| /* |
| * Copyright (C) 2007 The Android Open Source Project |
| * |
| * Licensed under the Apache License, Version 2.0 (the "License"); |
| * you may not use this file except in compliance with the License. |
| * You may obtain a copy of the License at |
| * |
| * http://www.apache.org/licenses/LICENSE-2.0 |
| * |
| * Unless required by applicable law or agreed to in writing, software |
| * distributed under the License is distributed on an "AS IS" BASIS, |
| * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. |
| * See the License for the specific language governing permissions and |
| * limitations under the License. |
| */ |
| |
| #ifndef ANDROID_AUDIOTRACK_H |
| #define ANDROID_AUDIOTRACK_H |
| |
| #include <stdint.h> |
| #include <sys/types.h> |
| |
| #include <media/IAudioFlinger.h> |
| #include <media/IAudioTrack.h> |
| #include <media/AudioSystem.h> |
| |
| #include <utils/RefBase.h> |
| #include <utils/Errors.h> |
| #include <binder/IInterface.h> |
| #include <binder/IMemory.h> |
| #include <cutils/sched_policy.h> |
| #include <utils/threads.h> |
| |
| namespace android { |
| |
| // ---------------------------------------------------------------------------- |
| |
| class audio_track_cblk_t; |
| class AudioTrackClientProxy; |
| |
| // ---------------------------------------------------------------------------- |
| |
| class AudioTrack : virtual public RefBase |
| { |
| public: |
| enum channel_index { |
| MONO = 0, |
| LEFT = 0, |
| RIGHT = 1 |
| }; |
| |
| /* Events used by AudioTrack callback function (audio_track_cblk_t). |
| * Keep in sync with frameworks/base/media/java/android/media/AudioTrack.java NATIVE_EVENT_*. |
| */ |
| enum event_type { |
| EVENT_MORE_DATA = 0, // Request to write more data to buffer. |
| // If this event is delivered but the callback handler |
| // does not want to write more data, the handler must explicitly |
| // ignore the event by setting frameCount to zero. |
| EVENT_UNDERRUN = 1, // Buffer underrun occurred. |
| EVENT_LOOP_END = 2, // Sample loop end was reached; playback restarted from |
| // loop start if loop count was not 0. |
| EVENT_MARKER = 3, // Playback head is at the specified marker position |
| // (See setMarkerPosition()). |
| EVENT_NEW_POS = 4, // Playback head is at a new position |
| // (See setPositionUpdatePeriod()). |
| EVENT_BUFFER_END = 5 // Playback head is at the end of the buffer. |
| }; |
| |
| /* Client should declare Buffer on the stack and pass address to obtainBuffer() |
| * and releaseBuffer(). See also callback_t for EVENT_MORE_DATA. |
| */ |
| |
| class Buffer |
| { |
| public: |
| 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 |
| |
| size_t size; // input/output in byte units |
| union { |
| void* raw; |
| short* i16; // signed 16-bit |
| int8_t* i8; // unsigned 8-bit, offset by 0x80 |
| }; |
| }; |
| |
| |
| /* 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 available or various conditions occur. |
| * Parameters: |
| * |
| * event: type of event notified (see enum AudioTrack::event_type). |
| * user: Pointer to context for use by the callback receiver. |
| * info: Pointer to optional parameter according to event type: |
| * - EVENT_MORE_DATA: pointer to AudioTrack::Buffer struct. The callback must not write |
| * more bytes than indicated by 'size' field and update 'size' if fewer bytes are |
| * written. |
| * - EVENT_UNDERRUN: unused. |
| * - EVENT_LOOP_END: pointer to an int indicating the number of loops remaining. |
| * - EVENT_MARKER: pointer to an uint32_t containing the marker position in frames. |
| * - EVENT_NEW_POS: pointer to an uint32_t containing the new position in frames. |
| * - EVENT_BUFFER_END: unused. |
| */ |
| |
| typedef void (*callback_t)(int event, void* user, void *info); |
| |
| /* Returns the minimum frame count required for the successful creation of |
| * an AudioTrack object. |
