Recherche avancée

Sur d’autres sites (11560)

• Pass embedded closed captions with FFmpeg

  4 décembre 2020, par Andrew

  I am taking an HLS stream and outputting it through a DeckLink card to convert it to SDI. I need to pass the CEA-708 closed captions that are embedded in the stream, and it has to happen live. This is the command I'm currently using :

  



  ffmpeg.exe -loglevel info -re -i https://my-url.m3u8 -c:v wrapped_avframe -pix_fmt uyvy422 -vf scale=1280x720 -c:a pcm_s16le -ar 48000 -ac 2 -f decklink -s 1280x720 -r 60000/1001 -b:v 4M "DeckLink SDI (2)"

  



  I have tried adding the -teletext all command to no avail.

  



  Here is the output of FFprobe :

  



  Input #0, hls, from 'https://my-url.m3u8':
  Duration: N/A, start: 83023.890600, bitrate: N/A
  Program 0
    Metadata:
      variant_bitrate : 998000
    Stream #0:0: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p(progressive), 640x360, Closed Captions, 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc
    Metadata:
      variant_bitrate : 998000
    Stream #0:1(eng): Audio: aac (LC) ([15][0][0][0] / 0x000F), 44100 Hz, stereo, fltp, 158 kb/s (default)
    Metadata:
      variant_bitrate : 998000
      comment         : Master Audio
    Stream #0:2: Data: timed_id3 (ID3  / 0x20334449)
    Metadata:
      variant_bitrate : 998000
  Program 1
    Metadata:
      variant_bitrate : 658000
    Stream #0:3: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p(progressive), 480x270, Closed Captions, 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc
    Metadata:
      variant_bitrate : 658000
    Stream #0:4(eng): Audio: aac (LC) ([15][0][0][0] / 0x000F), 44100 Hz, stereo, fltp, 158 kb/s (default)
    Metadata:
      variant_bitrate : 658000
      comment         : Master Audio
    Stream #0:5: Data: timed_id3 (ID3  / 0x20334449)
    Metadata:
      variant_bitrate : 658000
  Program 2
    Metadata:
      variant_bitrate : 1958000
    Stream #0:6: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p(progressive), 960x540, Closed Captions, 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc
    Metadata:
      variant_bitrate : 1958000
    Stream #0:7(eng): Audio: aac (LC) ([15][0][0][0] / 0x000F), 44100 Hz, stereo, fltp, 158 kb/s (default)
    Metadata:
      variant_bitrate : 1958000
      comment         : Master Audio
    Stream #0:8: Data: timed_id3 (ID3  / 0x20334449)
    Metadata:
      variant_bitrate : 1958000
  Program 3
    Metadata:
      variant_bitrate : 2658000
    Stream #0:9: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p(progressive), 1280x720, Closed Captions, 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc
    Metadata:
      variant_bitrate : 2658000
    Stream #0:10(eng): Audio: aac (LC) ([15][0][0][0] / 0x000F), 44100 Hz, stereo, fltp, 158 kb/s (default)
    Metadata:
      variant_bitrate : 2658000
      comment         : Master Audio
    Stream #0:11: Data: timed_id3 (ID3  / 0x20334449)
    Metadata:
      variant_bitrate : 2658000
Unsupported codec with id 100358 for input stream 2
Unsupported codec with id 100358 for input stream 5
Unsupported codec with id 100358 for input stream 8
Unsupported codec with id 100358 for input stream 11


  



  As you can see, closed captions are embedded in each of the VBR streams, but are getting stripped during the transcode to SDI. How can I pass these to view them ?

  


• AVFrame with NV12 format breaks using D3D11 HW acceleration

  5 décembre 2024, par faith0058

  I have a ffmpeg-autogen video decoder, im trying to use a D3D11 as HW accelerator, but the frame i get in result(with NV12 format) is broken. If im doing the same with Vulkan as accelerator, everything works fine, but vulkan uses more graphic card resources..

  


  For rendering i use OpenTK, received frames i convert to RGB with textures and shaders.

  


  Below my decoder initialization.

