+ * Before starting playback, specify the input audio format by calling + * {@link #configure(String, int, int, int, int, int[], int, int)}. + *
+ * Call {@link #handleBuffer(ByteBuffer, long)} to write data, and {@link #handleDiscontinuity()} + * when the data being fed is discontinuous. Call {@link #play()} to start playing the written data. + *
+ * Call {@link #configure(String, int, int, int, int, int[], int, int)} whenever the input format + * changes. The sink will be reinitialized on the next call to + * {@link #handleBuffer(ByteBuffer, long)}. + *
+ * Call {@link #reset()} to prepare the sink to receive audio data from a new playback position. + *
+ * Call {@link #playToEndOfStream()} repeatedly to play out all data when no more input buffers will + * be provided via {@link #handleBuffer(ByteBuffer, long)} until the next {@link #reset()}. Call + * {@link #release()} when the instance is no longer required. + *
+ * The implementation may be backed by a platform {@link AudioTrack}. In this case, + * {@link #setAudioSessionId(int)}, {@link #setAudioAttributes(AudioAttributes)}, + * {@link #enableTunnelingV21(int)} and/or {@link #disableTunneling()} may be called before writing + * data to the sink. These methods may also be called after writing data to the sink, in which case + * it will be reinitialized as required. For implementations that are not based on platform + * {@link AudioTrack}s, calling methods relating to audio sessions, audio attributes, and tunneling + * may have no effect. + */ +public interface AudioSink { + + /** + * Listener for audio sink events. + */ + interface Listener { + + /** + * Called if the audio sink has started rendering audio to a new platform audio session. + * + * @param audioSessionId The newly generated audio session's identifier. + */ + void onAudioSessionId(int audioSessionId); + + /** + * Called when the audio sink handles a buffer whose timestamp is discontinuous with the last + * buffer handled since it was reset. + */ + void onPositionDiscontinuity(); + + /** + * Called when the audio sink runs out of data. + *
+ * An audio sink implementation may never call this method (for example, if audio data is + * consumed in batches rather than based on the sink's own clock). + * + * @param bufferSize The size of the sink's buffer, in bytes. + * @param bufferSizeMs The size of the sink's buffer, in milliseconds, if it is configured for + * PCM output. {@link C#TIME_UNSET} if it is configured for passthrough output, as the + * buffered media can have a variable bitrate so the duration may be unknown. + * @param elapsedSinceLastFeedMs The time since the sink was last fed data, in milliseconds. + */ + void onUnderrun(int bufferSize, long bufferSizeMs, long elapsedSinceLastFeedMs); + + } + + /** + * Thrown when a failure occurs configuring the sink. + */ + final class ConfigurationException extends Exception { + + /** + * Creates a new configuration exception with the specified {@code cause} and no message. + */ + public ConfigurationException(Throwable cause) { + super(cause); + } + + /** + * Creates a new configuration exception with the specified {@code message} and no cause. + */ + public ConfigurationException(String message) { + super(message); + } + + } + + /** + * Thrown when a failure occurs initializing the sink. + */ + final class InitializationException extends Exception { + + /** + * The underlying {@link AudioTrack}'s state, if applicable. + */ + public final int audioTrackState; + + /** + * @param audioTrackState The underlying {@link AudioTrack}'s state, if applicable. + * @param sampleRate The requested sample rate in Hz. + * @param channelConfig The requested channel configuration. + * @param bufferSize The requested buffer size in bytes. + */ + public InitializationException(int audioTrackState, int sampleRate, int channelConfig, + int bufferSize) { + super("AudioTrack init failed: " + audioTrackState + ", Config(" + sampleRate + ", " + + channelConfig + ", " + bufferSize + ")"); + this.audioTrackState = audioTrackState; + } + + } + + /** + * Thrown when a failure occurs writing to the sink. + */ + final class WriteException extends Exception { + + /** + * The error value returned from the sink implementation. If the sink writes to a platform + * {@link AudioTrack}, this will be the error value returned from + * {@link AudioTrack#write(byte[], int, int)} or {@link AudioTrack#write(ByteBuffer, int, int)}. + * Otherwise, the meaning of the error code depends on the sink implementation. + */ + public final int errorCode; + + /** + * @param errorCode The error value returned from the sink implementation. + */ + public WriteException(int errorCode) { + super("AudioTrack write failed: " + errorCode); + this.errorCode = errorCode; + } + + } + + /** + * Returned by {@link #getCurrentPositionUs(boolean)} when the position is not set. + */ + long CURRENT_POSITION_NOT_SET = Long.MIN_VALUE; + + /** + * Sets the listener for sink events, which should be the audio renderer. + * + * @param listener The listener for sink events, which should be the audio renderer. + */ + void setListener(Listener listener); + + /** + * Returns whether it's possible to play audio in the specified format using encoded audio + * passthrough. + * + * @param mimeType The format mime type. + * @return Whether it's possible to play audio in the format using encoded audio passthrough. + */ + boolean isPassthroughSupported(String mimeType); + + /** + * Returns the playback position in the stream starting at zero, in microseconds, or + * {@link #CURRENT_POSITION_NOT_SET} if it is not yet available. + * + * @param sourceEnded Specify {@code true} if no more input buffers will be provided. + * @return