| * Returned status (from utils/Errors.h) can be: |
| * - NO_ERROR: successful operation |
| * - NO_INIT: audio server or audio hardware not initialized |
| */ |
| |
| static status_t getMinFrameCount(size_t* frameCount, |
| audio_stream_type_t streamType = AUDIO_STREAM_DEFAULT, |
| uint32_t sampleRate = 0); |
| |
| /* Constructs an uninitialized AudioTrack. No connection with |
| * AudioFlinger takes place. Use set() after this. |
| */ |
| AudioTrack(); |
| |
| /* 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 deprecated. |
| * |
| * Parameters: |
| * |
| * streamType: Select the type of audio stream this track is attached to |
| * (e.g. AUDIO_STREAM_MUSIC). |
| * sampleRate: Track sampling rate in Hz. |
| * format: Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed |
| * 16 bits per sample). |
| * channelMask: Channel mask. |
| * 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 |
| * larger if the requested size is not compatible with current audio HAL |
| * 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. |
| * 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. |
| * sessionId: Specific session ID, or zero to use default. |
| * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI. |
| * If not present in parameter list, then fixed at false. |
| */ |
| |
| AudioTrack( audio_stream_type_t streamType, |
| uint32_t sampleRate = 0, |
| audio_format_t format = AUDIO_FORMAT_DEFAULT, |
| audio_channel_mask_t channelMask = 0, |
| int frameCount = 0, |
| audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, |
| callback_t cbf = NULL, |
| void* user = NULL, |
| int notificationFrames = 0, |
| int sessionId = 0); |
| |
| /* 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. |
| * 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 |
| * EVENT_UNDERRUN event. |
| * FIXME EVENT_MORE_DATA still occurs; it must be ignored. |
| */ |
| |
| AudioTrack( audio_stream_type_t streamType, |
| uint32_t sampleRate = 0, |
| audio_format_t format = AUDIO_FORMAT_DEFAULT, |
| audio_channel_mask_t channelMask = 0, |
| const sp<IMemory>& sharedBuffer = 0, |
| audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, |
| callback_t cbf = NULL, |
| void* user = NULL, |
| int notificationFrames = 0, |
| int sessionId = 0); |
| |
| /* Terminates the AudioTrack and unregisters it from AudioFlinger. |
| * Also destroys all resources associated with the AudioTrack. |
| */ |
| ~AudioTrack(); |
| |
| /* Initialize an uninitialized AudioTrack. |
| * Returned status (from utils/Errors.h) can be: |
| * - NO_ERROR: successful initialization |
| * - INVALID_OPERATION: AudioTrack is already initialized |
| * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...) |
| * - NO_INIT: audio server or audio hardware not initialized |
| * If sharedBuffer is non-0, the frameCount parameter is ignored and |
| * replaced by the shared buffer's total allocated size in frame units. |
| */ |
| status_t set(audio_stream_type_t streamType = AUDIO_STREAM_DEFAULT, |
| uint32_t sampleRate = 0, |
| audio_format_t format = AUDIO_FORMAT_DEFAULT, |
| audio_channel_mask_t channelMask = 0, |
| int frameCount = 0, |
| audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, |
| callback_t cbf = NULL, |
| void* user = NULL, |
| int notificationFrames = 0, |
| const sp<IMemory>& sharedBuffer = 0, |
| bool threadCanCallJava = false, |
| int sessionId = 0); |
| |
| /* Result of constructing the AudioTrack. This must be checked |
| * before using any AudioTrack API (except for set()), because using |
| * an uninitialized AudioTrack produces undefined results. |
| * See set() method above for possible return codes. |
| */ |
| status_t initCheck() const { return mStatus; } |
| |
| /* Returns this track's estimated latency in milliseconds. |
| * This includes the latency due to AudioTrack buffer size, AudioMixer (if any) |
