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• When I append a silent audio (mp3) to an existing list of audio it garbles the final audio ?

  6 février 2020, par Marie

  After several hours I have narrowed down the issue with the garbled audio to be the 2-seconds silence audio mp3 I am appending (I think I had produced it once with Wavelab)

  However, I tried using ffmpeg according to a post to produce a similar 2 seconds audio but it too will corrupt/garble/chop voice in the final concatenation of audio files.

  ffmpeg -f lavfi -i anullsrc=r=44100:cl=mono -t 2 -q:a 9 -acodec libmp3lame SILENCE_2sec.MP3

  I typically will have several audio files to concatenate together but for simplicity I have able to narrow it to a couple of files simplifying to the following script. A simple Windows batch file you should be able to use and reproduce the issue at your end.

  rem 
  
  rem  
  
  SET EXE="S:\_BINS\FFmpeg 4.2.1 20200112\bin\ffmpeg.exe"
  
  
  
  SET ROOTPATH=.\
  
  
  
  SET IN_FILE="%ROOTPATH%MyList.txt"
  
  
  
  ECHO file '%ROOTPATH%HELLO.mp3' > MyList.txt
  
  ECHO file 'SILENCE_2sec.MP3' >> MyList.txt
  
  
  
  SET OPTIONS= -f concat -safe 0 -i  %IN_FILE%  -c copy -y
  
  
  
  SET OUT_FILE="%ROOTPATH%CONCATENATED_AUDIO_2.MP3"
  
  
  
  SET INFO_FILE="INFO.TXT"
  
  
  
  %EXE% %OPTIONS%  %OUT_FILE% 1> %INFO_FILE% 2>&1 
  
  
  
  ECHO ======================== >> %INFO_FILE%
  
  ECHO IN_FILE=%IN_FILE%  >> %INFO_FILE% 
  
  ECHO EXE=%EXE%  >> %INFO_FILE% 
  
  ECHO OPTIONS=%OPTIONS%  >> %INFO_FILE% 
  
  ECHO ======================== >> %INFO_FILE%

  Here is the console info output from the ffmpeg, let me know if you need other output include ones from ffprobe

  ffmpeg version git-2020-01-10-3d894db Copyright (c) 2000-2020 the FFmpeg developers
  
    built with gcc 9.2.1 (GCC) 20191125
  
    configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --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-libaom --enable-libmfx --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt --enable-amf
  
    libavutil      56. 38.100 / 56. 38.100
  
    libavcodec     58. 65.103 / 58. 65.103
  
    libavformat    58. 35.101 / 58. 35.101
  
    libavdevice    58.  9.103 / 58.  9.103
  
    libavfilter     7. 70.101 /  7. 70.101
  
    libswscale      5.  6.100 /  5.  6.100
  
    libswresample   3.  6.100 /  3.  6.100
  
    libpostproc    55.  6.100 / 55.  6.100
  
  [mp3 @ 000000000036af80] Estimating duration from bitrate, this may be inaccurate
  
  Input #0, concat, from '.\MyList.txt':
  
    Duration: N/A, start: 0.000000, bitrate: 32 kb/s
  
      Stream #0:0: Audio: mp3, 24000 Hz, mono, fltp, 32 kb/s
  
  Output #0, mp3, to '.\CONCATENATED_AUDIO_2.MP3':
  
    Metadata:
  
      TSSE            : Lavf58.35.101
  
      Stream #0:0: Audio: mp3, 24000 Hz, mono, fltp, 32 kb/s
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (copy)
  
  Press [q] to stop, [?] for help
  
  [mp3 @ 0000000000372d00] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 17280 >= 17255
  
  size=      11kB time=00:00:02.73 bitrate=  33.2kbits/s speed=2.73e+03x    
  
  video:0kB audio:11kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 2.137446%
  
  ======================== 
  
  IN_FILE=".\MyList.txt"   
  
  EXE="S:\_BINS\FFmpeg 4.2.1 20200112\bin\ffmpeg.exe"   
  
  OPTIONS= -f concat -safe 0 -i  ".\MyList.txt"  -c copy -y   
  
  ========================  

  I believe I am running FFmpeg 4.2.1, recently installed (20200112)

  You may produce the HELLO.mp3 by saving the following link

  https://translate.google.com.vn/translate_tts?en=UTF-8&q=Hello+&tl=en&client=tw-ob

  FYI, I am still a novice of ffmpeg and using it more like a black box with the help I received in this very super forum.
  Please be as explicit as you can with command line options on how I can fix this issue.
  Thank you.

  Additional Hints Debugging :

  If I append more files after the silence audio it seems that the silence audio impacts (garbles, chops) the previous audio.
  You may try the following for the list of audio files input.

  ECHO file '%ROOTPATH%HELLO.mp3' > MyList.txt
  
  ECHO file 'SILENCE_2sec.MP3' >> MyList.txt
  
  ECHO file '%ROOTPATH%HELLO.mp3' >> MyList.txt
  
  ECHO file '%ROOTPATH%HELLO.mp3' >> MyList.txt

  I typically add one or more silence file to derive a post silence effect after the actual audio. That’s my current logic. However if you have an alternative to appending a silence in the process of concatenating several audio files or appending x-seconds silence to an existing audio file. I can use that method as well from my coding.

  Thank you.