The playback position relative to the start of playback, in microseconds. + */ + long getCurrentPositionUs(boolean sourceEnded); + + /** + * Configures (or reconfigures) the sink. + * + * @param mimeType The MIME type of audio data provided in the input buffers. + * @param channelCount The number of channels. + * @param sampleRate The sample rate in Hz. + * @param pcmEncoding For PCM formats, the encoding used. One of {@link C#ENCODING_PCM_16BIT}, + * {@link C#ENCODING_PCM_16BIT}, {@link C#ENCODING_PCM_24BIT} and + * {@link C#ENCODING_PCM_32BIT}. + * @param specifiedBufferSize A specific size for the playback buffer in bytes, or 0 to infer a + * suitable buffer size. + * @param outputChannels A mapping from input to output channels that is applied to this sink's + * input as a preprocessing step, if handling PCM input. Specify {@code null} to leave the + * input unchanged. Otherwise, the element at index {@code i} specifies index of the input + * channel to map to output channel {@code i} when preprocessing input buffers. After the + * map is applied the audio data will have {@code outputChannels.length} channels. + * @param trimStartSamples The number of audio samples to trim from the start of data written to + * the sink after this call. + * @param trimEndSamples The number of audio samples to trim from data written to the sink + * immediately preceding the next call to {@link #reset()} or + * {@link #configure(String, int, int, int, int, int[], int, int)}. + * @throws ConfigurationException If an error occurs configuring the sink. + */ + void configure(String mimeType, int channelCount, int sampleRate, @C.PcmEncoding int pcmEncoding, + int specifiedBufferSize, @Nullable int[] outputChannels, int trimStartSamples, + int trimEndSamples) throws ConfigurationException; + + /** + * Starts or resumes consuming audio if initialized. + */ + void play(); + + /** + * Signals to the sink that the next buffer is discontinuous with the previous buffer. + */ + void handleDiscontinuity(); + + /** + * Attempts to process data from a {@link ByteBuffer}, starting from its current position and + * ending at its limit (exclusive). The position of the {@link ByteBuffer} is advanced by the + * number of bytes that were handled. {@link Listener#onPositionDiscontinuity()} will be called if + * {@code presentationTimeUs} is discontinuous with the last buffer handled since the last reset. + *
+ * Returns whether the data was handled in full. If the data was not handled in full then the same + * {@link ByteBuffer} must be provided to subsequent calls until it has been fully consumed, + * except in the case of an intervening call to {@link #reset()} (or to + * {@link #configure(String, int, int, int, int, int[], int, int)} that causes the sink to be + * reset). + * + * @param buffer The buffer containing audio data. + * @param presentationTimeUs The presentation timestamp of the buffer in microseconds. + * @return Whether the buffer was handled fully. + * @throws InitializationException If an error occurs initializing the sink. + * @throws WriteException If an error occurs writing the audio data. + */ + boolean handleBuffer(ByteBuffer buffer, long presentationTimeUs) + throws InitializationException, WriteException; + + /** + * Processes any remaining data. {@link #isEnded()} will return {@code true} when no data remains. + * + * @throws WriteException If an error occurs draining data to the sink. + */ + void playToEndOfStream() throws WriteException; + + /** + * Returns whether {@link #playToEndOfStream} has been called and all buffers have been processed. + */ + boolean isEnded(); + + /** + * Returns whether the sink has data pending that has not been consumed yet. + */ + boolean hasPendingData(); + + /** + * Attempts to set the playback parameters and returns the active playback parameters, which may + * differ from those passed in. + * + * @param playbackParameters The new playback parameters to attempt to set. + * @return The active playback parameters. + */ + PlaybackParameters setPlaybackParameters(PlaybackParameters playbackParameters); + + /** + * Gets the active {@link PlaybackParameters}. + */ + PlaybackParameters getPlaybackParameters(); + + /** + * Sets attributes for audio playback. If the attributes have changed and if the sink is not + * configured for use with tunneling, then it is reset and the audio session id is cleared. + *
+ * If the sink is configured for use with tunneling then the audio attributes are ignored. The + * sink is not reset and the audio session id is not cleared. The passed attributes will be used + * if the sink is later re-configured into non-tunneled mode. + * + * @param audioAttributes The attributes for audio playback. + */ + void setAudioAttributes(AudioAttributes audioAttributes); + + /** + * Sets the audio session id. + */ + void setAudioSessionId(int audioSessionId); + + /** + * Enables tunneling, if possible. The sink is reset if tunneling was previously disabled or if + * the audio session id has changed. Enabling tunneling is only possible if the sink is based on a + * platform {@link AudioTrack}, and requires platform API version 21 onwards. + * + * @param tunnelingAudioSessionId The audio session id to use. + * @throws IllegalStateException Thrown if enabling tunneling on platform API version < 21. + */ + void enableTunnelingV21(int tunnelingAudioSessionId); + + /** + * Disables tunneling. If tunneling was previously enabled then the sink is reset and any audio + * session id is cleared. + */ + void disableTunneling(); + + /** + * Sets the playback volume. + * + * @param volume A volume in the range [0.0, 1.0]. + */ + void setVolume(float volume); + + /** + * Pauses playback. + */ + void pause(); + + /** + * Resets the sink, after which it is ready to receive buffers from a new playback position. + *