| * and audio hardware driver. |
| */ |
| uint32_t latency() const { return mLatency; } |
| |
| /* getters, see constructors and set() */ |
| |
| audio_stream_type_t streamType() const { return mStreamType; } |
| audio_format_t format() const { return mFormat; } |
| |
| /* Return frame size in bytes, which for linear PCM is channelCount * (bit depth per channel / 8). |
| * channelCount is determined from channelMask, and bit depth comes from format. |
| * For non-linear formats, the frame size is typically 1 byte. |
| */ |
| uint32_t channelCount() const { return mChannelCount; } |
| |
| uint32_t frameCount() const { return mFrameCount; } |
| size_t frameSize() const { return mFrameSize; } |
| |
| /* Return the static buffer specified in constructor or set(), or 0 for streaming mode */ |
| sp<IMemory> sharedBuffer() const { return mSharedBuffer; } |
| |
| /* After it's created the track is not active. Call start() to |
| * make it active. If set, the callback will start being called. |
| * If the track was previously paused, volume is ramped up over the first mix buffer. |
| */ |
| void start(); |
| |
| /* Stop a track. |
| * In static buffer mode, the track is stopped immediately. |
| * In streaming mode, the callback will cease being called and |
| * obtainBuffer returns STOPPED. Note that obtainBuffer() still works |
| * and will fill up buffers until the pool is exhausted. |
| * The stop does not occur immediately: any data remaining in the buffer |
| * is first drained, mixed, and output, and only then is the track marked as stopped. |
| */ |
| void stop(); |
| bool stopped() const; |
| |
| /* Flush a stopped or paused track. All previously buffered data is discarded immediately. |
| * This has the effect of draining the buffers without mixing or output. |
| * Flush is intended for streaming mode, for example before switching to non-contiguous content. |
| * This function is a no-op if the track is not stopped or paused, or uses a static buffer. |
| */ |
| void flush(); |
| |
| /* Pause a track. After pause, the callback will cease being called and |
| * obtainBuffer returns STOPPED. Note that obtainBuffer() still works |
| * and will fill up buffers until the pool is exhausted. |
| * Volume is ramped down over the next mix buffer following the pause request, |
| * and then the track is marked as paused. It can be resumed with ramp up by start(). |
| */ |
| void pause(); |
| |
| /* Set volume for this track, mostly used for games' sound effects |
| * left and right volumes. Levels must be >= 0.0 and <= 1.0. |
| * This is the older API. New applications should use setVolume(float) when possible. |
| */ |
| status_t setVolume(float left, float right); |
| |
| /* Set volume for all channels. This is the preferred API for new applications, |
| * especially for multi-channel content. |
| */ |
| status_t setVolume(float volume); |
| |
| /* Set the send level for this track. An auxiliary effect should be attached |
| * to the track with attachEffect(). Level must be >= 0.0 and <= 1.0. |
| */ |
| status_t setAuxEffectSendLevel(float level); |
| void getAuxEffectSendLevel(float* level) const; |
| |
| /* Set sample rate for this track in Hz, mostly used for games' sound effects |
| */ |
| status_t setSampleRate(uint32_t sampleRate); |
| |
| /* Return current sample rate in Hz, or 0 if unknown */ |
| uint32_t getSampleRate() const; |
| |
| /* Enables looping and sets the start and end points of looping. |
| * Only supported for static buffer mode. |
| * |
| * Parameters: |
| * |
| * loopStart: loop start expressed as the number of PCM frames played since AudioTrack start. |
| * loopEnd: loop end expressed as the number of PCM frames played since AudioTrack start. |
| * loopCount: number of loops to execute. Calling setLoop() with loopCount == 0 cancels any |
| * pending or active loop. loopCount = -1 means infinite looping. |
