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• ffmpeg concatenation with -filter_complex

  16 octobre 2018, par Igniter

  I’ve seen several similar questions but none of them actually helped in my case.
  Getting this error while trying to join 1 audio and 4 video files of different nature and resolutions.

  ffmpeg -i 0.mp3 -i 1.mp4 -i 2.mkv -i 3.mkv -i 4.webm \
  
      -filter_complex [0:a:0][1:v:0][2:v:0][3:v:0][4:v:0]concat=n=5:v=1:a=1[outv][outa] \
  
      -map "[outv]" -map "[outa]" output.mp4

  All this gives the following error :

  Stream specifier ':a:0' in filtergraph description [0:a:0][1:v:0][2:v:0][3:v:0][4:v:0]concat=n=5:v=1:a=1[outv][outa] matches no streams.

  Straight concatenation -i "concat:0.mp3|1.mp4..." also doesn’t work as expected due to different resolutions and video formats. All methods syntax was taken from official documentation but there should be something that I’ve missed here.

  Full output log :

  ffmpeg version 3.4.4-0ubuntu0.18.04.1 Copyright (c) 2000-2018 the FFmpeg developers
  
    built with gcc 7 (Ubuntu 7.3.0-16ubuntu3)
  
    configuration: --prefix=/usr --extra-version=0ubuntu0.18.04.1 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-librsvg --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
  
    libavutil      55. 78.100 / 55. 78.100
  
    libavcodec     57.107.100 / 57.107.100
  
    libavformat    57. 83.100 / 57. 83.100
  
    libavdevice    57. 10.100 / 57. 10.100
  
    libavfilter     6.107.100 /  6.107.100
  
    libavresample   3.  7.  0 /  3.  7.  0
  
    libswscale      4.  8.100 /  4.  8.100
  
    libswresample   2.  9.100 /  2.  9.100
  
    libpostproc    54.  7.100 / 54.  7.100
  
  Input #0, mp3, from 'mp3/10.mp3':
  
    Metadata:
  
      album_artist    : artist
  
      title           : title
  
      artist          : 10
  
      album           : 12
  
      track           : 1
  
      VideoKind       : 2
  
      date            : 2009
  
    Duration: 00:06:00.44, start: 0.025056, bitrate: 64 kb/s
  
      Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16p, 64 kb/s
  
      Metadata:
  
        encoder         : LAME3.98r
  
      Stream #0:1: Video: mjpeg, yuvj420p(pc, bt470bg/unknown/unknown), 200x200 [SAR 72:72 DAR 1:1], 90k tbr, 90k tbn, 90k tbc
  
      Metadata:
  
        comment         : Cover (front)
  
  Input #1, matroska,webm, from '1.mp4':
  
    Metadata:
  
      MINOR_VERSION   : 0
  
      COMPATIBLE_BRANDS: iso6avc1mp41
  
      MAJOR_BRAND     : dash
  
      ENCODER         : Lavf57.83.100
  
    Duration: 00:01:53.05, start: 0.007000, bitrate: 2292 kb/s
  
      Stream #1:0: Video: h264 (High), yuv420p(tv, bt709, progressive), 1920x1080 [SAR 1:1 DAR 16:9], 24 fps, 24 tbr, 1k tbn, 48 tbc (default)
  
      Metadata:
  
        HANDLER_NAME    : VideoHandler
  
        DURATION        : 00:01:53.048000000
  
  Input #2, matroska,webm, from '2.mkv':
  
    Metadata:
  
      MINOR_VERSION   : 0
  
      COMPATIBLE_BRANDS: iso6avc1mp41
  
      MAJOR_BRAND     : dash
  
      ENCODER         : Lavf57.83.100
  
    Duration: 00:02:08.09, start: 0.007000, bitrate: 1607 kb/s
  
      Stream #2:0: Video: h264 (High), yuv420p(tv, bt709, progressive), 1920x1080 [SAR 1:1 DAR 16:9], 24 fps, 24 tbr, 1k tbn, 48 tbc (default)
  
