diff options
Diffstat (limited to 'include/media/AudioTrack.h')
| -rw-r--r-- | include/media/AudioTrack.h | 113 |
1 files changed, 55 insertions, 58 deletions
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h index 4736369..79db323 100644 --- a/include/media/AudioTrack.h +++ b/include/media/AudioTrack.h @@ -27,7 +27,7 @@ namespace android { // ---------------------------------------------------------------------------- -class audio_track_cblk_t; +struct audio_track_cblk_t; class AudioTrackClientProxy; class StaticAudioTrackClientProxy; @@ -36,11 +36,6 @@ class StaticAudioTrackClientProxy; class AudioTrack : public RefBase { public: - enum channel_index { - MONO = 0, - LEFT = 0, - RIGHT = 1 - }; /* Events used by AudioTrack callback function (callback_t). * Keep in sync with frameworks/base/media/java/android/media/AudioTrack.java NATIVE_EVENT_*. @@ -82,6 +77,7 @@ public: // (currently ignored, but will make the primary field in future) 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 @@ -91,7 +87,7 @@ public: void* raw; short* i16; // signed 16-bit int8_t* i8; // unsigned 8-bit, offset by 0x80 - }; + }; // input: unused, output: pointer to buffer }; /* As a convenience, if a callback is supplied, a handler thread @@ -123,6 +119,8 @@ 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, @@ -156,9 +154,10 @@ public: * 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 (e.g AUDIO_FORMAT_PCM_16_BIT for signed - * 16 bits per sample). - * channelMask: Channel mask. + * 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. + * 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 @@ -180,15 +179,16 @@ public: uint32_t sampleRate, audio_format_t format, audio_channel_mask_t, - int frameCount = 0, + size_t frameCount = 0, audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, callback_t cbf = NULL, void* user = NULL, - int notificationFrames = 0, - int sessionId = 0, + uint32_t notificationFrames = 0, + int sessionId = AUDIO_SESSION_ALLOCATE, transfer_type transferType = TRANSFER_DEFAULT, const audio_offload_info_t *offloadInfo = NULL, - int uid = -1); + int uid = -1, + pid_t pid = -1); /* Creates an audio track and registers it with AudioFlinger. * With this constructor, the track is configured for static buffer mode. @@ -209,11 +209,12 @@ public: audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, callback_t cbf = NULL, void* user = NULL, - int notificationFrames = 0, - int sessionId = 0, + uint32_t notificationFrames = 0, + int sessionId = AUDIO_SESSION_ALLOCATE, transfer_type transferType = TRANSFER_DEFAULT, const audio_offload_info_t *offloadInfo = NULL, - int uid = -1); + int uid = -1, + pid_t pid = -1); /* Terminates the AudioTrack and unregisters it from AudioFlinger. * Also destroys all resources associated with the AudioTrack. @@ -241,17 +242,18 @@ public: uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, - int frameCount = 0, + size_t frameCount = 0, audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, callback_t cbf = NULL, void* user = NULL, - int notificationFrames = 0, + uint32_t notificationFrames = 0, const sp<IMemory>& sharedBuffer = 0, bool threadCanCallJava = false, - int sessionId = 0, + int sessionId = AUDIO_SESSION_ALLOCATE, transfer_type transferType = TRANSFER_DEFAULT, const audio_offload_info_t *offloadInfo = NULL, - int uid = -1); + int uid = -1, + pid_t pid = -1); /* Result of constructing the AudioTrack. This must be checked for successful initialization * before using any AudioTrack API (except for set()), because using @@ -279,7 +281,7 @@ public: size_t frameSize() const { return mFrameSize; } uint32_t channelCount() const { return mChannelCount; } - uint32_t frameCount() const { return mFrameCount; } + size_t frameCount() const { return mFrameCount; } /* Return the static buffer specified in constructor or set(), or 0 for streaming mode */ sp<IMemory> sharedBuffer() const { return mSharedBuffer; } @@ -336,7 +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. @@ -361,7 +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. * @@ -450,9 +452,10 @@ public: * none. * * Returned value: - * handle on audio hardware output + * handle on audio hardware output, or AUDIO_IO_HANDLE_NONE if the + * track needed to be re-created but that failed */ - audio_io_handle_t getOutput(); + audio_io_handle_t getOutput() const; /* Returns the unique