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

[Jean-Daniel Dupas]

Le 10 juil. 2010 ? 11:14, Axel Holzinger a ?crit :

> 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"?
> 

Isn't sample_fmt equivalent to pix_fmt for images ?
I imagine you may have some negotiation between client and codec to get the best format, and also have a swscale library for audio that provide a simple way to  convert a sample from/to native codec format.


-- Jean-Daniel