iOS: Audio Unit RemoteIO not working on iPhone

audio, audiounit, ios, iphone, microphone

Solution

I solved my problem. I simply needed to initialize the AudioSession before playing/recording. I did so with the following code:

OSStatus status;

AudioSessionInitialize(NULL, NULL, NULL, self);
UInt32 sessionCategory = kAudioSessionCategory_PlayAndRecord;
status = AudioSessionSetProperty (kAudioSessionProperty_AudioCategory,
                               sizeof (sessionCategory),
                               &sessionCategory);

if (status != kAudioSessionNoError)
{
    if (status == kAudioServicesUnsupportedPropertyError) {
        NSLog(@"AudioSessionInitialize failed: unsupportedPropertyError");
    }else if (status == kAudioServicesBadPropertySizeError) {
        NSLog(@"AudioSessionInitialize failed: badPropertySizeError");
    }else if (status == kAudioServicesBadSpecifierSizeError) {
        NSLog(@"AudioSessionInitialize failed: badSpecifierSizeError");
    }else if (status == kAudioServicesSystemSoundUnspecifiedError) {
        NSLog(@"AudioSessionInitialize failed: systemSoundUnspecifiedError");
    }else if (status == kAudioServicesSystemSoundClientTimedOutError) {
        NSLog(@"AudioSessionInitialize failed: systemSoundClientTimedOutError");
    }else {
        NSLog(@"AudioSessionInitialize failed! %ld", status);
    }
}


AudioSessionSetActive(TRUE);

...

Problem

I'm trying to create my own custom sound effects Audio Unit based on the input from the mic. This application allows simultaneous input/output from the microphone to speaker. I can apply effects and work using the simulator, but when I try to test on the iPhone I can't hear anything. I paste my code if anyone can help me: ``` - (id) init{ self = [super init]; OSStatus status; // Describe audio component AudioComponentDescription desc; desc.componentType = kAudioUnitType_Output; desc.componentSubType = kAudioUnitSubType_RemoteIO; desc.componentFlags = 0; desc.componentFlagsMask = 0; desc.componentManufacturer = kAudioUnitManufacturer_Apple; // Get component AudioComponent inputComponent = AudioComponentFindNext(NULL, &desc); // Get audio units status = AudioComponentInstanceNew(inputComponent, &audioUnit); checkStatus(status); // Enable IO for recording UInt32 flag = 1; status = AudioUnitSetProperty(audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, kInputBus, &flag, sizeof(flag)); checkStatus(status); // Enable IO for playback status = AudioUnitSetProperty(audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, kOutputBus, &flag, sizeof(flag)); checkStatus(status); // Describe format AudioStreamBasicDescription audioFormat; audioFormat.mSampleRate = 44100.00; audioFormat.mFormatID = kAudioFormatLinearPCM; audioFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked; audioFormat.mFramesPerPacket = 1; audioFormat.mChannelsPerFrame = 1; audioFormat.mBitsPerChannel = 16; audioFormat.mBytesPerPacket = 2; audioFormat.mBytesPerFrame = 2; // Apply format status = AudioUnitSetProperty(audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, kInputBus, &audioFormat, sizeof(audioFormat)); checkStatus(status); status = AudioUnitSetProperty(audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, kOutputBus, &audioFormat, sizeof(audioFormat)); checkStatus(status); // Set input callback AURenderCallbackStruct callbackStruct; callbackStruct.inputProc = recordingCallback; callbackStruct.inputProcRefCon = self; status = AudioUnitSetProperty(audioUnit, kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global, kInputBus, &callbackStruct, sizeof(callbackStruct)); checkStatus(status); // Set output callback callbackStruct.inputProc = playbackCallback; callbackStruct.inputProcRefCon = self; status = AudioUnitSetProperty(audioUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Global, kOutputBus, &callbackStruct, sizeof(callbackStruct)); checkStatus(status); // Allocate our own buffers (1 channel, 16 bits per sample, thus 16 bits per frame, thus 2 bytes per frame). // Practice learns the buffers used contain 512 frames, if this changes it will be fixed in processAudio. tempBuffer.mNumberChannels = 1; tempBuffer.mDataByteSize = 512 * 2; tempBuffer.mData = malloc( 512 * 2 ); // Initialise status = AudioUnitInitialize(audioUnit); checkStatus(status); return self; } ``` This callback is called when new audio data from the microphone is available. But never enter here when I test on the iPhone: ``` static OSStatus recordingCallback(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags, const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList *ioData) { AudioBuffer buffer; buffer.mNumberChannels = 1; buffer.mDataByteSize = inNumberFrames * 2; buffer.mData = malloc( inNumberFrames * 2 ); // Put buffer in a AudioBufferList AudioBufferList bufferList; bufferList.mNumberBuffers = 1; bufferList.mBuffers[0] = buffer; // Then: // Obtain recorded samples OSStatus status; status = AudioUnitRender([iosAudio audioUnit], ioActionFlags, inTimeStamp, inBusNumber, inNumberFrames, &bufferList); checkStatus(status); // Now, we have the samples we just read sitting in buffers in bufferList // Process the new data [iosAudio processAudio:&bufferList]; // release the malloc'ed data in the buffer we created earlier free(bufferList.mBuffers[0].mData); return noErr; } ```

Original source