I had a need to transmit sound over the network and for this I chose libraries "PortAudio" and "Opus". I am new to working with sound and therefore i don’t know many thing.I am new to working with sound and therefore i don’t know many things, but i read the documentation and looked at some examples, but i still have some problems with encoding/decoding with Opus. I do not understand how to correctly restore the original encoded PСM.I have some sequence of actions:
Some consts
const int FRAMES_PER_BUFFER = 960;
const int SAMPLE_RATE = 48000;
int NUM_CHANNELS = 2;
int totalFrames = 2 * SAMPLE_RATE; /* Record for a few seconds. */
int numSamples = totalFrames * 2;
int numBytes = numSamples * sizeof(float);
float *sampleBlock = nullptr;
int bytesOfPacket = 0;
unsigned char *packet = nullptr;
I get PСM to sampleBlock
paError = Pa_ReadStream(**&stream, sampleBlock, totalFrames);
if (paError != paNoError) {
cout << "PortAudio error : " << Pa_GetErrorText(paError) << endl;
std::system("pause");
}
Encoding sampleBlock
OpusEncoder *encoder;
int error;
int size;
encoder = opus_encoder_create(SAMPLE_RATE, NUM_CHANNELS, OPUS_APPLICATION_VOIP, &error);
size = opus_encoder_get_size(NUM_CHANNELS);
encoder = (OpusEncoder *)malloc(size);
packet = new unsigned char[480];
error = opus_encoder_init(encoder, SAMPLE_RATE, NUM_CHANNELS, OPUS_APPLICATION_VOIP);
if (error == -1) {
return -1;
}
bytesOfPacket = opus_encode_float(encoder, sampleBlock, FRAMES_PER_BUFFER, packet, 480);
opus_encoder_destroy(encoder);
Ok, i received a encoded packet to Opus
Decoding
OpusDecoder *decoder;
int error;
int size;
decoder = opus_decoder_create(SAMPLE_RATE, NUM_CHANNELS, &error);
size = opus_decoder_get_size(NUM_CHANNELS);
decoder = (OpusDecoder *)malloc(size);
error = opus_decoder_init(decoder, SAMPLE_RATE, NUM_CHANNELS);
opus_decode_float(decoder, packet, bytesOfPacket, sampleBlock, 480, 0);
opus_decoder_destroy(decoder);
Here i am trying to decode the Opus back to the PCM and save the result to the sampleBlock
Playing the sound
paError = Pa_WriteStream(**&stream, sampleBlock, totalFrames);
if (paError != paNoError) {
cout << "PortAudio error : " << Pa_GetErrorText(paError) << endl;
std::system("pause");
}
I get silence. I don't really understand the subtleties in working with sound since i am new to this business. Help please understand what is wrong.
As for your settings you're encoding 20ms of audio per opus_encode_float call. I don't see any iteration over this call so I suppose you don't hear anything because you encode only 20ms of audio. You should pass to opus_encode_float 20ms worth of samples with your sampleBlock pointer incrementing it through the whole buffer x times.
Try to encode more audio and remember that you have to add some sort of framing to decode it. You cannot just feed the whole buffer to the decoder. You should feed the decoder one time for each encoder call with the same data that each encoder call outputs.
Damiano
Related
I write a voip app that uses "novocaine" library for recording and playback of sound. I set sample rate as 8kHz. This sample rate is set in novocaine in AudioStreamBasicDescription of audio unit and as audio session property kAudioSessionProperty_PreferredHardwareSampleRate. I understand that setting preferred hardware sample rate has no guarantee that actual hardware sample rate will be changed but it worked for all devices except iPhone6s and iPhone6s+ (when route is changed to speaker). With iPhone6s(+) and speaker route I receive 48kHz sound from microphone. So I need to somehow convert this 48 kHz sound to 8kHz. In documentation I found that AudioConverterRef can be used in this case but I have troubles with using it.
