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• Uncomplete path recognition (FFmpeg) [duplicate]

  23 septembre 2022, par Francesco Battisti

  The script should download an entire playlist (only one song for this test) from YT and convert all the downloaded MP4 to MP3 :

  


  from distutils import extension
from pytube import Playlist
import os

link = input("Enter YouTube Playlist URL: ")

yt_playlist = Playlist(link)

for video in yt_playlist.videos:
    downloaded_file = video.streams.filter(only_audio=True).first().download(r"C:\Users\Francesco\Desktop\Music\JC's\+++NEW+++")
    file, extension = os.path.splitext(downloaded_file)
    # Convert video into .mp3 file
    os.system('ffmpeg -i {file}{ext} {file}.mp3'.format(file=file, ext=extension))


  


  Now, when I put the playlist's url in input, the script downloads the song but it can't convert it because :

  


  C:\Users\Francesco\Desktop\Music\JC's\+++NEW+++\Ariete: No such file or directory


  


  but the right path is :

  


  C:\Users\Francesco\Desktop\Music\JC's\+++NEW+++\Ariete - LULTIMA NOTTE Testo  Lyrics


  


  so it stops when is there a space in directory name...

  


• MP3 files created using FFmpeg are not starting playback in browser immediately. Is there any major difference between FFmpeg and AVCONV ?

  23 janvier 2019, par AR5

  I am working on a website that streams music. We recently changed server from Debian (with avconv) to a Centos7 (with FFmpeg) server.
  The mp3 files created on Debian server start playback on browser (I have tested Chrome and Firefox) start almost at the same time they start loading into the browser (I used Network tab on Developer Tools to monitor this)

  Now after the switch to Centos/FFmpeg server, the files being created on this new server are displaying a strange behavior. They only start playback after about 1MB is loaded into the browser.

  I have used identical settings for converting original file into MP3 in both AVCONV and FFmpeg but the files created using FFmpeg are showing this issue. Is there something that might be causing such an issue ? Are there differences in terms of audio conversion between AVCONV and FFmpeg ?

  I have already tried

  I first found that the files created on old server (Debian/Avconv) were VBR (variable bitrate) and the ones created on new server were CBR (constant bitrate), so I tried switching to VBR but the issue still persisted.

  I checked the mp3 files using MediaInfo app and there seems to be no difference between the files.

  I also checked if both files were being served as 206 Partial Content and they both are indeed.

  I am trying to create mp3 files using FFmpeg that work exactly like the ones created before using avconv

  I am trying to make the streaming site work on the new server but the mp3 files created using FFmpeg are not playing back correctly as compared to the ones created on the old server. I am trying to figure out what I might be doing wrong ? or if there is a difference between avconv and FFmpeg that is causing this issue.

  I am really stuck on this issue, any help will be really appreciated.

  ---

  Edit

  I don’t have access to old server anymore so I couldn’t retrieve the log output of avconv. The command that I was using was as follows :

  avconv -y -i "/test/Track 01.mp3" -ac 2 -ar 44100 -acodec libmp3lame -b:a 128k "/test/Track 01 (converted).mp3"

  Here is the command and log output from new server :

  ffmpeg -y -i "/test/Track 01.mp3" -ac 2 -ar 44100 -acodec libmp3lame -b:a 128k "/test/Track 01 (converted).mp3"
  
  ffmpeg version 2.8.15 Copyright (c) 2000-2018 the FFmpeg developers
  
    built with gcc 4.8.5 (GCC) 20150623 (Red Hat 4.8.5-28)
  
    configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --optflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector-strong --param=ssp-buffer-size=4 -grecord-gcc-switches -m64 -mtune=generic' --extra-ldflags='-Wl,-z,relro ' --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvo-amrwbenc --enable-version3 --enable-bzlib --disable-crystalhd --enable-gnutls --enable-ladspa --enable-libass --enable-libcdio --enable-libdc1394 --disable-indev=jack --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-openal --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libv4l2 --enable-libx264 --enable-libx265 --enable-libxvid --enable-x11grab --enable-avfilter --enable-avresample --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 --enable-runtime-cpudetect
  
    libavutil      54. 31.100 / 54. 31.100
  
    libavcodec     56. 60.100 / 56. 60.100
  
    libavformat    56. 40.101 / 56. 40.101
  
    libavdevice    56.  4.100 / 56.  4.100
  
    libavfilter     5. 40.101 /  5. 40.101
  
    libavresample   2.  1.  0 /  2.  1.  0
  
    libswscale      3.  1.101 /  3.  1.101
  
    libswresample   1.  2.101 /  1.  2.101
  
    libpostproc    53.  3.100 / 53.  3.100
  
  [mp3 @ 0xd60be0] Skipping 0 bytes of junk at 240044.
  
