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• Impossible to convert between the formats supported by the filter '...' - Error reinitializing filters

  14 novembre 2023, par Fabien Biller

  I am using this ffmpeg command(values removed for simplicity)

  


  ffmpeg -hwaccel cuvid -c:v h264_cuvid -y -ss 1 -i "FILE0001.MOV" -ss 0 -i "GOPR0621.MP4" -filter_complex 
[0:v][1:v]
  midequalizer
[al];
[al]
  yadif
  lenscorrection
  scale
[vl];
[1:v]
  lenscorrection
  scale
[vr];
[vl][vr]
  hstack=shortest=1 
-an -c:v h264_nvenc -preset slow "output.mp4"


  


  on a machine with a cuda graphics card.

  


  I get

  


  ffmpeg version N-90979-g08032331ac Copyright (c) 2000-2018 the FFmpeg developers
  built with gcc 7.3.0 (GCC)
  configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-bzlib --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-amf --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth
  libavutil      56. 18.100 / 56. 18.100
  libavcodec     58. 19.100 / 58. 19.100
  libavformat    58. 13.101 / 58. 13.101
  libavdevice    58.  4.100 / 58.  4.100
  libavfilter     7. 21.100 /  7. 21.100
  libswscale      5.  2.100 /  5.  2.100
  libswresample   3.  2.100 /  3.  2.100
  libpostproc    55.  2.100 / 55.  2.100
[mov,mp4,m4a,3gp,3g2,mj2 @ 00000254a8afc0c0] st: 0 edit list: 1 Missing key frame while searching for timestamp: 6006
[mov,mp4,m4a,3gp,3g2,mj2 @ 00000254a8afc0c0] st: 0 edit list 1 Cannot find an index entry before timestamp: 6006.
....
Stream mapping:
  Stream #0:0 (h264_cuvid) -> midequalizer:in0
  Stream #1:0 (h264) -> midequalizer:in1
  Stream #1:0 (h264) -> lenscorrection
  hstack -> Stream #0:0 (h264_nvenc)
  
Impossible to convert between the formats supported by the filter 'graph 0 input from stream 0:0' and the filter 'auto_scaler_0'
Error reinitializing filters!


  


  The same command without CUDA works, ie

  


  ffmpeg -y -ss 1 -i "FILE0001.MOV" -ss 0 -i "GOPR0621.MP4" -filter_complex 
[0:v][1:v]
  midequalizer
[al];
[al]
  yadif
  lenscorrection
  scale
[vl];
[1:v]
  lenscorrection
  scale
[vr];
[vl][vr]
  hstack=shortest=1 
-an "output.mp4"


  


  How do I make it work on a Windows 10 machine with cuda ?

  


• ffmpeg configuration difficulty with filter_complex and hls

  4 février 2020, par akc42

  I am trying to set up ffmpeg so that it will record from a microphone and encode the results at the same time into a .flac file for later syncing up with some video I will be making.

  The microphone is plugged into a raspberry pi (4B) and I am currently trying it with a blue yeti mic, but I think I can do the same with a focusrite scarlett 2i2 plugged in instead. However I was puzzling about how to start the server recording and decided I could do it from a web browser if I made a simple nodejs server that spawned ffmpeg as a child process.

  But then I was inspired by this sample ffmpeg command which displays (on my desktop with an graphical interface) a volume meter

  ffmpeg -hide_banner -i 'http://distribution.bbb3d.renderfarming.net/video/mp4/bbb_sunflower_1080p_30fps_normal.mp4' -filter_complex "showvolume=rate=25:f=0.95:o=v:m=p:dm=3:h=80:w=480:ds=log:s=2" -c:v libx264 -c:a aac -f mpegts - | ffplay -window_title "Peak Volume" -i -

