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• Error Installing mobile-ffmpeg-full-gpl (4.4.LTS) via CocoaPods - 404 Not Found

  6 mai, par Muhammad Accucia

  I am trying to install mobile-ffmpeg-full-gpl (4.4.LTS) in my iOS project using CocoaPods, but I am encountering a 404 error when CocoaPods attempts to download the framework.

  Here is the error message:

  \[!\] Error installing mobile-ffmpeg-full-gpl
  \[!\] /usr/bin/curl -f -L -o /var/folders/78/lk7swzb97ml4dt3zd9grny0c0000gn/T/d20250201-6580-dtli8h/file.zip https://github.com/tanersener/mobile-ffmpeg/releases/download/v4.4.LTS/mobile-ffmpeg-full-gpl-4.4.LTS-ios-framework.zip --create-dirs --netrc-optional --retry 2 -A 'CocoaPods/1.15.2 cocoapods-downloader/2.1'
  
  % Total    % Received % Xferd  Average Speed   Time    Time     Time  Current
  Dload  Upload   Total   Spent    Left  Speed
  0     9    0     0    0     0      0      0 --:--:-- --:--:-- --:--:--     0
  curl: (56) The requested URL returned error: 404
  

  It seems like the requested file is no longer available at the specified URL.

  Checked the official GitHub releases page for mobile-ffmpeg-full-gpl (4.4.LTS), but I couldn't find the exact file.

• FFMPEG File Output is Still in Use By a Process

  6 mai, par Tyler Bacon

  I am trying to complete this part of my program. In this section, I am trying to speed up or slow down a video based on a factor variable. Once it's done, I use moviepy to turn it into a VideoFileClip, then I delete the file.

      if factor <= 2:
          system("ffmpeg -i " + paths[dex] + " -vf setpts=" + str(vfactor) + "*PTS -an ./Media/Videos/temp.mp4")
          system("ffmpeg -i " + paths[dex] + " -filter:a atempo=" + str(factor) + " -vn ./Media/ShortSounds/temp.mp3")
      elif 2 < factor < 4:
          factor = round(sqrt(factor), 1)
          system("ffmpeg -i " + paths[dex] + " -vf setpts=" + str(vfactor) + "*PTS,setpts=" + str(vfactor) + "*PTS  -an ./Media/Videos/temp.mp4")
          system("ffmpeg -i " + paths[dex] + " -filter:a atempo=" + str(factor) + ",atempo=" + str(factor) + " -vn ./Media/ShortSounds/temp.mp3")
      elif factor > 4:
          raise Exception("File " + paths[dex] + " is too long.")
      t = VideoFileClip("./Media/Videos/temp.mp4")
      t.audio = AudioFileClip("./Media/Videos/temp.mp3")
      templist.append(t)
      remove("./Media/Videos/temp.mp4")
  

  However, when the code gets to the deletion command, it has the following error:

  PermissionError: [WinError 32] The process cannot access the file because it is being used by another process: './Media/Videos/temp.mp4'
  

  What's strange is, I can see the temp.mp4 file, and it runs just fine. I never get this error while manually running the temp.mp4 file.

  I have tried the following:

  • Waiting 5, 10, and 20 seconds before deleting the file.
  • Running "taskkill -f -im ffmpeg.exe" before deleting the file
  • I went through the debugger, and right before the deletion, I checked in task manager to see if ffmpeg was still running, and it wasn't.

  Do you guys have any idea what could be holding this up? My code worked previously when I was trying to just do audio, but I am trying it with video and this is happening.

• ffmpeg pipe process ends right after writing first buffer data to input stream and does not keep running

  6 mai, par Taketo Matsunaga

  I have been trying to convert 16bit PCM (s16le) audio data to webm using ffmpeg in C#. But the process ends right after the writing the first buffer data to standard input. I has exited with the status 0, meaning success. But do not know why.... Could anyone tell me why?

