下面列出了怎么用android.media.AudioTrack的API类实例代码及写法，或者点击链接到github查看源代码。
public void initAudio(String vfsPath) throws Exception {
isAudio = new BufferedInputStream(new FileInputStream(vfsPath));
if (useAAC)
{
aac = new AACHelper();
aac.setDecoder(MediaConstants.sAudioSampleRate, MediaConstants.sAudioChannels, MediaConstants.sAudioBitRate);
}
else
{
int minBufferSize = AudioTrack.getMinBufferSize(MediaConstants.sAudioSampleRate,
MediaConstants.sChannelConfigOut, AudioFormat.ENCODING_PCM_16BIT)*8;
at = new AudioTrack(AudioManager.STREAM_MUSIC, MediaConstants.sAudioSampleRate,
MediaConstants.sChannelConfigOut, AudioFormat.ENCODING_PCM_16BIT,
minBufferSize, AudioTrack.MODE_STREAM);
}
}
@Override
public void start() {
minBufferSize =
AudioTrack.getMinBufferSize(
frameRate, AudioFormat.CHANNEL_OUT_STEREO, AudioFormat.ENCODING_PCM_16BIT);
System.out.println("Audio minBufferSize = " + minBufferSize);
bufferSize = (3 * (minBufferSize / 2)) & ~3;
System.out.println("Audio bufferSize = " + bufferSize);
audioTrack =
new AudioTrack(
AudioManager.STREAM_MUSIC,
frameRate,
AudioFormat.CHANNEL_OUT_STEREO,
AudioFormat.ENCODING_PCM_16BIT,
bufferSize,
AudioTrack.MODE_STREAM);
audioTrack.play();
}
@Override
public void onMarkerReached(AudioTrack track) {
Log.i(TAG, "onMarkerReached>>>" + track.getNotificationMarkerPosition());
if (playLock.tryLock()) {
try {
playCondition.signalAll();
} finally {
playLock.unlock();
}
}
Log.i(TAG, "PCM SIZE=" + pcms.size());
if (!pending.get() && pcms.size() == 0) {
play.set(false);
playListener.onCompleted();
}
}
private void initPlayer(){
stopTtsPlayer();
// IMPORTANT: minimum required buffer size for the successful creation of an AudioTrack instance in streaming mode.
int bufferSize = AudioTrack.getMinBufferSize(sampleRate, AudioFormat.CHANNEL_OUT_MONO, AudioFormat.ENCODING_PCM_16BIT);
synchronized (this) {
audioTrack = new AudioTrack(AudioManager.STREAM_MUSIC,
sampleRate,
AudioFormat.CHANNEL_OUT_MONO,
AudioFormat.ENCODING_PCM_16BIT,
bufferSize,
AudioTrack.MODE_STREAM);
if (audioTrack != null)
audioTrack.play();
}
}
private int getDefaultBufferSize() {
if (isInputPcm) {
int minBufferSize =
AudioTrack.getMinBufferSize(outputSampleRate, outputChannelConfig, outputEncoding);
Assertions.checkState(minBufferSize != ERROR_BAD_VALUE);
int multipliedBufferSize = minBufferSize * BUFFER_MULTIPLICATION_FACTOR;
int minAppBufferSize =
(int) durationUsToFrames(MIN_BUFFER_DURATION_US) * outputPcmFrameSize;
int maxAppBufferSize =
(int)
Math.max(
minBufferSize, durationUsToFrames(MAX_BUFFER_DURATION_US) * outputPcmFrameSize);
return Util.constrainValue(multipliedBufferSize, minAppBufferSize, maxAppBufferSize);
} else {
int rate = getMaximumEncodedRateBytesPerSecond(outputEncoding);
if (outputEncoding == C.ENCODING_AC3) {
rate *= AC3_BUFFER_MULTIPLICATION_FACTOR;
}
return (int) (PASSTHROUGH_BUFFER_DURATION_US * rate / C.MICROS_PER_SECOND);
}
}
@CalledByNative
private boolean configureAudio(MediaFormat format, MediaCrypto crypto, int flags,
boolean playAudio) {
try {
mMediaCodec.configure(format, null, crypto, flags);
if (playAudio) {
int sampleRate = format.getInteger(MediaFormat.KEY_SAMPLE_RATE);
int channelCount = format.getInteger(MediaFormat.KEY_CHANNEL_COUNT);
int channelConfig = (channelCount == 1) ? AudioFormat.CHANNEL_OUT_MONO :
AudioFormat.CHANNEL_OUT_STEREO;
// Using 16bit PCM for output. Keep this value in sync with
// kBytesPerAudioOutputSample in media_codec_bridge.cc.
