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author | Glenn Kasten <gkasten@google.com> | 2012-11-01 15:41:48 -0700 |
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committer | Glenn Kasten <gkasten@google.com> | 2012-11-02 10:56:20 -0700 |
commit | c4bae46d8bc833f200c2d460df73d42130efb5bc (patch) | |
tree | 2c800085866d236f1c0f599e12249826ed4e8016 /include | |
parent | 3d07702e3b95579370aa74d40b56c63685cbb518 (diff) | |
download | frameworks_av-c4bae46d8bc833f200c2d460df73d42130efb5bc.zip frameworks_av-c4bae46d8bc833f200c2d460df73d42130efb5bc.tar.gz frameworks_av-c4bae46d8bc833f200c2d460df73d42130efb5bc.tar.bz2 |
AudioRecord comments
Change-Id: Ibec910608948d778dc655d900255a80384e9b06f
Diffstat (limited to 'include')
-rw-r--r-- | include/media/AudioRecord.h | 75 |
1 files changed, 45 insertions, 30 deletions
diff --git a/include/media/AudioRecord.h b/include/media/AudioRecord.h index 156c592..f9f6e8d 100644 --- a/include/media/AudioRecord.h +++ b/include/media/AudioRecord.h @@ -43,15 +43,15 @@ public: */ enum event_type { EVENT_MORE_DATA = 0, // Request to read more data from PCM buffer. - EVENT_OVERRUN = 1, // PCM buffer overrun occured. + EVENT_OVERRUN = 1, // PCM 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 // (See setPositionUpdatePeriod()). }; - /* Create Buffer on the stack and pass it to obtainBuffer() - * and releaseBuffer(). + /* Client should declare Buffer on the stack and pass address to obtainBuffer() + * and releaseBuffer(). See also callback_t for EVENT_MORE_DATA. */ class Buffer @@ -63,26 +63,30 @@ public: uint32_t flags; int channelCount; audio_format_t format; - size_t frameCount; + + 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 + size_t size; // total size in bytes == frameCount * frameSize union { void* raw; - short* i16; - int8_t* i8; + 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 ready or an overrun condition occurs. + * invokes the callback when a new buffer becomes ready or various conditions occur. * Parameters: * * event: type of event notified (see enum AudioRecord::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 AudioRecord::Buffer struct. The callback must not read - * more bytes than indicated by 'size' field and update 'size' if less bytes are - * read. + * more bytes than indicated by 'size' field and update 'size' if fewer bytes are + * consumed. * - EVENT_OVERRUN: unused. * - EVENT_MARKER: pointer to const uint32_t containing the marker position in frames. * - EVENT_NEW_POS: pointer to const uint32_t containing the new position in frames. @@ -108,7 +112,7 @@ public: */ AudioRecord(); - /* Creates an AudioRecord track and registers it with AudioFlinger. + /* Creates an AudioRecord object and registers it with AudioFlinger. * Once created, the track needs to be started before it can be used. * Unspecified values are set to the audio hardware's current * values. @@ -120,10 +124,13 @@ public: * format: Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed * 16 bits per sample). * channelMask: Channel mask. - * frameCount: Total size of track PCM buffer in frames. This defines the - * latency of the track. + * 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 AudioRecord could + * 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 provide new PCM data. + * to consume new PCM data. * 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. @@ -154,7 +161,7 @@ public: * - BAD_VALUE: invalid parameter (channels, format, sampleRate...) * - NO_INIT: audio server or audio hardware not initialized * - PERMISSION_DENIED: recording is not allowed for the requesting process - * */ + */ status_t set(audio_source_t inputSource = AUDIO_SOURCE_DEFAULT, uint32_t sampleRate = 0, audio_format_t format = AUDIO_FORMAT_DEFAULT, @@ -168,14 +175,14 @@ public: /* Result of constructing the AudioRecord. This must be checked - * before using any AudioRecord API (except for set()), using + * before using any AudioRecord API (except for set()), because using * an uninitialized AudioRecord produces undefined results. * See set() method above for possible return codes. */ status_t initCheck() const; - /* Returns this track's latency in milliseconds. - * This includes the latency due to AudioRecord buffer size + /* Returns this track's estimated latency in milliseconds. + * This includes the latency due to AudioRecord buffer size, * and audio hardware driver. */ uint32_t latency() const; @@ -191,7 +198,7 @@ public: /* After it's created the track is not active. Call start() to * make it active. If set, the callback will start being called. - * if event is not AudioSystem::SYNC_EVENT_NONE, the capture start will be delayed until + * If event is not AudioSystem::SYNC_EVENT_NONE, the capture start will be delayed until * the specified event occurs on the specified trigger session. */ status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, @@ -199,12 +206,12 @@ public: /* Stop a track. If set, 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. + * and will drain buffers until the pool is exhausted. */ void stop(); bool stopped() const; - /* get sample rate for this record track + /* Get sample rate for this record track in Hz. */ uint32_t getSampleRate() const; @@ -258,7 +265,7 @@ public: */ status_t getPosition(uint32_t *position) const; - /* returns a handle on the audio input used by this AudioRecord. + /* Returns a handle on the audio input used by this AudioRecord. * * Parameters: * none. @@ -268,7 +275,7 @@ public: */ audio_io_handle_t getInput() const; - /* returns the audio session ID associated with this AudioRecord. + /* Returns the audio session ID associated with this AudioRecord. * * Parameters: * none. @@ -278,22 +285,30 @@ public: */ int getSessionId() const; - /* obtains a buffer of "frameCount" frames. The buffer must be - * filled entirely. If the track is stopped, obtainBuffer() returns + /* Obtains a buffer of "frameCount" frames. The buffer must be + * drained 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 (buffercount()) + * 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. + * + * Interpretation of waitCount: + * +n limits wait time to n * WAIT_PERIOD_MS, + * -1 causes an (almost) infinite wait time, + * 0 non-blocking. */ enum { - NO_MORE_BUFFERS = 0x80000001, + 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 an emptied buffer of "frameCount" frames for AudioFlinger to re-fill. */ void releaseBuffer(Buffer* audioBuffer); @@ -302,16 +317,16 @@ public: */ ssize_t read(void* buffer, size_t size); - /* Return the amount of input frames lost in the audio driver since the last call of this + /* Return the number of input frames lost in the audio driver since the last call of this * function. Audio driver is expected to reset the value to 0 and restart counting upon * 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. - * Unit: the number of input audio frames + * Units: the number of input audio frames. */ unsigned int getInputFramesLost() const; private: - /* copying audio tracks is not allowed */ + /* copying audio record objects is not allowed */ AudioRecord(const AudioRecord& other); AudioRecord& operator = (const AudioRecord& other); @@ -355,7 +370,7 @@ private: bool mActive; // protected by mLock // for client callback handler - callback_t mCbf; + callback_t mCbf; // callback handler for events, or NULL void* mUserData; // for notification APIs |