I have an audio interface with two inputs: microphone and instrument. In Windows it is represented by a single stereo audio input, left channel gets microphone input and right channel gets instrument input.
I open recording session like this:
void CALLBACK waveIn(HWAVEIN hwi, UINT uMsg, DWORD_PTR dwInstance, DWORD_PTR dwParam1, DWORD_PTR dwParam2);
// ...
const int nchannels = 2;
const int sr = 48000;
HWAVEIN hwi;
WAVEFORMATEX wfe =
{
WAVE_FORMAT_IEEE_FLOAT, // wFormatTag
nchannels, // nChannels
sr, // nSamplesPerSec
sr * sizeof(float) * nchannels, // nAvgBytesPerSec
sizeof(float) * nchannels, // nBlockAlign
32, // wBitsPerSample
0 // cbSize
};
MMRESULT openResult = waveInOpen(&hwi, 0, &wfe, (DWORD_PTR)waveIn, 0, CALLBACK_FUNCTION);
// ..set up buffers..
MMRESULT startResult = waveInStart(hwi);
Then process samples in waveIn function. It works fine, but I only need to work with instrument (right channel) data and not interested in microphone. I tried setting nchannels=1 in the code above, hoping it would mix both channels, but instead I only get data from microphone (left) channel. Is there a way to set up recording from a stereo input to get only right (or mix of both) audio channel data (as mono recording)?
Related
I am hearing a very loud and harsh distortion sound when I run this simple application. I am simply instantiating a default output unit and assign a render callback. And letting the program run in the runloop. I have detected no errors from Core Audio and everything works as usual except for this distortion.
#import <AudioToolbox/AudioToolbox.h>
OSStatus render1(void *inRefCon,
AudioUnitRenderActionFlags *ioActionFlags,
const AudioTimeStamp *inTimeStamp,
UInt32 inBusNumber,
UInt32 inNumberFrames,
AudioBufferList * ioData)
{
return noErr;
}
int main(int argc, const char * argv[]) {
AudioUnit timerAU;
UInt32 propsize = 0;
AudioComponentDescription outputUnitDesc;
outputUnitDesc.componentType = kAudioUnitType_Output;
outputUnitDesc.componentSubType = kAudioUnitSubType_DefaultOutput;
outputUnitDesc.componentManufacturer = kAudioUnitManufacturer_Apple;
outputUnitDesc.componentFlags = 0;
outputUnitDesc.componentFlagsMask = 0;
//Get RemoteIO AU from Audio Unit Component Manager
AudioComponent outputComp = AudioComponentFindNext(NULL, &outputUnitDesc);
if (outputComp == NULL) exit (-1);
CheckError(AudioComponentInstanceNew(outputComp, &timerAU), "comp");
//Set up render callback function for the RemoteIO AU.
AURenderCallbackStruct renderCallbackStruct;
renderCallbackStruct.inputProc = render1;
renderCallbackStruct.inputProcRefCon = nil;//(__bridge void *)(self);
propsize = sizeof(renderCallbackStruct);
CheckError(AudioUnitSetProperty(timerAU,
kAudioUnitProperty_SetRenderCallback,
kAudioUnitScope_Global,
0,
&renderCallbackStruct,
propsize), "set render");
CheckError(AudioUnitInitialize(timerAU), "init");
// tickMethod = completion;
CheckError(AudioOutputUnitStart(timerAU), "start");
CFRunLoopRunInMode(kCFRunLoopDefaultMode, 1000, false);
}
Your question does not seem complete. I don't know about the side effects of silencing the output noise which is probably just undefined behavior. I also don't know what your code would serve for as such. There is an unfinished render callback on the kAudioUnitSubType_DefaultOutput which does nothing (it is not generating silence!). I know for two ways of silencing it.