  What if I could stream the video produced by the showvolume filter to the web browser that I am using to control the ffmpeg process (NOTE I don’t want to send the audio with this). So I tried to read up on hls (since the control device will be an ipad - in fact that is what I will record the video on), and came up with this command

  ffmpeg -hide_banner -f alsa -ac 2 -ar 48k -i hw:CARD=Microphone -filter_complex "asplit=2[main][vol],[vol]showvolume=rate=25:f=0.95:o=v:m=p:dm=3:h=80:w=480:ds=log:s=2[vid]" -map [main] -c:a:0 flac recordings/session_$(date +%a_%d_%b_%Y___%H_%M_%S).flac -map [vid] -preset veryfast -g 25 -an -sc_threshold 0 -c:v:1 libx264 -b:v:1 2000k -maxrate:v:1 2200k -bufsize:v:3000k -f hls -hls_time 4 -hls_flags independent_segments delete_segments -strftime 1 -hls_segment_filename recordings/volume-%Y%m%d-%s.ts recordings/volume.m3u8

  The problem is I am finding the documentation a bit opaque as to what happens once I have generated two streams - the main audio and a video stream, and this command throws both a warning and an error :-

  The warning is Guessed Channel Layout for Input Stream #0.0 : stereo

  and the error is

  [NULL @ 0x1baa130] Unable to find a suitable output format for 'hls'
  
  hls: Invalid argument

  What I am trying to do is set up stream labels [main] and [vol] as I split the incoming audio into two parts, then I pass [vol] through the "showvolume" filter and end up with stream [vid].

  I think I need to then use -map to specify encoding the [main] stream down to flac and writing it out to file (The file exists after I run the command although they have zero length), and use another -map to pass through to the -f hls section. But I think I have something wrong at this stage.

  Can someone help me get this command right.

• How to encode 3840 nb_samples to a codec that asks for 1024 using ffmpeg

  26 juillet 2018, par Gabulit

  FFmpeg has an example muxing code on https://ffmpeg.org/doxygen/4.0/muxing_8c-example.html

  This code generates frame by frame video and audio. What I am trying to do is to change

  ost->tmp_frame = alloc_audio_frame(AV_SAMPLE_FMT_S16, c->channel_layout,
  
                                         c->sample_rate, nb_samples);

  to

  ost->tmp_frame = alloc_audio_frame(AV_SAMPLE_FMT_S16, c->channel_layout,
  
                                         c->sample_rate, 3840);

  so that it generates 3840 samples per channel instead of 1024 samples which is the default for nb_samples (aac codec).

  I tried to combine code from https://ffmpeg.org/doxygen/4.0/transcode_aac_8c-example.html which has an example on buffering the frames.

  My resulting program crashes when generating audio samples after a couple of frames when assigning *q++ a new value at the first iteration :

  /* Prepare a 16 bit dummy audio frame of 'frame_size' samples and
  
   * 'nb_channels' channels. */
  
  static AVFrame *get_audio_frame(OutputStream *ost)
  
  {
  
      AVFrame *frame = ost->tmp_frame;
  
      int j, i, v;
  
      int16_t *q = (int16_t*)frame->data[0];
  
      /* check if we want to generate more frames */
  
      if (av_compare_ts(ost->next_pts, ost->enc->time_base,
  
                        STREAM_DURATION, (AVRational){ 1, 1 }) >= 0)
  
          return NULL;
  
      for (j = 0; j nb_samples; j++) {
  
          v = (int)(sin(ost->t) * 10000);
  
          for (i = 0; i < ost->enc->channels; i++)
  
              *q++ = v;
  
          ost->t     += ost->tincr;
  
          ost->tincr += ost->tincr2;
  
      }
  
      frame->pts = ost->next_pts;
  
      ost->next_pts  += frame->nb_samples;
  
      return frame;
  
  }

  Maybe I don’t get the logic behind encoding.

  Here is the full source that i’ve come up with :

  https://paste.ee/p/b07qf

  The reason i am trying to accomplish this task is that I have a capture card sdk that outputs 2 channel 16 bit raw pcm 48000Hz which has 3840 samples per channel and I am trying to encode its output to aac. So basically if I get the muxing example to work with 3840 nb_samples this will help me understand the concept.

  I have already looked at How to encode resampled PCM-audio to AAC using ffmpeg-API when input pcm samples count not equal 1024 but the example uses "encodeFrame", which the examples on ffmpeg documentation doesn’t use or I am mistaken.

  Any help is greatly appreciated.