+ * The audio session may remain active until {@link #release()} is called. + */ + void reset(); + + /** + * Releases any resources associated with this instance. + */ + void release(); + +} diff --git a/library/core/src/main/java/com/google/android/exoplayer2/audio/ChannelMappingAudioProcessor.java b/library/core/src/main/java/com/google/android/exoplayer2/audio/ChannelMappingAudioProcessor.java index ef85985f1f..03bbd5817b 100644 --- a/library/core/src/main/java/com/google/android/exoplayer2/audio/ChannelMappingAudioProcessor.java +++ b/library/core/src/main/java/com/google/android/exoplayer2/audio/ChannelMappingAudioProcessor.java @@ -52,7 +52,7 @@ import java.util.Arrays; * Resets the channel mapping. After calling this method, call {@link #configure(int, int, int)} * to start using the new channel map. * - * @see AudioTrack#configure(String, int, int, int, int, int[], int, int) + * @see AudioSink#configure(String, int, int, int, int, int[], int, int) */ public void setChannelMap(int[] outputChannels) { pendingOutputChannels = outputChannels; diff --git a/library/core/src/main/java/com/google/android/exoplayer2/audio/AudioTrack.java b/library/core/src/main/java/com/google/android/exoplayer2/audio/DefaultAudioSink.java similarity index 76% rename from library/core/src/main/java/com/google/android/exoplayer2/audio/AudioTrack.java rename to library/core/src/main/java/com/google/android/exoplayer2/audio/DefaultAudioSink.java index 25813aefc0..34ea173deb 100644 --- a/library/core/src/main/java/com/google/android/exoplayer2/audio/AudioTrack.java +++ b/library/core/src/main/java/com/google/android/exoplayer2/audio/DefaultAudioSink.java @@ -20,6 +20,7 @@ import android.annotation.TargetApi; import android.media.AudioFormat; import android.media.AudioManager; import android.media.AudioTimestamp; +import android.media.AudioTrack; import android.os.ConditionVariable; import android.os.SystemClock; import android.support.annotation.IntDef; @@ -39,131 +40,19 @@ import java.util.ArrayList; import java.util.LinkedList; /** - * Plays audio data. The implementation delegates to an {@link android.media.AudioTrack} and handles - * playback position smoothing, non-blocking writes and reconfiguration. + * Plays audio data. The implementation delegates to an {@link AudioTrack} and handles playback + * position smoothing, non-blocking writes and reconfiguration. *
- * Before starting playback, specify the input format by calling - * {@link #configure(String, int, int, int, int, int[], int, int)}. Optionally call - * {@link #setAudioSessionId(int)}, {@link #setAudioAttributes(AudioAttributes)}, - * {@link #enableTunnelingV21(int)} and {@link #disableTunneling()} to configure audio playback. - * These methods may be called after writing data to the track, in which case it will be - * reinitialized as required. - *
- * Call {@link #handleBuffer(ByteBuffer, long)} to write data, and {@link #handleDiscontinuity()} - * when the data being fed is discontinuous. Call {@link #play()} to start playing the written data. - *
- * Call {@link #configure(String, int, int, int, int, int[], int, int)} whenever the input format - * changes. The track will be reinitialized on the next call to - * {@link #handleBuffer(ByteBuffer, long)}. - *
- * Calling {@link #reset()} releases the underlying {@link android.media.AudioTrack} (and so does - * calling {@link #configure(String, int, int, int, int, int[], int, int)} unless the format is - * unchanged). It is safe to call {@link #handleBuffer(ByteBuffer, long)} after {@link #reset()} - * without calling {@link #configure(String, int, int, int, int, int[], int, int)}. - *
- * Call {@link #playToEndOfStream()} repeatedly to play out all data when no more input buffers will - * be provided via {@link #handleBuffer(ByteBuffer, long)} until the next {@link #reset}. Call - * {@link #release()} when the instance is no longer required. + * If tunneling mode is enabled, care must be taken that audio processors do not output buffers with + * a different duration than their input, and buffer processors must produce output corresponding to + * their last input immediately after that input is queued. This means that, for example, speed + * adjustment is not possible while using tunneling. */ -public final class AudioTrack { +public final class DefaultAudioSink implements AudioSink { /** - * Listener for audio track events. - */ - public interface Listener { - - /** - * Called when the audio track has been initialized with a newly generated audio session id. - * - * @param audioSessionId The newly generated audio session id. - */ - void onAudioSessionId(int audioSessionId); - - /** - * Called when the audio track handles a buffer whose timestamp is discontinuous with the last - * buffer handled since it was reset. - */ - void onPositionDiscontinuity(); - - /** - * Called when the audio track underruns. - * - * @param bufferSize The size of the track's buffer, in bytes. - * @param bufferSizeMs The size of the track's buffer, in milliseconds, if it is configured for - * PCM output. {@link C#TIME_UNSET} if it is configured for passthrough output, as the - * buffered media can have a variable bitrate so the duration may be unknown. - * @param elapsedSinceLastFeedMs The time since the track was last fed data, in milliseconds. - */ - void onUnderrun(int bufferSize, long bufferSizeMs, long elapsedSinceLastFeedMs); - - } - - /** - * Thrown when a failure occurs configuring the track. - */ - public static final class ConfigurationException extends Exception { - - public