android.media.AudioTrack#getMinBufferSize ( )源码实例Demo

下面列出了android.media.AudioTrack#getMinBufferSize ( ) 实例代码，或者点击链接到github查看源代码，也可以在右侧发表评论。

源代码1 项目： Telegram 文件： DefaultAudioSink.java

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);
  }
}

源代码2 项目： media-samples 文件： SimpleAudioOutput.java

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;
}

源代码3 项目： media-samples 文件： SimpleAudioOutput.java

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;
}

源代码4 项目： CameraV 文件： AACHelper.java

public boolean setPlayer(int rate)
{
    int bufferSizePlayer = AudioTrack.getMinBufferSize(rate, AudioFormat.CHANNEL_OUT_MONO, audioFormat);
    Log.d("====buffer Size player ", String.valueOf(bufferSizePlayer));

    player= new AudioTrack(AudioManager.STREAM_MUSIC, rate, AudioFormat.CHANNEL_OUT_MONO, audioFormat, bufferSizePlayer, AudioTrack.MODE_STREAM);


    if (player.getState() == AudioTrack.STATE_INITIALIZED)
    {

        return true;
    }
    else
    {
        return false;
    }

}

源代码5 项目： android-MidiScope 文件： SimpleAudioOutput.java

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;
}

源代码6 项目： android-MidiSynth 文件： SimpleAudioOutput.java

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;
}

源代码7 项目： processing-sound 文件： JSynAndroidAudioDeviceManager.java

public void start() {
	this.minBufferSize = AudioTrack.getMinBufferSize(this.frameRate, AudioFormat.CHANNEL_OUT_STEREO,
			AudioFormat.ENCODING_PCM_16BIT);
	this.bufferSize = (3 * (this.minBufferSize / 2)) & ~3;
	this.audioTrack = new AudioTrack.Builder()
			.setAudioAttributes(new AudioAttributes.Builder()
					.setUsage(AudioAttributes.USAGE_MEDIA)
					.setContentType(AudioAttributes.CONTENT_TYPE_MUSIC)
					.build())
			.setAudioFormat(new AudioFormat.Builder()
					.setChannelMask(AudioFormat.CHANNEL_OUT_STEREO)
					.setEncoding(AudioFormat.ENCODING_PCM_16BIT)
					.setSampleRate(this.frameRate)
					.build())
			.setBufferSizeInBytes(this.bufferSize)
			.setTransferMode(AudioTrack.MODE_STREAM)
			.build();
	this.audioTrack.play();
}

源代码8 项目： video-player 文件： MediaPlayer.java

@SuppressLint("NewApi")
  
private int audioTrackInit(int sampleRateInHz, int channels) {
 //  this.sampleRateInHz=sampleRateInHz;
 //  this.channels=channels;
 //   return 0;
	
 audioTrackRelease();
    int channelConfig = channels >= 2 ? AudioFormat.CHANNEL_OUT_STEREO : AudioFormat.CHANNEL_OUT_MONO;
    try {
      mAudioTrackBufferSize = AudioTrack.getMinBufferSize(sampleRateInHz, channelConfig, AudioFormat.ENCODING_PCM_16BIT);
      mAudioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, sampleRateInHz, channelConfig, AudioFormat.ENCODING_PCM_16BIT, mAudioTrackBufferSize, AudioTrack.MODE_STREAM);
    } catch (Exception e) {
      mAudioTrackBufferSize = 0;
      Log.e("audioTrackInit", e);
    }
    return mAudioTrackBufferSize;
  }

源代码9 项目： K-Sonic 文件： Track.java

private void initDevice(int sampleRate, int numChannels) {
    if (isJMono)
        numChannels = 2;
    mLock.lock();
    try {
        final int format = findFormatFromChannels(numChannels);
        final int minSize = AudioTrack.getMinBufferSize(sampleRate, format,
                AudioFormat.ENCODING_PCM_16BIT);
        mTrack = new AudioTrack(AudioManager.STREAM_MUSIC, sampleRate, format,
                AudioFormat.ENCODING_PCM_16BIT, minSize * 4,
                AudioTrack.MODE_STREAM);
        mSonic = new Sonic(sampleRate, numChannels);
    } catch (Exception e) {//IllegalArgumentException
        throw e;
    } finally {
        mLock.unlock();
    }
}

