Re: [FFmpeg-user] ffmpeg Live Streaming Error : av_interleaved_write_frame(): Broken pipe

[KRISHNAKUMAR N K]

Can someone please help me with this?

On Thu, 18 Jun, 2020, 1:05 PM KRISHNAKUMAR N K, 
wrote:

> I am trying to stream video live using ffmpeg, Input is from a UDP Source.
> Most of the time it works fine, but I could see the following log lines and
> ffmpeg stop's the conversion/feed push to the rtmp server. Few log lines
> below for reference. Is this issue related to network on reaching the
> destination or source or something that i need to change in the ffmpeg
> command?
>
> *https://pastebin.com/WE0d6eN2 *
>
>
> *ffmpeg Command: *
> *ffmpeg -err_detect aggressive -fflags discardcorrupt -i
> udp://230.1.1.2:11000?fifo_size=500_nonfatal=1
> 
> -analyzeduration 25M -probesize 50M -threads 0 -vsync 1 -filter_complex
> [i:0x202]yadif,setdar=16/9[1out1];[i:0x202]yadif,setdar=16/9[1out2];[i:0x202]yadif,setdar=16/9[1out3];[i:0x202]yadif,setdar=16/9[1out4]
> -map [1out1] -map i:0x29e -c:v libx264 -x264-params nal-hrd=cbr -b:v 200k
> -maxrate 200k -bufsize 300k -r 15 -g 30 -profile:v baseline -level 3.0
> -pix_fmt yuv420p -c:a libfdk_aac -b:a 32k -ac 2 -ar 48000 -sc_threshold 0
> -s 256x144 -tune film -af
> aresample=async=1:min_hard_comp=0.10:first_pts=0 -x264opts opencl -f
> flv rtmp://ipaddress:1935/rtmp/push/test1-map [1out2] -map i:0x29e -c:v
> libx264 -x264-params nal-hrd=cbr -b:v 500k -maxrate 500k -bufsize 750k -r
> 25 -g 50 -profile:v baseline -level 3.0 -pix_fmt yuv420p -c:a libfdk_aac
> -b:a 64k -ac 2 -ar 48000 -sc_threshold 0 -s 512x288 -tune film -af
> aresample=async=1:min_hard_comp=0.10:first_pts=0 -x264opts opencl -f
> flv rtmp://ipaddress:1935/rtmp/push/test2 -map [1out3] -map i:0x29e -c:v
> libx264 -x264-params nal-hrd=cbr -b:v 1000k -maxrate 1000k -bufsize 1500k
> -r 25 -g 50 -profile:v main -level 3.0 -pix_fmt yuv420p -c:a libfdk_aac
> -b:a 64k -ac 2 -ar 48000 -sc_threshold 0 -s 1024x576 -tune film -af
> aresample=async=1:min_hard_comp=0.10:first_pts=0 -x264opts opencl -f
> flv rtmp://ipaddress:1935/rtmp/push/test3 -map [1out4] -map i:0x29e -c:v
> libx264 -x264-params nal-hrd=cbr -b:v 1800k -maxrate 1800k -bufsize 2700k
> -r 25 -g 50 -profile:v main -level 3.1 -pix_fmt yuv420p -c:a libfdk_aac
> -b:a 64k -ac 2 -ar 48000 -sc_threshold 0 -s 1280x720 -tune film -af
> aresample=async=1:min_hard_comp=0.10:first_pts=0 -x264opts opencl -f
> flv rtmp://ipaddress:1935/rtmp/push/test4*
>
> Regards
>
> *KrishnaKumar *
> *N K *
>
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[FFmpeg-user] Streaming quality

[Simon Roberts]

Greetings all,

I'm making my first experiments with streaming. Currently I do a live
capture of a screen region compress in real time and save the file. What I
want to do is to take that capture and send it to another machine for
storage where it will be embedded in a multi-stream container with other
sources.

So far in my experiments, I have a file I captured earlier and I'm
streaming that over to the remote system and using ffplay to view it,
rather than store it.

