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• Problems using Intel Quick Sync H.264 Encoder with FFMPEG (shared) [closed]

  28 mai 2024, par Michael Werner
  

  • OS : Windows 11 Pro

  


  • IDE : Visual Studio 2022

  


  • PL : C++

  


  • Libraries : FFMPEG shared (own x64 Windows build with libmfx)

  


  


  I am trying to encode frames from a framegrabber card with Intel Quick Sync Encoder using libmfx in FFMPEG shared / libav but I am getting an error always. The "same" procedure using ffmpeg.exe of same build works fine.

  


  I increased the FFMPEG logging level to the maximum level. The avcodec_send_frame function always returns -22 (which means "Invalid argument"). And I do not get more information from the log output /console. The only thing the encoder is reporting is :

  


  [h264_mp4toannexb @ 0000028e41f42e00] The input looks like it is Annex B already

[h264_qsv @ 0000028e41f46980] Invalid pkt_timebase, passing timestamps as-is.


  


  The frames from the grabber card are 1920x1080 pixel YUV420P. I send them as they are to the encoder setting following FFMPEG video codec context options :

  


  

  • bit_rate : 5000000

  


  • width : 1920

  


  • height : 1080

  


  • framerate : 50, 1

  


  • max_b_frames : 3

  


  • pix_fmt : AV_PIX_FMT_YUV420P

  


  


  Actually I do not set any private encoder options.

  


  Does anyone have an idea what I am doing wrong or what is missing ? Is there a way to increase the debug output of libmfx ? I do not find anything related in the encoders private options.

  


• DirectShow Capture Source and FFMPEG

  27 janvier 2015, par Juan Ayala

  I have an AJA Capture card. The drivers installed with the card include some DirectShow filter. If I pop the filter into GraphEdit I see this :

  

  and if I run the ffmpeg command

  ffmpeg -f dshow -list_options true -i video="AJA Capture Source"

  I see

  [dshow @ 0034eec0] DirectShow video device options
  
  [dshow @ 0034eec0]  Pin "Video"
  
  [dshow @ 0034eec0]   pixel_format=yuyv422  min s=720x486 fps=27.2604 max s=1024x
  
  486 fps=29.985
  
  ...
  
  [dshow @ 0034eec0]  Pin "Audio 1-2"
  
  [dshow @ 0034eec0]  Pin "Line21"
  
  video=AJA Capture Source: Immediate exit requested

  So I see the Video and Audio pins I need. But when I try to run an ffmpeg command to capture both, I can only figure out how to do the video part. How do I hook in to that audio pin ? It seems all the examples and documentation point to using a separate audio device, and nothing about hooking into the pins. I’m running it out of a batch file for now like this and I use the ^ to break the line

  ffmpeg.exe ^
  
  -y ^
  
  -rtbufsize 100M ^
  
  -f dshow ^
  
  -i video="AJA Capture Source" ^
  
  -t 00:00:10 ^
  
  -aspect 16:9 ^
  
  -c:v libx264 ^
  
  "C:\VCS_AUD_SAMPLE.mp4"

  Again, the command above will get me some beautiful video, but I can’t figure out the audio part. Is this even supported in ffmpeg or am I going to have to modify the ffmpeg dshow code ?

• ffmpeg not honoring sample rate in opus output

  28 août 2016, par Adam

  I am capturing a live audio stream to Opus, and no matter what I choose for the audio sample rate, I get 48khz output.

  This is my command line

  ./ffmpeg -f alsa -ar 16000 -i sysdefault:CARD=CODEC -f
  alsa -ar 16000 -i sysdefault:CARD=CODEC_1 -filter_complex
  join=inputs=2:channel_layout=stereo:map=0.1-FR|1.0-FL,asetpts=expr=N/SR/TB
  -ar 16000 -ab 64k -c:a opus -vbr off -compression_level 5 output.ogg

  And this is what ffmpeg responds with :

  Output #0, ogg, to ’output.ogg’ : Metadata :
  encoder : Lavf57.48.100
  Stream #0:0 : Audio : opus (libopus), 16000 Hz, stereo, s16, delay 104, padding 0, 64 kb/s (default)
  Metadata :
  encoder : Lavc57.54.100 libopus

  However, it appears that ffmpeg has lied, because when analysing the file again, I get :

  Input #0, ogg, from ’output.ogg’ : Duration : 00:00:03.21, start :
  0.000000, bitrate : 89 kb/s
  Stream #0:0 : Audio : opus, 48000 Hz, stereo, s16, delay 156, padding 0
  Metadata :
  ENCODER : Lavc57.54.100 libopus

  I have tried so many permutations of sample rate, simplifying down to a single audio input etc etc - always with the same result.

  Any ideas ?