[FFmpeg-user] Average a "rolling" N frames?

[Steven Kan]

I’m putting together a time-lapse video of bees building comb in the hive. I 
have 5,000+ jpgs (and growing!) in a directory that I process with:

ffmpeg -hwaccel videotoolbox -framerate 60 -pattern_type glob -i '*.jpg' -c:v 
h264_videotoolbox -b:v 100M CombLapse.mp4

which results in:

https://www.youtube.com/watch?v=CvGAHWVcbwY

Someone suggested that I try to “remove the bees” and get video of just the 
comb. Which got me thinking, what if I could do a rolling average of, say, 100 
frames? So frame 1 of my output would be the average of frames 1 - 100, and 
frame 2 of my output would be the average of frames 2 - 101, etc.

I’ve used -vf tmix=frames=10:weights=“1” to take 10 frames of input and output 
1 frame, but what syntax could I use to do a rolling average? 

Or would I have to loop through the jpgs twice? Once to get 4,900 averaged 
stills and then another run to combine those into my video?

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Re: [FFmpeg-user] Fluorescent White Balance Video Filters

[Roger]

>> Found through Google,
>> https://lutify.me/free-white-balance-correction-luts-for-everyone/
>> Download: Free White Balance Correction LUTs
>> Lutify-me-Free-White-Balance-Correction-3D-LUTs.zip
>> 
>> 2800 Kelvin - 3200 Kelvin - 0.34 CTO.cube
>> 2800 Kelvin - 4300 Kelvin - 0.95 CTO.cube
>> 2800 Kelvin - 5500 Kelvin - 1.34 CTO.cube
>> 2800 Kelvin - 6500 Kelvin - 1.56 CTO.cube
>> 3200 Kelvin - 2800 Kelvin - 0.34 CTB.cube
>> 3200 Kelvin - 4300 Kelvin - 0.61 CTO.cube
>> 3200 Kelvin - 5500 Kelvin - 1.00 CTO.cube
>> 3200 Kelvin - 6500 Kelvin - 1.21 CTO.cube
>> 4300 Kelvin - 2800 Kelvin - 0.95 CTB.cube
>> 4300 Kelvin - 3200 Kelvin - 0.61 CTB.cube
>> 4300 Kelvin - 5500 Kelvin - 0.39 CTO.cube
>> 4300 Kelvin - 6500 Kelvin - 0.60 CTO.cube
>> 5500 Kelvin - 2800 Kelvin - 1.34 CTB.cube
>> 5500 Kelvin - 3200 Kelvin - 1.00 CTB.cube
>> 5500 Kelvin - 4300 Kelvin - 0.39 CTB.cube
>> 5500 Kelvin - 6500 Kelvin - 0.21 CTO.cube
>> 6500 Kelvin - 2800 Kelvin - 1.56 CTB.cube
>> 6500 Kelvin - 3200 Kelvin - 1.21 CTB.cube
>> 6500 Kelvin - 4300 Kelvin - 0.60 CTB.cube
>> 6500 Kelvin - 5500 Kelvin - 0.21 CTB.cube
...
>> Are these LUTS my best options?  Or does the paid subscription offer better?
>> Or are there other white balance/color correcting LUTS elsewhere?
>

Think I figured-out, the previously listed kelvin cube files above, seem to be 
a compilation of gradual generic tint lut/cube files.

>You can make your own LUT, as described in chapter 2.28 in my book.
>-- Extract one frame from your video.
>-- Insert a haldclut in a corner of the image, or use xstack to attach it to
>the side.
>-- Use the program of your choice to correct the colors in this image. When
>done, save it lossless as 16-bit PNG.
>-- Use FFmpeg to separate the haldclut from the image.
>-- Apply the LUT to the whole video.
>
>Michael

Sorry.  I had a little difficulty with understanding the text within the 
Chapter 2.28.  Just couldn't scan and understand easily.  Although I thoroughly 
understand SH/Bash, the variable assignments for simple filenames also slowed 
reading, as I had to scan/reference back to the top of the page for the 
definition/intent of the variable used.  Easier understanding for the reader to 
read, "input_video.ext" and "output_video.ext", or use a similar named variable 
such as $_input_video.ext and $_output_video.ext. Note: using a prefixed 
underscore within SH/Bash variables separates/isolates from any possible 
conflicts.  Heard of using this syntax from the Bash mailing list when doing C 
style defines in SH/Bash.  However, for the sake of easy readability for 
readers, it's probably best to use simple non-isolated variables.

Some of the terminology used at this chapter is above my head, but I've been 
looking to do something exactly as you described when using my Nikon D5600 when 
taking raw photos and attaining similar results to it's Capture NX-D/Studio.

I've flagged this Email and will hopefully return to it when I get more free 
time.  I waste a full day on these larger tasks, especially on first time 
workflows.

Don't worry, I did figure-out how to apply LUTS, either from the chapter or 
from additional Google searching for ffmpeg Stack Exchange examples, etc.

Roger

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Re: [FFmpeg-user] Hardware acceleration for converting PNGs into VP9

[Dennis Mungai]

On Wed, 20 Apr 2022 at 14:22, Klimek, Denis <
dkli...@stadtwerke-norderstedt.de> wrote:

> Hi there,
>
> I’ve got an question regarding conversation from multiple PNGs into a VP9
> movie. Currently we are doing this job with our CPU and this is quite slow…
> about 2.5fps in average and we have multiple movies to create every hour.
> My idea was to accelerate it via an dedicated graphic card which can be
> installed into our server to speed this things up.
>
> Which graphic card can handle this work for me and which parameters or
> additional software pieces are required? (Vendor, Model)
>
> My current system is an Ubuntu 22.04 LTS 64bit and this is our current
> command line which converts the PNGs to an VP9:
> ffmpeg -y -r 4.86 -pattern_type glob -i $DIR'/*.png' -c:v libvpx-vp9 -b:v
> 0 -crf 30 -row-mt 1 -pix_fmt yuva420p -threads 8 -slices 16 -f webm
> $WORKINGDIR'/'$DIR'.webm.tmp'
> (Codeline is snipped from a bash script which fills the variables with the
> necessary values)
>
> Thanks for advance for any help 
>
> Mit freundlichem Gruß
> wilhelm.tel GmbH
>
> Denis Klimek
> Carrier Manager & Professional Network Engineer
> IP-Systemtechnik
> Tel:+49 (0) 40 / 521 04 – 1049 -> Work From Home
> Mobil: +49 (0) 151 / 652 219 06
>
> dkli...@stadtwerke-norderstedt.de
> www.wilhelm-tel.de
>
> __
>
> [cid:image005.png@01D854B9.A6B5E0D0]
>
> Postanschrift:
> wilhelm.tel GmbH
> Heidbergstraße 101-111
> 22846 Norderstedt
>
> Geschäftsführer: Jens Seedorff, Theo Weirich
> Vorsitzender des Aufsichtsrats: Christoph Mendel
> Handelsregister: HRB 4216 NO, Amtsgericht Kiel
> Umsatzsteuer ID: DE 81 299 7663
>
> [ig][fb]<
> https://www.facebook.com/wilhelmtel.norderstedt/>
>
>
>
Hello there,

The closest you can get would be with either:

1. Intel's VP9 encoder wrappers for VAAPI and QuickSync (with IceLake IGPs
and above), and this is only supported on Linux (at the moment).
The QSV encoder wrapper for VP9 isn't yet functional on Windows on any
tested hardware. VAAPI is also only available on Linux.
For the VP9 QSV encoder wrapper usage, see
https://superuser.com/a/1644371/473795 for a good starting point with known
limitations.
For the VP9 VAAPI encoder wrapper usage, see
https://stackoverflow.com/a/55036502/4675388 , and there are some known
limitations with that encoder too.
I'll retest this and report back as soon as I have access to Intel's
(release or engineering sample silicon's) Xe and Arc GPU products, in (both
discrete and) integrated where available.
Expect a transition towards OneVPL (from QuickSync) for these workloads as
from Intel's CometLake (10th Generation) and newer, and Intel Atom X moving
forward. See
https://www.intel.com/content/www/us/en/developer/articles/technical/onevpl-codecs-for-intel-hardware.html
for more details.

