Recherche avancée

Sur d’autres sites (7022)

• M4a (mp4) audio file encoded with pydub+ffmpeg doesn't play on Android

  6 juillet 2020, par Evelyn

  I have a python script to split up some wav files and export to m4a using pydub. I'm able to get these files to play on several devices, but not on an Android device (using Google Pixel 3). When I try encoding with straight ffmpeg in terminal, that works fine on the Android device.

  


  What is the difference in these two files, and since pydub is using ffmpeg, what do I need to change to make it do exactly the same as the ffmpeg command ?

  


  Not working

  


  from pydub import AudioSegment
>>> audio = AudioSegment.from_wav("input.wav")
>>> slice = audio[1000:3000]
>>> slice.export("pydub_export.m4a", format="mp4", parameters=["-ac", "1", "-c:a", "libfdk_aac", "-profile:a", "aac_he", "-vbr", "2"])


  


  mediainfo output :

  


  General
Complete name                            : pydub_export.m4a
Format                                   : MPEG-4
Format profile                           : Base Media
Codec ID                                 : isom (isom/iso2/mp41)
File size                                : 11.7 KiB
Duration                                 : 2 s 115 ms
Overall bit rate mode                    : Constant
Overall bit rate                         : 45.1 kb/s
Writing application                      : Lavf58.29.100

Audio
ID                                       : 1
Format                                   : AAC LC SBR
Format/Info                              : Advanced Audio Codec Low Complexity with Spectral Band Replication
Commercial name                          : HE-AAC
Format settings                          : NBC
Codec ID                                 : mp4a-40-5
Duration                                 : 2 s 115 ms
Duration_LastFrame                       : -22 ms
Bit rate mode                            : Constant
Bit rate                                 : 41.4 kb/s
Channel(s)                               : 1 channel
Channel layout                           : C
Sampling rate                            : 44.1 kHz
Frame rate                               : 21.533 FPS (2048 SPF)
Compression mode                         : Lossy
Stream size                              : 10.7 KiB (92%)
Default                                  : Yes
Alternate group                          : 1


  


  Working

  


  $ffmpeg -i input.wav -acodec copy -ss 1 -to 3 input_slice.wav
$ffmpeg -i input_slice.wav -ac 1 -c:a libfdk_aac -profile:a aac_he -vbr 2 ffmpeg_export.m4a


  


  mediainfo output :

  


  General
Complete name                            : ffmpeg_export.m4a
Format                                   : MPEG-4
Format profile                           : Apple audio with iTunes info
Codec ID                                 : M4A  (isom/iso2)
File size                                : 11.2 KiB
Duration                                 : 2 s 112 ms
Overall bit rate mode                    : Constant
Overall bit rate                         : 43.6 kb/s
Writing application                      : Lavf58.29.100

Audio
ID                                       : 1
Format                                   : AAC LC SBR
Format/Info                              : Advanced Audio Codec Low Complexity with Spectral Band Replication
Commercial name                          : HE-AAC
Format settings                          : NBC
Codec ID                                 : mp4a-40-5
Duration                                 : 2 s 112 ms
Duration_LastFrame                       : -25 ms
Bit rate mode                            : Constant
Bit rate                                 : 39.9 kb/s
Channel(s)                               : 1 channel
Channel layout                           : C
Sampling rate                            : 44.1 kHz
Frame rate                               : 21.533 FPS (2048 SPF)
Compression mode                         : Lossy
Stream size                              : 10.3 KiB (91%)
Default                                  : Yes
Alternate group                          : 1


  


  I already tried moving metadata to the front with -movflags faststart on the broken file and it didn't make a difference.

  


• Play playlist of audio on website using node to create a jukebox / radio app

  1er juillet 2022, par neff

  So I have some time on my hands and thought I would make myself a little jukebox / radio type app.

  



  It would be fairly simple, just a collection of MP3's on the server, one is chosen at random, it plays, on completion, the next one is chosen and plays. The front of this would just be a super simple page that has a player and displays the metadata.

  



  I don't really have any experience with server programming but I'm going to look in to Node, seems like it would be good for this. I've already written a little script in Python that chooses a song from a selection and plays it (using VLC at the moment) so it should be simple to port it to Node / js.

  



  Just wondering if someone could point me in the right direction for how to link the "player" with the "playlist".

  



  Looking in to it, I can only find solutions involving a client and server using shoutCast or ICEcast or similar - so the playlist streams audio to a shoutcast server, and the website is just a player looking at the shoutCast URL - that seems unnecessary for me, as the streaming and the site would be the same thing.

