Merge "Use fast mode with patch track and patch record if possible." (d02904c6) · Commits · e / os / android_frameworks_av

services/audioflinger/FastCapture.cpp

+28 −5

Original line number	Diff line number	Diff line
		@@ -20,6 +20,7 @@
		#define ATRACE_TAG ATRACE_TAG_AUDIO

		#include "Configuration.h"
		#include <audio_utils/format.h>
		#include <linux/futex.h>
		#include <sys/syscall.h>
		#include <media/AudioBufferProvider.h>
		@@ -161,7 +162,21 @@ void FastCapture::onWork()
		const FastCaptureState * const current = (const FastCaptureState *) mCurrent;
		FastCaptureDumpState * const dumpState = (FastCaptureDumpState *) mDumpState;
		const FastCaptureState::Command command = mCommand;
		const size_t frameCount = current->mFrameCount;
		size_t frameCount = current->mFrameCount;
		AudioBufferProvider* fastPatchRecordBufferProvider = current->mFastPatchRecordBufferProvider;
		AudioBufferProvider::Buffer patchBuffer;

		if (fastPatchRecordBufferProvider != 0) {
		patchBuffer.frameCount = ~0;
		status_t status = fastPatchRecordBufferProvider->getNextBuffer(&patchBuffer);
		if (status != NO_ERROR) {
		frameCount = 0;
		} else if (patchBuffer.frameCount < frameCount) {
		// TODO: Make sure that it doesn't cause any issues if we just get a small available
		// buffer from the buffer provider.
		frameCount = patchBuffer.frameCount;
		}
		}

		if ((command & FastCaptureState::READ) /&& isWarm/) {
		ALOG_ASSERT(mInputSource != NULL);
		@@ -176,6 +191,7 @@ void FastCapture::onWork()
		mTotalNativeFramesRead += framesRead;
		dumpState->mFramesRead = mTotalNativeFramesRead;
		mReadBufferState = framesRead;
		patchBuffer.frameCount = framesRead;
		} else {
		dumpState->mReadErrors++;
		mReadBufferState = 0;
		@@ -193,11 +209,18 @@ void FastCapture::onWork()
		}
		if (mReadBufferState > 0) {
		ssize_t framesWritten = mPipeSink->write(mReadBuffer, mReadBufferState);
		audio_track_cblk_t* cblk = current->mCblk;
		if (fastPatchRecordBufferProvider != 0) {
		// This indicates the fast track is a patch record, update the cblk by
		// calling releaseBuffer().
		memcpy_by_audio_format(patchBuffer.raw, current->mFastPatchRecordFormat,
		mReadBuffer, mFormat.mFormat, framesWritten * mFormat.mChannelCount);
		patchBuffer.frameCount = framesWritten;
		fastPatchRecordBufferProvider->releaseBuffer(&patchBuffer);
		} else if (cblk != NULL && framesWritten > 0) {
		// FIXME This supports at most one fast capture client.
		// To handle multiple clients this could be converted to an array,
		// or with a lot more work the control block could be shared by all clients.
		audio_track_cblk_t* cblk = current->mCblk;
		if (cblk != NULL && framesWritten > 0) {
		int32_t rear = cblk->u.mStreaming.mRear;
		android_atomic_release_store(framesWritten + rear, &cblk->u.mStreaming.mRear);
		cblk->mServer += framesWritten;

services/audioflinger/FastCaptureState.h

+5 −0

Original line number	Diff line number	Diff line
		@@ -18,6 +18,7 @@
		#define ANDROID_AUDIO_FAST_CAPTURE_STATE_H

		#include <media/nbaio/NBAIO.h>
		#include <media/AudioBufferProvider.h>
		#include "FastThreadState.h"
		#include <private/media/AudioTrackShared.h>

		@@ -37,6 +38,10 @@ struct FastCaptureState : FastThreadState {
		size_t mFrameCount; // number of frames per fast capture buffer
		audio_track_cblk_t* mCblk; // control block for the single fast client, or NULL

		audio_format_t mFastPatchRecordFormat = AUDIO_FORMAT_INVALID;
		AudioBufferProvider* mFastPatchRecordBufferProvider = nullptr; // a reference to a patch
		// record in fast mode