What I find is that the quality is trashed in the transmission, and I don't
know where to start looking for why. The original file is 720p screen
capture, and it's crisp and sharp. At the receiver, it shows what I might
describe as heavy "jpeg artifacts", that is, fuzzyness and penumbra around
all the edges of things like windows and text.

The commands I'm using are:

Receiver:
ffplay -i tcp://192.168.1.109:2000?listen

Transmitter:
ffmpeg -i  -f mpegts tcp://192.168.1.109:2000

Since no codecs are mentioned, I rather assumed that the file would be
streamed "as is", but that's clearly not the case.

Can anyone point me at what I need to know to make the transmission process
not alter the quality of the image?

Many thanks,
Simon


-- 
Simon Roberts
(303) 249 3613
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Re: [FFmpeg-user] 360 degree VR video streaming support in ffmpeg+HLS

[Rama Krishna Thelagathoti]

could someone help me with this?

On Thu, Jun 25, 2020 at 11:51 PM Rama Krishna Thelagathoti <
mrk.ramakris...@gmail.com> wrote:

> Hello ffmpeg team
> Does ffmpeg support 360 degree VR video streaming through HLS/DASH?
> I could see its being supported in libx264 under unofficial/experimental ,
> but I do not see this support in HLS.
>
> when transcoding to HLS, looks like ffmpeg drops spatial metadata. please
> confirm.
>
> --
> Best Regards
> *Rama krishna Thelagathoti*
>


-- 
Best Regards
*Rama krishna Thelagathoti*
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Re: [FFmpeg-user] newbie help - framerate

[Mark Foley]

On Sat, 27 Jun 2020 01:48:53 +0200 Carl Eugen Hoyos  wrote:
>
> > Am 27.06.2020 um 00:14 schrieb Mark Foley :
> > 
> > I am trying for the first time to use ffmpeg. I have a collection of .jpg 
> > files
> > starting at 000.jpg, numbered sequentially.
> > 
> > On Windows 10 I've tried:
> > 
> > ffmpeg.exe -start_number 000 -i %03d.jpg -r 1/6 mom.mp4
> > 
> > This works (images displayed for 10 seconds), but only shows 3 of the 300+
> > images. If I leave off the -r option it builds the .mp4 with all images, but
> > they flash by as fast as possible.
> > 
> > How do I get all images to show for 10 seconds?
>
> Funny that the option name “-framerate” is part of your subject.

Like I said, "newbie". I tried -framerate, but nothing happened. Then I tried
-r. Do you have suggestions on how to get all of the images processed to the
output and not just the first 3?

> > 
> > Problem #2.
> > 
> > The source .jpg images all vary in HxW size. How do I get them all to 
> > display
> > with the same height regardless of width?
>
> I believe you can only output one resolution but a shell script will help.
>
> Carl Eugen

How would a shell script help?

--Mark
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Re: [FFmpeg-user] newbie help - framerate

[Carl Eugen Hoyos]

> Am 27.06.2020 um 00:14 schrieb Mark Foley :
> 
> I am trying for the first time to use ffmpeg. I have a collection of .jpg 
> files
> starting at 000.jpg, numbered sequentially.
> 
> On Windows 10 I've tried:
> 
> ffmpeg.exe -start_number 000 -i %03d.jpg -r 1/6 mom.mp4
> 
> This works (images displayed for 10 seconds), but only shows 3 of the 300+
> images. If I leave off the -r option it builds the .mp4 with all images, but
> they flash by as fast as possible.
> 
> How do I get all images to show for 10 seconds?

Funny that the option name “-framerate” is part of your subject.

> 
> Problem #2.
> 
> The source .jpg images all vary in HxW size. How do I get them all to display
> with the same height regardless of width?

I believe you can only output one resolution but a shell script will help.

Carl Eugen
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[FFmpeg-user] newbie help - framerate

[Mark Foley]

I am trying for the first time to use ffmpeg. I have a collection of .jpg files
starting at 000.jpg, numbered sequentially.