| * |
| * For proper operation the following condition must be respected: |
| * (loopEnd-loopStart) <= framecount() |
| */ |
| status_t setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount); |
| |
| /* 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 -1 or 1. |
| * If the AudioTrack has been opened with no callback function associated, the operation will |
| * fail. |
| * |
| * Parameters: |
| * |
| * marker: marker position expressed in wrapping (overflow) frame units, |
| * like the return value of getPosition(). |
| * |
| * Returned status (from utils/Errors.h) can be: |
| * - NO_ERROR: successful operation |
| * - INVALID_OPERATION: the AudioTrack has no callback installed. |
| */ |
| status_t setMarkerPosition(uint32_t marker); |
| status_t getMarkerPosition(uint32_t *marker) const; |
| |
| /* Sets position update period. Every time the number of frames specified has been played, |
| * a callback with event type EVENT_NEW_POS is called. |
| * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification |
| * callback. |
| * If the AudioTrack has been opened with no callback function associated, the operation will |
| * fail. |
| * Extremely small values may be rounded up to a value the implementation can support. |
| * |
| * Parameters: |
| * |
| * updatePeriod: position update notification period expressed in frames. |
| * |
| * Returned status (from utils/Errors.h) can be: |
| * - NO_ERROR: successful operation |
| * - INVALID_OPERATION: the AudioTrack has no callback installed. |
| */ |
| status_t setPositionUpdatePeriod(uint32_t updatePeriod); |
| status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const; |
| |
| /* Sets playback head position within AudioTrack buffer. The new position is specified |
| * in number of frames. |
| * This method must be called with the AudioTrack in paused or stopped state. |
| * Note that the actual position set is <position> modulo the AudioTrack buffer size in frames. |
| * Therefore using this method makes sense only when playing a "static" audio buffer |
| * as opposed to streaming. |
| * The getPosition() method on the other hand returns the total number of frames played since |
| * playback start. |
| * |
| * Parameters: |
| * |
| * position: New playback head position within AudioTrack buffer. |
| * |
| * Returned status (from utils/Errors.h) can be: |
| * - NO_ERROR: successful operation |
| * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. |
| * - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack |
| * buffer |
| */ |
| status_t setPosition(uint32_t position); |
| |
| /* Return the total number of frames played since playback start. |
| * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz. |
| * It is reset to zero by flush(), reload(), and stop(). |
| */ |
| status_t getPosition(uint32_t *position); |
| |
| /* Forces AudioTrack buffer full condition. When playing a static buffer, this method avoids |
| * rewriting the buffer before restarting playback after a stop. |
| * This method must be called with the AudioTrack in paused or stopped state. |
| * Not allowed in streaming mode. |
| * |
| * Returned status (from utils/Errors.h) can be: |
| * - NO_ERROR: successful operation |
| * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. |
| */ |
| status_t reload(); |
| |
| /* Returns a handle on the audio output used by this AudioTrack. |
| * |
| * Parameters: |
| * none. |
| * |
| * Returned value: |
| * handle on audio hardware output |
| */ |
| audio_io_handle_t getOutput(); |
| |
| /* Returns the unique session ID associated with this track. |
| * |
| * Parameters: |
| * none. |
| * |
| * Returned value: |
| * AudioTrack session ID. |
| */ |
| int getSessionId() const { return mSessionId; } |
| |
| /* Attach track auxiliary output to specified effect. Use effectId = 0 |
| * to detach track from effect. |
| * |
