core audio offline rendering GenericOutput
Offline rendering Worked for me using GenericOutput AudioUnit.
I am sharing the working code here.
core-audio framework seems a little though. But small-small things in it like ASBD, parameters ...etc are making these issues. try hard it will work. Don't give-up :-). core-audio is very powerful and useful while dealing with low-level audio. Thats what I learned from these last weeks. Enjoy :-D ....
Declare these in .h
//AUGraph
AUGraph mGraph;
//Audio Unit References
AudioUnit mFilePlayer;
AudioUnit mFilePlayer2;
AudioUnit mReverb;
AudioUnit mTone;
AudioUnit mMixer;
AudioUnit mGIO;
//Audio File Location
AudioFileID inputFile;
AudioFileID inputFile2;
//Audio file refereces for saving
ExtAudioFileRef extAudioFile;
//Standard sample rate
Float64 graphSampleRate;
AudioStreamBasicDescription stereoStreamFormat864;
Float64 MaxSampleTime;
//in .m class
- (id) init
{
self = [super init];
graphSampleRate = 44100.0;
MaxSampleTime = 0.0;
UInt32 category = kAudioSessionCategory_MediaPlayback;
CheckError(AudioSessionSetProperty(kAudioSessionProperty_AudioCategory,
sizeof(category),
&category),
"Couldn't set category on audio session");
[self initializeAUGraph];
return self;
}
//ASBD setup
- (void) setupStereoStream864 {
// The AudioUnitSampleType data type is the recommended type for sample data in audio
// units. This obtains the byte size of the type for use in filling in the ASBD.
size_t bytesPerSample = sizeof (AudioUnitSampleType);
// Fill the application audio format struct's fields to define a linear PCM,
// stereo, noninterleaved stream at the hardware sample rate.
stereoStreamFormat864.mFormatID = kAudioFormatLinearPCM;
stereoStreamFormat864.mFormatFlags = kAudioFormatFlagsAudioUnitCanonical;
stereoStreamFormat864.mBytesPerPacket = bytesPerSample;
stereoStreamFormat864.mFramesPerPacket = 1;
stereoStreamFormat864.mBytesPerFrame = bytesPerSample;
stereoStreamFormat864.mChannelsPerFrame = 2; // 2 indicates stereo
stereoStreamFormat864.mBitsPerChannel = 8 * bytesPerSample;
stereoStreamFormat864.mSampleRate = graphSampleRate;
}
//AUGraph setup
- (void)initializeAUGraph
{
[self setupStereoStream864];
// Setup the AUGraph, add AUNodes, and make connections
// create a new AUGraph
CheckError(NewAUGraph(&mGraph),"Couldn't create new graph");
// AUNodes represent AudioUnits on the AUGraph and provide an
// easy means for connecting audioUnits together.
AUNode filePlayerNode;
AUNode filePlayerNode2;
AUNode mixerNode;
AUNode reverbNode;
AUNode toneNode;
AUNode gOutputNode;
// file player component
AudioComponentDescription filePlayer_desc;
filePlayer_desc.componentType = kAudioUnitType_Generator;
filePlayer_desc.componentSubType = kAudioUnitSubType_AudioFilePlayer;
filePlayer_desc.componentFlags = 0;
filePlayer_desc.componentFlagsMask = 0;
filePlayer_desc.componentManufacturer = kAudioUnitManufacturer_Apple;
// file player component2
AudioComponentDescription filePlayer2_desc;
filePlayer2_desc.componentType = kAudioUnitType_Generator;
filePlayer2_desc.componentSubType = kAudioUnitSubType_AudioFilePlayer;
filePlayer2_desc.componentFlags = 0;
filePlayer2_desc.componentFlagsMask = 0;
filePlayer2_desc.componentManufacturer = kAudioUnitManufacturer_Apple;
// Create AudioComponentDescriptions for the AUs we want in the graph
// mixer component
