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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