ConfigurationException(Throwable cause) { - super(cause); - } - - public ConfigurationException(String message) { - super(message); - } - - } - - /** - * Thrown when a failure occurs initializing an {@link android.media.AudioTrack}. - */ - public static final class InitializationException extends Exception { - - /** - * The state as reported by {@link android.media.AudioTrack#getState()}. - */ - public final int audioTrackState; - - /** - * @param audioTrackState The state as reported by {@link android.media.AudioTrack#getState()}. - * @param sampleRate The requested sample rate in Hz. - * @param channelConfig The requested channel configuration. - * @param bufferSize The requested buffer size in bytes. - */ - public InitializationException(int audioTrackState, int sampleRate, int channelConfig, - int bufferSize) { - super("AudioTrack init failed: " + audioTrackState + ", Config(" + sampleRate + ", " - + channelConfig + ", " + bufferSize + ")"); - this.audioTrackState = audioTrackState; - } - - } - - /** - * Thrown when a failure occurs writing to an {@link android.media.AudioTrack}. - */ - public static final class WriteException extends Exception { - - /** - * The error value returned from {@link android.media.AudioTrack#write(byte[], int, int)} or - * {@link android.media.AudioTrack#write(ByteBuffer, int, int)}. - */ - public final int errorCode; - - /** - * @param errorCode The error value returned from - * {@link android.media.AudioTrack#write(byte[], int, int)} or - * {@link android.media.AudioTrack#write(ByteBuffer, int, int)}. - */ - public WriteException(int errorCode) { - super("AudioTrack write failed: " + errorCode); - this.errorCode = errorCode; - } - - } - - /** - * Thrown when {@link android.media.AudioTrack#getTimestamp} returns a spurious timestamp, if - * {@code AudioTrack#failOnSpuriousAudioTimestamp} is set. + * Thrown when {@link AudioTrack#getTimestamp} returns a spurious timestamp, if + * {@link #failOnSpuriousAudioTimestamp} is set. */ public static final class InvalidAudioTrackTimestampException extends RuntimeException { @@ -177,61 +66,56 @@ public final class AudioTrack { } /** - * Returned by {@link #getCurrentPositionUs(boolean)} when the position is not set. - */ - public static final long CURRENT_POSITION_NOT_SET = Long.MIN_VALUE; - - /** - * A minimum length for the {@link android.media.AudioTrack} buffer, in microseconds. + * A minimum length for the {@link AudioTrack} buffer, in microseconds. */ private static final long MIN_BUFFER_DURATION_US = 250000; /** - * A maximum length for the {@link android.media.AudioTrack} buffer, in microseconds. + * A maximum length for the {@link AudioTrack} buffer, in microseconds. */ private static final long MAX_BUFFER_DURATION_US = 750000; /** - * The length for passthrough {@link android.media.AudioTrack} buffers, in microseconds. + * The length for passthrough {@link AudioTrack} buffers, in microseconds. */ private static final long PASSTHROUGH_BUFFER_DURATION_US = 250000; /** * A multiplication factor to apply to the minimum buffer size requested by the underlying - * {@link android.media.AudioTrack}. + * {@link AudioTrack}. */ private static final int BUFFER_MULTIPLICATION_FACTOR = 4; /** - * @see android.media.AudioTrack#PLAYSTATE_STOPPED + * @see AudioTrack#PLAYSTATE_STOPPED */ - private static final int PLAYSTATE_STOPPED = android.media.AudioTrack.PLAYSTATE_STOPPED; + private static final int PLAYSTATE_STOPPED = AudioTrack.PLAYSTATE_STOPPED; /** - * @see android.media.AudioTrack#PLAYSTATE_PAUSED + * @see AudioTrack#PLAYSTATE_PAUSED */ - private static final int PLAYSTATE_PAUSED = android.media.AudioTrack.PLAYSTATE_PAUSED; + private static final int PLAYSTATE_PAUSED = AudioTrack.PLAYSTATE_PAUSED; /** - * @see android.media.AudioTrack#PLAYSTATE_PLAYING + * @see AudioTrack#PLAYSTATE_PLAYING */ - private static final int PLAYSTATE_PLAYING = android.media.AudioTrack.PLAYSTATE_PLAYING; + private static final int PLAYSTATE_PLAYING = AudioTrack.PLAYSTATE_PLAYING; /** - * @see android.media.AudioTrack#ERROR_BAD_VALUE + * @see AudioTrack#ERROR_BAD_VALUE */ - private static final int ERROR_BAD_VALUE = android.media.AudioTrack.ERROR_BAD_VALUE; + private static final int ERROR_BAD_VALUE = AudioTrack.ERROR_BAD_VALUE; /** - * @see android.media.AudioTrack#MODE_STATIC + * @see AudioTrack#MODE_STATIC */ - private static final int MODE_STATIC = android.media.AudioTrack.MODE_STATIC; + private static final int MODE_STATIC = AudioTrack.MODE_STATIC; /** - * @see android.media.AudioTrack#MODE_STREAM + * @see AudioTrack#MODE_STREAM */ - private static final int MODE_STREAM = android.media.AudioTrack.MODE_STREAM; + private static final int MODE_STREAM = AudioTrack.MODE_STREAM; /** - * @see android.media.AudioTrack#STATE_INITIALIZED + * @see AudioTrack#STATE_INITIALIZED */ - private static final int STATE_INITIALIZED = android.media.AudioTrack.STATE_INITIALIZED; + private static final int STATE_INITIALIZED = AudioTrack.STATE_INITIALIZED; /** - * @see android.media.AudioTrack#WRITE_NON_BLOCKING + * @see AudioTrack#WRITE_NON_BLOCKING */ @SuppressLint("InlinedApi") - private static final int WRITE_NON_BLOCKING = android.media.AudioTrack.WRITE_NON_BLOCKING; + private static final int WRITE_NON_BLOCKING = AudioTrack.WRITE_NON_BLOCKING; private static final String TAG = "AudioTrack"; @@ -282,7 +166,7 @@ public final class AudioTrack { /** * Whether to throw an {@link InvalidAudioTrackTimestampException} when a spurious timestamp is - * reported from {@link android.media.AudioTrack#getTimestamp}. + * reported from {@link AudioTrack#getTimestamp}. *
* The flag must be set before creating a player. Should be set to {@code true} for testing and
* debugging purposes only.
