I'm trying to use a Low Pass Filter AU. I keep getting a kAudioUnitErr_FormatNotSupported (-10868) error when setting the stream format to the filter unit, but if I just use the Remote IO unit there's no error.
The stream format I'm using is (Updated):
myASBD.mSampleRate = hardwareSampleRate;
myASBD.mFormatID = kAudioFormatLinearPCM;
myASBD.mFormatFlags = kAudioFormatFlagIsSignedInteger;
myASBD.mBitsPerChannel = 8 * sizeof(float);
myASBD.mFramesPerPacket = 1;
myASBD.mChannelsPerFrame = 1;
myASBD.mBytesPerPacket = sizeof(float) * myASBD.mFramesPerPacket;
myASBD.mBytesPerFrame = sizeof(float) * myASBD.mChannelsPerFrame;
And I'm setting the filter stream like this:
// Sets input stream type to ASBD
setupErr = AudioUnitSetProperty(filterUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &myASBD, sizeof(myASBD));
NSLog(@"Filter in: %i", setupErr);
//NSAssert(setupErr == noErr, @"No ASBD on Finput");
//Sets output stream type to ASBD
setupErr = AudioUnitSetProperty(filterUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 0, &myASBD, sizeof(myASBD));
NSLog(@"Filter out: %i", setupErr);
NSAssert(setupErr == noErr, @"No ASBD on Foutput");
iOS过滤器音频单元的规范格式为8.24定点(线性PCM),每个通道32位,而不是16位。
what format is working wit the reverb unit??? I'm getting weird errors tryn to record a buffer....any news on this topic?
Try this for the canonical format.
size_t bytesPerSample = sizeof (AudioUnitSampleType); //Default is 4 bytes myASBD.mSampleRate = hardwareSampleRate; myASBD.mFormatID = kAudioFormatLinearPCM; myASBD.mFormatFlags = kAudioFormatFlagsAudioUnitCanonical; //Canonical AU format myASBD.mBytesPerPacket = bytesPerSample; myASBD.mFramesPerPacket = 1; myASBD.mBytesPerFrame = bytesPerSample; myASBD.mChannelsPerFrame = 2; //Stereo myASBD.mBitsPerChannel = 8 * bytesPerSample; //32bit integer
You will need to make sure all your AudioUnits ASBDs are configured uniformly.
If you are doing heavy audio processing, floats (supported in iOS5) is not a bad idea too.
size_t bytesPerSample = sizeof (float); //float is 4 bytes myASBD.mSampleRate = hardwareSampleRate; myASBD.mFormatID = kAudioFormatLinearPCM; myASBD.mFormatFlags = kAudioFormatFlagIsFloat; myASBD.mBytesPerPacket = bytesPerSample; myASBD.mFramesPerPacket = 1; myASBD.mBytesPerFrame = bytesPerSample; myASBD.mChannelsPerFrame = 2; myASBD.mBitsPerChannel = 8 * bytesPerSample; //32bit float
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