[FFmpeg-devel] avcodec_encode_audio() and libavcodec/pcm.c

[Stefano Sabatini]

On date Sunday 2008-05-25 20:36:25 +0200, Stefano Sabatini encoded:
> Hi all,
> 
> maybe this belongs to libav-user, anyway...
> 
> According to documentation for avcodec_encode_audio():
> 
>  * @param[in] samples the input buffer containing the samples
>  * The number of samples read from this buffer is frame_size*channels,
>  * both of which are defined in \p avctx.
> 
> This isn't true for PCM audio, for example with PCM 16 the function
> reads exactly a number of bytes which is buf_size/sample_size, and
> this causes also the strange switch in ffmpeg.c:
> 
>         /* output a pcm frame */
>         /* XXX: change encoding codec API to avoid this ? */
>         switch(enc->codec->id) {
>         case CODEC_ID_PCM_S32LE:
>         case CODEC_ID_PCM_S32BE:
>         case CODEC_ID_PCM_U32LE:
>         case CODEC_ID_PCM_U32BE:
>             size_out = size_out << 1;
>             break;
>         case CODEC_ID_PCM_S24LE:
>         case CODEC_ID_PCM_S24BE:
>         case CODEC_ID_PCM_U24LE:
>         case CODEC_ID_PCM_U24BE:
>         case CODEC_ID_PCM_S24DAUD:
>             size_out = size_out / 2 * 3;
>             break;
>         case CODEC_ID_PCM_S16LE:
>         case CODEC_ID_PCM_S16BE:
>         case CODEC_ID_PCM_U16LE:
>         case CODEC_ID_PCM_U16BE:
>             break;
>         default:
>             size_out = size_out >> 1;
>             break;
>         }
> 
> Can someone comment on this (pretty weird) behaviour, there is some
> hope to change it to make it match the documentation for
> avcodec_encode_audio() or application writers have to hardcode the
> above logic into every application which deals with PCM audio?
> 
> Thanks in advance for any clarification/suggestion.

Well, now I see it is required in order to be able to encode large
chunks of data at once, attched patch document it.

BTW sometime ago someone proposed to extend the API to encode audio
data using AVFrame, I think about something like:
int avcodec_encode_audio2(AVCodecContext *avctx, uint8_t *buf, int buf_size, const AVFrame *frame);

is there some progress with it?

Regards.
-- 
FFmpeg = Fantastic and Foolish MultiPurpose EniGma
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