延迟播放音频



目前我正在做一个关于"延迟听觉反馈"(DAF(的项目。基本上,我想用麦克风录制声音,延迟一段特定的时间,然后播放。使用大约200毫秒的延迟和一个戴耳机的人,这种反馈会关闭人们流利说话的能力。(相当有趣:DAF在youtube上(

现在,我正试图使用256字节的byte[]缓冲区,使用SourceDataLine和TargetDataLine进行循环。如果缓冲区变大,延迟也会变大。我现在的问题是:我不知道以毫秒为单位的延迟是多少

有没有办法根据缓冲区的大小来计算以毫秒为单位的实际延迟?或者有没有其他方法可以得到这个结果?

这就是我现在的循环:

private int mBufferSize; // 256
private TargetDataLine mLineOutput;
private SourceDataLine mLineInput;
public void run() {
    ... creating the DataLines and getting the lines from AudioSystem ...
    // byte buffer for audio
    byte[] data = new byte[mBufferSize];
    // start the data lines
    mLineOutput.start();
    mLineInput.start();
    // start recording and playing back
    while (running) {
        mLineOutput.read(data, 0, mBufferSize);
        mLineInput.write(data, 0, mBufferSize);
    }
    ... closing the lines and exiting ...
}

您可以很容易地计算延迟,因为它取决于音频的采样率。假设这是CD质量(单声道(音频,则采样率为每秒44100个采样。200毫秒是0.2秒,因此44100 X 0.2=8820。

因此,您的音频播放需要延迟8820个样本(或17640个字节(。如果您将录制和播放缓冲区设置为这个大小(17640字节(,那么代码就会变得非常简单。当每个录制缓冲区被填满时,您将其传递给播放;这将实现正好一个缓冲器的持续时间的回放滞后。

Android中存在一些固有的延迟,您应该考虑到这一点,但除此之外。。。

创建一个循环缓冲区。不管多大,只要它足够大,可以容纳N 0个样本。现在用N个"0"个样本来编写它。

在这种情况下,N是(以秒为单位的延迟(*(以赫兹为单位的采样率(。

示例:200ms,16kHz立体声:

0.2s*1600Hz*(2通道(=3200*2个样本=6400个样本

您可能也将使用pcm数据,它是16位的,所以使用short而不是byte。

在用适量的零填充缓冲区后,开始读取扬声器的数据,同时填充麦克风的数据。

PCM Fifo:

public class PcmQueue
{
    private short                mBuf[] = null;
    private int                  mWrIdx = 0;
    private int                  mRdIdx = 0;
    private int                  mCount = 0;
    private int                  mBufSz = 0;
    private Object               mSync  = new Object();
    private PcmQueue(){}
    public PcmQueue( int nBufSz )
    {
        try {
            mBuf = new short[nBufSz];
        } catch (Exception e) {
            Log.e(this.getClass().getName(), "AudioQueue allocation failed.", e);
            mBuf = null;
            mBufSz = 0;
        }
    }
    public int doWrite( final short pWrBuf[], final int nWrBufIdx, final int nLen )
    {
        int sampsWritten   = 0;
        if ( nLen > 0 ) {
            int toWrite;
            synchronized(mSync) {
                // Write nothing if there isn't room in the buffer.
                toWrite = (nLen <= (mBufSz - mCount)) ? nLen : 0;
            }
            // We can definitely read toWrite shorts.
            while (toWrite > 0)
            {
                // Calculate how many contiguous shorts to the end of the buffer
                final int sampsToCopy = Math.min( toWrite, (mBufSz - mWrIdx) );
                // Copy that many shorts.
                System.arraycopy(pWrBuf, sampsWritten + nWrBufIdx, mBuf, mWrIdx, sampsToCopy);
                // Circular buffering.
                mWrIdx += sampsToCopy;
                if (mWrIdx >= mBufSz) {
                    mWrIdx -= mBufSz;
                }
                // Increment the number of shorts sampsWritten.
                sampsWritten += sampsToCopy;
                toWrite -= sampsToCopy;
            }
            synchronized(mSync) {
                // Increment the count.
                mCount = mCount + sampsWritten;
            }
        }
        return sampsWritten;
    }
    public int doRead( short pcmBuffer[], final int nRdBufIdx, final int nRdBufLen )
    {
        int sampsRead   = 0;
        final int nSampsToRead = Math.min( nRdBufLen, pcmBuffer.length - nRdBufIdx );
        if ( nSampsToRead > 0 ) {
            int sampsToRead;
            synchronized(mSync) {
                // Calculate how many shorts can be read from the RdBuffer.
                sampsToRead = Math.min(mCount, nSampsToRead);
            }
            // We can definitely read sampsToRead shorts.
            while (sampsToRead > 0) 
            {
                // Calculate how many contiguous shorts to the end of the buffer
                final int sampsToCopy = Math.min( sampsToRead, (mBufSz - mRdIdx) );
                // Copy that many shorts.
                System.arraycopy( mBuf, mRdIdx, pcmBuffer, sampsRead + nRdBufIdx, sampsToCopy);
                // Circular buffering.
                mRdIdx += sampsToCopy;
                if (mRdIdx >= mBufSz)  {
                    mRdIdx -= mBufSz;
                }
                // Increment the number of shorts read.
                sampsRead += sampsToCopy;
                sampsToRead -= sampsToCopy;
            }
            // Decrement the count.
            synchronized(mSync) {
                mCount = mCount - sampsRead;
            }
        }
        return sampsRead;
    }
}

和你的代码,修改为先进先出。。。我没有使用TargetDataLine/SourceDataLine的经验,所以如果它们只处理字节数组,请将FIFO修改为字节而不是短字节。

private int mBufferSize; // 256
private TargetDataLine mLineOutput;
private SourceDataLine mLineInput;
public void run() {
    ... creating the DataLines and getting the lines from AudioSystem ...

    // short buffer for audio
    short[] data = new short[256];
    final int emptySamples = (int)(44100.0 * 0.2); 
    final int bufferSize = emptySamples*2; 
    PcmQueue pcmQueue = new PcmQueue( bufferSize );
    // Create a temporary empty buffer to write to the PCM queue
    {
        short[] emptyBuf = new short[emptySamples];
        Arrays.fill(emptyBuf, (short)emptySamples );
        pcmQueue.doWrite(emptyBuf, 0, emptySamples);
    }
    // start recording and playing back
    while (running) {
        mLineOutput.read(data, 0, mBufferSize);
        pcmQueue.doWrite(data, 0, mBufferSize);        
        pcmQueue.doRead(data, 0, mBufferSize);        
        mLineInput.write(data, 0, mBufferSize);
    }
    ... closing the lines and exiting ...
} 

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