mirror of
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Factor out AudioTrack from MediaCodecAudioTrackRenderer.
AudioTrack contains the portions of MediaCodecAudioTrackRenderer that handle the platform AudioTrack instance, including synchronization (playback position smoothing), non-blocking writes and releasing. This refactoring should not affect the behavior of audio playback, and is in preparation for adding an Ac3PassthroughAudioTrackRenderer that will use the AudioTrack.
This commit is contained in:
Oliver Woodman
parent
44bc01b28d
commit
bc303b730a
4 changed files with 800 additions and 561 deletions
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@ -16,8 +16,8 @@
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package com.google.android.exoplayer.demo.full;
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import com.google.android.exoplayer.ExoPlayer;
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import com.google.android.exoplayer.MediaCodecAudioTrackRenderer.AudioTrackInitializationException;
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import com.google.android.exoplayer.MediaCodecTrackRenderer.DecoderInitializationException;
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import com.google.android.exoplayer.audio.AudioTrack;
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import com.google.android.exoplayer.demo.full.player.DemoPlayer;
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import com.google.android.exoplayer.util.VerboseLogUtil;
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@ -149,7 +149,7 @@ public class EventLogger implements DemoPlayer.Listener, DemoPlayer.InfoListener
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}
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@Override
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public void onAudioTrackInitializationError(AudioTrackInitializationException e) {
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public void onAudioTrackInitializationError(AudioTrack.InitializationException e) {
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printInternalError("audioTrackInitializationError", e);
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}
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@ -19,10 +19,10 @@ import com.google.android.exoplayer.DummyTrackRenderer;
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import com.google.android.exoplayer.ExoPlaybackException;
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import com.google.android.exoplayer.ExoPlayer;
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import com.google.android.exoplayer.MediaCodecAudioTrackRenderer;
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import com.google.android.exoplayer.MediaCodecAudioTrackRenderer.AudioTrackInitializationException;
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import com.google.android.exoplayer.MediaCodecTrackRenderer.DecoderInitializationException;
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import com.google.android.exoplayer.MediaCodecVideoTrackRenderer;
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import com.google.android.exoplayer.TrackRenderer;
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import com.google.android.exoplayer.audio.AudioTrack;
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import com.google.android.exoplayer.chunk.ChunkSampleSource;
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import com.google.android.exoplayer.chunk.MultiTrackChunkSource;
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import com.google.android.exoplayer.drm.StreamingDrmSessionManager;
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@ -106,7 +106,7 @@ public class DemoPlayer implements ExoPlayer.Listener, ChunkSampleSource.EventLi
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*/
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public interface InternalErrorListener {
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void onRendererInitializationError(Exception e);
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void onAudioTrackInitializationError(AudioTrackInitializationException e);
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void onAudioTrackInitializationError(AudioTrack.InitializationException e);
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void onDecoderInitializationError(DecoderInitializationException e);
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void onCryptoError(CryptoException e);
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void onUpstreamError(int sourceId, IOException e);
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@ -424,7 +424,7 @@ public class DemoPlayer implements ExoPlayer.Listener, ChunkSampleSource.EventLi
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}
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@Override
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public void onAudioTrackInitializationError(AudioTrackInitializationException e) {
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public void onAudioTrackInitializationError(AudioTrack.InitializationException e) {
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if (internalErrorListener != null) {
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internalErrorListener.onAudioTrackInitializationError(e);
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}
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@ -15,28 +15,21 @@
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*/
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package com.google.android.exoplayer;
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import com.google.android.exoplayer.audio.AudioTrack;
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import com.google.android.exoplayer.drm.DrmSessionManager;
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import com.google.android.exoplayer.util.Assertions;
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import com.google.android.exoplayer.util.MimeTypes;
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import com.google.android.exoplayer.util.Util;
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import android.annotation.TargetApi;
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import android.media.AudioFormat;
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import android.media.AudioManager;
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import android.media.AudioTimestamp;
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import android.media.AudioTrack;
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import android.media.MediaCodec;
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import android.media.MediaFormat;
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import android.media.audiofx.Virtualizer;
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import android.os.ConditionVariable;
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import android.os.Handler;
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import android.util.Log;
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import java.lang.reflect.Method;
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import java.nio.ByteBuffer;
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/**
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* Decodes and renders audio using {@link MediaCodec} and {@link AudioTrack}.
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* Decodes and renders audio using {@link MediaCodec} and {@link android.media.AudioTrack}.
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*/
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@TargetApi(16)
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public class MediaCodecAudioTrackRenderer extends MediaCodecTrackRenderer {
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@ -52,26 +45,7 @@ public class MediaCodecAudioTrackRenderer extends MediaCodecTrackRenderer {
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*
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* @param e The corresponding exception.
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*/
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void onAudioTrackInitializationError(AudioTrackInitializationException e);
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}
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/**
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* Thrown when a failure occurs instantiating an audio track.
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*/
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public static class AudioTrackInitializationException extends Exception {
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/**
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* The state as reported by {@link AudioTrack#getState()}
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*/
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public final int audioTrackState;
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public AudioTrackInitializationException(int audioTrackState, int sampleRate,
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int channelConfig, int bufferSize) {
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super("AudioTrack init failed: " + audioTrackState + ", Config(" + sampleRate + ", "
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+ channelConfig + ", " + bufferSize + ")");
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this.audioTrackState = audioTrackState;
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}
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void onAudioTrackInitializationError(AudioTrack.InitializationException e);
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}
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@ -82,73 +56,12 @@ public class MediaCodecAudioTrackRenderer extends MediaCodecTrackRenderer {
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*/
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public static final int MSG_SET_VOLUME = 1;
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/**
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* The default multiplication factor used when determining the size of the underlying
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* {@link AudioTrack}'s buffer.
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*/
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public static final float DEFAULT_MIN_BUFFER_MULTIPLICATION_FACTOR = 4;
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private static final String TAG = "MediaCodecAudioTrackRenderer";
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private static final long MICROS_PER_SECOND = 1000000L;
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/**
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* AudioTrack timestamps are deemed spurious if they are offset from the system clock by more
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* than this amount.
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* <p>
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* This is a fail safe that should not be required on correctly functioning devices.
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*/
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private static final long MAX_AUDIO_TIMESTAMP_OFFSET_US = 10 * MICROS_PER_SECOND;
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/**
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* AudioTrack latencies are deemed impossibly large if they are greater than this amount.
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* <p>
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* This is a fail safe that should not be required on correctly functioning devices.
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*/
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private static final long MAX_AUDIO_TRACK_LATENCY_US = 10 * MICROS_PER_SECOND;
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private static final int MAX_PLAYHEAD_OFFSET_COUNT = 10;
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private static final int MIN_PLAYHEAD_OFFSET_SAMPLE_INTERVAL_US = 30000;
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private static final int MIN_TIMESTAMP_SAMPLE_INTERVAL_US = 500000;
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private static final int START_NOT_SET = 0;
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private static final int START_IN_SYNC = 1;
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private static final int START_NEED_SYNC = 2;
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private final EventListener eventListener;
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private final ConditionVariable audioTrackReleasingConditionVariable;
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private final AudioTimestampCompat audioTimestampCompat;
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private final long[] playheadOffsets;
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private final float minBufferMultiplicationFactor;
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private int nextPlayheadOffsetIndex;
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private int playheadOffsetCount;
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private long smoothedPlayheadOffsetUs;
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private long lastPlayheadSampleTimeUs;
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private boolean audioTimestampSet;
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private long lastTimestampSampleTimeUs;
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private long lastRawPlaybackHeadPosition;
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private long rawPlaybackHeadWrapCount;
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private int sampleRate;
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private int frameSize;
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private int channelConfig;
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private int minBufferSize;
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private int bufferSize;
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private AudioTrack audioTrack;
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private Method audioTrackGetLatencyMethod;
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private final AudioTrack audioTrack;
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private int audioSessionId;
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private long submittedBytes;
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private int audioTrackStartMediaTimeState;
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private long audioTrackStartMediaTimeUs;
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private long audioTrackResumeSystemTimeUs;
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private long lastReportedCurrentPositionUs;
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private long audioTrackLatencyUs;
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private float volume;
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private byte[] temporaryBuffer;
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private int temporaryBufferOffset;
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private int temporaryBufferSize;
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private long currentPositionUs;
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/**
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* @param source The upstream source from which the renderer obtains samples.
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@ -198,15 +111,16 @@ public class MediaCodecAudioTrackRenderer extends MediaCodecTrackRenderer {
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*/
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public MediaCodecAudioTrackRenderer(SampleSource source, DrmSessionManager drmSessionManager,
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boolean playClearSamplesWithoutKeys, Handler eventHandler, EventListener eventListener) {
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this(source, drmSessionManager, playClearSamplesWithoutKeys,
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DEFAULT_MIN_BUFFER_MULTIPLICATION_FACTOR, eventHandler, eventListener);
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this(source, drmSessionManager, playClearSamplesWithoutKeys, eventHandler, eventListener,
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new AudioTrack());
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}
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/**
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* @param source The upstream source from which the renderer obtains samples.
