Merge "audio flinger: fix sink metadata for telephony software patches" into tm-qpr-dev (9cace9f3) · Commits · e / os / android_frameworks_av

services/audioflinger/PatchPanel.cpp

+18 −3

Original line number	Diff line number	Diff line
		@@ -313,12 +313,19 @@ status_t AudioFlinger::PatchPanel::createAudioPatch(const struct audio_patch *pa
		patch->sources[0].config_mask & AUDIO_PORT_CONFIG_FLAGS ?
		patch->sources[0].flags.input : AUDIO_INPUT_FLAG_NONE;
		audio_io_handle_t input = AUDIO_IO_HANDLE_NONE;
		audio_source_t source = AUDIO_SOURCE_MIC;
		// For telephony patches, propagate voice communication use case to record side
		if (patch->num_sources == 2
		&& patch->sources[1].ext.mix.usecase.stream
		== AUDIO_STREAM_VOICE_CALL) {
		source = AUDIO_SOURCE_VOICE_COMMUNICATION;
		}
		sp<ThreadBase> thread = mAudioFlinger.openInput_l(srcModule,
		&input,
		&config,
		device,
		address,
		AUDIO_SOURCE_MIC,
		source,
		flags,
		outputDevice,
		outputDeviceAddress);
		@@ -516,9 +523,14 @@ status_t AudioFlinger::PatchPanel::Patch::createConnections(PatchPanel *panel)
		audio_output_flags_t outputFlags = mAudioPatch.sinks[0].config_mask & AUDIO_PORT_CONFIG_FLAGS ?
		mAudioPatch.sinks[0].flags.output : AUDIO_OUTPUT_FLAG_NONE;
		audio_stream_type_t streamType = AUDIO_STREAM_PATCH;
		audio_source_t source = AUDIO_SOURCE_DEFAULT;
		if (mAudioPatch.num_sources == 2 && mAudioPatch.sources[1].type == AUDIO_PORT_TYPE_MIX) {
		// "reuse one existing output mix" case
		streamType = mAudioPatch.sources[1].ext.mix.usecase.stream;
		// For telephony patches, propagate voice communication use case to record side
		if (streamType == AUDIO_STREAM_VOICE_CALL) {
		source = AUDIO_SOURCE_VOICE_COMMUNICATION;
		}
		}
		if (mPlayback.thread()->hasFastMixer()) {
		// Create a fast track if the playback thread has fast mixer to get better performance.
		@@ -546,7 +558,8 @@ status_t AudioFlinger::PatchPanel::Patch::createConnections(PatchPanel *panel)
		inChannelMask,
		format,
		frameCount,
		inputFlags);
		inputFlags,
		source);
		} else {
		// use a pseudo LCM between input and output framecount
		int playbackShift = __builtin_ctz(playbackFrameCount);
		@@ -566,7 +579,9 @@ status_t AudioFlinger::PatchPanel::Patch::createConnections(PatchPanel *panel)
		frameCount,
		nullptr,
		(size_t)0 /* bufferSize */,
		inputFlags);
		inputFlags,
		{} /* timeout */,
		source);
		}
		status = mRecord.checkTrack(tempRecordTrack.get());
		if (status != NO_ERROR) {

services/audioflinger/RecordTracks.h

+8 −2

Original line number	Diff line number	Diff line
		@@ -87,6 +87,10 @@ public:
		&& (flags & AUDIO_INPUT_FLAG_HW_AV_SYNC) == 0;
		}

		using SinkMetadatas = std::vector<record_track_metadata_v7_t>;
		using MetadataInserter = std::back_insert_iterator<SinkMetadatas>;
		virtual void copyMetadataTo(MetadataInserter& backInserter) const;

		private:
		friend class AudioFlinger; // for mState

		@@ -134,7 +138,8 @@ public:
		void *buffer,
		size_t bufferSize,
		audio_input_flags_t flags,
		const Timeout& timeout = {});
		const Timeout& timeout = {},
		audio_source_t source = AUDIO_SOURCE_DEFAULT);
		virtual ~PatchRecord();

		virtual Source* getSource() { return nullptr; }
		@@ -166,7 +171,8 @@ public:
		audio_channel_mask_t channelMask,
		audio_format_t format,
		size_t frameCount,
		audio_input_flags_t flags);
		audio_input_flags_t flags,
		audio_source_t source = AUDIO_SOURCE_DEFAULT);

		Source* getSource() override { return static_cast<Source*>(this); }

services/audioflinger/Threads.cpp

+2 −14

Original line number	Diff line number	Diff line
		@@ -8775,21 +8775,9 @@ void AudioFlinger::RecordThread::updateMetadata_l()
		return; // nothing to do
		}
		StreamInHalInterface::SinkMetadata metadata;
		auto backInserter = std::back_inserter(metadata.tracks);
		for (const sp<RecordTrack> &track : mActiveTracks) {
		// Do not forward PatchRecord metadata to audio HAL
		if (track->isPatchTrack()) {
		continue;
		}
		// No track is invalid as this is called after prepareTrack_l in the same critical section
		record_track_metadata_v7_t trackMetadata;
		trackMetadata.base = {
		.source = track->attributes().source,
		.gain = 1, // capture tracks do not have volumes
		};
		trackMetadata.channel_mask = track->channelMask(),
		strncpy(trackMetadata.tags, track->attributes().tags, AUDIO_ATTRIBUTES_TAGS_MAX_SIZE);

		metadata.tracks.push_back(trackMetadata);
		track->copyMetadataTo(backInserter);
		}
		mInput->stream->updateSinkMetadata(metadata);
		}

services/audioflinger/Tracks.cpp

+26 −7

Original line number	Diff line number	Diff line
		@@ -1480,7 +1480,7 @@ void AudioFlinger::PlaybackThread::Track::copyMetadataTo(MetadataInserter& backI
		}
		}

		metadata.channel_mask = mChannelMask,
		metadata.channel_mask = mChannelMask;
		strncpy(metadata.tags, mAttr.tags, AUDIO_ATTRIBUTES_TAGS_MAX_SIZE);
		*backInserter++ = metadata;
		}
		@@ -2024,7 +2024,6 @@ ssize_t AudioFlinger::PlaybackThread::OutputTrack::write(void* data, uint32_t fr
		{
		Buffer *pInBuffer;
		Buffer inBuffer;
		bool outputBufferFull = false;
		inBuffer.frameCount = frames;
		inBuffer.raw = data;