      Metadata:
  
        HANDLER_NAME    : VideoHandler
  
        DURATION        : 00:02:08.090000000
  
  Input #3, matroska,webm, from '3.mkv':
  
    Metadata:
  
      MINOR_VERSION   : 0
  
      COMPATIBLE_BRANDS: iso6avc1mp41
  
      MAJOR_BRAND     : dash
  
      ENCODER         : Lavf57.83.100
  
    Duration: 00:01:37.05, start: 0.007000, bitrate: 3525 kb/s
  
      Stream #3:0: Video: h264 (High), yuv420p(tv, bt709, progressive), 1920x1080 [SAR 1:1 DAR 16:9], 24 fps, 24 tbr, 1k tbn, 48 tbc (default)
  
      Metadata:
  
        HANDLER_NAME    : VideoHandler
  
        DURATION        : 00:01:37.048000000
  
  Input #4, matroska,webm, from '4.webm':
  
    Metadata:
  
      MINOR_VERSION   : 0
  
      COMPATIBLE_BRANDS: iso6avc1mp41
  
      MAJOR_BRAND     : dash
  
      ENCODER         : Lavf57.83.100
  
    Duration: 00:01:45.13, start: 0.007000, bitrate: 3685 kb/s
  
      Stream #4:0: Video: h264 (High), yuv420p(tv, bt709, progressive), 1920x1080 [SAR 1:1 DAR 16:9], 24 fps, 24 tbr, 1k tbn, 48 tbc (default)
  
      Metadata:
  
        HANDLER_NAME    : VideoHandler
  
        DURATION        : 00:01:45.131000000
  
  Stream specifier ':a:0' in filtergraph description [0:a:0][1:v:0][2:v:0][3:v:0][4:v:0]concat=n=5:v=1:a=1[outv][outa] matches no streams.

• Problems with Python's azure.cognitiveservices.speech when installing together with FFmpeg in a Linux web app

  15 mai 2024, par Kakobo kakobo

  I need some help.
I'm building an web app that takes any audio format, converts into a .wav file and then passes it to 'azure.cognitiveservices.speech' for transcription.I'm building the web app via a container Dockerfile as I need to install ffmpeg to be able to convert non ".wav" audio files to ".wav" (as azure speech services only process wav files). For some odd reason, the 'speechsdk' class of 'azure.cognitiveservices.speech' fails to work when I install ffmpeg in the web app. The class works perfectly fine when I install it without ffpmeg or when i build and run the container in my machine.

  


  I have placed debug print statements in the code. I can see the class initiating, for some reason it does not buffer in the same when when running it locally in my machine. The routine simply stops without any reason.

  


  Has anybody experienced a similar issue with azure.cognitiveservices.speech conflicting with ffmpeg ?

  


  Here's my Dockerfile :

  


  # Use an official Python runtime as a parent imageFROM python:3.11-slim

#Version RunRUN echo "Version Run 1..."

Install ffmpeg

RUN apt-get update && apt-get install -y ffmpeg && # Ensure ffmpeg is executablechmod a+rx /usr/bin/ffmpeg && # Clean up the apt cache by removing /var/lib/apt/lists saves spaceapt-get clean && rm -rf /var/lib/apt/lists/*

//Set the working directory in the container

WORKDIR /app

//Copy the current directory contents into the container at /app

COPY . /app

//Install any needed packages specified in requirements.txt

RUN pip install --no-cache-dir -r requirements.txt

//Make port 80 available to the world outside this container

EXPOSE 8000

//Define environment variable

ENV NAME World

//Run main.py when the container launches

CMD ["streamlit", "run", "main.py", "--server.port", "8000", "--server.address", "0.0.0.0"]`and here's my python code:


