Re: [xine-devel] ffmpeg audio plugin and 24 bits per sample problem

[Thibaut M. <thi...@gm...>]

On 5/10/06, Mike Melanson <mi...@mu...> wrote:
> Thibaut Mattern wrote:
> > Hi,
> >
> > currently 16 bits per sample is forced due to some obscure reason.
> >
> > Mike, do you understand this comment ?
> >
> >      /* Current ffmpeg audio decoders always use 16 bits/sample
> >       * buf->decoder_info[2] can't be used as it doesn't refer to the
> > output
> >       * bits/sample for some codecs (e.g. MS ADPCM) */
> >      this->audio_bits =3D 16;
> >
> >      this->context->bits_per_sample =3D this->audio_bits;
> >      this->context->sample_rate =3D this->audio_sample_rate;
> >      this->context->channels    =3D this->audio_channels;
> >      this->context->codec_id    =3D this->codec->id;
>
> I think I was only saying that FFmpeg only supports 16-bit output.
> Ordinarily, buf->decoder_info[2] contains the bits/sample as specified
> by the demuxer. But there is apparently no point in providing that to
> FFmpeg.

ok, do you know if ffmpeg handles 24 bits flac files ?
i don't care if the output is downsampled to 16 bits, it would be
better than the random noise we get right now.

>         Maybe... :)
> --
>         -Mike Melanson

Thibaut