On Windows 10 I've tried:

ffmpeg.exe -start_number 000 -i %03d.jpg -r 1/6 mom.mp4

This works (images displayed for 10 seconds), but only shows 3 of the 300+
images. If I leave off the -r option it builds the .mp4 with all images, but
they flash by as fast as possible.

How do I get all images to show for 10 seconds?

Problem #2.

The source .jpg images all vary in HxW size. How do I get them all to display
with the same height regardless of width?

Thanks (I need this for a presentation on Monday - gulp!)

--Mark
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[FFmpeg-user] Encoding Warnings

[Ruler2112]

Hello everybody.  Even though I've used ffmpeg for over 20 years, this will
be my first post to the mailing list.

I get videos from a variety of sources - youtube, vimeo, DVR, DVDs I've
purchased, etc.  They're in a wide variety of different formats, many of
which my TVs/DVRs over the years have refused to play.  The volume also
varies wildly between sources - DVR is fine, but anything from outside can
either be so quiet I have to crank the volume up to max to have any chance
at hearing it or be blasted out of the room if I don't have it set on
minimum.  Because of this, I wrote a batch file to process each video.  I
simply drop the raw files in a todo folder, run the batch file, and it would
do everything from there.  The resolution/frame rate is read with MediaInfo,
calculations done internally, then video is reprocessed with ffmpeg, the
audio is dumped to a wave with ffmpeg and then had processing done on it to
standardize the volume, and the two resulting files were then remuxed with
VirtualDub.  (I know ffmpeg should have been able to do it and frankly do
not remember why I made it like this - it was 15 years ago when I wrote it
after all.)  The output has XVID video with CBR MP3 audio with large
resolutions scaled down to a maximum I specified which has played perfectly
on everything I've thrown them at.  The script has worked awesome for years.

The problem now is that the windows machine I've been running it on has died
of a faulty motherboard.  Frankly, I moved on from windows many years ago
and that one XP box was kept solely for running this program.  I'd rather
not replace it with another windows machine if I can help it, so have been
working to re-write the batch files as a perl script and utilizing ffmpeg
exclusively to do the heavy lifting.  This has gone extremely well and
everything seems to be working fine with a couple exceptions.

First, the volume standardization program was also windows only.  I
contacted the author and got the source code.  Even though I hadn't touched
C since college, adapting it to work on Linux wasn't too bad and it's now
functional. :) 

Second are the ffmpeg warnings/errors.  I've spent over two days looking for
solutions to the various problems and have solved everything except a
couple.  I do not know if they're related to the version of ffmpeg being so
much newer than the one that was running on the XP machine or what, but I
don't know what else to try.  I'm hoping someone can help me iron them out.

The version of ffmpeg on the XP box (according to the output when running it
through WINE) is:

ffmpeg version N-78758-g5156578 Copyright (c) 2000-2016 the FFmpeg
developers
built with gcc 5.3.0 (GCC)
configuration: --enable-gpl --enable-version3 --disable-w32threads
--enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r
--enable-gnutls --enable-iconv --enable-libass --enable-libbluray
--enable-libbs2b
--enable-libcaca --enable-libdcadec --enable-libfreetype --enable-libgme
--enable-libgsm
--enable-libilbc --enable-libmodplug --enable-libmp3lame
--enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg
--enable-libopus
--enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex
--enable-libtheora --enable-libtwolame --enable-libvidstab
--enable-libvo-amrwbenc
--enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp
--enable-libx264
--enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg
--enable-lzma
--enable-decklink --enable-zlib
libavutil 55. 19.100 / 55. 19.100
libavcodec 57. 27.100 / 57. 27.100
libavformat 57. 26.100 / 57. 26.100
libavdevice 57. 0.101 / 57. 0.101
libavfilter 6. 37.100 / 6. 37.100
libswscale 4. 0.100 / 4. 0.100
libswresample 2. 0.101 / 2. 0.101
libpostproc 54. 0.100 / 54. 0.100

