fre 2020-02-28 klockan 01:37 +0100 skrev Marton Balint: > Signed-off-by: Marton Balint <c...@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 _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-devel To unsubscribe, visit link above, or email ffmpeg-devel-requ...@ffmpeg.org with subject "unsubscribe".