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• FFMpeg - how to encode a F4M manifest file to mp4

  20 novembre 2015, par Roee

  I’m working on a project, where we want to encode flash videos to mp4 (for example, we want to encode a f4v file), and live-stream them (which means we can’t just download all the f4f files, then encode them to a single mp4 and just then stream it. The encoding and streaming has to be done while downloading the files).

  If the format of the flash video is flv for example, FFMpeg can do what I’ve described without any problem. I just give the address of the flv file to FFMpeg, and it encodes and streams it as mp4.

  But, if the format of the flash video is something more complicated as f4v (which gets downloaded as many f4f files), I don’t have a single url to give to FFMpeg as input - I have many addresses of f4f files. I don’t even know how many f4f files the video has before playing it - it looks like the flash player just fetches the next f4f file when needed.

  I read online that there is a manifest file (its extension is f4m), that "describes" to the flash player which f4f files it should download and play, and what’s their playing-order and everything.

  My question is - if I have the url of this f4m file, what should I do in order to encode all its f4f files to mp4 ?
  I’ve tried to give to FFMpeg just the f4m file as input, but it doesn’t know what to do with it...

  I’ll really appreciate any response that might help, as I’ve been working on this issue for few days now and I still haven’t found any answer...

  Thanks,
  Roee.

• How do you use Node.js to stream an MP4 file with ffmpeg ?

  2 novembre 2016, par LaserJesus

  I’ve been trying to solve this problem for several days now and would really appreciate any help on the subject.

  I’m able to successfully stream an mp4 audio file stored on a Node.js server using fluent-ffmpeg by passing the location of the file as a string and transcoding it to mp3. If I create a file stream from the same file and pass that to fluent-ffmpeg instead it works for an mp3 input file, but not a mp4 file. In the case of the mp4 file no error is thrown and it claims the stream completed successfully, but nothing is playing in the browser. I’m guessing this has to do with the meta data being stored at the end of an mp4 file, but I don’t know how to code around this. This is the exact same file that works correctly when it’s location is passed to ffmpeg, rather than the stream. When I try and pass a stream to the mp4 file on s3, again no error is thrown, but nothing streams to the browser. This isn’t surprising as ffmpeg won’t work with the file locally as stream, so expecting it to handle the stream from s3 is wishful thinking.

  How can I stream the mp4 file from s3, without storing it locally as a file first ? How do I get ffmpeg to do this without transcoding the file too ? The following is the code I have at the moment which isn’t working. Note that it attempts to pass the s3 file as a stream to ffmpeg and it’s also transcoding it into an mp3, which I’d prefer not to do.

  .get(function(req,res) {
  
      aws.s3(s3Bucket).getFile(s3Path, function (err, result) {
  
          if (err) {
  
              return next(err);
  
          }
  
          var proc = new ffmpeg(result)
  
              .withAudioCodec('libmp3lame')
  
              .format('mp3')
  
              .on('error', function (err, stdout, stderr) {
  
                  console.log('an error happened: ' + err.message);
  
                  console.log('ffmpeg stdout: ' + stdout);
  
                  console.log('ffmpeg stderr: ' + stderr);
  
              })
  
              .on('end', function () {
  
                  console.log('Processing finished !');
  
              })
  
              .on('progress', function (progress) {
  
                  console.log('Processing: ' + progress.percent + '% done');
  
              })
  
              .pipe(res, {end: true});
  
      });
  
  });

  This is using the knox library when it calls aws.s3... I’ve also tried writing it using the standard aws sdk for Node.js, as shown below, but I get the same outcome as above.

  var AWS = require('aws-sdk');
  
  
  
  var s3 = new AWS.S3({
  
      accessKeyId: process.env.AWS_ACCESS_KEY_ID,
  
      secretAccessKey: process.env.AWS_SECRET_KEY,
  
      region: process.env.AWS_REGION_ID
  
  });
  
  var fileStream = s3.getObject({
  
          Bucket: s3Bucket,
  
          Key: s3Key
  
      }).createReadStream();
  
  var proc = new ffmpeg(fileStream)
  