@@ -294,18 +178,17 @@ public final class AudioTrack {
private final TrimmingAudioProcessor trimmingAudioProcessor;
private final SonicAudioProcessor sonicAudioProcessor;
private final AudioProcessor[] availableAudioProcessors;
- private final Listener listener;
private final ConditionVariable releasingConditionVariable;
private final long[] playheadOffsets;
private final AudioTrackUtil audioTrackUtil;
private final LinkedList If the device supports it, the method uses the playback timestamp from
- * {@link android.media.AudioTrack#getTimestamp}. Otherwise, it derives a smoothed position by
- * sampling the {@link android.media.AudioTrack}'s frame position.
- *
- * @param sourceEnded Specify {@code true} if no more input buffers will be provided.
- * @return The playback position relative to the start of playback, in microseconds.
- */
+ @Override
public long getCurrentPositionUs(boolean sourceEnded) {
+ // If the device supports it, use the playback timestamp from AudioTrack.getTimestamp.
+ // Otherwise, derive a smoothed position by sampling the track's frame position.
if (!hasCurrentPositionUs()) {
return CURRENT_POSITION_NOT_SET;
}
@@ -462,29 +335,7 @@ public final class AudioTrack {
return startMediaTimeUs + applySpeedup(positionUs);
}
- /**
- * Configures (or reconfigures) the audio track.
- *
- * @param mimeType The mime type.
- * @param channelCount The number of channels.
- * @param sampleRate The sample rate in Hz.
- * @param pcmEncoding For PCM formats, the encoding used. One of {@link C#ENCODING_PCM_16BIT},
- * {@link C#ENCODING_PCM_16BIT}, {@link C#ENCODING_PCM_24BIT} and
- * {@link C#ENCODING_PCM_32BIT}.
- * @param specifiedBufferSize A specific size for the playback buffer in bytes, or 0 to infer a
- * suitable buffer size automatically.
- * @param outputChannels A mapping from input to output channels that is applied to this track's
- * input as a preprocessing step, if handling PCM input. Specify {@code null} to leave the
- * input unchanged. Otherwise, the element at index {@code i} specifies index of the input
- * channel to map to output channel {@code i} when preprocessing input buffers. After the
- * map is applied the audio data will have {@code outputChannels.length} channels.
- * @param trimStartSamples The number of audio samples to trim from the start of data written to
- * the track after this call.
- * @param trimEndSamples The number of audio samples to trim from data written to the track
- * immediately preceding the next call to {@link #reset()} or
- * {@link #configure(String, int, int, int, int, int[], int, int)}.
- * @throws ConfigurationException If an error occurs configuring the track.
- */
+ @Override
public void configure(String mimeType, int channelCount, int sampleRate,
@C.PcmEncoding int pcmEncoding, int specifiedBufferSize, @Nullable int[] outputChannels,
int trimStartSamples, int trimEndSamples) throws ConfigurationException {
@@ -590,8 +441,7 @@ public final class AudioTrack {
bufferSize = (int) (PASSTHROUGH_BUFFER_DURATION_US * 192 * 1024 / C.MICROS_PER_SECOND);
}
} else {
- int minBufferSize =
- android.media.AudioTrack.getMinBufferSize(sampleRate, channelConfig, outputEncoding);
+ int minBufferSize = AudioTrack.getMinBufferSize(sampleRate, channelConfig, outputEncoding);
Assertions.checkState(minBufferSize != ERROR_BAD_VALUE);
int multipliedBufferSize = minBufferSize * BUFFER_MULTIPLICATION_FACTOR;
int minAppBufferSize = (int) durationUsToFrames(MIN_BUFFER_DURATION_US) * outputPcmFrameSize;
@@ -651,7 +501,9 @@ public final class AudioTrack {
}
if (this.audioSessionId != audioSessionId) {
this.audioSessionId = audioSessionId;
- listener.onAudioSessionId(audioSessionId);
+ if (listener != null) {
+ listener.onAudioSessionId(audioSessionId);
+ }
}
audioTrackUtil.reconfigure(audioTrack, needsPassthroughWorkarounds());
@@ -659,9 +511,7 @@ public final class AudioTrack {
hasData = false;
}
- /**
- * Starts or resumes playing audio if the audio track has been initialized.
- */
+ @Override
public void play() {
playing = true;
if (isInitialized()) {
@@ -670,9 +520,7 @@ public final class AudioTrack {
}
}
- /**
- * Signals to the audio track that the next buffer is discontinuous with the previous buffer.
- */
+ @Override
public void handleDiscontinuity() {
// Force resynchronization after a skipped buffer.
if (startMediaTimeState == START_IN_SYNC) {
@@ -680,24 +528,7 @@ public final class AudioTrack {
}
}
- /**
- * Attempts to process data from a {@link ByteBuffer}, starting from its current position and
- * ending at its limit (exclusive). The position of the {@link ByteBuffer} is advanced by the
- * number of bytes that were handled. {@link Listener#onPositionDiscontinuity()} will be called if
- * {@code presentationTimeUs} is discontinuous with the last buffer handled since the last reset.
- *
- * Returns whether the data was handled in full. If the data was not handled in full then the same
- * {@link ByteBuffer} must be provided to subsequent calls until it has been fully consumed,
- * except in the case of an interleaving call to {@link #reset()} (or an interleaving call to
- * {@link #configure(String, int, int, int, int, int[], int, int)} that caused the track to be
- * reset).
- *
- * @param buffer The buffer containing audio data.
- * @param presentationTimeUs The presentation timestamp of the buffer in microseconds.
- * @return Whether the buffer was handled fully.
- * @throws InitializationException If an error occurs initializing the track.
- * @throws WriteException If an error occurs writing the audio data.
