I would like to use AudioQueueNewInput to create a stereo recording. I configured it as follows:
audioFormat.mFormatID = kAudioFormatLinearPCM;
hardwareChannels = 2;
audioFormat.mChannelsPerFrame = hardwareChannels;
audioFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked | kAudioFormatFlagIsBigEndian;
audioFormat.mFramesPerPacket = 1;
audioFormat.mBitsPerChannel = 16;
audioFormat.mBytesPerPacket = (audioFormat.mBitsPerChannel / 8) * hardwareChannels;
audioFormat.mBytesPerFrame = audioFormat.mBytesPerPacket;
OSStatus result = AudioQueueNewInput(
&audioFormat,
recordingCallback,
(__bridge void *)(self), // userData
NULL, // run loop
NULL, // run loop mode
0, // flags
&queueObject
);
AudioQueueStart (
queueObject,
NULL // start time. NULL means as soon as possible.
);
I tested this code on an iPhone 6s plus with an external stereo microphone. It does not seem to record stereo. Both the left and right channels get identical streams of data. What else do I need to do to record stereo?
Related
I am trying to capture app sound and pass it to AVAssetWriter as input.
I am setting callback for audio unit to get AudioBufferList.
The problem starts with converting AudioBufferList to CMSampleBufferRef.
It always return error -12731 which indicates that required parameter is missing
Thanks Karol
-(OSStatus) recordingCallbackWithRef:(void*)inRefCon
flags:(AudioUnitRenderActionFlags*)flags
timeStamp:(const AudioTimeStamp*)timeStamp
busNumber:(UInt32)busNumber
framesNumber:(UInt32)numberOfFrames
data:(AudioBufferList*)data
{
AudioBufferList bufferList;
bufferList.mNumberBuffers = 1;
bufferList.mBuffers[0].mData = NULL;
OSStatus status;
status = AudioUnitRender(audioUnit,
flags,
timeStamp,
busNumber,
numberOfFrames,
&bufferList);
[self checkOSStatus:status];
AudioStreamBasicDescription audioFormat;
// Describe format
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;
CMSampleBufferRef buff = NULL;
CMFormatDescriptionRef format = NULL;
CMSampleTimingInfo timing = { CMTimeMake(1, 44100), kCMTimeZero, kCMTimeInvalid };
status = CMAudioFormatDescriptionCreate(kCFAllocatorDefault, &audioFormat, 0, NULL, 0, NULL, NULL, &format);
[self checkOSStatus:status];
status = CMSampleBufferCreate(kCFAllocatorDefault,NULL,false,NULL,NULL,format,1, 1, &timing, 0, NULL, &buff);
[self checkOSStatus:status];
status = CMSampleBufferSetDataBufferFromAudioBufferList(buff,
kCFAllocatorDefault,
kCFAllocatorDefault,
0,
&bufferList);
[self checkOSStatus:status]; //Status here is 12731
//Do something with the buffer
return noErr;
}
Edit:
I checked bufferList.mBuffers[0].mData and it is not null so probably that's not a problem.
Since there is a similar question without answer all over the internet.
I managed to solve it and the recording fully works.
My problem was wrong parameter passed to CMSampleBufferCreate.
numSamples instead of 1 should be equal to numberOfFrames.
