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• ffmpeg how to concat and COMPRESS at the same time ? [closed]

  17 juin 2022, par Alex Nox

  Concat command :

  


  ffmpeg -f concat -safe 0 -i list.txt -c copy output.mp4


  


  Content of list.txt

  


  file '2022-03-26-1.mp4'

file '2022-03-26-2.mp4'


  


  The following above seems working, but I want to compress files at the same time, so I type :

  


  ffmpeg concat -an -i list.txt -crf 23 -c:v libx264 out.mp4


  


  But it doesn't work ? Why ?

  


• ffmpeg taking so much time to extract frames from mpeg-ts file

  19 septembre 2016, par Syed Azaruddin

  I got a problem from one mpeg-ts video. Actually it was created by some one else and even I don’t know how they are created. The problem is that, ffmpeg is taking so much time to decode all frames from the mpeg-ts video. The command I used for this operation is...

  ffmpeg -i shame-run.mov -r 24/1 test/output%d.jpg

  Actually my application is integrated with ffmpeg v2.1.1. and I had a code for detecting black frames in a mpeg-ts video. Here, my code is not able to detect all black frames from ffmpeg for this mpeg-ts video. So, I taken standalone ffmpeg of same version as mentioned above and decoded standalone. But, it is taking so much time i.e., it is taken half an hour for decoding 1 min duration video.

  So, is there any reason is that ffmpeg not able to extract as fast as required, and is there any reason that video got some errors.

• Libav API : How to modify the start time of an audio stream ?

  24 avril 2022, par xtingray

  Currently, I am working on a new feature for my software using the Libav API. I was able to merge a video file with and audio file, the output is an MP4 file and the source code works perfectly.

  


  Right now, I have a new requirement : the start time for the video and the audio streams are the same. Both start at 00:00. My next challenge is to add an option to shift the start time for the audio stream based on a variable. For example, if audio start variable is equal to 10 seconds, then the audio stream should be played at time 00:10.

  


  So I wonder, what is the best approach to implement this feature ? Should I insert silent packets before the audio start time is reached ? Or should I modify the timestamp information for every audio stream packet before it is written into the output container ?

  


  Here is the piece of code I use to write the audio stream into the MP4 file. Right now it works like a charm, but I guess this is the place where I should implement the new requirement. I will appreciate any suggestion.

  


  while (1) {&#xA;    AVStream *in_stream;&#xA;&#xA;    int ret = av_read_frame(audioInputFormatContext, pkt);&#xA;    if (ret &lt; 0)&#xA;        break;&#xA;&#xA;    in_stream  = audioInputFormatContext->streams[pkt->stream_index];&#xA;    pkt->stream_index = outIndex;&#xA;&#xA;    AVRational out_time_base = audioStreamList.at(i)->time_base;&#xA;&#xA;    // copy packet&#xA;    pkt->pts = av_rescale_q_rnd(pkt->pts, in_stream->time_base, audioStreamList.at(i)->time_base,&#xA;                                static_cast<avrounding>(AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX));&#xA;    pkt->dts = av_rescale_q_rnd(pkt->dts, in_stream->time_base, audioStreamList.at(i)->time_base,&#xA;                                static_cast<avrounding>(AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX));&#xA;    pkt->duration = av_rescale_q(pkt->duration, in_stream->time_base, audioStreamList.at(i)->time_base);&#xA;    pkt->pos = -1;&#xA;&#xA;    ret = av_interleaved_write_frame(formatContext, pkt);&#xA;    if (ret &lt; 0) {&#xA;        fprintf(stderr, "Error muxing packet\n");&#xA;        break;&#xA;    }&#xA;    av_packet_unref(pkt);&#xA;}&#xA;</avrounding></avrounding>