*Special notes:* 7th Generation KBL (Kabylake) GPUs have support for VP9
encoding wired up and available via VAAPI and *unofficially* via the
media-driver package patched with the approriate hybrid encode mode
enabled, see
https://github.com/sreerenjb/media-driver/tree/vp9-enc-gen9-vme-pak
However, this comes with multiple caveats, as noted on these PRs and issues:
(a). https://github.com/intel/intel-vaapi-driver/pull/209
(b). https://github.com/intel/intel-vaapi-driver/issues/87
Which resulted in this decision not to wire up the functionality (by
default) in the media-driver package:
https://github.com/intel/media-driver/issues/630#issuecomment-495498209

2. On Mac OSX:

There is potential for VP9 hardware-accelerated encoding via videotoolbox
in future(?) M-series chips, considering that we now have H/W accelerated
decoding for the same, see
https://git.ffmpeg.org/gitweb/ffmpeg.git/commit/a41a2efc85f8c88caec10040ee437562f9d0b947
.
Wouldn't bet too much on this (for now), but its' very feasible that Apple
will eventually enable this function on their chips at some point.

Warm regards,

Dennis.

Warm regards,

Dennis.
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[FFmpeg-user] Normalize video on a section of the video

[Benjamin Hill]

I have a time-lapse of a beach scene.  The exposure wasn't locked.  So, I'd
like to keep the exposure as constant as possible, likely from using the
"normalize" filter. 

The problem is that it takes the entire frame into account - including the
palm tree that is waving in the wind and messing everything up.  Is there a
way to normalize on the upper-left quadrant of the scene (which is all blue
sky) and ignore the right side?  Maybe a way to temporarily mask out the
tree when picking normalization adjustments?
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Re: [FFmpeg-user] Large Sized output files recieved while encoding the audio

[Ferdi Scholten]

Rephrasing the requirement.

We have an audio pipeline which redacts the audios using the set of start
time and end time provided from our ML team. To achieve this, we execute
the ffmpeg command with filter
"volume=enable='between(t,starttime,endtime)':volume=0".
For some cases, we are receiving audio files with codecs which are causing
the outputs to have large file sizes. One such example is below,
ffprobe and ffmpeg output of the file is attached to the mail.
*command:* ffmpeg -i call.wav -af "volume=enable='between(t,0,1)':volume=0"
output.wav
*behaviour: *executed within a second, input file size 2.1 mb, output file
size 176 mb

As we can see here, the output file size is 88 times the original. The
objective here is to achieve the output file size of less than 2-3 times
the original without reducing the sound quality.
I went through the documentation but couldn't find any suitable way to fix
it. Let me know if anyone knows the issue in my command. Please point me to
some resource containing the details of audio codecs and filtering.

Regards,
Shubham

The input file is a 19 kb/s highly compressed audio file (opus codec 
lossy format variable bitrate)


The generated wav output is in pcm 1536 kb/s uncompressed cbr.

so 1536 / 19 = 80.84... times bigger output (at least) so your 88 times 
bigger is expected going from highly compressed variable bitrate audio 
to uncompressed constant bitrate audio.


You should not convert the audio, just use a universal container like 
matroska that can contain almost any existing codec.

Try this command:

    ffmpeg -i call.wav -af "volume=enable='between(t,0,1)':volume=0" 
-c:a copy output.mka


Learn about codecs, lossy and lossless encoding and differnt containers 
to know what is going on.
maybe this can help you learn: 
http://ce.sharif.edu/courses/91-92/2/ce873-1/resources/root/Class%20Notes/MMN-lec3-Audio-911121.pdf


Greetings
Ferdi

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Re: [FFmpeg-user] Large Sized output files recieved while encoding the audio

[Bouke / edit 'B]

> On 20 Apr 2022, at 19:50, Shubham Tiwari  wrote:
> 
> I think you are not reading the message properly.
 
I think I do.

> If you notice, the size
> of the file i am passing is 2 mb, the return file size is 176 mb. Does 88
> times increase seem reasonable to you?

Yes, as far as it goes for your command line, this is to be expected, and 
totally comprehensive.

> I want the file to be in approximately similar quality and file size should
> go at max 2 to 3 times. In this case upto 6 mb file size is fine.

State what you want, do you want to trim (cut out parts), just to forward it to 
other people?
So, what do you want, what are you after?
> 
> I am doing this because we have a redaction logic to be executed on
> approximately 1 million audio files per day.  And all these calls are
> stored in aws s3. so storing multiple files of size 176 mb doesn't seem to
> be a good idea.

Again, what are you after? If you need ’some’ kind of computer logic / 
analysis, uncompressed Wave / PCM CAN be faster / more efficient than 
compressing it.
In that case, file size does not matter if it’s local, it’s just math on what 
is cheaper / more efficient.

BUT, since you don’t give any clue what / who has to process the output of your 
work, trust me, hire a specialist to drag this out of you.

I’m trying to help you, but for that, you need to provide more input.

Bouke / edit 'B

videotoolshed.com
Van Oldenbarneveltstraat 33
6512 AS Nijmegen, the Netherlands
+31 6 21817248
If you want to send me large files, please use:
https://videotoolshed.wetransfer.com/


> 
> Concerning my lack of knowledge, I read the documentation page multiple
> times but couldn't find something that gives me the right information.  In
> the end I reached out here. Concerning my lack of knowledge, probably some
> help would be better instead of passing remarks.
> 
> On Wed, Apr 20, 2022 at 11:07 PM Bouke / Videotoolshed <
> bo...@videotoolshed.com> wrote:


Bouke / edit 'B

videotoolshed.com
Van Oldenbarneveltstraat 33
6512 AS Nijmegen, the Netherlands
+31 6 21817248
If you want to send me large files, please use:
https://videotoolshed.wetransfer.com/

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Re: [FFmpeg-user] Large Sized output files recieved while encoding the audio

[Carl Zwanzig]

Please stop top-posting.

On 4/20/2022 11:19 AM, Shubham Tiwari wrote:

*behaviour: *executed within a second, input file size 2.1 mb, output file
size 176 mb


Yep, 15 minutes at 48k samples/second is... 48000 * 60 * 15 = 43 million(!) 
samples. In 16 bit stereo at 4 bytes/sample, hmm, at least 173MB. Why? 
because you're outputting pcm format, which is the default for .wav files.


And it's right in the log-
  Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, 
s16, *1536 kb/s*

(emphasis added)

Don't want a big file? Use a different output codec. How? use "-f" to 
specify it (that's all in the doc and many web pages about ffmpeg) or use a 
different file extension.



I went through the documentation but couldn't find any suitable way to fix
it. 