In the callback the ioData buffers have to be explicitly filled with zeroes, because there's no guarantee they will be initialized empty:
Float32 * lBuffer0;
Float32 * lBuffer1;
lBuffer0 = (Float32 *)ioData->mBuffers[0].mData;
lBuffer1 = (Float32 *)ioData->mBuffers[1].mData;
memset(lBuffer0, 0, inNumberFrames*sizeof(Float32));
memset(lBuffer1, 0, inNumberFrames*sizeof(Float32));
Other possibility is to leave the unfinished callback as it is, but declare the timerAU to be of outputUnitDesc.componentSubType = kAudioUnitSubType_HALOutput; instead of
outputUnitDesc.componentSubType = kAudioUnitSubType_DefaultOutput;
and explicity disable I/O before setting the render callback by means of following code:
UInt32 lEnableIO = 0;
CheckError(AudioUnitSetProperty(timerAU,
kAudioOutputUnitProperty_EnableIO,
kAudioUnitScope_Output,
0, //output element
&lEnableIO,
sizeof(lEnableIO)),
"couldn't disable output");
I would strongly encourage into studying thoroughly the CoreAudio API and understanding how to set up an audio unit. This is crucial in understanding the matter. I've seen in your code a comment mentioning a RemoteIO AU. There is nothing like a RemoteIO AU in OSX. In case you're attempting a port from iOS code, please try learning the differences. They are well documented.
I am attempting to write incoming mic audio to a file. Because the audio samples are delivered 4096 frames (the set frame rate for my project) at a time in a time-critical callback I cannot simply write the bytes to a file with AudioFileWriteBytes. I also did not wish to go through the effort and complexity of setting up my own ring buffer to store samples to write elsewhere. So I am using Extended Audio File API for its ExtAudioFileWriteAsync function.
As per instructed by the documentation I create the ExtAudioFileRef with a CFURL and than run it once with a null buffer and 0 frames in main. Then I initiate my AUHAL unit and the input callback begins to be called.
ExtAudioFileWriteAsync(player.recordFile, 0, NULL);
There I have my code to write to this file asynchronously. I have the call nested in a dispatch queue so that it runs after the callback function exits scope (tho not sure if that is necessary but I get this error with all without the enclosing dispatch block). This is the callback as it is right now.
OSStatus InputRenderProc(void *inRefCon,
AudioUnitRenderActionFlags *ioActionFlags,
const AudioTimeStamp *inTimeStamp,
UInt32 inBusNumber,
UInt32 inNumberFrames,
AudioBufferList * ioData)
{
MyAUGraphPlayer *player = (MyAUGraphPlayer*) inRefCon;
// rendering incoming mic samples to player->inputBuffer
OSStatus inputProcErr = noErr;
inputProcErr = AudioUnitRender(player->inputUnit,
ioActionFlags,
inTimeStamp,
inBusNumber,
inNumberFrames,
player->inputBuffer);
printf("%i", inNumberFrames);
dispatch_async(player->fileWritingQueue, ^{
ExtAudioFileWriteAsync(player->recordFile, 4096, player->inputBuffer);
});
return inputProcErr;
}
It immediately bails out with the bad access exception on the first callback invocation. For clarity these are the settings I have for creating the file to begin with.
// describe a PCM format for audio file
AudioStreamBasicDescription format = { 0 };
format.mBytesPerFrame = 4;
format.mBytesPerPacket = 4;
format.mChannelsPerFrame = 2;
format.mBitsPerChannel = 16;
format.mFramesPerPacket = 1;
format.mFormatFlags = kAudioFormatFlagIsPacked | kAudioFormatFlagIsFloat;
format.mFormatID = kAudioFormatLinearPCM;
CFURLRef myFileURL = CFURLCreateWithFileSystemPath(kCFAllocatorDefault, CFSTR("./test2.wav"), kCFURLPOSIXPathStyle, false);
ExtAudioFileCreateWithURL(myFileURL,
kAudioFileWAVEType,
&format,
NULL,
kAudioFileFlags_EraseFile,
&player.recordFile);
player.fileWritingQueue = dispatch_queue_create("myQueue", NULL);
ExtAudioFileWriteAsync(player.recordFile, 0, NULL);
I need access to audio data from microphone on macbook. I have the an example program for recording microphone data based on the one in "Learning Core Audio". When I run this program and break on the call back routine I see the inBuffer pointer and the mAudioData pointer. However I am having a heck of a time making sense of the data. I've tried casting the void* pointer to mAudioData to SInt16, to SInt32 and to float and tried a number of endian conversions all with nonsense looking results. What I need to know definitively is the number format for the data in the buffer. The example actually works writing microphone data to a file which I can play so I know that real audio is being recorded.
