Re: [FFmpeg-user] Question

2020-02-17 Thread Carl Eugen Hoyos


> Am 17.02.2020 um 14:25 schrieb CheeZe AMV's :
> 
> ffmpeg -i "
> https://pb113f1714e.dmc.nico/hlsarchive/ht2_nicolive/nicolive-production-pg24538707395151_2cbb016187df709433a9e76484bb39250da544eb6b7683e0b9f7a020a3f05c65/master.m3u8?ht2_nicolive=94477233.8mqto3hvqg_q5phgz_3auh6nan7odup";
> -c copy -bsf:a aac_adtstoasc "YMZ.mp4"

Is there really no tool or script that allows downloading such streams without 
remuxing?

Carl Eugen
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Re: [FFmpeg-user] Using FFMpeg to produce LL-HLS

2020-02-17 Thread Gyan Doshi



On 17-02-2020 08:38 pm, Dennis Mungai wrote:

On Mon, 17 Feb 2020 at 17:26, Simon Brown  wrote:

On Mon, 17 Feb 2020 at 12:06, Dennis Mungai  wrote:


On Mon, 17 Feb 2020, 14:50 Simon Brown,  wrote:


I'm trying to reduce the latency of streaming video to a web page.
I have tried a few options:
1) WebRTC - great, but doesn't support interlaced video
2) fMP4 - couldn't get a working solution
3) HLS - great, plays interlaced video as well, but latency is appalling
4) LL-HLS - can't find anything that produces it.

Does FFMpeg support generating LL-HLS?

Regards,
Simon


Yes, though at the moment you'll have to use the dash muxer.

See ffmpeg -h muxer=dash

For usage.

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Thanks Dennis,
I've read through the options and looked at some examples and have
downloaded the latest FFMpeg build from zeranoe (nightly build).  That
produces this result:

C:\ffmpeg-20200216-8578433-win64-static\bin>ffmpeg.exe -i udp://
127.0.0.1:9034 -codec copy -f dash -seg_duration 1 -frag_duration 0.1
-window_size 5 -extra_window_size 5 -single_file 0 -lhls 1 -hls_playlist 1
-strict experimental -streaming 1 -adaptation_sets "id=0,streams=v
id=1,streams=a" c:\xampp\htdocs\video.m3u8
ffmpeg version git-2020-02-16-8578433 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. 41.100 / 56. 41.100
   libavcodec 58. 70.100 / 58. 70.100
   libavformat58. 38.101 / 58. 38.101
   libavdevice58.  9.103 / 58.  9.103
   libavfilter 7. 76.100 /  7. 76.100
   libswscale  5.  6.100 /  5.  6.100
   libswresample   3.  6.100 /  3.  6.100
   libpostproc55.  6.100 / 55.  6.100
Input #0, mpegts, from 'udp://127.0.0.1:9034':
   Duration: N/A, start: 68873.648600, bitrate: N/A
   Program 1
 Stream #0:0[0x1100]: Video: h264 (Main) ([27][0][0][0] / 0x001B),
yuv420p(tv, progressive), 1920x1080 [SAR 1:1 DAR 16:9], 50 fps, 50 tbr, 90k
tbn, 100 tbc
 Stream #0:1[0x1110]: Audio: aac (LC) ([15][0][0][0] / 0x000F), 48000
Hz, stereo, fltp, 192 kb/s
[dash @ 022f500c6a40] No bit rate set for stream 0
[dash @ 022f500c6a40] Opening 'init-stream0.m4s' for writing
[mp4 @ 022f4e21ac00] Could not find tag for codec h264 in stream #0,
codec not currently supported in container
Could not write header for output file #0 (incorrect codec parameters ?):
Invalid argument
Stream mapping:
   Stream #0:0 -> #0:0 (copy)
   Stream #0:1 -> #0:1 (copy)
 Last message repeated 1 times

What am I doing wrong?  I thought h264 was supported in HLS/Dash?

Regards,
Simon

Hey there,

Try this instead:

ffmpeg.exe -i udp://127.0.0.1:9034 -codec copy -bsf:a aac_adtstoasc
-flags +global_header -f dash ^
-seg_duration 1 -frag_duration 0.1 -window_size 5 -extra_window_size 5
-single_file 0 -lhls 1 -hls_playlist 1 ^
-strict experimental -streaming 1 -adaptation_sets "id=0,streams=v
id=1,streams=a" c:\xampp\htdocs\video.m3u8

Then report back.