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* @param minBufferMultiplicationFactor When instantiating an underlying {@link AudioTrack},
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* the size of the track's is calculated as this value multiplied by the minimum buffer size
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* obtained from {@link AudioTrack#getMinBufferSize(int, int, int)}. The multiplication
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* @param minBufferMultiplicationFactor When instantiating an underlying
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* {@link android.media.AudioTrack}, the size of the track is calculated as this value
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* multiplied by the minimum buffer size obtained from
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* {@link android.media.AudioTrack#getMinBufferSize(int, int, int)}. The multiplication
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* factor must be greater than or equal to 1.
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* @param eventHandler A handler to use when delivering events to {@code eventListener}. May be
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* null if delivery of events is not required.
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@ -226,9 +140,10 @@ public class MediaCodecAudioTrackRenderer extends MediaCodecTrackRenderer {
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* begin in parallel with key acquisision. This parameter specifies whether the renderer is
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* permitted to play clear regions of encrypted media files before {@code drmSessionManager}
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* has obtained the keys necessary to decrypt encrypted regions of the media.
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* @param minBufferMultiplicationFactor When instantiating an underlying {@link AudioTrack},
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* the size of the track's is calculated as this value multiplied by the minimum buffer size
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* obtained from {@link AudioTrack#getMinBufferSize(int, int, int)}. The multiplication
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* @param minBufferMultiplicationFactor When instantiating an underlying
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* {@link android.media.AudioTrack}, the size of the track is calculated as this value
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* multiplied by the minimum buffer size obtained from
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* {@link android.media.AudioTrack#getMinBufferSize(int, int, int)}. The multiplication
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* factor must be greater than or equal to 1.
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* @param eventHandler A handler to use when delivering events to {@code eventListener}. May be
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* null if delivery of events is not required.
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@ -237,25 +152,31 @@ public class MediaCodecAudioTrackRenderer extends MediaCodecTrackRenderer {
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public MediaCodecAudioTrackRenderer(SampleSource source, DrmSessionManager drmSessionManager,
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boolean playClearSamplesWithoutKeys, float minBufferMultiplicationFactor,
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Handler eventHandler, EventListener eventListener) {
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this(source, drmSessionManager, playClearSamplesWithoutKeys, eventHandler, eventListener,
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new AudioTrack(minBufferMultiplicationFactor));
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}
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/**
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* @param source The upstream source from which the renderer obtains samples.
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* @param drmSessionManager For use with encrypted content. May be null if support for encrypted
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* content is not required.
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* @param playClearSamplesWithoutKeys Encrypted media may contain clear (un-encrypted) regions.
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* For example a media file may start with a short clear region so as to allow playback to
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* begin in parallel with key acquisision. This parameter specifies whether the renderer is
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* permitted to play clear regions of encrypted media files before {@code drmSessionManager}
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* has obtained the keys necessary to decrypt encrypted regions of the media.
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* @param eventHandler A handler to use when delivering events to {@code eventListener}. May be
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* null if delivery of events is not required.
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* @param eventListener A listener of events. May be null if delivery of events is not required.
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* @param audioTrack Used for playing back decoded audio samples.
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*/
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public MediaCodecAudioTrackRenderer(SampleSource source, DrmSessionManager drmSessionManager,
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boolean playClearSamplesWithoutKeys, Handler eventHandler, EventListener eventListener,
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AudioTrack audioTrack) {
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super(source, drmSessionManager, playClearSamplesWithoutKeys, eventHandler, eventListener);
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Assertions.checkState(minBufferMultiplicationFactor >= 1);
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this.minBufferMultiplicationFactor = minBufferMultiplicationFactor;
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this.eventListener = eventListener;
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audioTrackReleasingConditionVariable = new ConditionVariable(true);
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if (Util.SDK_INT >= 19) {
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audioTimestampCompat = new AudioTimestampCompatV19();
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} else {
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audioTimestampCompat = new NoopAudioTimestampCompat();
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}
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if (Util.SDK_INT >= 18) {
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try {
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audioTrackGetLatencyMethod = AudioTrack.class.getMethod("getLatency", (Class<?>[]) null);
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} catch (NoSuchMethodException e) {
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// There's no guarantee this method exists. Do nothing.
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}
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}
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playheadOffsets = new long[MAX_PLAYHEAD_OFFSET_COUNT];
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volume = 1.0f;
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this.audioTrack = Assertions.checkNotNull(audioTrack);
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this.audioSessionId = AudioTrack.SESSION_ID_NOT_SET;
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}
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@Override
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@ -271,104 +192,12 @@ public class MediaCodecAudioTrackRenderer extends MediaCodecTrackRenderer {
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@Override
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protected void onEnabled(long positionUs, boolean joining) {
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super.onEnabled(positionUs, joining);
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lastReportedCurrentPositionUs = Long.MIN_VALUE;
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}
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@Override
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protected void doSomeWork(long positionUs, long elapsedRealtimeUs) throws ExoPlaybackException {
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super.doSomeWork(positionUs, elapsedRealtimeUs);
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maybeSampleSyncParams();
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currentPositionUs = Long.MIN_VALUE;
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}
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@Override
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protected void onOutputFormatChanged(MediaFormat format) {
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int channelCount = format.getInteger(MediaFormat.KEY_CHANNEL_COUNT);
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int channelConfig;
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switch (channelCount) {
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case 1:
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channelConfig = AudioFormat.CHANNEL_OUT_MONO;
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break;
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case 2:
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channelConfig = AudioFormat.CHANNEL_OUT_STEREO;
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break;
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case 6:
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channelConfig = AudioFormat.CHANNEL_OUT_5POINT1;
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break;
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case 8:
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channelConfig = AudioFormat.CHANNEL_OUT_7POINT1;
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break;
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default:
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throw new IllegalArgumentException("Unsupported channel count: " + channelCount);
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}
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int sampleRate = format.getInteger(MediaFormat.KEY_SAMPLE_RATE);
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if (audioTrack != null && this.sampleRate == sampleRate
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&& this.channelConfig == channelConfig) {
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// We already have an existing audio track with the correct sample rate and channel config.
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return;
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}
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releaseAudioTrack();
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this.sampleRate = sampleRate;
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this.channelConfig = channelConfig;
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this.minBufferSize = AudioTrack.getMinBufferSize(sampleRate, channelConfig,
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AudioFormat.ENCODING_PCM_16BIT);
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this.bufferSize = (int) (minBufferMultiplicationFactor * minBufferSize);
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this.frameSize = 2 * channelCount; // 2 bytes per 16 bit sample * number of channels.
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}
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private void initAudioTrack() throws ExoPlaybackException {
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// If we're asynchronously releasing a previous audio track then we block until it has been
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// released. This guarantees that we cannot end up in a state where we have multiple audio
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// track instances. Without this guarantee it would be possible, in extreme cases, to exhaust
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// the shared memory that's available for audio track buffers. This would in turn cause the
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// initialization of the audio track to fail.
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audioTrackReleasingConditionVariable.block();
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if (audioSessionId == 0) {
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audioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, sampleRate, channelConfig,
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AudioFormat.ENCODING_PCM_16BIT, bufferSize, AudioTrack.MODE_STREAM);
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checkAudioTrackInitialized();
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audioSessionId = audioTrack.getAudioSessionId();
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onAudioSessionId(audioSessionId);
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} else {
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// Re-attach to the same audio session.
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audioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, sampleRate, channelConfig,
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AudioFormat.ENCODING_PCM_16BIT, bufferSize, AudioTrack.MODE_STREAM, audioSessionId);
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checkAudioTrackInitialized();
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}
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setVolume(volume);
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if (getState() == TrackRenderer.STATE_STARTED) {
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audioTrackResumeSystemTimeUs = System.nanoTime() / 1000;
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audioTrack.play();
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}
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}
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/**
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* Checks that {@link #audioTrack} has been successfully initialized. If it has then calling this
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* method is a no-op. If it hasn't then {@link #audioTrack} is released and set to null, and an
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* exception is thrown.
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*
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* @throws ExoPlaybackException If {@link #audioTrack} has not been successfully initialized.
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*/
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private void checkAudioTrackInitialized() throws ExoPlaybackException {
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int audioTrackState = audioTrack.getState();
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if (audioTrackState == AudioTrack.STATE_INITIALIZED) {
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return;
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}
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// The track is not successfully initialized. Release and null the track.
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try {
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audioTrack.release();
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} catch (Exception e) {
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// The track has already failed to initialize, so it wouldn't be that surprising if release
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// were to fail too. Swallow the exception.
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} finally {
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audioTrack = null;
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}
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// Propagate the relevant exceptions.
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AudioTrackInitializationException exception = new AudioTrackInitializationException(
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audioTrackState, sampleRate, channelConfig, bufferSize);
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notifyAudioTrackInitializationError(exception);
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throw new ExoPlaybackException(exception);
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audioTrack.reconfigure(format);
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}
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/**
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@ -387,51 +216,15 @@ public class MediaCodecAudioTrackRenderer extends MediaCodecTrackRenderer {
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// Do nothing.
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}
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private void releaseAudioTrack() {
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if (audioTrack != null) {
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submittedBytes = 0;
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temporaryBufferSize = 0;
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lastRawPlaybackHeadPosition = 0;
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rawPlaybackHeadWrapCount = 0;
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audioTrackStartMediaTimeUs = 0;
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audioTrackStartMediaTimeState = START_NOT_SET;
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resetSyncParams();
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int playState = audioTrack.getPlayState();
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if (playState == AudioTrack.PLAYSTATE_PLAYING) {
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audioTrack.pause();
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}
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// AudioTrack.release can take some time, so we call it on a background thread.