AudioStreamBasicDescription recordFormat;
memset(&recordFormat,0,sizeof(recordFormat));
//recordFormat.mFormatID = kAudioFormatMPEG4AAC;
recordFormat.mFormatID = kAudioFormatLinearPCM;
recordFormat.mChannelsPerFrame = 2;
recordFormat.mBitsPerChannel = 16;
recordFormat.mBytesPerPacket = recordFormat.mBytesPerFrame = recordFormat.mChannelsPerFrame * sizeof(SInt16);
recordFormat.mFramesPerPacket = 1;
MyGetDefaultInputDeviceSampleRate(&recordFormat.mSampleRate);
UInt32 propSize = sizeof(recordFormat);
CheckError(AudioFormatGetProperty(kAudioFormatProperty_FormatInfo,
0,
NULL,
&propSize,
&recordFormat),
"AudioFormatProperty failed");
//set up queue
AudioQueueRef queue = {0};
CheckError(AudioQueueNewInput(&recordFormat,
MyAQInputCallback,
&recorder,
NULL,
kCFRunLoopCommonModes,
0,
&queue),
"AudioQueueNewInput failed");
UInt32 size = sizeof(recordFormat);
CheckError(AudioQueueGetProperty(queue,
kAudioConverterCurrentOutputStreamDescription,
&recordFormat,
&size), "Couldn't get queue's format");
In this sample app, I was able to load in a file of stereo data and play it using the simulator. But it doesn't work on the device. I tried using a sound editor and convert the stereo clip to mono and changing the descriptor settings and it will work mono only. I had a hard time trying to find out why, I am guessing it has to do with my descriptor configuration problems.
This sample app is at https://github.com/peter7777usa/TestAudio
The PlayBack Function
static OSStatus playbackCallback(void *inRefCon,
AudioUnitRenderActionFlags *ioActionFlags,
const AudioTimeStamp *inTimeStamp,
UInt32 inBusNumber,
UInt32 inNumberFrames,
AudioBufferList *ioData) {
// Notes: ioData contains buffers (may be more than one!)
// Fill them up as much as you can. Remember to set the size value in each buffer to match how
// much data is in the buffer.
NSLog(#"muffers %d", ioData->mNumberBuffers);
UInt32 size = 2048;
if (iosAudio->incomingCircularBuffer.fillCount>size){
NSLog(#"Playing %d", iosAudio->incomingCircularBuffer.fillCount);
iosAudio.pkgtotal -=2;
int32_t availableBytes;
SInt16 *databuffer = TPCircularBufferTail(&iosAudio->incomingCircularBuffer, &availableBytes);
memcpy(ioData->mBuffers[0].mData, databuffer, size);
ioData->mBuffers[0].mDataByteSize = size; // indicate how much data we wrote in the buffer
TPCircularBufferConsume(&iosAudio->incomingCircularBuffer, size);
}else{
}
return noErr;
}
The AudioStreamDescription
// Describe format
AudioStreamBasicDescription audioFormat;
bzero(&audioFormat, sizeof(AudioStreamBasicDescription));
UInt32 channelCount = 2;
UInt32 sampleSize = sizeof(UInt16);
audioFormat.mSampleRate = 44100.00;
audioFormat.mFormatID = kAudioFormatLinearPCM;
audioFormat.mFormatFlags = kAudioFormatFlagsCanonical;;
audioFormat.mFramesPerPacket = 1;
audioFormat.mChannelsPerFrame = channelCount;
audioFormat.mBitsPerChannel = sampleSize * 8;
audioFormat.mBytesPerPacket = sampleSize * channelCount;
audioFormat.mBytesPerFrame = sampleSize * channelCount;
// Apply format
status = AudioUnitSetProperty(audioUnit,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Output,
kInputBus,
&audioFormat,
sizeof(audioFormat));
I have a remoteIO application that loads and plays samples on iOS. It works fine when built with xcode5. I use iOS7 as a deployment target.
