I want to reroute the input from line-in to A2DP. But I can't set line-in as input unfortunately. Is there any way to force line-in as input then outputting to A2DP?
Configuration below. Note the code near "set line in as preferred". I do believe this should set the input correctly - if it was available...
- (id) init {
self = [super init];
sizeof(allowBluetoothInput), &allowBluetoothInput);
// You can adjust the latency of RemoteIO (and, in fact, any other audio framework) by setting the kAudioSessionProperty_PreferredHardwareIOBufferDuration property
float aBufferLength = 0.005; // In seconds
AudioSessionSetProperty(kAudioSessionProperty_PreferredHardwareIOBufferDuration, sizeof(aBufferLength), &aBufferLength);
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;
// set line in as preferred
AVAudioSession *audioSession = [AVAudioSession sharedInstance];
NSString *preferredPortType = AVAudioSessionPortLineIn;
for (AVAudioSessionPortDescription *desc in audioSession.availableInputs) {
if ([desc.portType isEqualToString: AVAudioSessionPortLineIn] ||
// [desc.portType isEqualToString: AVAudioSessionPortBuiltInMic] ||
[desc.portType isEqualToString: AVAudioSessionPortHeadsetMic])
{
[audioSession setPreferredInput:desc error:nil];
}
}
// 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; // kAudioFormatMPEG4AAC_ELD
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);
// Disable buffer allocation for the recorder (optional - do this if we want to pass in our own)
flag = 0;
status = AudioUnitSetProperty(audioUnit,
kAudioUnitProperty_ShouldAllocateBuffer,
kAudioUnitScope_Output,
kInputBus,
&flag,
sizeof(flag));
// 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 );
[[AVAudioSession sharedInstance] setCategory:AVAudioSessionCategoryPlayAndRecord withOptions:AVAudioSessionCategoryOptionAllowBluetoothA2DP error:NULL];
// Initialise
AudioSessionSetActive(true);
NSTimeInterval outLat = [[AVAudioSession sharedInstance] outputLatency];
NSTimeInterval inLat = [[AVAudioSession sharedInstance] inputLatency];
status = AudioUnitInitialize(audioUnit);
checkStatus(status);
return self;
}
Related
I am using following code to record audio and writing it in file
- (void) initializeOutputUnit
{
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, &mAudioUnit);
// Enable IO for recording
UInt32 flag = 1;
//bhoomi
status = AudioUnitSetProperty(mAudioUnit,
kAudioOutputUnitProperty_EnableIO,
kAudioUnitScope_Input,
kInputBus,
&flag,
sizeof(flag));
// Enable IO for playback
status = AudioUnitSetProperty(mAudioUnit,
kAudioOutputUnitProperty_EnableIO,
kAudioUnitScope_Output,
kOutputBus,
&flag,
sizeof(flag));
// Describe format
AudioStreamBasicDescription audioFormat={0};
audioFormat.mSampleRate = kSampleRate;
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(mAudioUnit,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Output,
kInputBus,
&audioFormat,
sizeof(audioFormat));
status = AudioUnitSetProperty(mAudioUnit,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Input,
kOutputBus,
&audioFormat,
sizeof(audioFormat));
// Set input callback
AURenderCallbackStruct callbackStruct;
callbackStruct.inputProc = recordingCallback;
callbackStruct.inputProcRefCon = (__bridge void *)self;
status = AudioUnitSetProperty(mAudioUnit,
kAudioOutputUnitProperty_SetInputCallback,
kAudioUnitScope_Global,
kInputBus,
&callbackStruct,
sizeof(callbackStruct));
// Disable buffer allocation for the recorder (optional - do this if we want to pass in our own)
flag = 0;
status = AudioUnitSetProperty(mAudioUnit,
kAudioUnitProperty_ShouldAllocateBuffer,
kAudioUnitScope_Output,
kInputBus,
&flag,
sizeof(flag));
// On initialise le fichier audio
NSArray *paths = NSSearchPathForDirectoriesInDomains(NSDocumentDirectory, NSUserDomainMask, YES);