- */
+ @Override
@SuppressWarnings("ReferenceEquality")
public boolean handleBuffer(ByteBuffer buffer, long presentationTimeUs)
throws InitializationException, WriteException {
@@ -729,7 +560,7 @@ public final class AudioTrack {
boolean hadData = hasData;
hasData = hasPendingData();
- if (hadData && !hasData && audioTrack.getPlayState() != PLAYSTATE_STOPPED) {
+ if (hadData && !hasData && audioTrack.getPlayState() != PLAYSTATE_STOPPED && listener != null) {
long elapsedSinceLastFeedMs = SystemClock.elapsedRealtime() - lastFeedElapsedRealtimeMs;
listener.onUnderrun(bufferSize, C.usToMs(bufferSizeUs), elapsedSinceLastFeedMs);
}
@@ -779,7 +610,9 @@ public final class AudioTrack {
// number of bytes submitted.
startMediaTimeUs += (presentationTimeUs - expectedPresentationTimeUs);
startMediaTimeState = START_IN_SYNC;
- listener.onPositionDiscontinuity();
+ if (listener != null) {
+ listener.onPositionDiscontinuity();
+ }
}
}
@@ -899,11 +732,7 @@ public final class AudioTrack {
return false;
}
- /**
- * Plays out remaining audio. {@link #isEnded()} will return {@code true} when playback has ended.
- *
- * @throws WriteException If an error occurs draining data to the track.
- */
+ @Override
public void playToEndOfStream() throws WriteException {
if (handledEndOfStream || !isInitialized()) {
return;
@@ -947,30 +776,19 @@ public final class AudioTrack {
return true;
}
- /**
- * Returns whether all buffers passed to {@link #handleBuffer(ByteBuffer, long)} have been
- * completely processed and played.
- */
+ @Override
public boolean isEnded() {
return !isInitialized() || (handledEndOfStream && !hasPendingData());
}
- /**
- * Returns whether the audio track has more data pending that will be played back.
- */
+ @Override
public boolean hasPendingData() {
return isInitialized()
&& (getWrittenFrames() > audioTrackUtil.getPlaybackHeadPosition()
|| overrideHasPendingData());
}
- /**
- * Attempts to set the playback parameters and returns the active playback parameters, which may
- * differ from those passed in.
- *
- * @param playbackParameters The new playback parameters to attempt to set.
- * @return The active playback parameters.
- */
+ @Override
public PlaybackParameters setPlaybackParameters(PlaybackParameters playbackParameters) {
if (passthrough) {
// The playback parameters are always the default in passthrough mode.
@@ -997,24 +815,12 @@ public final class AudioTrack {
return this.playbackParameters;
}
- /**
- * Gets the {@link PlaybackParameters}.
- */
+ @Override
public PlaybackParameters getPlaybackParameters() {
return playbackParameters;
}
- /**
- * Sets the attributes for audio playback. If the attributes have changed and if the audio track
- * is not configured for use with tunneling, then the audio track is reset and the audio session
- * id is cleared.
- *
- * If the audio track is configured for use with tunneling then the audio attributes are ignored.
- * The audio track is not reset and the audio session id is not cleared. The passed attributes
- * will be used if the audio track is later re-configured into non-tunneled mode.
- *
- * @param audioAttributes The attributes for audio playback.
- */
+ @Override
public void setAudioAttributes(AudioAttributes audioAttributes) {
if (this.audioAttributes.equals(audioAttributes)) {
return;
@@ -1028,9 +834,7 @@ public final class AudioTrack {
audioSessionId = C.AUDIO_SESSION_ID_UNSET;
}
- /**
- * Sets the audio session id. The audio track is reset if the audio session id has changed.
- */
+ @Override
public void setAudioSessionId(int audioSessionId) {
if (this.audioSessionId != audioSessionId) {
this.audioSessionId = audioSessionId;
@@ -1038,18 +842,7 @@ public final class AudioTrack {
}
}
- /**
- * Enables tunneling. The audio track is reset if tunneling was previously disabled or if the
- * audio session id has changed. Enabling tunneling requires platform API version 21 onwards.
- *
- * If this instance has {@link AudioProcessor}s and tunneling is enabled, care must be taken that
- * audio processors do not output buffers with a different duration than their input, and buffer
- * processors must produce output corresponding to their last input immediately after that input
- * is queued.
- *
- * @param tunnelingAudioSessionId The audio session id to use.
- * @throws IllegalStateException Thrown if enabling tunneling on platform API version < 21.
- */
+ @Override
public void enableTunnelingV21(int tunnelingAudioSessionId) {
Assertions.checkState(Util.SDK_INT >= 21);
if (!tunneling || audioSessionId != tunnelingAudioSessionId) {
@@ -1059,10 +852,7 @@ public final class AudioTrack {
}
}
- /**
- * Disables tunneling. If tunneling was previously enabled then the audio track is reset and the
- * audio session id is cleared.
- */
+ @Override
public void disableTunneling() {
if (tunneling) {
tunneling = false;
@@ -1071,11 +861,7 @@ public final class AudioTrack {
}
}
- /**
- * Sets the playback volume.
- *
- * @param volume A volume in the range [0.0, 1.0].
- */
+ @Override
public void setVolume(float volume) {
if (this.volume != volume) {
this.volume = volume;
@@ -1093,9 +879,7 @@ public final class AudioTrack {
}
}
- /**
- * Pauses playback.
- */
+ @Override
public void pause() {
playing = false;
if (isInitialized()) {
@@ -1104,13 +888,7 @@ public final class AudioTrack {
}
}
- /**
- * Releases the underlying audio track asynchronously.
- *
- * Calling {@link #handleBuffer(ByteBuffer, long)} will block until the audio track has been
- * released, so it is safe to use the audio track immediately after a reset. The audio session may
- * remain active until {@link #release()} is called.
