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https://github.com/samsonjs/media.git
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Clean up AudioTrack.
This commit is contained in:
Oliver Woodman
parent
a4f1e3ce53
commit
57ee90a96c
3 changed files with 131 additions and 96 deletions
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@ -385,7 +385,7 @@ public final class LibopusAudioTrackRenderer extends SampleSourceTrackRenderer
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int result = readSource(positionUs, formatHolder, null, false);
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if (result == SampleSource.FORMAT_READ) {
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format = formatHolder.format;
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audioTrack.reconfigure(format.getFrameworkMediaFormatV16(), false);
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audioTrack.configure(format.getFrameworkMediaFormatV16(), false);
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return true;
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}
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return false;
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@ -246,7 +246,7 @@ public class MediaCodecAudioTrackRenderer extends MediaCodecTrackRenderer implem
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@Override
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protected void onOutputFormatChanged(android.media.MediaFormat outputFormat) {
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boolean passthrough = passthroughMediaFormat != null;
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audioTrack.reconfigure(passthrough ? passthroughMediaFormat : outputFormat, passthrough);
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audioTrack.configure(passthrough ? passthroughMediaFormat : outputFormat, passthrough);
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}
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/**
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@ -36,19 +36,24 @@ import java.nio.ByteBuffer;
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/**
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* Plays audio data. The implementation delegates to an {@link android.media.AudioTrack} and handles
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* playback position smoothing, non-blocking writes and reconfiguration.
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*
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* <p>If {@link #isInitialized} returns {@code false}, the instance can be {@link #initialize}d.
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* After initialization, start playback by calling {@link #play}.
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*
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* <p>Call {@link #handleBuffer} to write data for playback.
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*
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* <p>Call {@link #handleDiscontinuity} when a buffer is skipped.
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*
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* <p>Call {@link #reconfigure} when the output format changes.
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*
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* <p>Call {@link #reset} to free resources. It is safe to re-{@link #initialize} the instance.
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*
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* <p>Call {@link #release} when the instance will no longer be used.
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* <p>
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* Before starting playback, specify the input audio format by calling one of the {@link #configure}
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* methods and {@link #initialize} the instance, optionally specifying an audio session.
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* <p>
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* Call {@link #handleBuffer(ByteBuffer, int, int, long)} to write data to play back, and
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* {@link #handleDiscontinuity()} when a buffer is skipped. Call {@link #play()} to start playing
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* back written data.
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* <p>
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* Call {@link #configure} again whenever the input format changes. If {@link #isInitialized()}
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* returns false after calling {@link #configure}, it is necessary to re-{@link #initialize} the
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* instance before writing more data.
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* <p>
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* The underlying framework audio track is created by {@link #initialize} and released
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* asynchronously by {@link #reset} (and {@link #configure}, unless the format is unchanged).
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* Reinitialization blocks until releasing the old audio track completes. It is safe to
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* re-{@link #initialize} the instance after calling {@link #reset()}, without reconfiguration.
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* <p>
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* Call {@link #release()} when the instance will no longer be used.
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*/
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@TargetApi(16)
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public final class AudioTrack {
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@ -58,7 +63,9 @@ public final class AudioTrack {
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*/
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public static final class InitializationException extends Exception {
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/** The state as reported by {@link android.media.AudioTrack#getState()}. */
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/**
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* The state as reported by {@link android.media.AudioTrack#getState()}.
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*/
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public final int audioTrackState;
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public InitializationException(
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@ -75,7 +82,9 @@ public final class AudioTrack {
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*/
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public static final class WriteException extends Exception {
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/** The value returned from {@link android.media.AudioTrack#write(byte[], int, int)}. */
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/**
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* The value returned from {@link android.media.AudioTrack#write(byte[], int, int)}.
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*/
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public final int errorCode;
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public WriteException(int errorCode) {
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@ -97,20 +106,32 @@ public final class AudioTrack {
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}
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/** Returned in the result of {@link #handleBuffer} if the buffer was discontinuous. */
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/**
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* Returned in the result of {@link #handleBuffer} if the buffer was discontinuous.