		// Extends FastThreadState::Command
		static const Command
		// The following commands also process configuration changes, and can be "or"ed:

services/audioflinger/PatchPanel.cpp

+21 −5

Original line number	Diff line number	Diff line
		@@ -431,14 +431,14 @@ status_t AudioFlinger::PatchPanel::Patch::createConnections(PatchPanel *panel)
		// use a pseudo LCM between input and output framecount
		size_t playbackFrameCount = mPlayback.thread()->frameCount();
		int playbackShift = __builtin_ctz(playbackFrameCount);
		size_t recordFramecount = mRecord.thread()->frameCount();
		int shift = __builtin_ctz(recordFramecount);
		size_t recordFrameCount = mRecord.thread()->frameCount();
		int shift = __builtin_ctz(recordFrameCount);
		if (playbackShift < shift) {
		shift = playbackShift;
		}
		size_t frameCount = (playbackFrameCount * recordFramecount) >> shift;
		ALOGV("%s() playframeCount %zu recordFramecount %zu frameCount %zu",
		__func__, playbackFrameCount, recordFramecount, frameCount);
		size_t frameCount = (playbackFrameCount * recordFrameCount) >> shift;
		ALOGV("%s() playframeCount %zu recordFrameCount %zu frameCount %zu",
		__func__, playbackFrameCount, recordFrameCount, frameCount);

		// create a special record track to capture from record thread
		uint32_t channelCount = mPlayback.thread()->channelCount();
		@@ -455,6 +455,17 @@ status_t AudioFlinger::PatchPanel::Patch::createConnections(PatchPanel *panel)
		}
		audio_input_flags_t inputFlags = mAudioPatch.sources[0].config_mask & AUDIO_PORT_CONFIG_FLAGS ?
		mAudioPatch.sources[0].flags.input : AUDIO_INPUT_FLAG_NONE;
		if (sampleRate == mRecord.thread()->sampleRate() &&
		inChannelMask == mRecord.thread()->channelMask() &&
		mRecord.thread()->fastTrackAvailable() &&
		mRecord.thread()->hasFastCapture()) {
		// Create a fast track if the record thread has fast capture to get better performance.
		// Only enable fast mode when there is no resample needed.
		inputFlags = (audio_input_flags_t) (inputFlags \| AUDIO_INPUT_FLAG_FAST);
		} else {
		// Fast mode is not available in this case.
		inputFlags = (audio_input_flags_t) (inputFlags & ~AUDIO_INPUT_FLAG_FAST);
		}
		sp<RecordThread::PatchRecord> tempRecordTrack = new (std::nothrow) RecordThread::PatchRecord(
		mRecord.thread().get(),
		sampleRate,
		@@ -476,6 +487,11 @@ status_t AudioFlinger::PatchPanel::Patch::createConnections(PatchPanel *panel)
		// "reuse one existing output mix" case
		streamType = mAudioPatch.sources[1].ext.mix.usecase.stream;
		}
		if (mPlayback.thread()->hasFastMixer()) {
		// Create a fast track if the playback thread has fast mixer to get better performance.
		outputFlags = (audio_output_flags_t) (outputFlags \| AUDIO_OUTPUT_FLAG_FAST);
		}

		// create a special playback track to render to playback thread.
		// this track is given the same buffer as the PatchRecord buffer
		sp<PlaybackThread::PatchTrack> tempPatchTrack = new (std::nothrow) PlaybackThread::PatchTrack(

services/audioflinger/Threads.cpp

+8 −0

Original line number	Diff line number	Diff line
		@@ -6700,6 +6700,14 @@ reacquire_wakelock:
		}
		didModify = true;
		}
		AudioBufferProvider* abp = (fastTrack != 0 && fastTrack->isPatchTrack()) ?
		reinterpret_cast<AudioBufferProvider*>(fastTrack.get()) : nullptr;
		if (state->mFastPatchRecordBufferProvider != abp) {
		state->mFastPatchRecordBufferProvider = abp;
		state->mFastPatchRecordFormat = fastTrack == 0 ?
		AUDIO_FORMAT_INVALID : fastTrack->format();
		didModify = true;
		}
		sq->end(didModify);
		if (didModify) {
		sq->push(block);

services/audioflinger/Threads.h

+2 −0

Original line number	Diff line number	Diff line
		@@ -1530,6 +1530,8 @@ public:

		void updateMetadata_l() override;

		bool fastTrackAvailable() const { return mFastTrackAvail; }

		private:
		// Enter standby if not already in standby, and set mStandby flag
		void standbyIfNotAlreadyInStandby();