| * Parameters: |
| * |
| * effectId: effectId obtained from AudioEffect::id(). |
| * |
| * Returned status (from utils/Errors.h) can be: |
| * - NO_ERROR: successful operation |
| * - INVALID_OPERATION: the effect is not an auxiliary effect. |
| * - BAD_VALUE: The specified effect ID is invalid |
| */ |
| status_t attachAuxEffect(int effectId); |
| |
| /* Obtains a buffer of "frameCount" frames. The buffer must be |
| * filled entirely, and then released with releaseBuffer(). |
| * If the track is stopped, obtainBuffer() returns |
| * STOPPED instead of NO_ERROR as long as there are buffers available, |
| * at which point NO_MORE_BUFFERS is returned. |
| * Buffers will be returned until the pool |
| * is exhausted, at which point obtainBuffer() will either block |
| * or return WOULD_BLOCK depending on the value of the "blocking" |
| * parameter. |
| * |
| * 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, |
| * -1 causes an (almost) infinite wait time, |
| * 0 non-blocking. |
| * |
| * Buffer fields |
| * On entry: |
| * frameCount number of frames requested |
| * After error return: |
| * frameCount 0 |
| * size 0 |
| * raw undefined |
| * After successful return: |
| * frameCount actual number of frames available, <= number requested |
| * size actual number of bytes available |
| * raw pointer to the buffer |
| */ |
| |
| enum { |
| NO_MORE_BUFFERS = 0x80000001, // same name in AudioFlinger.h, ok to be different value |
| STOPPED = 1 |
| }; |
| |
| status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount); |
| |
| /* Release a filled buffer of "frameCount" frames for AudioFlinger to process. */ |
| void releaseBuffer(Buffer* audioBuffer); |
| |
| /* As a convenience we provide a write() interface to the audio buffer. |
| * This is implemented on top of obtainBuffer/releaseBuffer. For best |
| * performance use callbacks. Returns actual number of bytes written >= 0, |
| * or one of the following negative status codes: |
| * INVALID_OPERATION AudioTrack is configured for shared buffer mode |
| * BAD_VALUE size is invalid |
| * STOPPED AudioTrack was stopped during the write |
| * NO_MORE_BUFFERS when obtainBuffer() returns same |
| * or any other error code returned by IAudioTrack::start() or restoreTrack_l(). |
| * Not supported for static buffer mode. |
| */ |
| ssize_t write(const void* buffer, size_t size); |
| |
| /* |
| * Dumps the state of an audio track. |
| */ |
| status_t dump(int fd, const Vector<String16>& args) const; |
| |
| protected: |
| /* copying audio tracks is not allowed */ |
| AudioTrack(const AudioTrack& other); |
| AudioTrack& operator = (const AudioTrack& other); |
| |
| /* a small internal class to handle the callback */ |
| class AudioTrackThread : public Thread |
| { |
| public: |
| AudioTrackThread(AudioTrack& receiver, bool bCanCallJava = false); |
| |
| // Do not call Thread::requestExitAndWait() without first calling requestExit(). |
| // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough. |
| virtual void requestExit(); |
| |
| void pause(); // suspend thread from execution at next loop boundary |
| void resume(); // allow thread to execute, if not requested to exit |
| |
| private: |
| friend class AudioTrack; |
| virtual bool threadLoop(); |
| AudioTrack& mReceiver; |
| ~AudioTrackThread(); |
| Mutex mMyLock; // Thread::mLock is private |
| Condition mMyCond; // Thread::mThreadExitedCondition is private |
| bool mPaused; // whether thread is currently paused |
| }; |
| |
| // body of AudioTrackThread::threadLoop() |
| bool processAudioBuffer(const sp<AudioTrackThread>& thread); |
| |
| // caller must hold lock on mLock for all _l methods |
| status_t createTrack_l(audio_stream_type_t streamType, |
| uint32_t sampleRate, |
| audio_format_t format, |
| size_t frameCount, |
| audio_output_flags_t flags, |
| const sp<IMemory>& sharedBuffer, |