NSString *documentsDirectory = [paths objectAtIndex:0];
NSString *destinationFilePath = [[NSString alloc] initWithFormat: #"%#/output.caf", documentsDirectory];
NSLog(#">>> %#\n", destinationFilePath);
CFURLRef destinationURL = CFURLCreateWithFileSystemPath(kCFAllocatorDefault, (__bridge CFStringRef)destinationFilePath, kCFURLPOSIXPathStyle, false);
OSStatus setupErr = ExtAudioFileCreateWithURL(destinationURL, kAudioFileCAFType, &audioFormat, NULL, kAudioFileFlags_EraseFile, &mAudioFileRef);
CFRelease(destinationURL);
NSAssert(setupErr == noErr, #"Couldn't create file for writing");
setupErr = ExtAudioFileSetProperty(mAudioFileRef, kExtAudioFileProperty_ClientDataFormat, sizeof(AudioStreamBasicDescription), &audioFormat);
NSAssert(setupErr == noErr, #"Couldn't create file for format");
setupErr = ExtAudioFileWriteAsync(mAudioFileRef, 0, NULL);
NSAssert(setupErr == noErr, #"Couldn't initialize write buffers for audio file");
CheckError(AudioUnitInitialize(mAudioUnit), "AudioUnitInitialize");
CheckError(AudioOutputUnitStart(mAudioUnit), "AudioOutputUnitStart");
// [NSTimer scheduledTimerWithTimeInterval:5 target:self selector:#selector(stopRecording:) userInfo:nil repeats:NO];
}
In my application I have video call functionality using webRTC. Above code
is working fine with mic audio, mic audio is recorded properly but i am not able to record speaker sound(remote audio stream). Is it possible to record both sound (mic and speaker) in same file?.If yes, then how to modify code so it will also record sound from internal iphone speakers.
I'm setting up my audio input and audio output stream as below. The problem is that no matter how I set format and buffer sizes, the output latency for A2DP is always around 0.15s. I've posted the configuration below for. Here outLat is the measured output latency.
How can I decrease audio output latency further, without sacrificing audio quality?
- (id) init {
self = [super init];
sizeof(allowBluetoothInput), &allowBluetoothInput);
// You can adjust the latency of RemoteIO (and, in fact, any other audio framework) by setting the kAudioSessionProperty_PreferredHardwareIOBufferDuration property
float aBufferLength = 0.005; // In seconds
AudioSessionSetProperty(kAudioSessionProperty_PreferredHardwareIOBufferDuration, sizeof(aBufferLength), &aBufferLength);
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; // kAudioFormatMPEG4AAC_ELD
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);
// Disable buffer allocation for the recorder (optional - do this if we want to pass in our own)
flag = 0;
status = AudioUnitSetProperty(audioUnit,
kAudioUnitProperty_ShouldAllocateBuffer,
kAudioUnitScope_Output,
kInputBus,
&flag,
sizeof(flag));
// 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 );
[[AVAudioSession sharedInstance] setCategory:AVAudioSessionCategoryPlayAndRecord withOptions:AVAudioSessionCategoryOptionAllowBluetoothA2DP error:NULL];
// Initialise
AudioSessionSetActive(true);
NSTimeInterval outLat = [[AVAudioSession sharedInstance] outputLatency];
NSTimeInterval inLat = [[AVAudioSession sharedInstance] inputLatency];
status = AudioUnitInitialize(audioUnit);
checkStatus(status);
return self;
}
Use AVAudioSession in Measurement Mode and be aware of the Categories it is intended for and its limitations.
AVAudio Session Mode Measurement
I have encountered an error.
I just want kAudioFormatXXX not PCM, but there are errors.
Error: Error Domain=NSOSStatusErrorDomain Code=1718449215 "The operation
couldn’t be completed. (OSStatus error 1718449215.)"
Error Code responded 1718449215 in file
/Users/breaklee/Documents/project/ginav/ginav-
mobile/ginav-mobile/AudioProcessor.m on line 281
281 line is the end of code.
how can i initialize this code properly?
I want to use iLBC or other codecs not PCM.