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final AudioTrack toRelease = audioTrack;
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audioTrack = null;
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audioTrackReleasingConditionVariable.close();
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new Thread() {
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@Override
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public void run() {
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try {
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toRelease.release();
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} finally {
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audioTrackReleasingConditionVariable.open();
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}
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}
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}.start();
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}
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}
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@Override
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protected void onStarted() {
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super.onStarted();
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if (audioTrack != null) {
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audioTrackResumeSystemTimeUs = System.nanoTime() / 1000;
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audioTrack.play();
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}
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audioTrack.play();
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}
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@Override
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protected void onStopped() {
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if (audioTrack != null) {
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resetSyncParams();
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audioTrack.pause();
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}
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audioTrack.pause();
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super.onStopped();
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}
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|
|
@ -439,149 +232,34 @@ public class MediaCodecAudioTrackRenderer extends MediaCodecTrackRenderer {
|
|||
protected boolean isEnded() {
|
||||
// We've exhausted the output stream, and the AudioTrack has either played all of the data
|
||||
// submitted, or has been fed insufficient data to begin playback.
|
||||
return super.isEnded() && (getPendingFrameCount() == 0 || submittedBytes < minBufferSize);
|
||||
return super.isEnded() && (!audioTrack.hasPendingData()
|
||||
|| !audioTrack.hasEnoughDataToBeginPlayback());
|
||||
}
|
||||
|
||||
@Override
|
||||
protected boolean isReady() {
|
||||
return getPendingFrameCount() > 0
|
||||
return audioTrack.hasPendingData()
|
||||
|| (super.isReady() && getSourceState() == SOURCE_STATE_READY_READ_MAY_FAIL);
|
||||
}
|
||||
|
||||
/**
|
||||
* This method uses a variety of techniques to compute the current position:
|
||||
*
|
||||
* 1. Prior to playback having started, calls up to the super class to obtain the pending seek
|
||||
* position.
|
||||
* 2. During playback, uses AudioTimestamps obtained from AudioTrack.getTimestamp on supported
|
||||
* devices.
|
||||
* 3. Else, derives a smoothed position by sampling the AudioTrack's frame position.
|
||||
*/
|
||||
@Override
|
||||
protected long getCurrentPositionUs() {
|
||||
long systemClockUs = System.nanoTime() / 1000;
|
||||
long currentPositionUs;
|
||||
if (audioTrack == null || audioTrackStartMediaTimeState == START_NOT_SET) {
|
||||
// The AudioTrack hasn't started.
|
||||
currentPositionUs = super.getCurrentPositionUs();
|
||||
} else if (audioTimestampSet) {
|
||||
// How long ago in the past the audio timestamp is (negative if it's in the future)
|
||||
long presentationDiff = systemClockUs - (audioTimestampCompat.getNanoTime() / 1000);
|
||||
long framesDiff = durationUsToFrames(presentationDiff);
|
||||
// The position of the frame that's currently being presented.
|
||||
long currentFramePosition = audioTimestampCompat.getFramePosition() + framesDiff;
|
||||
currentPositionUs = framesToDurationUs(currentFramePosition) + audioTrackStartMediaTimeUs;
|
||||
long audioTrackCurrentPositionUs = audioTrack.getCurrentPositionUs(isEnded());
|
||||
if (audioTrackCurrentPositionUs == AudioTrack.CURRENT_POSITION_NOT_SET) {
|
||||
// Use the super class position before audio playback starts.
|
||||
currentPositionUs = Math.max(currentPositionUs, super.getCurrentPositionUs());
|
||||
} else {
|
||||
if (playheadOffsetCount == 0) {
|
||||
// The AudioTrack has started, but we don't have any samples to compute a smoothed position.
|
||||
currentPositionUs = getPlayheadPositionUs() + audioTrackStartMediaTimeUs;
|
||||
} else {
|
||||
// getPlayheadPositionUs() only has a granularity of ~20ms, so we base the position off the
|
||||
// system clock (and a smoothed offset between it and the playhead position) so as to
|
||||
// prevent jitter in the reported positions.
|
||||
currentPositionUs = systemClockUs + smoothedPlayheadOffsetUs + audioTrackStartMediaTimeUs;
|
||||
}
|
||||
if (!isEnded()) {
|
||||
currentPositionUs -= audioTrackLatencyUs;
|
||||
}
|
||||
// Make sure we don't ever report time moving backwards.
|
||||
currentPositionUs = Math.max(currentPositionUs, audioTrackCurrentPositionUs);
|
||||
}
|
||||
// Make sure we don't ever report time moving backwards as a result of smoothing or switching
|
||||
// between the various code paths above.
|
||||
currentPositionUs = Math.max(lastReportedCurrentPositionUs, currentPositionUs);
|
||||
lastReportedCurrentPositionUs = currentPositionUs;
|
||||
return currentPositionUs;
|
||||
}
|
||||
|
||||
private void maybeSampleSyncParams() {
|
||||
if (audioTrack == null || audioTrackStartMediaTimeState == START_NOT_SET
|
||||
|| getState() != STATE_STARTED) {
|
||||
// The AudioTrack isn't playing.
|
||||
return;
|
||||
}
|
||||
long playheadPositionUs = getPlayheadPositionUs();
|
||||
if (playheadPositionUs == 0) {
|
||||
// The AudioTrack hasn't output anything yet.
|
||||
return;
|
||||
}
|
||||
long systemClockUs = System.nanoTime() / 1000;
|
||||
if (systemClockUs - lastPlayheadSampleTimeUs >= MIN_PLAYHEAD_OFFSET_SAMPLE_INTERVAL_US) {
|
||||
// Take a new sample and update the smoothed offset between the system clock and the playhead.
|
||||
playheadOffsets[nextPlayheadOffsetIndex] = playheadPositionUs - systemClockUs;
|
||||
nextPlayheadOffsetIndex = (nextPlayheadOffsetIndex + 1) % MAX_PLAYHEAD_OFFSET_COUNT;
|
||||
if (playheadOffsetCount < MAX_PLAYHEAD_OFFSET_COUNT) {
|
||||
playheadOffsetCount++;
|
||||
}
|
||||
lastPlayheadSampleTimeUs = systemClockUs;
|
||||
smoothedPlayheadOffsetUs = 0;
|
||||
for (int i = 0; i < playheadOffsetCount; i++) {
|
||||
smoothedPlayheadOffsetUs += playheadOffsets[i] / playheadOffsetCount;
|
||||
}
|
||||
}
|
||||
|
||||
if (systemClockUs - lastTimestampSampleTimeUs >= MIN_TIMESTAMP_SAMPLE_INTERVAL_US) {
|
||||
audioTimestampSet = audioTimestampCompat.update(audioTrack);
|
||||
if (audioTimestampSet) {
|
||||
// Perform sanity checks on the timestamp.
|
||||
long audioTimestampUs = audioTimestampCompat.getNanoTime() / 1000;
|
||||
if (audioTimestampUs < audioTrackResumeSystemTimeUs) {
|
||||
// The timestamp corresponds to a time before the track was most recently resumed.
|
||||
audioTimestampSet = false;
|
||||
} else if (Math.abs(audioTimestampUs - systemClockUs) > MAX_AUDIO_TIMESTAMP_OFFSET_US) {
|
||||
// The timestamp time base is probably wrong.
|
||||
audioTimestampSet = false;
|
||||
Log.w(TAG, "Spurious audio timestamp: " + audioTimestampCompat.getFramePosition() + ", "
|
||||
+ audioTimestampUs + ", " + systemClockUs);
|
||||
}
|
||||
}
|
||||
if (audioTrackGetLatencyMethod != null) {
|
||||
try {
|
||||
// Compute the audio track latency, excluding the latency due to the buffer (leaving
|
||||
// latency due to the mixer and audio hardware driver).
|
||||
audioTrackLatencyUs =
|
||||
(Integer) audioTrackGetLatencyMethod.invoke(audioTrack, (Object[]) null) * 1000L
|
||||
- framesToDurationUs(bufferSize / frameSize);
|
||||
// Sanity check that the latency is non-negative.
|
||||
audioTrackLatencyUs = Math.max(audioTrackLatencyUs, 0);
|
||||
// Sanity check that the latency isn't too large.
|
||||
if (audioTrackLatencyUs > MAX_AUDIO_TRACK_LATENCY_US) {
|
||||
Log.w(TAG, "Ignoring impossibly large audio latency: " + audioTrackLatencyUs);
|
||||
audioTrackLatencyUs = 0;
|
||||
}
|
||||
} catch (Exception e) {
|
||||
// The method existed, but doesn't work. Don't try again.
