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• Create HLS streamable audio file from mp3

  15 août 2023, par isADon

  I am using following command to create a hls aac audio file for web streaming

  



  ffmpeg -y -i song.mp3 -c:a aac -b:a 128k -f hls -hls_time 7 -hls_list_size 0 -hls_segment_filename file%d.m4a playlist.m3u8


  



  This command works only with some audio files. With many mp3 files I receive following output :

  



  C:\ffmpeg>ffmpeg -y -i song.mp3 -c:a aac -b:a 128k -f hls -hls_time 7 -hls_list_size 0 -hls_segment_filename file%d.m4a playlist.m3u8
ffmpeg version git-2020-01-31-62d92a8 Copyright (c) 2000-2020 the FFmpeg developers
  built with gcc 9.2.1 (GCC) 20200122
  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. 67.100 / 58. 67.100
  libavformat    58. 37.100 / 58. 37.100
  libavdevice    58.  9.103 / 58.  9.103
  libavfilter     7. 72.100 /  7. 72.100
  libswscale      5.  6.100 /  5.  6.100
  libswresample   3.  6.100 /  3.  6.100
  libpostproc    55.  6.100 / 55.  6.100
[mp3 @ 0000027d800babc0] Estimating duration from bitrate, this may be inaccurate
Input #0, mp3, from 'song.mp3':
  Metadata:
    TSS             : Logic Pro 8.0.2
    iTunNORM        :  000000EE 000000ED 00000C34 00001135 000088F0 0000B505 000080FA 00007577 00009B82 00018F49
    iTunSMPB        :  00000000 00000210 00000A07 00000000008783E9 00000000 007AD4E6 00000000 00000000 00000000 00000000 00000000 00000000
    genre           : Rock
    TCM             : Kevin MacLeod
    album           : Funk and Blues
    TKE             : C
    TBP             : 101
    title           : Funkorama
    artist          : Kevin MacLeod
    date            : 2008-06-16 18:35
  Duration: 00:03:21.46, start: 0.000000, bitrate: 325 kb/s
    Stream #0:0: Audio: mp3, 44100 Hz, stereo, fltp, 320 kb/s
    Stream #0:1: Video: mjpeg (Baseline), yuvj444p(pc, bt470bg/unknown/unknown), 400x400 [SAR 72:72 DAR 1:1], 90k tbr, 90k tbn, 90k tbc (attached pic)
    Metadata:
      comment         : Other
Stream mapping:
  Stream #0:1 -> #0:0 (mjpeg (native) -> h264 (libx264))
  Stream #0:0 -> #0:1 (mp3 (mp3float) -> aac (native))
Press [q] to stop, [?] for help
[hls @ 0000027d80100c40] Frame rate very high for a muxer not efficiently supporting it.
Please consider specifying a lower framerate, a different muxer or -vsync 2
[libx264 @ 0000027d800c1280] using SAR=1/1
[libx264 @ 0000027d800c1280] MB rate (56250000) > level limit (16711680)
[libx264 @ 0000027d800c1280] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX FMA3 BMI2 AVX2
[libx264 @ 0000027d800c1280] profile High 4:4:4 Predictive, level 6.2, 4:4:4, 8-bit
[libx264 @ 0000027d800c1280] 264 - core 159 - H.264/MPEG-4 AVC codec - Copyleft 2003-2019 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=4 threads=12 lookahead_threads=2 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
Output #0, hls, to 'playlist.m3u8':
  Metadata:
    TSS             : Logic Pro 8.0.2
    iTunNORM        :  000000EE 000000ED 00000C34 00001135 000088F0 0000B505 000080FA 00007577 00009B82 00018F49
    iTunSMPB        :  00000000 00000210 00000A07 00000000008783E9 00000000 007AD4E6 00000000 00000000 00000000 00000000 00000000 00000000
    genre           : Rock
    TCM             : Kevin MacLeod
    album           : Funk and Blues
    TKE             : C
    TBP             : 101
    title           : Funkorama
    artist          : Kevin MacLeod
    date            : 2008-06-16 18:35
    encoder         : Lavf58.37.100
    Stream #0:0: Video: h264 (libx264), yuvj444p(pc, progressive), 400x400 [SAR 72:72 DAR 1:1], q=-1--1, 90k fps, 90k tbn, 90k tbc (attached pic)
    Metadata:
      comment         : Other
      encoder         : Lavc58.67.100 libx264
    Side data:
      cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: N/A
    Stream #0:1: Audio: aac (LC), 44100 Hz, stereo, fltp, 128 kb/s
    Metadata:
      encoder         : Lavc58.67.100 aac
[mp3float @ 0000027d80146580] overread, skip -7 enddists: -6 -6 speed=68.6x
[mp3float @ 0000027d80146580] overread, skip -6 enddists: -5 -5
[mp3float @ 0000027d80146580] overread, skip -6 enddists: -4 -4
    Last message repeated 2 times
[mp3float @ 0000027d80146580] overread, skip -7 enddists: -6 -6
    Last message repeated 2 times
[mp3float @ 0000027d80146580] overread, skip -5 enddists: -2 -2
[mp3float @ 0000027d80146580] overread, skip -7 enddists: -6 -6
[mp3float @ 0000027d80146580] overread, skip -6 enddists: -4 -4
    Last message repeated 1 times
[mp3float @ 0000027d80146580] overread, skip -7 enddists: -6 -6
    Last message repeated 1 times
[mp3float @ 0000027d80146580] overread, skip -6 enddists: -4 -4
[mp3float @ 0000027d80146580] overread, skip -5 enddists: -3 -3
[mp3float @ 0000027d80146580] overread, skip -6 enddists: -4 -4
[mp3float @ 0000027d80146580] overread, skip -7 enddists: -6 -6
    Last message repeated 2 times
[mp3float @ 0000027d80146580] overread, skip -5 enddists: -4 -4
[hls @ 0000027d80100c40] Opening 'file0.m4a' for writingate=N/A speed=64.1x
[hls @ 0000027d80100c40] Opening 'playlist.m3u8.tmp' for writing
frame=    1 fps=0.3 q=33.0 Lsize=N/A time=00:03:21.45 bitrate=N/A speed=63.7x
video:7kB audio:3209kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
[libx264 @ 0000027d800c1280] frame I:1     Avg QP:34.64  size:  6567
[libx264 @ 0000027d800c1280] mb I  I16..4: 19.5% 53.0% 27.5%
[libx264 @ 0000027d800c1280] 8x8 transform intra:53.0%
[libx264 @ 0000027d800c1280] coded y,u,v intra: 46.8% 26.1% 15.3%
[libx264 @ 0000027d800c1280] i16 v,h,dc,p: 38% 39%  9% 14%
[libx264 @ 0000027d800c1280] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 21% 14% 26%  8%  5%  6%  5%  7%  7%
[libx264 @ 0000027d800c1280] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 42% 16% 14%  7%  4%  5%  3%  4%  4%
[libx264 @ 0000027d800c1280] kb/s:4728240.00
[aac @ 0000027d800bcc40] Qavg: 2138.508