"-f" on the output and a better choice of format. Is it really stereo 
content? Does it need to be? Could mix that to mono. Could drop the sample 
rate from 48k to 24k. If these are recordings of telephone calls, then 16k 
is more than enough (this should be implemented when the recording is made, 
not in later processing).


There are a lot of optimizations possible but it takes a greater 
understanding of the entire process (inputs, pipeline, and how the output is 
used) to properly attack this, with that much traffic simply adjusting one 
part is a bad engineering approach. As Bouke pretty much said, you need 
someone with experience in audio processing to analyze things and make 
recommendations.



You also have-
[opus @ 0x7f78d8009800] Too many errors when draining, this is a bug. Stop 
draining and force EOF.

Error while decoding stream #0:0: Internal bug, should not have happened

which might be cured by using a more recent ffmpeg build (which is 
recommended in the mailing list FAQ and is good practice in general).



_and_  to avoid confusion whatever is creating the files should be changed 
so the output files don't have a .wav extension-
"Opus was originally specified for encapsulation in Ogg containers, 
specified as audio/ogg; codecs=opus, and for Ogg Opus files the .opus 
filename extension is recommended. Opus streams are also supported in 
Matroska, WebM, MPEG-TS, and MP4."

https://en.wikipedia.org/wiki/Opus_(audio_format)



z!
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Re: [FFmpeg-user] Large Sized output files recieved while encoding the audio

[morgan holly via ffmpeg-user]

> On Apr 20, 2022, at 12:13 PM, morgan holly via ffmpeg-user 
>  wrote:
> 
> 
>> On Apr 20, 2022, at 12:04 PM, Shubham Tiwari  
>> wrote:
>> 
>> Thanks Bouke for the opinion.
>> Help from any other person is appreciated!!
>> 
>> Regards,
>> Shubham
>> 
> 
> I have not read the entire thread, but it looks like your source is highly 
> compressed, your output is uncompressed. So yes, in this scenario you should 
> expect a much larger file size on the output. Does your redaction software 
> only work with a wav input?
> ___


Shubham,

If you want to have compressed audio in a wav wrapper, try this (but the 
quality will be bad because it’s compressed)

ffmpeg -i source.wav -b:a 120k -c:a aac output.wav

The “-c:a aac” switch sets the codec to aac (compressed) rather than 
uncompressed PCM. “-b:a” to set the data rate for the audio, if needed.
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Re: [FFmpeg-user] Large Sized output files recieved while encoding the audio

[Shubham Tiwari]

Rephrasing the requirement.

We have an audio pipeline which redacts the audios using the set of start
time and end time provided from our ML team. To achieve this, we execute
the ffmpeg command with filter
"volume=enable='between(t,starttime,endtime)':volume=0".
For some cases, we are receiving audio files with codecs which are causing
the outputs to have large file sizes. One such example is below,
ffprobe and ffmpeg output of the file is attached to the mail.
*command:* ffmpeg -i call.wav -af "volume=enable='between(t,0,1)':volume=0"
output.wav
*behaviour: *executed within a second, input file size 2.1 mb, output file
size 176 mb

As we can see here, the output file size is 88 times the original. The
objective here is to achieve the output file size of less than 2-3 times
the original without reducing the sound quality.
I went through the documentation but couldn't find any suitable way to fix
it. Let me know if anyone knows the issue in my command. Please point me to
some resource containing the details of audio codecs and filtering.

Regards,
Shubham


On Wed, Apr 20, 2022 at 11:33 PM Shubham Tiwari 
wrote:

> Thanks Bouke for the opinion.
> Help from any other person is appreciated!!
>
> Regards,
> Shubham
>
> On Wed, Apr 20, 2022 at 11:27 PM Bouke / edit 'B  wrote:
>
>>
>>
>> > On 20 Apr 2022, at 19:50, Shubham Tiwari 
>> wrote:
>> >
>> > 1 million audio files per day.
>>
>> Hire a specialist. Now. Immediately. Pay him / her well. Do NOT hesitate.
>> Do NOT think, just do it. Right now.
>> It WILL pay off.
>> Did I mention you DO NOT HAVE time to waste?
>>
>>
>> Bouke / edit 'B
>>
>> videotoolshed.com
>> Van Oldenbarneveltstraat 33
>> 6512 AS Nijmegen, the Netherlands
>> +31 6 21817248
>> If you want to send me large files, please use:
>> https://videotoolshed.wetransfer.com/ <
>> https://videotoolshed.wetransfer.com/>
>>
>>
>>
>>
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>>
>
~/Downloads/ffprobe -i call.wav 
ffprobe version 4.4.1-tessus  https://evermeet.cx/ffmpeg/  Copyright (c) 
2007-2021 the FFmpeg developers
  built with Apple clang version 11.0.0 (clang-1100.0.33.17)
  configuration: --cc=/usr/bin/clang --prefix=/opt/ffmpeg 
--extra-version=tessus --enable-avisynth --enable-fontconfig --enable-gpl 
--enable-libaom --enable-libass --enable-libbluray --enable-libdav1d 
--enable-libfreetype --enable-libgsm --enable-libmodplug --enable-libmp3lame 
--enable-libmysofa --enable-libopencore-amrnb --enable-libopencore-amrwb 
--enable-libopenh264 --enable-libopenjpeg --enable-libopus 
--enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr 
--enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab 
--enable-libvmaf --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx 
--enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs 
--enable-libxvid --enable-libzimg --enable-libzmq --enable-libzvbi 
--enable-version3 --pkg-config-flags=--static --disable-ffplay
  libavutil  56. 70.100 / 56. 70.100
  libavcodec 58.134.100 / 58.134.100
  libavformat58. 76.100 / 58. 76.100
  libavdevice58. 13.100 / 58. 13.100
  libavfilter 7.110.100 /  7.110.100
  libswscale  5.  9.100 /  5.  9.100
  libswresample   3.  9.100 /  3.  9.100
  libpostproc55.  9.100 / 55.  9.100
Input #0, ogg, from 'call.wav':
  Duration: 00:15:36.90, start: 0.00, bitrate: 19 kb/s
  Stream #0:0: Audio: opus, 48000 Hz, stereo, fltp

--

With WAV output
 ~/Downloads/ffmpeg -i call.wav -af "volume=enable='between(t,0,1)':volume=0" 
output.wav
ffmpeg version 4.4.1-tessus  https://evermeet.cx/ffmpeg/  Copyright (c) 
2000-2021 the FFmpeg developers
  built with Apple clang version 11.0.0 (clang-1100.0.33.17)
  configuration: --cc=/usr/bin/clang --prefix=/opt/ffmpeg 
--extra-version=tessus --enable-avisynth --enable-fontconfig --enable-gpl 
--enable-libaom --enable-libass --enable-libbluray --enable-libdav1d 
--enable-libfreetype --enable-libgsm --enable-libmodplug --enable-libmp3lame 
--enable-libmysofa --enable-libopencore-amrnb --enable-libopencore-amrwb 
--enable-libopenh264 --enable-libopenjpeg --enable-libopus 
--enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr 
--enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab 
--enable-libvmaf --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx 
--enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs 
--enable-libxvid --enable-libzimg --enable-libzmq --enable-libzvbi 
--enable-version3 --pkg-config-flags=--static --disable-ffplay
  libavutil  56. 70.100 / 56. 70.100
  libavcodec 

Re: [FFmpeg-user] Large Sized output files recieved while encoding the audio

[morgan holly via ffmpeg-user]

> On Apr 20, 2022, at 12:04 PM, Shubham Tiwari  
> wrote:
> 
> Thanks Bouke for the opinion.
> Help from any other person is appreciated!!
> 
> Regards,
> Shubham
> 

I have not read the entire thread, but it looks like your source is highly 
compressed, your output is uncompressed. So yes, in this scenario you should 
expect a much larger file size on the output. Does your redaction software only 
work with a wav input?
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Re: [FFmpeg-user] Large Sized output files recieved while encoding the audio

[Shubham Tiwari]

Thanks Bouke for the opinion.
Help from any other person is appreciated!!

