On Mon, 2 Mar 2020, Tomas Härdin wrote:
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?)
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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