> On Sep 6, 2021, at 2:28 PM, Andreas Rheinhardt
> <andreas.rheinha...@outlook.com> wrote:
>
> Paul B Mahol:
>> If audio packet is present in DV stream it have duration of 1 in DV timebase
>> units.
>>
>> Signed-off-by: Paul B Mahol <one...@gmail.com>
>> ---
>> libavformat/dv.c | 4 ++++
>> 1 file changed, 4 insertions(+)
>>
>> diff --git a/libavformat/dv.c b/libavformat/dv.c
>> index d7909683c3..b2b74162df 100644
>> --- a/libavformat/dv.c
>> +++ b/libavformat/dv.c
>> @@ -48,6 +48,7 @@ struct DVPacket {
>> int stream_index;
>> int flags;
>> int64_t pos;
>> + int64_t duration;
>> };
>>
>> struct DVDemuxContext {
>> @@ -276,6 +277,7 @@ static int dv_extract_audio_info(DVDemuxContext *c,
>> const uint8_t *frame)
>> c->audio_pkt[i].stream_index = c->ast[i]->index;
>> c->audio_pkt[i].flags |= AV_PKT_FLAG_KEY;
>> c->audio_pkt[i].pts = AV_NOPTS_VALUE;
>> + c->audio_pkt[i].duration = 0;
>> c->audio_pkt[i].pos = -1;
>> }
>> c->ast[i]->codecpar->sample_rate = dv_audio_frequency[freq];
>> @@ -374,6 +376,7 @@ int avpriv_dv_get_packet(DVDemuxContext *c, AVPacket
>> *pkt)
>> pkt->stream_index = c->audio_pkt[i].stream_index;
>> pkt->flags = c->audio_pkt[i].flags;
>> pkt->pts = c->audio_pkt[i].pts;
>> + pkt->duration = c->audio_pkt[i].duration;
>> pkt->pos = c->audio_pkt[i].pos;
>>
>> c->audio_pkt[i].size = 0;
>> @@ -404,6 +407,7 @@ int avpriv_dv_produce_packet(DVDemuxContext *c, AVPacket
>> *pkt,
>> c->audio_pkt[i].pos = pos;
>> c->audio_pkt[i].size = size;
>> c->audio_pkt[i].pts = (c->sys->height == 720) ? (c->frames & ~1) :
>> c->frames;
>> + c->audio_pkt[i].duration = 1;
>> ppcm[i] = c->audio_buf[i];
>> }
>> if (c->ach)
>>
> Sure about that? According to the code, the packets contain slightly
> different amounts of samples; see dv_extract_audio_info(), in particular
> lines 242 and 289.
> (It seems to me that one audio packet is supposed to contain the audio
> for one video frame, yet the standard sample rates and standard video
> framerates (in particular NTSC) are incompatible, so a slight variation
> in the number of samples per frame is used.)
> IMO one should set the audio timebase to be the inverse of the
> samplerate and set the duration based upon the number of samples. But I
> have zero experience with dv.
Yes the audio duration is variable in NTSC DV, see page 17 of
http://web.archive.org/web/20060927044735/http://www.smpte.org/smpte_store/standards/pdf/s314m.pdf.
Each frame has either 1600 or 1602 samples, so while the video is 1001/30000
the audio is either 1600/48000 or 1602/48000. So video at ~0.03336666667 and
audio at either ~0.03333333333 or 0.033375.
I’ve tested this patch and it does correct some errors. For instance, before
this patch frame 3 of the output only offers half of the needed duration, so
the resulting audio track is (1001/30000)/2 shorter than the video. But after
the patch, the video and audio outputs have the same length.
ffmpeg -i https://archive.org/download/test_a_202108/TESTTHIS.dv
-filter_complex "[0:a:0]aresample=async=1:min_hard_comp=0.01[aud]" -c:v:0 copy
-map 0:v:0 -c:a pcm_s16le -map "[aud]” output.mov
Dave Rice
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