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• 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>

  


• FFMPEG streaming RTP : time base not set

  25 octobre 2020, par Managarm

  I'm trying to create a small demo to get a feeling for streaming programmatically with ffmpeg. I'm using the code from this question as a basis. I can compile my code, but when I try to run it I always get this error :

  



  


  [rtp @ 0xbeb480] time base not set

  


  



  The thing is, I have set the time base parameters. I even tried setting them for the stream (and the codec associated with the stream) as well, even though this should not be necessary as far as I understand it. This is the relevant section in my code :

  



  AVCodec* codec = avcodec_find_encoder(AV_CODEC_ID_H264);
AVCodecContext* c = avcodec_alloc_context3(codec);
c->pix_fmt = AV_PIX_FMT_YUV420P;
c->flags = CODEC_FLAG_GLOBAL_HEADER;
c->width = WIDTH;
c->height = HEIGHT;
c->time_base.den = FPS;
c->time_base.num = 1;
c->gop_size = FPS;
c->bit_rate = BITRATE;

avcodec_open2(c, codec, NULL);
struct AVStream* stream = avformat_new_stream(avctx, codec);

// TODO: causes an error
avformat_write_header(avctx, NULL);


  



  The error occurs when calling "avformat_write_header" near the end. All methods that can fail (like avcodec_open2) are checked, I just removed the checks to make the code more readable.

  



  Digging through google and the ffmpeg source code didn't yield any useful results. I think it's really basic, but I'm stuck. Who can help me ?