int minBufferSize = AudioTrack.getMinBufferSize(sampleRate, channelConfig,
AudioFormat.ENCODING_PCM_16BIT);
mAudioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, sampleRate, channelConfig,
AudioFormat.ENCODING_PCM_16BIT, minBufferSize, AudioTrack.MODE_STREAM);
}
return true;
} catch (IllegalStateException e) {
Log.e(TAG, "Cannot configure the audio codec " + e.toString());
}
return false;
}
/**
* Sets the {@link AudioTrack} to wrap. Subsequent method calls on this instance relate to this
* track's position, until the next call to {@link #reset()}.
*
* @param audioTrack The audio track to wrap.
* @param outputEncoding The encoding of the audio track.
* @param outputPcmFrameSize For PCM output encodings, the frame size. The value is ignored
* otherwise.
* @param bufferSize The audio track buffer size in bytes.
*/
public void setAudioTrack(
AudioTrack audioTrack,
@C.Encoding int outputEncoding,
int outputPcmFrameSize,
int bufferSize) {
this.audioTrack = audioTrack;
this.outputPcmFrameSize = outputPcmFrameSize;
this.bufferSize = bufferSize;
audioTimestampPoller = new AudioTimestampPoller(audioTrack);
outputSampleRate = audioTrack.getSampleRate();
needsPassthroughWorkarounds = needsPassthroughWorkarounds(outputEncoding);
isOutputPcm = Util.isEncodingLinearPcm(outputEncoding);
bufferSizeUs = isOutputPcm ? framesToDurationUs(bufferSize / outputPcmFrameSize) : C.TIME_UNSET;
lastRawPlaybackHeadPosition = 0;
rawPlaybackHeadWrapCount = 0;
passthroughWorkaroundPauseOffset = 0;
hasData = false;
stopTimestampUs = C.TIME_UNSET;
forceResetWorkaroundTimeMs = C.TIME_UNSET;
latencyUs = 0;
}
public boolean startPlayer(int streamType, int sampleRateInHz, int channelConfig, int audioFormat) {
if (mIsPlayStarted) {
Log.e(TAG, "Player already started !");
return false;
}
mMinBufferSize = AudioTrack.getMinBufferSize(sampleRateInHz,channelConfig,audioFormat);
if (mMinBufferSize == AudioTrack.ERROR_BAD_VALUE) {
Log.e(TAG, "Invalid parameter !");
return false;
}
Log.d(TAG , "getMinBufferSize = "+mMinBufferSize+" bytes !");
mAudioTrack = new AudioTrack(streamType,sampleRateInHz,channelConfig,audioFormat,mMinBufferSize,DEFAULT_PLAY_MODE);
if (mAudioTrack.getState() == AudioTrack.STATE_UNINITIALIZED) {
Log.e(TAG, "AudioTrack initialize fail !");
return false;
}
mIsPlayStarted = true;
Log.d(TAG, "Start audio player success !");
return true;
}
/**
* Stop stream started with @startStream.
