本文整理匯總了Java中android.media.AudioTrack類的典型用法代碼示例。如果您正苦於以下問題:Java AudioTrack類的具體用法?Java AudioTrack怎麽用?Java AudioTrack使用的例子?那麽, 這裏精選的類代碼示例或許可以為您提供幫助。
AudioTrack類屬於android.media包,在下文中一共展示了AudioTrack類的15個代碼示例,這些例子默認根據受歡迎程度排序。您可以為喜歡或者感覺有用的代碼點讚,您的評價將有助於係統推薦出更棒的Java代碼示例。
示例1: playSound
import android.media.AudioTrack; //導入依賴的package包/類
/**
* This method plays the sound data in the specified buffer.
*
* @param buffer specifies the sound data buffer.
*/
public void playSound(short[] buffer)
{
final String funcName = "playSound";
if (debugEnabled)
{
dbgTrace.traceEnter(funcName, TrcDbgTrace.TraceLevel.API);
dbgTrace.traceExit(funcName, TrcDbgTrace.TraceLevel.API);
}
audioTrack = new AudioTrack(
AudioManager.STREAM_MUSIC,
sampleRate,
AudioFormat.CHANNEL_OUT_MONO,
AudioFormat.ENCODING_PCM_16BIT,
buffer.length*2, //buffer length in bytes
AudioTrack.MODE_STATIC);
audioTrack.write(buffer, 0, buffer.length);
audioTrack.setNotificationMarkerPosition(buffer.length);
audioTrack.setPlaybackPositionUpdateListener(this);
audioTrack.play();
playing = true;
}
示例2: onCreate
import android.media.AudioTrack; //導入依賴的package包/類
@Override
public void onCreate() {
super.onCreate();
mHandler = new Handler();
fetchAccessToken();
int outputBufferSize = AudioTrack.getMinBufferSize(16000,
AudioFormat.CHANNEL_IN_STEREO,
AudioFormat.ENCODING_PCM_16BIT);
try {
mAudioTrack = new AudioTrack(AudioManager.USE_DEFAULT_STREAM_TYPE, 16000, AudioFormat.CHANNEL_OUT_MONO, AudioFormat.ENCODING_PCM_16BIT, outputBufferSize, AudioTrack.MODE_STREAM);
if (Build.VERSION.SDK_INT >= Build.VERSION_CODES.LOLLIPOP) {
mAudioTrack.setVolume(DEFAULT_VOLUME);
}
mAudioTrack.play();
}catch (Exception e){
e.printStackTrace();
}
}
示例3: stop
import android.media.AudioTrack; //導入依賴的package包/類
@Override
public void stop() {
getAudioTrackCurrentPosition();
mCurrentState = PlayState.STOPPED;
if (writeWorkThread != null) {
writeWorkThread.stopWrite();
}
try {
Log.d(TAG, "stop-PlayState:" + mAudioTrack.getPlayState());
if (mAudioTrack != null && mAudioTrack.getPlayState() != AudioTrack.STATE_UNINITIALIZED) {
mAudioTrack.pause();
mAudioTrack.flush();
Log.d(TAG, "stop-ok");
}
} catch (Exception e) {
e.printStackTrace();
Log.d(TAG, "stop()", e);
}
fireStopped();
}
示例4: AudioSink
import android.media.AudioTrack; //導入依賴的package包/類
/**
* Constructor. Will create a new AudioSink.