      .withAudioCodec('libmp3lame')
  
      .format('mp3')
  
      .on('error', function (err, stdout, stderr) {
  
          console.log('an error happened: ' + err.message);
  
          console.log('ffmpeg stdout: ' + stdout);
  
          console.log('ffmpeg stderr: ' + stderr);
  
      })
  
      .on('end', function () {
  
          console.log('Processing finished !');
  
      })
  
      .on('progress', function (progress) {
  
          console.log('Processing: ' + progress.percent + '% done');
  
      })
  
      .pipe(res, {end: true});

  =====================================

  Updated

  I placed an mp3 file in the same s3 bucket and the code I have here worked and was able to stream the file through to the browser without storing a local copy. So the streaming issues I face have something to do with the mp4/aac container/encoder format.

  I’m still interested in a way to bring the m4a file down from s3 to the Node.js server in it’s entirety, then pass it to ffmpeg for streaming without actually storing the file in the local file system.

  =====================================

  Updated Again

  I’ve managed to get the server streaming the file, as mp4, straight to the browser. This half answers my original question. My only issue now is that I have to download the file to a local store first, before I can stream it. I’d still like to find a way to stream from s3 without needing the temporary file.

  aws.s3(s3Bucket).getFile(s3Path, function(err, result){
  
      result.pipe(fs.createWriteStream(file_location));
  
      result.on('end', function() {
  
          console.log('File Downloaded!');
  
          var proc = new ffmpeg(file_location)
  
              .outputOptions(['-movflags isml+frag_keyframe'])
  
              .toFormat('mp4')
  
              .withAudioCodec('copy')
  
              .seekInput(offset)
  
              .on('error', function(err,stdout,stderr) {
  
                  console.log('an error happened: ' + err.message);
  
                  console.log('ffmpeg stdout: ' + stdout);
  
                  console.log('ffmpeg stderr: ' + stderr);
  
              })
  
              .on('end', function() {
  
                  console.log('Processing finished !');
  
              })
  
              .on('progress', function(progress) {
  
                  console.log('Processing: ' + progress.percent + '% done');
  
              })
  
              .pipe(res, {end: true});
  
      });
  
  });

  On the receiving side I just have the following javascript in an empty html page :

  window.AudioContext = window.AudioContext || window.webkitAudioContext;
  
  context = new AudioContext();
  
  
  
  function process(Data) {
  
      source = context.createBufferSource(); // Create Sound Source
  
      context.decodeAudioData(Data, function(buffer){
  
          source.buffer = buffer;
  
          source.connect(context.destination);
  
          source.start(context.currentTime);
  
      });
  
  };
  
  
  
  function loadSound() {
  
      var request = new XMLHttpRequest();
  
      request.open("GET", "/stream/", true);
  
      request.responseType = "arraybuffer";
  
  
  
      request.onload = function() {
  
          var Data = request.response;
  
          process(Data);
  
      };
  
  
  
      request.send();
  
  };
  
  
  
  loadSound()

  =====================================

  The Answer

  The code above under the title ’updated again’ will stream an mp4 file, from s3, via a Node.js server to a browser without using flash. It does require that the file be stored temporarily on the Node.js server so that the meta data in the file is moved from the end of the file to the front. In order to stream without storing the temporary file, you need to actual modify the file on S3 first and make this meta data change. If you have changed the file in this way on S3 then you can modify the code under the title ’updated again’ so that the result from S3 is piped straight into the ffmpeg constructor, rather than into a file stream on the Node.js server, then providing that file location to ffmepg, as the code does now. You can change the final ’pipe’ command to ’save(location)’ to get a version of the mp4 file locally with the meta data moved to the front. You can then upload that new version of the file to S3 and try out the end to end streaming. Personally I’m now going to create a task that modifies the files in this way as they are uploaded to s3 in the first place. This allows me to record and stream in mp4 without transcoding or storing a temp file on the Node.js server.