My application was originally built using the AudioUnitSampleType audio format and the kAudioFormatFlagsCanonical format flags. My sample files are 16 bits/44100Hz/Mono/Caf files.
Now I want to run it on iOS8.
Building my app with its original code on xcode6, the app runs fine on an iOS7 device but it produces no sounds on an iOS8 device.
As AudioUnitSampleType and kAudioFormatFlagsCanonical are deprecated in iOS8, I replaced them, after some researches, with float and kAudioFormatFlagIsFloat | kAudioFormatFlagIsPacked | kAudioFormatFlagIsNonInterleaved.
Now my app runs fine on iOS8 but the sounds are saturated on iOS7.
Has anyone experiences this? Any help ? Thanks, I am stuck here.
Pascal
ps : here is my sample loading method
#define AUDIO_SAMPLE_TYPE float
#define AUDIO_FORMAT_FLAGS kAudioFormatFlagIsFloat | kAudioFormatFlagIsPacked | kAudioFormatFlagIsNonInterleaved
-(void) load:(NSURL*)fileNameURL{
if (frameCount>0){
if (leftChannel!= NULL){
free (leftChannel);
leftChannel = 0;
}
if (rightChannel != NULL){
free (rightChannel);
rightChannel = 0;
}
}
soundFileURLRef=(CFURLRef)fileNameURL;
//----------------------------------------------
// 1.[OPEN AUDIO FILE] and associate it with the extended audio file object.
//----------------------------------------------
ExtAudioFileRef audioFileExtendedObject = 0;
log_if_err(ExtAudioFileOpenURL((CFURLRef)soundFileURLRef,
&audioFileExtendedObject),
#"ExtAudioFileOpenURL failed");
//----------------------------------------------
// 2.[AUDIO FILE LENGTH] Get the audio file's length in frames.
//----------------------------------------------
UInt64 totalFramesInFile = 0;
UInt32 frameLengthPropertySize = sizeof (totalFramesInFile);
log_if_err(ExtAudioFileGetProperty(audioFileExtendedObject,
kExtAudioFileProperty_FileLengthFrames,
&frameLengthPropertySize,
&totalFramesInFile),
#"ExtAudioFileGetProperty (audio file length in frames) failed");
frameCount = totalFramesInFile;
//----------------------------------------------
// 3.[AUDIO FILE FORMAT] Get the audio file's number of channels. Normally CAF.
//----------------------------------------------
AudioStreamBasicDescription fileAudioFormat = {0};
UInt32 formatPropertySize = sizeof (fileAudioFormat);
log_if_err(ExtAudioFileGetProperty(audioFileExtendedObject,
kExtAudioFileProperty_FileDataFormat,
&formatPropertySize,
&fileAudioFormat),
#"ExtAudioFileGetProperty (file audio format) failed");
//----------------------------------------------
// 4.[ALLOCATE AUDIO FILE MEMORY] Allocate memory in the soundFiles instance
// variable to hold the left channel, or mono, audio data
//----------------------------------------------
UInt32 channelCount = fileAudioFormat.mChannelsPerFrame;
// DLog(#"fileNameURL=%# | channelCount=%d",fileNameURL,(int)channelCount);
if (leftChannel != NULL){
free (leftChannel);
leftChannel = 0;
}
leftChannel =(AUDIO_UNIT_SAMPLE_TYPE *) calloc (totalFramesInFile, sizeof(AUDIO_UNIT_SAMPLE_TYPE));
AudioStreamBasicDescription importFormat = {0};
if (2==channelCount) {
isStereo = YES;
if (rightChannel != NULL){
free (rightChannel);
rightChannel = 0;
}
rightChannel = (AUDIO_UNIT_SAMPLE_TYPE *) calloc (totalFramesInFile, sizeof (AUDIO_UNIT_SAMPLE_TYPE));
importFormat = stereoStreamFormat;
} else if (1==channelCount) {
isStereo = NO;
importFormat = monoStreamFormat;
} else {
ExtAudioFileDispose (audioFileExtendedObject);
return;
}
//----------------------------------------------
// 5.[ASSIGN THE MIXER INPUT BUS STREAM DATA FORMAT TO THE AUDIO FILE]
// Assign the appropriate mixer input bus stream data format to the extended audio
// file object. This is the format used for the audio data placed into the audio
// buffer in the SoundStruct data structure, which is in turn used in the
// inputRenderCallback callback function.