AudioComponentDescription mixer_desc;
mixer_desc.componentType = kAudioUnitType_Mixer;
mixer_desc.componentSubType = kAudioUnitSubType_MultiChannelMixer;
mixer_desc.componentFlags = 0;
mixer_desc.componentFlagsMask = 0;
mixer_desc.componentManufacturer = kAudioUnitManufacturer_Apple;
// Create AudioComponentDescriptions for the AUs we want in the graph
// Reverb component
AudioComponentDescription reverb_desc;
reverb_desc.componentType = kAudioUnitType_Effect;
reverb_desc.componentSubType = kAudioUnitSubType_Reverb2;
reverb_desc.componentFlags = 0;
reverb_desc.componentFlagsMask = 0;
reverb_desc.componentManufacturer = kAudioUnitManufacturer_Apple;
//tone component
AudioComponentDescription tone_desc;
tone_desc.componentType = kAudioUnitType_FormatConverter;
//tone_desc.componentSubType = kAudioUnitSubType_NewTimePitch;
tone_desc.componentSubType = kAudioUnitSubType_Varispeed;
tone_desc.componentFlags = 0;
tone_desc.componentFlagsMask = 0;
tone_desc.componentManufacturer = kAudioUnitManufacturer_Apple;
AudioComponentDescription gOutput_desc;
gOutput_desc.componentType = kAudioUnitType_Output;
gOutput_desc.componentSubType = kAudioUnitSubType_GenericOutput;
gOutput_desc.componentFlags = 0;
gOutput_desc.componentFlagsMask = 0;
gOutput_desc.componentManufacturer = kAudioUnitManufacturer_Apple;
//Add nodes to graph
// Add nodes to the graph to hold our AudioUnits,
// You pass in a reference to the AudioComponentDescription
// and get back an AudioUnit
AUGraphAddNode(mGraph, &filePlayer_desc, &filePlayerNode );
AUGraphAddNode(mGraph, &filePlayer2_desc, &filePlayerNode2 );
AUGraphAddNode(mGraph, &mixer_desc, &mixerNode );
AUGraphAddNode(mGraph, &reverb_desc, &reverbNode );
AUGraphAddNode(mGraph, &tone_desc, &toneNode );
AUGraphAddNode(mGraph, &gOutput_desc, &gOutputNode);
//Open the graph early, initialize late
// open the graph AudioUnits are open but not initialized (no resource allocation occurs here)
CheckError(AUGraphOpen(mGraph),"Couldn't Open the graph");
//Reference to Nodes
// get the reference to the AudioUnit object for the file player graph node
AUGraphNodeInfo(mGraph, filePlayerNode, NULL, &mFilePlayer);
AUGraphNodeInfo(mGraph, filePlayerNode2, NULL, &mFilePlayer2);
AUGraphNodeInfo(mGraph, reverbNode, NULL, &mReverb);
AUGraphNodeInfo(mGraph, toneNode, NULL, &mTone);
AUGraphNodeInfo(mGraph, mixerNode, NULL, &mMixer);
AUGraphNodeInfo(mGraph, gOutputNode, NULL, &mGIO);
AUGraphConnectNodeInput(mGraph, filePlayerNode, 0, reverbNode, 0);
AUGraphConnectNodeInput(mGraph, reverbNode, 0, toneNode, 0);
AUGraphConnectNodeInput(mGraph, toneNode, 0, mixerNode,0);
AUGraphConnectNodeInput(mGraph, filePlayerNode2, 0, mixerNode, 1);
AUGraphConnectNodeInput(mGraph, mixerNode, 0, gOutputNode, 0);
UInt32 busCount = 2; // bus count for mixer unit input
//Setup mixer unit bus count
CheckError(AudioUnitSetProperty (
mMixer,
kAudioUnitProperty_ElementCount,
kAudioUnitScope_Input,
0,
&busCount,
sizeof (busCount)
),
"Couldn't set mixer unit's bus count");
//Enable metering mode to view levels input and output levels of mixer
UInt32 onValue = 1;
CheckError(AudioUnitSetProperty(mMixer,
kAudioUnitProperty_MeteringMode,
kAudioUnitScope_Input,
0,
&onValue,
sizeof(onValue)),
"error");
// Increase the maximum frames per slice allows the mixer unit to accommodate the
// larger slice size used when the screen is locked.
