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Re: AudioConverterNew() and 'perm' error on iPhone
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Re: AudioConverterNew() and 'perm' error on iPhone


  • Subject: Re: AudioConverterNew() and 'perm' error on iPhone
  • From: "Roni Music" <email@hidden>
  • Date: Fri, 14 Nov 2008 11:23:30 +0100


snip...

I need to get access to the decoded audio data for further DSP processing so I can't use the AudioQueue API.
Or is it possible to do this using the AudioQueue API?

AudioQueueRenderOffline is meant to enable this, but we are still looking at some issues here - I don't have any update to this at this stage

Do you mean there is a problem when using AudioQueueRenderOffline on the iPhone or also on Mac OS X?

To learn how to use it, I tested AudioQueueRenderOffline based on
aqplay from your AudioQueueTools sample. This is on OS X, not the iPhone.

Can't get it to work and it seems I'm in good company by looking in the archives
and would need some clarifications.


I replaced the AudioQueueStart() section with the code below.


Here is what I do:
1. create an audio channel layout
2. set up the output format: 44100 Hz, 16-bit, interleaved stereo, in this case same as the input file
3. call AudioQueueSetOfflineRenderFormat
4. create a new output buffer using AudioQueueAllocateBuffer
5. create a AudioTimeStamp
6. call AudioQueueStart (it isn't clear if this is needed but without this it's just silence)
7. read from the input file and and enqueue a new buffer by calling AQTestBufferCallback() from the sample
8. call AudioQueueOfflineRender() to get the output data
9. increase the absolute sample frame time for the AudioTimeStamp, then repeat 7,8, and 9 a couple of times


The first AudioQueueOfflineRender() call gives correct audio data back but after that is just silence.

I'm probably doing something wrong and would need some help.

Thanks,
Rolf

////////////////////////

code snippet from main() in aqplay.ccp

....

// set the volume of the queue
XThrowIfError (AudioQueueSetParameter(myInfo.mQueue, kAudioQueueParam_Volume, volume), "set queue volume");


XThrowIfError (AudioQueueAddPropertyListener (myInfo.mQueue, kAudioQueueProperty_IsRunning, MyAudioQueuePropertyListenerProc, NULL), "add listener");

// here is my off line rendering code

#if TEST_OFFLINE

// Set the output layout -> stereo
AudioChannelLayout layout;
memset(&layout, 0, sizeof(layout));
layout.mChannelLayoutTag = kAudioChannelLayoutTag_Stereo;


// Set the output format -> 44100 Hz, 16-bit, stereo (same as the input file used for testing)
CAStreamBasicDescription outFmt;
outFmt.mSampleRate = 44100.0;
outFmt.mFormatID = kAudioFormatLinearPCM;
outFmt.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked;
outFmt.mFramesPerPacket = 1;
outFmt.mChannelsPerFrame = 2;
outFmt.mBytesPerFrame = sizeof(SInt16) * outFmt.mChannelsPerFrame;
outFmt.mBytesPerPacket = outFmt.mBytesPerFrame * outFmt.mFramesPerPacket;
outFmt.mBitsPerChannel = (outFmt.mBytesPerFrame / outFmt.mChannelsPerFrame) * 8;


// change to offline rendering mode
err = AudioQueueSetOfflineRenderFormat(myInfo.mQueue, &outFmt, &layout);

// create an outbuffer
AudioQueueBufferRef outBuffer;
UInt32 outBufferByteSize = bufferByteSize; // same size as inbuffers
XThrowIfError(AudioQueueAllocateBuffer(myInfo.mQueue, outBufferByteSize, &outBuffer), "AudioQueueAllocateBuffer failed");


// create a timestamp
AudioTimeStamp timeStamp;
timeStamp.mFlags = kAudioTimeStampSampleTimeValid;
timeStamp.mSampleTime = 0;

myInfo.mCurrentPacket = 0; // read infile from start

// it seems the audio queue should bestarted also in offline mode, else we get silence only
XThrowIfError(AudioQueueStart(myInfo.mQueue, NULL), "AudioQueueStart failed");


// write the raw data to a file
static const char* outFile = "audio.raw";
FILE *fp = fopen(outFile, "w+");

for (int i = 0; i < 10; i++)
 {
 // use the input buffers
 int whatBuffer = i % kNumberBuffers;

// it seems the callback is not automatically called in off line mode, so read the infile and enqueue the data
AQTestBufferCallback(&myInfo, myInfo.mQueue, myInfo.mBuffers[whatBuffer]);


// the amount of data enqueued by the "callback"
UInt32 inNumberFrames = myInfo.mDataFormat.BytesToFrames(myInfo.mBuffers[whatBuffer]->mAudioDataByteSize);


// request the same number of samples frames -> inNumberFrames
// this call always renders the whole buffersize, the value of inNumberFrames doesn't matter. why??
err = AudioQueueOfflineRender(myInfo.mQueue,
&timeStamp,
outBuffer,
inNumberFrames);


// the number of frames we got
UInt32 outNumberFrames = outFmt.BytesToFrames(outBuffer->mAudioDataByteSize);


 // write to the file so we can listen to the result
 UInt32 nSamples = outNumberFrames*outFmt.NumberChannels();
 SInt16 *pBuf = (SInt16 *)outBuffer->mAudioData;
 fwrite(pBuf, 1, nSamples * sizeof(SInt16), fp);

   // increase the absolute sample frame time, is this correct??
 timeStamp.mSampleTime += inNumberFrames;
 }

fclose(fp);

#else

// lets start playing now - stop is called in the AQTestBufferCallback when there's
// no more to read from the file
XThrowIfError(AudioQueueStart(myInfo.mQueue, NULL), "AudioQueueStart failed");


 do
  {
  CFRunLoopRunInMode(kCFRunLoopDefaultMode, 0.25, false);
  } while (!myInfo.mDone /*|| gIsRunning*/);

 CFRunLoopRunInMode(kCFRunLoopDefaultMode, 1, false);

#endif

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References: 
 >AudioConverterNew() and 'perm' error on iPhone (From: "Roni Music" <email@hidden>)
 >Re: AudioConverterNew() and 'perm' error on iPhone (From: William Stewart <email@hidden>)

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