|
||||
audioTrackGetLatencyMethod = null;
|
||||
}
|
||||
}
|
||||
lastTimestampSampleTimeUs = systemClockUs;
|
||||
}
|
||||
}
|
||||
|
||||
private void resetSyncParams() {
|
||||
smoothedPlayheadOffsetUs = 0;
|
||||
playheadOffsetCount = 0;
|
||||
nextPlayheadOffsetIndex = 0;
|
||||
lastPlayheadSampleTimeUs = 0;
|
||||
audioTimestampSet = false;
|
||||
lastTimestampSampleTimeUs = 0;
|
||||
}
|
||||
|
||||
private long getPlayheadPositionUs() {
|
||||
return framesToDurationUs(getPlaybackHeadPosition());
|
||||
}
|
||||
|
||||
private long framesToDurationUs(long frameCount) {
|
||||
return (frameCount * MICROS_PER_SECOND) / sampleRate;
|
||||
}
|
||||
|
||||
private long durationUsToFrames(long durationUs) {
|
||||
return (durationUs * sampleRate) / MICROS_PER_SECOND;
|
||||
}
|
||||
|
||||
@Override
|
||||
protected void onDisabled() {
|
||||
audioSessionId = 0;
|
||||
audioSessionId = AudioTrack.SESSION_ID_NOT_SET;
|
||||
try {
|
||||
releaseAudioTrack();
|
||||
audioTrack.reset();
|
||||
} finally {
|
||||
super.onDisabled();
|
||||
}
|
||||
|
|
@ -591,8 +269,8 @@ public class MediaCodecAudioTrackRenderer extends MediaCodecTrackRenderer {
|
|||
protected void seekTo(long positionUs) throws ExoPlaybackException {
|
||||
super.seekTo(positionUs);
|
||||
// TODO: Try and re-use the same AudioTrack instance once [redacted] is fixed.
|
||||
releaseAudioTrack();
|
||||
lastReportedCurrentPositionUs = Long.MIN_VALUE;
|
||||
audioTrack.reset();
|
||||
currentPositionUs = Long.MIN_VALUE;
|
||||
}
|
||||
|
||||
@Override
|
||||
|
|
@ -602,74 +280,39 @@ public class MediaCodecAudioTrackRenderer extends MediaCodecTrackRenderer {
|
|||
if (shouldSkip) {
|
||||
codec.releaseOutputBuffer(bufferIndex, false);
|
||||
codecCounters.skippedOutputBufferCount++;
|
||||
if (audioTrackStartMediaTimeState == START_IN_SYNC) {
|
||||
// Skipping the sample will push track time out of sync. We'll need to sync again.
|
||||
audioTrackStartMediaTimeState = START_NEED_SYNC;
|
||||
}
|
||||
audioTrack.handleDiscontinuity();
|
||||
return true;
|
||||
}
|
||||
|
||||
if (temporaryBufferSize == 0) {
|
||||
// This is the first time we've seen this {@code buffer}.
|
||||
// Note: presentationTimeUs corresponds to the end of the sample, not the start.
|
||||
long bufferStartTime = bufferInfo.presentationTimeUs
|
||||
- framesToDurationUs(bufferInfo.size / frameSize);
|
||||
if (audioTrackStartMediaTimeState == START_NOT_SET) {
|
||||
audioTrackStartMediaTimeUs = Math.max(0, bufferStartTime);
|
||||
audioTrackStartMediaTimeState = START_IN_SYNC;
|
||||
} else {
|
||||
// Sanity check that bufferStartTime is consistent with the expected value.
|
||||
long expectedBufferStartTime = audioTrackStartMediaTimeUs
|
||||
+ framesToDurationUs(submittedBytes / frameSize);
|
||||
if (audioTrackStartMediaTimeState == START_IN_SYNC
|
||||
&& Math.abs(expectedBufferStartTime - bufferStartTime) > 200000) {
|
||||
Log.e(TAG, "Discontinuity detected [expected " + expectedBufferStartTime + ", got "
|
||||
+ bufferStartTime + "]");
|
||||
audioTrackStartMediaTimeState = START_NEED_SYNC;
|
||||
}
|
||||
if (audioTrackStartMediaTimeState == START_NEED_SYNC) {
|
||||
// Adjust audioTrackStartMediaTimeUs to be consistent with the current buffer's start
|
||||
// time and the number of bytes submitted. Also reset lastReportedCurrentPositionUs to
|
||||
// allow time to jump backwards if it really wants to.
|
||||
audioTrackStartMediaTimeUs += (bufferStartTime - expectedBufferStartTime);
|
||||
audioTrackStartMediaTimeState = START_IN_SYNC;
|
||||
lastReportedCurrentPositionUs = Long.MIN_VALUE;
|
||||
// Initialize and start the audio track now.
|
||||
if (!audioTrack.isInitialized()) {
|
||||
try {
|
||||
if (audioSessionId != AudioTrack.SESSION_ID_NOT_SET) {
|
||||
audioTrack.initialize(audioSessionId);
|
||||
} else {
|
||||
audioSessionId = audioTrack.initialize();
|
||||
onAudioSessionId(audioSessionId);
|
||||
}
|
||||
} catch (AudioTrack.InitializationException e) {
|
||||
notifyAudioTrackInitializationError(e);
|
||||
throw new ExoPlaybackException(e);
|
||||
}
|
||||
|
||||
temporaryBufferSize = bufferInfo.size;
|
||||
buffer.position(bufferInfo.offset);
|
||||
if (Util.SDK_INT < 21) {
|
||||
// Copy {@code buffer} into {@code temporaryBuffer}.
|
||||
if (temporaryBuffer == null || temporaryBuffer.length < bufferInfo.size) {
|
||||
temporaryBuffer = new byte[bufferInfo.size];
|
||||
}
|
||||
buffer.get(temporaryBuffer, 0, bufferInfo.size);
|
||||
temporaryBufferOffset = 0;
|
||||
if (getState() == TrackRenderer.STATE_STARTED) {
|
||||
audioTrack.play();
|
||||
}
|
||||
}
|
||||
|
||||
if (audioTrack == null) {
|
||||
initAudioTrack();
|
||||
int handleBufferResult = audioTrack.handleBuffer(
|
||||
buffer, bufferInfo.offset, bufferInfo.size, bufferInfo.presentationTimeUs);
|
||||
|
||||
// If we are out of sync, allow currentPositionUs to jump backwards.
|
||||
if ((handleBufferResult & AudioTrack.RESULT_POSITION_DISCONTINUITY) != 0) {
|
||||
currentPositionUs = Long.MIN_VALUE;
|
||||
}
|
||||
|
||||
int bytesWritten = 0;
|
||||
if (Util.SDK_INT < 21) {
|
||||
// Work out how many bytes we can write without the risk of blocking.
|
||||
int bytesPending = (int) (submittedBytes - getPlaybackHeadPosition() * frameSize);
|
||||
int bytesToWrite = bufferSize - bytesPending;
|
||||
if (bytesToWrite > 0) {
|
||||
bytesToWrite = Math.min(temporaryBufferSize, bytesToWrite);
|
||||
bytesWritten = audioTrack.write(temporaryBuffer, temporaryBufferOffset, bytesToWrite);
|
||||
temporaryBufferOffset += bytesWritten;
|
||||
}
|
||||
} else {
|
||||
bytesWritten = writeNonBlockingV21(audioTrack, buffer, temporaryBufferSize);
|
||||
}
|
||||
|
||||
temporaryBufferSize -= bytesWritten;
|
||||
submittedBytes += bytesWritten;
|
||||
if (temporaryBufferSize == 0) {
|
||||
// Release the buffer if it was consumed.
|
||||
if ((handleBufferResult & AudioTrack.RESULT_BUFFER_CONSUMED) != 0) {
|
||||
codec.releaseOutputBuffer(bufferIndex, false);
|
||||
codecCounters.renderedOutputBufferCount++;
|
||||
return true;
|
||||
|
|
@ -678,66 +321,16 @@ public class MediaCodecAudioTrackRenderer extends MediaCodecTrackRenderer {
|
|||
return false;
|
||||
}
|
||||
|
||||
@TargetApi(21)
|
||||
private static int writeNonBlockingV21(AudioTrack audioTrack, ByteBuffer buffer, int size) {
|
||||
return audioTrack.write(buffer, size, AudioTrack.WRITE_NON_BLOCKING);
|
||||
}
|
||||
|
||||
/**
|
||||
* {@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 {@link AudioTrack#getPlaybackHeadPosition()} of {@link #audioTrack} expressed as a
|
||||
* long.
|
||||
*/
|
||||
private long getPlaybackHeadPosition() {
|
||||
long rawPlaybackHeadPosition = 0xFFFFFFFFL & audioTrack.getPlaybackHeadPosition();
|
||||
if (lastRawPlaybackHeadPosition > rawPlaybackHeadPosition) {
|
||||
// The value must have wrapped around.