• FFmpeg : Concatenating two video files causing static output

  15 octobre 2015, par roy

  I have two .avi video files that I am trying to concat together.

  I have a files.txt that has in it :

  file './1_tmp.avi'
  
  file './2_tmp.avi'

  FFprobe for 1_tmp.avi gives me :

  ffprobe version 2.7.2 Copyright (c) 2007-2015 the FFmpeg developers
  
    built with gcc 4.8.5 (Gentoo 4.8.5 p1.3, pie-0.6.2)
  
    configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --enable-shared --cc=x86_64-pc-linux-gnu-gcc --cxx=x86_64-pc-linux-gnu-g++ --ar=x86_64-pc-linux-gnu-ar --optflags=' ' --disable-static --enable-avfilter --enable-avresample --disable-stripping --enable-version3 --disable-indev=v4l2 --disable-outdev=v4l2 --disable-indev=oss --disable-indev=jack --disable-outdev=oss --disable-outdev=sdl --enable-nonfree --enable-bzlib --disable-runtime-cpudetect --disable-debug --disable-doc --disable-gnutls --enable-gpl --enable-hardcoded-tables --enable-iconv --disable-lzma --enable-network --disable-openssl --enable-postproc --disable-libsmbclient --disable-ffplay --disable-vaapi --disable-vdpau --enable-xlib --disable-libxcb --disable-libxcb-shm --disable-libxcb-xfixes --enable-zlib --disable-libcdio --disable-libiec61883 --disable-libdc1394 --disable-libcaca --disable-openal --disable-opengl --disable-libv4l2 --disable-libpulse --disable-libopencore-amrwb --disable-libopencore-amrnb --enable-libfdk-aac --disable-libopenjpeg --disable-libbluray --disable-libcelt --disable-libgme --disable-libgsm --disable-libmodplug --disable-libopus --disable-libquvi --disable-librtmp --disable-libssh --disable-libschroedinger --disable-libspeex --disable-libvorbis --enable-libvpx --disable-libzvbi --disable-libbs2b --disable-libflite --disable-frei0r --disable-libfribidi --disable-fontconfig --disable-ladspa --disable-libass --disable-libfreetype --disable-libsoxr --enable-pthreads --enable-libvo-aacenc --disable-libvo-amrwbenc --enable-libmp3lame --disable-libaacplus --disable-libfaac --disable-libtheora --disable-libtwolame --disable-libwavpack --disable-libwebp --enable-libx264 --disable-libx265 --disable-libxvid --enable-x11grab --disable-amd3dnow --disable-amd3dnowext --disable-fma4 --disable-xop --cpu=core-avx-i
  
    libavutil      54. 27.100 / 54. 27.100
  
    libavcodec     56. 41.100 / 56. 41.100
  
    libavformat    56. 36.100 / 56. 36.100
  
    libavdevice    56.  4.100 / 56.  4.100
  
    libavfilter     5. 16.101 /  5. 16.101
  
    libavresample   2.  1.  0 /  2.  1.  0
  
    libswscale      3.  1.101 /  3.  1.101
  
    libswresample   1.  2.100 /  1.  2.100
  
    libpostproc    53.  3.100 / 53.  3.100
  
  Input #0, avi, from '1_tmp.avi':
  