*
* @param packetSize size of the incoming packets
* @param sampleRate sample rate of the audio signal
*/
public AudioSink (int packetSize, int sampleRate) {
this.packetSize = packetSize;
this.sampleRate = sampleRate;
// Create the queues and fill them with
this.inputQueue = new ArrayBlockingQueue<SamplePacket>(QUEUE_SIZE);
this.outputQueue = new ArrayBlockingQueue<SamplePacket>(QUEUE_SIZE);
for (int i = 0; i < QUEUE_SIZE; i++)
this.outputQueue.offer(new SamplePacket(packetSize));
// Create an instance of the AudioTrack class:
int bufferSize = AudioTrack.getMinBufferSize(sampleRate, AudioFormat.CHANNEL_OUT_MONO, AudioFormat.ENCODING_PCM_16BIT);
this.audioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, sampleRate, AudioFormat.CHANNEL_OUT_MONO,
AudioFormat.ENCODING_PCM_16BIT, bufferSize, AudioTrack.MODE_STREAM);
// Create the audio filters:
this.audioFilter1 = FirFilter.createLowPass(2, 1, 1, 0.1f, 0.15f, 30);
Log.d(LOGTAG,"constructor: created audio filter 1 with " + audioFilter1.getNumberOfTaps() + " Taps.");
this.audioFilter2 = FirFilter.createLowPass(4, 1, 1, 0.1f, 0.1f, 30);
Log.d(LOGTAG,"constructor: created audio filter 2 with " + audioFilter2.getNumberOfTaps() + " Taps.");
this.tmpAudioSamples = new SamplePacket(packetSize);
}
示例5: init_
import android.media.AudioTrack; //導入依賴的package包/類
private void init_(boolean eccEnabled) {
mEccEncoder = EccInstanceProvider.getEncoder(eccEnabled);
int minBufferSizeInBytes = AudioTrack.getMinBufferSize(
RATE,
AudioFormat.CHANNEL_OUT_MONO,
AudioFormat.ENCODING_PCM_16BIT);
// 44.1kHz mono 16bit
mAudioTrack = new AudioTrack(
AudioManager.STREAM_MUSIC,
RATE,
AudioFormat.CHANNEL_OUT_MONO,
AudioFormat.ENCODING_PCM_16BIT,
minBufferSizeInBytes,
AudioTrack.MODE_STREAM);
mExecutorService = Executors.newSingleThreadExecutor();
}
示例6: getMinBufferSize
import android.media.AudioTrack; //導入依賴的package包/類
private int getMinBufferSize(int sampleRate, int channelConfig, int audioFormat) {
minBufferSize = AudioTrack.getMinBufferSize(sampleRate, channelConfig, audioFormat);
// 解決異常IllegalArgumentException: Invalid audio buffer size
int channelCount = 1;
switch (channelConfig) {
// AudioFormat.CHANNEL_CONFIGURATION_DEFAULT
case AudioFormat.CHANNEL_OUT_DEFAULT:
case AudioFormat.CHANNEL_OUT_MONO:
case AudioFormat.CHANNEL_CONFIGURATION_MONO:
channelCount = 1;
break;
case AudioFormat.CHANNEL_OUT_STEREO:
case AudioFormat.CHANNEL_CONFIGURATION_STEREO:
channelCount = 2;
break;
default:
channelCount = Integer.bitCount(channelConfig);
}
// 判斷minBufferSize是否在範圍內,如果不在設定默認值為1152
int frameSizeInBytes = channelCount * (audioFormat == AudioFormat.ENCODING_PCM_8BIT ? 1 : 2);
if ((minBufferSize % frameSizeInBytes != 0) || (minBufferSize < 1)) {
minBufferSize = 1152;
}
return minBufferSize;
}
示例7: release
import android.media.AudioTrack; //導入依賴的package包/類
@Override
public void release() {
mCurrentState = PlayState.IDLE;
if (writeWorkThread != null) {
writeWorkThread.stopWrite();
}
try {
Log.d(TAG, "release-PlayState:" + mAudioTrack.getPlayState());
if (mAudioTrack != null && mAudioTrack.getPlayState() != AudioTrack.STATE_UNINITIALIZED) {
mAudioTrack.pause();
mAudioTrack.flush();
mAudioTrack.stop();
mAudioTrack.release();
Log.d(TAG, "release-ok");
}
} catch (Exception e) {
e.printStackTrace();
Log.d(TAG, "release()", e);
}
fireOnRelease();
mediaPlayerListeners.clear();
handlerMain.removeCallbacksAndMessages(null);
}
示例8: PcmPlayer
import android.media.AudioTrack; //導入依賴的package包/類
public PcmPlayer(Context context, Handler handler) {
this.mContext = context;
this.audioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, sampleRate, AudioFormat.CHANNEL_OUT_MONO, AudioFormat.ENCODING_PCM_16BIT, wBufferSize, AudioTrack.MODE_STREAM);
this.handler = handler;
audioTrack.setPlaybackPositionUpdateListener(this, handler);
cacheDir = context.getExternalFilesDir(Environment.DIRECTORY_MUSIC);
}
示例9: onMarkerReached
import android.media.AudioTrack; //導入依賴的package包/類
@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();
}
}
示例10: createAudioTrack
import android.media.AudioTrack; //導入依賴的package包/類
public AudioTrack createAudioTrack(int frameRate) {
int minBufferSizeBytes = AudioTrack.getMinBufferSize(frameRate,