Btw I'd recommend re-encoding the content wherever possible. A fixed
GOP size combined with the -flags +cgop with an encoder such as
libx264 produces the best results.
The GOP size should ideally be a fraction of the segment length, eg 2
seconds. The codec neutral option -g 2 should set that for you.


-g is in frames, so -g 50 for a 25 fps stream. Or you could use 
-force_key_frames


Gyan
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Re: [FFmpeg-user] Question

2020-02-17 Thread Ted Park
> It is a live TV Streaming site and the VOD remains for 7 days.

Then you’d probably have better luck downloading the VOD episodes when they 
become available, since in a VOD stream, segments don’t “expire” and the master 
playlist stays constant unlike a live “event” type stream. Unless you were 
already doing that? If you were, the problem might be something else.

> Download speed: 7.84
> Upload speed: 0.61

I mean honestly it does seem like it would be unbearable but that’s just 
because I’m spoiled. The download speed is around what you would get with a 
dedicated t2 circuit, I don’t think it’s considered too bad compared to the 
global average and encoding houses take that into account when delivering for 
web streams. 

But what’s more relevant is the speed you get when you’re downloading the 
actual stream, I assume the test was done on a different server, the speeds you 
get when you’re connected to the server with the media segments will be 
different. I don’t know what the CDN of niconico is like, but how far you are 
away from Japan might affect it a lot.

Your connection speeds shouldn’t even matter if you download a VOD stream 
(apart from the time it takes), the streaming service might still invalidate 
the URLs after a while, but usually not while you still have an open 
connection. If you can post the output of 
% ffmpeg -loglevel trace -i "m3u8 master playlist url" -c copy -bsf:a 
aac_adtstoasc “example.mp4”
We’d know more.
Try to post the output of the command you run when you post to this list in 
general, it includes a lot of relevant information.
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Re: [FFmpeg-user] Using FFMpeg to produce LL-HLS

2020-02-17 Thread Dennis Mungai
On Mon, 17 Feb 2020 at 17:26, Simon Brown  wrote:
>
> On Mon, 17 Feb 2020 at 12:06, Dennis Mungai  wrote:
>
> > On Mon, 17 Feb 2020, 14:50 Simon Brown,  wrote:
> >
> > > I'm trying to reduce the latency of streaming video to a web page.
> > > I have tried a few options:
> > > 1) WebRTC - great, but doesn't support interlaced video
> > > 2) fMP4 - couldn't get a working solution
> > > 3) HLS - great, plays interlaced video as well, but latency is appalling
> > > 4) LL-HLS - can't find anything that produces it.
> > >
> > > Does FFMpeg support generating LL-HLS?
> > >
> > > Regards,
> > > Simon
> > >
> >
> > Yes, though at the moment you'll have to use the dash muxer.
> >
> > See ffmpeg -h muxer=dash
> >
> > For usage.
> >
> > >
> > ___
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>
>
> Thanks Dennis,
> I've read through the options and looked at some examples and have
> downloaded the latest FFMpeg build from zeranoe (nightly build).  That
> produces this result:
>
> C:\ffmpeg-20200216-8578433-win64-static\bin>ffmpeg.exe -i udp://
> 127.0.0.1:9034 -codec copy -f dash -seg_duration 1 -frag_duration 0.1
> -window_size 5 -extra_window_size 5 -single_file 0 -lhls 1 -hls_playlist 1
> -strict experimental -streaming 1 -adaptation_sets "id=0,streams=v
> id=1,streams=a" c:\xampp\htdocs\video.m3u8
> ffmpeg version git-2020-02-16-8578433 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. 41.100 / 56. 41.100
>   libavcodec 58. 70.100 / 58. 70.100
>   libavformat58. 38.101 / 58. 38.101
>   libavdevice58.  9.103 / 58.  9.103
>   libavfilter 7. 76.100 /  7. 76.100
>   libswscale  5.  6.100 /  5.  6.100
>   libswresample   3.  6.100 /  3.  6.100
>   libpostproc55.  6.100 / 55.  6.100
> Input #0, mpegts, from 'udp://127.0.0.1:9034':
>   Duration: N/A, start: 68873.648600, bitrate: N/A
>   Program 1
> Stream #0:0[0x1100]: Video: h264 (Main) ([27][0][0][0] / 0x001B),
> yuv420p(tv, progressive), 1920x1080 [SAR 1:1 DAR 16:9], 50 fps, 50 tbr, 90k
> tbn, 100 tbc
> Stream #0:1[0x1110]: Audio: aac (LC) ([15][0][0][0] / 0x000F), 48000
> Hz, stereo, fltp, 192 kb/s
> [dash @ 022f500c6a40] No bit rate set for stream 0
> [dash @ 022f500c6a40] Opening 'init-stream0.m4s' for writing
> [mp4 @ 022f4e21ac00] Could not find tag for codec h264 in stream #0,
> codec not currently supported in container
> Could not write header for output file #0 (incorrect codec parameters ?):
> Invalid argument
> Stream mapping:
>   Stream #0:0 -> #0:0 (copy)
>   Stream #0:1 -> #0:1 (copy)
> Last message repeated 1 times
>
> What am I doing wrong?  I thought h264 was supported in HLS/Dash?
>
> Regards,
> Simon

