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• Mkv to Mp4 Converion using ffmpeg giving error

  17 avril 2019, par ashok patel

  I recently started to use Sony Vegas 13 which doesn’t accept MKV(mkv source file :-https://www.appsloveworld.com/2018/10/sample-mkv-video.html) file types so I decided to install FFMPEG to convert a MKV to MP4 which people say has almost no loss in quality, etc. However, when I try to convert a 900MB MKV file it returns a 0kb MP4 file.Image

  The line of code I use is "ffmpeg -i set.mkv -codec copy set.mp4"

  This is the error ? message I get
  error by ffmpeg

• add subtitle to video created from mp3 file and an image using ffmpeg

  1er février 2018, par no name

  I have a simple image and an mp3 file all I want to do is to add a lyrics to this mp3 file in a video as an output with fade out style.
  so i used this command to do so from this this blog
  

  ffmpeg -loop 1 -y -i test.jpg -i test.mp3 -vf subtitles=”f=test.srt:force_style=’FontName=Arial,FontSize=14″ -shortest test1.mp4

  when i used this command i got this eror

      ffmpeg -loop 1 -y -i test.jpg -i test.mp3 -vf subtitles="f=test.srt:force_style=’FontName=Arial,FontSize=14″ -shortest test1.mp4
  
  ffmpeg version N-89940-gb1af0e23a3 Copyright (c) 2000-2018 the FFmpeg developers
  
    built with gcc 7.2.0 (GCC)
  
    configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-bzlib --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libmfx --enable-amf --enable-cuda --enable-cuvid --enable-d3d11va --enable-nvenc --enable-dxva2 --enable-avisynth
  
    libavutil      56.  7.100 / 56.  7.100
  
    libavcodec     58.  9.100 / 58.  9.100
  
    libavformat    58.  7.100 / 58.  7.100
  
    libavdevice    58.  0.101 / 58.  0.101
  
    libavfilter     7. 11.101 /  7. 11.101
  
    libswscale      5.  0.101 /  5.  0.101
  
    libswresample   3.  0.101 /  3.  0.101
  
    libpostproc    55.  0.100 / 55.  0.100
  
  Trailing options were found on the commandline.
  
  Input #0, image2, from 'test.jpg':
  
    Duration: 00:00:00.04, start: 0.000000, bitrate: 17290 kb/s
  
      Stream #0:0: Video: mjpeg, yuvj420p(pc, bt470bg/unknown/unknown), 960x959 [SAR 1:1 DAR 960:959], 25 fps, 25 tbr, 25 tbn, 25 tbc
  
  Input #1, mp3, from 'test.mp3':
  
    Metadata:
  
      comment         : converted by convert2mp3.net
  
      title           : Celestial Empire [ Worlds 2017 ] [ Pick and b
  
      artist          : Raffael Gruber Matthias Ullrich
  
    Duration: 00:02:33.13, start: 0.025057, bitrate: 193 kb/s
  
      Stream #1:0: Audio: mp3, 44100 Hz, stereo, s16p, 192 kb/s
  
      Metadata:
  
        encoder         : Lavc57.28
  
  At least one output file must be specified

  iam using windows 10, last version of ffmpeg, so what makes this problem. also how to make the style of the lyrics to be something like fade in/out and so on. if it’s possible
  Thanks in advance.
  update
  when i use this command

  ffmpeg -loop 1 -y -i test.jpg -i test.mp3 -vf "subtitles=test.srt:force_style='FontName=Arial,FontSize=14'" -shortest test1.mp4

  it give me this error

  Press [q] to stop, [?] for help
  
  [Parsed_subtitles_0 @ 049af980] Shaper: FriBidi 0.19.7 (SIMPLE)
  
  [Parsed_subtitles_0 @ 049af980] Using font provider directwrite
  
  [libx264 @ 049d2440] height not divisible by 2 (960x959)
  
  Error initializing output stream 0:0 -- Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height
  
  Conversion failed!

  i searched this error and i found a solution by add this part before output file so the final command is

  ffmpeg -loop 1 -y -i test.jpg -i test.mp3 -vf "subtitles=test.srt:force_style='FontName=Arial,FontSize=14'" -shortest -vf "scale=trunc(iw/2)*2:trunc(ih/2)*2" test1.mp4

  i got no error and it worked but the output video not contains any lyrics which i added

