[Flac-dev] Using libFLAC++

[Josh Coalson]

samples come out of the decoder in 32-bit signed int.  the loop is
just converting them to 16-bit signed.

the aliasing of 'buffer' into 's16buffer' etc is just a hack to use
the same buffer space to convert to different target formats.

--- David Bishop <tech at bishop.dhs.org> wrote:
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> [meta: Thanks, Josh, for clearing that up]
> 
> Continuing my quest to make everyone think I'm an idiot, could
> someone 
> explain the following code to me?  I.e., what the two for loops are 
> accomplishing, and what, *exactly*, is in s16buffer when it's done. 
> Assume I want to write a .wav file from this decoded flac*.  Is it
> already in
> approximately the right format, or do I have to do some sort of 
> conversion? This code is ripped almost directly from decode.c in
> flac/, 
> just heavily cut down to deal with only 1 type of file (i.e., I'm not
> 
> dealing with differing endianess right now).
> 
> [this code is in my write callback]
> 
> static FLAC__int8 s8buffer[FLAC__MAX_BLOCK_SIZE * FLAC__MAX_CHANNELS
> * sizeof(FLAC__int32)];
> FLAC__int16  *s16buffer = (FLAC__int16  *)s8buffer;
> 
> if( wide_samples > 0 ) {
>     for(sample = wide_sample = 0; wide_sample < wide_samples;
> wide_sample++) {
>         for(channel = 0; channel < channels; channel++, sample++) {
>             s16buffer[sample] =
> (FLAC__int16)(buffer[channel][wide_sample]);
>         }
>     }
> }
> 
> Sorry for the newbie-style questions, I'm afraid I'm not that good,
> and
> it's showing through :-)
> 
> D.A.Bishop
> 
> *I don't, but assume I do :-)
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