I use AudioConverterFillComplexBuffer for sample rate conversion but it always returns -50 OSStatus (one or more parameters passed to the function were not valid). This is how I use audio converter:
// Setup AudioStreamBasicDescription for input
inputFormat.mSampleRate = 48000.0;
inputFormat.mFormatID = kAudioFormatLinearPCM;
inputFormat.mFormatFlags = kAudioFormatFlagIsFloat | kAudioFormatFlagsNativeEndian | kAudioFormatFlagIsPacked;
inputFormat.mChannelsPerFrame = 1;
inputFormat.mBitsPerChannel = 8 * sizeof(float);
inputFormat.mFramesPerPacket = 1;
inputFormat.mBytesPerFrame = sizeof(float) * inputFormat.mChannelsPerFrame;
inputFormat.mBytesPerPacket = inputFormat.mBytesPerFrame * inputFormat.mFramesPerPacket;
// Setup AudioStreamBasicDescription for output
outputFormat.mSampleRate = 8000.0;
outputFormat.mFormatID = kAudioFormatLinearPCM;
outputFormat.mFormatFlags = kAudioFormatFlagIsFloat | kAudioFormatFlagsNativeEndian | kAudioFormatFlagIsPacked;
outputFormat.mChannelsPerFrame = 1;
outputFormat.mBitsPerChannel = 8 * sizeof(float);
outputFormat.mFramesPerPacket = 1;
outputFormat.mBytesPerFrame = sizeof(float) * outputFormat.mChannelsPerFrame;
outputFormat.mBytesPerPacket = outputFormat.mBytesPerFrame * outputFormat.mFramesPerPacket;
// Create new instance of audio converter
AudioConverterNew(&inputFormat, &outputFormat, &converter);
// Set conversion quality
UInt32 tmp = kAudioConverterQuality_Medium;
AudioConverterSetProperty( converter, kAudioConverterCodecQuality,
sizeof( tmp ), &tmp );
AudioConverterSetProperty( converter, kAudioConverterSampleRateConverterQuality, sizeof( tmp ), &tmp );
// Get the size of the IO buffer(s)
UInt32 bufferSizeFrames = 0;
size = sizeof(UInt32);
AudioUnitGetProperty(self.inputUnit,
kAudioDevicePropertyBufferFrameSize,
kAudioUnitScope_Global,
0,
&bufferSizeFrames,
&size);
UInt32 bufferSizeBytes = bufferSizeFrames * sizeof(Float32);
// Allocate an AudioBufferList plus enough space for array of AudioBuffers
UInt32 propsize = offsetof(AudioBufferList, mBuffers[0]) + (sizeof(AudioBuffer) * outputFormat.mChannelsPerFrame);
// Malloc buffer lists
convertedInputBuffer = (AudioBufferList *)malloc(propsize);
convertedInputBuffer->mNumberBuffers = 1;
// Pre-malloc buffers for AudioBufferLists
convertedInputBuffer->mBuffers[0].mNumberChannels = outputFormat.mChannelsPerFrame;
convertedInputBuffer->mBuffers[0].mDataByteSize = bufferSizeBytes;
convertedInputBuffer->mBuffers[0].mData = malloc(bufferSizeBytes);
memset(convertedInputBuffer->mBuffers[0].mData, 0, bufferSizeBytes);
// Setup callback for converter
static OSStatus inputProcPtr(AudioConverterRef inAudioConverter,
UInt32* ioNumberDataPackets,
AudioBufferList* ioData,
AudioStreamPacketDescription* __nullable* __nullable outDataPacketDescription,
void* __nullable inUserData)
{
// Read data from buffer
}
// Perform actual sample rate conversion
AudioConverterFillComplexBuffer(converter, inputProcPtr, NULL, &numberOfFrames, convertedInputBuffer, NULL)
inputProcPtr callback is never called. I tried to set different number of frames but still receive OSStatus -50.
1) Is using AudioConverterRef is correct way to make sample rate conversion or it could be done in different way?
2) What is wrong with my conversion implementation?
Thank you all in advance
One problem is this:
AudioUnitGetProperty(self.inputUnit,
kAudioDevicePropertyBufferFrameSize,
kAudioUnitScope_Global,
0,
&bufferSizeFrames,
&size);
kAudioDevicePropertyBufferFrameSize is an OSX property, and doesn't exist on iOS. How is this code even compiling?
If you've somehow made it compile, check the return code from this function! I've got a feeling that it's failing, and bufferSizeFrames is zero. That would make AudioConverterFillComplexBuffer return -50 (kAudio_ParamError).
So on iOS, either pick a bufferSizeFrames yourself or base it on AVAudioSession's IOBufferDuration if you must.
Another problem: check your return codes. All of them!
e.g.
UInt32 tmp = kAudioConverterQuality_Medium;
AudioConverterSetProperty( converter, kAudioConverterCodecQuality,
sizeof( tmp ), &tmp );
I'm pretty sure there's no codec to speak of in LPCM->LPCM conversions, and that kAudioConverterQuality_Medium is not the right value to use with kAudioConverterCodecQuality in any case. I don't see how this call can succeed.
