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• When generating a video from images and adding audio the video isn't playable

  1er août 2017, par randommman

  So I have a bunch of images in a folder, with the following structure :

  image-0.png
  
  image-1.png
  
  image-2.png

  Sometimes the folder can have just 1 image which will be saved as :

  image-0.png

  My code to generate the video and to add the audio

  // create the file
  
  using (WaveFileReader wf = new WaveFileReader(Audio))
  
  {
  
      // the files
  
      OldVideo = "old.avi";
  
      NewVideo = "video.avi";
  
      Audio = "sound.wav";
  
  
  
      // Get the time of the audio and divide by images
  
      time = wf.TotalTime.TotalSeconds;
  
      mimtime = time / imagescount;
  
  
  
      ffmpegPath = "ffmpeg.exe";
  
      ffmpegParams = " -r 1/" + mimtime + " -i " + ImagesFolder + "image-%d.png -t " + time + " -y -vf scale=1280:-2 " + OldVideo;
  
      ffmaudioParams = " -i " + OldVideo + " -i " + Audio + " -c copy -shortest " + NewVideo;
  
  }
  
  
  
  
  
  using (Process ffmpeg = new Process())
  
  {
  
      //Generate video
  
      ffmpegstartInfo = new ProcessStartInfo();
  
      ffmpegstartInfo.FileName = ffmpegPath;
  
      ffmpegstartInfo.Arguments = ffmpegParams;
  
      ffmpegstartInfo.RedirectStandardOutput = true;
  
      ffmpegstartInfo.RedirectStandardError = true;
  
      ffmpegstartInfo.UseShellExecute = false;
  
      ffmpegstartInfo.CreateNoWindow = true;
  
  
  
      ffmpeg.StartInfo.FileName = ffmpegPath;
  
      ffmpeg.StartInfo.Arguments = ffmpegParams;
  
      ffmpeg.StartInfo = ffmpegstartInfo;
  
      ffmpeg.Start();
  
      ffmpeg.WaitForExit(30000);
  
  
  
  
  
      /*ffmpeg.StartInfo.FileName = "cmd.exe";
  
      ffmpeg.StartInfo.Arguments = "/k " + ffmpegPath + " " + ffmpegParams;
  
      ffmpeg.Start();
  
      ffmpeg.WaitForExit(30000);*/
  
  }
  
  
  
  using (Process ffmaudio = new Process())
  
  {
  
      //Add audio to video
  
      ffmaudiostartInfo = new ProcessStartInfo();
  
      ffmaudiostartInfo.FileName = ffmpegPath;
  
      ffmaudiostartInfo.Arguments = ffmaudioParams;
  
      ffmaudiostartInfo.RedirectStandardOutput = true;
  
      ffmaudiostartInfo.RedirectStandardError = true;
  
      ffmaudiostartInfo.UseShellExecute = false;
  
      ffmaudiostartInfo.CreateNoWindow = true;
  
  
  
      ffmaudio.StartInfo.FileName = ffmpegPath;
  
      ffmaudio.StartInfo.Arguments = ffmaudioParams;
  
      ffmaudio.StartInfo = ffmaudiostartInfo;
  
      ffmaudio.Start();
  
      ffmaudio.WaitForExit(30000);
  
  
  
      /*ffmaudio.StartInfo.FileName = "cmd.exe";
  
      ffmaudio.StartInfo.Arguments = "/k " + ffmpegPath + " " + ffmaudioParams;
  
      ffmaudio.Start();
  
      ffmaudio.WaitForExit(30000);*/
  
  }

  So, what my code does, with the wavfilereader it gets the total length of my audio file, and then splits that time up by the number of images I have, so I can generate a video with the images split equally. Then I generate the video to a file called ’old.avi’, which is then used with my wav file to generate a new file called video.avi (which I am trying to have the generated video and audio together)

  My problem is when I generate the video it becomes uplayable. It claims to be (for example) 1 minute 50 seconds long, but once I click it, it won’t play. It just becomes unplayable, as you can see below

  http://i.imgur.com/tsck2NX.gifv

  I am using C# with process and Winforms, which opens the cmd line to run ffmpeg, with the 2 commands called above with the named variables below :

  ffmpegParams;
  
  ffmaudioParams;

  Which has all the commands I am using

  -r 1/ (audio length /divided by total images) so they're equal
  
  -t (to limit to the total time which is the audio's total seconds)
  
  ... I am using more as you can see in my code above

  How can I fix my problem, that I have described ?