源代码10 项目： Telegram-FOSS 文件： DefaultAudioSink.java

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);
  }
}

源代码11 项目： sctalk 文件： SpeexDecoder.java

private void initializeAndroidAudio(int sampleRate) throws Exception {
    int minBufferSize = AudioTrack.getMinBufferSize(sampleRate,
            AudioFormat.CHANNEL_OUT_MONO, AudioFormat.ENCODING_PCM_16BIT);

    if (minBufferSize < 0) {
        throw new Exception("Failed to get minimum buffer size: "
                + Integer.toString(minBufferSize));
    }

    track = new AudioTrack(AudioManager.STREAM_MUSIC, sampleRate,
            AudioFormat.CHANNEL_OUT_MONO, AudioFormat.ENCODING_PCM_16BIT,
            minBufferSize, AudioTrack.MODE_STREAM);

}

源代码12 项目： jsyn 文件： AndroidAudioForJSyn.java

public void start() {
    Process.setThreadPriority(-5);
    minBufferSize = AudioTrack.getMinBufferSize(frameRate,
            AudioFormat.CHANNEL_OUT_STEREO,
            AudioFormat.ENCODING_PCM_FLOAT);
    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_FLOAT, bufferSize,
            AudioTrack.MODE_STREAM);
    audioTrack.play();
}

源代码13 项目： Vitamio 文件： MediaPlayer.java

private int audioTrackInit(int sampleRateInHz, int channels) {
  audioTrackRelease();
  int channelConfig = channels >= 2 ? AudioFormat.CHANNEL_OUT_STEREO : AudioFormat.CHANNEL_OUT_MONO;
  try {
    mAudioTrackBufferSize = AudioTrack.getMinBufferSize(sampleRateInHz, channelConfig, AudioFormat.ENCODING_PCM_16BIT);
    mAudioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, sampleRateInHz, channelConfig, AudioFormat.ENCODING_PCM_16BIT, mAudioTrackBufferSize, AudioTrack.MODE_STREAM);
  } catch (Exception e) {
    mAudioTrackBufferSize = 0;
    Log.e("audioTrackInit", e);
  }
  return mAudioTrackBufferSize;
}

源代码14 项目： BambooPlayer 文件： MediaPlayer.java

private int audioTrackInit(int sampleRateInHz, int channels) {
  audioTrackRelease();
  int channelConfig = channels >= 2 ? AudioFormat.CHANNEL_OUT_STEREO : AudioFormat.CHANNEL_OUT_MONO;
  try {
    mAudioTrackBufferSize = AudioTrack.getMinBufferSize(sampleRateInHz, channelConfig, AudioFormat.ENCODING_PCM_16BIT);
    mAudioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, sampleRateInHz, channelConfig, AudioFormat.ENCODING_PCM_16BIT, mAudioTrackBufferSize, AudioTrack.MODE_STREAM);
  } catch (Exception e) {
    mAudioTrackBufferSize = 0;
    Log.e("audioTrackInit", e);
  }
  return mAudioTrackBufferSize;
}

源代码15 项目： HPlayer 文件： MediaPlayer.java

public int audioTrackInit() {
//	  Log.e("  ffff mediaplayer audiotrackinit start .  sampleRateInHz:=" + sampleRateInHz + " channels:=" + channels );
	    audioTrackRelease();
	    int channelConfig = channels >= 2 ? AudioFormat.CHANNEL_OUT_STEREO : AudioFormat.CHANNEL_OUT_MONO;
	    try {
	      mAudioTrackBufferSize = AudioTrack.getMinBufferSize(sampleRateInHz, channelConfig, AudioFormat.ENCODING_PCM_16BIT);
	      mAudioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, sampleRateInHz, channelConfig, AudioFormat.ENCODING_PCM_16BIT, mAudioTrackBufferSize, AudioTrack.MODE_STREAM);
	    } catch (Exception e) {
	      mAudioTrackBufferSize = 0;
	      Log.e("audioTrackInit", e);
	    }
	    return mAudioTrackBufferSize;
	  }

源代码16 项目： 365browser 文件： AudioManagerAndroid.java

/**
* Returns the minimum frame size required for audio output.
*
* @param sampleRate sampling rate
* @param channels number of channels
*/
 @CalledByNative
 private static int getMinOutputFrameSize(int sampleRate, int channels) {
     int channelConfig;
     if (channels == 1) {
         channelConfig = AudioFormat.CHANNEL_OUT_MONO;
     } else if (channels == 2) {
         channelConfig = AudioFormat.CHANNEL_OUT_STEREO;
     } else {
         return -1;
     }
     return AudioTrack.getMinBufferSize(
             sampleRate, channelConfig, AudioFormat.ENCODING_PCM_16BIT) / 2 / channels;
 }

