[FFmpeg-devel] [PATCH 1/4] avformat/audiointerleave: disallow using a samples_per_frame array

[Tomas Härdin]

Sorry for replying a bit late, I've been sick

fre 2020-02-28 klockan 01:37 +0100 skrev Marton Balint:
> Only MXF used an actual sample array, and that is unneeded there
> because simple
> rounding rules can be used instead.

Does this produce the exact same rounding? Like 1602, 1601,
 1602, 1601, 1602, 1062 ... for 30/1.001 fps? If so then this is a nice
solution


> Signed-off-by: Marton Balint <cus at passwd.hu>
> ---
>  libavformat/audiointerleave.c | 24 ++++++++++--------------
>  libavformat/audiointerleave.h |  7 ++++---
>  libavformat/gxfenc.c          |  2 +-
>  libavformat/mxfenc.c          |  7 ++-----
>  4 files changed, 17 insertions(+), 23 deletions(-)
> 
> diff --git a/libavformat/audiointerleave.c b/libavformat/audiointerleave.c
> index 6797546a44..f9887c1f4a 100644
> --- a/libavformat/audiointerleave.c
> +++ b/libavformat/audiointerleave.c
> @@ -39,14 +39,11 @@ void ff_audio_interleave_close(AVFormatContext *s)
>  }
>  
>  int ff_audio_interleave_init(AVFormatContext *s,
> -                             const int *samples_per_frame,
> +                             const int samples_per_frame,
>                               AVRational time_base)
>  {
>      int i;
>  
> -    if (!samples_per_frame)
> -        return AVERROR(EINVAL);
> -
>      if (!time_base.num) {
>          av_log(s, AV_LOG_ERROR, "timebase not set for audio interleave\n");
>          return AVERROR(EINVAL);
> @@ -56,18 +53,18 @@ int ff_audio_interleave_init(AVFormatContext *s,
>          AudioInterleaveContext *aic = st->priv_data;
>  
>          if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
> +            int max_samples = samples_per_frame ? samples_per_frame : av_rescale_rnd(st->codecpar->sample_rate, time_base.num, time_base.den, AV_ROUND_UP);

Looks correct

>              aic->sample_size = (st->codecpar->channels *
>                                  av_get_bits_per_sample(st->codecpar->codec_id)) / 8;
>              if (!aic->sample_size) {
>                  av_log(s, AV_LOG_ERROR, "could not compute sample size\n");
>                  return AVERROR(EINVAL);
>              }
> -            aic->samples_per_frame = samples_per_frame;
> -            aic->samples = aic->samples_per_frame;
>              aic->time_base = time_base;
> +            aic->samples_per_frame = samples_per_frame;
>  
> -            aic->fifo_size = 100* *aic->samples;
> -            if (!(aic->fifo= av_fifo_alloc_array(100, *aic->samples)))
> +            aic->fifo_size = 100 * max_samples;
> +            if (!(aic->fifo = av_fifo_alloc_array(100, max_samples)))
>                  return AVERROR(ENOMEM);
>          }
>      }
> @@ -81,7 +78,8 @@ static int interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt,
>      AVStream *st = s->streams[stream_index];
>      AudioInterleaveContext *aic = st->priv_data;
>      int ret;
> -    int frame_size = *aic->samples * aic->sample_size;
> +    int nb_samples = aic->samples_per_frame ? aic->samples_per_frame : (av_rescale_q(aic->n + 1, av_make_q(st->codecpar->sample_rate, 1), av_inv_q(aic->time_base)) - aic->nb_samples);

Here's where the meat is. av_rescale_q() means AV_ROUND_NEAR_INF. So
basically round(). Fiddling around a bit in Octave to replicate the
logic:

  octave:13> round(48000*1001/30000)
  ans =  1602
  octave:14> round(48000*1001/30000*2)-1602
  ans =  1601
  octave:15> round(48000*1001/30000*3)-(1602+1601)
  ans =  1602
  octave:16> round(48000*1001/30000*4)-(1602+1601+1602)
  ans =  1601
  octave:17> 48000*1001/30000*5-(1602+1601+1602+1601)
  ans =  1602

Seems kosher. The rest of the patch looks OK.

mxfdec.c could also make use of this. Then we could remove
ff_mxf_get_samples_per_frame() and related code. Replacing
mxf_content_package_rates with a simple formula would also be nice.

/Tomas