  Update
  
  last thing i tried and it works first i use this command to convert the image and mp3 file to a mp4 video

  ffmpeg -loop 1 -i test.jpg -i test.mp3 -shortest -acodec copy -vf "scale=trunc(iw/2)*2:trunc(ih/2)*2" video.mp4

  but this command take a long time to process to convert mp3 of 2:55 minute it with image.
  then i start to add lyrics or subtitle using this command

  ffmpeg -i video.mp4 -vf "subtitles=test.srt:force_style='FontName=Arial,FontSize=14,PrimaryColour=&H0000ff&'" -c:a copy output.mp4

  this worked perfect and give the output but it take a long time to process i don’t know if this normal.
  now last thing how can i set the style of the subtitle so it be fade out or something

• How can I convert audio to WAVE_FORMAT_PCM using FFmpeg ?

  16 juin, par Eric Stdlib

  I am using Python's wave module to read audio, and using FFmpeg to convert audio from other types to wav. However, I am encountering some problem.

  



  I wrote v.py to generate an silence audio file a.wav

  



  import sys, wave, math
import numpy as np

wave_data = np.zeros(44100).astype(np.short)

f = wave.open('a.wav', 'wb')
f.setnchannels(1)
f.setsampwidth(2)
f.setframerate(96000)
f.writeframes(wave_data.tostring())
f.close()


  



  Then I used FFmpeg to "copy" a.wav to b.wav (though it seems to encode / decode the file), but I can only read a.wav with Python ; b.wav cannot be opened.

  



  [user@localhost tmp]$ ffmpeg -i a.wav b.wav&#xA;Guessed Channel Layout for Input Stream #0.0 : mono&#xA;Input #0, wav, from &#x27;a.wav&#x27;:&#xA;  Duration: 00:00:00.46, bitrate: 1536 kb/s&#xA;    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 96000 Hz, mono, s16, 1536 kb/s&#xA;Stream mapping:&#xA;  Stream #0:0 -> #0:0 (pcm_s16le (native) -> pcm_s16le (native))&#xA;Press [q] to stop, [?] for help&#xA;Output #0, wav, to &#x27;b.wav&#x27;:&#xA;  Metadata:&#xA;    ISFT            : Lavf57.71.100&#xA;    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 96000 Hz, mono, s16, 1536 kb/s&#xA;    Metadata:&#xA;      encoder         : Lavc57.89.100 pcm_s16le&#xA;size=      86kB time=00:00:00.45 bitrate=1537.8kbits/s speed= 706x    &#xA;video:0kB audio:86kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.115646%&#xA;[user@localhost tmp]$ python3&#xA;Python 3.6.4 (default, Jan 23 2018, 22:25:37) &#xA;[GCC 7.2.1 20170915 (Red Hat 7.2.1-2)] on linux&#xA;Type "help", "copyright", "credits" or "license" for more information.&#xA;>>> import wave&#xA;>>> wave.open(&#x27;a.wav&#x27;)&#xA;&#xA;>>> wave.open(&#x27;b.wav&#x27;)&#xA;Traceback (most recent call last):&#xA;  File "<stdin>", line 1, in <module>&#xA;  File "/usr/lib64/python3.6/wave.py", line 499, in open&#xA;    return Wave_read(f)&#xA;  File "/usr/lib64/python3.6/wave.py", line 163, in __init__&#xA;    self.initfp(f)&#xA;  File "/usr/lib64/python3.6/wave.py", line 143, in initfp&#xA;    self._read_fmt_chunk(chunk)&#xA;  File "/usr/lib64/python3.6/wave.py", line 260, in _read_fmt_chunk&#xA;    raise Error(&#x27;unknown format: %r&#x27; % (wFormatTag,))&#xA;wave.Error: unknown format: 65534&#xA;>>> &#xA;</module></stdin>

  



  How should I change the command of FFmpeg to convert the file to WAVE_FORMAT_PCM, so that I can read b.wav with Python ?