Hey there,

Try this instead:

ffmpeg.exe -i udp://127.0.0.1:9034 -codec copy -bsf:a aac_adtstoasc
-flags +global_header -f dash ^
-seg_duration 1 -frag_duration 0.1 -window_size 5 -extra_window_size 5
-single_file 0 -lhls 1 -hls_playlist 1 ^
-strict experimental -streaming 1 -adaptation_sets "id=0,streams=v
id=1,streams=a" c:\xampp\htdocs\video.m3u8

Then report back.

Btw I'd recommend re-encoding the content wherever possible. A fixed
GOP size combined with the -flags +cgop with an encoder such as
libx264 produces the best results.
The GOP size should ideally be a fraction of the segment length, eg 2
seconds. The codec neutral option -g 2 should set that for you.
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Re: [FFmpeg-user] Using FFMpeg to produce LL-HLS

2020-02-17 Thread Simon Brown
On Mon, 17 Feb 2020 at 12:06, Dennis Mungai  wrote:

> On Mon, 17 Feb 2020, 14:50 Simon Brown,  wrote:
>
> > I'm trying to reduce the latency of streaming video to a web page.
> > I have tried a few options:
> > 1) WebRTC - great, but doesn't support interlaced video
> > 2) fMP4 - couldn't get a working solution
> > 3) HLS - great, plays interlaced video as well, but latency is appalling
> > 4) LL-HLS - can't find anything that produces it.
> >
> > Does FFMpeg support generating LL-HLS?
> >
> > Regards,
> > Simon
> >
>
> Yes, though at the moment you'll have to use the dash muxer.
>
> See ffmpeg -h muxer=dash
>
> For usage.
>
> >
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Thanks Dennis,
I've read through the options and looked at some examples and have
downloaded the latest FFMpeg build from zeranoe (nightly build).  That
produces this result:

C:\ffmpeg-20200216-8578433-win64-static\bin>ffmpeg.exe -i udp://
127.0.0.1:9034 -codec copy -f dash -seg_duration 1 -frag_duration 0.1
-window_size 5 -extra_window_size 5 -single_file 0 -lhls 1 -hls_playlist 1
-strict experimental -streaming 1 -adaptation_sets "id=0,streams=v
id=1,streams=a" c:\xampp\htdocs\video.m3u8
ffmpeg version git-2020-02-16-8578433 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. 41.100 / 56. 41.100
  libavcodec 58. 70.100 / 58. 70.100
  libavformat58. 38.101 / 58. 38.101
  libavdevice58.  9.103 / 58.  9.103
  libavfilter 7. 76.100 /  7. 76.100
  libswscale  5.  6.100 /  5.  6.100
  libswresample   3.  6.100 /  3.  6.100
  libpostproc55.  6.100 / 55.  6.100
Input #0, mpegts, from 'udp://127.0.0.1:9034':
  Duration: N/A, start: 68873.648600, bitrate: N/A
  Program 1
Stream #0:0[0x1100]: Video: h264 (Main) ([27][0][0][0] / 0x001B),
yuv420p(tv, progressive), 1920x1080 [SAR 1:1 DAR 16:9], 50 fps, 50 tbr, 90k
tbn, 100 tbc
Stream #0:1[0x1110]: Audio: aac (LC) ([15][0][0][0] / 0x000F), 48000
Hz, stereo, fltp, 192 kb/s
[dash @ 022f500c6a40] No bit rate set for stream 0
[dash @ 022f500c6a40] Opening 'init-stream0.m4s' for writing
[mp4 @ 022f4e21ac00] Could not find tag for codec h264 in stream #0,
codec not currently supported in container
Could not write header for output file #0 (incorrect codec parameters ?):
Invalid argument
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #0:1 -> #0:1 (copy)
Last message repeated 1 times

What am I doing wrong?  I thought h264 was supported in HLS/Dash?