I am converting from the following format:
const int four_bytes_per_float = 4;
const int eight_bits_per_byte = 8;
_stereoGraphStreamFormat.mFormatID = kAudioFormatLinearPCM;
_stereoGraphStreamFormat.mFormatFlags = kAudioFormatFlagsNativeFloatPacked | kAudioFormatFlagIsNonInterleaved;
_stereoGraphStreamFormat.mBytesPerPacket = four_bytes_per_float;
_stereoGraphStreamFormat.mFramesPerPacket = 1;
_stereoGraphStreamFormat.mBytesPerFrame = four_bytes_per_float;
_stereoGraphStreamFormat.mChannelsPerFrame = 2;
_stereoGraphStreamFormat.mBitsPerChannel = eight_bits_per_byte * four_bytes_per_float;
_stereoGraphStreamFormat.mSampleRate = 44100;
to the following format:
interleavedAudioDescription.mFormatID = kAudioFormatLinearPCM;
interleavedAudioDescription.mFormatFlags = kAudioFormatFlagIsSignedInteger;
interleavedAudioDescription.mChannelsPerFrame = 2;
interleavedAudioDescription.mBytesPerPacket = sizeof(SInt16)*interleavedAudioDescription.mChannelsPerFrame;
interleavedAudioDescription.mFramesPerPacket = 1;
interleavedAudioDescription.mBytesPerFrame = sizeof(SInt16)*interleavedAudioDescription.mChannelsPerFrame;
interleavedAudioDescription.mBitsPerChannel = 8 * sizeof(SInt16);
interleavedAudioDescription.mSampleRate = 44100;
Using the following code:
int32_t availableBytes = 0;
void* tailL = TPCircularBufferTail(inputBufferL(), &availableBytes);
void* tailR = TPCircularBufferTail(inputBufferR(), &availableBytes);
// If we have no data in the buffer, we simply return
if (availableBytes <= 0)
{
return;
}
// ========== Non-Interleaved to Interleaved (Plus Samplerate Conversion) =========
// Get the number of frames available
UInt32 frames = availableBytes / this->mInputFormat.mBytesPerFrame;
pcmOutputBuffer->mBuffers[0].mDataByteSize = frames * interleavedAudioDescription.mBytesPerFrame;
struct complexInputDataProc_t data = (struct complexInputDataProc_t) { .self = this, .sourceL = tailL, .sourceR = tailR, .byteLength = availableBytes };
// Do the conversion
OSStatus result = AudioConverterFillComplexBuffer(interleavedAudioConverter,
complexInputDataProc,
&data,
&frames,
pcmOutputBuffer,
NULL);
// Tell the buffers how much data we consumed during the conversion so that it can be removed
TPCircularBufferConsume(inputBufferL(), availableBytes);
TPCircularBufferConsume(inputBufferR(), availableBytes);
// ========== Buffering Of Interleaved Samples =========
// If we got converted frames back from the converter, we want to add it to a separate buffer
if (frames > 0)
{
// Make sure we have enough space in the buffer to store the new data
TPCircularBufferHead(&pcmCircularBuffer, &availableBytes);
if (availableBytes > pcmOutputBuffer->mBuffers[0].mDataByteSize)
{
// Add the newly converted data to the buffer
TPCircularBufferProduceBytes(&pcmCircularBuffer, pcmOutputBuffer->mBuffers[0].mData, frames * interleavedAudioDescription.mBytesPerFrame);
}
else
{
printf("No Space in Buffer\n");
}
}
However I am getting the following output:
It should be a perfect sine wave, however as you can see it is not.
I have been working on this for days now and just can’t seem to find where it is going wrong.
Can anyone see something that I might be missing?