Regards,
Shubham

On Wed, Apr 20, 2022 at 11:27 PM Bouke / edit 'B  wrote:

>
>
> > On 20 Apr 2022, at 19:50, Shubham Tiwari 
> wrote:
> >
> > 1 million audio files per day.
>
> Hire a specialist. Now. Immediately. Pay him / her well. Do NOT hesitate.
> Do NOT think, just do it. Right now.
> It WILL pay off.
> Did I mention you DO NOT HAVE time to waste?
>
>
> Bouke / edit 'B
>
> videotoolshed.com
> Van Oldenbarneveltstraat 33
> 6512 AS Nijmegen, the Netherlands
> +31 6 21817248
> If you want to send me large files, please use:
> https://videotoolshed.wetransfer.com/ <
> https://videotoolshed.wetransfer.com/>
>
>
>
>
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Re: [FFmpeg-user] Large Sized output files recieved while encoding the audio

[Bouke / edit 'B]

> On 20 Apr 2022, at 19:50, Shubham Tiwari  wrote:
> 
> 1 million audio files per day.

Hire a specialist. Now. Immediately. Pay him / her well. Do NOT hesitate. Do 
NOT think, just do it. Right now.
It WILL pay off.
Did I mention you DO NOT HAVE time to waste?


Bouke / edit 'B

videotoolshed.com
Van Oldenbarneveltstraat 33
6512 AS Nijmegen, the Netherlands
+31 6 21817248
If you want to send me large files, please use:
https://videotoolshed.wetransfer.com/ 




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Re: [FFmpeg-user] Large Sized output files recieved while encoding the audio

[Shubham Tiwari]

I think you are not reading the message properly. If you notice, the size
of the file i am passing is 2 mb, the return file size is 176 mb. Does 88
times increase seem reasonable to you?
I want the file to be in approximately similar quality and file size should
go at max 2 to 3 times. In this case upto 6 mb file size is fine.

I am doing this because we have a redaction logic to be executed on
approximately 1 million audio files per day.  And all these calls are
stored in aws s3. so storing multiple files of size 176 mb doesn't seem to
be a good idea.

Concerning my lack of knowledge, I read the documentation page multiple
times but couldn't find something that gives me the right information.  In
the end I reached out here. Concerning my lack of knowledge, probably some
help would be better instead of passing remarks.

On Wed, Apr 20, 2022 at 11:07 PM Bouke / Videotoolshed <
bo...@videotoolshed.com> wrote:

>
> > On 20 Apr 2022, at 19:30, Shubham Tiwari 
> wrote:
> >
> > Please accept my apology for putting a large size msg. I appreciate all
> the
> > help received.
> >
> > I have another example with similar behavior. The exact commands used are
> > below. The detailed output is attached in this email.
> >
> > *command 1*: ffmpeg -i call.wav -af
> > "volume=enable='between(t,0,1)':volume=0" output.wav
> > *behaviour: *executed within a second, input file size 2.1 mb, output
> file
> > size 176 mb
> >
> > *command 2*: ffmpeg -i call.wav -af
> > "volume=enable='between(t,0,1)':volume=0" output.mp3
> > *behaviour: *executed time approx 40 seconds, input file size 2.1 mb,
> > output file size 15 mb
>
> You do not specify what kind of output file size you want, so yes, it will
> be ‘as it is’.
> Having a .wav extension on your input DOES NOT mean your input is
> uncompressed Wave PCM. (Wave Can be compressed, in your case, you have OGG
> compression with a .wav extension.)
>
> What do you expect / why are you doing this? The quality of your file will
> NOT get better, nor will the file size get significant down if you process
> it.
>
> Concerning your lack of knowledge, the only thing you can accomplish is
> making things worse.
>
> Bouke
>
>
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Re: [FFmpeg-user] Large Sized output files recieved while encoding the audio

[Bouke / Videotoolshed]

> On 20 Apr 2022, at 19:30, Shubham Tiwari  wrote:
> 
> Please accept my apology for putting a large size msg. I appreciate all the
> help received.
> 
> I have another example with similar behavior. The exact commands used are
> below. The detailed output is attached in this email.
> 
> *command 1*: ffmpeg -i call.wav -af
> "volume=enable='between(t,0,1)':volume=0" output.wav
> *behaviour: *executed within a second, input file size 2.1 mb, output file
> size 176 mb
> 
> *command 2*: ffmpeg -i call.wav -af
> "volume=enable='between(t,0,1)':volume=0" output.mp3
> *behaviour: *executed time approx 40 seconds, input file size 2.1 mb,
> output file size 15 mb

You do not specify what kind of output file size you want, so yes, it will be 
‘as it is’.
Having a .wav extension on your input DOES NOT mean your input is uncompressed 
Wave PCM. (Wave Can be compressed, in your case, you have OGG compression with 
a .wav extension.)

What do you expect / why are you doing this? The quality of your file will NOT 
get better, nor will the file size get significant down if you process it.

Concerning your lack of knowledge, the only thing you can accomplish is making 
things worse.

Bouke


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Re: [FFmpeg-user] Large Sized output files recieved while encoding the audio

[Shubham Tiwari]

Please accept my apology for putting a large size msg. I appreciate all the
help received.

I have another example with similar behavior. The exact commands used are
below. The detailed output is attached in this email.

*command 1*: ffmpeg -i call.wav -af
"volume=enable='between(t,0,1)':volume=0" output.wav
*behaviour: *executed within a second, input file size 2.1 mb, output file
size 176 mb

*command 2*: ffmpeg -i call.wav -af
"volume=enable='between(t,0,1)':volume=0" output.mp3
*behaviour: *executed time approx 40 seconds, input file size 2.1 mb,
output file size 15 mb

Thanks for the support.
Regards,
Shubham
~/Downloads/ffprobe -i call.wav 
ffprobe version 4.4.1-tessus  https://evermeet.cx/ffmpeg/  Copyright (c) 
2007-2021 the FFmpeg developers
  built with Apple clang version 11.0.0 (clang-1100.0.33.17)
  configuration: --cc=/usr/bin/clang --prefix=/opt/ffmpeg 
--extra-version=tessus --enable-avisynth --enable-fontconfig --enable-gpl 
--enable-libaom --enable-libass --enable-libbluray --enable-libdav1d 
--enable-libfreetype --enable-libgsm --enable-libmodplug --enable-libmp3lame 
--enable-libmysofa --enable-libopencore-amrnb --enable-libopencore-amrwb 
--enable-libopenh264 --enable-libopenjpeg --enable-libopus 
--enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr 
--enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab 
--enable-libvmaf --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx 
--enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs 
--enable-libxvid --enable-libzimg --enable-libzmq --enable-libzvbi 
--enable-version3 --pkg-config-flags=--static --disable-ffplay
  libavutil  56. 70.100 / 56. 70.100
  libavcodec 58.134.100 / 58.134.100
  libavformat58. 76.100 / 58. 76.100
  libavdevice58. 13.100 / 58. 13.100
  libavfilter 7.110.100 /  7.110.100
  libswscale  5.  9.100 /  5.  9.100
  libswresample   3.  9.100 /  3.  9.100
  libpostproc55.  9.100 / 55.  9.100
Input #0, ogg, from 'call.wav':
  Duration: 00:15:36.90, start: 0.00, bitrate: 19 kb/s
  Stream #0:0: Audio: opus, 48000 Hz, stereo, fltp

