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• avformat/smacker : Improve timestamps

  24 juin 2020, par Andreas Rheinhardt

  avformat/smacker : Improve timestamps
  A Smacker file can contain up to seven audio tracks. Up until now,
  
  the pts for the i. audio packet contained in a Smacker frame was
  
  simply the end timestamp of the last i. audio packet contained in
  
  an earlier Smacker frame.
  The problem with this is that a Smacker stream need not contain data in
  
  every Smacker frame and so the current i. audio packet present may come
  
  from a different underlying stream than the last i. audio packet
  
  contained in an earlier frame.
  The sample hypnotix.smk* exhibits this. It has three audio tracks and
  
  the first of the three has a longer first packet, so that the audio for
  
  the first track is contained in only 235 packets contained in the first
  
  235 Smacker frames ; the end timestamp of this track is 166696 (about 7.56s
  
  at a timebase of 1/22050) ; the other two audio tracks both have 253 packets
  
  contained in the first 253 Smacker frames. Up until now, the 236th
  
  packet of the second track being the first audio packet in the 236th
  
  Smacker frame would get the end timestamp of the last first audio packet
  
  from the last Smacker frame containing a first audio packet and said
  
  last audio packet is the first audio packet from the 235th Smacker frame
  
  from the first audio track, so that the timestamp is 166696. In contrast,
  
  the 236th packet from the third track (whose packets contain the same number
  
  of samples as the packets from the second track) has a timestamp of
  
  156116 (because its timestamp is derived from the end timestamp of the
  
  235th packet of the second audio track). In the end, the second track
  
  ended up being 177360/22050 s = 8.044s long ; in contrast, the third
  
  track was 166780/22050 s = 7.56s long which also coincided with the
  
  video.
  This commit fixes this by not using timestamps from other tracks for
  
  a packet's pts.
  * : https://samples.ffmpeg.org/game-formats/smacker/wetlands/hypnotix.smk
  Reviewed-by : Timotej Lazar <timotej.lazar@araneo.si>
  
  Signed-off-by : Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
  

  • [DH] libavformat/smacker.c

• Display FFMPEG decoded frame in a GLFW window

  17 juin 2020, par Infecto

  I am implementing the client program of a game where the server sends encoded frames of the game to the client (via UDP), while the client decodes them (via FFMPEG) and displays them in a GLFW window. 
My program has two threads :

  



  

  1. Thread 1 : renders the content of the uint8_t* variable dataToRender

  


  2. Thread 2 : keeps obtaining frames from the server, decodes them and updates dataToRender accordingly

  


  



  Thread 1 does the typical rendering of a GLFW window in a while-loop. I have already tried to display some dummy frame data (a completely red frame) and it worked :

  



  while (!glfwWindowShouldClose(window)) {
    glClear(GL_COLOR_BUFFER_BIT | GL_DEPTH_BUFFER_BIT);
    ...

    glBindTexture(GL_TEXTURE_2D, tex_handle);
    glTexImage2D(GL_TEXTURE_2D, 0, GL_RGB, window_width, window_height, 0, GL_RGB, GL_UNSIGNED_BYTE, dataToRender);
    ...
    glfwSwapBuffers(window);
}


  



  Thread 2 is where I am having trouble. I am unable to properly store the decoded frame into my dataToRender variable. On top if it, the frame data is originally in YUV format and needs to be converted to RGB. I use FFMPEG's sws_scale for that, which also gives me a bad dst image pointers error output in the console. Here's the code snippet responsible for that part :

  



          size_t data_size = frameBuffer.size();  // frameBuffer is a std::vector where I accumulate the frame data chunks
        uint8_t* data = frameBuffer.data();  // convert the vector to a pointer
        picture->format = AV_PIX_FMT_RGB24;
        av_frame_get_buffer(picture, 1);
        while (data_size > 0) {
            int ret = av_parser_parse2(parser, c, &pkt->data, &pkt->size,
                data, data_size, AV_NOPTS_VALUE, AV_NOPTS_VALUE, 0);
            if (ret < 0) {
                fprintf(stderr, "Error while parsing\n");
                exit(1);
            }
            data += ret;
            data_size -= ret;

            if (pkt->size) {
                swsContext = sws_getContext(
                    c->width, c->height,
                    AV_PIX_FMT_YUV420P, c->width, c->height,
                    AV_PIX_FMT_RGB24, SWS_BILINEAR, NULL, NULL, NULL
                );
                uint8_t* rgb24[1] = { data };
                int rgb24_stride[1] = { 3 * c->width };
                sws_scale(swsContext, rgb24, rgb24_stride, 0, c->height, picture->data, picture->linesize);

                decode(c, picture, pkt, outname);
                // TODO: copy content of picture->data[0] to "dataToRender" maybe?
            }
        }


  



  I have already tried doing another sws_scale to copy the content to dataToRender and I cannot get rid of the bad dst image pointers error. Any advice or solution to the problem would be greatly appreciated as I have been stuck for days on this.

  


• Set "start" field to 0 in mp3 with ffmpeg

  11 juin 2020, par TomatoCo

  I'm trying to change the bitrate and sample rate of an MP3 to match another to try and stop a small audio glitch from occurring when some game tries to play it. I've got the sample rate and bitrate right where I want them, but I can't get the "start" portion of

  



    Duration: 00:03:33.81, start: 0.025057, bitrate: 196 kb/s
    Stream #0:0: Audio: mp3, 44100 Hz, stereo, fltp, 196 kb/s


  



  to go to 0, like the mp3 I'm trying to replace. The target looks like :

  



    Duration: 00:06:47.59, start: 0.000000, bitrate: 196 kb/s
    Stream #0:0: Audio: mp3, 44100 Hz, stereo, fltp, 196 kb/s


  



  I've tried a variety of silenceremove filters and -ss flags to try and trim it, but I can't get rid of that "start" field. Google is failing me. What args am I looking for ?