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• ffmpeg - convert MP4 to Panasonic Lumix playable MJPEG [closed]

  21 août 2024, par Maro Natts

  I'm new to ffmpeg and ffprobe, I started using it to convert videos to show on 3DS and DSi for an art project.

  


  Now I want to show my own videos on a Panasonic LUMIX digicam (DMC-FT2)(JPN), but have had trouble getting the device to play the video.

  


  The digi takes videos in AVCHD Lite and Motion JPEG.

  


  I've tried online converting mp4 to the respective formats + changing file name, but the device only recognises that it's there — refuses to/can't play it.

  


  I also tried using this code to convert to MJPEG :
ffmpeg -i input.mp4 -c:v mjpeg -q:v 2 -an output.mjpeg

  


  But no luck ! Changed the filename to match the others, the camera recognised the file is there, but can't play it.

  


  So I used this code to get the information of a MJPEG video taken on the digi :
/opt/ffmpeg/ffprobe file.mp4 -show_streams  -select_streams v  -print_format json

  


  and got the following, is it possible to convert my own video (mp4) to these settings ? So that it can play on the device ? If so how ?

  


  Thanks !

  


    Metadata:
    major_brand     : qt  
    minor_version   : 537331972
    compatible_brands: qt  pana
    creation_time   : 2024-08-21T21:54:59.000000Z
  Duration: 00:00:06.00, start: 0.000000, bitrate: 12287 kb/s
  Stream #0:0[0x1](eng): Video: mjpeg (Baseline) (jpeg / 0x6765706A), yuvj420p(pc, bt470bg/unknown/unknown), 640x480, 11083 kb/s, 30 fps, 30 tbr, 30 tbn (default)
      Metadata:
        creation_time   : 2024-08-21T21:54:59.000000Z
        vendor_id       : pana
        encoder         : Photo - JPEG
  Stream #0:1[0x2](eng): Audio: pcm_s16be (twos / 0x736F7774), 16000 Hz, 1 channels, s16, 256 kb/s (default)
      Metadata:
        creation_time   : 2024-08-21T21:54:59.000000Z
        vendor_id       : pana
    "streams": [
        {
            "index": 0,
            "codec_name": "mjpeg",
            "codec_long_name": "Motion JPEG",
            "profile": "Baseline",
            "codec_type": "video",
            "codec_tag_string": "jpeg",
            "codec_tag": "0x6765706a",
            "width": 640,
            "height": 480,
            "coded_width": 640,
            "coded_height": 480,
            "closed_captions": 0,
            "film_grain": 0,
            "has_b_frames": 0,
            "pix_fmt": "yuvj420p",
            "level": -99,
            "color_range": "pc",
            "color_space": "bt470bg",
            "chroma_location": "center",
            "refs": 1,
            "id": "0x1",
            "r_frame_rate": "30/1",
            "avg_frame_rate": "30/1",
            "time_base": "1/30",
            "start_pts": 0,
            "start_time": "0.000000",
            "duration_ts": 180,
            "duration": "6.000000",
            "bit_rate": "11083093",
            "bits_per_raw_sample": "8",
            "nb_frames": "180",
            "disposition": {
                "default": 1,
                "dub": 0,
                "original": 0,
                "comment": 0,
                "lyrics": 0,
                "karaoke": 0,
                "forced": 0,
                "hearing_impaired": 0,
                "visual_impaired": 0,
                "clean_effects": 0,
                "attached_pic": 0,
                "timed_thumbnails": 0,
                "non_diegetic": 0,
                "captions": 0,
                "descriptions": 0,
                "metadata": 0,
                "dependent": 0,
                "still_image": 0
            },
            "tags": {
                "creation_time": "2024-08-21T21:54:59.000000Z",
                "language": "eng",
                "vendor_id": "pana",
                "encoder": "Photo - JPEG"
            }
        }
    ]
}



  


• Error audio loading when runing Whisper Open AI model

  9 juin 2024, par John mick

  The problem I'm trying to solve is that I can't run Whisper model for some audio, it says something related to audio decoding.

  


  payload.wav: Invalid data found when processing input.
raise RuntimeError(f"Failed to load audio: {e.stderr.decode()}") from e


  


  I tried using the micro-machines.wav and it works fine but when i used other audio it gives me an error.

