Hi I am trying to convert AAC buffer to PCM using AudioConverterFillComplexBuffer..Here is my code
-(void)initDecoder{
AudioStreamBasicDescription outAudioStreamBasicDescription;
outAudioStreamBasicDescription.mSampleRate = 44100.0;
outAudioStreamBasicDescription.mFormatID = kAudioFormatLinearPCM;
outAudioStreamBasicDescription.mFormatFlags = 0xc;
outAudioStreamBasicDescription.mBytesPerPacket = 2;
outAudioStreamBasicDescription.mFramesPerPacket = 1;
outAudioStreamBasicDescription.mBytesPerFrame = 2;
outAudioStreamBasicDescription.mChannelsPerFrame = 1;
outAudioStreamBasicDescription.mBitsPerChannel = 16;
AudioStreamBasicDescription inAudioStreamBasicDescription;
inAudioStreamBasicDescription.mSampleRate = 44100;
inAudioStreamBasicDescription.mFormatID = kAudioFormatMPEG4AAC;
inAudioStreamBasicDescription.mFormatFlags = kMPEG4Object_AAC_SSR;
inAudioStreamBasicDescription.mBytesPerPacket = 0;
inAudioStreamBasicDescription.mFramesPerPacket = 1024;
inAudioStreamBasicDescription.mBytesPerFrame = 0;
inAudioStreamBasicDescription.mChannelsPerFrame = 1;
inAudioStreamBasicDescription.mBitsPerChannel = 0;
inAudioStreamBasicDescription.mReserved = 0;
AudioClassDescription audioClassDescription;
memset(&audioClassDescription, 0, sizeof(audioClassDescription));
audioClassDescription.mManufacturer = kAppleSoftwareAudioCodecManufacturer;
audioClassDescription.mSubType = outAudioStreamBasicDescription.mFormatID;
audioClassDescription.mType = kAudioFormatLinearPCM;
NSAssert(audioClassDescription.mSubType == outAudioStreamBasicDescription.mFormatID && audioClassDescription.mManufacturer == kAppleSoftwareAudioCodecManufacturer, nil);
NSAssert(AudioConverterNewSpecific(&inAudioStreamBasicDescription, &outAudioStreamBasicDescription, 1, &audioClassDescription, &audioConverterDecode) == 0, nil);
}
OSStatus inInputDataProc(AudioConverterRef inAudioConverter, UInt32 *ioNumberDataPackets, AudioBufferList *ioData, AudioStreamPacketDescription **outDataPacketDescription, void *inUserData)
{
AudioBufferList audioBufferList = *(AudioBufferList *)inUserData;
ioData->mBuffers[0].mData = audioBufferList.mBuffers[0].mData;
ioData->mBuffers[0].mDataByteSize = audioBufferList.mBuffers[0].mDataByteSize;
return noErr;
}
-(void)decodeSample:(AudioBufferList)inAaudioBufferList{
//inAaudioBufferList is the AAC buffer
if (!audioConverterDecode) {
[self initDecoder];
}
NSAssert(inAaudioBufferList.mNumberBuffers == 1, nil);
uint32_t bufferSize = 1024*2;//inAaudioBufferList.mBuffers[0].mDataByteSize;
uint8_t *buffer = (uint8_t *)malloc(1024*2);
memset(buffer, 0, bufferSize);
AudioBufferList outAudioBufferList;
outAudioBufferList.mNumberBuffers = 1;
outAudioBufferList.mBuffers[0].mNumberChannels = 1;
outAudioBufferList.mBuffers[0].mDataByteSize = bufferSize;
outAudioBufferList.mBuffers[0].mData = buffer;
UInt32 ioOutputDataPacketSize = bufferSize;
OSStatus ret = AudioConverterFillComplexBuffer(audioConverterDecode, inInputDataProc, &inAaudioBufferList, &ioOutputDataPacketSize, &outAudioBufferList, NULL) ;//== 0, nil);
NSData *data = [NSData dataWithBytes:outAudioBufferList.mBuffers[0].mData length:outAudioBufferList.mBuffers[0].mDataByteSize];
DLog(#"Rev Size = %d",(unsigned int)outAudioBufferList.mBuffers[0].mDataByteSize);
free(buffer);
}
The decoded output length is zero and the OSStatus code for AudioConverterFillComplexBuffer is 561015652
So what could be wrong..?