  CMD line when generating the video :

  ffmpeg version N-86848-g03a9e6f Copyright (c) 2000-2017 the FFmpeg developers
  
    built with gcc 7.1.0 (GCC)
  
    configuration: --enable-gpl --enable-version3 --enable-cuda --enable-cuvid --enable-d3d11va --enable-dxva2 --enable-libmfx --enable-nvenc --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-zlib
  
    libavutil      55. 68.100 / 55. 68.100
  
    libavcodec     57.102.100 / 57.102.100
  
    libavformat    57. 76.100 / 57. 76.100
  
    libavdevice    57.  7.100 / 57.  7.100
  
    libavfilter     6. 95.100 /  6. 95.100
  
    libswscale      4.  7.101 /  4.  7.101
  
    libswresample   2.  8.100 /  2.  8.100
  
    libpostproc    54.  6.100 / 54.  6.100
  
  Input #0, image2, from 'C:\Users\Laptop\Documents\program/images/image-%d.png':
  
    Duration: 00:00:00.04, start: 0.000000, bitrate: N/A
  
      Stream #0:0: Video: png, rgba(pc), 800x800, 25 tbr, 25 tbn, 25 tbc
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (png (native) -> mpeg4 (native))
  
  Press [q] to stop, [?] for help
  
  [mpeg4 @ 00000000024f44a0] bitrate tolerance 4000000 too small for bitrate 200000, overriding
  
  Output #0, avi, to 'C:\Users\Laptop\Documents\program/old.avi':
  
    Metadata:
  
      ISFT            : Lavf57.76.100
  
      Stream #0:0: Video: mpeg4 (FMP4 / 0x34504D46), yuv420p, 1280x1280, q=2-31, 200 kb/s, 0.01 fps, 0.01 tbn, 0.01 tbc
  
      Metadata:
  
        encoder         : Lavc57.102.100 mpeg4
  
      Side data:
  
        cpb: bitrate max/min/avg: 0/0/200000 buffer size: 0 vbv_delay: -1
  
  frame=    1 fps=0.0 q=6.8 Lsize=      46kB time=00:01:37.57 bitrate=   3.9kbits/s speed=1.43e+003x
  
  video:41kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 13.641467%

  When adding the audio to the video

  ffmpeg version N-86848-g03a9e6f Copyright (c) 2000-2017 the FFmpeg developers
  
    built with gcc 7.1.0 (GCC)
  
    configuration: --enable-gpl --enable-version3 --enable-cuda --enable-cuvid --enable-d3d11va --enable-dxva2 --enable-libmfx --enable-nvenc --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-zlib
  
    libavutil      55. 68.100 / 55. 68.100
  
    libavcodec     57.102.100 / 57.102.100
  
    libavformat    57. 76.100 / 57. 76.100
  
    libavdevice    57.  7.100 / 57.  7.100
  
    libavfilter     6. 95.100 /  6. 95.100
  
    libswscale      4.  7.101 /  4.  7.101
  
    libswresample   2.  8.100 /  2.  8.100
  
    libpostproc    54.  6.100 / 54.  6.100
  
  Input #0, avi, from 'C:\Users\Laptop\Documents\program/old.avi':
  
    Metadata:
  
      encoder         : Lavf57.76.100
  
    Duration: 00:01:37.58, start: 0.000000, bitrate: 3 kb/s
  
      Stream #0:0: Video: mpeg4 (Simple Profile) (FMP4 / 0x34504D46), yuv420p, 1280x1280 [SAR 1:1 DAR 1:1], 0.01 fps, 0.01 tbr, 0.01 tbn, 40 tbc
  
