AudioTrack: Playing sound coming in over WiFi

Question

I've got an AudioTrack in my application, which is set to Stream mode. I want to write audio which I receive over a wireless connection. The AudioTrack is declared like this:

mPlayer = new AudioTrack(STREAM_TYPE,
                         FREQUENCY,
                         CHANNEL_CONFIG_OUT,
                         AUDIO_ENCODING,
                         PLAYER_CAPACITY,
                         PLAY_MODE);

Where the parameters are defined like:

private static final int FREQUENCY = 8000,
                         CHANNEL_CONFIG_OUT = AudioFormat.CHANNEL_OUT_MONO,
                         AUDIO_ENCODING = AudioFormat.ENCODING_PCM_16BIT,
                         PLAYER_CAPACITY = 2048,
                         STREAM_TYPE = AudioManager.STREAM_MUSIC,
                         PLAY_MODE = AudioTrack.MODE_STREAM;

However, when I write data to the AudioTrack with write(), it will play choppy... The call

byte[] audio = packet.getData();
mPlayer.write(audio, 0, audio.length);

is made whenever a packet is received over the network connection. Does anybody have an idea on why it sounds choppy? Maybe it has something to do with the WiFi connection itself? I don't think so, as the sound doesn't sound horrible the other way around, when I send data from the Android phone to another source over UDP. The sound then sounds complete and not choppy at all... So does anybody have an idea on why this is happening?

Answer 1

Do you know how many bytes per second you are recieving, the average time between packets compares, and the maximum time between packets? If not, can you add code to calculate it?

You need to be averaging 8000 samples/second * 2 bytes/sample = 16,000 bytes per second in order to keep the stream filled.

A gap of more than 2048 bytes / (16000 bytes/second) = 128 milliseconds between incoming packets will cause your stream to run dry and the audio to stutter.

One way to prevent it is to increase the buffer size (PLAYER_CAPACITY). A larger buffer will be more able to handle variation in the incoming packet size and rate. The cost of the extra stability is a larger delay in starting playback while you wait for the buffer to initially fill.