*/
public void stopStream() {
if (streaming) {
streaming = false;
stopStreamRtp();
}
if (!recordController.isRecording()) {
if (glInterface != null) {
glInterface.removeMediaCodecSurface();
glInterface.stop();
}
if (videoDecoder != null) videoDecoder.stop();
if (audioDecoder != null) audioDecoder.stop();
if (audioTrackPlayer != null
&& audioTrackPlayer.getPlayState() == AudioTrack.PLAYSTATE_PLAYING) {
audioTrackPlayer.stop();
}
audioTrackPlayer = null;
videoEncoder.stop();
audioEncoder.stop();
recordController.resetFormats();
}
}
@TargetApi(21)
private static AudioTrack createAudioTrackOnLollipopOrHigher(
int sampleRateInHz, int channelConfig, int bufferSizeInBytes) {
Logging.d(TAG, "createAudioTrackOnLollipopOrHigher");
// TODO(henrika): use setPerformanceMode(int) with PERFORMANCE_MODE_LOW_LATENCY to control
// performance when Android O is supported. Add some logging in the mean time.
final int nativeOutputSampleRate =
AudioTrack.getNativeOutputSampleRate(AudioManager.STREAM_VOICE_CALL);
Logging.d(TAG, "nativeOutputSampleRate: " + nativeOutputSampleRate);
if (sampleRateInHz != nativeOutputSampleRate) {
Logging.w(TAG, "Unable to use fast mode since requested sample rate is not native");
}
// Create an audio track where the audio usage is for VoIP and the content type is speech.
return new AudioTrack(new AudioAttributes.Builder()
.setUsage(DEFAULT_USAGE)
.setContentType(AudioAttributes.CONTENT_TYPE_SPEECH)
.build(),
new AudioFormat.Builder()
.setEncoding(AudioFormat.ENCODING_PCM_16BIT)
.setSampleRate(sampleRateInHz)
.setChannelMask(channelConfig)
.build(),
bufferSizeInBytes, AudioTrack.MODE_STREAM, AudioManager.AUDIO_SESSION_ID_GENERATE);
}
private static int writeToAudioTrack(AudioTrack audioTrack, byte[] bytes) {
if (audioTrack.getPlayState() != AudioTrack.PLAYSTATE_PLAYING) {
if (DBG) Log.d(TAG, "AudioTrack not playing, restarting : " + audioTrack.hashCode());
audioTrack.play();
}
int count = 0;
while (count < bytes.length) {
// Note that we don't take bufferCopy.mOffset into account because
// it is guaranteed to be 0.
int written = audioTrack.write(bytes, count, bytes.length);
if (written <= 0) {
break;
}
count += written;
}
return count;
}
private static void setupVolume(AudioTrack audioTrack, float volume, float pan) {
final float vol = clip(volume, 0.0f, 1.0f);
final float panning = clip(pan, -1.0f, 1.0f);
float volLeft = vol;
float volRight = vol;
if (panning > 0.0f) {
volLeft *= (1.0f - panning);
} else if (panning < 0.0f) {
volRight *= (1.0f + panning);
}
if (DBG) Log.d(TAG, "volLeft=" + volLeft + ",volRight=" + volRight);
if (audioTrack.setStereoVolume(volLeft, volRight) != AudioTrack.SUCCESS) {
Log.e(TAG, "Failed to set volume");
}
}
/**
* 指定されたサンプリングレートとチャンネル数で AudioTrack を作成します.
*/
private void createAudioTrack() {
int bufSize = AudioTrack.getMinBufferSize(mSamplingRate,
mChannel == 1 ? AudioFormat.CHANNEL_OUT_MONO : AudioFormat.CHANNEL_OUT_STEREO,
AudioFormat.ENCODING_PCM_16BIT) * 2;
if (DEBUG) {
Log.d(TAG, "OpusTrack::createAudioTrack");
Log.d(TAG, " SamplingRate: " + mSamplingRate);
Log.d(TAG, " Channels: " + mChannel);
Log.d(TAG, " AudioFormat: " + AudioFormat.ENCODING_PCM_16BIT);
Log.d(TAG, " BufSize: " + bufSize);
}
mAudioTrack = new AudioTrack(AudioManager.STREAM_MUSIC,
mSamplingRate,
mChannel == 1 ? AudioFormat.CHANNEL_OUT_MONO : AudioFormat.CHANNEL_OUT_STEREO,
AudioFormat.ENCODING_PCM_16BIT, bufSize,
AudioTrack.MODE_STREAM);
mAudioTrack.play();
}
/**
* Sets the {@link AudioTrack} to wrap. Subsequent method calls on this instance relate to this
* track's position, until the next call to {@link #reset()}.
