Merge "Improve documentation for AudioTrack buffer sizes" into nyc-dev

This commit is contained in:
Glenn Kasten
2016-05-02 21:28:36 +00:00
committed by Android (Google) Code Review

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@@ -233,9 +233,9 @@ public class AudioTrack extends PlayerBase
*/
private final Object mPlayStateLock = new Object();
/**
* Sizes of the native audio buffer.
* Sizes of the audio buffer.
* These values are set during construction and can be stale.
* To obtain the current native audio buffer frame count use {@link #getBufferSizeInFrames()}.
* To obtain the current audio buffer frame count use {@link #getBufferSizeInFrames()}.
*/
private int mNativeBufferSizeInBytes = 0;
private int mNativeBufferSizeInFrames = 0;
@@ -342,20 +342,22 @@ public class AudioTrack extends PlayerBase
* {@link AudioFormat#ENCODING_PCM_8BIT},
* and {@link AudioFormat#ENCODING_PCM_FLOAT}.
* @param bufferSizeInBytes the total size (in bytes) of the internal buffer where audio data is
* read from for playback. This should be a multiple of the frame size in bytes.
* read from for playback. This should be a nonzero multiple of the frame size in bytes.
* <p> If the track's creation mode is {@link #MODE_STATIC},
* this is the maximum length sample, or audio clip, that can be played by this instance.
* <p> If the track's creation mode is {@link #MODE_STREAM},
* this should be the desired buffer size
* for the <code>AudioTrack</code> to satisfy the application's
* natural latency requirements.
* latency requirements.
* If <code>bufferSizeInBytes</code> is less than the
* minimum buffer size for the output sink, it is automatically increased to the minimum
* minimum buffer size for the output sink, it is increased to the minimum
* buffer size.
* The method {@link #getBufferSizeInFrames()} returns the
* actual size in frames of the native buffer created, which
* determines the frequency to write
* actual size in frames of the buffer created, which
* determines the minimum frequency to write
* to the streaming <code>AudioTrack</code> to avoid underrun.
* See {@link #getMinBufferSize(int, int, int)} to determine the estimated minimum buffer size
* for an AudioTrack instance in streaming mode.
* @param mode streaming or static buffer. See {@link #MODE_STATIC} and {@link #MODE_STREAM}
* @throws java.lang.IllegalArgumentException
*/
@@ -392,13 +394,24 @@ public class AudioTrack extends PlayerBase
* See {@link AudioFormat#ENCODING_PCM_16BIT} and
* {@link AudioFormat#ENCODING_PCM_8BIT},
* and {@link AudioFormat#ENCODING_PCM_FLOAT}.
* @param bufferSizeInBytes the total size (in bytes) of the buffer where audio data is read
* from for playback. If using the AudioTrack in streaming mode, you can write data into
* this buffer in smaller chunks than this size. If using the AudioTrack in static mode,
* this is the maximum size of the sound that will be played for this instance.
* See {@link #getMinBufferSize(int, int, int)} to determine the minimum required buffer size
* for the successful creation of an AudioTrack instance in streaming mode. Using values
* smaller than getMinBufferSize() will result in an initialization failure.
* @param bufferSizeInBytes the total size (in bytes) of the internal buffer where audio data is
* read from for playback. This should be a nonzero multiple of the frame size in bytes.
* <p> If the track's creation mode is {@link #MODE_STATIC},
* this is the maximum length sample, or audio clip, that can be played by this instance.
* <p> If the track's creation mode is {@link #MODE_STREAM},
* this should be the desired buffer size
* for the <code>AudioTrack</code> to satisfy the application's
* latency requirements.
* If <code>bufferSizeInBytes</code> is less than the
* minimum buffer size for the output sink, it is increased to the minimum
* buffer size.
* The method {@link #getBufferSizeInFrames()} returns the
* actual size in frames of the buffer created, which
* determines the minimum frequency to write
* to the streaming <code>AudioTrack</code> to avoid underrun.
* You can write data into this buffer in smaller chunks than this size.
* See {@link #getMinBufferSize(int, int, int)} to determine the estimated minimum buffer size
* for an AudioTrack instance in streaming mode.
* @param mode streaming or static buffer. See {@link #MODE_STATIC} and {@link #MODE_STREAM}
* @param sessionId Id of audio session the AudioTrack must be attached to
* @throws java.lang.IllegalArgumentException
@@ -425,13 +438,23 @@ public class AudioTrack extends PlayerBase
* @param format a non-null {@link AudioFormat} instance describing the format of the data
* that will be played through this AudioTrack. See {@link AudioFormat.Builder} for
* configuring the audio format parameters such as encoding, channel mask and sample rate.
* @param bufferSizeInBytes the total size (in bytes) of the buffer where audio data is read
* from for playback. If using the AudioTrack in streaming mode, you can write data into
* this buffer in smaller chunks than this size. If using the AudioTrack in static mode,
* this is the maximum size of the sound that will be played for this instance.
* See {@link #getMinBufferSize(int, int, int)} to determine the minimum required buffer size
* for the successful creation of an AudioTrack instance in streaming mode. Using values
* smaller than getMinBufferSize() will result in an initialization failure.
* @param bufferSizeInBytes the total size (in bytes) of the internal buffer where audio data is
* read from for playback. This should be a nonzero multiple of the frame size in bytes.
* <p> If the track's creation mode is {@link #MODE_STATIC},
* this is the maximum length sample, or audio clip, that can be played by this instance.
