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

[Marton Balint]

On Mon, 2 Mar 2020, Tomas Härdin wrote:

> 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?)

I thought D10 can only be CBR and and it can only use a constant edit unit 
size, 1/1.001 audio packet size difference is handled using KLV 
padding. So what we compute here is a _maximum_ frame size.

Regards,
Marton

>
>>                  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
>
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