diff options
Diffstat (limited to 'include')
| -rw-r--r-- | include/media/AudioBufferProvider.h | 11 | ||||
| -rw-r--r-- | include/media/AudioRecord.h | 42 | ||||
| -rw-r--r-- | include/media/AudioSystem.h | 38 | ||||
| -rw-r--r-- | include/media/AudioTrack.h | 38 | ||||
| -rw-r--r-- | include/media/IAudioFlinger.h | 9 | ||||
| -rw-r--r-- | include/media/stagefright/MetaData.h | 3 |
6 files changed, 99 insertions, 42 deletions
diff --git a/include/media/AudioBufferProvider.h b/include/media/AudioBufferProvider.h index ef392f0..7be449c 100644 --- a/include/media/AudioBufferProvider.h +++ b/include/media/AudioBufferProvider.h @@ -61,6 +61,17 @@ public: // buffer->frameCount 0 virtual status_t getNextBuffer(Buffer* buffer, int64_t pts = kInvalidPTS) = 0; + // Release (a portion of) the buffer previously obtained by getNextBuffer(). + // It is permissible to call releaseBuffer() multiple times per getNextBuffer(). + // On entry: + // buffer->frameCount number of frames to release, must be <= number of frames + // obtained but not yet released + // buffer->raw unused + // On return: + // buffer->frameCount 0; implementation MUST set to zero + // buffer->raw undefined; implementation is PERMITTED to set to any value, + // so if caller needs to continue using this buffer it must + // keep track of the pointer itself virtual void releaseBuffer(Buffer* buffer) = 0; }; diff --git a/include/media/AudioRecord.h b/include/media/AudioRecord.h index fb47448..45134c4 100644 --- a/include/media/AudioRecord.h +++ b/include/media/AudioRecord.h @@ -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. @@ -155,7 +162,7 @@ public: callback_t cbf = NULL, void* user = NULL, int notificationFrames = 0, - int sessionId = 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: * @@ -188,11 +196,11 @@ public: void* user = NULL, int 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' @@ -452,7 +464,7 @@ private: // 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; @@ -476,7 +488,7 @@ 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() diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h index b96b8a1..5abab8a 100644 --- a/include/media/AudioSystem.h +++ b/include/media/AudioSystem.h @@ -67,20 +67,24 @@ public: // returns true in *state if tracks are active on the specified stream or have been active // in the past inPastMs milliseconds - static status_t isStreamActive(audio_stream_type_t stream, bool *state, uint32_t inPastMs = 0); + static status_t isStreamActive(audio_stream_type_t stream, bool *state, uint32_t inPastMs); // returns true in *state if tracks are active for what qualifies as remote playback // on the specified stream or have been active in the past inPastMs milliseconds. Remote // playback isn't mutually exclusive with local playback. static status_t isStreamActiveRemotely(audio_stream_type_t stream, bool *state, - uint32_t inPastMs = 0); + uint32_t inPastMs); // returns true in *state if a recorder is currently recording with the specified source static status_t isSourceActive(audio_source_t source, bool *state); // set/get audio hardware parameters. The function accepts a list of parameters // key value pairs in the form: key1=value1;key2=value2;... // Some keys are reserved for standard parameters (See AudioParameter class). + // The versions with audio_io_handle_t are intended for internal media framework use only. static status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); static String8 getParameters(audio_io_handle_t ioHandle, const String8& keys); + // The versions without audio_io_handle_t are intended for JNI. + static status_t setParameters(const String8& keyValuePairs); + static String8 getParameters(const String8& keys); static void setErrorCallback(audio_error_callback cb); @@ -90,12 +94,14 @@ public: static float linearToLog(int volume); static int logToLinear(float volume); + // Returned samplingRate and frameCount output values are guaranteed + // to be non-zero if status == NO_ERROR static status_t getOutputSamplingRate(uint32_t* samplingRate, - audio_stream_type_t stream = AUDIO_STREAM_DEFAULT); + audio_stream_type_t stream); static status_t getOutputFrameCount(size_t* frameCount, - audio_stream_type_t stream = AUDIO_STREAM_DEFAULT); + audio_stream_type_t stream); static status_t getOutputLatency(uint32_t* latency, - audio_stream_type_t stream = AUDIO_STREAM_DEFAULT); + audio_stream_type_t stream); static status_t getSamplingRate(audio_io_handle_t output, audio_stream_type_t streamType, uint32_t* samplingRate); @@ -132,7 +138,7 @@ public: audio_stream_type_t stream = AUDIO_STREAM_DEFAULT); // return the number of input frames lost by HAL implementation, or 0 if the handle is invalid - static size_t getInputFramesLost(audio_io_handle_t ioHandle); + static uint32_t getInputFramesLost(audio_io_handle_t ioHandle); static int newAudioSessionId(); static void acquireAudioSessionId(int audioSession); @@ -194,24 +200,32 @@ public: static status_t setPhoneState(audio_mode_t state); static