* <p> If the track's creation mode is {@link #MODE_STREAM},
* this should be the desired buffer size
* for the <code>AudioTrack</code> to satisfy the application's
* latency requirements.
* If <code>bufferSizeInBytes</code> is less than the
* minimum buffer size for the output sink, it is increased to the minimum
* buffer size.
* The method {@link #getBufferSizeInFrames()} returns the
* actual size in frames of the buffer created, which
* determines the minimum frequency to write
* to the streaming <code>AudioTrack</code> to avoid underrun.
* See {@link #getMinBufferSize(int, int, int)} to determine the estimated minimum buffer size
* for an AudioTrack instance in streaming mode.
* @param mode streaming or static buffer. See {@link #MODE_STATIC} and {@link #MODE_STREAM}.
* @param sessionId ID of audio session the AudioTrack must be attached to, or
* {@link AudioManager#AUDIO_SESSION_ID_GENERATE} if the session isn't known at construction
@@ -662,9 +685,8 @@ public class AudioTrack extends PlayerBase
* If using the {@link AudioTrack} in streaming mode
* (see {@link AudioTrack#MODE_STREAM}, you can write data into this buffer in smaller
* chunks than this size. See {@link #getMinBufferSize(int, int, int)} to determine
* the minimum required buffer size for the successful creation of an AudioTrack instance
* in streaming mode. Using values smaller than <code>getMinBufferSize()</code> will result
* in an exception when trying to build the <code>AudioTrack</code>.
* the estimated minimum buffer size for the creation of an AudioTrack instance
* in streaming mode.
* <br>If using the <code>AudioTrack</code> in static mode (see
* {@link AudioTrack#MODE_STATIC}), this is the maximum size of the sound that will be
* played by this instance.
@@ -1106,7 +1128,7 @@ public class AudioTrack extends PlayerBase
* size and capacity may enlarge to accommodate.
* <p> If the <code>AudioTrack</code> encoding indicates compressed data,
* e.g. {@link AudioFormat#ENCODING_AC3}, then the frame count returned is
* the size of the native <code>AudioTrack</code> buffer in bytes.
* the size of the <code>AudioTrack</code> buffer in bytes.
* <p> See also {@link AudioManager#getProperty(String)} for key
* {@link AudioManager#PROPERTY_OUTPUT_FRAMES_PER_BUFFER}.
* @return current size in frames of the <code>AudioTrack</code> buffer.
@@ -1120,7 +1142,7 @@ public class AudioTrack extends PlayerBase
* Limits the effective size of the <code>AudioTrack</code> buffer
* that the application writes to.
* <p> A write to this AudioTrack will not fill the buffer beyond this limit.
* If a blocking write is used then the write will block until the the data
* If a blocking write is used then the write will block until the data
* can fit within this limit.
* <p>Changing this limit modifies the latency associated with
* the buffer for this track. A smaller size will give lower latency
@@ -1134,7 +1156,7 @@ public class AudioTrack extends PlayerBase
* <p>This method is only supported for PCM audio.
* It is not supported for compressed audio tracks.
*
* @param bufferSizeInFrames requested buffer size
* @param bufferSizeInFrames requested buffer size in frames
* @return the actual buffer size in frames or an error code,
* {@link #ERROR_BAD_VALUE}, {@link #ERROR_INVALID_OPERATION}
* @throws IllegalStateException if track is not initialized.
@@ -1150,19 +1172,19 @@ public class AudioTrack extends PlayerBase
}
/**
* Returns the maximum size of the native <code>AudioTrack</code> buffer.
* Returns the maximum size of the <code>AudioTrack</code> buffer in frames.
* <p> If the track's creation mode is {@link #MODE_STATIC},
* it is equal to the specified bufferSizeInBytes on construction, converted to frame units.
* A static track's native frame count will not change.
* A static track's frame count will not change.
* <p> If the track's creation mode is {@link #MODE_STREAM},
* it is greater than or equal to the specified bufferSizeInBytes converted to frame units.
* For streaming tracks, this value may be rounded up to a larger value if needed by
* the target output sink, and
* if the track is subsequently routed to a different output sink, the native
* if the track is subsequently routed to a different output sink, the
* frame count may enlarge to accommodate.
* <p> If the <code>AudioTrack</code> encoding indicates compressed data,
* e.g. {@link AudioFormat#ENCODING_AC3}, then the frame count returned is
* the size of the native <code>AudioTrack</code> buffer in bytes.
* the size of the <code>AudioTrack</code> buffer in bytes.
* <p> See also {@link AudioManager#getProperty(String)} for key
* {@link AudioManager#PROPERTY_OUTPUT_FRAMES_PER_BUFFER}.
* @return maximum size in frames of the <code>AudioTrack</code> buffer.
@@ -1252,8 +1274,10 @@ public class AudioTrack extends PlayerBase
}
/**
* Returns the minimum buffer size required for the successful creation of an AudioTrack
* object to be created in the {@link #MODE_STREAM} mode. Note that this size doesn't
* Returns the estimated minimum buffer size required for an AudioTrack
* object to be created in the {@link #MODE_STREAM} mode.
* The size is an estimate because it does not consider either the route or the sink,
* since neither is known yet. Note that this size doesn't
* guarantee a smooth playback under load, and higher values should be chosen according to
* the expected frequency at which the buffer will be refilled with additional data to play.
* For example, if you intend to dynamically set the source sample rate of an AudioTrack