Java AudioTrack.getMinBufferSize方法代碼示例

import android.media.AudioTrack; //導入方法依賴的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();
    }
}
 

import android.media.AudioTrack; //導入方法依賴的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);
}
 

import android.media.AudioTrack; //導入方法依賴的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();
}
 

import android.media.AudioTrack; //導入方法依賴的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;
}
 

import android.media.AudioTrack; //導入方法依賴的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;
}
 

import android.media.AudioTrack; //導入方法依賴的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();
}
 

import android.media.AudioTrack; //導入方法依賴的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();
}
 

import android.media.AudioTrack; //導入方法依賴的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;
}
 

import android.media.AudioTrack; //導入方法依賴的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;
}
 

import android.media.AudioTrack; //導入方法依賴的package包/類
@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;
    }
 

import android.media.AudioTrack; //導入方法依賴的package包/類
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);
}
 

import android.media.AudioTrack; //導入方法依賴的package包/類
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();
    }
}
 

import android.media.AudioTrack; //導入方法依賴的package包/類
@Override
    protected void prepare() throws IOException {
        if (mState < STATE_PREPARED) {
            MediaFormat format;
            if (mState == STATE_UNINITIALIZED) {
                mTrackIndex = selectTrack();
                if (mTrackIndex < 0) {
                    setState(STATE_NO_TRACK_FOUND);
                    return;
                }
                mExtractor.selectTrack(mTrackIndex);
                format = mExtractor.getTrackFormat(mTrackIndex);
                mSampleRate = format.getInteger(MediaFormat.KEY_SAMPLE_RATE);
                int audioChannels = format.getInteger(MediaFormat.KEY_CHANNEL_COUNT);
                mAudioTrack = new AudioTrack(
                        AudioManager.STREAM_MUSIC,
                        mSampleRate,
                        (audioChannels == 1 ? AudioFormat.CHANNEL_OUT_MONO : AudioFormat.CHANNEL_OUT_STEREO),
                        AudioFormat.ENCODING_PCM_16BIT,
                        AudioTrack.getMinBufferSize(
                                mSampleRate,
                                (audioChannels == 1 ? AudioFormat.CHANNEL_OUT_MONO : AudioFormat.CHANNEL_OUT_STEREO),
                                AudioFormat.ENCODING_PCM_16BIT
                        ),
                        AudioTrack.MODE_STREAM
                );
                mState = STATE_INITIALIZED;
            } else {
                format = mExtractor.getTrackFormat(mTrackIndex);
            }

            String mime = format.getString(MediaFormat.KEY_MIME);
            Log.d(TAG, mime);
            mMediaCodec = MediaCodec.createDecoderByType(mime);
//            mMediaCodec.setCallback(mCallback);
            mMediaCodec.configure(format, null, null, 0);
            setState(STATE_PREPARED);
        }
        super.prepare();
    }
 

import android.media.AudioTrack; //導入方法依賴的package包/類
/**
 * 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();
}
 

import android.media.AudioTrack; //導入方法依賴的package包/類
public static void initAudioTrack(Object mediaplayer_ref, int sampleRateInHz, int channelConfig) throws IOException {
    FFMpegPlayer mp = (FFMpegPlayer) ((WeakReference) mediaplayer_ref).get();
    if (mp != null) {
        int bufferSizeInBytes = AudioTrack.getMinBufferSize(sampleRateInHz, channelConfig, 2);
        try {
            mp.mTrack = new AudioTrack(3, sampleRateInHz, channelConfig, 2, bufferSizeInBytes, 1);
        } catch (IllegalStateException e) {
            e.printStackTrace();
        }
        try {
            if (mp.mTrack != null) {
                mp.mTrack.play();
            }
        } catch (IllegalStateException e2) {
            LogTag.e("Error creating uninitialized AudioTrack, re-initial it");
            int tryCount = 0;
            while (mp.mTrack.getPlayState() == 0 && tryCount < 3) {
                if (mp.mTrack != null) {
                    mp.mTrack.stop();
                    mp.mTrack.release();
                    mp.mTrack = null;
                }
                mp.mTrack = new AudioTrack(3, sampleRateInHz, channelConfig, 2, bufferSizeInBytes, 1);
                tryCount++;
            }
            if (mp.mTrack != null) {
                mp.mTrack.play();
            }
        }
    }
}