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• FFmpeg's common filter "v360" missing

  26 septembre 2021, par Niklas E.

  I'm trying to convert a 180° fisheye video to a normal/regular video using the v360 filter of FFmpeg.

  


  This is the command I tried :
  
ffmpeg -i in.mp4 -vf "v360=input=fisheye:output=flat:iv_fov=180:v_fov=90" out.mp4

  


  But the output says clearly No such filter: 'v360', although v360 is a common filter listed in docs and other filters I used before worked just fine. I tried updating/reinstalling and looking for solutions, not fixing it.

  


  Why is the filter missing ? How can I debug this ? Should I doe the task using another program entirely ?

  


  Command output :

  


  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
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'in.mp4':
  Metadata:
    major_brand     : mp42
    minor_version   : 0
    compatible_brands: mp42mp41
    creation_time   : 2021-09-11T14:18:33.000000Z
  Duration: 00:02:48.02, start: 0.000000, bitrate: 26056 kb/s
    Stream #0:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, bt709), 2160x1080 [SAR 1:1 DAR 2:1], 25924 kb/s, 50 fps, 50 tbr, 50k tbn, 100 tbc (default)
    Metadata:
      creation_time   : 2021-09-11T14:18:33.000000Z
      handler_name    : Mainconcept MP4 Video Media Handler
      encoder         : AVC Coding
    Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 125 kb/s (default)
    Metadata:
      creation_time   : 2021-09-11T14:18:33.000000Z
      handler_name    : Mainconcept MP4 Sound Media Handler
Stream mapping:
  Stream #0:0 -> #0:0 (h264 (native) -> h264 (libx264))
  Stream #0:1 -> #0:1 (aac (native) -> aac (native))
Press [q] to stop, [?] for help
[AVFilterGraph @ 0x55ee57567340] No such filter: 'v360'
Error reinitializing filters!
Failed to inject frame into filter network: Invalid argument
Error while processing the decoded data for stream #0:0
Conversion failed!


  


• Start and end time of MoviePy's VideoClip not working

  21 mars 2024, par ernesto casco velazquez

  I'm trying to add captions to a video. The desired outcome is to show each word in the exact moment is being said.

  


  I have a method that gives me the accurate time start and end per each word :

  


  def get_words_per_time(audio_speech_file):
    model = whisper.load_model("base")
    transcribe = model.transcribe(
        audio=audio_speech_file, fp16=False, word_timestamps=True
    )
    segments = transcribe["segments"]
    words = []

    for seg in segments:
        for word in seg["words"]:
            words.append(
                {
                    "word": word["word"],
                    "start": word["start"],
                    "end": word["end"],
                    "prob": round(word["probability"], 4),
                }
            )
    return words


  


  Then I have a code that uses MoviePy to create TextClip and assing a given start and end time per pair of words (I know there are redundant statements, srry) :

  


  def generate_captions(
    words,
    font="Komika",
    fontsize=32,
    color="White",
    align="center",
    stroke_width=3,
    stroke_color="black",
):
    text_comp = []
    for i in track(range(0, len(words), 2), description="Creating captions..."):
        word1 = words[i]
        if i + 1 < len(words):
            word2 = words[i + 1]
        text_clip = TextClip(
            f"{word1['word']} {word2['word'] if i + 1 < len(words) else ''}",
            font=font,  # Change Font if not found
            fontsize=fontsize,
            color=color,
            align=align,
            method="caption",
            size=(660, None),
            stroke_width=stroke_width,
            stroke_color=stroke_color,
        )
        text_clip = text_clip.set_start(word1["start"])
        text_clip = text_clip.set_end(
            word2["end"] if i + 1 < len(words) else word1["end"]
        )
        text_comp.append(text_clip)
    return text_comp


  


  Finally, I concatenate the words into a single video :

  


  vid_clip = CompositeVideoClip(
    [vid_clip, concatenate_videoclips(text_comp).set_position(("center", 860))]
)


  


  The output is this, but you can clearly see the words are not flowing with the speech. They somehow move faster as if the start/end time did not matter. Here's the video

  


  The words with their respective start/end time, look like this :

  


  [
    {
        'word': 'This',
        'start': 0.0,
        'end': 0.22,
        'prob': 0.805
    },
    {
        'word': 'is',
        'start': 0.22,
        'end': 0.42,
        'prob': 0.9991
    },
    {
        'word': 'a',
        'start': 0.42,
        'end': 0.6,
        'prob': 0.999
    },
    {
        'word': 'test,
        ',
        'start': 0.6,
        'end': 1.04,
        'prob': 0.9939
    },
    {
        'word': 'to',
        'start': 1.18,
        'end': 1.3,
        'prob': 0.9847
    },
    {
        'word': 'show',
        'start': 1.3,
        'end': 1.54,
        'prob': 0.9971
    },
    {
        'word': 'words',
        'start': 1.54,
        'end': 1.9,
        'prob': 0.995
    },
    {
        'word': 'does',
        'start': 1.9,
        'end': 2.16,
        'prob': 0.997
    },
    {
        'word': 'not',
        'start': 2.16,
        'end': 2.4,
        'prob': 0.9978
    },
    {
        'word': 'appear.',
        'start': 2.4,
        'end': 2.82,
        'prob': 0.9984
    },
    {
        'word': 'At',
        'start': 3.46,
        'end': 3.6,
        'prob': 0.9793
    },
    {
        'word': 'their',
        'start': 3.6,
        'end': 3.8,
        'prob': 0.9984
    },
    {
        'word': 'proper',
        'start': 3.8,
        'end': 4.22,
        'prob': 0.9976
    },
    {
        'word': 'time.',
        'start': 4.22,
        'end': 4.72,
        'prob': 0.999
    },
    {
        'word': 'Thanks',
        'start': 5.04,
        'end': 5.4,
        'prob': 0.9662
    },
    {
        'word': 'for,
        ',
        'start': 5.4,
        'end': 5.66,
        'prob': 0.9941
    },
    {
        'word': 'watching.',
        'start': 5.94,
        'end': 6.36,
        'prob': 0.7701
    }
]


  


  What could be causing this ?

  


• ffmpeg silenceremove - hear what bits are removed

  7 avril 2020, par jimo

  ffmpeg silenceremove is pretty cool. im loving it. i can trim 3 second silences to 2 seconds and reduce a 1.5 hour file of spoken audio down 3 or 4 minutes (depending on the speaker).

  



  once in a while I do hear my choice for stop_threshold (ie-40dB on audio only analog file) does cause the end of a word to be clipped, just here and there when the speaker trails off softly at the end of the word.

  



  is there any way to output what is trimmed to a file ? so I can listen to it and get an idea of just how often this word clipping happens ?

  



  thanks !