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• Rotating a video during encoding with ffmpeg and libav API results in half of video corrupted

  11 mai 2020, par Daniel Kobe

  I'm using the C API for ffmpeg/libav to rotate a vertically filmed iphone video during the encoding step. There are other questions asking to do a similar thing but they are all using the CLI tool to do so.

  



  So far I was able to figure out how to use the AVFilter to rotate the video, base off this example https://github.com/FFmpeg/FFmpeg/blob/master/doc/examples/filtering_video.c

  



  The problem is that half the output file is corrupt.


  



  Here is the code for my encoding logic. Its written with GOLANG using CGO to interface with the C API.

  



  // Encode encode an AVFrame and return it
func Encode(enc Encoder, frame *C.AVFrame) (*EncodedFrame, error) {
    ctx := enc.Context()

    if ctx.buffersrcctx == nil {
        // initialize filter
        outputs := C.avfilter_inout_alloc()
        inputs  := C.avfilter_inout_alloc()
        m_pFilterGraph := C.avfilter_graph_alloc()
        buffersrc := C.avfilter_get_by_name(C.CString("buffer"))
        argsStr := fmt.Sprintf("video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:pixel_aspect=%d/%d", ctx.avctx.width, ctx.avctx.height, ctx.avctx.pix_fmt, ctx.avctx.time_base.num, ctx.avctx.time_base.den, ctx.avctx.sample_aspect_ratio.num, ctx.avctx.sample_aspect_ratio.den)
        Log.Info.Println("yakotest")
        Log.Info.Println(argsStr)
        args := C.CString(argsStr)
        ret := C.avfilter_graph_create_filter(&ctx.buffersrcctx, buffersrc, C.CString("my_buffersrc"), args, nil, m_pFilterGraph)
        if ret < 0 {
            Log.Info.Printf("\n problem creating filter %v\n", AVError(ret).Error())
        }

        buffersink := C.avfilter_get_by_name(C.CString("buffersink"))
        ret = C.avfilter_graph_create_filter(&ctx.buffersinkctx, buffersink, C.CString("my_buffersink"), nil, nil, m_pFilterGraph)
        if ret < 0 {
            Log.Info.Printf("\n problem creating filter %v\n", AVError(ret).Error())
        }

        /*
         * Set the endpoints for the filter graph. The filter_graph will
         * be linked to the graph described by filters_descr.
         */

        /*
         * The buffer source output must be connected to the input pad of
         * the first filter described by filters_descr; since the first
         * filter input label is not specified, it is set to "in" by
         * default.
         */
        outputs.name       = C.av_strdup(C.CString("in"))
        outputs.filter_ctx = ctx.buffersrcctx
        outputs.pad_idx    = 0
        outputs.next       = nil

        /*
         * The buffer sink input must be connected to the output pad of
         * the last filter described by filters_descr; since the last
         * filter output label is not specified, it is set to "out" by
         * default.
         */
        inputs.name       = C.av_strdup(C.CString("out"))
        inputs.filter_ctx = ctx.buffersinkctx
        inputs.pad_idx    = 0
        inputs.next       = nil

        ret = C.avfilter_graph_parse_ptr(m_pFilterGraph, C.CString("transpose=clock,scale=-2:1080"),
            &inputs, &outputs, nil)
        if ret < 0 {
            Log.Info.Printf("\n problem with avfilter_graph_parse %v\n", AVError(ret).Error())
        }

        ret = C.avfilter_graph_config(m_pFilterGraph, nil)
        if ret < 0 {
            Log.Info.Printf("\n problem with graph config %v\n", AVError(ret).Error())
        }
    }

    filteredFrame :=  C.av_frame_alloc()

    /* push the decoded frame into the filtergraph */
    ret := C.av_buffersrc_add_frame_flags(ctx.buffersrcctx, frame, C.AV_BUFFERSRC_FLAG_KEEP_REF)
    if ret < 0 {
        Log.Error.Printf("\nError while feeding the filter greaph, err = %v\n", AVError(ret).Error())
        return nil, errors.New(ErrorFFmpegCodecFailure)
    }

