Encoding 32bit AudioUnitCanonical stereo PCM samples to mp3 with lame

Question

I'm directly converting 32bit stereo samples from a digital line in on iOS devices using Mickey Blue microphone addon to mp3 with lame.

The problem is lame only seems to accept 16bit samples. Trying to directly convert the samples just leaves me with a lot of noise and no recognizable audio. The settings I use to record the audio on iOS are as follows

AudioStreamBasicDescription stereoStreamFormat;
stereoStreamFormat.mSampleRate          = 44100.00;
stereoStreamFormat.mFormatID            = kAudioFormatLinearPCM;
stereoStreamFormat.mFormatFlags         = kAudioFormatFlagsAudioUnitCanonical;
stereoStreamFormat.mBytesPerPacket      = 4;
stereoStreamFormat.mBytesPerFrame       = 4;
stereoStreamFormat.mFramesPerPacket     = 1;
stereoStreamFormat.mChannelsPerFrame    = 2;
stereoStreamFormat.mBitsPerChannel      = 32;//changing this to 16bit does not work

I then convert the samples from 2 individual buffers with lame:

-(void) encodeMicrophoneWithLeft :(NSData*) left andRight : (NSData *) right
{
    void *mp3 = malloc(MP3_SIZE_MIC);
    int write = lame_encode_buffer(microphoneLame, left.bytes, right.bytes, left.length/2, mp3, MP3_SIZE_MIC);
    [self.microphoneBuffer appendBytes:mp3 length:write];
}

With these settings:

lame_set_brate(microphoneLame, 128);// current streaming speed kbps
lame_set_in_samplerate(microphoneLame, 44100);
lame_set_VBR(microphoneLame, vbr_off);//set variable bitrate off
lame_set_num_channels(microphoneLame,2);

Is there any way I can convert the 32bit samples to 16bit or make lame work with 32bit samples? The recorder doesn't work with settings other then 32bit for stereo somehow, but if there's another way to initialise with 16bit that would be a solution for me too.

Thanks, I've tried that before but it didn't seem to work. Perhaps I made some mistakes so will try again following that path. — Dirk de Boer, May 27 '14 at 07:17
"That didn't seem to work" is not a mathematically correct response. — jaket, May 27 '14 at 07:26
IOW, if you have a 32 bit signal and drop the bottom 16 bits to make a 16 bit signal, then you have not done much but to throw away the insignificant portion of the signal. — jaket, May 27 '14 at 07:30
Well in previous attempts I would get a load of distortion lopping the lower 16bit and in other tries to get valid samples for lame. The distortion seemed completely random, nothing audible. I don't have the setup anymore so couldn't post a more detailed description atm. I also found [this](http://stackoverflow.com/questions/16478003/ios-core-audio-converting-between-kaudioformatflagscanonical-and-kaudioformatf) which seems like the format requires a bit more work. I'll post the results as soon as I get them. — Dirk de Boer, May 27 '14 at 10:16

score 0 · Answer 1 · answered May 29 '14 at 16:49

0

Dropping 16 of the bits won't work, because kAudioFormatFlagsAudioUnitCanonical makes your stream floating-point :) AudioUnit streams are 32 bit float.

Don't use kAudioFormatFlagsAudioUnitCanonical for your format flags. Read the header and see what LAME needs. kAudioFormatFlagsAreAllClear and mBitsPerChannel of 16 should work (also mutatis mutandis all the bytes-per-frame fields etc.). That'll give you 16 bit little-endian integer samples.

answered May 29 '14 at 16:49

iluvcapra

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Thanks, but it seems to only accepts this specific format with stereo recording using the Mickey Blue addon. It does record in mono with 16bit but I need stereo samples. – Dirk de Boer Jun 05 '14 at 08:23

Dirk de Boer · Accepted Answer · 2015-05-26T12:53:12.850

It seems like kAudioFormatFlagsAudioUnitCanonical is indeed not just 32bit, but a fixed point 24bit sample stored in a floating point notation. The most significant 7 bits can be dropped, only the least significant bit of the first 8 most significant bits functions to sign the value.

       sign    actual sample data
      /       /
|--8--|------24------|
     |----16----| <- part you need

This means I could bitshift the samples 9 to the right and then directly cast them to shorts, leaving me with 16bit signed samples.

Like this:

NSData *left =[[NSData alloc]initWithBytes:tempBufferLeft.mData length:tempBufferLeft.mDataByteSize ];
NSData *right =[[NSData alloc]initWithBytes:tempBufferRight.mData length:tempBufferRight.mDataByteSize ];

convertedArrayLeft = [[NSMutableData alloc]init];
convertedArrayRight = [[NSMutableData alloc]init];
for (int i = 0; i < left.length; i+=4)//steps of 4 bytes
{
    int tmpValueL, tmpValueR;
    [left getBytes:&tmpValueL range:NSMakeRange(i, 4)];//extract float samples into int for easier manipulation
    [right getBytes:&tmpValueR range:NSMakeRange(i, 4)];

    short endValueL = tmpValueL>>9, endValueR = tmpValueR>>9;//bitshift 9 to the right and save as short to lose most significant 16 bits that are now useless

    [convertedArrayLeft appendBytes:&endValueL length:sizeof(short)];
    [convertedArrayRight appendBytes:&endValueR length:sizeof(short)];
}

I couldn't find out why it would not accept an other format in stereo recording, but this works too with a little overhead.

Encoding 32bit AudioUnitCanonical stereo PCM samples to mp3 with lame

2 Answers2