status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config); static audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage); + + // Client must successfully hand off the handle reference to AudioFlinger via createTrack(), + // or release it with releaseOutput(). static audio_io_handle_t getOutput(audio_stream_type_t stream, uint32_t samplingRate = 0, audio_format_t format = AUDIO_FORMAT_DEFAULT, audio_channel_mask_t channelMask = AUDIO_CHANNEL_OUT_STEREO, audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, const audio_offload_info_t *offloadInfo = NULL); + static status_t startOutput(audio_io_handle_t output, audio_stream_type_t stream, - int session = 0); + int session); static status_t stopOutput(audio_io_handle_t output, audio_stream_type_t stream, - int session = 0); + int session); static void releaseOutput(audio_io_handle_t output); + + // Client must successfully hand off the handle reference to AudioFlinger via openRecord(), + // or release it with releaseInput(). static audio_io_handle_t getInput(audio_source_t inputSource, - uint32_t samplingRate = 0, - audio_format_t format = AUDIO_FORMAT_DEFAULT, - audio_channel_mask_t channelMask = AUDIO_CHANNEL_IN_MONO, - int sessionId = 0); + uint32_t samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + int sessionId); + static status_t startInput(audio_io_handle_t input); static status_t stopInput(audio_io_handle_t input); static void releaseInput(audio_io_handle_t input); diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h index e163f88..644e55c 100644 --- a/include/media/AudioTrack.h +++ b/include/media/AudioTrack.h @@ -123,6 +123,8 @@ 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, @@ -158,7 +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 @@ -185,7 +187,7 @@ public: callback_t cbf = NULL, void* user = NULL, int notificationFrames = 0, - int sessionId = 0, + int sessionId = AUDIO_SESSION_ALLOCATE, transfer_type transferType = TRANSFER_DEFAULT, const audio_offload_info_t *offloadInfo = NULL, int uid = -1); @@ -210,7 +212,7 @@ public: callback_t cbf = NULL, void* user = NULL, int notificationFrames = 0, - int sessionId = 0, + int sessionId = AUDIO_SESSION_ALLOCATE, transfer_type transferType = TRANSFER_DEFAULT, const audio_offload_info_t *offloadInfo = NULL, int uid = -1); @@ -248,7 +250,7 @@ public: int 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); @@ -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. * @@ -566,7 +568,7 @@ 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); @@ -626,9 +628,9 @@ 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 @@ -650,7 +652,7 @@ protected: // 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 @@ -659,19 +661,24 @@ protected: audio_track_cblk_t* mCblk; // re-load after mLock.unlock() 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. @@ -712,14 +719,17 @@ protected: 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; diff --git a/include/media/IAudioFlinger.h b/include/media/IAudioFlinger.h index 85298ec..4bd111a 100644 --- a/include/media/IAudioFlinger.h +++ b/include/media/IAudioFlinger.h @@ -67,6 +67,9 @@ public: size_t *pFrameCount, track_flags_t *flags, const sp<IMemory>& sharedBuffer, + // On successful return, AudioFlinger takes over the handle + // reference and will release it when the track is destroyed. + // However on failure, the client is responsible for release. audio_io_handle_t output, pid_t tid, // -1 means unused, otherwise must be valid non-0 int *sessionId, @@ -78,6 +81,9 @@ public: status_t *status) = 0; virtual sp<IAudioRecord> openRecord( + // On successful return, AudioFlinger takes over the handle + // reference and will release it when the track is destroyed. + // However on failure, the client is responsible for release. audio_io_handle_t input, uint32_t sampleRate, audio_format_t format, @@ -170,7 +176,7 @@ public: virtual status_t getRenderPosition(size_t *halFrames, size_t *dspFrames, audio_io_handle_t output) const = 0; - virtual size_t getInputFramesLost(audio_io_handle_t ioHandle) const = 0; + virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const = 0; virtual int newAudioSessionId() = 0; @@ -188,6 +194,7 @@ public: effect_descriptor_t *pDesc, const sp<IEffectClient>& client, int32_t priority, + // AudioFlinger doesn't take over handle reference from client audio_io_handle_t output, int sessionId, status_t *status, diff --git a/include/media/stagefright/MetaData.h b/include/media/stagefright/MetaData.h index de3fc36..db8216b 100644 --- a/include/media/stagefright/MetaData.h +++ b/include/media/stagefright/MetaData.h @@ -134,6 +134,7 @@ enum { kKeyRequiresSecureBuffers = 'secu', // bool (int32_t) kKeyIsADTS = 'adts', // bool (int32_t) + kKeyAACAOT = 'aaot', // int32_t // If a MediaBuffer's data represents (at least partially) encrypted // data, the following fields aid in decryption. @@ -214,6 +215,8 @@ public: bool findData(uint32_t key, uint32_t *type, const void **data, size_t *size) const; + bool hasData(uint32_t key) const; + void dumpToLog() const; protected: |