This is a shot in the dark and you probably already found the solution to this but I believe you need to change this
UInt32 ioOutputDataPacketSize = bufferSize;
to this
UInt32 ioOutputDataPacketSize = bufferSize/2;
Personally for me I do this in the inInputDataProc method and always sent
UInt32 ioOutputDataPacketSize = 1;
into the AudioConverterFillComplexBuffer method and then set it within the inInputDataProc like this.
UInt32 maxPackets = audioBufferList.mBuffers[0].mDataByteSize / 2;
*ioNumberDataPackets = maxPackets;
Hope this helps.
Related
I recevie audio data and size from outside, the audio appears to be linear PCM, signed int16, but when recording this using an AssetWriter it saves to the audio file highly distorted and higher pitch.
#define kSamplingRate 16000
#define kNumberChannels 1
UInt32 framesAlreadyWritten = 0;
-(AudioStreamBasicDescription) getAudioFormat {
AudioStreamBasicDescription format;
format.mSampleRate = kSamplingRate;
format.mFormatID = kAudioFormatLinearPCM;
format.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked;
format.mChannelsPerFrame = 1; // mono
format.mBitsPerChannel = 16;
format.mBytesPerFrame = sizeof(SInt16);
format.mFramesPerPacket = 1;
format.mBytesPerPacket = format.mBytesPerFrame * format.mFramesPerPacket;
format.mReserved = 0;
return format;
}
- (CMSampleBufferRef)createAudioSample:(const void *)audioData frames: (UInt32)len {
AudioStreamBasicDescription asbd = [self getAudioFormat];
CMSampleBufferRef buff = NULL;
static CMFormatDescriptionRef format = NULL;
OSStatus error = 0;
if(format == NULL) {
AudioChannelLayout acl;
bzero(&acl, sizeof(acl));
acl.mChannelLayoutTag = kAudioChannelLayoutTag_Mono;
error = CMAudioFormatDescriptionCreate(kCFAllocatorDefault, &asbd, sizeof(acl), &acl, 0, NULL, NULL, &format);
}
CMTime duration = CMTimeMake(1, kSamplingRate);
CMTime pts = CMTimeMake(framesAlreadyWritten, kSamplingRate);
NSLog(#"-----------pts");
CMTimeShow(pts);
CMSampleTimingInfo timing = {duration , pts, kCMTimeInvalid };
error = CMSampleBufferCreate(kCFAllocatorDefault, NULL, false, NULL, NULL, format, len, 1, &timing, 0, NULL, &buff);
framesAlreadyWritten += len;
if (error) {
NSLog(#"CMSampleBufferCreate returned error: %ld", (long)error);
return NULL;
}
AudioBufferList audioBufferList;
audioBufferList.mNumberBuffers = 1;
audioBufferList.mBuffers[0].mNumberChannels = asbd.mChannelsPerFrame;
audioBufferList.mBuffers[0].mDataByteSize = (UInt32)(number_of_frames * audioFormat.mBytesPerFrame);
audioBufferList.mBuffers[0].mData = audioData;
error = CMSampleBufferSetDataBufferFromAudioBufferList(buff, kCFAllocatorDefault, kCFAllocatorDefault, 0, &audioBufferList);
if(error) {
NSLog(#"CMSampleBufferSetDataBufferFromAudioBufferList returned error: %ld", (long)error);
return NULL;
}
return buff;
}
Not sure why you're dividing len by two, but your time should progress instead of being constant, something like
CMTime time = CMTimeMake(framesAlreadyWritten , kSamplingRate);
I want to use AudioConverterFillComplexBuffer to convert sample rate for a pcm buffer（32k to 44.1k）。But i didn't know why the voice seems changed（too many noise）。Here is the main code：
struct AudioFrame {
int samples; //number of samples in this frame. e.g. 320
int bytesPerSample; //number of bytes per sample: 2 for PCM16.