UInt32 maximumFramesPerSlice = 4096;
CheckError(AudioUnitSetProperty (
mMixer,
kAudioUnitProperty_MaximumFramesPerSlice,
kAudioUnitScope_Global,
0,
&maximumFramesPerSlice,
sizeof (maximumFramesPerSlice)
),
"Couldn't set mixer units maximum framers per slice");
// set the audio data format of tone Unit
AudioUnitSetProperty(mTone,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Global,
0,
&stereoStreamFormat864,
sizeof(AudioStreamBasicDescription));
// set the audio data format of reverb Unit
AudioUnitSetProperty(mReverb,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Global,
0,
&stereoStreamFormat864,
sizeof(AudioStreamBasicDescription));
// set initial reverb
AudioUnitParameterValue reverbTime = 2.5;
AudioUnitSetParameter(mReverb, 4, kAudioUnitScope_Global, 0, reverbTime, 0);
AudioUnitSetParameter(mReverb, 5, kAudioUnitScope_Global, 0, reverbTime, 0);
AudioUnitSetParameter(mReverb, 0, kAudioUnitScope_Global, 0, 0, 0);
AudioStreamBasicDescription auEffectStreamFormat;
UInt32 asbdSize = sizeof (auEffectStreamFormat);
memset (&auEffectStreamFormat, 0, sizeof (auEffectStreamFormat ));
// get the audio data format from reverb
CheckError(AudioUnitGetProperty(mReverb,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Input,
0,
&auEffectStreamFormat,
&asbdSize),
"Couldn't get aueffectunit ASBD");
auEffectStreamFormat.mSampleRate = graphSampleRate;
// set the audio data format of mixer Unit
CheckError(AudioUnitSetProperty(mMixer,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Output,
0,
&auEffectStreamFormat, sizeof(auEffectStreamFormat)),
"Couldn't set ASBD on mixer output");
CheckError(AUGraphInitialize(mGraph),"Couldn't Initialize the graph");
[self setUpAUFilePlayer];
[self setUpAUFilePlayer2];
}
//Audio file playback setup here i am setting the voice file
-(OSStatus) setUpAUFilePlayer{
NSString *songPath = [[NSBundle mainBundle] pathForResource: @"testVoice" ofType:@".m4a"];
CFURLRef songURL = ( CFURLRef) [NSURL fileURLWithPath:songPath];
// open the input audio file
CheckError(AudioFileOpenURL(songURL, kAudioFileReadPermission, 0, &inputFile),
"setUpAUFilePlayer AudioFileOpenURL failed");
AudioStreamBasicDescription fileASBD;
// get the audio data format from the file
UInt32 propSize = sizeof(fileASBD);
CheckError(AudioFileGetProperty(inputFile, kAudioFilePropertyDataFormat,
&propSize, &fileASBD),
"setUpAUFilePlayer couldn't get file's data format");
// tell the file player unit to load the file we want to play
CheckError(AudioUnitSetProperty(mFilePlayer, kAudioUnitProperty_ScheduledFileIDs,
kAudioUnitScope_Global, 0, &inputFile, sizeof(inputFile)),
"setUpAUFilePlayer AudioUnitSetProperty[kAudioUnitProperty_ScheduledFileIDs] failed");
UInt64 nPackets;
UInt32 propsize = sizeof(nPackets);
CheckError(AudioFileGetProperty(inputFile, kAudioFilePropertyAudioDataPacketCount,
&propsize, &nPackets),
"setUpAUFilePlayer AudioFileGetProperty[kAudioFilePropertyAudioDataPacketCount] failed");
// tell the file player AU to play the entire file
ScheduledAudioFileRegion rgn;
memset (&rgn.mTimeStamp, 0, sizeof(rgn.mTimeStamp));
rgn.mTimeStamp.mFlags = kAudioTimeStampSampleTimeValid;
