AudioTrack lag: obtainBuffer timed out

问题

I'm playing WAVs on my Android phone by loading the file and feeding the bytes into AudioTrack.write() via the FileInputStream > BufferedInputStream > DataInputStream method. The audio plays fine and when it is, I can easily adjust sample rate, volume, etc on the fly with nice performance. However, it's taking about two full seconds for a track to start playing. I know AudioTrack has an inescapable delay, but this is ridiculous. Every time I play a track, I get this:

03-13 14:55:57.100: WARN/AudioTrack(3454): obtainBuffer timed out (is the CPU pegged?) 0x2e9348 user=00000960,     server=00000000
03-13 14:55:57.340: WARN/AudioFlinger(72): write blocked for 233 msecs, 9 delayed writes, thread 0xba28

I've noticed that the delayed write count increases by one every time I play a track -- even across multiple sessions -- from the time the phone has been turned on. The block time is always 230 - 240ms, which makes sense considering a minimum buffer size of 9600 on this device (9600 / 44100). I've seen this message in countless searches on the Internet, but it usually seems to be related to not playing audio at all or skipping audio. In my case, it's just a delayed start.

I'm running all my code in a high priority thread. Here's a truncated-yet-functional version of what I'm doing. This is the thread callback in my playback class. Again, this works (only playing 16-bit, 44.1kHz, stereo files right now), it just takes forever to start and has that obtainBuffer/delayed write message every time.

public void run() {

    // Load file
    FileInputStream mFileInputStream;
    try {
        // mFile is instance of custom file class -- this is correct, 
        // so don't sweat this line
        mFileInputStream = new FileInputStream(mFile.path());
    } catch (FileNotFoundException e) {
        // log
    }

    BufferedInputStream mBufferedInputStream = new BufferedInputStream(mFileInputStream, mBufferLength);
    DataInputStream mDataInputStream = new DataInputStream(mBufferedInputStream);

    // Skip header
    try {
        if (mDataInputStream.available() > 44) {
            mDataInputStream.skipBytes(44);
        }
    } catch (IOException e) {
        // log
    }

    // Initialize device
    mAudioTrack = new AudioTrack(
        AudioManager.STREAM_MUSIC, 
        ConfigManager.SAMPLE_RATE, 
        AudioFormat.CHANNEL_CONFIGURATION_STEREO, 
        AudioFormat.ENCODING_PCM_16BIT, 
        ConfigManager.AUDIO_BUFFER_LENGTH,
        AudioTrack.MODE_STREAM
    );
    mAudioTrack.play();

    // Initialize buffer
    byte[] mByteArray = new byte[mBufferLength];
    int mBytesToWrite = 0;
    int mBytesWritten = 0;

    // Loop to keep thread running
    while (mRun) {

        // This flag is turned on when the user presses "play"
        while (mPlaying) {

            try {
                // Check if data is available
                if (mDataInputStream.available() > 0) {

                    // Read data from file and write to audio device
                    mBytesToWrite = mDataInputStream.read(mByteArray, 0, mBufferLength);
                    mBytesWritten += mAudioTrack.write(mByteArray, 0, mBytesToWrite);

                }
            }
            catch (IOException e){
                // log
            }                
        }   
    }
}

If I can get past the artificially long lag, I can easily deal with the inherit latency by starting my write at a later, predictable position (ie, skip past the minimum buffer length when I start playing a file).

回答1:

I ran into a similar problem, although I was using a RandomAccessFile, instead of a BufferedInputStream, to read the PCM data. The issue was that the file I/O was too slow. I suspect you will have this problem even with a buffered stream, because the I/O is still taking place on the same thread as audio processing.

The solution is to have two threads: A thread that reads buffers from a file and queues them into memory, and another thread that reads from this queue and writes to the audio hardware. I used a ConcurrentLinkedQueue to accomplish this.

I used the same technique for recording, using AudioRecord, but in the reverse direction. The key is to place the file I/O on a separate thread.

回答2:

A bit late to the party answering this, but in case it helps anyone in the future - I ran into this exact problem with code pretty similar to the code in the question, where the AudioTrack is created and set to play, but not written to immediately.

I found that creating the AudioTrack immediately before you start writing to it made the delay go away. For some reason AudioTrack doesn't seem to like sitting around with an empty buffer.

In terms of the code above, you'd want to do something like

mAudioTrack=null;
while (mRun) 
{ 

    // This flag is turned on when the user presses "play" 
    while (mPlaying) 
    { 

        try 
        { 
            if (mAudioTrack==null) {   
                mAudioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, ConfigManager.SAMPLE_RATE,AudioFormat.CHANNEL_CONFIGURATION_STEREO, AudioFormat.ENCODING_PCM_16BIT,ConfigManager.AUDIO_BUFFER_LENGTH,AudioTrack.MODE_STREAM);   
                mAudioTrack.play();   
            }
            // Rest of the playback code here
        }
    }
    mAudioTrack=null;
}

来源：https://stackoverflow.com/questions/5293025/audiotrack-lag-obtainbuffer-timed-out