Merge "Unify comments between AudioTrack and AudioRecord" (32a7e0a5) · Commits · e / os / android_frameworks_av

include/media/AudioRecord.h

+19 −10

Original line number	Diff line number	Diff line
		@@ -39,8 +39,12 @@ public:
		* Keep in sync with frameworks/base/media/java/android/media/AudioRecord.java NATIVE_EVENT_*.
		*/
		enum event_type {
		EVENT_MORE_DATA = 0, // Request to read more data from PCM buffer.
		EVENT_OVERRUN = 1, // PCM buffer overrun occurred.
		EVENT_MORE_DATA = 0, // Request to read available data from buffer.
		// If this event is delivered but the callback handler
		// does not want to read the available data, the handler must
		// explicitly
		// ignore the event by setting frameCount to zero.
		EVENT_OVERRUN = 1, // Buffer overrun occurred.
		EVENT_MARKER = 2, // Record head is at the specified marker position
		// (See setMarkerPosition()).
		EVENT_NEW_POS = 3, // Record head is at a new position
		@@ -63,6 +67,7 @@ public:
		// (currently ignored but will make the primary field in future)

		size_t size; // input/output in bytes == frameCount * frameSize
		// on output is the number of bytes actually drained
		// FIXME this is redundant with respect to frameCount,
		// and TRANSFER_OBTAIN mode is broken for 8-bit data
		// since we don't define the frame format
		@@ -76,7 +81,7 @@ public:

		/* As a convenience, if a callback is supplied, a handler thread
		* is automatically created with the appropriate priority. This thread
		* invokes the callback when a new buffer becomes ready or various conditions occur.
		* invokes the callback when a new buffer becomes available or various conditions occur.
		* Parameters:
		*
		* event: type of event notified (see enum AudioRecord::event_type).
		@@ -99,6 +104,8 @@ public:
		* - NO_ERROR: successful operation
		* - NO_INIT: audio server or audio hardware not initialized
		* - BAD_VALUE: unsupported configuration
		* frameCount is guaranteed to be non-zero if status is NO_ERROR,
		* and is undefined otherwise.
		*/

		static status_t getMinFrameCount(size_t* frameCount,
		@@ -109,7 +116,7 @@ public:
		/* How data is transferred from AudioRecord
		*/
		enum transfer_type {
		TRANSFER_DEFAULT, // not specified explicitly; determine from other parameters
		TRANSFER_DEFAULT, // not specified explicitly; determine from the other parameters
		TRANSFER_CALLBACK, // callback EVENT_MORE_DATA
		TRANSFER_OBTAIN, // FIXME deprecated: call obtainBuffer() and releaseBuffer()
		TRANSFER_SYNC, // synchronous read()
		@@ -137,7 +144,7 @@ public:
		* be larger if the requested size is not compatible with current audio HAL
		* latency. Zero means to use a default value.
		* cbf: Callback function. If not null, this function is called periodically
		* to consume new PCM data and inform of marker, position updates, etc.
		* to consume new data and inform of marker, position updates, etc.
		* user: Context for use by the callback receiver.
		* notificationFrames: The callback function is called each time notificationFrames PCM
		* frames are ready in record track output buffer.
		@@ -171,9 +178,10 @@ public:
		* Returned status (from utils/Errors.h) can be:
		* - NO_ERROR: successful intialization
		* - INVALID_OPERATION: AudioRecord is already initialized or record device is already in use
		* - BAD_VALUE: invalid parameter (channels, format, sampleRate...)
		* - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...)
		* - NO_INIT: audio server or audio hardware not initialized
		* - PERMISSION_DENIED: recording is not allowed for the requesting process
		* If status is not equal to NO_ERROR, don't call any other APIs on this AudioRecord.
		*
		* Parameters not listed in the AudioRecord constructors above:
		*
		@@ -192,7 +200,7 @@ public:
		transfer_type transferType = TRANSFER_DEFAULT,
		audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE);

		/* Result of constructing the AudioRecord. This must be checked
		/* Result of constructing the AudioRecord. This must be checked for successful initialization
		* before using any AudioRecord API (except for set()), because using
		* an uninitialized AudioRecord produces undefined results.
		* See set() method above for possible return codes.
		@@ -221,7 +229,7 @@ public:
		status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE,
		int triggerSession = 0);

		/* Stop a track. If set, the callback will cease being called. Note that obtainBuffer() still
		/* Stop a track. The callback will cease being called. Note that obtainBuffer() still
		* works and will drain buffers until the pool is exhausted, and then will return WOULD_BLOCK.
		*/
		void stop();
		@@ -236,7 +244,7 @@ public:
		* a callback with event type EVENT_MARKER is called. Calling setMarkerPosition
		* with marker == 0 cancels marker notification callback.
		* To set a marker at a position which would compute as 0,
		* a workaround is to the set the marker at a nearby position such as ~0 or 1.
		* a workaround is to set the marker at a nearby position such as ~0 or 1.
		* If the AudioRecord has been opened with no callback function associated,
		* the operation will fail.
		*
		@@ -426,6 +434,7 @@ private:
		nsecs_t processAudioBuffer();