    /* pull filtered frames from the filtergraph */
    for {
        ret = C.av_buffersink_get_frame(ctx.buffersinkctx, filteredFrame)
        if ret == C.AVERROR_EAGAIN || ret == C.AVERROR_EOF {
            break
        }
        if ret < 0 {
            Log.Error.Printf("\nCouldnt find a frame, err = %v\n", AVError(ret).Error())
            return nil, errors.New(ErrorFFmpegCodecFailure)
        }

        filteredFrame.pts = frame.pts
        frame = filteredFrame
        defer C.av_frame_free(&filteredFrame)
    }

    if frame != nil {
        frame.pict_type = 0 // reset pict type for the encoder
        if C.avcodec_send_frame(ctx.avctx, frame) != 0 {
            Log.Error.Printf("%+v\n", StackErrorf("codec error, could not send frame"))
            return nil, errors.New(ErrorFFmpegCodecFailure)
        }
    }

    for {
        ret := C.avcodec_receive_packet(ctx.avctx, ctx.pkt)
        if ret == C.AVERROR_EAGAIN {
            break
        }
        if ret == C.AVERROR_EOF {
            return nil, fmt.Errorf("EOF")
        }
        if ret < 0 {
            Log.Error.Printf("%+v\n", StackErrorf("codec error, receiving packet"))
            return nil, errors.New(ErrorFFmpegCodecFailure)
        }

        data := C.GoBytes(unsafe.Pointer(ctx.pkt.data), ctx.pkt.size)
        return &EncodedFrame{data, int64(ctx.pkt.pts), int64(ctx.pkt.dts),
            (ctx.pkt.flags & C.AV_PKT_FLAG_KEY) != 0}, nil
    }

    return nil, nil
}


  



  It seems like I need to do something with the scaling here but I'm struggling to find helpful information online.

  


• Does PTS have to start at 0 ?

  5 juillet 2018, par stevendesu

  I’ve seen a number of questions regarding video PTS values not starting at zero, or asking how to make them start at zero. I’m aware that using ffmpeg I can do something like ffmpeg -i <video> -vf="setpts=PTS-STARTPTS" <output></output></video> to fix this kind of thing

  However it’s my understanding that PTS values don’t have to start at zero. For instance, if you join a live stream then odds are it has been going on for an hour and the PTS is already somewhere around 3600000+ but your video player faithfully displays everything just fine. Therefore I would expect there to be no problem if I intentionally created a video with a PTS value starting at, say, the current wall clock time.

  I want to send a live stream using ffmpeg, but embed the current time into the stream. This can be used both for latency calculation while the stream is live, and later to determine when the stream was originally aired. From my understanding of PTS, something as simple as this should probably work :

  ffmpeg -i video.flv -vf="setpts=RTCTIME" rtmp://<output>
  
  </output>

  When I try this, however, ffmpeg outputs the following :

  frame=   93 fps= 20 q=-1.0 Lsize=    9434kB time=535020:39:58.70 bitrate=   0.0kbits/s speed=1.35e+11x

  Note the extremely large value for "time", the bitrate (0.0kbits), and the speed (135000000000x !!!)

  At first I thought the issue might be my timebase, so I tried the following :

  ffmpeg -i video.flv -vf="settb=1/1K,setpts=RTCTIME/1K" rtmp://<output>
  
  </output>

  This puts everything in terms of milliseconds (1 PTS = 1 ms) but I had the same issue (massive time, zero bitrate, and massive speed)

  Am I misunderstanding something about PTS ? Is it not allowed to start at non-zero values ? Or am I just doing something wrong ?