int channels; //number of channels (data are interleaved if stereo)
int samplesPerSec; //sampling rate
void* buffer; //data buffer
};
-(void)convertAudioFrame:(AudioFrame *)buffer outPutData:(unsigned char **)outPutData outPutDataSize:(UInt32 *)outPutDataSize{
if (buffer->bytesPerSample != self.unitDescription.mBitsPerChannel ||
buffer->channels != self.unitDescription.mChannelsPerFrame ||
buffer->samplesPerSec != self.unitDescription.mSampleRate){
// describe the input format's description
AudioStreamBasicDescription inputDescription = {0};
inputDescription.mFormatID = kAudioFormatLinearPCM;
inputDescription.mFormatFlags = kLinearPCMFormatFlagIsPacked | kLinearPCMFormatFlagIsSignedInteger;
inputDescription.mChannelsPerFrame = buffer->channels;
inputDescription.mSampleRate = buffer->samplesPerSec;
inputDescription.mBitsPerChannel = 16;
inputDescription.mBytesPerFrame = (inputDescription.mBitsPerChannel/8) * inputDescription.mChannelsPerFrame;
inputDescription.mFramesPerPacket = 1;
inputDescription.mBytesPerPacket = inputDescription.mBytesPerFrame;
AudioStreamBasicDescription outputDescription = {0};
outputDescription.mSampleRate = 44100;
outputDescription.mFormatID = kAudioFormatLinearPCM;
outputDescription.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked;
outputDescription.mChannelsPerFrame = 1;
outputDescription.mFramesPerPacket = 1;
outputDescription.mBitsPerChannel = 16;
outputDescription.mBytesPerFrame = (outputDescription.mBitsPerChannel/8) * outputDescription.mChannelsPerFrame;
outputDescription.mBytesPerPacket = outputDescription.mBytesPerFrame;
// create an audio converter
AudioConverterRef audioConverter;
OSStatus status = AudioConverterNew(&inputDescription, &outputDescription, &audioConverter);
[self checkError:status errorMsg:#"AudioConverterNew error"];
if(!audioConverter)
{
*outPutDataSize = 0;
return;
}
UInt32 outputBytes = outputDescription.mBytesPerPacket * (buffer->samples*buffer->bytesPerSample / inputDescription.mBytesPerPacket);
unsigned char *outputBuffer = (unsigned char*)malloc(outputBytes);
memset(outputBuffer, 0, outputBytes);
AudioBuffer inputBuffer;
inputBuffer.mNumberChannels = inputDescription.mChannelsPerFrame;
inputBuffer.mDataByteSize = buffer->samples*buffer->bytesPerSample;
inputBuffer.mData = buffer->buffer;
AudioBufferList outputBufferList;
outputBufferList.mNumberBuffers = 1;
outputBufferList.mBuffers[0].mNumberChannels = outputDescription.mChannelsPerFrame;
outputBufferList.mBuffers[0].mDataByteSize = outputBytes;
outputBufferList.mBuffers[0].mData = outputBuffer;
UInt32 outputDataPacketSize = outputBytes / outputDescription.mBytesPerPacket;
self.currentBuffer = &inputBuffer;
self.currentInputDescription = inputDescription;
// convert
OSStatus result = AudioConverterFillComplexBuffer(audioConverter,
converterComplexInputDataProc,
(__bridge void*)self,
&outputDataPacketSize,
&outputBufferList,
NULL);
[self checkError:result errorMsg:#"AudioConverterConvertBuffer error"];
*outPutData = outputBuffer;
*outPutDataSize = outputBytes;
AudioConverterDispose(audioConverter);
}
}
//convert callback
OSStatus converterComplexInputDataProc(AudioConverterRef inAudioConverter,
UInt32* ioNumberDataPackets, AudioBufferList* ioData, AudioStreamPacketDescription** ioDataPacketDescription, void* inUserData){
XMMicAudioManager *self = (__bridge XMMicAudioManager *)inUserData;
ioData->mNumberBuffers = 1;
ioData->mBuffers[0] = *(self.currentBuffer);
*ioNumberDataPackets = ioData->mBuffers[0].mDataByteSize / self.currentInputDescription.mBytesPerPacket;
return 0;
}
I'm trying to decode aac data with AudioToolbox in iOS environment. I consulted this thread.
'AudioConverterNew' function call succeed but AudioConverterFillComplexBuffer returns error code 1852797029, kAudioCodecIllegalOperationError.
I'm trying to find my mistakes. Thank you for reading.