  



  Notice the "mp3float overread" message.

  



  It results in a single file0.m4a file without splitting it up after every 7 seconds as specified.
This is an example audio file I am trying to convert to a aac hls stream that results the mentioned problem : https://incompetech.com/music/royalty-free/index.html?isrc=USUAN1100474

  



  How can I convert an audio file to a web friendly hls stream with ffmpeg ?

  


• ffmpeg : Downmixing 5.1 to stereo with phase invert

  1er novembre 2022, par anti-hero

  I'd like to use ffmpeg and Dolby Logic Pro II formula to downmix a 5.1 song to stereo :

  


  


  Lt = L + (–3 dB × C) – (–1.2 dB × Ls) – (–6.2 dB × Rs)

  


  Rt = R + (–3 dB × C) + (–6.2 dB × Ls) + (–1.2 dB × Rs)

  


  


  I came up with this long and probably redundant bash script :

  


  # L channel
ffmpeg -i "$filename" -map_channel 0.1.0 -c:a pcm_f32le L.wav
ffmpeg -i "$filename" -map_channel 0.1.2 -filter:a "volume=-3dB" -c:a pcm_f32le C.wav
ffmpeg -i "$filename" -map_channel 0.1.4 -af "volume=-1.2dB" -c:a pcm_f32le Ls.wav
ffmpeg -i "$filename" -map_channel 0.1.5 -af "volume=-6.2dB" -c:a pcm_f32le Rs.wav
ffmpeg -i Ls.wav -af "aeval=-val(0)" -c:a pcm_f32le Lsi.wav
ffmpeg -i Rs.wav -af "aeval=-val(0)" -c:a pcm_f32le Rsi.wav
ffmpeg -i L.wav -i C.wav -i Lsi.wav -i Rsi.wav -filter_complex "amix=inputs=4" -c:a pcm_f32le L_final.wav
rm L.wav Ls.wav Rs.wav Lsi.wav Rsi.wav

