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• 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 !

• ffmpeg trouble with NetMaui and Android [closed]

  15 juin 2024, par Billy Vanegas

  I find myself in a following predicament. While I work with Visual Studio 2022, ffmpeg performs flawlessly on Windows platform. However, I'm encountering difficulties when attempting to replicate this functionality on Android. Despite exploring NuGet packages like FFMpegCore, which seem to be ffmpeg wrappers but lack ffmpeg itself, I'm still struggling to find a clear path forward. I've even tried integrating ffmpeg-kit for Android as per the instructions, only to face repeated failures and a sense of confusion. I must admit my puzzlement : why isn't there a straightforward method to seamlessly add ffmpeg to a .NET MAUI project that functions consistently across both iOS and Android platforms ?

  


  I want to convert MP3 files to WAV format using FFmpeg on the Android platform within a .NET MAUI project.

  


  I am using the FFmpegCore library and have downloaded the FFmpeg binaries from the official FFmpeg website. However, I encountered issues when attempting to pass the binary folder on the emulator, when I try to pass it, I get permission denied in working folder :
/data/user/0/com.companyname.projectname/files/ffmpeg on this part of code :

  


           await FFMpegArguments
               .FromFileInput(mp3Path)
               .OutputToFile(wavPath, true, options => options
                   .WithAudioCodec("pcm_s16le")
                   .WithAudioSamplingRate(44100)
                   .WithAudioBitrate(320000)
                   )
               .ProcessAsynchronously();


  


  This is the AndroidManifest.xml

  


  &lt;?xml version="1.0" encoding="utf-8"?>&#xA;<manifest>&#xA;    <application></application>&#xA;    &#xA;    &#xA;    &#xA;    &#xA;</manifest>&#xA;

  


   private async Task ConvertMp3ToWav(string mp3Path, string wavPath)
 {
     try
     {
         var directory = Path.GetDirectoryName(wavPath);
         if (!Directory.Exists(directory))
             Directory.CreateDirectory(directory!);
         if (!File.Exists(wavPath))
             Console.WriteLine($"File not found {wavPath}, creating empty file.");
             using var fs = new FileStream(wavPath, FileMode.CreateNew);
         if (!File.Exists(mp3Path))
             Console.WriteLine($"File not found {mp3Path}");

         string? ffmpegBinaryPath = await ExtractFFmpegBinaries(Platform.AppContext);
         FFMpegCore.GlobalFFOptions.Configure(new FFOptions { BinaryFolder = Path.GetDirectoryName(ffmpegBinaryPath!)! });

         await FFMpegArguments
               .FromFileInput(mp3Path)
               .OutputToFile(wavPath, true, options => options
                   .WithAudioCodec("pcm_s16le")
                   .WithAudioSamplingRate(44100)
                   .WithAudioBitrate(320000)
                   )
               .ProcessAsynchronously();
     }
     catch (Exception ex)
     {
         Console.WriteLine($"An error occurred during the conversion process: {ex.Message}");
         throw;
     }
 }


  


      private async Task<string> ExtractFFmpegBinaries(Context context)&#xA;    {&#xA;        var architectureFolder = "x86"; // "armeabi-v7a";&#xA;        var ffmpegBinaryName = "ffmpeg"; &#xA;        var ffmpegBinaryPath = Path.Combine(context.FilesDir!.AbsolutePath, ffmpegBinaryName);&#xA;        var tempFFMpegFileName = Path.Combine(FileSystem.AppDataDirectory, ffmpegBinaryName);&#xA;&#xA;        if (!File.Exists(ffmpegBinaryPath))&#xA;        {&#xA;            try&#xA;            {&#xA;                var assetPath = $"Libs/{architectureFolder}/{ffmpegBinaryName}";&#xA;                using var assetStream = context.Assets!.Open(assetPath);&#xA;               &#xA;                await using var tempFFMpegFile = File.OpenWrite(tempFFMpegFileName);&#xA;                await assetStream.CopyToAsync(tempFFMpegFile);&#xA;&#xA;                //new MainActivity().RequestStoragePermission();&#xA;                Java.Lang.Runtime.GetRuntime()!.Exec($"chmod 755 {tempFFMpegFileName}");&#xA;            }&#xA;            catch (Exception ex)&#xA;            {&#xA;                Console.WriteLine($"An error occurred while extracting FFmpeg binaries: {ex.Message}");&#xA;                throw;&#xA;            }&#xA;        }&#xA;        else&#xA;        {&#xA;            Console.WriteLine($"FFmpeg binaries already extracted to: {ffmpegBinaryPath}");&#xA;        }&#xA;&#xA;        return tempFFMpegFileName!;&#xA;    }&#xA;</string>

  


• 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"