|
||||
rawPlaybackHeadWrapCount++;
|
||||
}
|
||||
lastRawPlaybackHeadPosition = rawPlaybackHeadPosition;
|
||||
return rawPlaybackHeadPosition + (rawPlaybackHeadWrapCount << 32);
|
||||
}
|
||||
|
||||
private int getPendingFrameCount() {
|
||||
return audioTrack == null
|
||||
? 0 : (int) (submittedBytes / frameSize - getPlaybackHeadPosition());
|
||||
}
|
||||
|
||||
@Override
|
||||
public void handleMessage(int messageType, Object message) throws ExoPlaybackException {
|
||||
if (messageType == MSG_SET_VOLUME) {
|
||||
setVolume((Float) message);
|
||||
audioTrack.setVolume((Float) message);
|
||||
} else {
|
||||
super.handleMessage(messageType, message);
|
||||
}
|
||||
}
|
||||
|
||||
private void setVolume(float volume) {
|
||||
this.volume = volume;
|
||||
if (audioTrack != null) {
|
||||
if (Util.SDK_INT >= 21) {
|
||||
setVolumeV21(audioTrack, volume);
|
||||
} else {
|
||||
setVolumeV3(audioTrack, volume);
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
@TargetApi(21)
|
||||
private static void setVolumeV21(AudioTrack audioTrack, float volume) {
|
||||
audioTrack.setVolume(volume);
|
||||
}
|
||||
|
||||
@SuppressWarnings("deprecation")
|
||||
private static void setVolumeV3(AudioTrack audioTrack, float volume) {
|
||||
audioTrack.setStereoVolume(volume, volume);
|
||||
}
|
||||
|
||||
private void notifyAudioTrackInitializationError(final AudioTrackInitializationException e) {
|
||||
private void notifyAudioTrackInitializationError(final AudioTrack.InitializationException e) {
|
||||
if (eventHandler != null && eventListener != null) {
|
||||
eventHandler.post(new Runnable() {
|
||||
@Override
|
||||
|
|
@ -748,74 +341,4 @@ public class MediaCodecAudioTrackRenderer extends MediaCodecTrackRenderer {
|
|||
}
|
||||
}
|
||||
|
||||
/**
|
||||
* Interface exposing the {@link AudioTimestamp} methods we need that were added in SDK 19.
|
||||
*/
|
||||
private interface AudioTimestampCompat {
|
||||
|
||||
/**
|
||||
* Returns true if the audioTimestamp was retrieved from the audioTrack.
|
||||
*/
|
||||
boolean update(AudioTrack audioTrack);
|
||||
|
||||
long getNanoTime();
|
||||
|
||||
long getFramePosition();
|
||||
|
||||
}
|
||||
|
||||
/**
|
||||
* The AudioTimestampCompat implementation for SDK < 19 that does nothing or throws an exception.
|
||||
*/
|
||||
private static final class NoopAudioTimestampCompat implements AudioTimestampCompat {
|
||||
|
||||
@Override
|
||||
public boolean update(AudioTrack audioTrack) {
|
||||
return false;
|
||||
}
|
||||
|
||||
@Override
|
||||
public long getNanoTime() {
|
||||
// Should never be called if initTimestamp() returned false.
|
||||
throw new UnsupportedOperationException();
|
||||
}
|
||||
|
||||
@Override
|
||||
public long getFramePosition() {
|
||||
// Should never be called if initTimestamp() returned false.
|
||||
throw new UnsupportedOperationException();
|
||||
}
|
||||
|
||||
}
|
||||
|
||||
/**
|
||||
* The AudioTimestampCompat implementation for SDK >= 19 that simply calls through to the actual
|
||||
* implementations added in SDK 19.
|
||||
*/
|
||||
@TargetApi(19)
|
||||
private static final class AudioTimestampCompatV19 implements AudioTimestampCompat {
|
||||
|
||||
private final AudioTimestamp audioTimestamp;
|
||||
|
||||
public AudioTimestampCompatV19() {
|
||||
audioTimestamp = new AudioTimestamp();
|
||||
}
|
||||
|
||||
@Override
|
||||
public boolean update(AudioTrack audioTrack) {
|
||||
return audioTrack.getTimestamp(audioTimestamp);
|
||||
}
|
||||
|
||||
@Override
|
||||
public long getNanoTime() {
|
||||
return audioTimestamp.nanoTime;
|
||||
}
|
||||
|
||||
@Override
|
||||
public long getFramePosition() {
|
||||
return audioTimestamp.framePosition;
|
||||
}
|
||||
|
||||
}
|
||||
|
||||
}
|
||||
|
|
|
|||
|
|
@ -0,0 +1,716 @@
|
|||
/*
|
||||
* Copyright (C) 2014 The Android Open Source Project
|
||||
*
|
||||
* Licensed under the Apache License, Version 2.0 (the "License");
|
||||
* you may not use this file except in compliance with the License.
|
||||
* You may obtain a copy of the License at
|
||||
*
|
||||
* http://www.apache.org/licenses/LICENSE-2.0
|
||||
*
|
||||
* Unless required by applicable law or agreed to in writing, software
|
||||
* distributed under the License is distributed on an "AS IS" BASIS,
|
||||
* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
|
||||
* See the License for the specific language governing permissions and
|
||||
* limitations under the License.
|
||||
*/
|
||||
package com.google.android.exoplayer.audio;
|
||||
|
||||
import com.google.android.exoplayer.ExoPlaybackException;
|
||||
import com.google.android.exoplayer.util.Assertions;
|
||||
import com.google.android.exoplayer.util.Util;
|
||||
|
||||
import android.annotation.TargetApi;
|
||||
import android.media.AudioFormat;
|
||||
import android.media.AudioManager;
|
||||
import android.media.AudioTimestamp;
|
||||
import android.media.MediaFormat;
|
||||
import android.os.ConditionVariable;
|
||||
import android.util.Log;
|
||||
|
||||
import java.lang.reflect.Method;
|
||||
import java.nio.ByteBuffer;
|
||||
|
||||
/**
|
||||
* Plays audio data. The implementation delegates to an {@link android.media.AudioTrack} and handles
|
||||
* playback position smoothing, non-blocking writes and reconfiguration.
|
||||
*
|
||||
* <p>If {@link #isInitialized} returns {@code false}, the instance can be {@link #initialize}d.
|
||||
* After initialization, start playback by calling {@link #play}.
|
||||
*
|
||||
* <p>Call {@link #handleBuffer} to write data for playback.
|
||||
*
|
||||
* <p>Call {@link #handleDiscontinuity} when a buffer is skipped.
|
||||
*
|
||||
* <p>Call {@link #reconfigure} when the output format changes.
|
||||
*
|
||||
* <p>Call {@link #reset} to free resources. It is safe to re-{@link #initialize} the instance.
|
||||
*/
|
||||
@TargetApi(16)
|
||||
public final class AudioTrack {
|
||||
|
||||
/**
|
||||
* Thrown when a failure occurs instantiating an {@link android.media.AudioTrack}.
|
||||
*/
|
||||
public static class InitializationException extends Exception {
|
||||
|
||||
/** The state as reported by {@link android.media.AudioTrack#getState()}. */
|
||||
public final int audioTrackState;
|
||||
|
||||
public InitializationException(
|
||||
int audioTrackState, int sampleRate, int channelConfig, int bufferSize) {
|
||||
super("AudioTrack init failed: " + audioTrackState + ", Config(" + sampleRate + ", "
|
||||
+ channelConfig + ", " + bufferSize + ")");
|
||||
this.audioTrackState = audioTrackState;
|
||||
}
|
||||
|
||||
}
|
||||
|
||||
/** Returned in the result of {@link #handleBuffer} if the buffer was discontinuous. */
|
||||
public static final int RESULT_POSITION_DISCONTINUITY = 1;
|
||||
/** Returned in the result of {@link #handleBuffer} if the buffer can be released. */
|
||||
public static final int RESULT_BUFFER_CONSUMED = 2;
|
||||
|
||||
/** Represents an unset {@link android.media.AudioTrack} session identifier. */
|
||||
public static final int SESSION_ID_NOT_SET = 0;
|
||||
|
||||
/** The default multiplication factor used when determining the size of the track's buffer. */
|
||||
public static final float DEFAULT_MIN_BUFFER_MULTIPLICATION_FACTOR = 4;
|
||||
|
||||
/** Returned by {@link #getCurrentPositionUs} when the position is not set. */
|
||||
public static final long CURRENT_POSITION_NOT_SET = Long.MIN_VALUE;
|
||||
|
||||
private static final String TAG = "AudioTrack";
|
||||
|
||||
private static final long MICROS_PER_SECOND = 1000000L;
|
||||
|
||||
/**
|
||||
* AudioTrack timestamps are deemed spurious if they are offset from the system clock by more
|
||||
* than this amount.
|
||||
*
|
||||
* <p>This is a fail safe that should not be required on correctly functioning devices.
|
||||
*/
|
||||
private static final long MAX_AUDIO_TIMESTAMP_OFFSET_US = 10 * MICROS_PER_SECOND;
|
||||
|
||||
/**
|
||||
* AudioTrack latencies are deemed impossibly large if they are greater than this amount.
|
||||
*
|
||||
* <p>This is a fail safe that should not be required on correctly functioning devices.