The version on the Linux machine is:

ffmpeg version 4.3-static https://johnvansickle.com/ffmpeg/  Copyright (c)
2000-2020 the FFmpeg developers
  built with gcc 8 (Debian 8.3.0-6)
  configuration: --enable-gpl --enable-version3 --enable-static
--disable-debug --disable-ffplay --disable-indev=sndio
--disable-outdev=sndio --cc=gcc --enable-fontconfig --enable-frei0r
--enable-gnutls --enable-gmp --enable-libgme --enable-gray --enable-libaom
--enable-libfribidi --enable-libass --enable-libvmaf --enable-libfreetype
--enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb
--enable-libopenjpeg --enable-librubberband --enable-libsoxr
--enable-libspeex --enable-libsrt --enable-libvorbis --enable-libopus
--enable-libtheora --enable-libvidstab --enable-libvo-amrwbenc
--enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265
--enable-libxml2 --enable-libdav1d --enable-libxvid --enable-libzvbi
--enable-libzimg
  libavutil  56. 51.100 / 56. 51.100
  libavcodec 58. 91.100 / 58. 91.100
  libavformat58. 45.100 / 58. 45.100
  libavdevice58. 10.100 / 58. 10.100
  libavfilter 7. 85.100 /  7. 85.100
  libswscale  5.  7.100 /  5.  7.100
  libswresample   3.  7.100 

Re: [FFmpeg-user] Frequency response of an ffmpeg equalizer filter

[Michael Koch]

        => How do I get the frequency graph of such a filter ? Is it 
mandatory to get a video stream (and how to get / display it) ? 
Can't I get command with the frequency as input and the gain as 
output ? Of course, such a command would means I need to query the 
db output for each interesting frequency.


Make a test file with 2kHz sine:
ffmpeg -f lavfi -i sine=2000 -t 5 -y test.wav

Analyze the volume of this file:
ffmpeg -i test.wav -af volumedetect -y out1.wav
The volume is written to the log file.

Then apply your filter and analyze the output again.

Michael



        => Thank you again for your answer Michael but, I think I 
probably didn't expressed my question with the right words. Let's 
forget the file in and the file out to focus only on the filter : 
"anequalizer=c0 f=200 w=100 g=-10 t=0|c1 f=200 w=100 g=-10 t=0"


        => What I want to know, before applying any input signal is 
the result of this mathematical function : f(frequency) = gain. For 
example, this filter is centered on 200 Hz with a bandwith of 100Hz 
and a gain of -10 db. So, all the 200Hz frequencies will be attenuated 
by -10 db. But what for the 250Hz frequencies in the signal ? Still 
-10 db ? Or -9 ? Or -7.5 ? Same question for the 350 Hz frequencies in 
the signal.




Unfortunately the filter steepness is undocumented. I don't know if 
ffmpeg can analyze the filter curve for all frequencies at the same 
time. I have no other idea than analyzing one frequency after the other.


Michael









=> The result given like a mathematical function, a cloud of dots, an 
image or anything else that could help me to draw such a graph : 
https://s1.qwant.com/thumbr/0x380/0/5/cfb0ceac6e3b4fdd73923eaaf68e76c58991b4de8ac088f4dfad665d5b8c34/parametric.png?u=https%3A%2F%2Fcdn.tutsplus.com%2Faudio%2Fuploads%2Flegacy%2Fqt_040_eq%2Fparametric.png=0=1=0=1


        => I wish my question is more clear now.

        => And thank you again for your answers and your help :-)



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--
**
  ASTRO ELECTRONIC   Dipl.-Ing. Michael Koch
   Raabestr. 43   37412 Herzberg
  www.astro-electronic.de
  Tel. +49 5521 854265   Fax +49 5521 854266
**

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[FFmpeg-user] Decode H264 RTP/UDP without SDP? Guessing at sprop-parameter-sets?

[Michael McConnell]

Hello all,

Does anyone know of a way to decode H264 streaming over RTP/UDP without an SDP 
file? I realize the answer is basically no. My question is more, is there a way 
to intelligently guess or get clues from the RTP / H264 itself to generate a 
SDP (sprop-parameter-sets) file? 