--

With WAV output
 ~/Downloads/ffmpeg -i call.wav -af "volume=enable='between(t,0,1)':volume=0" 
output.wav
ffmpeg version 4.4.1-tessus  https://evermeet.cx/ffmpeg/  Copyright (c) 
2000-2021 the FFmpeg developers
  built with Apple clang version 11.0.0 (clang-1100.0.33.17)
  configuration: --cc=/usr/bin/clang --prefix=/opt/ffmpeg 
--extra-version=tessus --enable-avisynth --enable-fontconfig --enable-gpl 
--enable-libaom --enable-libass --enable-libbluray --enable-libdav1d 
--enable-libfreetype --enable-libgsm --enable-libmodplug --enable-libmp3lame 
--enable-libmysofa --enable-libopencore-amrnb --enable-libopencore-amrwb 
--enable-libopenh264 --enable-libopenjpeg --enable-libopus 
--enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr 
--enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab 
--enable-libvmaf --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx 
--enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs 
--enable-libxvid --enable-libzimg --enable-libzmq --enable-libzvbi 
--enable-version3 --pkg-config-flags=--static --disable-ffplay
  libavutil  56. 70.100 / 56. 70.100
  libavcodec 58.134.100 / 58.134.100
  libavformat58. 76.100 / 58. 76.100
  libavdevice58. 13.100 / 58. 13.100
  libavfilter 7.110.100 /  7.110.100
  libswscale  5.  9.100 /  5.  9.100
  libswresample   3.  9.100 /  3.  9.100
  libpostproc55.  9.100 / 55.  9.100
Input #0, ogg, from 'call.wav':
  Duration: 00:15:36.90, start: 0.00, bitrate: 19 kb/s
  Stream #0:0: Audio: opus, 48000 Hz, stereo, fltp
Stream mapping:
  Stream #0:0 -> #0:0 (opus (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
Output #0, wav, to 'output.wav':
  Metadata:
ISFT: Lavf58.76.100
  Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, s16, 
1536 kb/s
Metadata:
  encoder : Lavc58.134.100 pcm_s16le
[opus @ 0x7f78d8009800] Error parsing the packet header.peed= 609x
Error while decoding stream #0:0: Invalid data found when processing input
[opus @ 0x7f78d8009800] Error parsing the packet header.
Error while decoding stream #0:0: Invalid data found when processing input
[opus @ 0x7f78d8009800] Error parsing the packet header.
Error while decoding stream #0:0: Invalid data found when processing input
[opus @ 0x7f78d8009800] Error parsing the packet header.
Error while decoding stream #0:0: Invalid data found when processing input
[opus @ 0x7f78d8009800] Error parsing the packet header.
Error while decoding stream #0:0: Invalid data found when processing input
[opus @ 0x7f78d8009800] Error parsing the packet header.
Error while decoding stream #0:0: 

Re: [FFmpeg-user] Large Sized output files recieved while encoding the audio

[Bouke / edit 'B]

Hi Shubham,
When I copy your mail below (not done on this list, mind you!), it’s exactly as 
long as it was before. What do you expect?
Bouke



> On 20 Apr 2022, at 15:01, Shubham Tiwari  wrote:
> 
> Hi,
> 
> When I run ffmpeg command on a wav file, the received output file is of
> large size as compared to the original file. The command and the output are
> below,
> 
> *FFMPEG command*
> 
> % ~/Downloads/audio/ffmpeg -i call-redacted.wav output.wav
> 
> ffmpeg version 4.4.1-tessus  https://evermeet.cx/ffmpeg/  Copyright (c)
> 2000-2021 the FFmpeg developers
> 
>  built with Apple clang version 11.0.0 (clang-1100.0.33.17)
> 
>  configuration: --cc=/usr/bin/clang --prefix=/opt/ffmpeg
> --extra-version=tessus --enable-avisynth --enable-fontconfig --enable-gpl
> --enable-libaom --enable-libass --enable-libbluray --enable-libdav1d
> --enable-libfreetype --enable-libgsm --enable-libmodplug
> --enable-libmp3lame --enable-libmysofa --enable-libopencore-amrnb
> --enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg
> --enable-libopus --enable-librubberband --enable-libshine
> --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora
> --enable-libtwolame --enable-libvidstab --enable-libvmaf
> --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwebp
> --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid
> --enable-libzimg --enable-libzmq --enable-libzvbi --enable-version3
> --pkg-config-flags=--static --disable-ffplay
> 
>  libavutil  56. 70.100 / 56. 70.100
> 
>  libavcodec 58.134.100 / 58.134.100
> 
>  libavformat58. 76.100 / 58. 76.100
> 
>  libavdevice58. 13.100 / 58. 13.100
> 
>  libavfilter 7.110.100 /  7.110.100
> 
>  libswscale  5.  9.100 /  5.  9.100
> 
>  libswresample   3.  9.100 /  3.  9.100
> 
>  libpostproc55.  9.100 / 55.  9.100
> 
> Input #0, mp3, from 'call-redacted.wav':
> 
>  Metadata:
> 
>encoder : Lavf58.45.100
> 
>  Duration: 00:13:06.38, start: 0.138125, bitrate: 64 kb/s
> 
>  Stream #0:0: Audio: mp3, 8000 Hz, stereo, fltp, 64 kb/s
> 
> Stream mapping:
> 
>  Stream #0:0 -> #0:0 (mp3 (mp3float) -> pcm_s16le (native))
> 
> Press [q] to stop, [?] for help
> 
> Output #0, wav, to 'output.wav':
> 
>  Metadata:
> 
>ISFT: Lavf58.76.100
> 
>  Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 8000 Hz, stereo,
> s16, 256 kb/s
> 
>Metadata:
> 
>  encoder : Lavc58.134.100 pcm_s16le
> 
> size=   24569kB time=00:13:06.17 bitrate= 256.0kbits/s speed=2.97e+03x
> 
> video:0kB audio:24569kB subtitle:0kB other streams:0kB global headers:0kB
> muxing overhead: 0.000310
> 
> 
> *Input and output file sizes*
> 
> du -sh call-redacted.wav
> 
> 6.5M call-redacted.wav
> 
> du -sh output.wav
> 
> 24M output.wav
> 
> 
> Alternatively, when I run the same command but change the output
> file's extension to mp3, the correct sized file is returned. But the time
> taken (6 seconds) in this case is quite higher than the previous case(less
> than 1 second). The command and output is below,
> 
> 
> 
> ~/Downloads/audio/ffmpeg -i call-redacted.wav output.mp3
> 
> ffmpeg version 4.4.1-tessus  https://evermeet.cx/ffmpeg/  Copyright (c)
> 2000-2021 the FFmpeg developers
> 
>  built with Apple clang version 11.0.0 (clang-1100.0.33.17)
> 
>  configuration: --cc=/usr/bin/clang --prefix=/opt/ffmpeg
> --extra-version=tessus --enable-avisynth --enable-fontconfig --enable-gpl
> --enable-libaom --enable-libass --enable-libbluray --enable-libdav1d
> --enable-libfreetype --enable-libgsm --enable-libmodplug
> --enable-libmp3lame --enable-libmysofa --enable-libopencore-amrnb
> --enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg
> --enable-libopus --enable-librubberband --enable-libshine
> --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora
> --enable-libtwolame --enable-libvidstab --enable-libvmaf
> --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwebp
> --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid
> --enable-libzimg --enable-libzmq --enable-libzvbi --enable-version3
> --pkg-config-flags=--static --disable-ffplay
> 
>  libavutil  56. 70.100 / 56. 70.100
> 
>  libavcodec 58.134.100 / 58.134.100
> 
>  libavformat58. 76.100 / 58. 76.100
> 
>  libavdevice58. 13.100 / 58. 13.100
> 
>  libavfilter 7.110.100 /  7.110.100
> 
>  libswscale  5.  9.100 /  5.  9.100
> 
>  libswresample   3.  9.100 /  3.  9.100
> 
>  libpostproc55.  9.100 / 55.  9.100
> 
> Input #0, mp3, from 'call-redacted.wav':
> 
>  Metadata:
> 
>encoder : Lavf58.45.100
> 
>  Duration: 00:13:06.38, start: 0.138125, bitrate: 64 kb/s
> 
>  Stream #0:0: Audio: mp3, 8000 Hz, stereo, fltp, 64 kb/s
> 
> Stream mapping:
> 
>  Stream #0:0 -> #0:0 (mp3 (mp3float) -> mp3 (libmp3lame))
> 
> Press [q] to stop, [?] for help
> 
> Output #0, mp3, to 'output.mp3':
> 
>  Metadata:
> 
>TSSE: Lavf58.76.100
> 
>  Stream #0:0: 

