Regulate Android AudioTrack playback speed

Question

I'm currently trying to playback audio using AudioTrack. Audio is received over the network and application continuously read data and add to an internal buffer. A separate thread is consuming data and using AudioTrack to playback.

Problems:

1. Audio playback fluctuate (feels like audio drop at a regular interval) continuously making it unclear.
2. Playback speed is too high or too low making them unrealistic.

In order to avoid the network latency and other factors I made the application to wait till it read enough data and playback at the end.

This makes the audio to play really fast. Here is a basic sample of logic I use.

    sampleRate = AudioTrack.getNativeOutputSampleRate(AudioManager.STREAM_MUSIC);
    audioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, sampleRate,
                    AudioFormat.CHANNEL_OUT_STEREO,
                    AudioFormat.ENCODING_PCM_16BIT,
                    AudioTrack.getMinBufferSize(sampleRate, AudioFormat.CHANNEL_OUT_STEREO, AudioFormat.ENCODING_PCM_16BIT),
                    AudioTrack.MODE_STREAM);
audioTrack.play();

short shortBuffer[] = new short[AudioTrack.getMinBufferSize(sampleRate, AudioFormat.CHANNEL_OUT_STEREO, AudioFormat.ENCODING_PCM_16BIT)];
while (!stopRequested){
    readData(shortBuffer);
    audioTrack.write(shortBuffer, 0, shortBuffer.length, AudioTrack.WRITE_BLOCKING);
}

Is it correct to say that Android AudiTrack class doesn't have in built functionality to control the audio playback based on environment conditions? If so, are there better libraries available with a simplified way for audio playback?

Answer 1

The first issue that I see, it is an arbitrary sampling rate.

AudioTrack.getNativeOutputSampleRate will return the sampling rate that used by the sound system. It may be 44100, 48000, 96000, 192000 or whatever. But looks like you have audio data from some independent source, which produces the data on the very exact sampling rate.

Let's say audio data from the source is sampled at 44100 samples per second. If you start playing it at 96000 it will be speeded up and higher pitched.

So, use the sampling rate setting, along with the number of channels, sample format etc, as it given by the source, not relying on system defaults.

The second: are you sure the readData procedure always will be fast enough to successfully fill the buffer, whatever small the buffer is, and return back faster than the buffer is played?

You have created AudioTrack with AudioTrack.getMinBufferSize passed as bufferSizeInBytes parameter.

The getMinBufferSize function returns a minimum possible size of the buffer that can be used at this parameter. Let's say it returned the size corresponding to a buffer of 10ms length. That means the new data should be prepared within this time interval. I.e. The time interval between previous write returned control and new write is performed should be less than the time size of the buffer.

So, if the readData function may delay for some reason longer than that time interval, the playback will be paused for that time, you'll hear small gaps in the playback.

The reasons why readData may delay could be various: if it's reading data from the file, then it may delay waiting for IO operations; if it allocates java objects, it may be bumped into garbage collector's delay; if it uses some kind of decoder of another kind of audio source which uses it's own buffering, it may periodically delay refilling the buffer.

But anyway, if you're not creating some kind of real-time synthesizer which should react as soon as possible to the user input, always use the buffer size reasonably high, but not less than getMinBufferSize returned. I.e.:

sampleRate = 44100;// sampling rate of the source

int bufSize = sampleRate * 4; // 1 second length; 4 - is the frame size: 2 chanels * 2 bytes per each sample
bufSize = max(bufSize, AudioTrack.getMinBufferSize(sampleRate, AudioFormat.CHANNEL_OUT_STEREO, AudioFormat.ENCODING_PCM_16BIT)); // Not less than getMinBufferSize returns
audioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, sampleRate,
                AudioFormat.CHANNEL_OUT_STEREO,
                AudioFormat.ENCODING_PCM_16BIT,
                bufSize,
                AudioTrack.MODE_STREAM);