Question

Been banging my head against this problem all morning.

I have setup a connection to a datasource which returns audio data (It is a recording device, so there is no set length on the data. the data just streams in. Like, if you would open a stream to a radio)

and I have managed to receive all the packets of data in my code. Now I just need to play it. I want to play the data that is coming in, so I do not want to queue a few minutes or anything, I want to use the data I am recieving at that exact moment and play it.

Now I been searching all morning finding different examples but none were really layed out.

in the

• (void)connection:(NSURLConnection )connection didReceiveData:(NSData)data {

function, the “data” package is the audio package. I tried streaming it with AVPlayer, MFVideoPlayer but nothing has worked for me so far. Also tried looking at mattgallagher’s Audiostreamer but still was unable to achieve it.

Anyone here can help, has some (preferably) working examples?

Answer 1

Careful: The answer below is only valid if you receive PCM data from the server. This is of course never happens. That’s why between rendering the audio and receiving the data you need another step: data conversion.

Depending on format, this could be more or less tricky, but in general you should use Audio Converter Services for this step.

You should use -(void)connection:(NSURLConnection )connection didReceiveData:(NSData)data only to fill a buffer with the data that comes from the server, playing it should not have anything to do with this method.

Now, to play the data you ‘stored’ in memory using the buffer you need to use RemoteIO and audio units. Here is a good, comprehensive tutorial. You can remove the “record” part from the tutorial as you don’t really need it.

As you can see, they define a callback for playback:

callbackStruct.inputProc = playbackCallback;
callbackStruct.inputProcRefCon = self;
status = AudioUnitSetProperty(audioUnit, 
                              kAudioUnitProperty_SetRenderCallback, 
                              kAudioUnitScope_Global, 
                              kOutputBus,
                              &callbackStruct, 
                              sizeof(callbackStruct));

and playbackCallback function looks like this:

static OSStatus playbackCallback(void *inRefCon, 
                          AudioUnitRenderActionFlags *ioActionFlags, 
                          const AudioTimeStamp *inTimeStamp, 
                          UInt32 inBusNumber, 
                          UInt32 inNumberFrames, 
                          AudioBufferList *ioData) {

    for (int i = 0 ; i < ioData->mNumberBuffers; i++){      
        AudioBuffer buffer = ioData->mBuffers[i];
        unsigned char *frameBuffer = buffer.mData;
        for (int j = 0; j < inNumberFrames*2; j++){
            frameBuffer[j] = getNextPacket();//this here is a function you have to make to get the next chunk of bytes available in the stream buffer
        }
    }

    return noErr;
}

Basically what it does is to fill up the ioData buffer with the next chunk of bytes that need to be played. Be sure to zero out (silence) the ioData buffer if there is no new data to play (the player is silenced if not enough data is in the stream buffer).

Also, you can achieve the same thing with OpenAL using alSourceQueueBuffers and alSourceUnqueueBuffers to queue buffers one after the other.

That’s it. Happy codding!