session ID associated with this track. * @@ -528,15 +531,6 @@ private: struct timespec *elapsed = NULL, size_t *nonContig = NULL); public: -//EL_FIXME to be reconciled with new obtainBuffer() return codes and control block proxy -// enum { -// NO_MORE_BUFFERS = 0x80000001, // same name in AudioFlinger.h, ok to be different value -// TEAR_DOWN = 0x80000002, -// STOPPED = 1, -// STREAM_END_WAIT, -// STREAM_END -// }; - /* 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); @@ -551,8 +545,11 @@ 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 + * false for the method to return immediately without waiting to try multiple times to write + * the full content of the buffer. */ - ssize_t write(const void* buffer, size_t size); + ssize_t write(const void* buffer, size_t size, bool blocking = true); /* * Dumps the state of an audio track. @@ -566,7 +563,7 @@ public: uint32_t getUnderrunFrames() const; /* Get the flags */ - audio_output_flags_t getFlags() const { return mFlags; } + audio_output_flags_t getFlags() const { AutoMutex _l(mLock); return mFlags; } /* Set parameters - only possible when using direct output */ status_t setParameters(const String8& keyValuePairs); @@ -626,53 +623,50 @@ protected: // NS_INACTIVE inactive so don't run again until re-started // NS_NEVER never again static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3; - nsecs_t processAudioBuffer(const sp<AudioTrackThread>& thread); - status_t processStreamEnd(int32_t waitCount); + nsecs_t processAudioBuffer(); + bool isOffloaded() const; // 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, - size_t epoch); + status_t createTrack_l(size_t epoch); // can only be called when mState != STATE_ACTIVE void flush_l(); void setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount); - audio_io_handle_t getOutput_l(); // FIXME enum is faster than strcmp() for parameter 'from' status_t restoreTrack_l(const char *from); - bool isOffloaded() const + bool isOffloaded_l() const { return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; } - // Next 3 fields may be changed if IAudioTrack is re-created, but always != 0 + // Next 4 fields may be changed if IAudioTrack is re-created, but always != 0 sp<IAudioTrack> mAudioTrack; sp<IMemory> mCblkMemory; audio_track_cblk_t* mCblk; // re-load after mLock.unlock() + audio_io_handle_t mOutput; // returned by AudioSystem::getOutput() sp<AudioTrackThread> mAudioTrackThread; + float mVolume[2]; float mSendLevel; mutable uint32_t mSampleRate; // mutable because getSampleRate() can update it. - size_t mFrameCount; // corresponds to current IAudioTrack - size_t mReqFrameCount; // frame count to request the next time a new - // IAudioTrack is needed - + 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 + // a new IAudioTrack is needed, non-decreasing // constant after constructor or set() 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; + sp<IMemory> mSharedBuffer; transfer_type mTransfer; + audio_offload_info_t mOffloadInfoCopy; + const audio_offload_info_t* mOffloadInfo; // 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. @@ -705,21 +699,25 @@ protected: uint32_t mNotificationFramesAct; // actual number of frames between each // notification callback, // at initial source sample rate - bool mRefreshRemaining; // processAudioBuffer() should refresh next 2 + bool mRefreshRemaining; // processAudioBuffer() should refresh + // mRemainingFrames and mRetryOnPartialBuffer // 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 - 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 uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS audio_output_flags_t mFlags; + // const after set(), except for bits AUDIO_OUTPUT_FLAG_FAST and AUDIO_OUTPUT_FLAG_OFFLOAD. + // mLock must be held to read or write those bits reliably. + int mSessionId; int mAuxEffectId; @@ -739,7 +737,6 @@ protected: sp<AudioTrackClientProxy> mProxy; // primary owner of the memory bool mInUnderrun; // whether track is currently in underrun state - String8 mName; // server's name for this IAudioTrack uint32_t mPausedPosition; private: @@ -754,8 +751,8 @@ private: sp<DeathNotifier> mDeathNotifier; uint32_t mSequence; // incremented for each new IAudioTrack attempt - audio_io_handle_t mOutput; // cached output io handle int mClientUid; + pid_t mClientPid; }; class TimedAudioTrack : public AudioTrack |