  I apprecite it if you could support me.

      public class SpeechService : ISpeechService
      {
          
          /// 
  
          /// Defines the _audioInputStream
          /// 
          private readonly MemoryStream _audioInputStream = new MemoryStream();
  
          public async Task SendPcmAsWebmViaWebSocketAsync(
              MemoryStream pcmAudioStream,
              int sampleRate,
              int channels) 
          {
              string inputFormat = "s16le";
  
              var ffmpegProcessInfo = new ProcessStartInfo
              {
                  FileName = _ffmpegPath,
                  Arguments =
                      $"-f {inputFormat} -ar {sampleRate} -ac {channels} -i pipe:0 " +
                      $"-f webm pipe:1",
                  RedirectStandardInput = true,
                  RedirectStandardOutput = true,
                  RedirectStandardError = true,
                  UseShellExecute = false,
                  CreateNoWindow = true,
              };
  
              _ffmpegProcess = new Process { StartInfo = ffmpegProcessInfo };
  
              Console.WriteLine("Starting FFmpeg process...");
              try
              {
  
                  if (!await Task.Run(() => _ffmpegProcess.Start()))
                  {
                      Console.Error.WriteLine("Failed to start FFmpeg process.");
                      return;
                  }
                  Console.WriteLine("FFmpeg process started.");
  
              }
              catch (Exception ex)
              {
                  Console.Error.WriteLine($"Error starting FFmpeg process: {ex.Message}");
                  throw;
              }
  
              var encodeAndSendTask = Task.Run(async () =>
              {
                  try
                  {
                      using var ffmpegOutputStream = _ffmpegProcess.StandardOutput.BaseStream;
                      byte[] buffer = new byte[8192]; // Temporary buffer to read data
                      byte[] sendBuffer = new byte[8192]; // Buffer to accumulate data for sending
                      int sendBufferIndex = 0; // Tracks the current size of sendBuffer
                      int bytesRead;
  
                      Console.WriteLine("Reading WebM output from FFmpeg and sending via WebSocket...");
                      while (true)
                      {
                          if ((bytesRead = await ffmpegOutputStream.ReadAsync(buffer, 0, buffer.Length)) > 0)
                          {
                              // Copy data to sendBuffer
                              Array.Copy(buffer, 0, sendBuffer, sendBufferIndex, bytesRead);
                              sendBufferIndex += bytesRead;
  
                              // If sendBuffer is full, send it via WebSocket
                              if (sendBufferIndex >= sendBuffer.Length)
                              {
                                  var segment = new ArraySegment(sendBuffer, 0, sendBuffer.Length);
                                  _ws.SendMessage(segment);
                                  sendBufferIndex = 0; // Reset the index after sending
                              }
                          }
                      }
                  }
                  catch (OperationCanceledException)
                  {
                      Console.WriteLine("Encode/Send operation cancelled.");
                  }
                  catch (IOException ex) when (ex.InnerException is ObjectDisposedException)
                  {
                      Console.WriteLine("Stream was closed, likely due to process exit or cancellation.");
                  }
                  catch (Exception ex)
                  {
                      Console.Error.WriteLine($"Error during encoding/sending: {ex}");
                  }
              });
  
              var errorReadTask = Task.Run(async () =>
              {
                  Console.WriteLine("Starting to read FFmpeg stderr...");
                  using var errorReader = _ffmpegProcess.StandardError;
                  try
                  {
                      string? line;
                      while ((line = await errorReader.ReadLineAsync()) != null) 
                      {
                          Console.WriteLine($"[FFmpeg stderr] {line}");
                      }
                  }
                  catch (OperationCanceledException) { Console.WriteLine("FFmpeg stderr reading cancelled."); }
                  catch (TimeoutException) { Console.WriteLine("FFmpeg stderr reading timed out (due to cancellation)."); }
                  catch (Exception ex) { Console.Error.WriteLine($"Error reading FFmpeg stderr: {ex.Message}"); }
                  Console.WriteLine("Finished reading FFmpeg stderr.");
              });
  
          }
  
          public async Task AppendAudioBuffer(AudioMediaBuffer audioBuffer)
          {
              try
              {
                  // audio for a 1:1 call
                  var bufferLength = audioBuffer.Length;
                  if (bufferLength > 0)
                  {
                      var buffer = new byte[bufferLength];
                      Marshal.Copy(audioBuffer.Data, buffer, 0, (int)bufferLength);
  
