Recherche avancée

Sur d’autres sites (10772)

• Documented ffmpeg commands not recognized by ffmpeg

  2 avril 2020, par agconti

  I'm trying to use options like, ldash and http_opts, as the dash muxer docs describe but FFmpeg doesn't recognize them. I'm on the latest released version of ffmpeg, v4.2.2. I see the changes in the ffmpeg master branch but not in the v4.2 release branch. Does ffmpeg not recognize them because they haven't been released yet ?

  



  Here's the dash muxer docs for reference : https://ffmpeg.org/ffmpeg-all.html#dash-2

  



  Here's a minimal example command with uncut output :

  



  Andrews-MacBook-Pro :: dev/test ‹master› » ffmpeg -re -i test.mp4 \                                   
-map 0 -map 0 -c:a libfdk_aac -c:v libx264 \
-b:v:0 800k -b:v:1 300k -s:v:1 320x170 -profile:v:1 baseline \
-profile:v:0 main -bf 1 \
-b_strategy 0 -ar:a:1 22050 \
-adaptation_sets "id=0,streams=v id=1,streams=a" \
-ldash 1 \
-f dash ./output/out.mpd

ffmpeg version 4.2.2 Copyright (c) 2000-2019 the FFmpeg developers
  built with Apple clang version 11.0.0 (clang-1100.0.33.17)
  configuration: --prefix=/usr/local/Cellar/ffmpeg/4.2.2_2 --enable-shared --enable-pthreads --enable-version3 --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libbluray --enable-libmp3lame --enable-libopus --enable-librubberband --enable-libsnappy --enable-libtesseract --enable-libtheora --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxvid --enable-lzma --enable-libfontconfig --enable-libfreetype --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librtmp --enable-libspeex --enable-libsoxr --enable-videotoolbox --disable-libjack --disable-indev=jack
  libavutil      56. 31.100 / 56. 31.100
  libavcodec     58. 54.100 / 58. 54.100
  libavformat    58. 29.100 / 58. 29.100
  libavdevice    58.  8.100 / 58.  8.100
  libavfilter     7. 57.100 /  7. 57.100
  libavresample   4.  0.  0 /  4.  0.  0
  libswscale      5.  5.100 /  5.  5.100
  libswresample   3.  5.100 /  3.  5.100
  libpostproc    55.  5.100 / 55.  5.100
Unrecognized option 'ldash'.
Error splitting the argument list: Option not found


  


• Efficient real-time video stream processing and forwarding with RTMP servers

  19 mai 2023, par dumbQuestions

  I have a scenario where I need to retrieve a video stream from an RTMP server, apply image processing (specifically, adding blur to frames), and then forward the processed stream to another RTMP server (in this case, Twitch).

  


  Currently, I'm using ffmpeg in conjunction with cv2 to retrieve and process the stream. However, this approach introduces significant lag when applying the blur. I'm seeking an alternative method that can achieve the desired result more efficiently. I did attempt to solely rely on ffmpeg for the entire process, but I couldn't find a way to selectively process frames based on a given condition and subsequently transmit only those processed frames.

  


  Is there a more efficient approach or alternative solution that can address this issue and enable real-time video stream processing with minimal lag ?

  


  Thanks in advance !

  


  def forward_stream(server_url, stream_key, twitch_stream_key):
    get_ffmpeg_command = [...]

    send_ffmpeg_command [...]

    # Start get FFmpeg process
    read_process = subprocess.Popen(get_ffmpeg_command, stdout=subprocess.PIPE, stderr=subprocess.DEVNULL)

    # Start send FFmpeg process
    send_process = send_process = subprocess.Popen(send_ffmpeg_command, stdin=subprocess.PIPE, stderr=subprocess.DEVNULL)

    # Open video capture
    cap = cv2.VideoCapture(f'{server_url}')

    while True:
        # Read the frame
        ret, frame = cap.read()
        if ret:
            # Apply machine learning algorithm
            should_blur = machine_learning_algorithm(frame)

            # Apply blur if necessary
            if machine_learning_algorithm(frame):
                frame = cv2.blur(frame, (25, 25))

            # Write the frame to FFmpeg process
            send_process.stdin.write(frame.tobytes())
        else:
            break

    # Release resources
    cap.release()
    read_process.stdin.close()
    read_process.wait()



  


• Transcoding WAV audio to AAC in an MP4 container using FFmpeg C API

  7 novembre 2022, par vstrom coder

  I'm trying to compose source video and audio into a final MP4 video.
I have a problem with the WAV audio. After decoding and filtering, I'm getting an error from the output encoder : [aac @ 0x145e04c40] more samples than frame size

  


  I initially used the following filter graph (minimal reproducible example) :

  


      abuffer -> aformat -> abuffersink


  


  At this point I was getting the error mentioned above.

  


  Then, I tried to insert a aresample filter to the graph :

  


      abuffer -> aresample -> aformat -> abuffersink


  


  But still getting the same error.
This was based on the fact that the ffmpeg CLI uses this filter when converting WAV to MP4 :

  


  Command :

  


      ffmpeg -i source.wav output.mp4 -loglevel debug


  


  Output contains :

  


      [graph_0_in_0_0 @ 0x138f06200] Setting 'time_base' to value '1/44100'
    [graph_0_in_0_0 @ 0x138f06200] Setting 'sample_rate' to value '44100'
    [graph_0_in_0_0 @ 0x138f06200] Setting 'sample_fmt' to value 's16'
    [graph_0_in_0_0 @ 0x138f06200] Setting 'channel_layout' to value 'mono'
    [graph_0_in_0_0 @ 0x138f06200] tb:1/44100 samplefmt:s16 samplerate:44100 chlayout:mono
    [format_out_0_0 @ 0x138f06620] Setting 'sample_fmts' to value 'fltp'
    [format_out_0_0 @ 0x138f06620] Setting 'sample_rates' to value '96000|88200|64000|48000|44100|32000|24000|22050|16000|12000|11025|8000|7350'
    [format_out_0_0 @ 0x138f06620] auto-inserting filter 'auto_aresample_0' between the filter 'Parsed_anull_0' and the filter 'format_out_0_0'
    [AVFilterGraph @ 0x138f060f0] query_formats: 4 queried, 6 merged, 3 already done, 0 delayed
    [auto_aresample_0 @ 0x138f06c30] [SWR @ 0x120098000] Using s16p internally between filters
    [auto_aresample_0 @ 0x138f06c30] ch:1 chl:mono fmt:s16 r:44100Hz -> ch:1 chl:mono fmt:fltp r:44100Hz
    Output #0, mp4, to 'output.mp4':
      Metadata:
        encoder         : Lavf59.27.100
      Stream #0:0, 0, 1/44100: Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, mono, fltp, delay 1024, 69 kb/s
        Metadata:
          encoder         : Lavc59.37.100 aac


  


  I'm trying to figure out whether I should use the SWR library directly as exemplified in the transcode_aac example.