  Update

  After reviewing @Gyan’s answer, I formatted my command like so :

  ffmpeg -re -i video.flv -vf="settb=1/1K, setpts=(RTCTIME-RTCSTART)/1K" -output_ts_offset $(date +%s.%N) rtmp://<output>
  
  </output>

  This way the PTS values would match up to "milliseconds since the stream started" and would be offset by the start time of the stream (theoretically making PTS = timestamp on the server)

  This looked like it was encoding better :

  frame=  590 fps=7.2 q=22.0 size=   25330kB time=00:01:21.71 bitrate=2539.5kbits/s dup=0 drop=1350 speed=   1x 

  Bitrate was now correct, time was accurate, and speed was not outrageous. The frames per second was still a bit off, though (the source video is 24 fps but it’s reporting 7.2 frames per second)

  When I tried watching the stream from the other end, the video was out of sync with the audio and played at about double normal speed for a while, then the video froze and the audio continued without it

  Furthermore, when I dumped the stream to a file (ffmpeg -i rtmp://<output> dump.mp4</output>) and look at the PTS timestamps with ffprobe (ffprobe -show_entries packet=codec_type,pts dump.mp4 | grep "video" -B 1 -A 2) the timestamps didn’t seem to show server time at all :

  ...
  
  --
  
  [PACKET]
  
  codec_type=video
  
  pts=131072
  
  [/PACKET]
  
  [PACKET]
  
  codec_type=video
  
  pts=130048
  
  [/PACKET]
  
  --
  
  [PACKET]
  
  codec_type=video
  
  pts=129536
  
  [/PACKET]
  
  [PACKET]
  
  codec_type=video
  
  pts=130560
  
  [/PACKET]
  
  --
  
  [PACKET]
  
  codec_type=video
  
  pts=131584
  
  [/PACKET]

  Is the problem just an incompatibility with RTMP ?

  Update 2

  I’ve removed the video filter and I’m now encoding like so :

  ffmpeg -re -i video.flv -output_ts_offset $(date +%s.%N) rtmp://<output>
  
  </output>

  This is encoding correctly :

  frame=  910 fps= 23 q=25.0 size=   12027kB time=00:00:38.97 bitrate=2528.2kbits/s speed=0.981x 

  In order to verify that the PTS values are correct, I’m dumping the output to a file like so :

  ffmpeg -i rtmp://<output> -copyts -write_tmcd 0 dump.mp4
  
  </output>

  I tried saving it as dump.flv (since it’s RTMP) however this threw the error :

  [flv @ 0x5600f24b4620] Audio codec mp3 not compatible with flv

  This is a bit weird since the video isn’t mp3-encoded (it’s speex) - but whatever.

  While dumping this file the following error pops up repeatedly :

  frame=    1 fps=0.0 q=0.0 size=       0kB time=00:00:09.21 bitrate=   0.0kbits/s dup=0 dr
  
  43090023 frame duplication too large, skipping
  
  43090027 frame duplication too large, skipping
  
      Last message repeated 3 times
  
  43090031 frame duplication too large, skipping
  
      Last message repeated 3 times
  
  43090035 frame duplication too large, skipping

  Playing the resulting video in VLC plays an audio stream but displays no video. I then attempt to probe this video with ffprobe to look at the video PTS values :

  ffprobe -show_entries packet=codec_type,pts dump.mp4 | grep "video" -B 1 -A 2

  This returns only a single video frame whose PTS is not large like I would expect :

  [PACKET]
  
  codec_type=video
  
  pts=1020
  
  [/PACKET]

  This has been a surprisingly difficult task

• ffmpeg issue with Jupyter notebook

  16 novembre 2022, par Nanda

  I am trying to extract frames from a video file using ffmpeg in Python. I installed ffmpeg using Homebrew and ffmpeg-python on the Anaconda-Navigator. Yet when I call ffmpeg on Jupyter notebook as follows

  


  !ffmpeg -i "$file" "$rootdir"/"$folder_name"/frame%04d.png


  


  I get an error saying

  


  zsh:1: command not found: ffmpeg


  


  I clearly see ffmpeg in my usr/local/bin. Can someone please assist me in sorting this ? I am able to use ffmpeg in Google Colab, though.