Re: [FFmpeg-user] Large Sized output files recieved while encoding the audio

[Carl Zwanzig]

On 4/20/2022 9:31 AM, Shubham Tiwari wrote:


My use case is redaction of audio. To achieve this, our backend adds the
filter to mute the audio in the ffmpeg command and then executes on audio
files. 


What are those commands? The original email only shows basic file conversion.


 When we use the extension of the output
audio file as mp3, then the time taken is high. When we use the original
extension (wav) in the output audio file, then the file size is high (for
some cases, the size goes upto 250mb+).


To expand on what Ferdi said- on input the file extension is only used to 
make a guess about the format; on output it's used to select that format 
unless told otherwise. So if the output file is called file.wav, ffmpeg is 
going to write an uncompressed wav; if it's file.mp3, ffmpeg will write an 
mp3 file using default parameters. All of that can be overridden using 
command line parameters.


Please read parts 1-7 of the doc at https://ffmpeg.org/ffmpeg-all.html to 
see how the parameters & options behave.




Do you know the reason why reducing the quality even further takes
longer?
That's decoding the compressed data into raw/uncompressed date, the 
re-compressing/encoding; more steps, more work.


Also, please read https://ffmpeg.org/mailing-list-faq.html

Later,

z!
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Re: [FFmpeg-user] Large Sized output files recieved while encoding the audio

[Shubham Tiwari]

Hi Ferdi,

My use case is redaction of audio. To achieve this, our backend adds the
filter to mute the audio in the ffmpeg command and then executes on audio
files. For some audio files it's taking more than 30 seconds which is
causing high cpu usage and low throughput. The problem in our scenario is
the same as I mentioned above. When we use the extension of the output
audio file as mp3, then the time taken is high. When we use the original
extension (wav) in the output audio file, then the file size is high (for
some cases, the size goes upto 250mb+).

Do you know what we are doing wrong? Any help would be appreciated.



*Your second command actually encodes the already encoded file
again,reducing the quality of the audio even further as 64 kb is already
verylow quality for an mp3 file. This is also why it takes longer.*

Do you know the reason why reducing the quality even further takes longer?

Regards,
Shubham

On Wed, Apr 20, 2022 at 8:08 PM Ferdi Scholten  wrote:

> Hi,
> > When I run ffmpeg command on a wav file, the received output file is of
> > large size as compared to the original file. The command and the output
> are
> > below,
> >
> > *FFMPEG command*
> >
> >   % ~/Downloads/audio/ffmpeg -i call-redacted.wav output.wav
> >
> > ffmpeg version 4.4.1-tessus  https://evermeet.cx/ffmpeg/  Copyright (c)
> > 2000-2021 the FFmpeg developers
> >
> >built with Apple clang version 11.0.0 (clang-1100.0.33.17)
> >
> >configuration: --cc=/usr/bin/clang --prefix=/opt/ffmpeg
> > --extra-version=tessus --enable-avisynth --enable-fontconfig --enable-gpl
> > --enable-libaom --enable-libass --enable-libbluray --enable-libdav1d
> > --enable-libfreetype --enable-libgsm --enable-libmodplug
> > --enable-libmp3lame --enable-libmysofa --enable-libopencore-amrnb
> > --enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg
> > --enable-libopus --enable-librubberband --enable-libshine
> > --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora
> > --enable-libtwolame --enable-libvidstab --enable-libvmaf
> > --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx
> --enable-libwebp
> > --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid
> > --enable-libzimg --enable-libzmq --enable-libzvbi --enable-version3
> > --pkg-config-flags=--static --disable-ffplay
> >
> >libavutil  56. 70.100 / 56. 70.100
> >
> >libavcodec 58.134.100 / 58.134.100
> >
> >libavformat58. 76.100 / 58. 76.100
> >
> >libavdevice58. 13.100 / 58. 13.100
> >
> >libavfilter 7.110.100 /  7.110.100
> >
> >libswscale  5.  9.100 /  5.  9.100
> >
> >libswresample   3.  9.100 /  3.  9.100
> >
> >libpostproc55.  9.100 / 55.  9.100
> >
> > Input #0, mp3, from 'call-redacted.wav':
> >
> >Metadata:
> >
> >  encoder : Lavf58.45.100
> >
> >Duration: 00:13:06.38, start: 0.138125, bitrate: 64 kb/s
> >
> >Stream #0:0: Audio: mp3, 8000 Hz, stereo, fltp, 64 kb/s
> >
> > Stream mapping:
> >
> >Stream #0:0 -> #0:0 (mp3 (mp3float) -> pcm_s16le (native))
> >
> > Press [q] to stop, [?] for help
> >
> > Output #0, wav, to 'output.wav':
> >
> >Metadata:
> >
> >  ISFT: Lavf58.76.100
> >
> >Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 8000 Hz,
> stereo,
> > s16, 256 kb/s
> >
> >  Metadata:
> >
> >encoder : Lavc58.134.100 pcm_s16le
> >
> > size=   24569kB time=00:13:06.17 bitrate= 256.0kbits/s speed=2.97e+03x
> >
> > video:0kB audio:24569kB subtitle:0kB other streams:0kB global headers:0kB
> > muxing overhead: 0.000310
> >
> >
> > *Input and output file sizes*
> >
> > du -sh call-redacted.wav
> >
> > 6.5M call-redacted.wav
> >
> > du -sh output.wav
> >
> >   24M output.wav
> >
> >
> > Alternatively, when I run the same command but change the output
> > file's extension to mp3, the correct sized file is returned. But the time
> > taken (6 seconds) in this case is quite higher than the previous
> case(less
> > than 1 second). The command and output is below,
> >
> >
> >
> > ~/Downloads/audio/ffmpeg -i call-redacted.wav output.mp3
> >
> > ffmpeg version 4.4.1-tessus  https://evermeet.cx/ffmpeg/  Copyright (c)
> > 2000-2021 the FFmpeg developers
> >
> >built with Apple clang version 11.0.0 (clang-1100.0.33.17)
> >
> >configuration: --cc=/usr/bin/clang --prefix=/opt/ffmpeg
> > --extra-version=tessus --enable-avisynth --enable-fontconfig --enable-gpl
> > --enable-libaom --enable-libass --enable-libbluray --enable-libdav1d
> > --enable-libfreetype --enable-libgsm --enable-libmodplug
> > --enable-libmp3lame --enable-libmysofa --enable-libopencore-amrnb
> > --enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg
> > --enable-libopus --enable-librubberband --enable-libshine
> > --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora
> > --enable-libtwolame --enable-libvidstab --enable-libvmaf
> > --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx
> 

