core-audio

I have used to following code the stream the i/o of audio from microphone. What I want to do is want to select the rear microphone for recording. I have read that setting kAudioSessionProperty_Mode to kAudioSessionMode_VideoRecording can do the work but I am not sure how to use this with my code. Can any one help me in successfully setting this parameter. I have these lines for setting the property status = AudioUnitSetProperty(audioUnit, kAudioSessionProperty_Mode, kAudioSessionMode_VideoRecording, kOutputBus, &audioFormat, sizeof(audioFormat)); checkStatus(status); but its not working. in

Previously I read audio samples from a complete audio file using CMSampleBufferGetAudioBufferListWithRetainedBlockBuffer . Right now I would like to do the same using ranges (ie i specify the range in time.. read a small chunk of audio as per the time, and then go back and read again). The reason why I want to use time range is b/c I want to control the size of each read (to fit in a packet with a max size). for some reason, there is always a bump between each read. In my code you'll notice that I start the AVAssetReader and end it every time I set a time range, and that's b/c I cannot

I set up an AVAudioRecorder instance the following way when my view loads: AVAudioSession *audioSession = [AVAudioSession sharedInstance]; audioSession.delegate = self; [audioSession setActive:YES error:nil]; [audioSession setCategory:AVAudioSessionCategoryRecord error:nil]; NSString *tempDir = NSTemporaryDirectory(); NSString *soundFilePath = [tempDir stringByAppendingPathComponent:@"sound.m4a"]; NSURL *soundFileURL = [NSURL fileURLWithPath:soundFilePath]; NSLog(@"%@", soundFileURL); NSDictionary *recordSettings = [NSDictionary dictionaryWithObjectsAndKeys: [NSNumber numberWithInt

I have these 6 coefficients for peaking EQ: b0 = 1 + (α ⋅ A) b1 = −2⋅ωC b2 = 1 - (α ⋅ A) a0 = 1 + (α / A) a1 = −2 ⋅ ωC a2 = 1 − (α / A) With these intermediate variables: ωc = 2 ⋅ π ⋅ fc / fs ωS = sin(ωc) ωC = cos(ωc) A = sqrt(10^(G/20)) α = ωS / (2Q) The documentation of vDSP_deq22() states that "5 single-precision inputs, filter coefficients" should be passed but I have 6 coefficients! Also, in what order do I pass them to vDSP_deq22() ? Update (17/05): I recommend everyone to use my DSP class I released on github: https://github.com/bartolsthoorn/NVDSP It'll probably save you quite some

I'd like to record the some audio using AVAudioEngine and the users Microphone. I already have a working sample, but just can't figure out how to specify the format of the output that I want... My requirement would be that I need the AVAudioPCMBuffer as I speak which it currently does... Would I need to add a seperate node that does some transcoding? I can't find much documentation/samples on that problem... And I am also a noob when it comes to Audio-Stuff. I know that I want NSData containing PCM-16bit with a max sample-rate of 16000 (8000 would be better) Here's my working sample: private