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• FFMpeg C# wrapper "MediaToolkit 1.0.4.11". How can I do a conversion in memory ?

  8 décembre 2016, par Quantum_Kernel

  I am currently using the MediaToolkit 1.0.4.11 C# wrapper for FFMPEG to extract wav audio from a range of video files.

  Instead of creating a wav file on disk, I would like to create it in memory (as a MemoryStream). This will save me from creating temp files which I need to delete and should speed up the various analyses that I am doing on the audio data.

  Because of the way the call to the library works, I’m not sure how to ’trick’ it into doing this. Is there a way to do it or will I have to obtain and edit the wrapper source code ?

  Here is what I have for doing the conversion on disk :

  var inputFile = new MediaToolkit.Model.MediaFile { Filename = mediaFilePath };
  
              var outputFile = new MediaToolkit.Model.MediaFile { Filename = @"C:\Temp\audio.wav" };
  
  
  
              var conversionOptions = new MediaToolkit.Options.ConversionOptions
  
              {
  
                  MaxVideoDuration = TimeSpan.FromSeconds(30),
  
                  VideoAspectRatio = MediaToolkit.Options.VideoAspectRatio.R16_9,
  
                  VideoSize = MediaToolkit.Options.VideoSize.Hd1080,
  
                  AudioSampleRate = MediaToolkit.Options.AudioSampleRate.Hz48000
  
              };
  
  
  
              using (var engine = new Engine())
  
              {
  
                  engine.Convert(inputFile, outputFile, conversionOptions);
  
              }

• Watson NarrowBand Speech to Text not accepting ogg file

  19 janvier 2017, par Bob Dill

  NodeJS app using ffmpeg to create ogg files from mp3 & mp4. If the source file is broadband, Watson Speech to Text accepts the file with no issues. If the source file is narrow band, Watson Speech to Text fails to read the ogg file. I’ve tested the output from ffmpeg and the narrowband ogg file has the same audio content (e.g. I can listen to it and hear the same people) as the mp3 file. Yes, in advance, I am changing the call to Watson to correctly specify the model and content_type. Code follows :

  exports.createTranscript = function(req, res, next)
  
  { var _name = getNameBase(req.body.movie);
  
    var _type = getType(req.body.movie);
  
    var _voice = (_type == "mp4") ? "en-US_BroadbandModel" : "en-US_NarrowbandModel" ;
  
    var _contentType = (_type == "mp4") ? "audio/ogg" : "audio/basic" ;
  
    var _audio = process.cwd()+"/HTML/movies/"+_name+'ogg';
  
    var transcriptFile = process.cwd()+"/HTML/movies/"+_name+'json';
  
  
  
    speech_to_text.createSession({model: _voice}, function(error, session) {
  
      if (error) {console.log('error:', error);}
  
      else
  
        {
  
          var params = { content_type: _contentType, continuous: true,
  
           audio: fs.createReadStream(_audio),
  
            session_id: session.session_id
  
            };
  
            speech_to_text.recognize(params, function(error, transcript) {
  
              if (error) {console.log('error:', error);}
  
              else
  
                { fs.writeFile(transcriptFile, JSON.stringify(transcript), function(err) {if (err) {console.log(err);}});
  
                  res.send(transcript);
  
                }
  
            });
  
        }
  
    });
  
  }

  _type is either mp3 (narrowband from phone recording) or mp4 (broadband)
  model: _voice has been traced to ensure correct setting
  content_type: _contentType has been traced to ensure correct setting

  Any ogg file submitted to Speech to Text with narrowband settings fails with Error: No speech detected for 30s. Tested with both real narrowband files and asking Watson to read a broadband ogg file (created from mp4) as narrowband. Same error message. What am I missing ?

• FPS drop in FFMPEG streaming processes to FB from production server

  30 janvier 2017, par Aakash Gupta

  I have made a rails app that can stream live videos to facebook rtmp server and deployed it on AWS. I have used nginx as web server. The major problem that I am encountering after viewing log files of FFMpeg processes is that sometimes the FPS of FFmpeg process starts to drop. In some cases, it remains stable at 25 FPS but in some cases, it remains at 25 only for sometime, and after that it starts to drop and sometimes it falls to even 3-4 FPS which is unacceptable during live streaming. As FFMpeg process is quite heavy, I would also like to share my CPU info as well.

  CPU information is :

  cat /proc/cpuinfo
processor   : 0
vendor_id   : GenuineIntel
cpu family  : 6
model       : 63
model name  : Intel(R) Xeon(R) CPU E5-2676 v3 @ 2.40GHz
stepping    : 2
microcode   : 0x25
cpu MHz     : 2400.070
cache size  : 30720 KB
physical id : 0
siblings    : 1
core id     : 0
cpu cores   : 1
apicid      : 0
initial apicid  : 0
fpu     : yes
fpu_exception   : yes
cpuid level : 13
wp      : yes
flags       : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush mmx fxsr sse sse2 ht syscall nx rdtscp lm constant_tsc rep_good nopl xtopology eagerfpu pni pclmulqdq ssse3 fma cx16 pcid sse4_1 sse4_2 x2apic movbe popcnt tsc_deadline_timer aes xsave avx f16c rdrand hypervisor lahf_lm abm xsaveopt fsgsbase bmi1 avx2 smep bmi2 erms invpcid
bogomips    : 4800.14
clflush size    : 64
cache_alignment : 64
address sizes   : 46 bits physical, 48 bits virtual
power management:

  FFMPEG log file with unstable fps : https://drive.google.com/open?id=0B1gtp1iXJppkUndFamk4M0lRYzA

  FFMPEG log file with stable fps : https://drive.google.com/open?id=0B1gtp1iXJppkMkVCZEJjYWJrVTA

  When FPS was stable, I also tried to run another parallel FFMpeg process from the same server which resulted in FPS dropping of both the processes to 13-14 FPS.

  I am currently using this FFMPEG command :

  ffmpeg -loop 1 -re -y -f image2 -i "image_path" -i "audio_path.aac" -acodec copy -bsf:a aac_adtstoasc -pix_fmt yuv420p -profile:v high -s 1280x720 -vb 400k -maxrate 400k -minrate 400k -bufsize 600k -deinterlace -vcodec libx264 -preset veryfast -g 30 -r 30 -t 14400 -strict -2 -f flv "rtmp_server_link"

  I never face this problem when I try to stream to FB using app on my localhost.

  So, my questions are :

  1. What can be the reason for this FPS drop ?
  2. Can upscaling production server help me fix this issue ?
  3. Can I run multiple FFMpeg processes for streaming from same server without performance drop ?

  Thanks in advance :)