		// caller must hold lock on mLock for all _l methods

		status_t openRecord_l(size_t epoch);

		// FIXME enum is faster than strcmp() for parameter 'from'
		@@ -477,7 +486,7 @@ private:

		audio_io_handle_t mInput; // returned by AudioSystem::getInput()

		// may be changed if IAudioRecord object is re-created
		// Next 3 fields may be changed if IAudioRecord is re-created, but always != 0
		sp<IAudioRecord> mAudioRecord;
		sp<IMemory> mCblkMemory;
		audio_track_cblk_t* mCblk; // re-load after mLock.unlock()

include/media/AudioTrack.h

+4 −3

Original line number	Diff line number	Diff line
		@@ -160,7 +160,7 @@ public:
		* sampleRate: Data source sampling rate in Hz.
		* format: Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed
		* 16 bits per sample).
		* channelMask: Channel mask.
		* channelMask: Channel mask, such that audio_is_output_channel(channelMask) is true.
		* frameCount: Minimum size of track PCM buffer in frames. This defines the
		* application's contribution to the
		* latency of the track. The actual size selected by the AudioTrack could be
		@@ -338,7 +338,7 @@ public:
		*/
		status_t setSampleRate(uint32_t sampleRate);

		/* Return current source sample rate in Hz, or 0 if unknown */
		/* Return current source sample rate in Hz */
		uint32_t getSampleRate() const;

		/* Enables looping and sets the start and end points of looping.
		@@ -363,7 +363,7 @@ public:
		/* Sets marker position. When playback reaches the number of frames specified, a callback with
		* event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker
		* notification callback. To set a marker at a position which would compute as 0,
		* a workaround is to the set the marker at a nearby position such as ~0 or 1.
		* a workaround is to set the marker at a nearby position such as ~0 or 1.
		* If the AudioTrack has been opened with no callback function associated, the operation will
		* fail.
		*
		@@ -717,6 +717,7 @@ protected:

		sp<IMemory> mSharedBuffer;
		uint32_t mLoopPeriod; // in frames, zero means looping is disabled

		uint32_t mMarkerPosition; // in wrapping (overflow) frame units
		bool mMarkerReached;
		uint32_t mNewPosition; // in frames

media/libmedia/AudioRecord.cpp

+5 −3

Original line number	Diff line number	Diff line
		@@ -352,6 +352,7 @@ bool AudioRecord::stopped() const

		status_t AudioRecord::setMarkerPosition(uint32_t marker)
		{
		// The only purpose of setting marker position is to get a callback
		if (mCbf == NULL) {
		return INVALID_OPERATION;
		}
		@@ -377,6 +378,7 @@ status_t AudioRecord::getMarkerPosition(uint32_t *marker) const

		status_t AudioRecord::setPositionUpdatePeriod(uint32_t updatePeriod)
		{
		// The only purpose of setting position update period is to get a callback
		if (mCbf == NULL) {
		return INVALID_OPERATION;
		}
		@@ -770,7 +772,7 @@ nsecs_t AudioRecord::processAudioBuffer()
		int32_t sequence = mSequence;

		// These fields don't need to be cached, because they are assigned only by set():
		// mTransfer, mCbf, mUserData, mSampleRate
		// mTransfer, mCbf, mUserData, mSampleRate, mFrameSize

		mLock.unlock();

		@@ -844,8 +846,8 @@ nsecs_t AudioRecord::processAudioBuffer()
		"obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount);
		requested = &ClientProxy::kNonBlocking;
		size_t avail = audioBuffer.frameCount + nonContig;
		ALOGV("obtainBuffer(%u) returned %u = %u + %u",
		mRemainingFrames, avail, audioBuffer.frameCount, nonContig);
		ALOGV("obtainBuffer(%u) returned %u = %u + %u err %d",
		mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err);
		if (err != NO_ERROR) {
		if (err == TIMED_OUT \|\| err == WOULD_BLOCK \|\| err == -EINTR) {
		break;

media/libmedia/AudioTrack.cpp

+3 −2

Original line number	Diff line number	Diff line
		@@ -698,6 +698,7 @@ status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod)
		AutoMutex lock(mLock);
		mNewPosition = mProxy->getPosition() + updatePeriod;
		mUpdatePeriod = updatePeriod;

		return NO_ERROR;
		}