That being said, is there a database of SDP files or an SDP generator or 
anything of the sort that someone might be aware of? Looking at various cameras 
and h264 sources, there appears to be a wide variety of options; however, when 
you decode sprop-parameter-sets from base64 it seems most of these parameters 
are unset in everyday use cases.

Perhaps my real question is, using a dummy SDP file, is there a way to 
generator a sprop-parameter-sets and cycle through? Would mode 96 work as a 
generic?

Thanks,
Mike


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Re: [FFmpeg-user] Frequency response of an ffmpeg equalizer filter

[Nicolas VAMBAIRGUE]

Le 26/06/2020 à 19:11, Michael Koch a écrit :

Am 26.06.2020 um 19:00 schrieb Nicolas VAMBAIRGUE:


Le 26/06/2020 à 18:15, Paul B Mahol a écrit :

On 6/26/20, Nicolas VAMBAIRGUE  wrote:

Hello,


I'm starting to use the ffmpeg filters, using the documentation. It
seems working but, I don't find the way to get the frequency 
response of

a filter. For example, let's say I'm applying this filter :

equalizer=f=7000:t=q:w=2:g=2,equalizer=f=1000:t=q:w=1:g=3,equalizer=f=80:t=q:w=2:g=-30 



in the following command :

ffmpeg -re -i /home/dr_click/live.wav -acodec pcm_s16be -ar 44100 
-ac 2

-f rtp -af
"equalizer=f=7000:t=q:w=2:g=2,equalizer=f=1000:t=q:w=1:g=3,equalizer=f=80:t=q:w=2:g=-30" 


rtp://127.0.0.1:1234

It works, I'm hearing a distorted sound. And what I'd like to know, 
with

such a filter, is the db gain for the 2000 Hz signal or for the 150Hz
signal or any other frequency with a query like : give me your 
frequency

and I'll tell you which gain is applied at this frequency with the
defined filter.

Yes, with anequalizer filter.


        => Ok, so let's say I'm using this command with an anequaliwer

        =>_ffmpeg -re -i /home/dr_click/live.wav -af "anequalizer=c0 
f=200 w=100 g=-10 t=0|c1 f=200 w=100 g=-10 t=0 curves=true" -acodec 
pcm_s16be -ar 44100 -ac 2 -f rtp rtp://127.0.0.1:1234


        => How do I get the frequency graph of such a filter ? Is it 
mandatory to get a video stream (and how to get / display it) ? Can't 
I get command with the frequency as input and the gain as output ? Of 
course, such a command would means I need to query the db output for 
each interesting frequency.


Make a test file with 2kHz sine:
ffmpeg -f lavfi -i sine=2000 -t 5 -y test.wav

Analyze the volume of this file:
ffmpeg -i test.wav -af volumedetect -y out1.wav
The volume is written to the log file.

Then apply your filter and analyze the output again.

Michael



        => Thank you again for your answer Michael but, I think I 
probably didn't expressed my question with the right words. Let's forget 
the file in and the file out to focus only on the filter : 
"anequalizer=c0 f=200 w=100 g=-10 t=0|c1 f=200 w=100 g=-10 t=0"


        => What I want to know, before applying any input signal is the 
result of this mathematical function : f(frequency) = gain. For example, 
this filter is centered on 200 Hz with a bandwith of 100Hz and a gain of 
-10 db. So, all the 200Hz frequencies will be attenuated by -10 db. But 
what for the 250Hz frequencies in the signal ? Still -10 db ? Or -9 ? Or 
-7.5 ? Same question for the 350 Hz frequencies in the signal.


        => The result given like a mathematical function, a cloud of 
dots, an image or anything else that could help me to draw such a graph 
: 
https://s1.qwant.com/thumbr/0x380/0/5/cfb0ceac6e3b4fdd73923eaaf68e76c58991b4de8ac088f4dfad665d5b8c34/parametric.png?u=https%3A%2F%2Fcdn.tutsplus.com%2Faudio%2Fuploads%2Flegacy%2Fqt_040_eq%2Fparametric.png=0=1=0=1


        => I wish my question is more clear now.