*
* @param audioTrack The audio track to wrap.
* @param outputEncoding The encoding of the audio track.
* @param outputPcmFrameSize For PCM output encodings, the frame size. The value is ignored
* otherwise.
* @param bufferSize The audio track buffer size in bytes.
*/
public void setAudioTrack(
AudioTrack audioTrack,
@C.Encoding int outputEncoding,
int outputPcmFrameSize,
int bufferSize) {
this.audioTrack = audioTrack;
this.outputPcmFrameSize = outputPcmFrameSize;
this.bufferSize = bufferSize;
audioTimestampPoller = new AudioTimestampPoller(audioTrack);
outputSampleRate = audioTrack.getSampleRate();
needsPassthroughWorkarounds = needsPassthroughWorkarounds(outputEncoding);
isOutputPcm = Util.isEncodingLinearPcm(outputEncoding);
bufferSizeUs = isOutputPcm ? framesToDurationUs(bufferSize / outputPcmFrameSize) : C.TIME_UNSET;
lastRawPlaybackHeadPosition = 0;
rawPlaybackHeadWrapCount = 0;
passthroughWorkaroundPauseOffset = 0;
hasData = false;
stopTimestampUs = C.TIME_UNSET;
forceResetWorkaroundTimeMs = C.TIME_UNSET;
latencyUs = 0;
}
@Override
public boolean stopRenderer() {
Log.d("AUDIO_FOCUS", "Stop Renderer");
rendererLock.lock();
try {
// only stop if we are playing
if (audioTrack.getPlayState() == AudioTrack.PLAYSTATE_PLAYING) {
// stop playout
audioTrack.stop();
}
// flush the buffers
audioTrack.flush();
} catch (Exception e) {
throw new RuntimeException(e.getMessage());
} finally {
// Ensure we always unlock, both for success, exception or error
// return.
isRendering = false;
rendererLock.unlock();
}
unregisterHeadsetReceiver();
unregisterBtReceiver();
return true;
}
@WorkerThread
public synchronized boolean play(byte[] data, int size) {
if (mAudioTrack != null) {
try {
int ret = mAudioTrack.write(data, 0, size);
switch (ret) {
case AudioTrack.ERROR_INVALID_OPERATION:
Log.w(TAG, "play fail: ERROR_INVALID_OPERATION");
return false;
case AudioTrack.ERROR_BAD_VALUE:
Log.w(TAG, "play fail: ERROR_BAD_VALUE");
return false;
case AudioManager.ERROR_DEAD_OBJECT:
Log.w(TAG, "play fail: ERROR_DEAD_OBJECT");
return false;
default:
return true;
}
} catch (IllegalStateException e) {
Log.w(TAG, "play fail: " + e.getMessage());
return false;
}
}
Log.w(TAG, "play fail: null mAudioTrack");
return false;
}
public void initAudio(String vfsPath) throws Exception {
isAudio = new BufferedInputStream(new FileInputStream(vfsPath));
if (useAAC)
{
aac = new AACHelper();
aac.setDecoder(MediaConstants.sAudioSampleRate, MediaConstants.sAudioChannels, MediaConstants.sAudioBitRate);
}
else
{
int minBufferSize = AudioTrack.getMinBufferSize(MediaConstants.sAudioSampleRate,
MediaConstants.sChannelConfigOut, AudioFormat.ENCODING_PCM_16BIT)*8;
at = new AudioTrack(AudioManager.STREAM_MUSIC, MediaConstants.sAudioSampleRate,
MediaConstants.sChannelConfigOut, AudioFormat.ENCODING_PCM_16BIT,
minBufferSize, AudioTrack.MODE_STREAM);
}
}
public AudioThread(int sampleRateInHz, int channel, long streamId, long decoderId, Media media)
{
if (channel == 1)
{