Edit:
Buffer initialisation:
TPCircularBuffer pcmCircularBuffer;
static SInt16 pcmOutputBuf[BUFFER_SIZE];
pcmOutputBuffer = (AudioBufferList*)malloc(sizeof(AudioBufferList));
pcmOutputBuffer->mNumberBuffers = 1;
pcmOutputBuffer->mBuffers[0].mNumberChannels = 2;
pcmOutputBuffer->mBuffers[0].mData = pcmOutputBuf;
TPCircularBufferInit(&pcmCircularBuffer, BUFFER_SIZE);
Complex input data proc:
static OSStatus complexInputDataProc(AudioConverterRef inAudioConverter,
UInt32 *ioNumberDataPackets,
AudioBufferList *ioData,
AudioStreamPacketDescription **outDataPacketDescription,
void *inUserData) {
struct complexInputDataProc_t *arg = (struct complexInputDataProc_t*)inUserData;
BroadcastingServices::MP3Encoder *self = (BroadcastingServices::MP3Encoder*)arg->self;
if ( arg->byteLength <= 0 )
{
*ioNumberDataPackets = 0;
return 100; //kNoMoreDataErr;
}
UInt32 framesAvailable = arg->byteLength / self->interleavedAudioDescription.mBytesPerFrame;
if (*ioNumberDataPackets > framesAvailable)
{
*ioNumberDataPackets = framesAvailable;
}
ioData->mBuffers[0].mData = arg->sourceL;
ioData->mBuffers[0].mDataByteSize = arg->byteLength;
ioData->mBuffers[1].mData = arg->sourceR;
ioData->mBuffers[1].mDataByteSize = arg->byteLength;
arg->byteLength = 0;
return noErr;
}
I see a few things that raise a red flag.
1) as mentioned in a comment above, the fact that you are overwriting availableBytes for the left input with that from the right:
void* tailL = TPCircularBufferTail(inputBufferL(), &availableBytes);
void* tailR = TPCircularBufferTail(inputBufferR(), &availableBytes);
If the two input streams are changing asynchronously to this code then most certainly you have a race condition.
2) Truncation errors: availableBytes is not necessarily a multiple of bytes per frame. If not then the following bit of code could cause you to consume more bytes than you actually converted.
void* tailL = TPCircularBufferTail(inputBufferL(), &availableBytes);
void* tailR = TPCircularBufferTail(inputBufferR(), &availableBytes);
...
UInt32 frames = availableBytes / this->mInputFormat.mBytesPerFrame;
...
TPCircularBufferConsume(inputBufferL(), availableBytes);
TPCircularBufferConsume(inputBufferR(), availableBytes);
3) If the output buffer is not ready to consume all of the input you just throw the input buffer away. That happens in this code.
if (availableBytes > pcmOutputBuffer->mBuffers[0].mDataByteSize)
{
...
}
else
{
printf("No Space in Buffer\n");
}
I'd be really curious if your seeing the print output.
Here's is how I would suggest doing it. It's going to be pseudo-codeish since I don't have anything necessary to compile and test it.
int32_t availableBytesInL = 0;
int32_t availableBytesInR = 0;
int32_t availableBytesOut = 0;
// figure out how many bytes are available in each buffer.
void* tailL = TPCircularBufferTail(inputBufferL(), &availableBytesInL);
void* tailR = TPCircularBufferTail(inputBufferR(), &availableBytesInR);
TPCircularBufferHead(&pcmCircularBuffer, &availableBytesOut);
// figure out how many full frames are available
UInt32 framesInL = availableBytesInL / mInputFormat.mBytesPerFrame;
UInt32 framesInR = availableBytesInR / mInputFormat.mBytesPerFrame;
UInt32 framesOut = availableBytesOut / interleavedAudioDescription.mBytesPerFrame;
// figure out how many frames to process this time.
UInt32 frames = min(min(framesInL, framesInL), framesOut);
if (frames == 0)
return;
int32_t bytesConsumed = frames * mInputFormat.mBytesPerFrame;
struct complexInputDataProc_t data = (struct complexInputDataProc_t) {
.self = this, .sourceL = tailL, .sourceR = tailR, .byteLength = bytesConsumed };
// Do the conversion
OSStatus result = AudioConverterFillComplexBuffer(interleavedAudioConverter,
complexInputDataProc,
&data,
&frames,
pcmOutputBuffer,
NULL);
int32_t bytesProduced = frames * interleavedAudioDescription.mBytesPerFrame;
// Tell the buffers how much data we consumed during the conversion so that it can be removed
TPCircularBufferConsume(inputBufferL(), bytesConsumed);
TPCircularBufferConsume(inputBufferR(), bytesConsumed);
TPCircularBufferProduceBytes(&pcmCircularBuffer, pcmOutputBuffer->mBuffers[0].mData, bytesProduced);
Basically what I've done here is to figure out up front how many frames should be processed making sure I'm only processing as many frames as the output buffer can handle. If it were me I'd also add some checking for buffer underruns on the output and buffer overruns on the input. Finally, I'm not exactly sure of the semantics of AudioConverterFillComplexBuffer wrt the frame parameter that is passing in and out. It's conceivable that the # frames out would be less or more than the number of frames in. Although, since your not doing sample rate conversion that's probably not going to happen. I've attempted to account for that condition by assigning bytesProduced after the conversion.