  


  def transcribe_audio_continuous_old(temp_dir, audio_file, language):
    speech_key = azure_speech_key
    service_region = azure_speech_region

    time.sleep(5)
    print(f"DEBUG TIME BEFORE speechconfig")

    ran = generate_random_string(length=5)
    temp_file = f"transcript_key_{ran}.txt"
    output_text_file = os.path.join(temp_dir, temp_file)
    speech_recognition_language = set_language_to_speech_code(language)
    
    speech_config = speechsdk.SpeechConfig(subscription=speech_key, region=service_region)
    speech_config.speech_recognition_language = speech_recognition_language
    audio_input = speechsdk.AudioConfig(filename=os.path.join(temp_dir, audio_file))
        
    speech_recognizer = speechsdk.SpeechRecognizer(speech_config=speech_config, audio_config=audio_input, language=speech_recognition_language)
    done = False
    transcript_contents = ""

    time.sleep(5)
    print(f"DEBUG TIME AFTER speechconfig")
    print(f"DEBUG FIle about to be passed {audio_file}")

    try:
        with open(output_text_file, "w", encoding=encoding) as file:
            def recognized_callback(evt):
                print("Start continuous recognition callback.")
                print(f"Recognized: {evt.result.text}")
                file.write(evt.result.text + "\n")
                nonlocal transcript_contents
                transcript_contents += evt.result.text + "\n"

            def stop_cb(evt):
                print("Stopping continuous recognition callback.")
                print(f"Event type: {evt}")
                speech_recognizer.stop_continuous_recognition()
                nonlocal done
                done = True
            
            def canceled_cb(evt):
                print(f"Recognition canceled: {evt.reason}")
                if evt.reason == speechsdk.CancellationReason.Error:
                    print(f"Cancellation error: {evt.error_details}")
                nonlocal done
                done = True

            speech_recognizer.recognized.connect(recognized_callback)
            speech_recognizer.session_stopped.connect(stop_cb)
            speech_recognizer.canceled.connect(canceled_cb)

            speech_recognizer.start_continuous_recognition()
            while not done:
                time.sleep(1)
                print("DEBUG LOOPING TRANSCRIPT")

    except Exception as e:
        print(f"An error occurred: {e}")

    print("DEBUG DONE TRANSCRIPT")

    return temp_file, transcript_contents


  


  The transcript this callback works fine locally, or when installed without ffmpeg in the linux web app. Not sure why it conflicts with ffmpeg when installed via container dockerfile. The code section that fails can me found on note #NOTE DEBUG"

  


• Mix additional audio file with video(+audio) in ffmpeg

  21 mai 2019, par Serg

  I’m trying to mix additional audio file with video which has also audio within. But the problem is that I already have complex ffmpeg command and don’t know how to combine them together.

  This is my existing ffmpeg command which uses some offsets and replaces additional audio file with embedded one (audio inside video) and also overlays few gauges and watermark to the video.

  ffmpeg -y 
  
  -ss 00:00:01:213 -i videoFile.mp4 
  
  -ss 00:00:03:435 -i audioFile.wav 
  
  -i watermark.png 
  
  -framerate 6 -i gauge1_path/img-%04d.png 
  
  -framerate 1 -i gauge2_path/img-%04d.png 
  
  -framerate 2 -i gauge3_path/img-%04d.png 
  
  -framerate 2 -i gauge4_path/img-%04d.png 
  
  -framerate 2 -i gauge5_path/img-%04d.png 
  
  -framerate 2 -i gauge6_path/img-%04d.png 
  
  -filter_complex [0][2]overlay=(21):(H-h-21)[ovr0];
  
  [ovr0][3]overlay=(W-w-21):(H-h-21)[ovr1];
  
  [ovr1][4]overlay=(W-w-21):(H-h-333)[ovr2];
  
  [ovr2][5]overlay=(W-w-21):(H-h-418)[ovr3];
  
  [ovr3][6]overlay=(W-w-21):(H-h-503)[ovr4];
  
  [ovr4][7]overlay=(W-w-21):(H-h-588)[ovr5];
  
  [ovr5][8]overlay=(W-w-21):(H-h-673)
  
  -map 0:v -map 1:a -c:v libx264 -preset ultrafast -crf 23 -t 00:5:10:000 output.mp4

  Now I would like to use ffmpeg's amix in order to mix both audios instead of replacing them, if possible with ability to set volumes. But official documentation amix says nothing about volume.