channel_configuration = AudioFormat.CHANNEL_CONFIGURATION_MONO;
} else
{
channel_configuration = AudioFormat.CHANNEL_CONFIGURATION_STEREO;
}
this.mediaStreamId = streamId;
this.decoderId = decoderId;
this.media = media;
int minBufferSize = AudioTrack.getMinBufferSize(sampleRateInHz, channel_configuration, AudioFormat.ENCODING_PCM_16BIT);
if (minBufferSize > audioLength)
{
audioLength = minBufferSize;
}
mAudioBuffer = new byte[audioLength];
mAudio = new AudioTrack(AudioManager.STREAM_MUSIC, sampleRateInHz, channel_configuration, AudioFormat.ENCODING_PCM_16BIT, audioLength, AudioTrack.MODE_STREAM);
}
public void pauseAudioTrack() {
if ((mAudioTrack != null) && (mAudioTrack.getPlayState() == AudioTrack.PLAYSTATE_PLAYING)) {
try {
mAudioTrack.pause();
} catch (IllegalStateException e) {
e.printStackTrace();
}
// release audio track focus
releaseAudioTrackFocus();
}
}
@Override
public boolean onInitRenderer() {
int bytesPerFrame = getRendererFormat().getChannelCount() * (BITS_PER_SAMPLE / 8);
readByteBuffer = ByteBuffer.allocateDirect(bytesPerFrame * (getRendererFormat().getSampleRate() / BUFFERS_PER_SECOND));
int channelConfig = channelCountToConfiguration(getRendererFormat().getChannelCount());
int minBufferSize = AudioRecord.getMinBufferSize(getRendererFormat().getSampleRate(), channelConfig, android.media.AudioFormat.ENCODING_PCM_16BIT);
audioTrack = new AudioTrack(AudioManager.STREAM_VOICE_CALL, getRendererFormat().getSampleRate(), channelConfig,
android.media.AudioFormat.ENCODING_PCM_16BIT, minBufferSize, AudioTrack.MODE_STREAM);
keepAliveRendererRunnable = true;
return true;
}
private void reInitAudioTrack() {
try {
mMusicAudioTrack =
new AudioTrack(AudioManager.STREAM_MUSIC, mPreMediaSampleRate, mPreMediaChannelConfig,
mPreMediaFormate, mPreMinBuffSize, AudioTrack.MODE_STREAM);
} catch (IllegalArgumentException e) {
informMusicPause();
mMusicAudioTrack = null;
e.printStackTrace();
}
}
@CalledByNative
private void playOutputBuffer(byte[] buf) {
if (mAudioTrack != null) {
if (AudioTrack.PLAYSTATE_PLAYING != mAudioTrack.getPlayState()) {
mAudioTrack.play();
}
int size = mAudioTrack.write(buf, 0, buf.length);
if (buf.length != size) {
Log.i(TAG, "Failed to send all data to audio output, expected size: " +
buf.length + ", actual size: " + size);
}
}
}
private static AudioTrack initializeKeepSessionIdAudioTrack(int audioSessionId) {
int sampleRate = 4000; // Equal to private AudioTrack.MIN_SAMPLE_RATE.
int channelConfig = AudioFormat.CHANNEL_OUT_MONO;
@C.PcmEncoding int encoding = C.ENCODING_PCM_16BIT;
int bufferSize = 2; // Use a two byte buffer, as it is not actually used for playback.
return new AudioTrack(C.STREAM_TYPE_DEFAULT, sampleRate, channelConfig, encoding, bufferSize,
MODE_STATIC, audioSessionId);
}
private void audioTrackRelease() {
if (mAudioTrack != null) {
if (mAudioTrack.getState() == AudioTrack.STATE_INITIALIZED)
mAudioTrack.stop();
mAudioTrack.release();
}
mAudioTrack = null;
}
/**
* Creates a new audio timestamp poller.