Hope this helps. If not you have 2 other clues. One is that the drop outs are periodic and two is that the size of the drop outs looks to be about the same. If you can figure out how many samples each are then you can look for those numbers in your code.
I don't think your output buffer, pcmCircularBuffer, is big enough.
Try replacing
TPCircularBufferInit(&pcmCircularBuffer, BUFFER_SIZE);
with
TPCircularBufferInit(&pcmCircularBuffer, sizeof(pcmOutputBuf));
Even if that is the solution, I think there are some problems with your code. I don't know exactly what you're doing, I guess encoding mp3 (which by itself is an uphill battle on iOS, why not use hardware AAC?), but unless you have realtime demands on both input and output, why use ring buffers at all? Also, I recommend using units to visually catch type frame/byte size mismatches: e.g. BUFFER_SIZE_IN_FRAMES
If it's not the solution, then I want to see the sine generator.
So ,I need to reverse some audio *.caf file,
I have seen that the way to do it should be:
You cannot just reverse the byte data. I have achieved the same
effect using CoreAudio and AudioUnits. Use ExtFileReader C API to read
the file into lPCM buffers and then you can reverse the buffers as
needed.
But I cannot find any documentation of the use of
ExtFileReader C API
So if I have a *.caf file, how can I read it in to a linear PCM, I have checked the Core Audio overview but cant find how to accomplish this?
How can i then, read my caf file to linear PCM?
thanks!
ExtendedAudioFile is in the AudioToolbox framework. It's pretty straightforward to read in a file to whatever format you'd like. Here's a quick (compiles, but not tested) example of reading in to 32-bit float non-interleaved Linear PCM:
#import <AudioToolbox/AudioToolbox.h>
...
ExtAudioFileRef audioFile = NULL;
CFURLRef url = NULL;
OSStatus err = ExtAudioFileOpenURL(url, &audioFile);
AudioStreamBasicDescription asbd;
UInt32 dataSize = sizeof(asbd);
// get the audio file's format
err = ExtAudioFileGetProperty(audioFile, kExtAudioFileProperty_FileDataFormat, &dataSize, &asbd);
// now set the client format to what we want on read (LPCM, 32-bit floating point)
AudioStreamBasicDescription clientFormat = asbd;
clientFormat.mFormatID = kAudioFormatLinearPCM;
clientFormat.mFormatFlags = kAudioFormatFlagIsFloat | kAudioFormatFlagIsNonInterleaved | kAudioFormatFlagIsPacked;
clientFormat.mBitsPerChannel = 32;
clientFormat.mBytesPerPacket = 4;
clientFormat.mFramesPerPacket = 1;
clientFormat.mBytesPerFrame = 4;
err = ExtAudioFileSetProperty(audioFile, kExtAudioFileProperty_ClientDataFormat, sizeof(clientFormat), &clientFormat);
// okay, now the ext audio file is set up to convert samples to LPCM on read
// get the total number of samples
SInt64 numFrames = 0;
dataSize = sizeof(numFrames);
err = ExtAudioFileGetProperty(audioFile, kExtAudioFileProperty_FileLengthFrames, &dataSize, &numFrames);
// prepare an audio buffer list to hold the data when we read it from the file
UInt32 maxReadFrames = 4096; // how many samples will we read in at a time?
AudioBufferList *bufferList = (AudioBufferList *)malloc(sizeof(AudioBufferList) + sizeof(AudioBuffer) * (asbd.mChannelsPerFrame - 1));
bufferList->mNumberBuffers = asbd.mChannelsPerFrame;
for (int ii = 0; ii < bufferList->mNumberBuffers; ++ii) {
bufferList->mBuffers[ii].mDataByteSize = maxReadFrames * sizeof(float);
bufferList->mBuffers[ii].mData = malloc(bufferList->mBuffers[ii].mDataByteSize);
bzero(bufferList->mBuffers[ii].mData, bufferList->mBuffers[ii].mDataByteSize);
bufferList->mBuffers[ii].mNumberChannels = 1;
}
while(numFrames > 0) {
UInt32 framesToRead = (maxReadFrames > numFrames) ? numFrames : maxReadFrames;
err = ExtAudioFileRead(audioFile, &framesToRead, bufferList);
// okay, your LPCM audio data is in `bufferList` -- do whatever processing you'd like!