AudioFormat.CHANNEL_OUT_STEREO, AudioFormat.ENCODING_PCM_FLOAT);
Log.i(TAG, "AudioTrack.minBufferSize = " + minBufferSizeBytes
+ " bytes = " + (minBufferSizeBytes / BYTES_PER_FRAME)
+ " frames");
int bufferSize = 8 * minBufferSizeBytes / 8;
int outputBufferSizeFrames = bufferSize / BYTES_PER_FRAME;
Log.i(TAG, "actual bufferSize = " + bufferSize + " bytes = "
+ outputBufferSizeFrames + " frames");
AudioTrack player = new AudioTrack(AudioManager.STREAM_MUSIC,
mFrameRate, AudioFormat.CHANNEL_OUT_STEREO,
AudioFormat.ENCODING_PCM_FLOAT, bufferSize,
AudioTrack.MODE_STREAM);
Log.i(TAG, "created AudioTrack");
return player;
}
示例11: run
import android.media.AudioTrack; //導入依賴的package包/類
@Override
public void run() {
super.run();
isRunning = true;
int buffsize = AudioTrack.getMinBufferSize(sr,
AudioFormat.CHANNEL_OUT_MONO, AudioFormat.ENCODING_PCM_16BIT);
// create an audiotrack object
AudioTrack audioTrack = new AudioTrack(AudioManager.STREAM_MUSIC,
sr, AudioFormat.CHANNEL_OUT_MONO,
AudioFormat.ENCODING_PCM_16BIT, buffsize,
AudioTrack.MODE_STREAM);
short samples[] = new short[buffsize];
int amp = 10000;
double twopi = 8.*Math.atan(1.);
double ph = 0.0;
// start audio
audioTrack.play();
// synthesis loop
while(isRunning){
double fr = tuneFreq;
for(int i=0; i < buffsize; i++){
samples[i] = (short) (amp*Math.sin(ph));
ph += twopi*fr/sr;
}
audioTrack.write(samples, 0, buffsize);
}
audioTrack.stop();
audioTrack.release();
}
示例12: PWave
import android.media.AudioTrack; //導入依賴的package包/類
public PWave(AppRunner appRunner) {
super(appRunner);
appRunner.whatIsRunning.add(this);
// set the buffer size
buffsize = AudioTrack.getMinBufferSize(mSampleRate,
AudioFormat.CHANNEL_OUT_MONO, AudioFormat.ENCODING_PCM_16BIT);
samples = new short[buffsize];
// create an audiotrack object
audioTrack = new AudioTrack(AudioManager.STREAM_MUSIC,
mSampleRate, AudioFormat.CHANNEL_OUT_MONO,
AudioFormat.ENCODING_PCM_16BIT, buffsize,
AudioTrack.MODE_STREAM);
// start audio
audioTrack.play();
}
示例13: startPlayer
import android.media.AudioTrack; //導入依賴的package包/類
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;
}
示例14: handlePrepare
import android.media.AudioTrack; //導入依賴的package包/類
@TargetApi(Build.VERSION_CODES.JELLY_BEAN)
@Override
protected int handlePrepare(MediaExtractor media_extractor) {
int track_index = selectTrack(media_extractor, "audio/");
if (track_index >= 0) {
final MediaFormat format = media_extractor.getTrackFormat(track_index);
mAudioChannels = format.getInteger(MediaFormat.KEY_CHANNEL_COUNT);
mAudioSampleRate = format.getInteger(MediaFormat.KEY_SAMPLE_RATE);
final int min_buf_size = AudioTrack.getMinBufferSize(mAudioSampleRate,
(mAudioChannels == 1 ? AudioFormat.CHANNEL_OUT_MONO : AudioFormat.CHANNEL_OUT_STEREO),
AudioFormat.ENCODING_PCM_16BIT);
final int max_input_size = format.getInteger(MediaFormat.KEY_MAX_INPUT_SIZE);
mAudioInputBufSize = min_buf_size > 0 ? min_buf_size * mAudioChannels * 2 : max_input_size;
if (mAudioInputBufSize > max_input_size) mAudioInputBufSize = max_input_size;
final int frameSizeInBytes = mAudioChannels * 2;
mAudioInputBufSize = (mAudioInputBufSize / frameSizeInBytes) * frameSizeInBytes;
if (DEBUG) Log.v(TAG, String.format("getMinBufferSize=%d,max_input_size=%d,mAudioInputBufSize=%d",min_buf_size, max_input_size, mAudioInputBufSize));
}
return track_index;
}
示例15: generateTrack
import android.media.AudioTrack; //導入依賴的package包/類
public AudioTrack generateTrack(int sampleRate, short[] buf, int len) {
int end = len;
int c = 0;
if (RawSamples.CHANNEL_CONFIG == AudioFormat.CHANNEL_IN_MONO)
c = AudioFormat.CHANNEL_OUT_MONO;
if (RawSamples.CHANNEL_CONFIG == AudioFormat.CHANNEL_IN_STEREO)
c = AudioFormat.CHANNEL_OUT_STEREO;
// old phones bug.
// http://stackoverflow.com/questions/27602492
//
// with MODE_STATIC setNotificationMarkerPosition not called
AudioTrack track = new AudioTrack(AudioManager.STREAM_MUSIC, sampleRate,
c, RawSamples.AUDIO_FORMAT,
len * (Short.SIZE / 8), AudioTrack.MODE_STREAM);
track.write(buf, 0, len);
if (track.setNotificationMarkerPosition(end) != AudioTrack.SUCCESS)
throw new RuntimeException("unable to set marker");
return track;
}