[FFmpeg-devel] [PATCH 2/4] avformat/mxf: get rid of samples per frame array usage

[Tomas Härdin]

fre 2020-02-28 klockan 01:37 +0100 skrev Marton Balint:
> Signed-off-by: Marton Balint <cus at passwd.hu>
> ---
>  libavformat/mxf.c    | 44 ++++----------------------------------------
>  libavformat/mxf.h    |  6 ------
>  libavformat/mxfdec.c | 23 +++--------------------
>  libavformat/mxfenc.c | 24 ++++++------------------
>  4 files changed, 13 insertions(+), 84 deletions(-)

>  int ff_mxf_get_content_package_rate(AVRational time_base)
>  {
> -    int idx = av_find_nearest_q_idx(time_base, mxf_time_base);
> -    AVRational diff = av_sub_q(time_base, mxf_time_base[idx]);
> -
> -    diff.num = FFABS(diff.num);
> -
> -    if (av_cmp_q(diff, (AVRational){1, 1000}) >= 0)
> -        return -1;
> -
> -    return mxf_content_package_rates[idx];
> +    for (int i = 0; mxf_time_base[i].num; i++)
> +        if (!av_cmp_q(time_base, mxf_time_base[i]))

I see this gets rid of the inexact check for an exact one. Approve!

> @@ -3307,20 +3307,17 @@ static int mxf_get_next_track_edit_unit(MXFContext *mxf, MXFTrack *track, int64_
>  static int64_t mxf_compute_sample_count(MXFContext *mxf, AVStream *st,
>                                          int64_t edit_unit)
>  {
> -    int i, total = 0, size = 0;
>      MXFTrack *track = st->priv_data;
>      AVRational time_base = av_inv_q(track->edit_rate);
>      AVRational sample_rate = av_inv_q(st->time_base);
> -    const MXFSamplesPerFrame *spf = NULL;
> -    int64_t sample_count;
>  
>      // For non-audio sample_count equals current edit unit
>      if (st->codecpar->codec_type != AVMEDIA_TYPE_AUDIO)
>          return edit_unit;
>  
> -    if ((sample_rate.num / sample_rate.den) == 48000)
> -        spf = ff_mxf_get_samples_per_frame(mxf->fc, time_base);
> -    if (!spf) {
> +    if ((sample_rate.num / sample_rate.den) == 48000) {
> +        return av_rescale_q(edit_unit, sample_rate, track->edit_rate);

Should be OK, per previous rounding argument

>                  }
>                  sc->index = INDEX_D10_AUDIO;
>                  sc->container_ul = ((MXFStreamContext*)s->streams[0]->priv_data)->container_ul;
> -                sc->frame_size = 4 + 8 * spf[0].samples_per_frame[0] * 4;
> +                sc->frame_size = 4 + 8 * av_rescale_rnd(st->codecpar->sample_rate, mxf->time_base.num, mxf->time_base.den, AV_ROUND_UP) * 4;

I was going to say this is only used for CBR video, but closer
inspection reveals it's used to prevent 1/1.001 rate audio packets from
making their way into CBR files. This is a bit surprising to me, since
D-10 supports NTSC (without audio?)

>                  sc->index = INDEX_WAV;
>              } else {
>                  mxf->slice_count = 1;
> -                sc->frame_size = (st->codecpar->channels * spf[0].samples_per_frame[0] *
> -                                  av_get_bits_per_sample(st->codecpar->codec_id)) / 8;
> +                sc->frame_size = st->codecpar->channels *
> +                                 av_rescale_rnd(st->codecpar->sample_rate, mxf->time_base.num, mxf->time_base.den, AV_ROUND_UP) *
> +                                 av_get_bits_per_sample(st->codecpar->codec_id) / 8;

Looks similarly OK

/Tomas