diff options
Diffstat (limited to 'include/media/AudioTrack.h')
| -rw-r--r-- | include/media/AudioTrack.h | 136 |
1 files changed, 85 insertions, 51 deletions
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h index fd51b8f..d9b7057 100644 --- a/include/media/AudioTrack.h +++ b/include/media/AudioTrack.h @@ -63,7 +63,7 @@ public: // See AudioTimestamp for the information included with event. }; - /* Client should declare Buffer on the stack and pass address to obtainBuffer() + /* Client should declare a Buffer and pass the address to obtainBuffer() * and releaseBuffer(). See also callback_t for EVENT_MORE_DATA. */ @@ -72,22 +72,26 @@ public: public: // FIXME use m prefix size_t frameCount; // number of sample frames corresponding to size; - // on input it is the number of frames desired, - // on output is the number of frames actually filled - // (currently ignored, but will make the primary field in future) + // on input to obtainBuffer() it is the number of frames desired, + // on output from obtainBuffer() it is the number of available + // [empty slots for] frames to be filled + // on input to releaseBuffer() it is currently ignored size_t size; // input/output in bytes == frameCount * frameSize - // on input it is unused - // on output is the number of bytes actually filled - // 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 + // on input to obtainBuffer() it is ignored + // on output from obtainBuffer() it is the number of available + // [empty slots for] bytes to be filled, + // which is frameCount * frameSize + // on input to releaseBuffer() it is the number of bytes to + // release + // FIXME This is redundant with respect to frameCount. Consider + // removing size and making frameCount the primary field. union { void* raw; short* i16; // signed 16-bit int8_t* i8; // unsigned 8-bit, offset by 0x80 - }; // input: unused, output: pointer to buffer + }; // input to obtainBuffer(): unused, output: pointer to buffer }; /* As a convenience, if a callback is supplied, a handler thread @@ -121,6 +125,7 @@ public: * - BAD_VALUE: unsupported configuration * frameCount is guaranteed to be non-zero if status is NO_ERROR, * and is undefined otherwise. + * FIXME This API assumes a route, and so should be deprecated. */ static status_t getMinFrameCount(size_t* frameCount, @@ -132,7 +137,7 @@ public: enum transfer_type { 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_OBTAIN, // call obtainBuffer() and releaseBuffer() TRANSFER_SYNC, // synchronous write() TRANSFER_SHARED, // shared memory }; @@ -145,18 +150,15 @@ public: /* Creates an AudioTrack object and registers it with AudioFlinger. * Once created, the track needs to be started before it can be used. * Unspecified values are set to appropriate default values. - * With this constructor, the track is configured for streaming mode. - * Data to be rendered is supplied by write() or by the callback EVENT_MORE_DATA. - * Intermixing a combination of write() and non-ignored EVENT_MORE_DATA is not allowed. * * Parameters: * * streamType: Select the type of audio stream this track is attached to * (e.g. AUDIO_STREAM_MUSIC). * sampleRate: Data source sampling rate in Hz. - * format: Audio format. For mixed tracks, any PCM format supported by server is OK - * or AUDIO_FORMAT_PCM_8_BIT which is handled on client side. For direct - * and offloaded tracks, the possible format(s) depends on the output sink. + * format: Audio format. For mixed tracks, any PCM format supported by server is OK. + * For direct and offloaded tracks, the possible format(s) depends on the + * output sink. * 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 @@ -165,20 +167,28 @@ public: * configuration. Zero means to use a default value. * flags: See comments on audio_output_flags_t in <system/audio.h>. * cbf: Callback function. If not null, this function is called periodically - * to provide new data and inform of marker, position updates, etc. + * to provide new data in TRANSFER_CALLBACK mode + * 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 have been consumed from track input buffer. * This is expressed in units of frames at the initial source sample rate. * sessionId: Specific session ID, or zero to use default. * transferType: How data is transferred to AudioTrack. + * offloadInfo: If not NULL, provides offload parameters for + * AudioSystem::getOutputForAttr(). + * uid: User ID of the app which initially requested this AudioTrack + * for power management tracking, or -1 for current user ID. + * pid: Process ID of the app which initially requested this AudioTrack + * for power management tracking, or -1 for current process ID. + * pAttributes: If not NULL, supersedes streamType for use case selection. * threadCanCallJava: Not present in parameter list, and so is fixed at false. */ AudioTrack( audio_stream_type_t streamType, uint32_t