Regulate Android Audiotrack Playback Speed

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

Solution 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 sourceint 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 = newAudioTrack(AudioManager.STREAM_MUSIC, sampleRate,
                AudioFormat.CHANNEL_OUT_STEREO,
                AudioFormat.ENCODING_PCM_16BIT,
                bufSize,
                AudioTrack.MODE_STREAM);

Solution 2:

Like user @pskink said,

Most likely your sampleRate (or any other parameter passed to the AudioTrack constructor) is invalid.

So I would start by checking what value you are actually setting the sample rate.

For reference, you can also set the speed of AudioTrack by calling the setPlayBackParams method:

publicvoidsetPlaybackParams (PlaybackParams params)

If you check the AudioTrack docs, you can see the PlaybackParams docs and can set the speed and pitch of the output audio. This object can then be passed to set the playback parameters within your AudioTrack object.

However, it is unlikely that you will need to use this if your only issue is the original constructor sampleRate (since we cannot see where the variable sampleRate comes from).