Regards,
Simon
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Re: [FFmpeg-user] Question

2020-02-17 Thread CheeZe AMV's
It is a live TV Streaming site and the VOD remains for 7 days. My internet
speed is utter garbage because for some reason the area in which I live in
still doesn't have Fibre Optics. So, it makes sense that those missed
segments are to do with buffering, because I'm trying to get the best
possible quality.

What bitrate would you suggest & how would I work that in a command line?

Here are my speed test results (horrible, right?):
Download speed: 7.84
Upload speed: 0.61

On Mon, Feb 17, 2020 at 1:42 AM Ted Park  wrote:

> > Would you be able to help me with why there are skipped segments and how
> I
> > can avoid them? Because after it's downloaded there are frames in the
> video
> > that are frozen. The website I am referring to is:
> > https://ch.nicovideo.jp/nicopro if that's of any assistance. Also, if
> you
> > don't mind would you be able to template the solution in a similar
> > format that I have, that would be of great help.
>
> Is that a live TV streaming site? (I think I’ve heard it being mentioned
> in Chihayafuru a couple weeks ago)
> If you’re downloading a live stream, it most likely means that a segment
> was taken off the master playlist before it was downloaded, as in ffmpeg
> didn’t reach it in time. Something like forcing an HD resolution in a
> browser and missing a few seconds of the stream while the player buffers.
> The first workaround that comes to mind is to choose a lower bitrate
> alternative to download than the default, with -map 0:p:# (replacing # with
> the program number). I can’t think of a way to make it switch streams
> depending on bandwidth requirements like what would happen in a browser
> player implementation.
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Re: [FFmpeg-user] Question

2020-02-17 Thread CheeZe AMV's
here is the exact command line: ffmpeg -i "
https://pb113f1714e.dmc.nico/hlsarchive/ht2_nicolive/nicolive-production-pg24538707395151_2cbb016187df709433a9e76484bb39250da544eb6b7683e0b9f7a020a3f05c65/master.m3u8?ht2_nicolive=94477233.8mqto3hvqg_q5phgz_3auh6nan7odup";
-c copy -bsf:a aac_adtstoasc "YMZ.mp4"

Also, I am using the latest version & by template I did mean a command
line, thank you.

On Sun, Feb 16, 2020 at 1:03 PM Moritz Barsnick  wrote:

> On Sat, Feb 15, 2020 at 11:11:53 +, CheeZe AMV's wrote:
> > Would you be able to help me with why there are skipped segments and how
> I
> > can avoid them? Because after it's downloaded there are frames in the
> video
> > that are frozen. The website I am referring to is:
> > https://ch.nicovideo.jp/nicopro if that's of any assistance.
>
> On this list, it is of utmost assistance if you provide us with the
> actual (real, minus private credentials) ffmpeg command line you used,
> and its complete, uncut console output. There is lots of valuable
> information in there.
>
> Also please be sure to use the most latest version of ffmpeg. Please
> retry with a version from git master to see whether your issue
> persists.
>
> (Windows:  https://ffmpeg.zeranoe.com/builds/, choose the "Nightly
> builds (2020021x)"
>  Linux x86_64: https://johnvansickle.com/ffmpeg/,  choose from the "git
> master" column)
>
> > Also, if you don't mind would you be able to template the solution in
> > a similar format that I have, that would be of great help.
>
> What do you mean by template? A command line?
>
> Thanks,
> Moritz
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Re: [FFmpeg-user] Using FFMpeg to produce LL-HLS

2020-02-17 Thread Dennis Mungai
On Mon, 17 Feb 2020, 14:50 Simon Brown,  wrote:

> I'm trying to reduce the latency of streaming video to a web page.
> I have tried a few options:
> 1) WebRTC - great, but doesn't support interlaced video
> 2) fMP4 - couldn't get a working solution
> 3) HLS - great, plays interlaced video as well, but latency is appalling
> 4) LL-HLS - can't find anything that produces it.
>
> Does FFMpeg support generating LL-HLS?
>
> Regards,
> Simon
>

Yes, though at the moment you'll have to use the dash muxer.