AudioStreamBasicDescription audioFormat;
audioFormat.mSampleRate = SAMPLE_RATE;
audioFormat.mFormatID = kAudioFormatAC3;
audioFormat.mFormatFlags = kAudioFormatFlagIsPacked |
kAudioFormatFlagIsSignedInteger;
audioFormat.mFramesPerPacket = 1;
audioFormat.mChannelsPerFrame = 1;
audioFormat.mBitsPerChannel = 8;
audioFormat.mBytesPerPacket = 1;
audioFormat.mBytesPerFrame = 1;
// set the format on the output stream
status = AudioUnitSetProperty(_audioUnit,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Output,
kInputBus,
&audioFormat,
sizeof(audioFormat));
[self hasError:status file:__FILE__ line:__LINE__];
// set the format on the input stream
status = AudioUnitSetProperty(_audioUnit,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Input,
kOutputBus,
&audioFormat,
sizeof(audioFormat));
[self hasError:status file:__FILE__ line:__LINE__];
/**
We need to define a callback structure which holds
a pointer to the recordingCallback and a reference to
the audio processor object
*/
AURenderCallbackStruct callbackStruct;
// set recording callback
callbackStruct.inputProc = recordingCallback; // recordingCallback pointer
callbackStruct.inputProcRefCon = (__bridge void*)(self);
// set input callback to recording callback on the input bus
status = AudioUnitSetProperty(_audioUnit,
kAudioOutputUnitProperty_SetInputCallback,
kAudioUnitScope_Global,
kInputBus,
&callbackStruct,
sizeof(callbackStruct));
[self hasError:status file:__FILE__ line:__LINE__];
/*
We do the same on the output stream to hear what is coming
from the input stream
*/
callbackStruct.inputProc = playbackCallback;
callbackStruct.inputProcRefCon = (__bridge void*)(self);
// set playbackCallback as callback on our renderer for the output bus
status = AudioUnitSetProperty(_audioUnit,
kAudioUnitProperty_SetRenderCallback,
kAudioUnitScope_Global,
kOutputBus,
&callbackStruct,
sizeof(callbackStruct));
[self hasError:status file:__FILE__ line:__LINE__];
// reset flag to 0
flag = 0;
/*
we need to tell the audio unit to allocate the render buffer,
that we can directly write into it.
*/
status = AudioUnitSetProperty(_audioUnit,
kAudioUnitProperty_ShouldAllocateBuffer,
kAudioUnitScope_Output,
kInputBus,
&flag,
sizeof(flag));
// kAudioFormatUnsupportedDataFormatError
// Initialize the Audio Unit and cross fingers =)
status = AudioUnitInitialize(_audioUnit);
NSError *error = [NSError errorWithDomain:NSOSStatusErrorDomain
code:status
userInfo:nil];
NSLog(#"Error: %#", [error description]);
[self hasError:status file:__FILE__ line:__LINE__];
NSLog(#"Started");
I am working on a app which has following requirements:
Record real time audio from iOS device (iPhone/iPad) and send to server over network
Play received audio from network server on iOS device(iPhone/iPad)
Above mentioned things need to be done simultaneously.
I have used AudioUnit for this.
I have run into a problem where I am hearing same audio what i speak into iPhone Mic instead of audio received from network server.
I have searched a lot on how to avoid this but haven't got the solution.
If anyone has had same problem and found any solution, sharing it will help a lot.
here is my code for initializing audio Unit
-(void)initializeAudioUnit
{
audioUnit = NULL;
// Describe audio component
AudioComponentDescription desc;
desc.componentType = kAudioUnitType_Output;
desc.componentSubType = kAudioUnitSubType_VoiceProcessingIO;
desc.componentFlags = 0;
desc.componentFlagsMask = 0;
desc.componentManufacturer = kAudioUnitManufacturer_Apple;
// Get component
AudioComponent inputComponent = AudioComponentFindNext(NULL, &desc);
// Get audio units
status = AudioComponentInstanceNew(inputComponent, &audioUnit);
UInt32 flag = 1;
//enable IO for recording
status = AudioUnitSetProperty(audioUnit,
kAudioOutputUnitProperty_EnableIO,
kAudioUnitScope_Input,
kInputBus,
&flag,
sizeof(flag));
status = AudioUnitSetProperty(audioUnit, kAudioOutputUnitProperty_EnableIO,
kAudioUnitScope_Output,
kOutputBus,
&flag,
sizeof(flag));
AudioStreamBasicDescription audioStreamBasicDescription;
// Describe format
audioStreamBasicDescription.mSampleRate = 16000;
audioStreamBasicDescription.mFormatID = kAudioFormatLinearPCM;
audioStreamBasicDescription.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked |kLinearPCMFormatFlagIsNonInterleaved;