                      _logger.Info("_ffmpegProcess.HasExited:" + _ffmpegProcess.HasExited);
                      using var ffmpegInputStream = _ffmpegProcess.StandardInput.BaseStream;
                      await ffmpegInputStream.WriteAsync(buffer, 0, buffer.Length);
                      await ffmpegInputStream.FlushAsync(); // バッファをフラッシュ
                      _logger.Info("Wrote buffer data.");
  
                  }
              }
              catch (Exception e)
              {
                  _logger.Error(e, "Exception happend writing to input stream");
              }
          }
  
  

  Starting FFmpeg process...
  FFmpeg process started.
  Starting to read FFmpeg stderr...
  Reading WebM output from FFmpeg and sending via WebSocket...
  [FFmpeg stderr] ffmpeg version 7.1.1-essentials_build-www.gyan.dev Copyright (c) 2000-2025 the FFmpeg developers
  [FFmpeg stderr]   built with gcc 14.2.0 (Rev1, Built by MSYS2 project)
  [FFmpeg stderr]   configuration: --enable-gpl --enable-version3 --enable-static --disable-w32threads --disable-autodetect --enable-fontconfig --enable-iconv --enable-gnutls --enable-libxml2 --enable-gmp --enable-bzlib --enable-lzma --enable-zlib --enable-libsrt --enable-libssh --enable-libzmq --enable-avisynth --enable-sdl2 --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxvid --enable-libaom --enable-libopenjpeg --enable-libvpx --enable-mediafoundation --enable-libass --enable-libfreetype --enable-libfribidi --enable-libharfbuzz --enable-libvidstab --enable-libvmaf --enable-libzimg --enable-amf --enable-cuda-llvm --enable-cuvid --enable-dxva2 --enable-d3d11va --enable-d3d12va --enable-ffnvcodec --enable-libvpl --enable-nvdec --enable-nvenc --enable-vaapi --enable-libgme --enable-libopenmpt --enable-libopencore-amrwb --enable-libmp3lame --enable-libtheora --enable-libvo-amrwbenc --enable-libgsm --enable-libopencore-amrnb --enable-libopus --enable-libspeex --enable-libvorbis --enable-librubberband
  [FFmpeg stderr]   libavutil      59. 39.100 / 59. 39.100
  [FFmpeg stderr]   libavcodec     61. 19.101 / 61. 19.101
  [FFmpeg stderr]   libavformat    61.  7.100 / 61.  7.100
  [FFmpeg stderr]   libavdevice    61.  3.100 / 61.  3.100
  [FFmpeg stderr]   libavfilter    10.  4.100 / 10.  4.100
  [FFmpeg stderr]   libswscale      8.  3.100 /  8.  3.100
  [FFmpeg stderr]   libswresample   5.  3.100 /  5.  3.100
  [FFmpeg stderr]   libpostproc    58.  3.100 / 58.  3.100
  