Re: [FFmpeg-user] Large Sized output files recieved while encoding the audio

[Ferdi Scholten]

Hi,

When I run ffmpeg command on a wav file, the received output file is of
large size as compared to the original file. The command and the output are
below,

*FFMPEG command*

  % ~/Downloads/audio/ffmpeg -i call-redacted.wav output.wav

ffmpeg version 4.4.1-tessus  https://evermeet.cx/ffmpeg/  Copyright (c)
2000-2021 the FFmpeg developers

   built with Apple clang version 11.0.0 (clang-1100.0.33.17)

   configuration: --cc=/usr/bin/clang --prefix=/opt/ffmpeg
--extra-version=tessus --enable-avisynth --enable-fontconfig --enable-gpl
--enable-libaom --enable-libass --enable-libbluray --enable-libdav1d
--enable-libfreetype --enable-libgsm --enable-libmodplug
--enable-libmp3lame --enable-libmysofa --enable-libopencore-amrnb
--enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg
--enable-libopus --enable-librubberband --enable-libshine
--enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora
--enable-libtwolame --enable-libvidstab --enable-libvmaf
--enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwebp
--enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid
--enable-libzimg --enable-libzmq --enable-libzvbi --enable-version3
--pkg-config-flags=--static --disable-ffplay

   libavutil  56. 70.100 / 56. 70.100

   libavcodec 58.134.100 / 58.134.100

   libavformat58. 76.100 / 58. 76.100

   libavdevice58. 13.100 / 58. 13.100

   libavfilter 7.110.100 /  7.110.100

   libswscale  5.  9.100 /  5.  9.100

   libswresample   3.  9.100 /  3.  9.100

   libpostproc55.  9.100 / 55.  9.100

Input #0, mp3, from 'call-redacted.wav':

   Metadata:

 encoder : Lavf58.45.100

   Duration: 00:13:06.38, start: 0.138125, bitrate: 64 kb/s

   Stream #0:0: Audio: mp3, 8000 Hz, stereo, fltp, 64 kb/s

Stream mapping:

   Stream #0:0 -> #0:0 (mp3 (mp3float) -> pcm_s16le (native))

Press [q] to stop, [?] for help

Output #0, wav, to 'output.wav':

   Metadata:

 ISFT: Lavf58.76.100

   Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 8000 Hz, stereo,
s16, 256 kb/s

 Metadata:

   encoder : Lavc58.134.100 pcm_s16le

size=   24569kB time=00:13:06.17 bitrate= 256.0kbits/s speed=2.97e+03x

video:0kB audio:24569kB subtitle:0kB other streams:0kB global headers:0kB
muxing overhead: 0.000310


*Input and output file sizes*

du -sh call-redacted.wav

6.5M call-redacted.wav

du -sh output.wav

  24M output.wav


Alternatively, when I run the same command but change the output
file's extension to mp3, the correct sized file is returned. But the time
taken (6 seconds) in this case is quite higher than the previous case(less
than 1 second). The command and output is below,



~/Downloads/audio/ffmpeg -i call-redacted.wav output.mp3

ffmpeg version 4.4.1-tessus  https://evermeet.cx/ffmpeg/  Copyright (c)
2000-2021 the FFmpeg developers

   built with Apple clang version 11.0.0 (clang-1100.0.33.17)

   configuration: --cc=/usr/bin/clang --prefix=/opt/ffmpeg
--extra-version=tessus --enable-avisynth --enable-fontconfig --enable-gpl
--enable-libaom --enable-libass --enable-libbluray --enable-libdav1d
--enable-libfreetype --enable-libgsm --enable-libmodplug
--enable-libmp3lame --enable-libmysofa --enable-libopencore-amrnb
--enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg
--enable-libopus --enable-librubberband --enable-libshine
--enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora
--enable-libtwolame --enable-libvidstab --enable-libvmaf
--enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwebp
--enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid
--enable-libzimg --enable-libzmq --enable-libzvbi --enable-version3
--pkg-config-flags=--static --disable-ffplay

   libavutil  56. 70.100 / 56. 70.100

   libavcodec 58.134.100 / 58.134.100

   libavformat58. 76.100 / 58. 76.100

   libavdevice58. 13.100 / 58. 13.100

   libavfilter 7.110.100 /  7.110.100

   libswscale  5.  9.100 /  5.  9.100

   libswresample   3.  9.100 /  3.  9.100

   libpostproc55.  9.100 / 55.  9.100

Input #0, mp3, from 'call-redacted.wav':

   Metadata:

 encoder : Lavf58.45.100

   Duration: 00:13:06.38, start: 0.138125, bitrate: 64 kb/s

   Stream #0:0: Audio: mp3, 8000 Hz, stereo, fltp, 64 kb/s

Stream mapping:

   Stream #0:0 -> #0:0 (mp3 (mp3float) -> mp3 (libmp3lame))

Press [q] to stop, [?] for help

Output #0, mp3, to 'output.mp3':

   Metadata:

 TSSE: Lavf58.76.100

   Stream #0:0: Audio: mp3, 8000 Hz, stereo, fltp

 Metadata:

   encoder : Lavc58.134.100 libmp3lame

size=2304kB time=00:13:06.17 bitrate=  24.0kbits/s speed= 130x

video:0kB audio:2304kB subtitle:0kB other streams:0kB global headers:0kB
muxing overhead: 0.011063

du -sh output.mp3

3.3M output.mp3


Please help me out in identifying the right command options for optimum
time and filesize.


Regards,

Shubham

[FFmpeg-user] What versions get point releases (active support)

[B S]

Hi

I am a maintainer of https://endoflife.date/. It is a community-maintained 
project to document end-of-life dates, and support lifecycles of various 
products.

We have ffmpeg listed there as well. With the latest stable releases of ffmpeg 
a discussion started about what versions get point releases (active support) at 
the moment.

I think all releases on https://ffmpeg.org/download.html are getting active 
support:

FFmpeg 5.0.1 "Lorentz"
FFmpeg 4.4.2 "Rao"
FFmpeg 4.3.4 "4:3"
FFmpeg 4.2.6 "Ada"
FFmpeg 4.1.9 "al-Khwarizmi"
FFmpeg 3.4.9 "Cantor"
FFmpeg 3.2.16 "Hypatia"
FFmpeg 2.8.18 "Feynman"

And all releases on https://ffmpeg.org/olddownload.html are no longer supported 
(end of life and no longer recommended for use).