  


  import whisper

model = whisper.load_model("base")
text=model.transcribe('micro-machines.wav',fp16=False)
print(text)
text=model.transcribe('payload.wav',fp16=False)
print(text)


  


  Error I'm getting for payload :

  


  d:\...\venv\lib\site-packages\whisper\transcribe.py:79: UserWarning: FP16 is not supported on CPU; using FP32 instead&#xA;  warnings.warn("FP16 is not supported on CPU; using FP32 instead")                                                                                        &#xA;Traceback (most recent call last):&#xA;  File "d:\...\venv\lib\site-packages\whisper\audio.py", line 42, in load_audio&#xA;    ffmpeg.input(file, threads=0)                                                                                    &#xA;  File "d:\...\venv\lib\site-packages\ffmpeg\_run.py", line 325, in run        &#xA;    raise Error(&#x27;ffmpeg&#x27;, out, err)                                                                                  &#xA;ffmpeg._run.Error: ffmpeg error (see stderr output for detail)                                                       &#xA;&#xA;The above exception was the direct cause of the following exception:&#xA;&#xA;Traceback (most recent call last):&#xA;  File "C:\....\Python\Python39\lib\runpy.py", line 197, in _run_module_as_main&#xA;    return _run_code(code, main_globals, None,&#xA;  File "C:\.....\Python\Python39\lib\runpy.py", line 87, in _run_code&#xA;    exec(code, run_globals)&#xA;  File "D:\...\venv\Scripts\whisper.exe\__main__.py", line 7, in <module>&#xA;  File "d:\...\venv\lib\site-packages\whisper\transcribe.py", line 314, in cli&#xA;    result = transcribe(model, audio_path, temperature=temperature, **args)&#xA;  File "d:\...\venv\lib\site-packages\whisper\transcribe.py", line 85, in transcribe&#xA;    mel = log_mel_spectrogram(audio)&#xA;  File "d:\...\venv\lib\site-packages\whisper\audio.py", line 111, in log_mel_spectrogram&#xA;    audio = load_audio(audio)&#xA;  File "d:\...\venv\lib\site-packages\whisper\audio.py", line 47, in load_audio&#xA;    raise RuntimeError(f"Failed to load audio: {e.stderr.decode()}") from e&#xA;RuntimeError: Failed to load audio: ffmpeg version 6.0-essentials_build-www.gyan.dev Copyright (c) 2000-2023 the FFmpeg developers&#xA;  built with gcc 12.2.0 (Rev10, Built by MSYS2 project)&#xA;  configuration: --enable-gpl --enable-version3 --enable-static --disable-w32threads --disable-autodetect --enable-fontconfig --enable-iconv --enable-gnutls --enable-libxml2 --enab&#xA;le-gmp --enable-lzma --enable-zlib --enable-libsrt --enable-libssh --enable-libzmq --enable-avisynth --enable-sdl2 --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxv&#xA;id --enable-libaom --enable-libopenjpeg --enable-libvpx --enable-libass --enable-libfreetype --enable-libfribidi --enable-libvidstab --enable-libvmaf --enable-libzimg --enable-amf &#xA;--enable-cuda-llvm --enable-cuvid --enable-ffnvcodec --enable-nvdec --enable-nvenc --enable-d3d11va --enable-dxva2 --enable-libmfx --enable-libgme --enable-libopenmpt --enable-libo&#xA;pencore-amrwb --enable-libmp3lame --enable-libtheora --enable-libvo-amrwbenc --enable-libgsm --enable-libopencore-amrnb --enable-libopus --enable-libspeex --enable-libvorbis --enab&#xA;le-librubberband&#xA;  libavutil      58.  2.100 / 58.  2.100&#xA;  libavcodec     60.  3.100 / 60.  3.100&#xA;  libavformat    60.  3.100 / 60.  3.100&#xA;  libavdevice    60.  1.100 / 60.  1.100&#xA;  libavfilter     9.  3.100 /  9.  3.100&#xA;  libswscale      7.  1.100 /  7.  1.100&#xA;  libswresample   4. 10.100 /  4. 10.100&#xA;  libpostproc    57.  1.100 / 57.  1.100&#xA;payload.wav: Invalid data found when processing input&#xA;</module>

  


  I tried searching for solutions and I found one which says It appears that the code failed to load the audio file for some reason and even failed to display that error because e.stderr did not contain a valid UTF-8 string

  


• configure : Separate subsystem for Immersive Audio Model

  3 avril 2024, par Eugene Zemtsov

  configure : Separate subsystem for Immersive Audio Model
  This change allows users to build libavfomat without support
  
  for Immersive Audio Model by specifying —disable-iamf.
  
  It helps to save on binary size in cases where it's important.
  Co-authored-by : James Almer <jamrial@gmail.com>
  
  Signed-off-by : Eugene Zemtsov <eugene@chromium.org>
  
  Signed-off-by : James Almer <jamrial@gmail.com>
  

  • [DH] configure
  • [DH] libavformat/mov.c
  • [DH] libavformat/movenc.c