| audio_io_handle_t output); |
| |
| // can only be called when !mActive |
| void flush_l(); |
| |
| status_t setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount); |
| audio_io_handle_t getOutput_l(); |
| status_t restoreTrack_l(audio_track_cblk_t*& cblk, bool fromStart); |
| bool stopped_l() const { return !mActive; } |
| |
| sp<IAudioTrack> mAudioTrack; |
| sp<IMemory> mCblkMemory; |
| sp<AudioTrackThread> mAudioTrackThread; |
| |
| float mVolume[2]; |
| float mSendLevel; |
| uint32_t mSampleRate; |
| size_t mFrameCount; // corresponds to current IAudioTrack |
| size_t mReqFrameCount; // frame count to request the next time a new |
| // IAudioTrack is needed |
| |
| audio_track_cblk_t* mCblk; // re-load after mLock.unlock() |
| |
| // Starting address of buffers in shared memory. If there is a shared buffer, mBuffers |
| // is the value of pointer() for the shared buffer, otherwise mBuffers points |
| // immediately after the control block. This address is for the mapping within client |
| // address space. AudioFlinger::TrackBase::mBuffer is for the server address space. |
| void* mBuffers; |
| |
| audio_format_t mFormat; // as requested by client, not forced to 16-bit |
| audio_stream_type_t mStreamType; |
| uint32_t mChannelCount; |
| audio_channel_mask_t mChannelMask; |
| |
| // mFrameSize is equal to mFrameSizeAF for non-PCM or 16-bit PCM data. |
| // For 8-bit PCM data, mFrameSizeAF is |
| // twice as large because data is expanded to 16-bit before being stored in buffer. |
| size_t mFrameSize; // app-level frame size |
| size_t mFrameSizeAF; // AudioFlinger frame size |
| |
| status_t mStatus; |
| uint32_t mLatency; |
| |
| bool mActive; // protected by mLock |
| |
| callback_t mCbf; // callback handler for events, or NULL |
| void* mUserData; // for client callback handler |
| |
| // for notification APIs |
| uint32_t mNotificationFramesReq; // requested number of frames between each |
| // notification callback |
| uint32_t mNotificationFramesAct; // actual number of frames between each |
| // notification callback |
| sp<IMemory> mSharedBuffer; |
| int mLoopCount; |
| uint32_t mRemainingFrames; |
| uint32_t mMarkerPosition; // in wrapping (overflow) frame units |
| bool mMarkerReached; |
| uint32_t mNewPosition; // in frames |
| uint32_t mUpdatePeriod; // in frames |
| |
| bool mFlushed; // FIXME will be made obsolete by making flush() synchronous |
| audio_output_flags_t mFlags; |
| int mSessionId; |
| int mAuxEffectId; |
| |
| // When locking both mLock and mCblk->lock, must lock in this order to avoid deadlock: |
| // 1. mLock |
| // 2. mCblk->lock |
| // It is OK to lock only mCblk->lock. |
| mutable Mutex mLock; |
| |
| bool mIsTimed; |
| int mPreviousPriority; // before start() |
| SchedPolicy mPreviousSchedulingGroup; |
| AudioTrackClientProxy* mProxy; |
| }; |
| |
| class TimedAudioTrack : public AudioTrack |
| { |
| public: |
| TimedAudioTrack(); |
| |
| /* allocate a shared memory buffer that can be passed to queueTimedBuffer */ |
| status_t allocateTimedBuffer(size_t size, sp<IMemory>* buffer); |
| |
| /* queue a buffer obtained via allocateTimedBuffer for playback at the |
| given timestamp. PTS units are microseconds on the media time timeline. |
| The media time transform (set with setMediaTimeTransform) set by the |
| audio producer will handle converting from media time to local time |
| (perhaps going through the common time timeline in the case of |
| synchronized multiroom audio case) */ |
| status_t queueTimedBuffer(const sp<IMemory>& buffer, int64_t pts); |
| |
| /* define a transform between media time and either common time or |
| local time */ |
| enum TargetTimeline {LOCAL_TIME, COMMON_TIME}; |
| status_t setMediaTimeTransform(const LinearTransform& xform, |
| TargetTimeline target); |
| }; |
| |
| }; // namespace android |
| |
| #endif // ANDROID_AUDIOTRACK_H |
