Multi-client recording (6dd62fb9) · Commits · e / os / android_frameworks_av

media/libmedia/AudioRecord.cpp

+3 −1

Original line number	Diff line number	Diff line
		@@ -462,7 +462,9 @@ status_t AudioRecord::openRecord_l(size_t epoch)
		audio_io_handle_t input = AudioSystem::getInput(mInputSource, mSampleRate, mFormat,
		mChannelMask, mSessionId);
		if (input == 0) {
		ALOGE("Could not get audio input for record source %d", mInputSource);
		ALOGE("Could not get audio input for record source %d, sample rate %u, format %#x, "
		"channel mask %#x, session %d",
		mInputSource, mSampleRate, mFormat, mChannelMask, mSessionId);
		return BAD_VALUE;
		}
		{

services/audioflinger/RecordTracks.h

+29 −0

Original line number	Diff line number	Diff line
		@@ -59,4 +59,33 @@ private:
		// releaseBuffer() not overridden

		bool mOverflow; // overflow on most recent attempt to fill client buffer

		// updated by RecordThread::readInputParameters()
		AudioResampler *mResampler;

		// interleaved stereo pairs of fixed-point signed Q19.12
		int32_t *mRsmpOutBuffer;
		// current allocated frame count for the above, which may be larger than needed
		size_t mRsmpOutFrameCount;

		size_t mRsmpInUnrel; // unreleased frames remaining from
		// most recent getNextBuffer
		// for debug only

		// rolling counter that is never cleared
		int32_t mRsmpInFront; // next available frame

		AudioBufferProvider::Buffer mSink; // references client's buffer sink in shared memory

		// sync event triggering actual audio capture. Frames read before this event will
		// be dropped and therefore not read by the application.
		sp<SyncEvent> mSyncStartEvent;

		// number of captured frames to drop after the start sync event has been received.
		// when < 0, maximum frames to drop before starting capture even if sync event is
		// not received
		ssize_t mFramesToDrop;

		// used by resampler to find source frames
		ResamplerBufferProvider *mResamplerBufferProvider;
		};

services/audioflinger/Threads.cpp

+334 −287

File changed.

Preview size limit exceeded, changes collapsed.

services/audioflinger/Threads.h

+19 −26

Original line number	Diff line number	Diff line
		@@ -839,10 +839,23 @@ public:


		// record thread
		class RecordThread : public ThreadBase, public AudioBufferProvider
		class RecordThread : public ThreadBase
		{
		public:

		class RecordTrack;
		class ResamplerBufferProvider : public AudioBufferProvider
		// derives from AudioBufferProvider interface for use by resampler
		{
		public:
		ResamplerBufferProvider(RecordTrack* recordTrack) : mRecordTrack(recordTrack) { }
		virtual ~ResamplerBufferProvider() { }
		// AudioBufferProvider interface
		virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts);
		virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
		private:
		RecordTrack * const mRecordTrack;
		};

		#include "RecordTracks.h"

		@@ -898,9 +911,6 @@ public:
		AudioStreamIn* clearInput();
		virtual audio_stream_t* stream() const;

		// AudioBufferProvider interface
		virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts);
		virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);

		virtual bool checkForNewParameters_l();
		virtual String8 getParameters(const String8& keys);
		@@ -921,13 +931,13 @@ public:
		virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const;

		static void syncStartEventCallback(const wp<SyncEvent>& event);
		void handleSyncStartEvent(const sp<SyncEvent>& event);
		void handleSyncStartEvent(RecordTrack *recordTrack, const sp<SyncEvent>& event);

		virtual size_t frameCount() const { return mFrameCount; }
		bool hasFastRecorder() const { return false; }

		private:
		void clearSyncStartEvent();
		void clearSyncStartEvent(RecordTrack* recordTrack);

		// Enter standby if not already in standby, and set mStandby flag
		void standbyIfNotAlreadyInStandby();
		@@ -944,35 +954,18 @@ private:
		int mActiveTracksGen;
		Condition mStartStopCond;

		// updated by RecordThread::readInputParameters()
		AudioResampler *mResampler;
		// interleaved stereo pairs of fixed-point signed Q19.12
		int32_t *mRsmpOutBuffer;

		// resampler converts input at HAL Hz to output at AudioRecord client Hz
		int16_t *mRsmpInBuffer; // see new[] for details on the size
		size_t mRsmpInFrames; // size of resampler input in frames
		size_t mRsmpInFramesP2;// size rounded up to a power-of-2
		size_t mRsmpInUnrel; // unreleased frames remaining from
		// most recent getNextBuffer
		// these are rolling counters that are never cleared
		int32_t mRsmpInFront; // next available frame

		// rolling index that is never cleared
		int32_t mRsmpInRear; // last filled frame + 1
		size_t mRsmpInIndex; // FIXME legacy

		// client's requested configuration, which may differ from the HAL configuration
		const uint32_t mReqChannelCount;
		const uint32_t mReqSampleRate;

		ssize_t mBytesRead;
		// sync event triggering actual audio capture. Frames read before this event will
		// be dropped and therefore not read by the application.
		sp<SyncEvent> mSyncStartEvent;
		// number of captured frames to drop after the start sync event has been received.
		// when < 0, maximum frames to drop before starting capture even if sync event is
		// not received
		ssize_t mFramestoDrop;

		// For dumpsys
		const sp<NBAIO_Sink> mTeeSink;
		};

services/audioflinger/TrackBase.h

+3 −1

Original line number	Diff line number	Diff line
		@@ -34,7 +34,9 @@ public:
		RESUMING,
		ACTIVE,
		PAUSING,
		PAUSED
		PAUSED,
		STARTING_1, // for RecordTrack only
		STARTING_2, // for RecordTrack only
		};

		TrackBase(ThreadBase *thread,