		@@ -2054,7 +2053,6 @@ ssize_t AudioFlinger::PlaybackThread::OutputTrack::write(void* data, uint32_t fr
		ALOGV("%s(%d): thread %d no more output buffers; status %d",
		__func__, mId,
		(int)mThreadIoHandle, status);
		outputBufferFull = true;
		break;
		}
		uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime);
		@@ -2750,6 +2748,25 @@ status_t AudioFlinger::RecordThread::RecordTrack::shareAudioHistory(
		}
		}

		void AudioFlinger::RecordThread::RecordTrack::copyMetadataTo(MetadataInserter& backInserter) const
		{

		// Do not forward PatchRecord metadata with unspecified audio source
		if (mAttr.source == AUDIO_SOURCE_DEFAULT) {
		return;
		}

		// No track is invalid as this is called after prepareTrack_l in the same critical section
		record_track_metadata_v7_t metadata;
		metadata.base = {
		.source = mAttr.source,
		.gain = 1, // capture tracks do not have volumes
		};
		metadata.channel_mask = mChannelMask;
		strncpy(metadata.tags, mAttr.tags, AUDIO_ATTRIBUTES_TAGS_MAX_SIZE);

		*backInserter++ = metadata;
		}

		// ----------------------------------------------------------------------------
		#undef LOG_TAG
		@@ -2763,9 +2780,10 @@ AudioFlinger::RecordThread::PatchRecord::PatchRecord(RecordThread *recordThread,
		void *buffer,
		size_t bufferSize,
		audio_input_flags_t flags,
		const Timeout& timeout)
		const Timeout& timeout,
		audio_source_t source)
		: RecordTrack(recordThread, NULL,
		audio_attributes_t{} /* currently unused for patch track */,
		audio_attributes_t{ .source = source } ,
		sampleRate, format, channelMask, frameCount,
		buffer, bufferSize, AUDIO_SESSION_NONE, getpid(),
		audioServerAttributionSource(getpid()), flags, TYPE_PATCH),
		@@ -2876,9 +2894,10 @@ AudioFlinger::RecordThread::PassthruPatchRecord::PassthruPatchRecord(
		audio_channel_mask_t channelMask,
		audio_format_t format,
		size_t frameCount,
		audio_input_flags_t flags)
		audio_input_flags_t flags,
		audio_source_t source)
		: PatchRecord(recordThread, sampleRate, channelMask, format, frameCount,
		nullptr /buffer/, 0 /bufferSize/, flags),
		nullptr /buffer/, 0 /bufferSize/, flags, {} /* timeout */, source),
		mPatchRecordAudioBufferProvider(*this),
		mSinkBuffer(allocAligned(32, mFrameCount * mFrameSize)),
		mStubBuffer(allocAligned(32, mFrameCount * mFrameSize))

services/audiopolicy/managerdefault/AudioPolicyManager.cpp

+5 −3

Original line number	Diff line number	Diff line
		@@ -788,7 +788,8 @@ void AudioPolicyManager::connectTelephonyTxAudioSource(
		ALOGV("%s between source %s and sink %s", __func__,
		srcDevice->toString().c_str(), sinkDevice->toString().c_str());
		auto callTxSourceClientPortId = PolicyAudioPort::getNextUniqueId();
		const audio_attributes_t aa = { .source = AUDIO_SOURCE_VOICE_COMMUNICATION };
		const auto aa = mEngine->getAttributesForStreamType(AUDIO_STREAM_VOICE_CALL);

		struct audio_port_config source = {};
		srcDevice->toAudioPortConfig(&source);
		mCallTxSourceClient = new InternalSourceClientDescriptor(
		@@ -4534,7 +4535,7 @@ status_t AudioPolicyManager::createAudioPatchInternal(const struct audio_patch *
		// In case of Hw bridge, it is a Work Around. The mixPort used is the one declared
		// in config XML to reach the sink so that is can be declared as available.
		audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
		sp<SwAudioOutputDescriptor> outputDesc = nullptr;
		sp<SwAudioOutputDescriptor> outputDesc;
		if (!sourceDesc->isInternal()) {
		// take care of dynamic routing for SwOutput selection,
		audio_attributes_t attributes = sourceDesc->attributes();
		@@ -4604,7 +4605,8 @@ status_t AudioPolicyManager::createAudioPatchInternal(const struct audio_patch *
		audio_port_config srcMixPortConfig = {};
		outputDesc->toAudioPortConfig(&srcMixPortConfig, nullptr);
		// for volume control, we may need a valid stream
		srcMixPortConfig.ext.mix.usecase.stream = !sourceDesc->isInternal() ?
		srcMixPortConfig.ext.mix.usecase.stream =
		(!sourceDesc->isInternal() \|\| isCallTxAudioSource(sourceDesc)) ?
		mEngine->getStreamTypeForAttributes(sourceDesc->attributes()) :
		AUDIO_STREAM_PATCH;
		patchBuilder.addSource(srcMixPortConfig);