- (void)initAudioToolBox {
HCAudioAsset* asset = [self.provider getAudioAsset];
AudioStreamBasicDescription outFormat;
memset(&outFormat, 0, sizeof(outFormat));
outFormat.mSampleRate = 44100;
outFormat.mFormatID = kAudioFormatLinearPCM;
outFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger;
outFormat.mBytesPerPacket = 2;
outFormat.mFramesPerPacket = 1;
outFormat.mBytesPerFrame = 2;
outFormat.mChannelsPerFrame = 1;
outFormat.mBitsPerChannel = 16;
outFormat.mReserved = 0;
AudioStreamBasicDescription inFormat;
memset(&inFormat, 0, sizeof(inFormat));
inFormat.mSampleRate = [asset sampleRate];
inFormat.mFormatID = kAudioFormatMPEG4AAC;
inFormat.mFormatFlags = kMPEG4Object_AAC_LC;
inFormat.mBytesPerPacket = 0;
inFormat.mFramesPerPacket = (UInt32)[asset framePerPacket];
inFormat.mBytesPerFrame = 0;
inFormat.mChannelsPerFrame = (UInt32)[asset channelCount];
inFormat.mBitsPerChannel = 0;
inFormat.mReserved = 0;
OSStatus status = AudioConverterNew(&inFormat, &outFormat, &audioConverter);
if (status != noErr) {
NSLog(#"setup converter error, status: %i\n", (int)status);
} else {
NSLog(#"Audio Converter is initialized successfully.");
}
}
typedef struct _PassthroughUserData PassthroughUserData;
struct _PassthroughUserData {
UInt32 mChannels;
UInt32 mDataSize;
const void* mData;
AudioStreamPacketDescription mPacket;
};
int inInputDataProc(AudioConverterRef aAudioConverter,
UInt32* aNumDataPackets,
AudioBufferList* aData,
AudioStreamPacketDescription** aPacketDesc,
void* aUserData)
{
PassthroughUserData* userData = (PassthroughUserData*)aUserData;
if (!userData->mDataSize) {
*aNumDataPackets = 0;
NSLog(#"inInputDataProc returns -1");
return -1;
}
if (aPacketDesc) {
userData->mPacket.mStartOffset = 0;
userData->mPacket.mVariableFramesInPacket = 0;
userData->mPacket.mDataByteSize = userData->mDataSize;
NSLog(#"mDataSize:%d", userData->mDataSize);
*aPacketDesc = &userData->mPacket;
}
aData->mBuffers[0].mNumberChannels = userData->mChannels;
aData->mBuffers[0].mDataByteSize = userData->mDataSize;
aData->mBuffers[0].mData = (void*)(userData->mData);
NSLog(#"buffer[0] - channel:%d, byte size:%u, data:%p",
aData->mBuffers[0].mNumberChannels,
(unsigned int)aData->mBuffers[0].mDataByteSize,
aData->mBuffers[0].mData);
// No more data to provide following this run.
userData->mDataSize = 0;
NSLog(#"inInputDataProc returns 0");
return 0;
}
- (void)decodeAudioFrame:(NSData *)frame withPts:(NSInteger)pts{
if(!audioConverter){
[self initAudioToolBox];
}
HCAudioAsset* asset = [self.provider getAudioAsset];
PassthroughUserData userData = { (UInt32)[asset channelCount], (UInt32)frame.length, [frame bytes]};
NSMutableData *decodedData = [NSMutableData new];
const uint32_t MAX_AUDIO_FRAMES = 128;
const uint32_t maxDecodedSamples = MAX_AUDIO_FRAMES * 1;
do {
uint8_t *buffer = (uint8_t *)malloc(maxDecodedSamples * sizeof(short int));
AudioBufferList decBuffer;
memset(&decBuffer, 0, sizeof(AudioBufferList));
decBuffer.mNumberBuffers = 1;
decBuffer.mBuffers[0].mNumberChannels = 2;
decBuffer.mBuffers[0].mDataByteSize = maxDecodedSamples * sizeof(short int);
decBuffer.mBuffers[0].mData = buffer;
UInt32 numFrames = MAX_AUDIO_FRAMES;
AudioStreamPacketDescription outPacketDescription;
memset(&outPacketDescription, 0, sizeof(AudioStreamPacketDescription));
outPacketDescription.mDataByteSize = MAX_AUDIO_FRAMES;
outPacketDescription.mStartOffset = 0;
outPacketDescription.mVariableFramesInPacket = 0;
NSLog(#"frame - size:%lu, buffer:%p", [frame length], [frame bytes]);
OSStatus rv = AudioConverterFillComplexBuffer(audioConverter,
inInputDataProc,
&userData,
&numFrames,
&decBuffer,
&outPacketDescription);
NSLog(#"num frames:%d, dec buffer [0] channels:%d, dec buffer [0] data byte size:%d, rv:%d",
numFrames, decBuffer.mBuffers[0].mNumberChannels,
decBuffer.mBuffers[0].mDataByteSize, (int)rv);
if (rv && rv != noErr) {
NSLog(#"Error decoding audio stream: %d\n", rv);
break;
}
if (numFrames) {
[decodedData appendBytes:decBuffer.mBuffers[0].mData length:decBuffer.mBuffers[0].mDataByteSize];
}
} while (true);
//void *pData = (void *)[decodedData bytes];
//audioRenderer->Render(&pData, decodedData.length, pts);
}
I have successfully generated a tone using iOS with the following code. After that, I want to save the generated tone to an audio file. How can I do this?