  Input #0, mp3, from '/test/Track 01.mp3':
  
    Metadata:
  
      album           : Future Hndrxx Presents: The WIZRD
  
      artist          : Future
  
      genre           : Hip-Hop
  
      title           : Never Stop
  
      track           : 1
  
      lyrics-eng      : rgf.is
  
      WEB SITE        : rgf.is
  
      TAGGINGTIME     : rgf.is
  
      WEB             : rgf.is
  
      date            : 2019
  
      encoder         : Lavf56.40.101
  
    Duration: 00:04:51.40, start: 0.025056, bitrate: 121 kb/s
  
      Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16p, 114 kb/s
  
      Metadata:
  
        encoder         : Lavc56.60
  
      Stream #0:1: Video: png, rgb24(pc), 333x333 [SAR 1:1 DAR 1:1], 90k tbr, 90k tbn, 90k tbc
  
      Metadata:
  
        comment         : Cover (front)
  
  [mp3 @ 0xd66ec0] Frame rate very high for a muxer not efficiently supporting it.
  
  Please consider specifying a lower framerate, a different muxer or -vsync 2
  
  Output #0, mp3, to '/test/Track 01 (converted).mp3':
  
    Metadata:
  
      TALB            : Future Hndrxx Presents: The WIZRD
  
      TPE1            : Future
  
      TCON            : Hip-Hop
  
      TIT2            : Never Stop
  
      TRCK            : 1
  
      lyrics-eng      : rgf.is
  
      WEB SITE        : rgf.is
  
      TAGGINGTIME     : rgf.is
  
      WEB             : rgf.is
  
      TDRC            : 2019
  
      TSSE            : Lavf56.40.101
  
      Stream #0:0: Video: png, rgb24, 333x333 [SAR 1:1 DAR 1:1], q=2-31, 200 kb/s, 90k fps, 90k tbn, 90k tbc
  
      Metadata:
  
        comment         : Cover (front)
  
        encoder         : Lavc56.60.100 png
  
      Stream #0:1: Audio: mp3 (libmp3lame), 44100 Hz, stereo, s16p, 128 kb/s
  
      Metadata:
  
        encoder         : Lavc56.60.100 libmp3lame
  
  Stream mapping:
  
    Stream #0:1 -> #0:0 (png (native) -> png (native))
  
    Stream #0:0 -> #0:1 (mp3 (native) -> mp3 (libmp3lame))
  
  Press [q] to stop, [?] for help
  
  [libmp3lame @ 0xd9b0c0] Trying to remove 1152 samples, but the queue is emptys/s
  
  frame=    1 fps=0.1 q=-0.0 Lsize=    4788kB time=00:04:51.39 bitrate= 134.6kbits/s
  
  video:234kB audio:4553kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.014809%

  Samples of MP3 files

  I have uploaded samples of mp3 files created using both avconv and FFmpeg. Please find these here : https://drive.google.com/drive/folders/1gRTmMM2iSK0VWQ4Zaf_iBNQe5laFJl08?usp=sharing

• How to stop ffmpeg from manipulating mp3 metadata ?

  13 septembre 2024, par he rambled

  I'm using ffmpeg to change bitrate of my mp3 files. It works well, but one thing is very frustrating.

  



  ffmpeg automatically changes some of metadata fields. Specifically it converts ID3v2.3 to ID3v2.4, and it does it incorrectly. For example, it writes TYER field that actually does not exist in ID3v2.4. But the most frustrating thing is, it converts USLT field to lyrics-LANGCODE(like lyrics-eng). Most of music players does not recognise this tag !

  



  I don't want ffmpeg to mess up with metadata fields. I just want it to change bitrate. Is there anyway to tell ffmpeg that it should not touch any metadata fields ?

  



  I'm running ffmpeg 4.0.2 in windows 64bit. Options are :

  



  ffmpeg -i input.mp3 -codec:a libmp3lame -b:a 128k output.mp3


  



  And no, -id3v2_version 3 did not help. It corrected TYER problem, but not lyrics problem.