Merge "AudioRecord comments"
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 @@
*/
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 @@
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 @@
*/
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 @@
* 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 @@
* - 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 @@
/* 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 @@
/* 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 @@
/* 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 @@
*/
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 @@
*/
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 @@
*/
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 @@
*/
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 @@
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