how can I create a function of sound volue from time??

[Dev04]

I have a question how can I analyze class AudioQueueBufferRef, for creating a function of sound volue from time?? Here is what I get . there is AudioQueueBufferRef fillBuf = audioQueueBuffer[fillBufferIndex]; volume height is 2000 elements from SInt16* coreAudioBuffer = (SInt16*)fillBuf->mAudioData. so function looks like H(t*i)=coreAudioBuffer[i] where t = 1/sampleRate = 1/22050 but here is a problem. my program gets sound and uses a class AudioStreamer for this. AudioStreamer has 3000 lines when I play music from Free Internet Radio - SHOUTcast Radio - Thousands of Free Online Radio Stations. internet radio - my problem is as follows either I dont know where 85 % of sound information is or I dont know how I can analyze class AudioQueueBufferRef
Here is the code where I analyze Buffer.

{@synchronized(self)
{
if ([self isFinishing] || stream == 0)
{
return;
}

inuse[fillBufferIndex] = true; // set in use flag
buffersUsed++;

// enqueue buffer
AudioQueueBufferRef fillBuf = audioQueueBuffer[fillBufferIndex];
fillBuf->mAudioDataByteSize = bytesFilled;

// ======>in this place I analyze Buffer
if (packetsFilled)
{
err = AudioQueueEnqueueBuffer(audioQueue, fillBuf, packetsFilled, packetDescs);
}
else
{
err = AudioQueueEnqueueBuffer(audioQueue, fillBuf, 0, NULL);
}
..............
..............}




when bitRate = 24 buffer has the following options int size=(fillBuf->mAudioDataByteSize) == 2000 double sampleRate=asbd.mSampleRate == 22050 numberOfChannels = asbd.mChannelsPerFrame == 1 it turns out that duration of play buffer float bufferTime =(size/numberOfChannels)/sampleRate == 0.1 number of buffers per second float numBuffersInOneSeconds == 1,5 duration of play all buffers per one second numBuffersInOneSeconds * time == 0.15 so it is 15 % of all information

as a result If buffer comes at 0.0 seconds he lasts up to 0.1 seconds.farther in my function there is no volume. second buffer comes in 0.7 seconds and lasts up to 0.8 seconds. but in reality the sound doesnt breaks. Maybe I'm doing something wrong .please tell me.

just for comparison

when bitRate = 32 buffer has the following options int size=(fillBuf->mAudioDataByteSize) == 2000 double sampleRate=asbd.mSampleRate == 22050 numberOfChannels = asbd.mChannelsPerFrame == 1 it turns out that duration of play buffer float bufferTime =(size/numberOfChannels)/sampleRate == 0.1 number of buffers per second float numBuffersInOneSeconds == 2 duration of play all buffers per one second numBuffersInOneSeconds * time == 0.2 so it is 20 % of all information

when bitRate = 32 buffer has the following options int size=(fillBuf->mAudioDataByteSize) == 1660 double sampleRate=asbd.mSampleRate == 44100 numberOfChannels = asbd.mChannelsPerFrame == 2 it turns out that duration of play buffer float bufferTime =(size/numberOfChannels)/sampleRate == 0.02 number of buffers per second float numBuffersInOneSeconds == 10 duration of play all buffers per one second numBuffersInOneSeconds * time == 0.2 so it is 20 % of all information