}
// clean up
for (int ii = 0; ii < bufferList->mNumberBuffers; ++ii) {
free(bufferList->mBuffers[ii].mData);
}
free(bufferList);
ExtAudioFileDispose(audioFile);
Im stuck on an issue on my objective C App.
I'm reading a byte array from a serveur (Socket c#) who send me an PCM encoded sound, and i'm currently looking for a sample code that decode for me this byte array (NSData), and play it.
Does anyone know a solution ? Or how can I read a u-Law audio?
Thanks a lot ! :D
This link has information about mu-law encoding and decoding:
http://dystopiancode.blogspot.com.es/2012/02/pcm-law-and-u-law-companding-algorithms.html
#define MULAW_BIAS 33
/*
* Description:
* Decodes an 8-bit unsigned integer using the mu-Law.
* Parameters:
* number - the number who will be decoded
* Returns:
* The decoded number
*/
int16_t MuLaw_Decode(int8_t number)
{
uint8_t sign = 0, position = 0;
int16_t decoded = 0;
number=~number;
if(number&0x80)
{
number&=~(1<<7);
sign = -1;
}
position = ((number & 0xF0) >>4) + 5;
decoded = ((1<<position)|((number&0x0F)<<(position-4))|(1<<(position-5)))
- MULAW_BIAS;
return (sign==0)?(decoded):(-(decoded));
}
When you have the uncompressed audio you should be able to play it using the Audio Queue API.
Good luck!
I am trying to access the raw data for an audio file on the iPhone/iPad. I have the following code which is a basic start down the path I need. However I am stumped at what to do once I have an AudioBuffer.
AVAssetReader *assetReader = [AVAssetReader assetReaderWithAsset:urlAsset error:nil];
AVAssetReaderTrackOutput *assetReaderOutput = [AVAssetReaderTrackOutput assetReaderTrackOutputWithTrack:[[urlAsset tracks] objectAtIndex:0] outputSettings:nil];
[assetReader addOutput:assetReaderOutput];
[assetReader startReading];
CMSampleBufferRef ref;
NSArray *outputs = assetReader.outputs;
AVAssetReaderOutput *output = [outputs objectAtIndex:0];
int y = 0;
while (ref = [output copyNextSampleBuffer]) {
AudioBufferList audioBufferList;
CMBlockBufferRef blockBuffer;
CMSampleBufferGetAudioBufferListWithRetainedBlockBuffer(ref, NULL, &audioBufferList, sizeof(audioBufferList), NULL, NULL, 0, &blockBuffer);
for (y=0; y<audioBufferList.mNumberBuffers; y++) {
AudioBuffer audioBuffer = audioBufferList.mBuffers[y];
SInt16 *frames = audioBuffer.mData;
for(int i = 0; i < 24000; i++) { // This sometimes crashes
Float32 currentFrame = frames[i] / 32768.0f;
}
}
}
Essentially I don't know how to tell how many frames each buffer contains so I can't reliably extract the data from them. I am new to working with raw audio data so I'm open to any suggestions in how to best read the mData property of the AudioBuffer struct. I also haven't done much with void pointers in the past so help with that in this context would be great too!
audioBuffer.mDataByteSize tells you the size of the buffer. Did you know this? Just incase you didn't you can't have looked at the declaration of struct AudioBuffer. You should always look at the header files as well as the docs.
For the mDataByteSize to make sense you must know the format of the data. The count of output values is mDataByteSize/sizeof(outputType). However, you seem confused about the format - you must have specified it somewhere. First of all you treat it as a 16bit signed int
SInt16 *frames = audioBuffer.mData
then you treat it as 32 bit float
Float32 currentFrame = frames[i] / 32768.0f
inbetween you assume that there are 24000 values, of course this will crash if there aren't exactly 24000 16bit values. Also, you refer to the data as 'frames' but what you really mean is samples. Each value you call 'currentFrame' is one sample of the audio. 'Frame' would typically refer to a block of samples like .mData
So, assuming the data format is 32bit Float (and please note, i have no idea if it is, it could be 8 bit int, or 32bit Fixed for all i know)
for( int y=0; y<audioBufferList.mNumberBuffers; y++ )
{
AudioBuffer audioBuffer = audioBufferList.mBuffers[y];
int bufferSize = audioBuffer.mDataByteSize / sizeof(Float32);
Float32 *frame = audioBuffer.mData;
for( int i=0; i<bufferSize; i++ ) {
Float32 currentSample = frame[i];
}
}
Note, sizeof(Float32) is always 4, but i left it in to be clear.