|
||||
*/
|
||||
private static final long MAX_LATENCY_US = 10 * MICROS_PER_SECOND;
|
||||
|
||||
private static final int START_NOT_SET = 0;
|
||||
private static final int START_IN_SYNC = 1;
|
||||
private static final int START_NEED_SYNC = 2;
|
||||
|
||||
private static final int MAX_PLAYHEAD_OFFSET_COUNT = 10;
|
||||
private static final int MIN_PLAYHEAD_OFFSET_SAMPLE_INTERVAL_US = 30000;
|
||||
private static final int MIN_TIMESTAMP_SAMPLE_INTERVAL_US = 500000;
|
||||
|
||||
private final ConditionVariable releasingConditionVariable;
|
||||
private final AudioTimestampCompat audioTimestampCompat;
|
||||
private final long[] playheadOffsets;
|
||||
private final float minBufferMultiplicationFactor;
|
||||
|
||||
private android.media.AudioTrack audioTrack;
|
||||
private int sampleRate;
|
||||
private int channelConfig;
|
||||
private int encoding;
|
||||
private int frameSize;
|
||||
private int minBufferSize;
|
||||
private int bufferSize;
|
||||
|
||||
private int nextPlayheadOffsetIndex;
|
||||
private int playheadOffsetCount;
|
||||
private long smoothedPlayheadOffsetUs;
|
||||
private long lastPlayheadSampleTimeUs;
|
||||
private boolean audioTimestampSet;
|
||||
private long lastTimestampSampleTimeUs;
|
||||
private long lastRawPlaybackHeadPosition;
|
||||
private long rawPlaybackHeadWrapCount;
|
||||
|
||||
private Method getLatencyMethod;
|
||||
private long submittedBytes;
|
||||
private int startMediaTimeState;
|
||||
private long startMediaTimeUs;
|
||||
private long resumeSystemTimeUs;
|
||||
private long latencyUs;
|
||||
private float volume;
|
||||
|
||||
private byte[] temporaryBuffer;
|
||||
private int temporaryBufferOffset;
|
||||
private int temporaryBufferSize;
|
||||
|
||||
/** Constructs an audio track using the default minimum buffer size multiplier. */
|
||||
public AudioTrack() {
|
||||
this(DEFAULT_MIN_BUFFER_MULTIPLICATION_FACTOR);
|
||||
}
|
||||
|
||||
/** Constructs an audio track using the specified minimum buffer size multiplier. */
|
||||
public AudioTrack(float minBufferMultiplicationFactor) {
|
||||
Assertions.checkArgument(minBufferMultiplicationFactor >= 1);
|
||||
this.minBufferMultiplicationFactor = minBufferMultiplicationFactor;
|
||||
releasingConditionVariable = new ConditionVariable(true);
|
||||
if (Util.SDK_INT >= 19) {
|
||||
audioTimestampCompat = new AudioTimestampCompatV19();
|
||||
} else {
|
||||
audioTimestampCompat = new NoopAudioTimestampCompat();
|
||||
}
|
||||
if (Util.SDK_INT >= 18) {
|
||||
try {
|
||||
getLatencyMethod =
|
||||
android.media.AudioTrack.class.getMethod("getLatency", (Class<?>[]) null);
|
||||
} catch (NoSuchMethodException e) {
|
||||
// There's no guarantee this method exists. Do nothing.
|
||||
}
|
||||
}
|
||||
playheadOffsets = new long[MAX_PLAYHEAD_OFFSET_COUNT];
|
||||
volume = 1.0f;
|
||||
startMediaTimeState = START_NOT_SET;
|
||||
}
|
||||
|
||||
/**
|
||||
* Returns whether the audio track has been successfully initialized via {@link #initialize} and
|
||||
* not yet {@link #reset}.
|
||||
*/
|
||||
public boolean isInitialized() {
|
||||
return audioTrack != null;
|
||||
}
|
||||
|
||||
/**
|
||||
* Returns the playback position in the stream starting at zero, in microseconds, or
|
||||
* {@link #CURRENT_POSITION_NOT_SET} if it is not yet available.
|
||||
*
|
||||
* <p>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.
|
||||
*/
|
||||
public long getCurrentPositionUs(boolean sourceEnded) {
|
||||
if (!hasCurrentPositionUs()) {
|
||||
return CURRENT_POSITION_NOT_SET;
|
||||
}
|
||||
|
||||
long systemClockUs = System.nanoTime() / 1000;
|
||||
long currentPositionUs;
|
||||
maybeSampleSyncParams();
|
||||
if (audioTimestampSet) {
|
||||
// How long ago in the past the audio timestamp is (negative if it's in the future).
|
||||
long presentationDiff = systemClockUs - (audioTimestampCompat.getNanoTime() / 1000);
|
||||
long framesDiff = durationUsToFrames(presentationDiff);
|
||||
// The position of the frame that's currently being presented.
|
||||
long currentFramePosition = audioTimestampCompat.getFramePosition() + framesDiff;
|
||||
currentPositionUs = framesToDurationUs(currentFramePosition) + startMediaTimeUs;
|
||||
} else {
|
||||
if (playheadOffsetCount == 0) {
|
||||
// The AudioTrack has started, but we don't have any samples to compute a smoothed position.
|
||||
currentPositionUs = getPlaybackPositionUs() + startMediaTimeUs;
|
||||
} else {
|
||||
// getPlayheadPositionUs() only has a granularity of ~20ms, so we base the position off the
|
||||
// system clock (and a smoothed offset between it and the playhead position) so as to
|
||||
// prevent jitter in the reported positions.
|
||||
currentPositionUs = systemClockUs + smoothedPlayheadOffsetUs + startMediaTimeUs;
|
||||
}
|
||||
if (!sourceEnded) {
|
||||
currentPositionUs -= latencyUs;
|
||||
}
|
||||
}
|
||||
|
||||
return currentPositionUs;
|
||||
}
|
||||
|
||||
/**
|
||||
* Initializes the audio track for writing new buffers using {@link #handleBuffer}.
|
||||
*
|
||||
* @return The audio track session identifier.
|
||||
*/
|
||||
public int initialize() throws InitializationException {
|
||||
return initialize(SESSION_ID_NOT_SET);
|
||||
}
|
||||
|
||||
/**
|
||||
* Initializes the audio track for writing new buffers using {@link #handleBuffer}.
|
||||
*
|
||||
* @param sessionId Audio track session identifier to re-use, or {@link #SESSION_ID_NOT_SET} to
|
||||
* create a new one.
|
||||
* @return The new (or re-used) session identifier.
|
||||
*/
|
||||
public int initialize(int sessionId) throws InitializationException {
|
||||
// If we're asynchronously releasing a previous audio track then we block until it has been
|
||||
// released. This guarantees that we cannot end up in a state where we have multiple audio
|
||||
// track instances. Without this guarantee it would be possible, in extreme cases, to exhaust
|
||||
// the shared memory that's available for audio track buffers. This would in turn cause the
|
||||
// initialization of the audio track to fail.
|
||||
releasingConditionVariable.block();
|
||||
|
||||
if (sessionId == SESSION_ID_NOT_SET) {
|
||||
audioTrack = new android.media.AudioTrack(AudioManager.STREAM_MUSIC, sampleRate,
|
||||
channelConfig, encoding, bufferSize, android.media.AudioTrack.MODE_STREAM);
|
||||
} else {
|
||||
// Re-attach to the same audio session.
|
||||
audioTrack = new android.media.AudioTrack(AudioManager.STREAM_MUSIC, sampleRate,
|
||||
channelConfig, encoding, bufferSize, android.media.AudioTrack.MODE_STREAM, sessionId);
|
||||
}
|
||||
checkAudioTrackInitialized();
|
||||
setVolume(volume);
|
||||
return audioTrack.getAudioSessionId();
|
||||
}
|
||||
|
||||
/**
|
||||
* Reconfigures the audio track to play back media in {@code format}. The encoding is assumed to
|
||||
* be {@link AudioFormat#ENCODING_PCM_16BIT}.
|
||||
*/
|
||||
public void reconfigure(MediaFormat format) {
|
||||
reconfigure(format, AudioFormat.ENCODING_PCM_16BIT, 0);
|
||||
}
|
||||
|
||||
/**
|
||||
* Reconfigures the audio track to play back media in {@code format}. Buffers passed to
|
||||
* {@link #handleBuffer} must using the specified {@code encoding}, which should be a constant
|
||||
* from {@link AudioFormat}.
|
||||
*
|
||||
* @param format Specifies the channel count and sample rate to play back.
|
||||
* @param encoding The format in which audio is represented.
|
||||
* @param bufferSize The total size of the playback buffer in bytes. Specify 0 to use a buffer
|
||||
* size based on the minimum for format.
|
||||
*/
|
||||
public void reconfigure(MediaFormat format, int encoding, int bufferSize) {
|
||||
int channelCount = format.getInteger(MediaFormat.KEY_CHANNEL_COUNT);
|
||||
int channelConfig;
|
||||
switch (channelCount) {
|
||||
case 1:
|
||||
channelConfig = AudioFormat.CHANNEL_OUT_MONO;
|
||||
break;
|
||||
case 2:
|
||||
channelConfig = AudioFormat.CHANNEL_OUT_STEREO;
|
||||
break;
|
||||
case 6:
|
||||
channelConfig = AudioFormat.CHANNEL_OUT_5POINT1;
|
||||
break;
|
||||
case 8:
|
||||
channelConfig = AudioFormat.CHANNEL_OUT_7POINT1;
|
||||
break;
|
||||
default:
|
||||
throw new IllegalArgumentException("Unsupported channel count: " + channelCount);
|
||||
}
|
||||
|
||||
int sampleRate = format.getInteger(MediaFormat.KEY_SAMPLE_RATE);
|
||||
|
||||
// TODO: Does channelConfig determine channelCount?
|
||||
if (audioTrack != null && this.sampleRate == sampleRate
|
||||
&& this.channelConfig == channelConfig) {
|
||||
// We already have an existing audio track with the correct sample rate and channel config.