audioStreamBasicDescription.mFramesPerPacket = 1;
audioStreamBasicDescription.mChannelsPerFrame = 1;
audioStreamBasicDescription.mBitsPerChannel = 16;
audioStreamBasicDescription.mBytesPerPacket = 2;
audioStreamBasicDescription.mBytesPerFrame = 2;
status = AudioUnitSetProperty(audioUnit,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Output,
kInputBus,
&audioStreamBasicDescription,
sizeof(audioStreamBasicDescription));
NSLog(#"Status[%d]",(int)status);
status = AudioUnitSetProperty(audioUnit,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Input,
kOutputBus,
&audioStreamBasicDescription,
sizeof(audioStreamBasicDescription));
NSLog(#"Status[%d]",(int)status);
AURenderCallbackStruct callbackStruct;
// Set input callback
callbackStruct.inputProc = recordingCallback;
callbackStruct.inputProcRefCon = (__bridge void *)(self);
status = AudioUnitSetProperty(audioUnit,
kAudioOutputUnitProperty_SetInputCallback,
kAudioUnitScope_Global,
kInputBus,
&callbackStruct,
sizeof(callbackStruct));
callbackStruct.inputProc = playbackCallback;
callbackStruct.inputProcRefCon = (__bridge void *)(self);
status = AudioUnitSetProperty(audioUnit,
kAudioUnitProperty_SetRenderCallback,
kAudioUnitScope_Global,
kOutputBus,
&callbackStruct,
sizeof(callbackStruct));
flag=0;
status = AudioUnitSetProperty(audioUnit,
kAudioUnitProperty_ShouldAllocateBuffer,
kAudioUnitScope_Output,
kInputBus,
&flag,
sizeof(flag));
}
Recording Call Back
static OSStatus recordingCallback (void *inRefCon,AudioUnitRenderActionFlags *ioActionFlags,const AudioTimeStamp *inTimeStamp,UInt32 inBusNumber,UInt32 inNumberFrames,AudioBufferList *ioData)
{
MyAudioViewController *THIS = (__bridge MyAudioViewController *)inRefCon;
AudioBuffer tempBuffer;
tempBuffer.mNumberChannels = 1;
tempBuffer.mDataByteSize = inNumberFrames * 2;
tempBuffer.mData = malloc(inNumberFrames *2);
AudioBufferList bufferList;
bufferList.mNumberBuffers = 1;
bufferList.mBuffers[0] = tempBuffer;
OSStatus status;
status = AudioUnitRender(THIS->audioUnit,
ioActionFlags,
inTimeStamp,
kInputBus,
inNumberFrames,
&bufferList);
if (noErr != status) {
printf("AudioUnitRender error: %d", (int)status);
return noErr;
}
tempBuffer.mDataByteSize, &encodedSize,(__bridge void *)(THIS));
[THIS processAudio:&bufferList];
free(bufferList.mBuffers[0].mData);
return noErr;
}
Playback Call Back
static OSStatus playbackCallback(void *inRefCon,AudioUnitRenderActionFlags *ioActionFlags,const AudioTimeStamp *inTimeStamp,UInt32 inBusNumber,UInt32 inNumberFrames,AudioBufferList *ioData) {
NSLog(#"In play back call back");
MyAudioViewController *THIS = (__bridge MyAudioViewController *)inRefCon;
int32_t availableBytes=0;
char *inBuffer = GetDataFromCircularBuffer(&THIS->mybuffer, &availableBytes);
NSLog(#"bytes available in buffer[%d]",availableBytes);
decodeSpeexData(inBuffer, availableBytes,(__bridge void *)(THIS));
ConsumeReadBytes(&(THIS->mybuffer), availableBytes);
memcpy(targetBuffer, THIS->outTemp, inNumberFrames*2);
return noErr;
}
Process Audio recorded from MIC
- (void) processAudio: (AudioBufferList*) bufferList
{
AudioBuffer sourceBuffer = bufferList->mBuffers[0];
// NSLog(#"Origin size: %d", (int)sourceBuffer.mDataByteSize);
int size = 0;
encodeAudioDataSpeex((spx_int16_t*)sourceBuffer.mData, sourceBuffer.mDataByteSize, &size, (__bridge void *)(self));
[self performSelectorOnMainThread:#selector(SendAudioData:) withObject:[NSData dataWithBytes:self->jitterBuffer length:size] waitUntilDone:NO];
NSLog(#"Encoded size: %i", size);
}
Your playbackCallback render callback, which you have not shown, is responsible for the audio that is sent to the RemoteIO speaker output. If this RemoteIO render callback puts no data in its callback buffers, whatever junk that was left in buffers (stuff that was previously in the record callback buffers perhaps) might be sent to the speaker instead.
Also, it is strongly recommended by Apple DTS that your recordingCallback not include any memory management calls, such as malloc(). So this may be a bug helping cause the problem as well.
I am developing an application in which i want to mix multiple sounds to create a single audio file. I'm able to mix the multiple audio files to generate single audio using apple's MixerHost example, but am not able to record this audio to generate single audio file.