• Picture size 0x0 is invalid : trying to downsample 4k raw video into 1080 using ffmpeg

  25 février 2021, par neetu sigger

  ffmpeg -i CityAlley_3840x2160_50fps_8bit.yuv -vf scale=1920:1080 CityAlly.yuv


  


  But I got the error :

  


  ffmpeg version 4.2.4-1ubuntu0.1 Copyright (c) 2000-2020 the FFmpeg developers
  built with gcc 9 (Ubuntu 9.3.0-10ubuntu2)
  configuration: --prefix=/usr --extra-version=1ubuntu0.1 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --arch=amd64 --enable-gpl --disable-stripping --enable-avresample --disable-filter=resample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libcodec2 --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librsvg --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-lv2 --enable-omx --enable-openal --enable-opencl --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-nvenc --enable-chromaprint --enable-frei0r --enable-libx264 --enable-shared
  libavutil      56. 31.100 / 56. 31.100
  libavcodec     58. 54.100 / 58. 54.100
  libavformat    58. 29.100 / 58. 29.100
  libavdevice    58.  8.100 / 58.  8.100
  libavfilter     7. 57.100 /  7. 57.100
  libavresample   4.  0.  0 /  4.  0.  0
  libswscale      5.  5.100 /  5.  5.100
  libswresample   3.  5.100 /  3.  5.100
  libpostproc    55.  5.100 / 55.  5.100
[IMGUTILS @ 0x7fffc0509b50] Picture size 0x0 is invalid
CityAlley_3840x2160_50fps_8bit.yuv: Invalid argument


  


• Manim error : It's a sound problem that I haven't seen anywhere else

  2 avril 2024, par Nεo Pλατo

  I've been using Manim for about a month now, and with already existing impediments to my progress none has been more painful than the lack of sound when I got to this part of Theorem of Beethoven's tutorial. I've been struggling with it for three weeks and opened issues on GitHub and Reddit without any success.

  


  So the code looks like this :

  


  class AudioTest(Scene):
    def construct(self):
        group_dots=VGroup(*[Dot()for _ in range(3)])
        group_dots.arrange_submobjects(RIGHT)
        for dot in group_dots:
            self.add_sound("click_this_and_nothing_else",gain=-10)
            self.add(dot)
            self.wait()
            self.wait()


  


  And the error message is a bit long and I'm still cracking at what it actually means. Here it is :

  


  C:\Manim\manim-18june>python manim.py tutorial\7_add_audio.py AudioTest -pm
Media will be written to C:/Users/NeoPlato/Videos/manim_videos\. You can change this behavior with the --media_dir flag.


Traceback (most recent call last):
  File "C:\Manim\manim-18june\manimlib\extract_scene.py", line 155, in main
    scene = SceneClass(**scene_kwargs)
  File "C:\Manim\manim-18june\manimlib\scene\scene.py", line 79, in __init__
    self.file_writer.finish()
  File "C:\Manim\manim-18june\manimlib\scene\scene_file_writer.py", line 379, in finish
    self.combine_movie_files()
  File "C:\Manim\manim-18june\manimlib\scene\scene_file_writer.py", line 506, in combine_movie_files
    self.audio_segment.export(
  File "C:\Users\allan\AppData\Local\Programs\Python\Python38-32\lib\site-packages\pydub\audio_segment.py", line 911, in export
    raise CouldntEncodeError(
pydub.exceptions.CouldntEncodeError: Encoding failed. ffmpeg/avlib returned error code: 1

Command:['ffmpeg', '-y', '-f', 'wav', '-i', 'C:\\Users\\NEOPLATO~1.000\\AppData\\Local\\Temp\\tmpk6j8yj98', '-b:a', '312k', '-f', 'mp3', 'C:\\Users\\NEOPLATO~1.000\\AppData\\Local\\Temp\\tmpyn225uxk']

Output from ffmpeg/avlib:

ffmpeg version git-2020-06-17-0b3bd00 Copyright (c) 2000-2020 the FFmpeg developers
  built with gcc 9.3.1 (GCC) 20200523
  configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libsrt --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-libvmaf --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --disable-w32threads --enable-libmfx --enable-ffnvcodec --enable-cuda-llvm --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt
--enable-amf
  libavutil      56. 55.100 / 56. 55.100
  libavcodec     58. 92.100 / 58. 92.100
  libavformat    58. 46.101 / 58. 46.101
  libavdevice    58. 11.100 / 58. 11.100
  libavfilter     7. 86.100 /  7. 86.100
  libswscale      5.  8.100 /  5.  8.100
  libswresample   3.  8.100 /  3.  8.100
  libpostproc    55.  8.100 / 55.  8.100
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, wav, from 'C:\Users\NEOPLATO~1.000\AppData\Local\Temp\tmpk6j8yj98':
  Duration: 00:00:04.23, bitrate: 2822 kb/s
    Stream #0:0: Audio: pcm_s32le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s32, 2822 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s32le (native) -> mp3 (mp3_mf))
Press [q] to stop, [?] for help
[mp3_mf @ 011be000] could not find any MFT for the given media type
[mp3_mf @ 011be000] could not create MFT
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 do have a larger documentation of the problem with current progress [over at GitHub](https://github.com/3b1b/manim/issues/1152)

>Help
>>Me
>>>Please :(
>>>>Pardon my naive looking formatting, I'm new to Markdown