- */
+ @Override
public void reset() {
if (isInitialized()) {
submittedPcmBytes = 0;
@@ -1146,7 +924,7 @@ public final class AudioTrack {
audioTrack.pause();
}
// AudioTrack.release can take some time, so we call it on a background thread.
- final android.media.AudioTrack toRelease = audioTrack;
+ final AudioTrack toRelease = audioTrack;
audioTrack = null;
audioTrackUtil.reconfigure(null, false);
releasingConditionVariable.close();
@@ -1164,9 +942,7 @@ public final class AudioTrack {
}
}
- /**
- * Releases all resources associated with this instance.
- */
+ @Override
public void release() {
reset();
releaseKeepSessionIdAudioTrack();
@@ -1186,7 +962,7 @@ public final class AudioTrack {
}
// AudioTrack.release can take some time, so we call it on a background thread.
- final android.media.AudioTrack toRelease = keepSessionIdAudioTrack;
+ final AudioTrack toRelease = keepSessionIdAudioTrack;
keepSessionIdAudioTrack = null;
new Thread() {
@Override
@@ -1367,19 +1143,19 @@ public final class AudioTrack {
&& audioTrack.getPlaybackHeadPosition() == 0;
}
- private android.media.AudioTrack initializeAudioTrack() throws InitializationException {
- android.media.AudioTrack audioTrack;
+ private AudioTrack initializeAudioTrack() throws InitializationException {
+ AudioTrack audioTrack;
if (Util.SDK_INT >= 21) {
audioTrack = createAudioTrackV21();
} else {
int streamType = Util.getStreamTypeForAudioUsage(audioAttributes.usage);
if (audioSessionId == C.AUDIO_SESSION_ID_UNSET) {
- audioTrack = new android.media.AudioTrack(streamType, sampleRate, channelConfig,
- outputEncoding, bufferSize, MODE_STREAM);
+ audioTrack = new AudioTrack(streamType, sampleRate, channelConfig, outputEncoding,
+ bufferSize, MODE_STREAM);
} else {
// Re-attach to the same audio session.
- audioTrack = new android.media.AudioTrack(streamType, sampleRate, channelConfig,
- outputEncoding, bufferSize, MODE_STREAM, audioSessionId);
+ audioTrack = new AudioTrack(streamType, sampleRate, channelConfig, outputEncoding,
+ bufferSize, MODE_STREAM, audioSessionId);
}
}
@@ -1397,7 +1173,7 @@ public final class AudioTrack {
}
@TargetApi(21)
- private android.media.AudioTrack createAudioTrackV21() {
+ private AudioTrack createAudioTrackV21() {
android.media.AudioAttributes attributes;
if (tunneling) {
attributes = new android.media.AudioAttributes.Builder()
@@ -1415,17 +1191,16 @@ public final class AudioTrack {
.build();
int audioSessionId = this.audioSessionId != C.AUDIO_SESSION_ID_UNSET ? this.audioSessionId
: AudioManager.AUDIO_SESSION_ID_GENERATE;
- return new android.media.AudioTrack(attributes, format, bufferSize, MODE_STREAM,
- audioSessionId);
+ return new AudioTrack(attributes, format, bufferSize, MODE_STREAM, audioSessionId);
}
- private android.media.AudioTrack initializeKeepSessionIdAudioTrack(int audioSessionId) {
- int sampleRate = 4000; // Equal to private android.media.AudioTrack.MIN_SAMPLE_RATE.
+ private AudioTrack initializeKeepSessionIdAudioTrack(int audioSessionId) {
+ int sampleRate = 4000; // Equal to private AudioTrack.MIN_SAMPLE_RATE.
int channelConfig = AudioFormat.CHANNEL_OUT_MONO;
@C.PcmEncoding int encoding = C.ENCODING_PCM_16BIT;
int bufferSize = 2; // Use a two byte buffer, as it is not actually used for playback.
- return new android.media.AudioTrack(C.STREAM_TYPE_DEFAULT, sampleRate, channelConfig, encoding,
- bufferSize, MODE_STATIC, audioSessionId);
+ return new AudioTrack(C.STREAM_TYPE_DEFAULT, sampleRate, channelConfig, encoding, bufferSize,
+ MODE_STATIC, audioSessionId);
}
@C.Encoding
@@ -1457,14 +1232,13 @@ public final class AudioTrack {
}
@TargetApi(21)
- private static int writeNonBlockingV21(android.media.AudioTrack audioTrack, ByteBuffer buffer,
- int size) {
+ private static int writeNonBlockingV21(AudioTrack audioTrack, ByteBuffer buffer, int size) {
return audioTrack.write(buffer, size, WRITE_NON_BLOCKING);
}
@TargetApi(21)
- private int writeNonBlockingWithAvSyncV21(android.media.AudioTrack audioTrack,
- ByteBuffer buffer, int size, long presentationTimeUs) {
+ private int writeNonBlockingWithAvSyncV21(AudioTrack audioTrack, ByteBuffer buffer, int size,
+ long presentationTimeUs) {
// TODO: Uncomment this when [Internal ref: b/33627517] is clarified or fixed.
// if (Util.SDK_INT >= 23) {
// // The underlying platform AudioTrack writes AV sync headers directly.
