[FFmpeg-devel] Audio conversion and floating-point codecs

[Axel Holzinger]

Hi Peter et al,

On Sat, Jul 10, 2010 at 10:51 AM, Peter Ross wrote:

[...]

> > Regarding audio sample format, wouldn't an approach be nice 
> > where the user (the one using libav...) can define the native 
> > audio sample format from a supported list (i.e. uint8_t, 
> > int16_t, int32_t, float, ...) as the default sample format 
> > that all audio functions will then use? Like a C++ template 
> > that can be instatiated with uint8_t, int16_t, etc.
> > 
> > I know this is a bunch of work, because it concerns so many 
> > parts in the code. But if thinking about adding more support 
> > than sole int16_t (which is a good idea I think and high 
> > time), all the possibilities should be on the table.
> 
> we already support this via sample_fmt.
> 
> (in the proposed implementation, frame->data[n] would by 
> typecast to the datatype used by codec->sample_fmt. e.g. 
> 16-bit signed interlaved, 32-bit float planar..)

sample_fmt is a codec property, would this be FFmpeg system wide or
only codec wide?

Is the idea that every codec should be able to support all the formats
that are considered being standard formats (enum SampleFormat) or is
it up to the user to check the format the codec is
accepting/delivering and converting it from/to the "system format"?

Cheers
Axel