So the final call is:
CMSampleBufferCreate(kCFAllocatorDefault,NULL,false,NULL,NULL,format,
(CMItemCount)numberOfFrames, 1, &timing, 0, NULL, &buff);
I have an Audio App that from time to time is need to encode audio data from PCM to AAC format. I'm using software decoder (Actually I don't care what encoder is used, but I've checked twice, and there's software decoder). I'm using (https://github.com/TheAmazingAudioEngine/TheAmazingAudioEngine library)
I setup Audio session category as kAudioSessionCategory_PlayAndRecord
Have these formats
// Output format
outputFormat.mFormatID = kAudioFormatMPEG4AAC;
outputFormat.mSampleRate = 44100;
outputFormat.mFormatFlags = kMPEG4Object_AAC_Scalable;
outputFormat.mChannelsPerFrame = 2;
outputFormat.mBitsPerChannel = 0;
outputFormat.mBytesPerFrame = 0;
outputFormat.mBytesPerPacket = 0;
outputFormat.mFramesPerPacket = 1024;
// Input format
audioDescription.mFormatID = kAudioFormatLinearPCM;
audioDescription.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked | kAudioFormatFlagsNativeEndian | kAudioFormatFlagIsNonInterleaved;
audioDescription.mChannelsPerFrame = 2;
audioDescription.mBytesPerPacket = sizeof(SInt16);
audioDescription.mFramesPerPacket = 1;
audioDescription.mBytesPerFrame = sizeof(SInt16);
audioDescription.mBitsPerChannel = 8 * sizeof(SInt16);
audioDescription.mSampleRate = 44100.0;
And All works perfectly with kAudioSessionProperty_OverrideCategoryMixWithOthers == YES
But when I setup kAudioSessionProperty_OverrideCategoryMixWithOthers with NO then:
iOS Simulator - OK
iPod (6.1.3) - OK
iPhone 4S (7.0.3) - Fail with ('!sst' error on `AudioConverterFillComplexBuffer`) call
iPad 3(7.0.3) - Fail
As I already said all works fine until I change audio session property kAudioSessionProperty_OverrideCategoryMixWithOthers to NO
So the question is:
What is this error about? (I didn't found any clues in Headers and documentation what does this error mean (There's also no '!sst' string in any public framework header))
How can I fix it?
If you have any other Ideas that you think I need to try - Feel free to comment in.
The title pretty much sums up what I'm trying to achieve. I am trying to use Michael Tyson's TPCircularBuffer inside of a render callback while the circular buffer is getting filled with incoming audio data. I want to send the audio from the render callback to the output element of the RemoteIO audio unit so I can hear it through the device speakers.
The audio is interleaved stereo 16 bit coming in as packets of 2048 frames. Here's how I've set up my audio session:
#define kInputBus 1
#define kOutputBus 0
NSError *err = nil;
NSTimeInterval ioBufferDuration = 46;
AVAudioSession *session = [AVAudioSession sharedInstance];
[session setCategory:AVAudioSessionCategoryPlayback withOptions:AVAudioSessionCategoryOptionMixWithOthers error:&err];
[session setPreferredIOBufferDuration:ioBufferDuration error:&err];
[session setActive:YES error:&err];
AudioComponentDescription defaultOutputDescription;
defaultOutputDescription.componentType = kAudioUnitType_Output;
defaultOutputDescription.componentSubType = kAudioUnitSubType_RemoteIO;
defaultOutputDescription.componentManufacturer = kAudioUnitManufacturer_Apple;
defaultOutputDescription.componentFlags = 0;
defaultOutputDescription.componentFlagsMask = 0;
AudioComponent defaultOutput = AudioComponentFindNext(NULL, &defaultOutputDescription);
NSAssert(defaultOutput, #"Can't find default output.");
AudioComponentInstanceNew(defaultOutput, &remoteIOUnit);
UInt32 flag = 0;
OSStatus status = AudioUnitSetProperty(remoteIOUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, kOutputBus, &flag, sizeof(flag));
size_t bytesPerSample = sizeof(AudioUnitSampleType);
AudioStreamBasicDescription streamFormat = {0};
streamFormat.mSampleRate = 44100.00;
streamFormat.mFormatID = kAudioFormatLinearPCM;
streamFormat.mFormatFlags = kAudioFormatFlagsCanonical;
streamFormat.mBytesPerPacket = bytesPerSample;
streamFormat.mFramesPerPacket = 1;
streamFormat.mBytesPerFrame = bytesPerSample;
streamFormat.mChannelsPerFrame = 2;
streamFormat.mBitsPerChannel = bytesPerSample * 8;
streamFormat.mReserved = 0;
status = AudioUnitSetProperty(remoteIOUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, kInputBus, &streamFormat, sizeof(streamFormat));
