After filing a DTS, found out the problem.
The AudioSettings property is not enough to convert the output to the
necessary format. Though I needed to add "[NSNumber
numberWithBool:NO],
AVLinearPCMIsNonInterleaved," to the dictionary.
Upon receiving a sample, there is still further PCM format conversion that
is required. This can be done using the "AudioConverterConvertBuffer"
function.
This function requires a source and destination
"AudioStreamBasicDescription".
The source can be obtained using
"CMAudioFormatDescriptionGetStreamBasicDescription" and
"CMSampleBufferGetFormatDescription".
The destination is setup by the user, mine is set as follow:
_destinationDescription.mFormatID =
kAudioFormatLinearPCM;
_destinationDescription.mSampleRate =
AV::kSampleRate;
_destinationDescription.mBitsPerChannel = 16;
_destinationDescription.mBytesPerFrame = (2 * 16) / 8;
_destinationDescription.mBytesPerPacket = (2 * 16) / 8;
_destinationDescription.mChannelsPerFrame = 2;
_destinationDescription.mFormatFlags =
kLinearPCMFormatFlagIsPacked |
kAudioFormatFlagsNativeEndian |
kAudioFormatFlagIsSignedInteger;
_destinationDescription.mFramesPerPacket = 1;
_destinationDescription.mReserved = 0;
Once the conversion was done, the audio sounded perfectly!
Hope anyone who comes across this problem finds this useful in the future.
-San Saeteurn
On 5/30/13 3:22 PM, "San Saeteurn" <s...@telestream.net> wrote:
>Hello
>
>I am trying to write an external plugin that simply reads audio from the
>built in microphone, obtains the raw audio data, and passes it to an SDK
>API.
>My setup is very simple, I have a single session, with a single input
>device (built in microphone) and a single output
>(AVCaptureAudioDataOutput)
>I Initialize the Session, Input, and Output as follow:
>
> // Obtain the device and retain it
>
> _pAudioCaptureInput = [pAVInputDevice retain];
>
>
> // Initialize Capture Session
>
> _dispatchQueue = dispatch_get_main_queue();
>
> _pCaptureSession = [[AVCaptureSession alloc] init];
>
> _pCaptureSession.sessionPreset = AVCaptureSessionPresetHigh;
>
>
>
> // Put the Device inside an Input
>
> pInputDevice = [[AVCaptureDeviceInput alloc]
>initWithDevice:_pAudioCaptureInput error:&pError];
>
> [_pCaptureSession addInput:pInputDevice]; // Add Audio Input
>to Session
>
> [pInputDevice release];
>
> pInputDevice = nil;
>
>
> // Setup the Output Configurations
>
> NSDictionary *pAudioSettings = [NSDictionary
>dictionaryWithObjectsAndKeys:
>
> [NSNumber
>numberWithInt:kAudioFormatLinearPCM], AVFormatIDKey,
>
> [NSNumber
>numberWithInt:44100], AVSampleRateKey,
>
> [NSNumber
>numberWithInt:2], AVNumberOfChannelsKey,
>
> [NSNumber
>numberWithInt:16], AVLinearPCMBitDepthKey,
>
> [NSNumber
>numberWithBool:NO], AVLinearPCMIsFloatKey,
>
> [NSNumber
>numberWithBool:NO], AVLinearPCMIsBigEndianKey,
>
> nil];
>
>
>
>
> // Create the Output
>
> _pAudioCaptureOutput = [[AVCaptureAudioDataOutput alloc]
>init];
>
> [_pAudioCaptureOutput setAudioSettings:pAudioSettings]; //
>Apply Output Settings
>
> [_pCaptureSession addOutput:_pAudioCaptureOutput]; // Add
>Output to Session
>
>
> // Create the Output Delegate that will receive the