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*/
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public static final int RESULT_POSITION_DISCONTINUITY = 1;
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/** Returned in the result of {@link #handleBuffer} if the buffer can be released. */
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/**
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* Returned in the result of {@link #handleBuffer} if the buffer can be released.
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*/
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public static final int RESULT_BUFFER_CONSUMED = 2;
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/** Represents an unset {@link android.media.AudioTrack} session identifier. */
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/**
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* Represents an unset {@link android.media.AudioTrack} session identifier.
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*/
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public static final int SESSION_ID_NOT_SET = 0;
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/** Returned by {@link #getCurrentPositionUs} when the position is not set. */
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/**
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* Returned by {@link #getCurrentPositionUs} when the position is not set.
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*/
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public static final long CURRENT_POSITION_NOT_SET = Long.MIN_VALUE;
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/** A minimum length for the {@link android.media.AudioTrack} buffer, in microseconds. */
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/**
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* A minimum length for the {@link android.media.AudioTrack} buffer, in microseconds.
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*/
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private static final long MIN_BUFFER_DURATION_US = 250000;
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/** A maximum length for the {@link android.media.AudioTrack} buffer, in microseconds. */
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/**
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* A maximum length for the {@link android.media.AudioTrack} buffer, in microseconds.
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*/
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private static final long MAX_BUFFER_DURATION_US = 750000;
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/**
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* A multiplication factor to apply to the minimum buffer size requested by the underlying
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@ -171,7 +192,9 @@ public final class AudioTrack {
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private final long[] playheadOffsets;
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private final AudioTrackUtil audioTrackUtil;
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/** Used to keep the audio session active on pre-V21 builds (see {@link #initialize()}). */
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/**
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* Used to keep the audio session active on pre-V21 builds (see {@link #initialize()}).
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*/
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private android.media.AudioTrack keepSessionIdAudioTrack;
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private android.media.AudioTrack audioTrack;
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@ -306,6 +329,89 @@ public final class AudioTrack {
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return currentPositionUs;
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}
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/**
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* Configures (or reconfigures) the audio track to play back media in {@code format}, inferring a
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* buffer size from the format.
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*
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* @param format Specifies the channel count and sample rate to play back.
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* @param passthrough Whether to play back using a passthrough encoding.
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*/
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public void configure(MediaFormat format, boolean passthrough) {
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configure(format, passthrough, 0);
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}
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/**
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* Configures (or reconfigures) the audio track to play back media in {@code format}.
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*
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* @param format Specifies the channel count and sample rate to play back.
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* @param passthrough Whether to playback using a passthrough encoding.
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* @param specifiedBufferSize A specific size for the playback buffer in bytes, or 0 to use a
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* size inferred from the format.
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*/
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public void configure(MediaFormat format, boolean passthrough, int specifiedBufferSize) {
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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 3:
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channelConfig = AudioFormat.CHANNEL_OUT_STEREO | AudioFormat.CHANNEL_OUT_FRONT_CENTER;
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break;
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case 4:
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channelConfig = AudioFormat.CHANNEL_OUT_QUAD;
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break;
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case 5:
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channelConfig = AudioFormat.CHANNEL_OUT_QUAD | AudioFormat.CHANNEL_OUT_FRONT_CENTER;
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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 7:
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channelConfig = AudioFormat.CHANNEL_OUT_5POINT1 | AudioFormat.CHANNEL_OUT_BACK_CENTER;
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break;
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case 8:
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channelConfig = C.CHANNEL_OUT_7POINT1_SURROUND;
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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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String mimeType = format.getString(MediaFormat.KEY_MIME);
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int encoding = passthrough ? getEncodingForMimeType(mimeType) : AudioFormat.ENCODING_PCM_16BIT;
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if (isInitialized() && this.sampleRate == sampleRate && this.channelConfig == channelConfig
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&& this.encoding == encoding) {
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// We already have an audio track with the correct sample rate, encoding and channel config.
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return;
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}
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reset();
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this.encoding = encoding;
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this.sampleRate = sampleRate;
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this.channelConfig = channelConfig;
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passthroughBitrate = UNKNOWN_BITRATE;
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frameSize = 2 * channelCount; // 2 bytes per 16 bit sample * number of channels.