AURenderCallbackStruct callbackStruct;
callbackStruct.inputProc = render;
callbackStruct.inputProcRefCon = self;
status = AudioUnitSetProperty(remoteIOUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Global, kOutputBus, &callbackStruct, sizeof(callbackStruct));
And here's where the audio data gets loaded into the circular buffer and used in the render callback:
#define kBufferLength 2048
-(void)loadBytes:(Byte *)byteArrPtr{
TPCircularBufferProduceBytes(&buffer, byteArrPtr, kBufferLength);
}
OSStatus render(
void *inRefCon,
AudioUnitRenderActionFlags *ioActionFlags,
const AudioTimeStamp *inTimeStamp,
UInt32 inBusNumber,
UInt32 inNumberFrames,
AudioBufferList *ioData)
{
AUDIOIO *audio = (__bridge AUDIOIO *)inRefCon;
AudioSampleType *outSample = (AudioSampleType *)ioData->mBuffers[0].mData;
//Zero outSample
memset(outSample, 0, kBufferLength);
int bytesToCopy = ioData->mBuffers[0].mDataByteSize;
SInt16 *targetBuffer = (SInt16 *)ioData->mBuffers[0].mData;
//Pull audio
int32_t availableBytes;
SInt16 *buffer = TPCircularBufferTail(&audio->buffer, &availableBytes);
memcpy(targetBuffer, buffer, MIN(bytesToCopy, availableBytes));
TPCircularBufferConsume(&audio->buffer, MIN(bytesToCopy, availableBytes));
return noErr;
}
There is something wrong with this setup because I am not getting any audio through the speakers, but I'm also not getting any errors when I test on my device. As far as I can tell the TPCircularBuffer is being filled and read from correctly. I've followed the Apple documentation for setting up the audio session. I am considering trying to set up an AUGraph next but I want to see if anyone could suggest a solution for what I'm trying to do here. Thanks!
For stereo (2 channels per frame), your bytes per frame and bytes per packet have to be twice your sample size in bytes. Same with bits per channel in terms of bits.
Added: If availableBytes/yourFrameSize isn't almost always as large or larger than inNumberFrames, you won't get much continuous sound.
At a glance, it looks like you've got everything set up correctly. You're missing a call to AudioOutputUnitStart() though:
...
// returns an OSStatus indicating success / fail
AudioOutputUnitStart(remoteIOUnit);
// now your callback should be being called
...
I believe one your problem is with using streamFormat.mBitsPerChannel = bytesPerSample * 8;
You assign bytesPerSample to be sizeof(AudioUnitSampleType) which is essentially 4 bytes.
So streamFormat.mBytesPerPacket = bytesPerSample; is ok.
But the assignment streamFormat.mBitsPerChannel = bytesPerSample * 8; is saying that you want 32 bits per sample instead of 16 bits per sample.
I would not create your audio format based on AudioUnitSampleType because this has nothing to do with your personal format that you want to utilize. I would create defines and do something like this:
#define BITS_PER_CHANNEL 16
#define SAMPLE_RATE 44100.0
#define CHANNELS_PER_FRAME 2
#define BYTES_PER_FRAME CHANNELS_PER_FRAME * (BITS_PER_CHANNEL / 8) //ie 4
#define FRAMES_PER_PACKET 1
#define BYTES_PER_PACKET FRAMES_PER_PACKET * BYTES_PER_FRAME
streamFormat.mSampleRate = SAMPLE_RATE; // 44100.0
streamFormat.mBitsPerChannel = BITS_PER_CHANNEL; //16
streamFormat.mChannelsPerFrame = CHANNELS_PER_FRAME; // 2
streamFormat.mFramesPerPacket = FRAMES_PER_PACKET; //1
streamFormat.mBytesPerFrame = BYTES_PER_FRAME; // 4 total, 2 for left ch, 2 for right ch
streamFormat.mBytesPerPacket = BYTES_PER_PACKET;
streamFormat.mReserved = 0;
streamFormat.mFormatID = kAudioFormatLinearPCM; // double check this also
streamFormat.mFormatFlags = kAudioFormatFlagsCanonical;`
You also need to look at the return values set to err and status immediately after each are run. You still need to add error checking at some of the calls as well such as
checkMyReturnValueToo = AudioComponentInstanceNew(defaultOutput, &remoteIOUnit);
You also have an extremely high value for your buffer duration. You have 46 and I am not sure where that came from. That means you want 46 seconds worth of audio during each audio callback. Usually you want something less than one second depending on your latency requirements. Most likely iOS will not use anything that high but you should try setting it to say 0.025 or so (25ms). You can try to lower it if you need faster latency.