        => And thank you again for your answers and your help :-)



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Re: [FFmpeg-user] Frequency response of an ffmpeg equalizer filter

[Michael Koch]

Am 26.06.2020 um 19:00 schrieb Nicolas VAMBAIRGUE:


Le 26/06/2020 à 18:15, Paul B Mahol a écrit :

On 6/26/20, Nicolas VAMBAIRGUE  wrote:

Hello,


I'm starting to use the ffmpeg filters, using the documentation. It
seems working but, I don't find the way to get the frequency 
response of

a filter. For example, let's say I'm applying this filter :

equalizer=f=7000:t=q:w=2:g=2,equalizer=f=1000:t=q:w=1:g=3,equalizer=f=80:t=q:w=2:g=-30 



in the following command :

ffmpeg -re -i /home/dr_click/live.wav -acodec pcm_s16be -ar 44100 -ac 2
-f rtp -af
"equalizer=f=7000:t=q:w=2:g=2,equalizer=f=1000:t=q:w=1:g=3,equalizer=f=80:t=q:w=2:g=-30" 


rtp://127.0.0.1:1234

It works, I'm hearing a distorted sound. And what I'd like to know, 
with

such a filter, is the db gain for the 2000 Hz signal or for the 150Hz
signal or any other frequency with a query like : give me your 
frequency

and I'll tell you which gain is applied at this frequency with the
defined filter.

Yes, with anequalizer filter.


        => Ok, so let's say I'm using this command with an anequaliwer

        =>_ffmpeg -re -i /home/dr_click/live.wav -af "anequalizer=c0 
f=200 w=100 g=-10 t=0|c1 f=200 w=100 g=-10 t=0 curves=true" -acodec 
pcm_s16be -ar 44100 -ac 2 -f rtp rtp://127.0.0.1:1234


        => How do I get the frequency graph of such a filter ? Is it 
mandatory to get a video stream (and how to get / display it) ? Can't 
I get command with the frequency as input and the gain as output ? Of 
course, such a command would means I need to query the db output for 
each interesting frequency.


Make a test file with 2kHz sine:
ffmpeg -f lavfi -i sine=2000 -t 5 -y test.wav

Analyze the volume of this file:
ffmpeg -i test.wav -af volumedetect -y out1.wav
The volume is written to the log file.

Then apply your filter and analyze the output again.

Michael

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Re: [FFmpeg-user] Frequency response of an ffmpeg equalizer filter

[Nicolas VAMBAIRGUE]

Le 26/06/2020 à 18:15, Paul B Mahol a écrit :

On 6/26/20, Nicolas VAMBAIRGUE  wrote:

Hello,


I'm starting to use the ffmpeg filters, using the documentation. It
seems working but, I don't find the way to get the frequency response of
a filter. For example, let's say I'm applying this filter :

equalizer=f=7000:t=q:w=2:g=2,equalizer=f=1000:t=q:w=1:g=3,equalizer=f=80:t=q:w=2:g=-30

in the following command :

ffmpeg -re -i /home/dr_click/live.wav -acodec pcm_s16be -ar 44100 -ac 2
-f rtp -af
"equalizer=f=7000:t=q:w=2:g=2,equalizer=f=1000:t=q:w=1:g=3,equalizer=f=80:t=q:w=2:g=-30"
rtp://127.0.0.1:1234

It works, I'm hearing a distorted sound. And what I'd like to know, with
such a filter, is the db gain for the 2000 Hz signal or for the 150Hz
signal or any other frequency with a query like : give me your frequency
and I'll tell you which gain is applied at this frequency with the
defined filter.

Yes, with anequalizer filter.


        => Ok, so let's say I'm using this command with an anequaliwer

        =>_ffmpeg -re -i /home/dr_click/live.wav -af "anequalizer=c0 
f=200 w=100 g=-10 t=0|c1 f=200 w=100 g=-10 t=0 curves=true" -acodec 
pcm_s16be -ar 44100 -ac 2 -f rtp rtp://127.0.0.1:1234


        => How do I get the frequency graph of such a filter ? