> On Sep 5, 2018, at 16:06, Marton Balint <c...@passwd.hu> wrote:
>
>
>
>> On Tue, 4 Sep 2018, Markus Schumann wrote:
>>
>> ---
>> libavformat/mxfdec.c | 67 +++++++++++++++++++++++++++++++++++++++++++-
>> 1 file changed, 66 insertions(+), 1 deletion(-)
>>
>> diff --git a/libavformat/mxfdec.c b/libavformat/mxfdec.c
>> index 8e1089620f..adfffd954a
>> --- a/libavformat/mxfdec.c
>> +++ b/libavformat/mxfdec.c
>> @@ -3266,9 +3266,63 @@ static int64_t mxf_set_current_edit_unit(MXFContext
>> *mxf, AVStream *st, int64_t
>> return mxf_set_current_edit_unit(mxf, st, current_offset, 0);
>> }
>>
>> +static int mxf_aac_adts_count_frames(MXFContext *mxf, AVCodecParameters
>> *par,
>> + AVPacket *pkt)
>> +{
>> + const uint8_t *end_ptr;
>> + uint8_t *data_ptr;
>> + int32_t aac_frame_length, adts_frame_count;
>> + uint8_t sample_rate_index;
>> + uint8_t channel_configuration;
>> +
>> + data_ptr = pkt->data;
>> + end_ptr = pkt->data + pkt->size;
>> + adts_frame_count = 0;
>> +
>> + /* ADTS header is 7 bytes */
>> + if (pkt->size < 7)
>> + return 0;
>> +
>> + /* check for sync bits 0xfff */
>> + if (data_ptr[0] != 0xff || (data_ptr[1] & 0xf0) != 0xf0) return
>> AVERROR(EINVAL);
>> +
>> + sample_rate_index = (data_ptr[2] & 0x3c) >> 2;
>> +
>> + channel_configuration = ((data_ptr[2] & 0x01) << 2) | ((data_ptr[3]
>> & 0xc0) >> 6);
>> +
>> + adts_frame_count = 1;
>> +
>> + data_ptr += 7;
>> +
>> + for (; data_ptr + 7 < end_ptr; ++data_ptr)
>> + {
>> + /* check for sync bits 0xfff */
>> + if (data_ptr[0] != 0xff || (data_ptr[1] & 0xf0) != 0xf0)
>> continue;
>> +
>> + /* make sure sample rate is identical */
>> + if (sample_rate_index != (data_ptr[2] & 0x3c) >> 2) continue;
>> +
>> + /* make sure channel configuration is identical */
>> + if (channel_configuration != (((data_ptr[2] & 0x01) << 2) |
>> ((data_ptr[3] & 0xc0) >> 6))) continue;
>> +
>> + aac_frame_length = ((int32_t)(data_ptr[3] & 0x03) << 11) |
>> ((int32_t)data_ptr[4] << 3) | ((int32_t)(data_ptr[5] & 0xe0) >> 5);
>> +
>> + /* sanity check on the frame length */
>> + if (aac_frame_length < 1 || aac_frame_length > 8 * 768 + 7 +
>> 2) continue;
>> +
>> + ++adts_frame_count;
>> +
>> + data_ptr += 7;
>> + }
>> +
>> + return adts_frame_count;
>> +}
>> +
>> static int mxf_set_audio_pts(MXFContext *mxf, AVCodecParameters *par,
>> AVPacket *pkt)
>> {
>> + int frame_count;
>> +
>> MXFTrack *track = mxf->fc->streams[pkt->stream_index]->priv_data;
>> int64_t bits_per_sample = par->bits_per_coded_sample;
>>
>> @@ -3281,7 +3335,18 @@ static int mxf_set_audio_pts(MXFContext *mxf,
>> AVCodecParameters *par,
>> || bits_per_sample <= 0
>> || par->channels * (int64_t)bits_per_sample < 8)
>> return AVERROR(EINVAL);
>> - track->sample_count += pkt->size / (par->channels *
>> (int64_t)bits_per_sample / 8);
>> +
>> + switch (par->codec_id) {
>> + case AV_CODEC_ID_AAC:
>> + frame_count = mxf_aac_adts_count_frames(mxf, par, pkt);
>> + if (frame_count < 0) return AVERROR(EINVAL);
>> + track->sample_count += 1024 * frame_count;
>> + break;
>> + default:
>> + /* here we assume PCM */
>> + track->sample_count += pkt->size / (par->channels *
>> (int64_t)bits_per_sample / 8);
>> + break;
>> + }
>> return 0;
>
> I just pushed a patch which fixes the error messages decoding frame wrapped
> (as in 25fps frame wrapped) AAC packets. Obviously audio packet pts-es will
> not be sample correct, but they should be accurate enough for average use
> cases. (and it is not uncommon that demuxers are unable to provide sample
> accurate timestamps).
>
> In general it is not a good idea to put codec parsing code in a demuxer, we
> have parsers for that. So an alternative approach might be to return
> AV_NOPTS_VALUE from the mxf demuxer and let the aac parser fill in the
> timestamps, but in this case, since we can determine a rough timestamp based
> on the format, I think it is better if we return that, because that is what
> the other demuxers are doing as well.
>
> Is there a use case which does not work for you using the current git master?
>
> Thanks,
> Marton
> _______________________________________________
> ffmpeg-devel mailing list
> ffmpeg-devel@ffmpeg.org
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I just built FFmpeg master with your changes and then VLC using FFmpeg master.
My sample from ticket #7366 plays fine in VLC with your changes.
So your changes should be good enough.
Thanks Markus.
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