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");
The canonical format for iOS filter audio units is 8.24 fixed-point (linear PCM), which is 32 bits per channel, not 16.
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
I am new to core audio and remote io. I need data of size 320 bytes which I encode and send. Also a minimum of 50 frames per second. Here is what I have done:
AudioComponentDescription desc;
desc.componentType = kAudioUnitType_Output;
desc.componentSubType = kAudioUnitSubType_RemoteIO;
desc.componentFlags = 0;
desc.componentFlagsMask = 0;
desc.componentManufacturer = 0;
// Get component
AudioComponent inputComponent = AudioComponentFindNext(NULL, &desc);
// Get audio units
AudioComponentInstanceNew(inputComponent, &audioUnit);
// Enable IO for recording
UInt32 flag = 1; AudioUnitSetProperty(audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, kInputBus, &flag, sizeof(flag));
// Enable IO for playback AudioUnitSetProperty(audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, kOutputBus, &flag, sizeof(flag));
UInt32 shouldAllocateBuffer = 1;
AudioUnitSetProperty(audioUnit, kAudioUnitProperty_ShouldAllocateBuffer, kAudioUnitScope_Global, 1, &shouldAllocateBuffer, sizeof(shouldAllocateBuffer));
// Describe format
audioFormat.mSampleRate = 8000.00;
audioFormat.mFormatID = kAudioFormatLinearPCM;
audioFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger|kAudioFormatFlagIsPacked;
audioFormat.mFramesPerPacket = 1;
audioFormat.mChannelsPerFrame = 1;
audioFormat.mBitsPerChannel = 16;
audioFormat.mBytesPerPacket = 2;
audioFormat.mBytesPerFrame = 2;
// Apply format AudioUnitSetProperty(audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 1, &audioFormat, sizeof(audioFormat));
AudioUnitSetProperty(audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &audioFormat, sizeof(audioFormat));
// Set input callback
AURenderCallbackStruct callbackStruct;
callbackStruct.inputProc = recordingCallback;
callbackStruct.inputProcRefCon = self;
AudioUnitSetProperty(audioUnit, kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global, 1, &callbackStruct, sizeof(callbackStruct));
// Set output callback
callbackStruct.inputProc = playbackCallback;
callbackStruct.inputProcRefCon = self;
AudioUnitSetProperty(audioUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Global, 0, &callbackStruct, sizeof(callbackStruct));
// Initialise
AudioUnitInitialize(audioUnit);
AudioOutputUnitStart(audioUnit);
With this settings, i get 186 frames in the callback method when tried with device.
I have allocated by buffer:
bufferList = (AudioBufferList*) malloc(sizeof(AudioBufferList));
bufferList->mNumberBuffers = 1; //mono input
for(UInt32 i=0;i<bufferList->mNumberBuffers;i++)
{
bufferList->mBuffers[i].mNumberChannels = 1;
bufferList->mBuffers[i].mDataByteSize = 2*186;
bufferList->mBuffers[i].mData = malloc(bufferList->mBuffers[i].mDataByteSize);
}
From this 372(2 x 186) bytes in the callback, i took 320 byte data and used as per my requirement. It is working, but very noisy.
Someone please help me. I am in big trouble.
A couple of suggestions -
Sample rate and buffer size get set using the AVAudioSession class.
386 is an unusual number of frames. Your callback is probably asking for 512 or 1024. You might try using a ring buffer to allow varying varying buffer size/ frame rates to suit your needs.
Here are some examples:
MixerHost
TimeCode