diff options
Diffstat (limited to 'include/media/AudioRecord.h')
| -rw-r--r-- | include/media/AudioRecord.h | 66 |
1 files changed, 41 insertions, 25 deletions
diff --git a/include/media/AudioRecord.h b/include/media/AudioRecord.h index 052064d..b3c44a8 100644 --- a/include/media/AudioRecord.h +++ b/include/media/AudioRecord.h @@ -26,7 +26,7 @@ namespace android { // ---------------------------------------------------------------------------- -class audio_track_cblk_t; +struct audio_track_cblk_t; class AudioRecordClientProxy; // ---------------------------------------------------------------------------- @@ -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 @@ -60,9 +64,10 @@ public: size_t frameCount; // number of sample frames corresponding to size; // on input it is the number of frames available, // on output is the number of frames actually drained - // (currently ignored, but will make the primary field in future) + // (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. @@ -151,11 +158,11 @@ public: uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, - int frameCount = 0, + size_t frameCount = 0, 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, audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE); @@ -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: * @@ -183,16 +191,16 @@ public: uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, - int frameCount = 0, + size_t frameCount = 0, callback_t cbf = NULL, void* user = NULL, - int notificationFrames = 0, + uint32_t notificationFrames = 0, bool threadCanCallJava = false, - int sessionId = 0, + int sessionId = AUDIO_SESSION_ALLOCATE, 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. * @@ -378,8 +386,10 @@ public: * returning the current value by this function call. Such loss typically occurs when the * user space process is blocked longer than the capacity of audio driver buffers. * Units: the number of input audio frames. + * FIXME The side-effect of resetting the counter may be incompatible with multi-client. + * Consider making it more like AudioTrack::getUnderrunFrames which doesn't have side effects. */ - unsigned int getInputFramesLost() const; + uint32_t getInputFramesLost() const; private: /* copying audio record objects is not allowed */ @@ -412,6 +422,7 @@ private: 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 }; // body of AudioRecordThread::threadLoop() @@ -422,9 +433,10 @@ private: // 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<AudioRecordThread>& thread); + 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' @@ -446,12 +458,13 @@ private: // notification callback uint32_t mNotificationFramesAct; // actual number of frames between each // notification callback - 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() - int mObservedSequence; // last observed value of mSequence + uint32_t mObservedSequence; // last observed value of mSequence uint32_t mMarkerPosition; // in wrapping (overflow) frame units bool mMarkerReached; @@ -460,9 +473,13 @@ private: status_t mStatus; + size_t mFrameCount; // corresponds to current IAudioRecord, value is + // reported back by AudioFlinger to the client + size_t mReqFrameCount; // frame count to request the first or next time + // a new IAudioRecord is needed, non-decreasing + // constant after constructor or set() uint32_t mSampleRate; - size_t mFrameCount; audio_format_t mFormat; uint32_t mChannelCount; size_t mFrameSize; // app-level frame size == AudioFlinger frame size @@ -473,12 +490,11 @@ private: int mSessionId; transfer_type mTransfer; - audio_io_handle_t mInput; // returned by AudioSystem::getInput() - - // may be changed if IAudioRecord object is re-created + // Next 4 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() + audio_io_handle_t mInput; // returned by AudioSystem::getInput() int mPreviousPriority; // before start() SchedPolicy mPreviousSchedulingGroup; |