*
* @param audioTrack The audio track that will provide timestamps, if the platform supports it.
*/
public AudioTimestampPoller(AudioTrack audioTrack) {
if (Util.SDK_INT >= 19) {
audioTimestamp = new AudioTimestampV19(audioTrack);
reset();
} else {
audioTimestamp = null;
updateState(STATE_NO_TIMESTAMP);
}
}
private void audioTrackRelease() {
if (mAudioTrack != null) {
if (mAudioTrack.getState() == AudioTrack.STATE_INITIALIZED)
mAudioTrack.stop();
mAudioTrack.release();
}
mAudioTrack = null;
}
public void write(double[] buffer, int start, int count) {
// Allocate buffer if needed.
if ((floatBuffer == null) || (floatBuffer.length < count)) {
floatBuffer = new float[count];
}
// Convert float samples to shorts.
for (int i = 0; i < count; i++) {
floatBuffer[i] = (float) buffer[i + start];
}
audioTrack.write(floatBuffer, 0, count, AudioTrack.WRITE_BLOCKING);
}
@TargetApi(21)
private static AudioTrack createAudioTrackOnLollipopOrHigher(
int sampleRateInHz, int channelConfig, int bufferSizeInBytes) {
Logging.d(TAG, "createAudioTrackOnLollipopOrHigher");
// TODO(henrika): use setPerformanceMode(int) with PERFORMANCE_MODE_LOW_LATENCY to control
// performance when Android O is supported. Add some logging in the mean time.
final int nativeOutputSampleRate =
AudioTrack.getNativeOutputSampleRate(AudioManager.STREAM_VOICE_CALL);
Logging.d(TAG, "nativeOutputSampleRate: " + nativeOutputSampleRate);
if (sampleRateInHz != nativeOutputSampleRate) {
Logging.w(TAG, "Unable to use fast mode since requested sample rate is not native");
}
if (usageAttribute != DEFAULT_USAGE) {
Logging.w(TAG, "A non default usage attribute is used: " + usageAttribute);
}
// Create an audio track where the audio usage is for VoIP and the content type is speech.
return new AudioTrack(
new AudioAttributes.Builder()
.setUsage(usageAttribute)
.setContentType(AudioAttributes.CONTENT_TYPE_SPEECH)
.build(),
new AudioFormat.Builder()
.setEncoding(AudioFormat.ENCODING_PCM_16BIT)
.setSampleRate(sampleRateInHz)
.setChannelMask(channelConfig)
.build(),
bufferSizeInBytes,
AudioTrack.MODE_STREAM,
AudioManager.AUDIO_SESSION_ID_GENERATE);
}
private void audioTrackRelease() {
if (mAudioTrack != null) {
if (mAudioTrack.getState() == AudioTrack.STATE_INITIALIZED)
mAudioTrack.stop();
mAudioTrack.release();
}
mAudioTrack = null;
}
@SuppressWarnings("unused")
private int StopPlayback() {
_playLock.lock();
try {
// only stop if we are playing
if (_audioTrack.getPlayState() == AudioTrack.PLAYSTATE_PLAYING) {
// stop playout
try {
_audioTrack.stop();
} catch (IllegalStateException e) {
e.printStackTrace();
return -1;
}
// flush the buffers
_audioTrack.flush();
}
// release the object
_audioTrack.release();
_audioTrack = null;
} finally {
// Ensure we always unlock, both for success, exception or error
// return.
_doPlayInit = true;
_playLock.unlock();
}
_isPlaying = false;
return 0;
}