    Metadata:
  
      encoder         : Lavf56.36.100
  
    Duration: 00:00:19.00, start: 0.000000, bitrate: 179869 kb/s
  
      Stream #0:0: Video: huffyuv (HFYU / 0x55594648), bgr0, 1024x768, 180390 kb/s, 18 fps, 18 tbr, 18 tbn, 18 tbc

  and for 2_tmp.avi :

  ffprobe version 2.7.2 Copyright (c) 2007-2015 the FFmpeg developers
  
    built with gcc 4.8.5 (Gentoo 4.8.5 p1.3, pie-0.6.2)
  
    configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --enable-shared --cc=x86_64-pc-linux-gnu-gcc --cxx=x86_64-pc-linux-gnu-g++ --ar=x86_64-pc-linux-gnu-ar --optflags=' ' --disable-static --enable-avfilter --enable-avresample --disable-stripping --enable-version3 --disable-indev=v4l2 --disable-outdev=v4l2 --disable-indev=oss --disable-indev=jack --disable-outdev=oss --disable-outdev=sdl --enable-nonfree --enable-bzlib --disable-runtime-cpudetect --disable-debug --disable-doc --disable-gnutls --enable-gpl --enable-hardcoded-tables --enable-iconv --disable-lzma --enable-network --disable-openssl --enable-postproc --disable-libsmbclient --disable-ffplay --disable-vaapi --disable-vdpau --enable-xlib --disable-libxcb --disable-libxcb-shm --disable-libxcb-xfixes --enable-zlib --disable-libcdio --disable-libiec61883 --disable-libdc1394 --disable-libcaca --disable-openal --disable-opengl --disable-libv4l2 --disable-libpulse --disable-libopencore-amrwb --disable-libopencore-amrnb --enable-libfdk-aac --disable-libopenjpeg --disable-libbluray --disable-libcelt --disable-libgme --disable-libgsm --disable-libmodplug --disable-libopus --disable-libquvi --disable-librtmp --disable-libssh --disable-libschroedinger --disable-libspeex --disable-libvorbis --enable-libvpx --disable-libzvbi --disable-libbs2b --disable-libflite --disable-frei0r --disable-libfribidi --disable-fontconfig --disable-ladspa --disable-libass --disable-libfreetype --disable-libsoxr --enable-pthreads --enable-libvo-aacenc --disable-libvo-amrwbenc --enable-libmp3lame --disable-libaacplus --disable-libfaac --disable-libtheora --disable-libtwolame --disable-libwavpack --disable-libwebp --enable-libx264 --disable-libx265 --disable-libxvid --enable-x11grab --disable-amd3dnow --disable-amd3dnowext --disable-fma4 --disable-xop --cpu=core-avx-i
  
    libavutil      54. 27.100 / 54. 27.100
  
    libavcodec     56. 41.100 / 56. 41.100
  
    libavformat    56. 36.100 / 56. 36.100
  
    libavdevice    56.  4.100 / 56.  4.100
  
    libavfilter     5. 16.101 /  5. 16.101
  
    libavresample   2.  1.  0 /  2.  1.  0
  
    libswscale      3.  1.101 /  3.  1.101
  
    libswresample   1.  2.100 /  1.  2.100
  
    libpostproc    53.  3.100 / 53.  3.100
  
  Input #0, avi, from '2_tmp.avi':
  
    Metadata:
  
      encoder         : Lavf56.36.100
  
    Duration: 00:00:56.00, start: 0.000000, bitrate: 160560 kb/s
  
      Stream #0:0: Video: huffyuv (HFYU / 0x55594648), bgr0, 1024x768, 160714 kb/s, 18 fps, 18 tbr, 18 tbn, 18 tbc

  There are only video streams, they look to be the same codecs/fps

  I am using this command to concatenate them :

  ffmpeg -f concat -i files.txt -c:v copy -y output.avi

  I don’t get any output from the terminal and it outputs a file.

  The file opens and the content in 1_tmp.avi plays fine, at the same quality, but when it gets to the point in the video where 2_tmp.avi should be, it’s pixelated static all the way through (see image below).

  

  If I use a command like :

  ffmpeg -i 1_tmp.avi -i 2_tmp.avi -filter_complex "[0:0] [1:0] concat=n=2:v=1 [v]" -map "[v]" output.avi

  I do get full output of both videos merged together, but it is dropped to an mpeg codec and the quality loss is noticeably prominent.