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minBufferSize = android.media.AudioTrack.getMinBufferSize(sampleRate, channelConfig, encoding);
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Assertions.checkState(minBufferSize != android.media.AudioTrack.ERROR_BAD_VALUE);
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if (specifiedBufferSize != 0) {
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bufferSize = specifiedBufferSize;
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} else {
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int multipliedBufferSize = minBufferSize * BUFFER_MULTIPLICATION_FACTOR;
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int minAppBufferSize = (int) durationUsToFrames(MIN_BUFFER_DURATION_US) * frameSize;
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int maxAppBufferSize = (int) Math.max(minBufferSize,
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durationUsToFrames(MAX_BUFFER_DURATION_US) * frameSize);
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bufferSize = multipliedBufferSize < minAppBufferSize ? minAppBufferSize
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: multipliedBufferSize > maxAppBufferSize ? maxAppBufferSize
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: multipliedBufferSize;
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}
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bufferSizeUs = framesToDurationUs(bytesToFrames(bufferSize));
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}
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/**
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* Initializes the audio track for writing new buffers using {@link #handleBuffer}.
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*
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@ -366,83 +472,12 @@ public final class AudioTrack {
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return sessionId;
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}
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/**
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* Reconfigures the audio track to play back media in {@code format}, inferring a buffer size from
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* the format.
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*
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* @param format Specifies the channel count and sample rate to play back.
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* @param passthrough Whether to play back using a passthrough encoding.
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*/
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public void reconfigure(MediaFormat format, boolean passthrough) {
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reconfigure(format, passthrough, 0);
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}
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/**
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* Reconfigures the audio track to play back media in {@code format}.
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*
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* @param format Specifies the channel count and sample rate to play back.
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* @param passthrough Whether to playback using a passthrough encoding.
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* @param specifiedBufferSize A specific size for the playback buffer in bytes, or 0 to use a
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* size inferred from the format.
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*/
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public void reconfigure(MediaFormat format, boolean passthrough, int specifiedBufferSize) {
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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 = C.CHANNEL_OUT_7POINT1_SURROUND;
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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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String mimeType = format.getString(MediaFormat.KEY_MIME);
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int encoding = passthrough ? getEncodingForMimeType(mimeType) : AudioFormat.ENCODING_PCM_16BIT;
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if (isInitialized() && this.sampleRate == sampleRate && this.channelConfig == channelConfig
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&& this.encoding == encoding) {
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// We already have an audio track with the correct sample rate, encoding and channel config.
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return;
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}
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reset();
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this.encoding = encoding;
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this.sampleRate = sampleRate;
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this.channelConfig = channelConfig;
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passthroughBitrate = UNKNOWN_BITRATE;
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frameSize = 2 * channelCount; // 2 bytes per 16 bit sample * number of channels.
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minBufferSize = android.media.AudioTrack.getMinBufferSize(sampleRate, channelConfig, encoding);
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Assertions.checkState(minBufferSize != android.media.AudioTrack.ERROR_BAD_VALUE);
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if (specifiedBufferSize != 0) {
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bufferSize = specifiedBufferSize;
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} else {
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int multipliedBufferSize = minBufferSize * BUFFER_MULTIPLICATION_FACTOR;
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int minAppBufferSize = (int) durationUsToFrames(MIN_BUFFER_DURATION_US) * frameSize;
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int maxAppBufferSize = (int) Math.max(minBufferSize,
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durationUsToFrames(MAX_BUFFER_DURATION_US) * frameSize);
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bufferSize = multipliedBufferSize < minAppBufferSize ? minAppBufferSize
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: multipliedBufferSize > maxAppBufferSize ? maxAppBufferSize
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: multipliedBufferSize;
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}
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bufferSizeUs = framesToDurationUs(bytesToFrames(bufferSize));
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}
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/**
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* Returns the size of this {@link AudioTrack}'s buffer in microseconds, given its current
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* configuration.
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* <p>
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* The duration returned from this method may change as a result of calling one of the
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* {@link #reconfigure} methods.
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* {@link #configure} methods.
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*
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* @return The size of the buffer in microseconds.
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*/
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