sampleRate, audio_format_t format, - audio_channel_mask_t, + audio_channel_mask_t channelMask, size_t frameCount = 0, audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, callback_t cbf = NULL, @@ -193,9 +203,10 @@ public: /* Creates an audio track and registers it with AudioFlinger. * With this constructor, the track is configured for static buffer mode. - * The format must not be 8-bit linear PCM. * Data to be rendered is passed in a shared memory buffer - * identified by the argument sharedBuffer, which must be non-0. + * identified by the argument sharedBuffer, which should be non-0. + * If sharedBuffer is zero, this constructor is equivalent to the previous constructor + * but without the ability to specify a non-zero value for the frameCount parameter. * The memory should be initialized to the desired data before calling start(). * The write() method is not supported in this case. * It is recommended to pass a callback function to be notified of playback end by an @@ -227,6 +238,7 @@ public: /* Initialize an AudioTrack that was created using the AudioTrack() constructor. * Don't call set() more than once, or after the AudioTrack() constructors that take parameters. + * set() is not multi-thread safe. * Returned status (from utils/Errors.h) can be: * - NO_ERROR: successful initialization * - INVALID_OPERATION: AudioTrack is already initialized @@ -461,7 +473,9 @@ public: * handle on audio hardware output, or AUDIO_IO_HANDLE_NONE if the * track needed to be re-created but that failed */ +private: audio_io_handle_t getOutput() const; +public: /* Returns the unique session ID associated with this track. * @@ -487,10 +501,18 @@ public: */ status_t attachAuxEffect(int effectId); - /* Obtains a buffer of up to "audioBuffer->frameCount" empty slots for frames. + /* Public API for TRANSFER_OBTAIN mode. + * Obtains a buffer of up to "audioBuffer->frameCount" empty slots for frames. * After filling these slots with data, the caller should release them with releaseBuffer(). * If the track buffer is not full, obtainBuffer() returns as many contiguous * [empty slots for] frames as are available immediately. + * + * If nonContig is non-NULL, it is an output parameter that will be set to the number of + * additional non-contiguous frames that are predicted to be available immediately, + * if the client were to release the first frames and then call obtainBuffer() again. + * This value is only a prediction, and needs to be confirmed. + * It will be set to zero for an error return. + * * If the track buffer is full and track is stopped, obtainBuffer() returns WOULD_BLOCK * regardless of the value of waitCount. * If the track buffer is full and track is not stopped, obtainBuffer() blocks with a @@ -499,10 +521,6 @@ public: * is exhausted, at which point obtainBuffer() will either block * or return WOULD_BLOCK depending on the value of the "waitCount" * parameter. - * Each sample is 16-bit signed PCM. - * - * obtainBuffer() and releaseBuffer() are deprecated for direct use by applications, - * which should use write() or callback EVENT_MORE_DATA instead. * * Interpretation of waitCount: * +n limits wait time to n * WAIT_PERIOD_MS, @@ -511,24 +529,27 @@ public: * * Buffer fields * On entry: - * frameCount number of frames requested + * frameCount number of [empty slots for] frames requested + * size ignored + * raw ignored * After error return: * frameCount 0 * size 0 * raw undefined * After successful return: - * frameCount actual number of frames available, <= number requested + * frameCount actual number of [empty slots for] frames available, <= number requested * size actual number of bytes available * raw pointer to the buffer */ - - /* FIXME Deprecated public API for TRANSFER_OBTAIN mode */ - status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount) - __attribute__((__deprecated__)); + status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount, + size_t *nonContig = NULL); private: /* If nonContig is non-NULL, it is an output parameter that will be set to the number of - * additional non-contiguous frames that are available immediately. + * additional non-contiguous frames that are predicted to be available immediately, + * if the client were to release the first frames and then call obtainBuffer() again. + * This value is only a prediction, and needs to be confirmed. + * It will be set to zero for an error return. * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(), * in case the requested amount of frames is in two or more non-contiguous regions. * FIXME requested and elapsed are both relative times. Consider changing to absolute time. @@ -537,9 +558,15 @@ private: struct timespec *elapsed = NULL, size_t *nonContig = NULL); public: - /* Release a filled buffer of "audioBuffer->frameCount" frames for AudioFlinger to process. */ - // FIXME make private when