I have an audio app i need to capture mic samples to encode into mp3 with ffmpeg
First configure the audio:
/**
* We need to specifie our format on which we want to work.
* We use Linear PCM cause its uncompressed and we work on raw data.
* for more informations check.
*
* We want 16 bits, 2 bytes (short bytes) per packet/frames at 8khz
*/
AudioStreamBasicDescription audioFormat;
audioFormat.mSampleRate = SAMPLE_RATE;
audioFormat.mFormatID = kAudioFormatLinearPCM;
audioFormat.mFormatFlags = kAudioFormatFlagIsPacked | kAudioFormatFlagIsSignedInteger;
audioFormat.mFramesPerPacket = 1;
audioFormat.mChannelsPerFrame = 1;
audioFormat.mBitsPerChannel = audioFormat.mChannelsPerFrame*sizeof(SInt16)*8;
audioFormat.mBytesPerPacket = audioFormat.mChannelsPerFrame*sizeof(SInt16);
audioFormat.mBytesPerFrame = audioFormat.mChannelsPerFrame*sizeof(SInt16);
The recording callback is:
static OSStatus recordingCallback(void *inRefCon,
AudioUnitRenderActionFlags *ioActionFlags,
const AudioTimeStamp *inTimeStamp,
UInt32 inBusNumber,
UInt32 inNumberFrames,
AudioBufferList *ioData)
{
NSLog(@"Log record: %lu", inBusNumber);
NSLog(@"Log record: %lu", inNumberFrames);
NSLog(@"Log record: %lu", (UInt32)inTimeStamp);
// the data gets rendered here
AudioBuffer buffer;
// a variable where we check the status
OSStatus status;
/**
This is the reference to the object who owns the callback.
*/
AudioProcessor *audioProcessor = (__bridge AudioProcessor*) inRefCon;
/**
on this point we define the number of channels, which is mono
for the iphone. the number of frames is usally 512 or 1024.
*/
buffer.mDataByteSize = inNumberFrames * sizeof(SInt16); // sample size
buffer.mNumberChannels = 1; // one channel
buffer.mData = malloc( inNumberFrames * sizeof(SInt16) ); // buffer size
// we put our buffer into a bufferlist array for rendering
AudioBufferList bufferList;
bufferList.mNumberBuffers = 1;
bufferList.mBuffers[0] = buffer;
// render input and check for error
status = AudioUnitRender([audioProcessor audioUnit], ioActionFlags, inTimeStamp, inBusNumber, inNumberFrames, &bufferList);
[audioProcessor hasError:status:__FILE__:__LINE__];
// process the bufferlist in the audio processor
[audioProcessor processBuffer:&bufferList];
// clean up the buffer
free(bufferList.mBuffers[0].mData);
//NSLog(@"RECORD");
return noErr;
}
With data:
inBusNumber = 1
inNumberFrames = 1024
inTimeStamp = 80444304 // All the time same inTimeStamp, this is strange
However, the framesize that i need to encode mp3 is 1152. How can i configure it?
If i do buffering, that implies a delay, but i would like to avoid this because is a real time app. If i use this configuration, each buffer i get trash trailing samples, 1152 - 1024 = 128 bad samples. All samples are SInt16.