Re: OpenAL for voicechat

Hi Peter,

I think you should probably change ...

> if(processed > 0) //If there is a free buffer
> {
>    ALuint buffer;
>    alSourceUnqueueBuffers(soundsOut_current->source, 1, &buffer);
>    //Fill the buffer with the received data
>    alBufferData(buffer, AL_FORMAT_MONO16, decoded_buffer,
> voice_Bufsize, voice_SamplesPerSecond);
>         alSourceQueueBuffers(soundsOut_current->source, 1, &buffer);
> }

to

while (processed) // while there are processed buffers
{
    ALuint buffer;
    alSourceUnqueueBuffers(soundsOut_current->source, 1, &buffer);
    //Fill the buffer with the received data
    alBufferData(buffer, AL_FORMAT_MONO16, decoded_buffer, voice_Bufsize,
voice_SamplesPerSecond);
    alSourceQueueBuffers(soundsOut_current->source, 1, &buffer);
    processed--;
}

(You will need to have more data to fill each buffer of course)

What values are using for the Buffer Size ( voice_Bufsize ) and sample-rate
( voice_SamplesPerSecond ) ?

> received per client). If a 3rd client connects and sends his stream
> (=> 2 streams are received per client) the final result is that both
> streams are played somehow cut (it's difficult to explain).

It would be helpful to know if OpenAL is being starved of data.  You can
detect this by adding a check after this statement ...

> alGetSourcei(soundsOut_current->source, AL_BUFFERS_PROCESSED,
&processed);

if (processed == outputBuffer_nBufs)
      print("Ooops - AL ran out of data");

Dan
Creative Labs (UK) Ltd.

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[email protected] wrote on 01/25/2010 09:36:05 PM:

> Hi!
>
> I'm using OpenAL for the output of a voicechat. It works great with
> 2 participants that talk to each other (=> only one stream is
> received per client). If a 3rd client connects and sends his stream
> (=> 2 streams are received per client) the final result is that both
> streams are played somehow cut (it's difficult to explain).
>
> The technical stuff:
> I'm using one source per received stream and each source uses a
> buffer queue with 4 buffers assigned. The stream is sent with udp
> over lan (100 Mbit => the problem is not caused by latency). I'm
> using OpenAL 1.1.
>
> Ok, first of all I thought the packets aren't received correct
> anymore after the 3rd client connects. But then I wrote a [ms]
> timestamp into a logfile everytime I received a packat and they are
> still coming correctly.
>
> => The problem must be the output code somehow. Maybe I missed
> something so that the sources aren't correctly mixed together or
something.
>
> Here the important OpenAL code:
> //------------------------------------------------------------
> //Initializes OpenAL:
> ////////////////////////
> pOutputDevice = alcOpenDevice(NULL); //Get handle to device
> pOutputContext = alcCreateContext(pOutputDevice, NULL); //Create audio
context
> alcMakeContextCurrent(pOutputContext); //Set active context
> if(alcGetError(pOutputDevice) != ALC_NO_ERROR) //Do an error check
> {
>    closesocket(voice_clientsock);
>    voice_clientsock = 0;
>    waveInClose(voice_recHandle);
>    return _VAR(-1);
> }
> alGetError(); //Clear the bit
>
> //Initializes the Listener:
> ALfloat ListenerPos[] = {0.0, 0.0, 0.0};
> ALfloat ListenerVel[] = {0.0, 0.0, 0.0};
> ALfloat ListenerOri[] = {0.0, 0.0, -1.0, 0.0, 1.0, 0.0};
> alListenerfv(AL_POSITION, ListenerPos);
> alListenerfv(AL_VELOCITY, ListenerVel);
> alListenerfv(AL_ORIENTATION, ListenerOri);
>
> //Initializes a OpenAL Source:
> soundsOut_current->next = 0;
> soundsOut_current->radioid = radioid;
> soundsOut_current->clientid = sender_clientid;
> //Create the buffers for playing the sound
> alGenBuffers(outputBuffer_nBufs, soundsOut_current->buffers);
> if (alGetError() != AL_NO_ERROR) return 0;
>
> //Initialize all Buffers
> for(int i=0;i<outputBuffer_nBufs;i++)
> alBufferData(soundsOut_current->buffers[i], AL_FORMAT_MONO16,
> decoded_buffer, voice_Bufsize, voice_SamplesPerSecond);
>
> //Generate the sound source
> alGenSources(1, &soundsOut_current->source);
> if(alGetError() != AL_NO_ERROR) return 0;
>
> //Inizializes the Bufferqueue
> alSourceQueueBuffers(soundsOut_current->source, outputBuffer_nBufs,
> soundsOut_current->buffers);
> alSourcef(soundsOut_current->source, AL_PITCH, 1.0f);
> alSourcef(soundsOut_current->source, AL_GAIN, 1.0f);
> //Create a normal 2D Sound
> alSource3f(soundsOut_current->source, AL_POSITION, 0.0, 0.0, 0.0);
> alSource3f(soundsOut_current->source, AL_VELOCITY, 0.0, 0.0, 0.0);
> alSourcei(soundsOut_current->source, AL_SOURCE_RELATIVE, AL_TRUE);
> alSourcei(soundsOut_current->source, AL_LOOPING, AL_FALSE);
>
>
> //Updates the buffers everytime a packet is received:
> //////////////////////////////////////////////////////
>
> ALint processed, state;
> alGetSourcei(soundsOut_current->source, AL_SOURCE_STATE, &state);
> alGetSourcei(soundsOut_current->source, AL_BUFFERS_PROCESSED,
&processed);
>
> if(processed > 0) //If there is a free buffer
> {
>    ALuint buffer;
>    alSourceUnqueueBuffers(soundsOut_current->source, 1, &buffer);
>    //Fill the buffer with the received data
>    alBufferData(buffer, AL_FORMAT_MONO16, decoded_buffer,
> voice_Bufsize, voice_SamplesPerSecond);
>         alSourceQueueBuffers(soundsOut_current->source, 1, &buffer);
> }
> // Restart in case we didn't get more data in time
> if(state != AL_PLAYING) alSourcePlay(soundsOut_current->source);
>
> //--------------------------------------------------------------
>
>
> Any ideas? I'm really getting stuck at this bug. I tested every
> possible reason of this problem with no success.
>
> Thanks for your help!
>
> Best regards,
> Peter Soxberger
>
> --
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