  Separately both seems to work ok.

  ffmpeg -y -i video.mp4 -i audio.mp3 -filter_complex [0][1]amix=inputs=2[a] -map 0:v -map [a] -c:v copy output.mp4

  and

  ffmpeg -y -i video.mp4 -i audio.mp3 -i watermark.png -filter_complex [0][2]overlay=(21):(H-h-21)[ovr0] -map [ovr0]:v -map 1:a -c:v libx264 -preset ultrafast -crf 23 output.mp4

  but together

  ffmpeg -y -i video.mp4 -i audio.mp3 -i watermark.png -filter_complex [0][1]amix=inputs=2[a];[a][2]overlay=(21):(H-h-21)[ovr0] -map [ovr0]:v -map [a] -c:v libx264 -preset ultrafast -crf 23 output.mp4

  I’m getting an error :

  ffmpeg version N-93886-gfbdb3aa179 Copyright (c) 2000-2019 the FFmpeg developers
  
    built with gcc 8.3.1 (GCC) 20190414
  
    configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --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 --enable-libopenmpt
  
    libavutil      56. 28.100 / 56. 28.100
  
    libavcodec     58. 52.101 / 58. 52.101
  
    libavformat    58. 27.103 / 58. 27.103
  
    libavdevice    58.  7.100 / 58.  7.100
  
    libavfilter     7. 53.101 /  7. 53.101
  
    libswscale      5.  4.101 /  5.  4.101
  
    libswresample   3.  4.100 /  3.  4.100
  
    libpostproc    55.  4.100 / 55.  4.100
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'video.mp4':
  
    Metadata:
  
      major_brand     : isom
  
      minor_version   : 512
  
      compatible_brands: isomiso2avc1mp41
  
      creation_time   : 1970-01-01T00:00:00.000000Z
  
      encoder         : Lavf53.24.2
  
    Duration: 00:00:29.57, start: 0.000000, bitrate: 1421 kb/s
  
      Stream #0:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 1032 kb/s, 25 fps, 25 tbr, 12800 tbn, 50 tbc (default)
  
      Metadata:
  
        creation_time   : 1970-01-01T00:00:00.000000Z
  
        handler_name    : VideoHandler
  
      Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, 5.1, fltp, 383 kb/s (default)
  
      Metadata:
  
        creation_time   : 1970-01-01T00:00:00.000000Z
  
        handler_name    : SoundHandler
  
  [mp3 @ 0000015e2f934ec0] Estimating duration from bitrate, this may be inaccurate
  
  Input #1, mp3, from 'audio.mp3':
  
    Duration: 00:00:45.33, start: 0.000000, bitrate: 128 kb/s
  
      Stream #1:0: Audio: mp3, 44100 Hz, stereo, fltp, 128 kb/s
  
  Input #2, png_pipe, from 'watermark.png':
  
    Duration: N/A, bitrate: N/A
  
      Stream #2:0: Video: png, rgb24(pc), 100x56 [SAR 3779:3779 DAR 25:14], 25 tbr, 25 tbn, 25 tbc
  
  [Parsed_amix_0 @ 0000015e2ff2e940] Media type mismatch between the 'Parsed_amix_0' filter output pad 0 (audio) and the 'Parsed_overlay_1' filter input pad 0 (video)
  
  [AVFilterGraph @ 0000015e2f91c600] Cannot create the link amix:0 -> overlay:0
  
  Error initializing complex filters.
  
  Invalid argument

  So my question : whether it’s possible to combine amix and overlay together and how and in which order they should be used ? Or should I look something different because amix unable to set volume levels ?

  Thanks in advance !