obtainBuffer() for TRANSFER_OBTAIN is removed - void releaseBuffer(Buffer* audioBuffer); + /* Public API for TRANSFER_OBTAIN mode. + * Release a filled buffer of frames for AudioFlinger to process. + * + * Buffer fields: + * frameCount currently ignored but recommend to set to actual number of frames filled + * size actual number of bytes filled, must be multiple of frameSize + * raw ignored + */ + void releaseBuffer(const Buffer* audioBuffer); /* As a convenience we provide a write() interface to the audio buffer. * Input parameter 'size' is in byte units. @@ -551,7 +578,7 @@ public: * WOULD_BLOCK when obtainBuffer() returns same, or * AudioTrack was stopped during the write * or any other error code returned by IAudioTrack::start() or restoreTrack_l(). - * Default behavior is to only return until all data has been transferred. Set 'blocking' to + * Default behavior is to only return when all data has been transferred. Set 'blocking' to * false for the method to return immediately without waiting to try multiple times to write * the full content of the buffer. */ @@ -559,6 +586,7 @@ public: /* * Dumps the state of an audio track. + * Not a general-purpose API; intended only for use by media player service to dump its tracks. */ status_t dump(int fd, const Vector<String16>& args) const; @@ -600,8 +628,6 @@ protected: AudioTrack(const AudioTrack& other); AudioTrack& operator = (const AudioTrack& other); - void setAttributesFromStreamType(audio_stream_type_t streamType); - /* a small internal class to handle the callback */ class AudioTrackThread : public Thread { @@ -614,6 +640,7 @@ protected: void pause(); // suspend thread from execution at next loop boundary void resume(); // allow thread to execute, if not requested to exit + void wake(); // wake to handle changed notification conditions. private: void pauseInternal(nsecs_t ns = 0LL); @@ -628,7 +655,9 @@ protected: bool mPaused; // whether thread is requested to pause at next loop entry bool mPausedInt; // whether thread internally requests pause nsecs_t mPausedNs; // if mPausedInt then associated timeout, otherwise ignored - bool mIgnoreNextPausedInt; // whether to ignore next mPausedInt request + bool mIgnoreNextPausedInt; // skip any internal pause and go immediately + // to processAudioBuffer() as state may have changed + // since pause time calculated. }; // body of AudioTrackThread::threadLoop() @@ -641,10 +670,6 @@ protected: static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3; nsecs_t processAudioBuffer(); - bool isOffloaded() const; - bool isDirect() const; - bool isOffloadedOrDirect() const; - // caller must hold lock on mLock for all _l methods status_t createTrack_l(); @@ -657,6 +682,10 @@ protected: // FIXME enum is faster than strcmp() for parameter 'from' status_t restoreTrack_l(const char *from); + bool isOffloaded() const; + bool isDirect() const; + bool isOffloadedOrDirect() const; + bool isOffloaded_l() const { return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; } @@ -680,7 +709,7 @@ protected: float mVolume[2]; float mSendLevel; - mutable uint32_t mSampleRate; // mutable because getSampleRate() can update it. + mutable uint32_t mSampleRate; // mutable because getSampleRate() can update it size_t mFrameCount; // corresponds to current IAudioTrack, value is // reported back by AudioFlinger to the client size_t mReqFrameCount; // frame count to request the first or next time @@ -698,10 +727,7 @@ protected: const audio_offload_info_t* mOffloadInfo; audio_attributes_t mAttributes; - // mFrameSize is equal to mFrameSizeAF for non-PCM or 16-bit PCM data. For 8-bit PCM data, it's - // twice as large as mFrameSize because data is expanded to 16-bit before it's stored in buffer. - size_t mFrameSize; // app-level frame size - size_t mFrameSizeAF; // AudioFlinger frame size + size_t mFrameSize; // frame size in bytes status_t mStatus; @@ -732,17 +758,25 @@ protected: bool mRefreshRemaining; // processAudioBuffer() should refresh // mRemainingFrames and mRetryOnPartialBuffer + // used for static track cbf and restoration + int32_t mLoopCount; // last setLoop loopCount; zero means disabled + uint32_t mLoopStart; // last setLoop loopStart + uint32_t mLoopEnd; // last setLoop loopEnd + int32_t mLoopCountNotified; // the last loopCount notified by callback. + // mLoopCountNotified counts down, matching + // the remaining loop count for static track + // playback. + // These are private to processAudioBuffer(), and are not protected by a lock uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() uint32_t mObservedSequence; // last observed value of mSequence - 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 uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS + uint32_t mServer; // in frames, last known mProxy->getPosition() // which is count of frames consumed by server, // reset by new IAudioTrack, |