[FFmpeg-devel] [PATCH] avformat/wavdec: More accurate duration

[Andreas Rheinhardt]

Danivy:
> Using data size and nAvgBytesPerSec to calculate the duration will be more accurate
> ---
>  libavformat/wavdec.c | 7 +++++--
>  1 file changed, 5 insertions(+), 2 deletions(-)
> 
> diff --git a/libavformat/wavdec.c b/libavformat/wavdec.c
> index d2fb81ca7f..50b1f81e62 100644
> --- a/libavformat/wavdec.c
> +++ b/libavformat/wavdec.c
> @@ -185,7 +185,8 @@ static int wav_parse_fmt_tag(AVFormatContext *s, int64_t size, AVStream *st)
>  
>      st->internal->need_parsing = AVSTREAM_PARSE_FULL_RAW;
>  
> -    avpriv_set_pts_info(st, 64, 1, st->codecpar->sample_rate);
> +    uint64_t nAvgBytesPerSec = st->codecpar->bit_rate / 8;
> +    avpriv_set_pts_info(st, 64, 1, nAvgBytesPerSec);
>  
>      return 0;
>  }
> @@ -637,7 +638,9 @@ break_loop:
>                                    /
>                  (st->codecpar->channels * (uint64_t)av_get_bits_per_sample(st->codecpar->codec_id));
>  
> -    if (sample_count)
> +    if (data_size)
> +        st->duration = data_size;
> +    else if (sample_count)
>          st->duration = sample_count;
>  
>      if (st->codecpar->codec_id == AV_CODEC_ID_PCM_S32LE &&
> 
1. What is the point of using a finer timestamp scale than the sample rate?
2. There are scenarios where this leads to a coarser timestamp scale;
e.g. for mono adpcm yamaha files.
3. Parts of the code apparently still presume that the timebase is the
sample rate; therefore FATE (our regression testing system) fails with
your patch: https://patchwork.ffmpeg.org/check/36501/

- Andreas