rgn.mTimeStamp.mSampleTime = 0;
rgn.mCompletionProc = NULL;
rgn.mCompletionProcUserData = NULL;
rgn.mAudioFile = inputFile;
rgn.mLoopCount = -1;
rgn.mStartFrame = 0;
rgn.mFramesToPlay = nPackets * fileASBD.mFramesPerPacket;
if (MaxSampleTime < rgn.mFramesToPlay)
{
MaxSampleTime = rgn.mFramesToPlay;
}
CheckError(AudioUnitSetProperty(mFilePlayer, kAudioUnitProperty_ScheduledFileRegion,
kAudioUnitScope_Global, 0,&rgn, sizeof(rgn)),
"setUpAUFilePlayer1 AudioUnitSetProperty[kAudioUnitProperty_ScheduledFileRegion] failed");
// prime the file player AU with default values
UInt32 defaultVal = 0;
CheckError(AudioUnitSetProperty(mFilePlayer, kAudioUnitProperty_ScheduledFilePrime,
kAudioUnitScope_Global, 0, &defaultVal, sizeof(defaultVal)),
"setUpAUFilePlayer AudioUnitSetProperty[kAudioUnitProperty_ScheduledFilePrime] failed");
// tell the file player AU when to start playing (-1 sample time means next render cycle)
AudioTimeStamp startTime;
memset (&startTime, 0, sizeof(startTime));
startTime.mFlags = kAudioTimeStampSampleTimeValid;
startTime.mSampleTime = -1;
CheckError(AudioUnitSetProperty(mFilePlayer, kAudioUnitProperty_ScheduleStartTimeStamp,
kAudioUnitScope_Global, 0, &startTime, sizeof(startTime)),
"setUpAUFilePlayer AudioUnitSetProperty[kAudioUnitProperty_ScheduleStartTimeStamp]");
return noErr;
}
//Audio file playback setup here i am setting the BGMusic file
-(OSStatus) setUpAUFilePlayer2{
NSString *songPath = [[NSBundle mainBundle] pathForResource: @"BGmusic" ofType:@".mp3"];
CFURLRef songURL = ( CFURLRef) [NSURL fileURLWithPath:songPath];
// open the input audio file
CheckError(AudioFileOpenURL(songURL, kAudioFileReadPermission, 0, &inputFile2),
"setUpAUFilePlayer2 AudioFileOpenURL failed");
AudioStreamBasicDescription fileASBD;
// get the audio data format from the file
UInt32 propSize = sizeof(fileASBD);
CheckError(AudioFileGetProperty(inputFile2, kAudioFilePropertyDataFormat,
&propSize, &fileASBD),
"setUpAUFilePlayer2 couldn't get file's data format");
// tell the file player unit to load the file we want to play
CheckError(AudioUnitSetProperty(mFilePlayer2, kAudioUnitProperty_ScheduledFileIDs,
kAudioUnitScope_Global, 0, &inputFile2, sizeof(inputFile2)),
"setUpAUFilePlayer2 AudioUnitSetProperty[kAudioUnitProperty_ScheduledFileIDs] failed");
UInt64 nPackets;
UInt32 propsize = sizeof(nPackets);
CheckError(AudioFileGetProperty(inputFile2, kAudioFilePropertyAudioDataPacketCount,
&propsize, &nPackets),
"setUpAUFilePlayer2 AudioFileGetProperty[kAudioFilePropertyAudioDataPacketCount] failed");
// tell the file player AU to play the entire file
ScheduledAudioFileRegion rgn;
memset (&rgn.mTimeStamp, 0, sizeof(rgn.mTimeStamp));
rgn.mTimeStamp.mFlags = kAudioTimeStampSampleTimeValid;
rgn.mTimeStamp.mSampleTime = 0;
rgn.mCompletionProc = NULL;
rgn.mCompletionProcUserData = NULL;
rgn.mAudioFile = inputFile2;
rgn.mLoopCount = -1;
rgn.mStartFrame = 0;
rgn.mFramesToPlay = nPackets * fileASBD.mFramesPerPacket;
if (MaxSampleTime < rgn.mFramesToPlay)
{
MaxSampleTime = rgn.mFramesToPlay;
}
CheckError(AudioUnitSetProperty(mFilePlayer2, kAudioUnitProperty_ScheduledFileRegion,
kAudioUnitScope_Global, 0,&rgn, sizeof(rgn)),