See ffmpeg -h muxer=dash

For usage.

>
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[FFmpeg-user] Using FFMpeg to produce LL-HLS

2020-02-17 Thread Simon Brown
I'm trying to reduce the latency of streaming video to a web page.
I have tried a few options:
1) WebRTC - great, but doesn't support interlaced video
2) fMP4 - couldn't get a working solution
3) HLS - great, plays interlaced video as well, but latency is appalling
4) LL-HLS - can't find anything that produces it.

Does FFMpeg support generating LL-HLS?

Regards,
Simon
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Re: [FFmpeg-user] Low-quality transcode

2020-02-17 Thread Wouter Verhelst
Hi Ted,

On Mon, Feb 17, 2020 at 03:03:26AM -0500, Ted Park wrote:
> Those settings seem like it would set up the encoder for constrained quality
> mode rather than VBR as you are expecting, setting a maximum quality. So as
> long as you kept the crf same in all your attempts not seeing an increase in
> quality is expected.

Ah, right. Thanks; that explains it.

> Check out this article on vp9 rc. It uses ffmpeg as an example.
> https://developers.google.com/media/vp9/bitrate-modes 
> 

Yeah, I'm aware of that article, I based those settings on that
documentation.  I should have just gone there again ;-)

Thanks for reminding me of crf,

-- 
 Home is where you have to wash the dishes.
  -- #debian-devel, Freenode, 2004-09-22
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Re: [FFmpeg-user] Low-quality transcode

2020-02-17 Thread Ted Park
> For one particular talk, the quality of the transcode from MP4 to WebM
> is very low. I use VBR encoding, but in most cases I end up with a
> transcoded file that is about 60-70% of the size of the original file.

> The command line that was originally used is this one:
> 
> 'ffmpeg' '-loglevel' 'warning' '-y' '-f' 'concat' '-safe' '0' '-i'
> '/tmp/transSnsDON/concat.txt' '-c:v' 'libvpx-vp9' '-b:v' '1024k'
> '-minrate' '512k' '-maxrate' '1484.8k' '-r:v' '25/1' '-crf' '32'
> '-speed' '4' '-pass' '1' '-passlogfile'
> '/srv/sreview/output/2020/AW1.125/ada_rad.webm-multipass' '-c:a'
> 'libopus' '-b:a' '128k' '-ar' '48000' '-t' '1331.934' '-pix_fmt'
> 'yuv420p' '-metadata' 'title=On Rapid Application Development in Ada'
> '-metadata' 'event=FOSDEM 2020' '-metadata' 'speakers=Tomasz Maluszycki'
> '-metadata' 'date=2020-02-01'
> '/srv/sreview/output/2020/AW1.125/ada_rad.webm'
> 
> 'ffmpeg' '-loglevel' 'warning' '-y' '-f' 'concat' '-safe' '0' '-i'
> '/tmp/transSnsDON/concat.txt' '-c:v' 'libvpx-vp9' '-b:v' '1024k'
> '-minrate' '512k' '-maxrate' '1484.8k' '-r:v' '25/1' '-crf' '32'
> '-speed' '2' '-pass' '2' '-passlogfile'
> '/srv/sreview/output/2020/AW1.125/ada_rad.webm-multipass' '-c:a'
> 'libopus' '-b:a' '128k' '-ar' '48000' '-t' '1331.934' '-pix_fmt'
> 'yuv420p' '-metadata' 'title=On Rapid Application Development in Ada'
> '-metadata' 'event=FOSDEM 2020' '-metadata' 'speakers=Tomasz Maluszycki'
> '-metadata' 'date=2020-02-01'
> '/srv/sreview/output/2020/AW1.125/ada_rad.webm’


Those settings seem like it would set up the encoder for constrained quality 
mode rather than VBR as you are expecting, setting a maximum quality. So as 
long as you kept the crf same in all your attempts not seeing an increase in 
quality is expected.

Check out this article on vp9 rc. It uses ffmpeg as an example.
https://developers.google.com/media/vp9/bitrate-modes 


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