- (void)createToneUnit
{
// Configure the search parameters to find the default playback output unit
// (called the kAudioUnitSubType_RemoteIO on iOS but
// kAudioUnitSubType_DefaultOutput on Mac OS X)
AudioComponentDescription defaultOutputDescription;
defaultOutputDescription.componentType = kAudioUnitType_Output;
defaultOutputDescription.componentSubType = kAudioUnitSubType_RemoteIO;
defaultOutputDescription.componentManufacturer = kAudioUnitManufacturer_Apple;
defaultOutputDescription.componentFlags = 0;
defaultOutputDescription.componentFlagsMask = 0;
// Get the default playback output unit
AudioComponent defaultOutput = AudioComponentFindNext(NULL, &defaultOutputDescription);
NSAssert(defaultOutput, #"Can't find default output");
// Create a new unit based on this that we'll use for output
OSErr err = AudioComponentInstanceNew(defaultOutput, &toneUnit);
NSAssert1(toneUnit, #"Error creating unit: %ld", err);
// Set our tone rendering function on the unit
AURenderCallbackStruct input;
input.inputProc = RenderTone;
input.inputProcRefCon = (__bridge void *)(self);
err = AudioUnitSetProperty(toneUnit,
kAudioUnitProperty_SetRenderCallback,
kAudioUnitScope_Input,
0,
&input,
sizeof(input));
NSAssert1(err == noErr, #"Error setting callback: %ld", err);
// Set the format to 32 bit, single channel, floating point, linear PCM
const int four_bytes_per_float = 4;
const int eight_bits_per_byte = 8;
AudioStreamBasicDescription streamFormat;
streamFormat.mSampleRate = sampleRate;
streamFormat.mFormatID = kAudioFormatLinearPCM;
streamFormat.mFormatFlags =
kAudioFormatFlagsNativeFloatPacked | kAudioFormatFlagIsNonInterleaved;
streamFormat.mBytesPerPacket = four_bytes_per_float;
streamFormat.mFramesPerPacket = 1;
streamFormat.mBytesPerFrame = four_bytes_per_float;
streamFormat.mChannelsPerFrame = 1;
streamFormat.mBitsPerChannel = four_bytes_per_float * eight_bits_per_byte;
err = AudioUnitSetProperty (toneUnit,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Input,
0,
&streamFormat,
sizeof(AudioStreamBasicDescription));
NSAssert1(err == noErr, #"Error setting stream format: %ld", err);
}
The Render Code :
OSStatus RenderTone(
void *inRefCon,
AudioUnitRenderActionFlags *ioActionFlags,
const AudioTimeStamp *inTimeStamp,
UInt32 inBusNumber,
UInt32 inNumberFrames,
AudioBufferList *ioData)
{
// Fixed amplitude is good enough for our purposes
const double amplitude = 0.25;
// Get the tone parameters out of the view controller
ViewController *viewController =
(__bridge ViewController *)inRefCon;
double theta = viewController->theta;
double theta_increment = 2.0 * M_PI * viewController->frequency / viewController->sampleRate;
// This is a mono tone generator so we only need the first buffer
const int channel = 0;
Float32 *buffer = (Float32 *)ioData->mBuffers[channel].mData;
// Generate the samples
for (UInt32 frame = 0; frame < inNumberFrames; frame++)
{
buffer[frame] = sin(theta) * amplitude;
theta += theta_increment;
if (theta > 2.0 * M_PI)
{
theta -= 2.0 * M_PI;
}
}
// Store the theta back in the view controller
viewController->theta = theta;
return noErr;
}
And to play the generated tone, I just :
OSErr err = AudioUnitInitialize(toneUnit);
err = AudioOutputUnitStart(toneUnit);
The extended audio file api provides an easy way to write audio files to disk.