  


  public VideoStreamDecoder(AVCodecParameters* parameters, AVCodec* codec)
{
    AVCodec* _codec = codec;
    
    _pCodecContext = ffmpeg.avcodec_alloc_context3(_codec);

    ffmpeg.av_hwdevice_ctx_create(&_pCodecContext->hw_device_ctx, AVHWDeviceType.AV_HWDEVICE_TYPE_D3D11VA, null, null, 0).ThrowExceptionIfError();

    ffmpeg.avcodec_parameters_to_context(_pCodecContext, parameters).ThrowExceptionIfError();
    ffmpeg.avcodec_open2(_pCodecContext, _codec, null).ThrowExceptionIfError();

    CodecName = ffmpeg.avcodec_get_name(_codec->id);
    FrameSize = new Size(_pCodecContext->width, _pCodecContext->height);
    PixelFormat = _pCodecContext->pix_fmt;

    _pFrame = ffmpeg.av_frame_alloc();
    _receivedFrame = ffmpeg.av_frame_alloc();
}


  


  And frames reading function

  


  public bool TryReadNextFrame(out AVFrame frame, AVPacket packet)
{
    int error;

    do
    {
        ffmpeg.avcodec_send_packet(_pCodecContext, &packet).ThrowExceptionIfError();

        error = ffmpeg.avcodec_receive_frame(_pCodecContext, _pFrame);

    } while (error == ffmpeg.AVERROR(ffmpeg.EAGAIN));

    error.ThrowExceptionIfError();

    ffmpeg.av_hwframe_transfer_data(_receivedFrame, _pFrame, 0);
    //here i get NV12 frame

    var clonedFrame = ffmpeg.av_frame_clone(_receivedFrame);

    frame = *clonedFrame;

    return true;
}


  


  And here is result i get with both accelerators :

  


  Vulkan
normally working

  


  The same camera, but D3D11
broken

  


  One more intresting moment, is that cameras, that dont work have yuv420p format, the only one working with D3D11 has yuvj420.

  


  I tried changing format using sws_scale from NV12 to NV12, and it worked, but it uses too much CPU.
Also tried changin format to rgb before av_hwframe_transfer_data function and rendering it using Vulkan rendering, it sort of works with Vulkan, but with D3D11 it doesnt.

  


  UPDATE

  


  I noticed, that frame linesize was bigger, than width and tried

  


  _receivedFrame->linesize[0] = _receivedFrame->width;


  


  And it helped, now image looks better, but still not perfect.

  


  UPDATE 2

  


  I triead also
_receivedFrame->linesize[1] = _receivedFrame->width; and now everything works just as it should, took me whole day to write two lines of code :)

  


• FFMPEG stereo track stops capturing at random times during a capture session

  26 mai 2022, par mrwassen

  I am currently working on building a workflow to capture and archive a large stash of family and friends PAL and NTSC VHS tapes. The hardware setup is as follows :

  


  

  • JVC HR-7860S VCR

  


  • s-video / RCA audio >

  


  • ADVC-3000 converter

  


  • SDI / BNC cable >

  


  • Blackmagic Decklink Mini Recorder 4K PCIe card

  


  • installed in a fairly hi-spec windows machine : AMD Ryzen 9 5900X 3.7 Ghz base 12 core, GEFORCE RTX 3060 12 gB, 32 gB ram

  


  


  The plan is to capture to lossless AVI, then drop into an NLE (Vegas Pro v.16) to do a minimal amount of cleanup / trimming, then render to a more compressed video format (TBD) for upload to AWS S3 accessible through a family website.

  


  The issue I am having is that when I run the capture using ffmpeg/directshow e.g. for a perfectly fine 90 min. PAL tape, at some random point of time during the capture one of the 2 stereo channels just stops capturing. This has happened with all of the tapes I have tested so far, and it happens at different times during the same video. I have examined the frames surrounding points in time when this happens, and it doesn't correlate to any transitions or jitter, but often just randomly in the middle of a perfectly smooth scene. Once the one channel stops capturing it never starts back up again during that capture session.

  


  The ADVC-3000 and the VCR are both showing both stereo channels playing normally throughout the capture. The windows machine running the capture hardly breaks a sweat at any time, and the transfer easily keeps up constantly showing a speed = 1x which I assume means nothing lagging. Also there are no video/audio sync issues at any point in time even towards the end of long tapes e.g. 90 mins.

  


  I am fairly new at ffmpeg, so I have spent extensive amounts of time reading up on forum posts and experimenting and have ended up with the following syntax :

  


  ffmpeg -y -f dshow -rtbufsize 2000M -i video="Blackmagic WDM Capture":audio="Blackmagic WDM Capture" -codec:v v210 -pix_fmt yuv422p -codec:a pcm_s16le -b:a 128k -t 02:00:00 -r 25 -threads 4 -maxrate 2500k -filter:a "volume=1.5" output_v210_audio.avi


  


  The capture runs without a single dropped frame, the only error I am getting when launching (and perhaps this is a smoking gun ?) is :

  


  


  "Non-monotonous DTS in output stream 0:1 ; previous : 0, current : -30 ;
changing to 1. This may result in incorrect timestamps in the output
file."

  


  


  I have tried to troubleshoot this in the hopes that it is tied to my issue but so far without luck.

  


  Hoping somebody can help correct or modify my command line or perhaps other ideas to help resolve the issue.