源代码17 项目： actor-platform 文件： OpusPlayerActor.java

protected void onPlayMessage(String fileName, float seek) {
    if (state != STATE_NONE) {
        destroyPlayer();
    }
    state = STATE_NONE;
    currentFileName = fileName;

    int res = opusLib.openOpusFile(currentFileName);
    if (res == 0) {
        callback.onError(currentFileName);
        return;
    }

    duration = opusLib.getTotalPcmDuration();
    offset = 0;

    try {
        bufferSize = AudioTrack.getMinBufferSize(48000, AudioFormat.CHANNEL_OUT_MONO, AudioFormat.ENCODING_PCM_16BIT);
        audioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, 48000, AudioFormat.CHANNEL_OUT_MONO, AudioFormat.ENCODING_PCM_16BIT, bufferSize, AudioTrack.MODE_STREAM);
        audioTrack.play();
    } catch (Exception e) {
        e.printStackTrace();
        destroyPlayer();
        callback.onError(currentFileName);
        return;
    }

    state = STATE_STARTED;
    if (seek != 0) {
        opusLib.seekOpusFile(seek);
    }
    callback.onStart(fileName);
    self().send(new Iterate());
}

源代码18 项目： cythara 文件： AndroidAudioPlayer.java

/**
 * Constructs a new AndroidAudioPlayer from an audio format, default buffer size and stream type.
 *
 * @param audioFormat The audio format of the stream that this AndroidAudioPlayer will process.
 *                    This can only be 1 channel, PCM 16 bit.
 * @param bufferSizeInSamples  The requested buffer size in samples.
 * @param streamType  The type of audio stream that the internal AudioTrack should use. For
 *                    example, {@link AudioManager#STREAM_MUSIC}.
 * @throws IllegalArgumentException if audioFormat is not valid or if the requested buffer size is invalid.
 * @see AudioTrack
 */
public AndroidAudioPlayer(TarsosDSPAudioFormat audioFormat, int bufferSizeInSamples, int streamType) {
    if (audioFormat.getChannels() != 1) {
        throw new IllegalArgumentException("TarsosDSP only supports mono audio channel count: " + audioFormat.getChannels());
    }

    // The requested sample rate
    int sampleRate = (int) audioFormat.getSampleRate();

    //The buffer size in bytes is twice the buffer size expressed in samples if 16bit samples are used:
    int bufferSizeInBytes = bufferSizeInSamples * audioFormat.getSampleSizeInBits()/8;

    // From the Android API about getMinBufferSize():
    // The total size (in bytes) of the internal buffer where audio data is read from for playback.
    // If track's creation mode is MODE_STREAM, you can write data into this buffer in chunks less than or equal to this size,
    // and it is typical to use chunks of 1/2 of the total size to permit double-buffering. If the track's creation mode is MODE_STATIC,
    // this is the maximum length sample, or audio clip, that can be played by this instance. See getMinBufferSize(int, int, int) to determine
    // the minimum required buffer size for the successful creation of an AudioTrack instance in streaming mode. Using values smaller
    // than getMinBufferSize() will result in an initialization failure.
    int minBufferSizeInBytes = AudioTrack.getMinBufferSize(sampleRate, AudioFormat.CHANNEL_OUT_MONO,  AudioFormat.ENCODING_PCM_16BIT);
    if(minBufferSizeInBytes > bufferSizeInBytes){
        throw new IllegalArgumentException("The buffer size should be at least " + (minBufferSizeInBytes/(audioFormat.getSampleSizeInBits()/8)) + " (samples) according to  AudioTrack.getMinBufferSize().");
    }

    //http://developer.android.com/reference/android/media/AudioTrack.html#AudioTrack(int, int, int, int, int, int)
    audioTrack = new AudioTrack(streamType, sampleRate, AudioFormat.CHANNEL_OUT_MONO, AudioFormat.ENCODING_PCM_16BIT, bufferSizeInBytes,AudioTrack.MODE_STREAM);

    audioTrack.play();
}

源代码19 项目： CameraV 文件： FullScreenMJPEGPlayerFragment.java

public void initAudio(String vfsPath) throws Exception {

    	isAudio = new BufferedInputStream(new FileInputStream(vfsPath));

	
        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);
        
	     
    }

源代码20 项目： CameraV 文件： FullScreenMJPEGViewFragment.java

public void initAudio(String vfsPath) throws Exception {

    	isAudio = new BufferedInputStream(new FileInputStream(vfsPath));

	
        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);
        
	     
    }

方法所在类

android.media.AudioTrack