  



  New to a lot of this :) but I have time at the moment so happy to get stuck in !

  



  Thanks in advance

  


• FFMPEG output truncated when adding watermark

  17 juin 2020, par user1452030

  I'm using the following command to generate a watermarked 320Kbps MP3 preview of a wav file :

  



  ffmpeg -i /path/input.wav -y -filter_complex "amovie=/path/wm_padded.wav:loop=0,asetpts=N/SR/TB,adelay=3000|3000[beep];[0][beep]amix=duration=shortest,volume=2"  -b:a 320k  /path/preview.mp3


  



  (wm_padded.wav is the watermark file padded to 10 seconds and I'm using a Mac. The command was structured based on this post)

  



  While this works as expected at times, it produces a short, garbled preview some other times. Any help in debugging this issue would be greatly appreciated. I've verified the input file and it seems to be fine and here's the FFMPEG command output :

  



      $ ffmpeg -i /path/input.wav -y -filter_complex "amovie=/path/wm_padded.wav:loop=0,asetpts=N/SR/TB,adelay=3000|3000[beep];[0][beep]amix=duration=shortest,volume=2"  -b:a 320k -vsync 2 /path/test-Preview.mp3
ffmpeg version 4.2.1 Copyright (c) 2000-2019 the FFmpeg developers
  built with Apple clang version 11.0.0 (clang-1100.0.33.8)
  configuration: --prefix=/usr/local/Cellar/ffmpeg/4.2.1_2 --enable-shared --enable-pthreads --enable-version3 --enable-avresample --cc=clang --host-cflags='-I/Library/Java/JavaVirtualMachines/adoptopenjdk-13.jdk/Contents/Home/include -I/Library/Java/JavaVirtualMachines/adoptopenjdk-13.jdk/Contents/Home/include/darwin -fno-stack-check' --host-ldflags= --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libbluray --enable-libmp3lame --enable-libopus --enable-librubberband --enable-libsnappy --enable-libtesseract --enable-libtheora --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-libxvid --enable-lzma --enable-libfontconfig --enable-libfreetype --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librtmp --enable-libspeex --enable-libsoxr --enable-videotoolbox --disable-libjack --disable-indev=jack
  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
[wav @ 0x7ff2ba801a00] Discarding ID3 tags because more suitable tags were found.
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, wav, from '/path/input.wav':
  Metadata:
    comment         :  motion graphics, motion, textures
    time_reference  : 0
    coding_history  : 
  Duration: 00:00:01.75, bitrate: 1520 kb/s
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
    Stream #0:1: Video: mjpeg (Baseline), yuvj420p(pc, bt470bg/unknown/unknown), 288x288 [SAR 72:72 DAR 1:1], 90k tbr, 90k tbn, 90k tbc (attached pic)
    Metadata:
      comment         : Cover (front)
[Parsed_amovie_0 @ 0x7ff2b9d0c040] Channel layout is not set in output stream 0, guessed channel layout is 'stereo'
Stream mapping:
  Stream #0:0 (pcm_s16le) -> amix:input0 (graph 0)
  volume (graph 0) -> Stream #0:0 (libmp3lame)
  Stream #0:1 -> #0:1 (mjpeg (native) -> png (native))
Press [q] to stop, [?] for help
[swscaler @ 0x108d00000] deprecated pixel format used, make sure you did set range correctly
[Parsed_amovie_0 @ 0x7ff2b9e09840] Channel layout is not set in output stream 0, guessed channel layout is 'stereo'
Output #0, mp3, to '/path/preview.mp3':
  Metadata:
    comment         :  motion graphics, motion, textures
    time_reference  : 0
    coding_history  : 
    TSSE            : Lavf58.29.100
    Stream #0:0: Audio: mp3 (libmp3lame), 44100 Hz, stereo, fltp, 320 kb/s
    Metadata:
      encoder         : Lavc58.54.100 libmp3lame
    Stream #0:1: Video: png, rgb24(progressive), 288x288 [SAR 1:1 DAR 1:1], q=2-31, 200 kb/s, 90k fps, 90k tbn, 90k tbc (attached pic)
    Metadata:
      comment         : Cover (front)
      encoder         : Lavc58.54.100 png
frame=    1 fps=0.0 q=-0.0 Lsize=     206kB time=00:00:00.10 bitrate=15994.8kbits/s speed=3.62x    
video:200kB audio:5kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.598223%


  



  Thanks in advance !