"setUpAUFilePlayer2 AudioUnitSetProperty[kAudioUnitProperty_ScheduledFileRegion] failed");
// prime the file player AU with default values
UInt32 defaultVal = 0;
CheckError(AudioUnitSetProperty(mFilePlayer2, kAudioUnitProperty_ScheduledFilePrime,
kAudioUnitScope_Global, 0, &defaultVal, sizeof(defaultVal)),
"setUpAUFilePlayer2 AudioUnitSetProperty[kAudioUnitProperty_ScheduledFilePrime] failed");
// tell the file player AU when to start playing (-1 sample time means next render cycle)
AudioTimeStamp startTime;
memset (&startTime, 0, sizeof(startTime));
startTime.mFlags = kAudioTimeStampSampleTimeValid;
startTime.mSampleTime = -1;
CheckError(AudioUnitSetProperty(mFilePlayer2, kAudioUnitProperty_ScheduleStartTimeStamp,
kAudioUnitScope_Global, 0, &startTime, sizeof(startTime)),
"setUpAUFilePlayer2 AudioUnitSetProperty[kAudioUnitProperty_ScheduleStartTimeStamp]");
return noErr;
}
//Start Saving File
- (void)startRecordingAAC{
AudioStreamBasicDescription destinationFormat;
memset(&destinationFormat, 0, sizeof(destinationFormat));
destinationFormat.mChannelsPerFrame = 2;
destinationFormat.mFormatID = kAudioFormatMPEG4AAC;
UInt32 size = sizeof(destinationFormat);
OSStatus result = AudioFormatGetProperty(kAudioFormatProperty_FormatInfo, 0, NULL, &size, &destinationFormat);
if(result) printf("AudioFormatGetProperty %ld \n", result);
NSArray *paths = NSSearchPathForDirectoriesInDomains(NSDocumentDirectory, NSUserDomainMask, YES);
NSString *documentsDirectory = [paths objectAtIndex:0];
NSString *destinationFilePath = [[NSString alloc] initWithFormat: @"%@/output.m4a", documentsDirectory];
CFURLRef destinationURL = CFURLCreateWithFileSystemPath(kCFAllocatorDefault,
(CFStringRef)destinationFilePath,
kCFURLPOSIXPathStyle,
false);
[destinationFilePath release];
// specify codec Saving the output in .m4a format
result = ExtAudioFileCreateWithURL(destinationURL,
kAudioFileM4AType,
&destinationFormat,
NULL,
kAudioFileFlags_EraseFile,
&extAudioFile);
if(result) printf("ExtAudioFileCreateWithURL %ld \n", result);
CFRelease(destinationURL);
// This is a very important part and easiest way to set the ASBD for the File with correct format.
AudioStreamBasicDescription clientFormat;
UInt32 fSize = sizeof (clientFormat);
memset(&clientFormat, 0, sizeof(clientFormat));
// get the audio data format from the Output Unit
CheckError(AudioUnitGetProperty(mGIO,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Output,
0,
&clientFormat,
&fSize),"AudioUnitGetProperty on failed");
// set the audio data format of mixer Unit
CheckError(ExtAudioFileSetProperty(extAudioFile,
kExtAudioFileProperty_ClientDataFormat,
sizeof(clientFormat),
&clientFormat),
"ExtAudioFileSetProperty kExtAudioFileProperty_ClientDataFormat failed");
// specify codec
UInt32 codec = kAppleHardwareAudioCodecManufacturer;
CheckError(ExtAudioFileSetProperty(extAudioFile,
kExtAudioFileProperty_CodecManufacturer,
sizeof(codec),
&codec),"ExtAudioFileSetProperty on extAudioFile Faild");
CheckError(ExtAudioFileWriteAsync(extAudioFile, 0, NULL),"ExtAudioFileWriteAsync Failed");
[self pullGenericOutput];
}
// Manual Feeding and getting data/Buffer from the GenericOutput Node.