@@ -1502,21 +1276,21 @@ public final class AudioTrack {
}
@TargetApi(21)
- private static void setVolumeInternalV21(android.media.AudioTrack audioTrack, float volume) {
+ private static void setVolumeInternalV21(AudioTrack audioTrack, float volume) {
audioTrack.setVolume(volume);
}
@SuppressWarnings("deprecation")
- private static void setVolumeInternalV3(android.media.AudioTrack audioTrack, float volume) {
+ private static void setVolumeInternalV3(AudioTrack audioTrack, float volume) {
audioTrack.setStereoVolume(volume, volume);
}
/**
- * Wraps an {@link android.media.AudioTrack} to expose useful utility methods.
+ * Wraps an {@link AudioTrack} to expose useful utility methods.
*/
private static class AudioTrackUtil {
- protected android.media.AudioTrack audioTrack;
+ protected AudioTrack audioTrack;
private boolean needsPassthroughWorkaround;
private int sampleRate;
private long lastRawPlaybackHeadPosition;
@@ -1534,8 +1308,7 @@ public final class AudioTrack {
* @param needsPassthroughWorkaround Whether to workaround issues with pausing AC-3 passthrough
* audio tracks on platform API version 21/22.
*/
- public void reconfigure(android.media.AudioTrack audioTrack,
- boolean needsPassthroughWorkaround) {
+ public void reconfigure(AudioTrack audioTrack, boolean needsPassthroughWorkaround) {
this.audioTrack = audioTrack;
this.needsPassthroughWorkaround = needsPassthroughWorkaround;
stopTimestampUs = C.TIME_UNSET;
@@ -1574,9 +1347,9 @@ public final class AudioTrack {
}
/**
- * {@link android.media.AudioTrack#getPlaybackHeadPosition()} returns a value intended to be
- * interpreted as an unsigned 32 bit integer, which also wraps around periodically. This method
- * returns the playback head position as a long that will only wrap around if the value exceeds
+ * {@link AudioTrack#getPlaybackHeadPosition()} returns a value intended to be interpreted as an
+ * unsigned 32 bit integer, which also wraps around periodically. This method returns the
+ * playback head position as a long that will only wrap around if the value exceeds
* {@link Long#MAX_VALUE} (which in practice will never happen).
*
* @return The playback head position, in frames.
@@ -1676,8 +1449,7 @@ public final class AudioTrack {
}
@Override
- public void reconfigure(android.media.AudioTrack audioTrack,
- boolean needsPassthroughWorkaround) {
+ public void reconfigure(AudioTrack audioTrack, boolean needsPassthroughWorkaround) {
super.reconfigure(audioTrack, needsPassthroughWorkaround);
rawTimestampFramePositionWrapCount = 0;
lastRawTimestampFramePosition = 0;
diff --git a/library/core/src/main/java/com/google/android/exoplayer2/audio/MediaCodecAudioRenderer.java b/library/core/src/main/java/com/google/android/exoplayer2/audio/MediaCodecAudioRenderer.java
index cbb3a4944d..f8206e94cf 100644
--- a/library/core/src/main/java/com/google/android/exoplayer2/audio/MediaCodecAudioRenderer.java
+++ b/library/core/src/main/java/com/google/android/exoplayer2/audio/MediaCodecAudioRenderer.java
@@ -40,13 +40,13 @@ import com.google.android.exoplayer2.util.Util;
import java.nio.ByteBuffer;
/**
- * Decodes and renders audio using {@link MediaCodec} and {@link AudioTrack}.
+ * Decodes and renders audio using {@link MediaCodec} and an {@link AudioSink}.
*/
@TargetApi(16)
public class MediaCodecAudioRenderer extends MediaCodecRenderer implements MediaClock {
private final EventDispatcher eventDispatcher;
- private final AudioTrack audioTrack;
+ private final AudioSink audioSink;
private boolean passthroughEnabled;
private boolean codecNeedsDiscardChannelsWorkaround;
@@ -110,7 +110,7 @@ public class MediaCodecAudioRenderer extends MediaCodecRenderer implements Media
boolean playClearSamplesWithoutKeys, @Nullable Handler eventHandler,
@Nullable AudioRendererEventListener eventListener) {
this(mediaCodecSelector, drmSessionManager, playClearSamplesWithoutKeys, eventHandler,
- eventListener, null);
+ eventListener, (AudioCapabilities) null);
}
/**
@@ -135,9 +135,32 @@ public class MediaCodecAudioRenderer extends MediaCodecRenderer implements Media
boolean playClearSamplesWithoutKeys, @Nullable Handler eventHandler,
@Nullable AudioRendererEventListener eventListener,
@Nullable AudioCapabilities audioCapabilities, AudioProcessor... audioProcessors) {
+ this(mediaCodecSelector, drmSessionManager, playClearSamplesWithoutKeys,
+ eventHandler, eventListener, new DefaultAudioSink(audioCapabilities, audioProcessors));
+ }
+
+ /**
+ * @param mediaCodecSelector A decoder selector.
+ * @param drmSessionManager For use with encrypted content. May be null if support for encrypted
+ * content is not required.
+ * @param playClearSamplesWithoutKeys Encrypted media may contain clear (un-encrypted) regions.
+ * For example a media file may start with a short clear region so as to allow playback to
+ * begin in parallel with key acquisition. This parameter specifies whether the renderer is
+ * permitted to play clear regions of encrypted media files before {@code drmSessionManager}
+ * has obtained the keys necessary to decrypt encrypted regions of the media.
+ * @param eventHandler A handler to use when delivering events to {@code eventListener}. May be
+ * null if delivery of events is not required.
+ * @param eventListener A listener of events. May be null if delivery of events is not required.
+ * @param audioSink The sink to which audio will be output.
+ */
+ public MediaCodecAudioRenderer(MediaCodecSelector mediaCodecSelector,
+ @Nullable DrmSessionManager