I referred This link to record the generated audio. With reference to the mentioned link i am able to generate audio file, but its silent(there is not sound).
Follwing is my code :
-(void) initializeOutputUnit
{
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);
// Enable IO for recording
UInt32 flag = 1;
status = AudioUnitSetProperty(audioUnit,
kAudioOutputUnitProperty_EnableIO,
kAudioUnitScope_Input,
kInputBus,
&flag,
sizeof(flag));
// Enable IO for playback
status = AudioUnitSetProperty(audioUnit,
kAudioOutputUnitProperty_EnableIO,
kAudioUnitScope_Output,
kOutputBus,
&flag,
sizeof(flag));
// Describe format
AudioStreamBasicDescription audioFormat={0};
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));
status = AudioUnitSetProperty(audioUnit,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Input,
kOutputBus,
&audioFormat,
sizeof(audioFormat));
// Set input callback
AURenderCallbackStruct callbackStruct;
callbackStruct.inputProc = recordingCallback;
callbackStruct.inputProcRefCon = self;
status = AudioUnitSetProperty(audioUnit,
kAudioOutputUnitProperty_SetInputCallback,
kAudioUnitScope_Global,
kInputBus,
&callbackStruct,
sizeof(callbackStruct));
// Disable buffer allocation for the recorder (optional - do this if we want to pass in our own)
flag = 0;
status = AudioUnitSetProperty(audioUnit,
kAudioUnitProperty_ShouldAllocateBuffer,
kAudioUnitScope_Output,
kInputBus,
&flag,
sizeof(flag));
AudioUnitInitialize(audioUnit);
AudioOutputUnitStart(audioUnit);
// On initialise le fichier audio
NSArray *paths = NSSearchPathForDirectoriesInDomains(NSDocumentDirectory, NSUserDomainMask, YES);
NSString *documentsDirectory = [paths objectAtIndex:0];
NSString *destinationFilePath = [[NSString alloc] initWithFormat: #"%#/output.caf", documentsDirectory];
NSLog(#">>> %#\n", destinationFilePath);
CFURLRef destinationURL = CFURLCreateWithFileSystemPath(kCFAllocatorDefault, (__bridge CFStringRef)destinationFilePath, kCFURLPOSIXPathStyle, false);
OSStatus setupErr = ExtAudioFileCreateWithURL(destinationURL, kAudioFileCAFType, &audioFormat, NULL, kAudioFileFlags_EraseFile, &mAudioFileRef);
CFRelease(destinationURL);
NSAssert(setupErr == noErr, #"Couldn't create file for writing");
setupErr = ExtAudioFileSetProperty(mAudioFileRef, kExtAudioFileProperty_ClientDataFormat, sizeof(AudioStreamBasicDescription), &audioFormat);
NSAssert(setupErr == noErr, #"Couldn't create file for format");
setupErr = ExtAudioFileWriteAsync(mAudioFileRef, 0, NULL);
NSAssert(setupErr == noErr, #"Couldn't initialize write buffers for audio file");}
static OSStatus recordingCallback (void * inRefCon,
AudioUnitRenderActionFlags * ioActionFlags,
const AudioTimeStamp * inTimeStamp,
UInt32 inBusNumber,
UInt32 inNumberFrames,
AudioBufferList * ioData)
{
AudioBufferList bufferList;
SInt16 samples[inNumberFrames]; // A large enough size to not have to worry about buffer overrun
memset (&samples, 0, sizeof (samples));
bufferList.mNumberBuffers = 1;
bufferList.mBuffers[0].mData = samples;
bufferList.mBuffers[0].mNumberChannels = 1;
bufferList.mBuffers[0].mDataByteSize = inNumberFrames*sizeof(SInt16);
AudioMixerViewController* THIS = (__bridge AudioMixerViewController *)inRefCon;
OSStatus status;
status = AudioUnitRender(THIS->audioUnit,
ioActionFlags,
inTimeStamp,
kInputBus,
inNumberFrames,
&bufferList);
if (noErr != status) {
printf("AudioUnitRender error: %ld", status);
return noErr;
}
// Now, we have the samples we just read sitting in buffers in bufferList
ExtAudioFileWriteAsync(THIS->mAudioFileRef, inNumberFrames, &bufferList);
return noErr;
}
I am getting OSStatus(ExtAudioFileDispose): 0 and OSStatus(ExtAudioFileDispose): -50
Please help me. Thanks!