# R channel
ffmpeg -i "$filename" -map_channel 0.1.1 -c:a pcm_f32le R.wav
ffmpeg -i "$filename" -map_channel 0.1.4 -filter:a "volume=-6.2dB" -c:a pcm_f32le Ls.wav
ffmpeg -i "$filename" -map_channel 0.1.5 -filter:a "volume=-1.2dB" -c:a pcm_f32le Rs.wav
ffmpeg -i R.wav -i C.wav -i Ls.wav -i Rs.wav -filter_complex "amix=inputs=4" -c:a pcm_f32le R_final.wav
rm R.wav C.wav Ls.wav Rs.wav

# Mix
ffmpeg -i L_final.wav -i R_final.wav -filter_complex "[0:0][1:0] amerge=inputs=2" -c:a pcm_f32le "${filename%.*}_dm.wav"
rm L_final.wav R_final.wav


  


  Would this work ? And is there a simpler way to do this ?

  


  update : I also wrote Python code for the 90 degree phase shift version.

  


  import scipy, os, numpy as np

os.system(f'ffmpeg -i "{filename}" -c:a pcm_f32le temp.wav')
sample_rate, waves = scipy.io.wavfile.read("temp.wav")
os.remove("temp.wav")

L, R, C, LFE, LS, RS = waves.T

LS_shifted = -np.imag(scipy.signal.hilbert(LS))
RS_shifted = np.imag(scipy.signal.hilbert(RS))
L_out = L + 0.708 * C - 0.871 * LS_shifted - 0.49 * RS_shifted
R_out = R + 0.708 * C + 0.49 * LS_shifted + 0.871 * RS_shifted

scipy.io.wavfile.write("out.wav", sample_rate, np.vstack((L_out, R_out)).T)


  


  Hope someone finds this useful ! (and that someone verifies it's correct)

  


• cannot find ffmpeg in videoshow in nodejs

  6 novembre 2019, par Sunil Garg

  I want to create video from image files. So I have installed videoshow module. And configured the same as per the documentaion.

  var videoOptions = {
  
    fps: 25,
  
    loop: 5, // seconds 
  
    transition: true,
  
    transitionDuration: 1, // seconds 
  
    videoBitrate: 1024,
  
    videoCodec: 'libx264',
  
    size: '640x?',
  
    audioBitrate: '128k',
  
    audioChannels: 2,
  
    format: 'mp4',
  
    pixelFormat: 'yuv420p'
  
  }
  
  
  
  var images = [
  
    "D:/PROJECTS/Video/storage/1.jpg",
  
    "D:/PROJECTS/Video/storage/2.jpg"
  
  ];
  
  
  
  app.get("/video", function () {
  
    videoshow(images, videoOptions)
  
      // .audio('song.mp3')
  
      .save('video.mp4')
  
      .on('start', function (command) {
  
        console.log('ffmpeg process started:', command)
  
      })
  
      .on('error', function (err, stdout, stderr) {
  
        console.error('Error:', err)
  
        console.error('ffmpeg stderr:', stderr)
  
      })
  
      .on('end', function (output) {
  
        console.error('Video created in:', output)
  
      })
  
  });

  But When I run it shows the error on server

  Error: Error: Cannot find ffmpeg
  
      at D:\PROJECTS\Video\node_modules\videoshow\node_modules\fluent-ffmpeg\lib\processor.js:136:22
  
      at D:\PROJECTS\Video\node_modules\videoshow\node_modules\fluent-ffmpeg\lib\capabilities.js:123:9
  
      at D:\PROJECTS\Video\node_modules\videoshow\node_modules\async\dist\async.js:473:16
  
      at next (D:\PROJECTS\Video\node_modules\videoshow\node_modules\async\dist\async.js:5315:29)
  
      at D:\PROJECTS\Video\node_modules\videoshow\node_modules\async\dist\async.js:958:16
  
      at D:\PROJECTS\Video\node_modules\videoshow\node_modules\fluent-ffmpeg\lib\capabilities.js:116:11
  
      at D:\PROJECTS\Video\node_modules\videoshow\node_modules\fluent-ffmpeg\lib\utils.js:223:16
  
      at F (D:\PROJECTS\Video\node_modules\videoshow\node_modules\which\which.js:68:16)
  
      at E (D:\PROJECTS\Video\node_modules\videoshow\node_modules\which\which.js:80:29)
  
      at D:\PROJECTS\Video\node_modules\videoshow\node_modules\which\which.js:89:16

  Then I installed ffmpeg using

  npm install ffmpeg --save

  but not worked. So I tried installing at the global level using

  npm install ffmpeg -g

  Even installing on my window machine and setting the path of its bin folder in environment variables did not work ?

  What could be the issue ?