|
||||
return;
|
||||
}
|
||||
|
||||
reset();
|
||||
|
||||
minBufferSize = android.media.AudioTrack.getMinBufferSize(sampleRate, channelConfig, encoding);
|
||||
|
||||
this.encoding = encoding;
|
||||
this.bufferSize =
|
||||
bufferSize == 0 ? (int) (minBufferMultiplicationFactor * minBufferSize) : bufferSize;
|
||||
this.sampleRate = sampleRate;
|
||||
this.channelConfig = channelConfig;
|
||||
|
||||
frameSize = 2 * channelCount; // 2 bytes per 16 bit sample * number of channels.
|
||||
}
|
||||
|
||||
/** Starts/resumes playing audio if the audio track has been initialized. */
|
||||
public void play() {
|
||||
if (isInitialized()) {
|
||||
resumeSystemTimeUs = System.nanoTime() / 1000;
|
||||
audioTrack.play();
|
||||
}
|
||||
}
|
||||
|
||||
/** Signals to the audio track that the next buffer is discontinuous with the previous buffer. */
|
||||
public void handleDiscontinuity() {
|
||||
// Force resynchronization after a skipped buffer.
|
||||
if (startMediaTimeState == START_IN_SYNC) {
|
||||
startMediaTimeState = START_NEED_SYNC;
|
||||
}
|
||||
}
|
||||
|
||||
/**
|
||||
* Attempts to write {@code size} bytes from {@code buffer} at {@code offset} to the audio track.
|
||||
* Returns a bit field containing {@link #RESULT_BUFFER_CONSUMED} if the buffer can be released
|
||||
* (due to having been written), and {@link #RESULT_POSITION_DISCONTINUITY} if the buffer was
|
||||
* discontinuous with previously written data.
|
||||
*
|
||||
* @param buffer The buffer containing audio data to play back.
|
||||
* @param offset The offset in the buffer from which to consume data.
|
||||
* @param size The number of bytes to consume from {@code buffer}.
|
||||
* @param presentationTimeUs Presentation timestamp of the next buffer in microseconds.
|
||||
* @return A bit field with {@link #RESULT_BUFFER_CONSUMED} if the buffer can be released, and
|
||||
* {@link #RESULT_POSITION_DISCONTINUITY} if the buffer was not contiguous with previously
|
||||
* written data.
|
||||
*/
|
||||
public int handleBuffer(ByteBuffer buffer, int offset, int size, long presentationTimeUs) {
|
||||
int result = 0;
|
||||
|
||||
if (temporaryBufferSize == 0 && size != 0) {
|
||||
// This is the first time we've seen this {@code buffer}.
|
||||
// Note: presentationTimeUs corresponds to the end of the sample, not the start.
|
||||
long bufferStartTime = presentationTimeUs - framesToDurationUs(bytesToFrames(size));
|
||||
if (startMediaTimeUs == START_NOT_SET) {
|
||||
startMediaTimeUs = Math.max(0, bufferStartTime);
|
||||
startMediaTimeState = START_IN_SYNC;
|
||||
} else {
|
||||
// Sanity check that bufferStartTime is consistent with the expected value.
|
||||
long expectedBufferStartTime = startMediaTimeUs
|
||||
+ framesToDurationUs(bytesToFrames(submittedBytes));
|
||||
if (startMediaTimeState == START_IN_SYNC
|
||||
&& Math.abs(expectedBufferStartTime - bufferStartTime) > 200000) {
|
||||
Log.e(TAG, "Discontinuity detected [expected " + expectedBufferStartTime + ", got "
|
||||
+ bufferStartTime + "]");
|
||||
startMediaTimeState = START_NEED_SYNC;
|
||||
}
|
||||
if (startMediaTimeState == START_NEED_SYNC) {
|
||||
// Adjust startMediaTimeUs to be consistent with the current buffer's start time and the
|
||||
// number of bytes submitted.
|
||||
startMediaTimeUs += (bufferStartTime - expectedBufferStartTime);
|
||||
startMediaTimeState = START_IN_SYNC;
|
||||
result = RESULT_POSITION_DISCONTINUITY;
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
if (size == 0) {
|
||||
return result;
|
||||
}
|
||||
|
||||
if (temporaryBufferSize == 0) {
|
||||
temporaryBufferSize = size;
|
||||
buffer.position(offset);
|
||||
if (Util.SDK_INT < 21) {
|
||||
// Copy {@code buffer} into {@code temporaryBuffer}.
|
||||
if (temporaryBuffer == null || temporaryBuffer.length < size) {
|
||||
temporaryBuffer = new byte[size];
|
||||
}
|
||||
buffer.get(temporaryBuffer, 0, size);
|
||||
temporaryBufferOffset = 0;
|
||||
}
|
||||
}
|
||||
|
||||
int bytesWritten = 0;
|
||||
if (Util.SDK_INT < 21) {
|
||||
// Work out how many bytes we can write without the risk of blocking.
|
||||
int bytesPending = (int) (submittedBytes - framesToBytes(getPlaybackPositionFrames()));
|
||||
int bytesToWrite = bufferSize - bytesPending;
|
||||
if (bytesToWrite > 0) {
|
||||
bytesToWrite = Math.min(temporaryBufferSize, bytesToWrite);
|
||||
bytesWritten = audioTrack.write(temporaryBuffer, temporaryBufferOffset, bytesToWrite);
|
||||
if (bytesWritten < 0) {
|
||||
Log.w(TAG, "AudioTrack.write returned error code: " + bytesWritten);
|
||||
} else {
|
||||
temporaryBufferOffset += bytesWritten;
|
||||
}
|
||||
}
|
||||
} else {
|
||||
bytesWritten = writeNonBlockingV21(audioTrack, buffer, temporaryBufferSize);
|
||||
}
|
||||
|
||||
temporaryBufferSize -= bytesWritten;
|
||||
submittedBytes += bytesWritten;
|
||||
if (temporaryBufferSize == 0) {
|
||||
result |= RESULT_BUFFER_CONSUMED;
|
||||
}
|
||||
|
||||
return result;
|
||||
}
|
||||
|
||||
@TargetApi(21)
|
||||
private static int writeNonBlockingV21(
|
||||
android.media.AudioTrack audioTrack, ByteBuffer buffer, int size) {
|
||||
return audioTrack.write(buffer, size, android.media.AudioTrack.WRITE_NON_BLOCKING);
|
||||
}
|
||||
|
||||
/** Returns whether the audio track has more data pending that will be played back. */
|
||||
public boolean hasPendingData() {
|
||||
return audioTrack != null && bytesToFrames(submittedBytes) > getPlaybackPositionFrames();
|
||||
}
|
||||
|
||||
/** Returns whether enough data has been supplied via {@link #handleBuffer} to begin playback. */
|
||||
public boolean hasEnoughDataToBeginPlayback() {
|
||||
return submittedBytes >= minBufferSize;
|
||||
}
|
||||
|
||||
/** Sets the playback volume. */
|
||||
public void setVolume(float volume) {
|
||||
this.volume = volume;
|
||||
if (audioTrack != null) {
|
||||
if (Util.SDK_INT >= 21) {
|
||||
setVolumeV21(audioTrack, volume);
|
||||
} else {
|
||||
setVolumeV3(audioTrack, volume);
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
@TargetApi(21)
|
||||
private static void setVolumeV21(android.media.AudioTrack audioTrack, float volume) {
|
||||
audioTrack.setVolume(volume);
|
||||
}
|
||||
|
||||
@SuppressWarnings("deprecation")
|
||||
private static void setVolumeV3(android.media.AudioTrack audioTrack, float volume) {
|
||||
audioTrack.setStereoVolume(volume, volume);
|
||||
}
|
||||
|
||||
/** Pauses playback. */
|
||||
public void pause() {
|
||||
if (audioTrack != null) {
|
||||
resetSyncParams();
|
||||
audioTrack.pause();
|
||||
}
|
||||
}
|
||||
|
||||
/**
|
||||
* Releases resources associated with this instance asynchronously. Calling {@link #initialize}
|
||||
* will block until the audio track has been released, so it is safe to initialize immediately
|
||||
* after resetting.
|
||||
*/
|
||||
public void reset() {
|
||||
if (audioTrack != null) {
|
||||
submittedBytes = 0;
|
||||
temporaryBufferSize = 0;
|
||||
lastRawPlaybackHeadPosition = 0;
|
||||
rawPlaybackHeadWrapCount = 0;
|
||||
startMediaTimeUs = START_NOT_SET;
|
||||
resetSyncParams();
|
||||
int playState = audioTrack.getPlayState();
|
||||
if (playState == android.media.AudioTrack.PLAYSTATE_PLAYING) {
|
||||
audioTrack.pause();
|
||||
}
|
||||
// AudioTrack.release can take some time, so we call it on a background thread.
|
||||
final android.media.AudioTrack toRelease = audioTrack;
|
||||
audioTrack = null;
|
||||
releasingConditionVariable.close();
|
||||
new Thread() {
|
||||
@Override
|
||||
public void run() {
|
||||
try {
|
||||
toRelease.release();
|
||||
} finally {
|
||||
releasingConditionVariable.open();
|
||||
}
|
||||
}
|
||||
}.start();
|
||||
}
|
||||
}
|
||||
|
||||
/** Returns whether {@link #getCurrentPositionUs} can return the current playback position. */
|
||||
private boolean hasCurrentPositionUs() {
|
||||
return isInitialized() && startMediaTimeUs != START_NOT_SET;
|
||||
}
|
||||
|
||||
/** Updates the audio track latency and playback position parameters. */
|
||||
private void maybeSampleSyncParams() {
|
||||
if (!hasCurrentPositionUs()) {
|
||||
return;
|
||||
}
|
||||
|
||||
long playbackPositionUs = getPlaybackPositionUs();
|
||||
if (playbackPositionUs == 0) {
|
||||
// The AudioTrack hasn't output anything yet.
