AudioTrack: Playing sound coming in over WiFi

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,
  开发者_运维百科                       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?

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.

I have partially solved it by placing the mPlayer.write(audio, 0, audio.length); in it's own Thread. This does take away some of the choppy-ness (due to the fact that write is a blocking call), but it still sounds choppy after a good second or 2. It still has a significant delay of 2-3 seconds.

new Thread(){
    public void run(){
        byte[] audio = packet.getData();
        mPlayer.write(audio, 0, audio.length);
    }
}.start();

Just a little anonymous Thread that does the writing now...

Anybody have an idea on how to solve this issue?

Edit:

After some further checking and debugging, I've noticed that this is an issue with obtainBuffer. I've looked at the java code of the AudioTrack and the C++ code of AudioTrack And I've noticed that it only can appear in the C++ code.

if (__builtin_expect(result!=NO_ERROR, false)) {
    LOGW(   "obtainBuffer timed out (is the CPU pegged?) "
            "user=%08x, server=%08x", u, s);
    mAudioTrack->start(); // FIXME: Wake up audioflinger
    timeout = 1;
}

I've noticed that there is a FIXME in this piece of code. :< But anyway, could anybody explain how this C++ code works? I've had some experience with it, but it was never as complicated as this...

Edit 2:

I've tried somewhat different now, the difference being that I buffer the data I receive, and then when the buffer is filled with some data, it is being written to the player. However, the player keeps up with consuming for a few cycles, then the obtainBuffer timed out (is the CPU pegged?) warning kicks in, and there is no data at all written to the player untill it is kick started back to life... After that, it will continually get data written to it untill the buffer is emptied.

Another slight difference is that I stream a file to the player now. That is, reading it in chunks, the writing those chunks to the buffer. This simulates the packages being received over wifi...

I am beginning to wonder if this is just an OS issue that Android has, and it isn't something I can solve on my own... Anybody got any ideas on that?

Edit 3:

I've done more testing, but this doesn't help me any further. This test shows me that I only get lag when I try to write to the AudioTrack for the first time. This takes somewhat between 1 and 3 seconds to complete. I did this by using the following bit of code:

long beforeTime = Utilities.getCurrentTimeMillis(), afterTime = 0;
mPlayer.write(data, 0, data.length);
afterTime = Utilities.getCurrentTimeMillis();
Log.e("WriteToPlayerThread", "Writing a package took " + (afterTime - beforeTime) + " milliseconds");

However, I get the following results: Logcat Image http://img810.imageshack.us/img810/3453/logcatimage.png

These show that the lag initially occurs at the beginning, after which the AudioTrack keeps getting data continuously... I really need to get this one fixed...