-(void)pullGenericOutput{
AudioUnitRenderActionFlags flags = 0;
AudioTimeStamp inTimeStamp;
memset(&inTimeStamp, 0, sizeof(AudioTimeStamp));
inTimeStamp.mFlags = kAudioTimeStampSampleTimeValid;
UInt32 busNumber = 0;
UInt32 numberFrames = 512;
inTimeStamp.mSampleTime = 0;
int channelCount = 2;
NSLog(@"Final numberFrames :%li",numberFrames);
int totFrms = MaxSampleTime;
while (totFrms > 0)
{
if (totFrms < numberFrames)
{
numberFrames = totFrms;
NSLog(@"Final numberFrames :%li",numberFrames);
}
else
{
totFrms -= numberFrames;
}
AudioBufferList *bufferList = (AudioBufferList*)malloc(sizeof(AudioBufferList)+sizeof(AudioBuffer)*(channelCount-1));
bufferList->mNumberBuffers = channelCount;
for (int j=0; j<channelCount; j++)
{
AudioBuffer buffer = {0};
buffer.mNumberChannels = 1;
buffer.mDataByteSize = numberFrames*sizeof(AudioUnitSampleType);
buffer.mData = calloc(numberFrames, sizeof(AudioUnitSampleType));
bufferList->mBuffers[j] = buffer;
}
CheckError(AudioUnitRender(mGIO,
&flags,
&inTimeStamp,
busNumber,
numberFrames,
bufferList),
"AudioUnitRender mGIO");
CheckError(ExtAudioFileWrite(extAudioFile, numberFrames, bufferList),("extaudiofilewrite fail"));
}
[self FilesSavingCompleted];
}
//FilesSavingCompleted
-(void)FilesSavingCompleted{
OSStatus status = ExtAudioFileDispose(extAudioFile);
printf("OSStatus(ExtAudioFileDispose): %ld\n", status);
}
offline rendering with a lowpass filter causes aliasing and clipping
I am using converters at both ends with ASBD set to 8000 samplerate mono floats for input of input converter and output of output converter while using 44100.0 stereo for input and output of the low pass unit, and calling AudioUnitRender on the end converter with no io unit for the offline render. For the online render I put a converter unit before the io unit so the render callback will pull from buffers at 8K for playback too. It appears that the lower sample rate on the output ASBD requires a higher maximum frames per slice and a smaller slice (AudioUnitRender inNumberFrames) and that's why it wouldn't render.
