_offload"/>
+ <route type="mix" sink="Speaker Safe"
+ sources="primary output,direct,deep buffer,haptic,raw,mmap_no_irq_out,voip_rx,compressed_offload"/>
+ <route type="mix" sink="Earpiece"
+ sources="primary output,direct,deep buffer,haptic,raw,mmap_no_irq_out,voip_rx,compressed_offload"/>
+ <route type="mix" sink="BT A2DP Out"
+ sources="primary output,direct,deep buffer,haptic,voip_rx,compressed_offload,raw,mmap_no_irq_out,immersive_out"/>
+ <route type="mix" sink="BT A2DP Headphones"
+ sources="primary output,direct,deep buffer,haptic,voip_rx,compressed_offload,raw,mmap_no_irq_out,immersive_out"/>
+ <route type="mix" sink="BT A2DP Speaker"
+ sources="primary output,direct,deep buffer,haptic,voip_rx,compressed_offload,raw,mmap_no_irq_out,immersive_out"/>
+ <route type="mix" sink="BT BLE Headset"
+ sources="primary output,direct,deep buffer,haptic,voip_rx,compressed_offload,raw,mmap_no_irq_out,immersive_out"/>
+ <route type="mix" sink="BT BLE Speaker"
+ sources="primary output,direct,deep buffer,haptic,voip_rx,compressed_offload,raw,mmap_no_irq_out,immersive_out"/>
+ <route type="mix" sink="BT BLE Broadcast"
+ sources="primary output,direct,deep buffer,haptic,voip_rx,compressed_offload,raw,mmap_no_irq_out,immersive_out"/>
+ <route type="mix" sink="USB Device Out"
+ sources="primary output,direct,deep buffer,haptic,raw,mmap_no_irq_out,voip_rx,hifi_playback,compressed_offload,immersive_out"/>
+ <route type="mix" sink="USB Headset Out"
+ sources="primary output,direct,deep buffer,haptic,raw,mmap_no_irq_out,voip_rx,hifi_playback,compressed_offload,immersive_out"/>
+ <route type="mix" sink="HDMI Out"
+ sources="primary output,direct,deep buffer,haptic,raw,mmap_no_irq_out,voip_rx,compressed_offload"/>
+ <route type="mix" sink="BT SCO"
+ sources="primary output,direct,deep buffer,haptic,voip_rx,compressed_offload,raw,mmap_no_irq_out"/>
+ <route type="mix" sink="BT SCO Headset"
+ sources="primary output,direct,deep buffer,haptic,voip_rx,compressed_offload,raw,mmap_no_irq_out"/>
+ <route type="mix" sink="BT SCO Car Kit"
+ sources="primary output,direct,deep buffer,haptic,voip_rx,compressed_offload,raw,mmap_no_irq_out"/>
+ <route type="mix" sink="Telephony Tx" sources="incall playback,voice call tx" />
+ <route type="mix" sink="primary input"
+ sources="Built-In Mic,Built-In Back Mic,USB Device In,USB Headset In,BT SCO Headset Mic,BLE Headset Mic"/>
+ <route type="mix" sink="hotword input"
+ sources="Built-In Mic,Built-In Back Mic,USB Device In,USB Headset In,BT SCO Headset Mic,BLE Headset Mic"/>
+ <route type="mix" sink="incall capture" sources="Voice Call And Telephony Rx" />
+ <route type="mix" sink="voice call rx" sources="Voice Call And Telephony Rx" />
+ <route type="mix" sink="voip_tx"
+ sources="Built-In Mic,Built-In Back Mic,USB Device In,USB Headset In,BT SCO Headset Mic,BLE Headset Mic"/>
+ <route type="mix" sink="fast input"
+ sources="Built-In Mic,Built-In Back Mic,USB Device In,USB Headset In,BT SCO Headset Mic,BLE Headset Mic"/>
+ <route type="mix" sink="mmap_no_irq_in"
+ sources="Built-In Mic,Built-In Back Mic,USB Device In,USB Headset In,BT SCO Headset Mic,BLE Headset Mic"/>
+ <route type="mix" sink="hifi_input" sources="USB Device In,USB Headset In" />
+ <route type="mix" sink="echo_ref_input" sources="Echo Ref In"/>
+ </routes>
+ </module>
+ <!-- Usb Audio HAL -->
+ <module name="usbv2" halVersion="2.0">
+ <mixPorts>
+ <mixPort name="usb_accessory output" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="44100" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ </mixPorts>
+ <devicePorts>
+ <devicePort tagName="USB Host Out" type="AUDIO_DEVICE_OUT_USB_ACCESSORY" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="44100" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </devicePort>
+ </devicePorts>
+ <routes>
+ <route type="mix" sink="USB Host Out"
+ sources="usb_accessory output"/>
+ </routes>
+ </module>
+ </modules>
+ <!-- End of Modules section -->
+</audioPolicyConfiguration>