Question

I would like to extract a channel audio from the an LPCM raw file ie extract left and right channel of a stereo LPCM file. The LPCM is 16 bit depth,interleaved, 2 channels,litle endian. From what I gather the order of byte is {LeftChannel,RightChannel,LeftChannel,RightChannel…} and since it is 16 bit depth there will be 2 bytes of sample for each channel right?

So my question is if i want to extract the left channel then I would take the bytes in 0,2,4,6…n*2 address? while the right channel would be 1,3,4,…(n*2+1).

Also after extracting the audio channel, should i set the format of the extracted channel as 16 bit depth ,1 channel?

Thanks in advance

This is the code that I currently use to extract PCM audio from AssetReader.. This code works fine with writing a music file without its channel being extracted so I it might be caused by the format or something…

    NSURL *assetURL = [song valueForProperty:MPMediaItemPropertyAssetURL];
AVURLAsset *songAsset = [AVURLAsset URLAssetWithURL:assetURL options:nil];
NSDictionary *outputSettings = [NSDictionary dictionaryWithObjectsAndKeys:
                                [NSNumber numberWithInt:kAudioFormatLinearPCM], AVFormatIDKey, 
                                [NSNumber numberWithFloat:44100.0], AVSampleRateKey,
                                [NSNumber numberWithInt:2], AVNumberOfChannelsKey,
                            //  [NSData dataWithBytes:&channelLayout length:sizeof(AudioChannelLayout)], AVChannelLayoutKey,
                                [NSNumber numberWithInt:16], AVLinearPCMBitDepthKey,
                                [NSNumber numberWithBool:NO], AVLinearPCMIsNonInterleaved,
                                [NSNumber numberWithBool:NO],AVLinearPCMIsFloatKey,
                                [NSNumber numberWithBool:NO], AVLinearPCMIsBigEndianKey,
                                nil];
NSError *assetError = nil;
AVAssetReader *assetReader = [[AVAssetReader assetReaderWithAsset:songAsset
                                                            error:&assetError]
                              retain];
if (assetError) {
    NSLog (@"error: %@", assetError);
    return;
}

AVAssetReaderOutput *assetReaderOutput = [[AVAssetReaderAudioMixOutput 
                                           assetReaderAudioMixOutputWithAudioTracks:songAsset.tracks
                                           audioSettings: outputSettings]
                                          retain];
if (! [assetReader canAddOutput: assetReaderOutput]) {
    NSLog (@"can't add reader output... die!");
    return;
}
[assetReader addOutput: assetReaderOutput];


NSArray *dirs = NSSearchPathForDirectoriesInDomains(NSDocumentDirectory, NSUserDomainMask, YES);
NSString *documentsDirectoryPath = [dirs objectAtIndex:0];

//CODE TO SPLIT STEREO
[self setupAudioWithFormatMono:kAudioFormatLinearPCM];
NSString *splitExportPath = [[documentsDirectoryPath stringByAppendingPathComponent:@"monoleft.caf"] retain];
if ([[NSFileManager defaultManager] fileExistsAtPath:splitExportPath]) {
    [[NSFileManager defaultManager] removeItemAtPath:splitExportPath error:nil];
}

AudioFileID mRecordFile;
NSURL *splitExportURL = [NSURL fileURLWithPath:splitExportPath];


OSStatus status =  AudioFileCreateWithURL(splitExportURL, kAudioFileCAFType, &_streamFormat, kAudioFileFlags_EraseFile,
                                          &mRecordFile);

NSLog(@"status os %d",status);

[assetReader startReading];

CMSampleBufferRef sampBuffer = [assetReaderOutput copyNextSampleBuffer];
UInt32 countsamp= CMSampleBufferGetNumSamples(sampBuffer);
NSLog(@"number of samples %d",countsamp);

SInt64 countByteBuf = 0;
SInt64 countPacketBuf = 0;
UInt32 numBytesIO = 0;
UInt32 numPacketsIO = 0;
NSMutableData * bufferMono = [NSMutableData new];
while (sampBuffer) {


    AudioBufferList  audioBufferList;
    CMBlockBufferRef blockBuffer;
    CMSampleBufferGetAudioBufferListWithRetainedBlockBuffer(sampBuffer, NULL, &audioBufferList, sizeof(audioBufferList), NULL, NULL, 0, &blockBuffer);
    for (int y=0; y<audioBufferList.mNumberBuffers; y++) {
        AudioBuffer audioBuffer = audioBufferList.mBuffers[y];
        //frames = audioBuffer.mData;
        NSLog(@"the number of channel for buffer number %d is %d",y,audioBuffer.mNumberChannels);
        NSLog(@"The buffer size is %d",audioBuffer.mDataByteSize);






        //Append mono left to buffer data
        for (int i=0; i<audioBuffer.mDataByteSize; i= i+4) {
            [bufferMono appendBytes:(audioBuffer.mData+i) length:2];
        }

        //the number of bytes in the mutable data containing mono audio file
        numBytesIO = [bufferMono length];
        numPacketsIO = numBytesIO/2;
        NSLog(@"numpacketsIO %d",numPacketsIO);
        status = AudioFileWritePackets(mRecordFile, NO, numBytesIO, &_packetFormat, countPacketBuf, &numPacketsIO, audioBuffer.mData);
        NSLog(@"status for writebyte %d, packets written %d",status,numPacketsIO);
        if(numPacketsIO != (numBytesIO/2)){
            NSLog(@"Something wrong");
            assert(0);
        }


        countPacketBuf = countPacketBuf + numPacketsIO;
        [bufferMono setLength:0];


    }

    sampBuffer = [assetReaderOutput copyNextSampleBuffer];
    countsamp= CMSampleBufferGetNumSamples(sampBuffer);
    NSLog(@"number of samples %d",countsamp);
}
AudioFileClose(mRecordFile);
[assetReader cancelReading];
[self performSelectorOnMainThread:@selector(updateCompletedSizeLabel:)
                       withObject:0
                    waitUntilDone:NO];

The output format with audiofileservices is as follows:

        _streamFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked;
    _streamFormat.mBitsPerChannel = 16;
    _streamFormat.mChannelsPerFrame = 1;
    _streamFormat.mBytesPerPacket = 2;
    _streamFormat.mBytesPerFrame = 2;// (_streamFormat.mBitsPerChannel / 8) * _streamFormat.mChannelsPerFrame;
    _streamFormat.mFramesPerPacket = 1;
    _streamFormat.mSampleRate = 44100.0;

    _packetFormat.mStartOffset = 0;
    _packetFormat.mVariableFramesInPacket = 0;
    _packetFormat.mDataByteSize = 2;

Answer 1

Sounds almost right – you have a 16 bit depth, so that means each sample will take 2 bytes. That means the left channel data will be in bytes {0,1}, {4,5}, {8,9} and so on. Interleaved means the samples are interleaved, not the bytes.
Other than that I would try it out and see if you have any problems with your code.

Also after extracting the audio
channel, should i set the format of
the extracted channel as 16 bit depth
,1 channel?

Only one of the two channels is remaining after your extraction, so yes, this is correct.