I also saw the page https://ffmpeg.org/security.html where all versions with 
the vulnerabilities are listed.

It would be helpful if someone from the team can confirm or correct my above 
assumtions.

Thanks and have a nice day
BiNZGi
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[FFmpeg-user] Large Sized output files recieved while encoding the audio

[Shubham Tiwari]

Hi,

When I run ffmpeg command on a wav file, the received output file is of
large size as compared to the original file. The command and the output are
below,

*FFMPEG command*

 % ~/Downloads/audio/ffmpeg -i call-redacted.wav output.wav

ffmpeg version 4.4.1-tessus  https://evermeet.cx/ffmpeg/  Copyright (c)
2000-2021 the FFmpeg developers

  built with Apple clang version 11.0.0 (clang-1100.0.33.17)

  configuration: --cc=/usr/bin/clang --prefix=/opt/ffmpeg
--extra-version=tessus --enable-avisynth --enable-fontconfig --enable-gpl
--enable-libaom --enable-libass --enable-libbluray --enable-libdav1d
--enable-libfreetype --enable-libgsm --enable-libmodplug
--enable-libmp3lame --enable-libmysofa --enable-libopencore-amrnb
--enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg
--enable-libopus --enable-librubberband --enable-libshine
--enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora
--enable-libtwolame --enable-libvidstab --enable-libvmaf
--enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwebp
--enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid
--enable-libzimg --enable-libzmq --enable-libzvbi --enable-version3
--pkg-config-flags=--static --disable-ffplay

  libavutil  56. 70.100 / 56. 70.100

  libavcodec 58.134.100 / 58.134.100

  libavformat58. 76.100 / 58. 76.100

  libavdevice58. 13.100 / 58. 13.100

  libavfilter 7.110.100 /  7.110.100

  libswscale  5.  9.100 /  5.  9.100

  libswresample   3.  9.100 /  3.  9.100

  libpostproc55.  9.100 / 55.  9.100

Input #0, mp3, from 'call-redacted.wav':

  Metadata:

encoder : Lavf58.45.100

  Duration: 00:13:06.38, start: 0.138125, bitrate: 64 kb/s

  Stream #0:0: Audio: mp3, 8000 Hz, stereo, fltp, 64 kb/s

Stream mapping:

  Stream #0:0 -> #0:0 (mp3 (mp3float) -> pcm_s16le (native))

Press [q] to stop, [?] for help

Output #0, wav, to 'output.wav':

  Metadata:

ISFT: Lavf58.76.100

  Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 8000 Hz, stereo,
s16, 256 kb/s

Metadata:

  encoder : Lavc58.134.100 pcm_s16le

size=   24569kB time=00:13:06.17 bitrate= 256.0kbits/s speed=2.97e+03x

video:0kB audio:24569kB subtitle:0kB other streams:0kB global headers:0kB
muxing overhead: 0.000310


*Input and output file sizes*

du -sh call-redacted.wav

6.5M call-redacted.wav

du -sh output.wav

 24M output.wav


Alternatively, when I run the same command but change the output
file's extension to mp3, the correct sized file is returned. But the time
taken (6 seconds) in this case is quite higher than the previous case(less
than 1 second). The command and output is below,



~/Downloads/audio/ffmpeg -i call-redacted.wav output.mp3

ffmpeg version 4.4.1-tessus  https://evermeet.cx/ffmpeg/  Copyright (c)
2000-2021 the FFmpeg developers

  built with Apple clang version 11.0.0 (clang-1100.0.33.17)

  configuration: --cc=/usr/bin/clang --prefix=/opt/ffmpeg
--extra-version=tessus --enable-avisynth --enable-fontconfig --enable-gpl
--enable-libaom --enable-libass --enable-libbluray --enable-libdav1d
--enable-libfreetype --enable-libgsm --enable-libmodplug
--enable-libmp3lame --enable-libmysofa --enable-libopencore-amrnb
--enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg
--enable-libopus --enable-librubberband --enable-libshine
--enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora
--enable-libtwolame --enable-libvidstab --enable-libvmaf
--enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwebp
--enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid
--enable-libzimg --enable-libzmq --enable-libzvbi --enable-version3
--pkg-config-flags=--static --disable-ffplay

  libavutil  56. 70.100 / 56. 70.100

  libavcodec 58.134.100 / 58.134.100

  libavformat58. 76.100 / 58. 76.100

  libavdevice58. 13.100 / 58. 13.100

  libavfilter 7.110.100 /  7.110.100

  libswscale  5.  9.100 /  5.  9.100

  libswresample   3.  9.100 /  3.  9.100

  libpostproc55.  9.100 / 55.  9.100

Input #0, mp3, from 'call-redacted.wav':

  Metadata:

encoder : Lavf58.45.100

  Duration: 00:13:06.38, start: 0.138125, bitrate: 64 kb/s

  Stream #0:0: Audio: mp3, 8000 Hz, stereo, fltp, 64 kb/s

Stream mapping:

  Stream #0:0 -> #0:0 (mp3 (mp3float) -> mp3 (libmp3lame))

Press [q] to stop, [?] for help

Output #0, mp3, to 'output.mp3':

  Metadata:

TSSE: Lavf58.76.100

  Stream #0:0: Audio: mp3, 8000 Hz, stereo, fltp

Metadata:

  encoder : Lavc58.134.100 libmp3lame

size=2304kB time=00:13:06.17 bitrate=  24.0kbits/s speed= 130x

video:0kB audio:2304kB subtitle:0kB other streams:0kB global headers:0kB
muxing overhead: 0.011063

du -sh output.mp3

3.3M output.mp3


Please help me out in identifying the right command options for optimum
time and filesize.


Regards,

Shubham
___
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[FFmpeg-user] Hardware acceleration for converting PNGs into VP9

[Klimek, Denis]

Hi there,

I’ve got an question regarding conversation from multiple PNGs into a VP9 
movie. Currently we are doing this job with our CPU and this is quite slow… 
about 2.5fps in average and we have multiple movies to create every hour.
My idea was to accelerate it via an dedicated graphic card which can be 
installed into our server to speed this things up.

Which graphic card can handle this work for me and which parameters or 
additional software pieces are required? (Vendor, Model)

My current system is an Ubuntu 22.04 LTS 64bit and this is our current command 
line which converts the PNGs to an VP9:
ffmpeg -y -r 4.86 -pattern_type glob -i $DIR'/*.png' -c:v libvpx-vp9 -b:v 0 
-crf 30 -row-mt 1 -pix_fmt yuva420p -threads 8 -slices 16 -f webm 
$WORKINGDIR'/'$DIR'.webm.tmp'
(Codeline is snipped from a bash script which fills the variables with the 
necessary values)

Thanks for advance for any help 

Mit freundlichem Gruß
wilhelm.tel GmbH

Denis Klimek
Carrier Manager & Professional Network Engineer
IP-Systemtechnik
Tel:+49 (0) 40 / 521 04 – 1049 -> Work From Home
Mobil: +49 (0) 151 / 652 219 06

dkli...@stadtwerke-norderstedt.de
www.wilhelm-tel.de

__

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Postanschrift:
wilhelm.tel GmbH
Heidbergstraße 101-111
22846 Norderstedt

Geschäftsführer: Jens Seedorff, Theo Weirich
Vorsitzender des Aufsichtsrats: Christoph Mendel
Handelsregister: HRB 4216 NO, Amtsgericht Kiel
Umsatzsteuer ID: DE 81 299 7663

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