  Guessed Channel Layout for Input Stream #1.0 : mono
  
  Input #1, wav, from 'C:\Users\Laptop\Documents\program/audio.wav':
  
    Duration: 00:01:37.57, bitrate: 352 kb/s
  
      Stream #1:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 22050 Hz, mono, s16, 352 kb/s
  
  Output #0, avi, to 'C:\Users\Laptop\Documents\program/video.avi':
  
    Metadata:
  
      ISFT            : Lavf57.76.100
  
      Stream #0:0: Video: mpeg4 (Simple Profile) (FMP4 / 0x34504D46), yuv420p, 1280x1280 [SAR 1:1 DAR 1:1], q=2-31, 0.01 fps, 0.01 tbr, 0.01 tbn, 0.01 tbc
  
      Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 22050 Hz, mono, s16, 352 kb/s
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (copy)
  
    Stream #1:0 -> #0:1 (copy)
  
  Press [q] to stop, [?] for help
  
  frame=    1 fps=0.0 q=-1.0 Lsize=      55kB time=00:01:37.57 bitrate=   4.6kbits/s speed=1.95e+005x
  
  video:41kB audio:4kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 21.946869%

• aaccoder : rewrite PNS implementation

  6 septembre 2015, par Rostislav Pehlivanov

  aaccoder : rewrite PNS implementation
  This commit rewrites the PNS implementation and significantly
  
  improves sonic quality.
  The previous implementation marked an incredibly big amount
  
  of SFBs to predict when there was no need for this and this
  
  resulted in quite a large amount of artifacts. Also the
  
  quantization was incorrect (av_clip(4+log2f(...))) which
  
  led to 3x the intensity for PNS values leading to even more
  
  artifacts.
  This commit rewrites the PNS search function and introduces
  
  a major change : the PNS values are synthesized and are compared
  
  to the current coefficients in addition to passing through
  
  the revised checks to see whether PNS can be used.
  This decreases distortions and makes the current PNS implementation
  
  mainly focused on replacing any low-power non-zero bands as well
  
  as adding any zeroed bands back.
  The current encoder’s performance is enough (especially with
  
  IS) so PNS isn’t really required except to fill in the occasional
  
  few bands as well as extend any zeroed high frequency, so this
  
  combination which is already enabled by default works
  
  to get as much quality as it can within the bits allowed.
  Signed-off-by : Rostislav Pehlivanov <atomnuker@gmail.com>
  

  • [DH] libavcodec/aaccoder.c

• aacenc_pred : rework the way prediction is done

  29 août 2015, par Rostislav Pehlivanov

  aacenc_pred : rework the way prediction is done
  This commit completely alters the algorithm of prediction.
  
  The original commit which introduced prediction was completely
  
  incorrect to even remotely care about what the actual coefficients
  
  contain or whether any options were enabled. Not my actual fault.
  This commit treats prediction the way the decoder does and expects
  
  to do : like lossy encryption. Everything related to prediction now
  
  happens at the very end but just before quantization and encoding
  
  of coefficients. On the decoder side, prediction happens before
  
  anything has had a chance to even access the coefficients.
  Also the original implementation had problems because it actually
  
  touched the band_type of special bands which already had their
  
  scalefactor indices marked and it’s a wonder the asserion wasn’t
  
  triggered when transmitting those.
  Overall, this now drastically increases audio quality and you should
  
  think about enabling it if you don’t plan on playing anything encoded
  
  on really old low power ultra-embedded devices since they might not
  
  support decoding of prediction or AAC-Main. Though the specifications
  
  were written ages ago and as times change so do the FLOPS.
  Signed-off-by : Rostislav Pehlivanov <atomnuker@gmail.com>
  

  • [DH] libavcodec/aac.h
  • [DH] libavcodec/aaccoder.c
  • [DH] libavcodec/aacenc.c
  • [DH] libavcodec/aacenc.h
  • [DH] libavcodec/aacenc_pred.c
  • [DH] libavcodec/aacenc_pred.h