//----------------------------------------------
UInt32 importFormatPropertySize = (UInt32) sizeof (importFormat);
log_if_err(ExtAudioFileSetProperty(audioFileExtendedObject,
kExtAudioFileProperty_ClientDataFormat,
importFormatPropertySize,
&importFormat),
#"ExtAudioFileSetProperty (client data format) failed");
//----------------------------------------------
// 6.[SET THE AUDIBUFFER LIST STRUCT] which has two roles:
//
// 1. It gives the ExtAudioFileRead function the configuration it
// needs to correctly provide the data to the buffer.
//
// 2. It points to the soundFiles[soundFile].leftChannel buffer, so
// that audio data obtained from disk using the ExtAudioFileRead function
// goes to that buffer
//
// Allocate memory for the buffer list struct according to the number of
// channels it represents.
//----------------------------------------------
AudioBufferList *bufferList;
bufferList = (AudioBufferList *) malloc(sizeof(AudioBufferList)+sizeof(AudioBuffer)*(channelCount-1));
if (NULL==bufferList){
NSLog(#"*** malloc failure for allocating bufferList memory");
return;
}
//----------------------------------------------
// 7.initialize the mNumberBuffers member
//----------------------------------------------
bufferList->mNumberBuffers = channelCount;
//----------------------------------------------
// 8.initialize the mBuffers member to 0
//----------------------------------------------
AudioBuffer emptyBuffer = {0};
size_t arrayIndex;
for (arrayIndex = 0; arrayIndex < channelCount; arrayIndex++) {
bufferList->mBuffers[arrayIndex] = emptyBuffer;
}
//----------------------------------------------
// 9.set up the AudioBuffer structs in the buffer list
//----------------------------------------------
bufferList->mBuffers[0].mNumberChannels = 1;
bufferList->mBuffers[0].mDataByteSize = totalFramesInFile * sizeof (AUDIO_UNIT_SAMPLE_TYPE);
bufferList->mBuffers[0].mData = leftChannel;
if (channelCount==2){
bufferList->mBuffers[1].mNumberChannels = 1;
bufferList->mBuffers[1].mDataByteSize = totalFramesInFile * sizeof (AUDIO_UNIT_SAMPLE_TYPE);
bufferList->mBuffers[1].mData = rightChannel;
}
//----------------------------------------------
// 10.Perform a synchronous, sequential read of the audio data out of the file and
// into the "soundFiles[soundFile].leftChannel" and (if stereo) ".rightChannel" members.
//----------------------------------------------
UInt32 numberOfPacketsToRead = (UInt32) totalFramesInFile;
OSStatus result = ExtAudioFileRead (audioFileExtendedObject,
&numberOfPacketsToRead,
bufferList);
free (bufferList);
if (noErr != result) {
log_if_err(result,#"ExtAudioFileRead failure");
//
// If reading from the file failed, then free the memory for the sound buffer.
//
free (leftChannel);
leftChannel = 0;
if (2==channelCount) {
free (rightChannel);
rightChannel = 0;
}
frameCount = 0;
}
//----------------------------------------------
// Dispose of the extended audio file object, which also
// closes the associated file.
//----------------------------------------------
ExtAudioFileDispose (audioFileExtendedObject);
return;
}