#import "ViewController.h"
#import <AudioToolbox/AudioToolbox.h>
@implementation ViewController{
int sampleCount;
int renderBufferHead;
float *renderBuffer;
}
- (void)viewDidLoad {
[super viewDidLoad];
float sampleRate = 8000;
int bufferSeconds = 3;
sampleCount = sampleRate * bufferSeconds;//seconds
float *originalSaw = generateSawWaveBuffer(440, sampleRate, sampleCount);
renderBuffer = originalSaw;
renderBufferHead = 0;
AURenderCallbackStruct cbStruct = {renderCallback,(__bridge void *)self};
//this will do offline render using the render callback, callback just reads from renderBuffer at samplerate
float *processedBuffer = offlineRender(sampleCount, sampleRate, &cbStruct);
renderBufferHead = 0;//rewind render buffer after processing
//set up audio units to do live render using the render callback at sample rate then self destruct after delay
//it will play originalSaw for bufferSeconds, then after delay will switch renderBuffer to point at processedBuffer
float secondsToPlayAudio = (bufferSeconds + 1) * 2;
onlineRender(sampleRate, &cbStruct,secondsToPlayAudio);
//wait for original to finish playing, then change render callback source buffer to processed buffer
dispatch_after(dispatch_time(DISPATCH_TIME_NOW, (int64_t)((secondsToPlayAudio / 2) * NSEC_PER_SEC)), dispatch_get_main_queue(), ^{
renderBuffer = processedBuffer;
renderBufferHead = 0;//rewind render buffer
});
//destroy after all rendering done
dispatch_after(dispatch_time(DISPATCH_TIME_NOW, (int64_t)(secondsToPlayAudio * NSEC_PER_SEC)), dispatch_get_main_queue(), ^{
free(originalSaw);
free(processedBuffer);
});
}
float * offlineRender(int count, double sampleRate, AURenderCallbackStruct *cbStruct){
AudioComponentInstance inConverter = getComponentInstance(kAudioUnitType_FormatConverter, kAudioUnitSubType_AUConverter);
AudioComponentInstance lowPass = getComponentInstance(kAudioUnitType_Effect, kAudioUnitSubType_LowPassFilter);
AudioComponentInstance outConverter = getComponentInstance(kAudioUnitType_FormatConverter, kAudioUnitSubType_AUConverter);
AudioStreamBasicDescription asbd = getMonoFloatASBD(sampleRate);
AudioUnitSetProperty(inConverter, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &asbd, sizeof(AudioStreamBasicDescription));
AudioUnitSetProperty(outConverter, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 0, &asbd, sizeof(AudioStreamBasicDescription));
AudioUnitSetProperty(inConverter, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, 0, cbStruct, sizeof(AURenderCallbackStruct));
formatAndConnect(inConverter, lowPass);
formatAndConnect(lowPass, outConverter);
UInt32 maxFramesPerSlice = 4096;
AudioUnitSetProperty(inConverter, kAudioUnitProperty_MaximumFramesPerSlice, kAudioUnitScope_Global, 0, &maxFramesPerSlice, sizeof(UInt32));
AudioUnitSetProperty(lowPass, kAudioUnitProperty_MaximumFramesPerSlice, kAudioUnitScope_Global, 0, &maxFramesPerSlice, sizeof(UInt32));
AudioUnitSetProperty(outConverter, kAudioUnitProperty_MaximumFramesPerSlice, kAudioUnitScope_Global, 0, &maxFramesPerSlice, sizeof(UInt32));
AudioUnitInitialize(inConverter);
AudioUnitInitialize(lowPass);
AudioUnitInitialize(outConverter);
AudioUnitSetParameter(lowPass, kLowPassParam_CutoffFrequency, kAudioUnitScope_Global, 0, 500, 0);
AudioBufferList *bufferlist = malloc(sizeof(AudioBufferList) + sizeof(AudioBufferList));//stereo bufferlist + sizeof(AudioBuffer)
float *left = malloc(sizeof(float) * 4096);
bufferlist->mBuffers[0].mData = left;
bufferlist->mNumberBuffers = 1;
AudioTimeStamp inTimeStamp;
memset(&inTimeStamp, 0, sizeof(AudioTimeStamp));
inTimeStamp.mFlags = kAudioTimeStampSampleTimeValid;
inTimeStamp.mSampleTime = 0;
float *buffer = malloc(sizeof(float) * count);
int inNumberframes = 512;
AudioUnitRenderActionFlags flag = 0;