I can't seem to find what I'm looking for in the documentation. This code works great, but I want stereo output.
- (void)createToneUnit
{
// Configure the search parameters to find the default playback output unit
// (called the kAudioUnitSubType_RemoteIO on iOS but
// kAudioUnitSubType_DefaultOutput on Mac OS X)
AudioComponentDescription defaultOutputDescription;
defaultOutputDescription.componentType = kAudioUnitType_Output;
defaultOutputDescription.componentSubType = kAudioUnitSubType_RemoteIO;
defaultOutputDescription.componentManufacturer = kAudioUnitManufacturer_Apple;
defaultOutputDescription.componentFlags = 0;
defaultOutputDescription.componentFlagsMask = 0;
// Get the default playback output unit
AudioComponent defaultOutput = AudioComponentFindNext(NULL, &defaultOutputDescription);
NSAssert(defaultOutput, #"Can't find default output");
// Create a new unit based on this that we'll use for output
OSErr err = AudioComponentInstanceNew(defaultOutput, &_toneUnit);
NSAssert1(_toneUnit, #"Error creating unit: %d", err);
// Set our tone rendering function on the unit
AURenderCallbackStruct input;
input.inputProc = RenderTone;
input.inputProcRefCon = (__bridge void*)self;
err = AudioUnitSetProperty(_toneUnit,
kAudioUnitProperty_SetRenderCallback,
kAudioUnitScope_Input,
0,
&input,
sizeof(input));
NSAssert1(err == noErr, #"Error setting callback: %d", err);
// Set the format to 32 bit, single channel, floating point, linear PCM
const int four_bytes_per_float = 4;
const int eight_bits_per_byte = 8;
AudioStreamBasicDescription streamFormat;
streamFormat.mSampleRate = kSampleRate;
streamFormat.mFormatID = kAudioFormatLinearPCM;
streamFormat.mFormatFlags =
kAudioFormatFlagsNativeFloatPacked | kAudioFormatFlagIsNonInterleaved;
streamFormat.mBytesPerPacket = four_bytes_per_float;
streamFormat.mFramesPerPacket = 1;
streamFormat.mBytesPerFrame = four_bytes_per_float;
streamFormat.mChannelsPerFrame = 1;
streamFormat.mBitsPerChannel = four_bytes_per_float * eight_bits_per_byte;
err = AudioUnitSetProperty (_toneUnit,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Input,
0,
&streamFormat,
sizeof(AudioStreamBasicDescription));
NSAssert1(err == noErr, #"Error setting stream format: %dd", err);
}
And here is the callback:
OSStatus RenderTone( void* inRefCon,
AudioUnitRenderActionFlags *ioActionFlags,
const AudioTimeStamp *inTimeStamp,
UInt32 inBusNumber,
UInt32 inNumberFrames,
AudioBufferList *ioData){
// Get the tone parameters out of the view controller
VWWSynthesizerC *synth = (__bridge VWWSynthesizerC *)inRefCon;
double theta = synth.theta;
double theta_increment = 2.0 * M_PI * synth.frequency / kSampleRate;
// This is a mono tone generator so we only need the first buffer
const int channel = 0;
Float32 *buffer = (Float32 *)ioData->mBuffers[channel].mData;
// Generate the samples
for (UInt32 frame = 0; frame < inNumberFrames; frame++)
{
if(synth.muted){
buffer[frame] = 0;
}
else{
switch(synth.waveType){
case VWWWaveTypeSine:{
buffer[frame] = sin(theta) * synth.amplitude;
break;
}
case VWWWaveTypeSquare:{
buffer[frame] = square(theta) * synth.amplitude;
break;
}
case VWWWaveTypeSawtooth:{
buffer[frame] = sawtooth(theta) * synth.amplitude;
break;
}
case VWWWaveTypeTriangle:{
buffer[frame] = triangle(theta) * synth.amplitude;
break;
}
default:
break;
}
}
theta += theta_increment;
if (theta > 2.0 * M_PI)
{
theta -= 2.0 * M_PI;
}
}
synth.theta = theta;
return noErr;
}
If there is a different or better way to render this data, I'm open to suggestions. I'm rendering sine, square, triangle, sawtooth, etc... waves.