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• Is there a way to change the ffplay playback speed while running

  20 mai 2024, par richjhart

  I am trying to modify ffplay 7.0 to allow us to modify the playback speed while running. Our use-case only involves basic video-only mp4 files.

  


  We can set the playback speed with something like the following :

  


  ffplay -i ..\Video\SampleVideoLong.mp4 -vf "setpts=5.0*PTS" -loglevel debug -sync video

  


  But I don't know how to change that option "live" (note if I don't choose -sync video, it gets very laggy - but that's fine as our use-case is video only.

  


  I have tried the following (with just a fixed rate at the moment) :

  


          {
            static double l_CurrentSpeed = 1.0;
            double l_NewSpeed = 2.0;
            double l_Diff = l_NewSpeed / l_CurrentSpeed;
            double l_ClockSpeed = cur_stream->extclk.speed;
            double l_NewClockSpeed = l_ClockSpeed * l_Diff;
            av_log(NULL, AV_LOG_DEBUG,
                "Changing speed from %f to %f\n",
                l_ClockSpeed, l_NewClockSpeed
                );

            set_clock_speed(&cur_stream->vidclk, l_NewClockSpeed);
            //set_clock(&cur_stream->vidclk, l_NewClockSpeed, cur_stream->extclk.serial);
            l_CurrentSpeed = l_NewSpeed;
        }


  


  I've set both the pts (set_clock()) and the "speed" (set_clock_speed()), but neither of these had any effect.

  


  Would I need to something extra, or is there a way to update the setpts expression in the video filter from ffplay ?

  


  Note the metadata of the file we are trying with is :

  


  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '..\Video\SampleVideoLong.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2mp41
    encoder         : Lavf61.1.100
  Duration: 00:18:10.56, start: 0.160000, bitrate: 1755 kb/s
  Stream #0:0[0x1](und), 1, 1/90000: Video: mpeg2video (Main), 1 reference frame (mp4v / 0x7634706D), yuv420p(tv, bt709, progressive, left), 1920x1080 [SAR 1:1 DAR 16:9], 0/1, 1754 kb/s, 25 fps, 25 tbr, 90k tbn (default)
      Metadata:
        handler_name    : VideoHandler
        vendor_id       : [0][0][0][0]
        encoder         : XDCAM EX 1080p25
      Side data:
        cpb: bitrate max/min/avg: 0/0/0 buffer size: 278528 vbv_delay: N/A


  


  I believe the files we'll be using should have identical or similar metadata.

  


• mencoder. Encoding from multiple input image files compatible with web browser (No video support and MIME type) [duplicate]

  23 juin 2020, par iblasi

  I have multiple JPG files that I want to use to make a TimeLapse video compatible with the web browser to upload it on my web page.
Create a video with mencoder from multiple images is explained in some webpages such us here, that shows how to create a video.

  


  ls -Ltr my_Pics/*.jpg >files.txt
mencoder -nosound -ovc lavc -lavcopts vcodec=mpeg4 -o video.avi -mf type=jpeg:fps=4 mf://@files.txt


  


  The video is set with no sound and to have one picture every 250ms (4 fps).
These command lines create an AVI video that I can see correctly with the VLC video tool. However, if I try to open it in a web browser it shows an error :

  


  


  No Video with Supported Format and MIME type found

  


  


  So, based on other similar comments (as here), I tryed to use ffmpeg renaming all my files as ffmpeg requires a number serial format. But it happens the same, that I can see it in VLC but not in the browser.

  


  ffmpeg -r 4 -i ./output/%04d.jpg -vcodec libx264 video.mp4


  


  Based on research made on internet I am quite sure that it is due the the encoding and/or container. I tryed multiple options of codecs nd containers existing on documentation (here) but still not able to find a way to work.

  


  If, once I create the video, I use the VLC tool to manually convert the video to ".m4v" I was able to create a video that the web browser recognizes. But I would like to do it with command lines to automate it.

  


• Encode Ogg/Opus with very specific structure

  28 septembre 2022, par sonovice

  I have an ogg/opus file with a pretty special structure that I would like to reproduce.

  


  opusinfo returns a few hints of how it was made :

  


  New logical stream (#1, serial: 59d24a85): type opus
Encoded with Lavf56.40.101
User comments section follows...
        encoder=Lavc56.60.100 libopus
        encoder_options=--quiet --bitrate 96 --vbr
        pad=00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000
Opus stream 1:
        Pre-skip: 312
        Playback gain: 0 dB
        Channels: 2
        Original sample rate: 48000Hz
        Packet duration:   60.0ms (max),   56.6ms (avg),   20.0ms (min)
        Page duration:   1460.0ms (max),  323.6ms (avg),  120.0ms (min)
        Total data length: 38872802 bytes (overhead: 1.12%)
        Playback length: 51m:11.473s
        Average bitrate: 101.2 kb/s, w/o overhead: 100.1 kb/s
Logical stream 1 ended


  


  Lavf/Lavc suggests that it was made with FFMPEG, but I am not able to find suitable settings.

  


  Looking at the actual bytes, it starts with an 0x200 (512) byte long ogg page containing metadata, followed by an 0xE00 (3584) byte long ogg page with actual audio content. Both pages sum up exactly to 0x1000 (4096) bytes.


  


  After that, each page is exactly 0x1000 (4096) bytes long with potential zero padding, so that the pages start at offsets 0x1000, 0x2000, 0x3000 etc.


  


  Is there any known tool or library that can produce such ogg/opus files from PCM wave audio ?

  


  (Unfortunately I cannot share the file as it contains proprietary content.)