  [2025-05-06 15:44:43,598][INFO][XbLogger.cs:85] _ffmpegProcess.HasExited:False
  [2025-05-06 15:44:43,613][INFO][XbLogger.cs:85] Wrote buffer data.
  [2025-05-06 15:44:43,613][INFO][XbLogger.cs:85] Wrote buffer data.
  [FFmpeg stderr] [aist#0:0/pcm_s16le @ 0000025ec8d36040] Guessed Channel Layout: mono
  [FFmpeg stderr] Input #0, s16le, from 'pipe:0':
  [FFmpeg stderr]   Duration: N/A, bitrate: 256 kb/s
  [FFmpeg stderr]   Stream #0:0: Audio: pcm_s16le, 16000 Hz, mono, s16, 256 kb/s
  [FFmpeg stderr] Stream mapping:
  [FFmpeg stderr]   Stream #0:0 -> #0:0 (pcm_s16le (native) -> opus (libopus))
  [FFmpeg stderr] [libopus @ 0000025ec8d317c0] No bit rate set. Defaulting to 64000 bps.
  [FFmpeg stderr] Output #0, webm, to 'pipe:1':
  [FFmpeg stderr]   Metadata:
  [FFmpeg stderr]     encoder         : Lavf61.7.100
  [FFmpeg stderr]   Stream #0:0: Audio: opus, 16000 Hz, mono, s16, 64 kb/s
  [FFmpeg stderr]       Metadata:
  [FFmpeg stderr]         encoder         : Lavc61.19.101 libopus
  [FFmpeg stderr] [out#0/webm @ 0000025ec8d36200] video:0KiB audio:1KiB subtitle:0KiB other streams:0KiB global headers:0KiB muxing overhead: 67.493113%
  [FFmpeg stderr] size=       1KiB time=00:00:00.04 bitrate= 243.2kbits/s speed=2.81x
  Finished reading FFmpeg stderr.
  [2025-05-06 15:44:44,101][INFO][XbLogger.cs:85] _ffmpegProcess.HasExited:True
  [2025-05-06 15:44:44,132][ERROR][XbLogger.cs:67] Exception happend writing to input stream
  System.ObjectDisposedException: Cannot access a closed file.
     at System.IO.FileStream.WriteAsync(Byte[] buffer, Int32 offset, Int32 count, CancellationToken cancellationToken)
     at System.IO.Stream.WriteAsync(Byte[] buffer, Int32 offset, Int32 count)
     at EchoBot.Media.SpeechService.AppendAudioBuffer(AudioMediaBuffer audioBuffer) in C:\Users\tm068\Documents\workspace\myprj\xbridge-teams-bot\src\EchoBot\Media\SpeechService.cs:line 242
  

  I am expecting the ffmpeg process keep running.

• MoviePY write_videofile using GPU for faster encoding [closed]

  5 mai, par kaushal

  I'm creating a video from scratch using moviePY. I am generating all the required frames, adding required audio (including a voiceover and background music), adding a logo and finally writing the video file in 4K.

  Everything works fine, except the write_videofile takes a lot of time.

  I have read many related posts which mentions using the right codec etc. I have NVidia card, so tried both h264_nvenc and hevc_nvenc. Quality of the output video dropped with the first one, so I'm sticking to hevc_nvenc. I'm using below line to write the file.

          video_clip.write_videofile(targetfile, codec="hevc_nvenc", threads=32, fps=24)
  

  What I have noticed is, it does seem to be using the gpu, but very little. Compared to this, when I run stable diffusion or vegas rendering, it uses gpu a lot more.

  That's why I think there is definitely some scope of improvement here. If you see below screenshot, when the video file write starts, the gpu utilisation increases a tiny bit, but it can take a lot more I think, isn't it?

  I can try various parameters that I've seen in other threads, like logger=None, progress_bar = False, ffmpeg_params=['-b:v','10000k'] etc., but they are not going to improve gpu utilisation in any shape or form. I've been wondering what am I missing.

  Any ideas or suggestions please?

  

• Live streaming webvtt subtitles with HLS protocol

  5 mai, par Victor Ruiz

  I need a tool to generate HLS subtitles in live mode. Specifically, I want to create an HTTP server that serves .m3u8 and .webvtt files which are continuously updated over time. Such HLS stream could be consumed via HTTP requests by HLS JS/ffplay multimedia players.

  The .webvtt files will be generated by an automatic transcriber, so the program must update the .m3u8 playlist accordingly whenever a new subtitle is produced.

  I only want to stream subtitle channels—no audio. The video channel can simply display a black chroma background.

  I attempted to use FFmpeg with a Linux pipe as input for the streaming .webvtt subtitles, along with a video file for the video stream. The output .webvtt and .m3u8 files were written to a folder and served via an NGINX server. However, FFmpeg fails after it reads the initial content of the .webvtt input from the pipe. If I inject more content afterward, it gets skipped.

  How can I achieve HLS subtitle streaming in live mode? Can FFmpeg be used for this purpose, or do I need a different tool?