mirror of
https://github.com/samsonjs/media.git
synced 2026-04-27 15:07:40 +00:00
Rm Timescale from AudioTrack. Not required.
This commit is contained in:
parent
72d42fbc9f
commit
2b0f68a0ab
2 changed files with 2 additions and 15 deletions
|
|
@ -68,7 +68,7 @@ public class MediaCodecAudioTrackRenderer extends MediaCodecTrackRenderer implem
|
||||||
private static final String RAW_DECODER_NAME = "OMX.google.raw.decoder";
|
private static final String RAW_DECODER_NAME = "OMX.google.raw.decoder";
|
||||||
|
|
||||||
private final EventListener eventListener;
|
private final EventListener eventListener;
|
||||||
protected final AudioTrack audioTrack;
|
private final AudioTrack audioTrack;
|
||||||
|
|
||||||
private int audioSessionId;
|
private int audioSessionId;
|
||||||
private long currentPositionUs;
|
private long currentPositionUs;
|
||||||
|
|
|
||||||
|
|
@ -145,8 +145,6 @@ public final class AudioTrack {
|
||||||
private static final int MIN_PLAYHEAD_OFFSET_SAMPLE_INTERVAL_US = 30000;
|
private static final int MIN_PLAYHEAD_OFFSET_SAMPLE_INTERVAL_US = 30000;
|
||||||
private static final int MIN_TIMESTAMP_SAMPLE_INTERVAL_US = 500000;
|
private static final int MIN_TIMESTAMP_SAMPLE_INTERVAL_US = 500000;
|
||||||
|
|
||||||
private static final int DEFAULT_TIMESCALE_PERCENT = 100;
|
|
||||||
|
|
||||||
/**
|
/**
|
||||||
* Whether to enable a workaround for an issue where an audio effect does not keep its session
|
* Whether to enable a workaround for an issue where an audio effect does not keep its session
|
||||||
* active across releasing/initializing a new audio track, on platform API version < 21.
|
* active across releasing/initializing a new audio track, on platform API version < 21.
|
||||||
|
|
@ -193,7 +191,6 @@ public final class AudioTrack {
|
||||||
private long resumeSystemTimeUs;
|
private long resumeSystemTimeUs;
|
||||||
private long latencyUs;
|
private long latencyUs;
|
||||||
private float volume;
|
private float volume;
|
||||||
private int timeScalePercent;
|
|
||||||
|
|
||||||
private byte[] temporaryBuffer;
|
private byte[] temporaryBuffer;
|
||||||
private int temporaryBufferOffset;
|
private int temporaryBufferOffset;
|
||||||
|
|
@ -221,7 +218,6 @@ public final class AudioTrack {
|
||||||
}
|
}
|
||||||
playheadOffsets = new long[MAX_PLAYHEAD_OFFSET_COUNT];
|
playheadOffsets = new long[MAX_PLAYHEAD_OFFSET_COUNT];
|
||||||
volume = 1.0f;
|
volume = 1.0f;
|
||||||
timeScalePercent = DEFAULT_TIMESCALE_PERCENT;
|
|
||||||
startMediaTimeState = START_NOT_SET;
|
startMediaTimeState = START_NOT_SET;
|
||||||
}
|
}
|
||||||
|
|
||||||
|
|
@ -481,7 +477,7 @@ public final class AudioTrack {
|
||||||
} else {
|
} else {
|
||||||
// Sanity check that bufferStartTime is consistent with the expected value.
|
// Sanity check that bufferStartTime is consistent with the expected value.
|
||||||
long expectedBufferStartTime = startMediaTimeUs
|
long expectedBufferStartTime = startMediaTimeUs
|
||||||
+ (framesToDurationUs(bytesToFrames(submittedBytes)) * timeScalePercent) / 100;
|
+ framesToDurationUs(bytesToFrames(submittedBytes));
|
||||||
if (startMediaTimeState == START_IN_SYNC
|
if (startMediaTimeState == START_IN_SYNC
|
||||||
&& Math.abs(expectedBufferStartTime - bufferStartTime) > 200000) {
|
&& Math.abs(expectedBufferStartTime - bufferStartTime) > 200000) {
|
||||||
Log.e(TAG, "Discontinuity detected [expected " + expectedBufferStartTime + ", got "
|
Log.e(TAG, "Discontinuity detected [expected " + expectedBufferStartTime + ", got "
|
||||||
|
|
@ -590,15 +586,6 @@ public final class AudioTrack {
|
||||||
}
|
}
|
||||||
}
|
}
|
||||||
|
|
||||||
/**
|
|
||||||
* Updates the timescale percent for reporting current time
|
|
||||||
*
|
|
||||||
* @param timeScalePercent The new percent multiplier
|
|
||||||
*/
|
|
||||||
public void setTimeScalePercent(int timeScalePercent) {
|
|
||||||
this.timeScalePercent = timeScalePercent;
|
|
||||||
}
|
|
||||||
|
|
||||||
/**
|
/**
|
||||||
* Releases the underlying audio track asynchronously. Calling {@link #initialize} will block
|
* Releases the underlying audio track asynchronously. Calling {@link #initialize} will block
|
||||||
* until the audio track has been released, so it is safe to initialize immediately after
|
* until the audio track has been released, so it is safe to initialize immediately after
|
||||||
|
|
|
||||||
Loading…
Reference in a new issue