int framesRead = 0;
while (count){
inNumberframes = MIN(inNumberframes, count);
bufferlist->mBuffers[0].mDataByteSize = sizeof(float) * inNumberframes;
printf("Offline Render %i frames\n",inNumberframes);
AudioUnitRender(outConverter, &flag, &inTimeStamp, 0, inNumberframes, bufferlist);
memcpy(buffer + framesRead, left, sizeof(float) * inNumberframes);
inTimeStamp.mSampleTime += inNumberframes;
count -= inNumberframes;
framesRead += inNumberframes;
}
free(left);
// free(right);
free(bufferlist);
AudioUnitUninitialize(inConverter);
AudioUnitUninitialize(lowPass);
AudioUnitUninitialize(outConverter);
return buffer;
}
OSStatus renderCallback(void * inRefCon,
AudioUnitRenderActionFlags * ioActionFlags,
const AudioTimeStamp * inTimeStamp,
UInt32 inBusNumber,
UInt32 inNumberFrames,
AudioBufferList * ioData){
ViewController *self = (__bridge ViewController*)inRefCon;
float *left = ioData->mBuffers[0].mData;
for (int i = 0; i < inNumberFrames; i++) {
if (self->renderBufferHead >= self->sampleCount) {
left[i] = 0;
}
else{
left[i] = self->renderBuffer[self->renderBufferHead++];
}
}
if(ioData->mNumberBuffers == 2){
memcpy(ioData->mBuffers[1].mData, left, sizeof(float) * inNumberFrames);
}
printf("render %f to %f\n",inTimeStamp->mSampleTime,inTimeStamp->mSampleTime + inNumberFrames);
return noErr;
}
void onlineRender(double sampleRate, AURenderCallbackStruct *cbStruct,float duration){
AudioComponentInstance converter = getComponentInstance(kAudioUnitType_FormatConverter, kAudioUnitSubType_AUConverter);
AudioComponentInstance ioUnit = getComponentInstance(kAudioUnitType_Output, kAudioUnitSubType_DefaultOutput);
AudioStreamBasicDescription asbd = getMonoFloatASBD(sampleRate);
AudioUnitSetProperty(converter, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &asbd, sizeof(AudioStreamBasicDescription));
AudioUnitSetProperty(converter, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, 0, cbStruct, sizeof(AURenderCallbackStruct));
formatAndConnect(converter, ioUnit);
AudioUnitInitialize(converter);
AudioUnitInitialize(ioUnit);
AudioOutputUnitStart(ioUnit);
dispatch_after(dispatch_time(DISPATCH_TIME_NOW, (int64_t)(duration * NSEC_PER_SEC)), dispatch_get_main_queue(), ^{
AudioOutputUnitStop(ioUnit);
AudioUnitUninitialize(ioUnit);
AudioUnitUninitialize(converter);
});
}
float * generateSawWaveBuffer(float frequency,float sampleRate, int sampleCount){
float *buffer = malloc(sizeof(float) * sampleCount);
float increment = (frequency / sampleRate) * 2;
int increasing = 1;
float sample = 0;
for (int i = 0; i < sampleCount; i++) {
if (increasing) {
sample += increment;
if (sample >= 1) {
increasing = 0;
}
}
else{
sample -= increment;
if (sample < -1) {
increasing = 1;
}
}
buffer[i] = sample;
}
return buffer;
}
AudioComponentInstance getComponentInstance(OSType type,OSType subType){
AudioComponentDescription desc = {0};
desc.componentFlags = 0;
desc.componentFlagsMask = 0;
desc.componentManufacturer = kAudioUnitManufacturer_Apple;
desc.componentSubType = subType;
desc.componentType = type;
AudioComponent ioComponent = AudioComponentFindNext(NULL, &desc);
AudioComponentInstance unit;
AudioComponentInstanceNew(ioComponent, &unit);
return unit;
}
AudioStreamBasicDescription getMonoFloatASBD(double sampleRate){
AudioStreamBasicDescription asbd = {0};
asbd.mSampleRate = sampleRate;
asbd.mFormatID = kAudioFormatLinearPCM;
asbd.mFormatFlags = kAudioFormatFlagIsFloat | kAudioFormatFlagIsNonInterleaved | kAudioFormatFlagIsPacked;
asbd.mFramesPerPacket = 1;
asbd.mChannelsPerFrame = 1;
asbd.mBitsPerChannel = 32;
asbd.mBytesPerPacket = 4;
asbd.mBytesPerFrame = 4;
return asbd;
}
void formatAndConnect(AudioComponentInstance src,AudioComponentInstance dst);Related Topics
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