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  • Download highest quality audio from YouTube using youtube-dl

    3 juin 2020, par darvast

    I'm using this command :

    



    youtube-dl -f bestaudio --extract-audio --audio-format "opus" --add-metadata -o "%(playlist_index)s-%(title)s.%(ext)s" "https://www.youtube.com/playlist?list=OLAK5uy_lWRq5MhPNthDDe1nYXtlekDA40wtrpKE0"


    



    Here's the available streams :

    



    [info] Available formats for 6t1dErgAglk:
format code  extension  resolution note
249          webm       audio only tiny   58k , opus @ 50k (48000Hz), 416.34KiB
250          webm       audio only tiny   72k , opus @ 70k (48000Hz), 516.52KiB
140          m4a        audio only tiny  130k , m4a_dash container, mp4a.40.2@128k (44100Hz), 1.06MiB
251          webm       audio only tiny  131k , opus @160k (48000Hz), 923.79KiB
278          webm       140x144    144p   32k , webm container, vp9, 25fps, video only, 159.74KiB
160          mp4        140x144    144p   54k , avc1.4d400b, 25fps, video only, 278.62KiB
242          webm       232x240    240p   71k , vp9, 25fps, video only, 321.29KiB
134          mp4        350x360    360p   96k , avc1.4d4015, 25fps, video only, 303.75KiB
133          mp4        232x240    240p  124k , avc1.4d400c, 25fps, video only, 651.46KiB
243          webm       350x360    360p  126k , vp9, 25fps, video only, 545.77KiB
135          mp4        466x480    360p  174k , avc1.4d401e, 25fps, video only, 534.97KiB
244          webm       466x480    360p  215k , vp9, 25fps, video only, 1003.20KiB
136          mp4        698x720    720p  305k , avc1.4d401f, 25fps, video only, 942.76KiB
137          mp4        1048x1080  1080p  494k , avc1.640020, 25fps, video only, 1.49MiB
247          webm       698x720    720p  593k , vp9, 25fps, video only, 1.97MiB
248          webm       1048x1080  1080p  768k , vp9, 25fps, video only, 3.81MiB
18           mp4        350x360    360p  213k , avc1.42001E, 25fps, mp4a.40.2@ 96k (44100Hz), 1.73MiB
22           mp4        698x720    720p  242k , avc1.64001F, 25fps, mp4a.40.2@192k (44100Hz) (best)


    



    When I run the above command it seems to be converting m4a to opus : https://prnt.sc/st2l8u

    



    I'm wondering why it's doing that instead of getting it from the webm container ?

    


  • FFMpeg WebVTT hls playlist out of sync [closed]

    29 mai 2023, par user1503606

    I am using the following commands with FFmpeg to encode a .mov video to hls and then also create a playlist for subtitles converting a .srt file to .vtt.

    



    // Creating hls playlist 

ffmpeg -hide_banner -y -i /Users/dave/mermaid.mov -c:a aac -ar 48000 -c:v h264 -profile:v baseline -avoid_negative_ts 0 -start_number 0 -hls_time 10 -hls_playlist_type vod -hls_list_size 0 -f hls -vf scale=w=640:h=360:force_original_aspect_ratio=decrease -b:v 800k -maxrate 856k -bufsize 1200k -b:a 96k -hls_segment_filename /Users/dave/mermaid/360/360p_%03d.ts /Users/dave/dave/mermaid/360/360p.m3u8 
-c:a aac -ar 48000 -c:v h264 -profile:v baseline -avoid_negative_ts 0 -start_number 0 -hls_time 10 -hls_playlist_type vod -hls_list_size 0 -f hls -vf scale=w=842:h=480:force_original_aspect_ratio=decrease -b:v 1400k -maxrate 1498k -bufsize 2100k -b:a 128k -hls_segment_filename /Users/dave/Desktop/mermaid/480/480p_%03d.ts /Users/dave/Desktop/mermaid/480/480p.m3u8 
-c:a aac -ar 48000 -c:v h264 -profile:v baseline -avoid_negative_ts 0 -start_number 0 -hls_time 10 -hls_playlist_type vod -hls_list_size 0 -f hls -vf scale=w=1280:h=720:force_original_aspect_ratio=decrease -b:v 2800k -maxrate 2996k -bufsize 4200k -b:a 128k -hls_segment_filename /Users/dave/Desktop/mermaid/720/720p_%03d.ts /Users/dave/Desktop/mermaid/720/720p.m3u8 
-c:a aac -ar 48000 -c:v h264 -profile:v baseline -avoid_negative_ts 0 -start_number 0 -hls_time 10 -hls_playlist_type vod -hls_list_size 0 -f hls -vf scale=w=1920:h=1080:force_original_aspect_ratio=decrease -b:v 5000k -maxrate 5350k -bufsize 7500k -b:a 192k -hls_segment_filename /Users/dave/Desktop/mermaid/1080/1080p_%03d.ts /Users/dave/Desktop/mermaid/1080/1080p.m3u8

// Creating subtitles

ffmpeg -i /Users/dave/mermaid.mov -i /Users/dave/little-mermaid.srt -c copy -c:s webvtt -start_number 0 -hls_time 10000000 -hls_flags single_file -hls_playlist_type vod -f hls -avoid_negative_ts 0 /Users/dave/mermaid/English/master.m3u8


    



    I am also doing this with AWS Media Convert the AWS conversion works fine the subtitles are perfectly in time but the FFMPEG version seems to be out of sync by about 1 second.

    



    Here are both files for to compare the exact same files are used for both.

    



    AWS Media Convert : https://d25hd5yfabpc2n.cloudfront.net/c0017bfe-3617-47a5-9525-86d69c73cf57/master.m3u8

    



    Local FFMPEG : https://d25hd5yfabpc2n.cloudfront.net/stackoverflow/master.m3u8

    



    The only thing I can see if I download both playlists is the AWS one has FRAME-RATE set.

    



    Here is an example of both videos playing the FFMEPG one is noticeably out of sync

    



    https://output.jsbin.com/xopohupivu/

    



    Another thing I have noticed is the AWS webvtt version has this at the top of the webvtt file.

    



    WEBVTT  #Elemental Media Engine(TM) 2.16.0.602399
X-TIMESTAMP-MAP=LOCAL:00:00:00.000,MPEGTS:191970


    



    Not sure what is happening here can't find any information on how to implement this with FFMPEG.

    



    Thanks

    


  • Extra frame added after ffmpeg concat

    7 mai 2020, par megapotz

    I have a bunch of mp4 files (h.264, 25 fps), which I want to concat. Concat file is :

    



    file 'clip1.mp4'
outpoint 4

file 'clip2.mp4'
outpoint 4

file 'clip3.mp4'
outpoint 4

file 'clip4.mp4'
outpoint 4

file 'clip5.mp4'
outpoint 4


    



    Then I run

    



    \ffmpeg -f concat -i listd.txt -vcodec copy -y testtttt.mp4


    



    Output is :

    



    ffmpeg -f concat -i listd.txt -vcodec copy -y testtttt.mp4
ffmpeg version 4.2.2 Copyright (c) 2000-2019 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-amf --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt
  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
  libswscale      5.  5.100 /  5.  5.100
  libswresample   3.  5.100 /  3.  5.100
  libpostproc    55.  5.100 / 55.  5.100
[mov,mp4,m4a,3gp,3g2,mj2 @ 000002027d363080] Auto-inserting h264_mp4toannexb bitstream filter
Input #0, concat, from 'listd.txt':
  Duration: N/A, start: 0.000000, bitrate: 193 kb/s
    Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 193 kb/s, 25 fps, 25 tbr, 12800 tbn, 50 tbc
    Metadata:
      handler_name    : VideoHandler
Output #0, mp4, to 'testtttt.mp4':
  Metadata:
    encoder         : Lavf58.29.100
    Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 193 kb/s, 25 fps, 25 tbr, 12800 tbn, 12800 tbc
    Metadata:
      handler_name    : VideoHandler
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
Press [q] to stop, [?] for help
[mov,mp4,m4a,3gp,3g2,mj2 @ 000002027d36b900] Auto-inserting h264_mp4toannexb bitstream filter
[mp4 @ 000002027d376500] Non-monotonous DTS in output stream 0:0; previous: 50688, current: 50688; changing to 50689. This may result in incorrect timestamps in the output file.
[mov,mp4,m4a,3gp,3g2,mj2 @ 000002027d36b900] Auto-inserting h264_mp4toannexb bitstream filter
[mp4 @ 000002027d376500] Non-monotonous DTS in output stream 0:0; previous: 101888, current: 101888; changing to 101889. This may result in incorrect timestamps in the output file.
[mov,mp4,m4a,3gp,3g2,mj2 @ 000002027d36b900] Auto-inserting h264_mp4toannexb bitstream filter
[mp4 @ 000002027d376500] Non-monotonous DTS in output stream 0:0; previous: 153088, current: 153088; changing to 153089. This may result in incorrect timestamps in the output file.
[mov,mp4,m4a,3gp,3g2,mj2 @ 000002027d36b900] Auto-inserting h264_mp4toannexb bitstream filter
[mp4 @ 000002027d376500] Non-monotonous DTS in output stream 0:0; previous: 204288, current: 204288; changing to 204289. This may result in incorrect timestamps in the output file.
frame=  505 fps=0.0 q=-1.0 Lsize=     661kB time=00:00:19.96 bitrate= 271.4kbits/s speed= 238x
video:656kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.823749%


    



    All input files are 25 fps, so I expect 500 frames 20seconds long file in output. However I get 505.

    



    The mediainfo looks like this :

    



    enter image description here

    



    Investigating further :

    



    ffprobe -select_streams v -show_frames -show_entries frame=pict_type -of csv testtttt.mp4 >1.txt


    



    I see that keyframes are 0, 101, 202, 303, 404.
Looks like ffmpeg adds an extra frame to each series of 100 frames.

    



    Here's my setup (only 2MB)

    



    https://www.dropbox.com/s/fjvhcoxywhw1hb3/1080p.zip?dl=0

    



    How do I get 500-frames, 20 seconds long file 25fps ?

    



    UPD :

    



    I read ffmpeg docs

    



    


    This directive works best with intra frame codecs and formats where
 all streams are tightly interleaved. For non-intra frame codecs you
 will usually get additional packets with presentation timestamp after
 Out point therefore the decoded content will most likely contain
 frames after Out point too. If your streams are not tightly
 interleaved you may not get all the packets from all streams before
 Out point and you may only will be able to decode the earliest stream
 until Out point.

    


    



    Does that mean it's normal behaviour ? Than how do I get the result I want ?

    


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