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• h264 lossless coding

  29 septembre 2014, par cloudraven

  Is it possible to do completely lossless encoding in h264 ? By lossless, I mean that if I feed it a series of frames and encode them, and then if I extract all the frames from the encoded video, I will get the exact same frames as in the input, pixel by pixel, frame by frame. Is that actually possible ?
  Take this example :

  I generate a bunch of frames, then I encode the image sequence to an uncompressed AVI (with something like virtualdub), I then apply lossless h264 (the help files claim that setting —qp 0 makes lossless compression, but I am not sure if that means that there is no loss at any point of the process or that just the quantization is lossless). I can then extract the frames from the resulting h264 video with something like mplayer.

  I tried with Handbrake first, but it turns out it doesn’t support lossless encoding. I tried x264 but it crashes. It may be because my source AVI file is in RGB colorspace instead of YV12. I don’t know how to feed a series of YV12 bitmaps and in what format to x264 anyway, so I cannot even try.

  In summary what I want to know if that is there a way to go from

  Series of lossless bitmaps (in any colorspace) -> some transformation -> h264 encode -> h264 decode -> some transformation -> the original series of lossless bitmaps

  If there a way to achieve this ?

  EDIT : There is a VERY valid point about lossless H264 not making too much sense. I am well aware that there is no way I could tell (with just my eyes) the difference between and uncompressed clip and another compressed at a high rate in H264, but I don’t think it is not without uses. For example, it may be useful for storing video for editing without taking huge amounts of space and not losing quality and spending too much encoding time every time the file is saved.

  UPDATE 2 : Now x264 doesn’t crash. I can use as sources either avisynth or lossless yv12 lagarith (to avoid the colorspace compression warning). Howerver, even with —qp 0 and a rgb or yv12 source I still get some differences, minimal but present. This is troubling, because all the information I have found on lossless predictive coding (—qp 0) claims that the whole encoding should be lossless, but I am unable to verifiy this.

• Bluemix node js build pack to support webm audio conversion

  31 août 2018, par Nimmy Mohandas

  Even after adding ffmpeg to bluemix node js build pack(I tried this https://github.com/BlueChasm/nodejs-buildpack-ffmpeg), it doesn’t support webm audio format conversion.Could anyone please suggest alternate ways to support this issue ?

  Description : I want to do speech to text conversion using Google speech recognition. I am using react native for front-end development. I recorded the voice and generated a webm audio file. Since the google speech recognition doesn’t support webm file format, I converted it into wav format using fluent-ffmpeg. It is working perfectly in my local system, but when I deploy it to the IBM cloud foundry using ffmpeg buildpack(https://github.com/BlueChasm/nodejs-buildpack-ffmpeg), it generates a file without extension. Following are my code,

  router.post('/st', audioUpload.single('audio'), function(req, res, next) {
  
          if (!req.file) {
  
              return res.json({ status: false, error: 'No input given!' });
  
          }
  
  
  
          /**
  
           * convert to .wav
  
           */
  
  
  
  
  
          ffmpeg(req.file.path)
  
              .toFormat('wav')
  
              .inputOptions(['-r 32000',
  
                  '-ac 1'
  
              ])
  
  
  
              .on('error', (err) => {
  
                  console.log('An error occurred: ' + err.message);
  
              })
  
              .save(path.join(__dirname, '..', 'uploads/audio/file.wav'))
  
  
  
          .on('end', () => {
  
  
  
              st.detectAudioFile(path.join(__dirname, '..', 'uploads/audio/file.wav')).then(data => {
  
                      fs.unlink(path.join(__dirname, '..', 'uploads/audio/file.wav'), function(err) {
  
                          if (err) return res.send(err);
  
                          const response = data[0];
  
                          const transcription = response.results
  
                              .map(result => result.alternatives[0].transcript)
  
                              .join('\n');
  
                          return res.json({ transcription });
  
                      }), fs.unlink(req.file.path, function(err) {
  
                          if (err) return res.json(err);
  
                      })
  
  
  
                  })
  
                  .catch(err => {
  
                      res.send({
  
  
  
                          status: false,
  
                          error_code: 400,
  
                          err: err.error || err.message
  
                      });
  
                  });
  
          })
  
      })

  //speech-to-text.js

  const fs = require('fs');
  
  const speech = require('@google-cloud/speech');
  
  
  
  
  
  var detectAudioFile = function(fileName) {
  
  
  
      const Speechclient = new speech.SpeechClient({})
  
  
  
      // Reads a local audio file and converts it to base64
  
      const file = fs.readFileSync(fileName);
  
  
  
      const audioBytes = file.toString('base64');
  
      const audio = {
  
          content: audioBytes,
  
      };
  
      const config = {
  
          // encoding: "FLAC",
  
          // sampleRateHertz: 44100,
  
  
  
          languageCode: 'en-US',
  
      };
  
      const request = {
  
          audio: audio,
  
          config: config,
  
      };
  
      return Speechclient
  
          .recognize(request);
  
  
  
  }
  
  
  
  
  
  
  
  module.exports = {
  
      detectAudioFile
  
  }

  `

• avconv file protocol with rw_timeout and follow

  29 août 2016, par Kasper Veenvliet

  I use avconv to convert a webm (still being recorded while converting) to h264.
  I want avconv to wait while being written to the file.

  https://libav.org/avconv.html#file says I can use the follow and rw_timeout option.
  But when I use them with avconv 11.7 I always get the unregonized option response.

  avconv  -y -follow 1 -i file:/tmp/video/krtv/5405457303486272/recordings/video.webm -s hd720 -r 15 -strict experimental -c:v libx264 -c:a aac /tmp/video/krtv/5405457303486272/sized/video.mp4
  
  avconv version 11.7, Copyright (c) 2000-2016 the Libav developers
  
    built on Aug 29 2016 10:40:47 with gcc 4.7 (Debian 4.7.2-5)
  
  Unrecognized option 'follow'.
  
  Error splitting the argument list: Option not found

  How to use these options on the file protocol with avconv 11.7 ?