>SampleBuffer callbacks
>
> _pAudioCaptureDelegate = [[ACaptureDelegate alloc] init];
>
> [_pAudioCaptureOutput
>setSampleBufferDelegate:_pAudioCaptureDelegate
>
> queue:_dispatchQueue];
>
>
>
> // Remember the Device Name
>
> _deviceName = [_pAudioCaptureInput.localizedName UTF8String];
>
>I start the session running by calling startRunning on the session. My
>callback delegate is notified with Audio Samples. The follow is the
>callback implementation:
>
>
>- (void)captureOutput:(AVCaptureOutput *)captureOutput
>didOutputSampleBuffer:(CMSampleBufferRef)sampleBuffer
>fromConnection:(AVCaptureConnection *)connection
>
>{
>
>// Save the Timescale
>
> CMTime timestamp =
>CMSampleBufferGetOutputPresentationTimeStamp(sampleBuffer);
>
> if(_timescale == -1) {
>
> _timescale = timestamp.timescale;
>
> }
>
> CMItemCount numSamples = CMSampleBufferGetNumSamples(sampleBuffer);
>
>
> // Get Audio Data
>
> CMBlockBufferRef dataBuffer =
>CMSampleBufferGetDataBuffer(sampleBuffer);
>
> zuint32 dataLength = CMBlockBufferGetDataLength(dataBuffer);
>
>
>
> AVCaptureAudioDataOutput *pOutput =
>static_cast<AVCaptureAudioDataOutput *>(captureOutput);
>
> NSDictionary *pAudioSettings = pOutput.audioSettings;
>
>
>
> // Retrieve Audio Settings Information from AVOutputDevice
>
> NSNumber *audioSettingValue = nil;
>
> zint32 sampleRate = 44100;
>
> audioSettingValue = [pAudioSettings objectForKey:AVSampleRateKey];
>
> if(audioSettingValue) {
>
> sampleRate = [audioSettingValue intValue];
>
> }
>
>
> zint32 bitDepth = 16;
>
> audioSettingValue = [pAudioSettings
>objectForKey:AVLinearPCMBitDepthKey];
>
> if(audioSettingValue) {
>
> bitDepth = [audioSettingValue intValue];
>
> }
>
>
>
> zint32 channels = 2;
>
> audioSettingValue = [pAudioSettings
>objectForKey:AVNumberOfChannelsKey];
>
> if(audioSettingValue) {
>
> channels = [audioSettingValue intValue];
>
> }
>
>
>
>
> zuint8 *pAudioData = new zuint8[dataLength];
>
> OSStatus err = CMBlockBufferCopyDataBytes(dataBuffer, 0, dataLength,
>pAudioData);
>
> if(err == noErr) {
>
>// Populate SDK Audio structure
>
> _audioBuffer.format = kFormatPCM;
>
> _audioBuffer.dataLength = dataLength;
>
> _audioBuffer.timeStamp = 0;
>
> _audioBuffer.flags = kBufferFlagTimeValid;
>
> _audioBuffer.sampleRate = sampleRate;
>
> _audioBuffer.bitsPerSample = bitDepth;
>
> _audioBuffer.channels = channels;
>
> _audioBuffer.data = pAudioData; // The Raw Audio Data
>
>
>
>// Send Audio Data to SDK API
>
> _pAudioDataArriveCallback(_pAudioDataArriveContext,
>&_audioBuffer);
>
>
>
> delete [] pAudioData;
>
> pAudioData = NULL;
>
> }
>
>}
>
>The Problem is, the audio is playing back with a higher pitch. Like
>someone is fast forwarding. If I adjust the SampleRate in the Output
>Audio Settings, it affects the play out.
>Higher I set it, the less squeaky it becomes, but never correct. Is
>there any initialization step I am missing?
>
>I have not been able to play the raw audio data back myself without the
>main app as I am new to AVFoundation and CoreAudio.
>If someone has sample code of getting this up, I would much appreciate it.
>
>Thanks,
>-San Saeteurn
>
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