If I open an audio file with extended audio file services, using the following client data format...
AudioStreamBasicDescription audioFormat;
memset(&audioFormat, 0, sizeof(audioFormat));
audioFormat.mSampleRate = 44100.0;
audioFormat.mFormatID = kAudioFormatLinearPCM;
audioFormat.mFormatFlags = kAudioFormatFlagIsBigEndian |
kAudioFormatFlagIsSignedInteger |
kAudioFormatFlagIsPacked;
audioFormat.mBytesPerPacket = 4;
audioFormat.mFramesPerPacket = 1;
audioFormat.mChannelsPerFrame = 2;
audioFormat.mBytesPerFrame = 4;
audioFormat.mBitsPerChannel = 16;
And configure an AudioBufferList like so....
AudioBufferList bufferList;
bufferList.mNumberBuffers = 1;
bufferList.mBuffers[0].mDataByteSize = bufferSize;
bufferList.mBuffers[0].mNumberChannels = audioFormat.mChannelsPerFrame;
bufferList.mBuffers[0].mData = buffer; //malloc(sizeof(UInt8) * 1024 * audioFormat.mBytesPerPacket)
How, then, is the data arranged in mData? If I iterate through the data like so
for (int i = 0; i < frameCount; i++) {
UInt8 somePieceOfAudioData = buffer[i];
}
then what is somePieceOfAudioData.
Is it a sample or a frame (left and right channels together)? If it's a sample then what channel is it a sample for? If for example it's a sample from the right channel, will buffer[i + 1] be a sample for the left channel?
Any ideas, links? Thank you!
Audio data is expected to be interleaved unless the kAudioFormatFlagIsNonInterleaved
is set. I've found that for Core Audio questions the best source of documentation is usually the headers. CoreAudioTypes.h
contains the following comment:
Typically, when an ASBD is being used, the fields describe the complete layout of the sample data in the buffers that are represented by this description - where typically those buffers are represented by an AudioBuffer that is contained in an AudioBufferList.
However, when an ASBD has the kAudioFormatFlagIsNonInterleaved flag, the AudioBufferList has a different structure and semantic. In this case, the ASBD fields will describe the format of ONE of the AudioBuffers that are contained in the list, AND each AudioBuffer in the list is determined to have a single (mono) channel of audio data. Then, the ASBD's mChannelsPerFrame will indicate the total number of AudioBuffers that are contained within the AudioBufferList - where each buffer contains one channel. This is used primarily with the AudioUnit (and AudioConverter) representation of this list - and won't be found in the AudioHardware usage of this structure.
In your particular case, the buffer will consist of interleaved shorts starting with the left channel.
Yeah, you're reading a frame and it's two 16-bit samples, Left and Right. (Actually, I'm not certain which is Left and which is Right. Hmmm.)
In addition to the header files, the class references built into Xcode are helpful. I find I'm using "option-click" and "command-click" in my code a lot when I'm sorting out these kinds of details. (for those new to Xcode.. these clicks get you the info and docs, and jump-to-source location, respectively.)
The upcoming book "Learning Core Audio: A Hands-on Guide to Audio Programming for Mac and iOS" by Kevin Avila and Chris Adamson does a nice job of explaining how all this works. It's available in "Rough Cut" form now at Safari Books Online:
http://my.safaribooksonline.com/book/audio/9780321636973
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