|
||||
return;
|
||||
}
|
||||
long systemClockUs = System.nanoTime() / 1000;
|
||||
if (systemClockUs - lastPlayheadSampleTimeUs >= MIN_PLAYHEAD_OFFSET_SAMPLE_INTERVAL_US) {
|
||||
// Take a new sample and update the smoothed offset between the system clock and the playhead.
|
||||
playheadOffsets[nextPlayheadOffsetIndex] = playbackPositionUs - systemClockUs;
|
||||
nextPlayheadOffsetIndex = (nextPlayheadOffsetIndex + 1) % MAX_PLAYHEAD_OFFSET_COUNT;
|
||||
if (playheadOffsetCount < MAX_PLAYHEAD_OFFSET_COUNT) {
|
||||
playheadOffsetCount++;
|
||||
}
|
||||
lastPlayheadSampleTimeUs = systemClockUs;
|
||||
smoothedPlayheadOffsetUs = 0;
|
||||
for (int i = 0; i < playheadOffsetCount; i++) {
|
||||
smoothedPlayheadOffsetUs += playheadOffsets[i] / playheadOffsetCount;
|
||||
}
|
||||
}
|
||||
|
||||
if (systemClockUs - lastTimestampSampleTimeUs >= MIN_TIMESTAMP_SAMPLE_INTERVAL_US) {
|
||||
audioTimestampSet = audioTimestampCompat.update(audioTrack);
|
||||
if (audioTimestampSet) {
|
||||
// Perform sanity checks on the timestamp.
|
||||
long audioTimestampUs = audioTimestampCompat.getNanoTime() / 1000;
|
||||
if (audioTimestampUs < resumeSystemTimeUs) {
|
||||
// The timestamp corresponds to a time before the track was most recently resumed.
|
||||
audioTimestampSet = false;
|
||||
} else if (Math.abs(audioTimestampUs - systemClockUs) > MAX_AUDIO_TIMESTAMP_OFFSET_US) {
|
||||
// The timestamp time base is probably wrong.
|
||||
audioTimestampSet = false;
|
||||
Log.w(TAG, "Spurious audio timestamp: " + audioTimestampCompat.getFramePosition() + ", "
|
||||
+ audioTimestampUs + ", " + systemClockUs);
|
||||
}
|
||||
}
|
||||
if (getLatencyMethod != null) {
|
||||
try {
|
||||
// Compute the audio track latency, excluding the latency due to the buffer (leaving
|
||||
// latency due to the mixer and audio hardware driver).
|
||||
latencyUs = (Integer) getLatencyMethod.invoke(audioTrack, (Object[]) null) * 1000L
|
||||
- framesToDurationUs(bytesToFrames(bufferSize));
|
||||
// Sanity check that the latency is non-negative.
|
||||
latencyUs = Math.max(latencyUs, 0);
|
||||
// Sanity check that the latency isn't too large.
|
||||
if (latencyUs > MAX_LATENCY_US) {
|
||||
Log.w(TAG, "Ignoring impossibly large audio latency: " + latencyUs);
|
||||
latencyUs = 0;
|
||||
}
|
||||
} catch (Exception e) {
|
||||
// The method existed, but doesn't work. Don't try again.
|
||||
getLatencyMethod = null;
|
||||
}
|
||||
}
|
||||
lastTimestampSampleTimeUs = systemClockUs;
|
||||
}
|
||||
}
|
||||
|
||||
/**
|
||||
* Checks that {@link #audioTrack} has been successfully initialized. If it has then calling this
|
||||
* method is a no-op. If it hasn't then {@link #audioTrack} is released and set to null, and an
|
||||
* exception is thrown.
|
||||
*
|
||||
* @throws ExoPlaybackException If {@link #audioTrack} has not been successfully initialized.
|
||||
*/
|
||||
private void checkAudioTrackInitialized() throws InitializationException {
|
||||
int state = audioTrack.getState();
|
||||
if (state == android.media.AudioTrack.STATE_INITIALIZED) {
|
||||
return;
|
||||
}
|
||||
// The track is not successfully initialized. Release and null the track.
|
||||
try {
|
||||
audioTrack.release();
|
||||
} catch (Exception e) {
|
||||
// The track has already failed to initialize, so it wouldn't be that surprising if release
|
||||
// were to fail too. Swallow the exception.
|
||||
} finally {
|
||||
audioTrack = null;
|
||||
}
|
||||
|
||||
throw new InitializationException(state, sampleRate, channelConfig, bufferSize);
|
||||
}
|
||||
|
||||
/**
|
||||
* {@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 Long#MAX_VALUE} (which in practice will never happen).
|
||||
*
|
||||
* @return {@link android.media.AudioTrack#getPlaybackHeadPosition()} of {@link #audioTrack}
|
||||
* expressed as a long.
|
||||
*/
|
||||
private long getPlaybackPositionFrames() {
|
||||
long rawPlaybackHeadPosition = 0xFFFFFFFFL & audioTrack.getPlaybackHeadPosition();
|
||||
if (lastRawPlaybackHeadPosition > rawPlaybackHeadPosition) {
|
||||
// The value must have wrapped around.
|
||||
rawPlaybackHeadWrapCount++;
|
||||
}
|
||||
lastRawPlaybackHeadPosition = rawPlaybackHeadPosition;
|
||||
return rawPlaybackHeadPosition + (rawPlaybackHeadWrapCount << 32);
|
||||
}
|
||||
|
||||
private long getPlaybackPositionUs() {
|
||||
return framesToDurationUs(getPlaybackPositionFrames());
|
||||
}
|
||||
|
||||
private long framesToBytes(long frameCount) {
|
||||
return frameCount * frameSize;
|
||||
}
|
||||
|
||||
private long bytesToFrames(long byteCount) {
|
||||
return byteCount / frameSize;
|
||||
}
|
||||
|
||||
private long framesToDurationUs(long frameCount) {
|
||||
return (frameCount * MICROS_PER_SECOND) / sampleRate;
|
||||
}
|
||||
|
||||
private long durationUsToFrames(long durationUs) {
|
||||
return (durationUs * sampleRate) / MICROS_PER_SECOND;
|
||||
}
|
||||
|
||||
private void resetSyncParams() {
|
||||
smoothedPlayheadOffsetUs = 0;
|
||||
playheadOffsetCount = 0;
|
||||
nextPlayheadOffsetIndex = 0;
|
||||
lastPlayheadSampleTimeUs = 0;
|
||||
audioTimestampSet = false;
|
||||
lastTimestampSampleTimeUs = 0;
|
||||
}
|
||||
|
||||
/**
|
||||
* Interface exposing the {@link android.media.AudioTimestamp} methods we need that were added in
|
||||
* SDK 19.
|
||||
*/
|
||||
private interface AudioTimestampCompat {
|
||||
|
||||
/**
|
||||
* Returns true if the audioTimestamp was retrieved from the audioTrack.
|
||||
*/
|
||||
boolean update(android.media.AudioTrack audioTrack);
|
||||
|
||||
long getNanoTime();
|
||||
|
||||
long getFramePosition();
|
||||
|
||||
}
|
||||
|
||||
/**
|
||||
* The AudioTimestampCompat implementation for SDK < 19 that does nothing or throws an exception.
|
||||
*/
|
||||
private static final class NoopAudioTimestampCompat implements AudioTimestampCompat {
|
||||
|
||||
@Override
|
||||
public boolean update(android.media.AudioTrack audioTrack) {
|
||||
return false;
|
||||
}
|
||||
|
||||
@Override
|
||||
public long getNanoTime() {
|
||||
// Should never be called if initTimestamp() returned false.
|
||||
throw new UnsupportedOperationException();
|
||||
}
|
||||
|
||||
@Override
|
||||
public long getFramePosition() {
|
||||
// Should never be called if initTimestamp() returned false.
|
||||
throw new UnsupportedOperationException();
|
||||
}
|
||||
|
||||
}
|
||||
|
||||
/**
|
||||
* The AudioTimestampCompat implementation for SDK >= 19 that simply calls through to the actual
|
||||
* implementations added in SDK 19.
|
||||
*/
|
||||
@TargetApi(19)
|
||||
private static final class AudioTimestampCompatV19 implements AudioTimestampCompat {
|
||||
|
||||
private final AudioTimestamp audioTimestamp;
|
||||
|
||||
public AudioTimestampCompatV19() {
|
||||
audioTimestamp = new AudioTimestamp();
|
||||
}
|
||||
|
||||
@Override
|
||||
public boolean update(android.media.AudioTrack audioTrack) {
|
||||
return audioTrack.getTimestamp(audioTimestamp);
|
||||
}
|
||||
|
||||
@Override
|
||||
public long getNanoTime() {
|
||||
return audioTimestamp.nanoTime;
|
||||
}
|
||||
|
||||
@Override
|
||||
public long getFramePosition() {
|
||||
return audioTimestamp.framePosition;
|
||||
}
|
||||
|
||||
}
|
||||
|
||||
}
|
||||
Loading…
Reference in a new issue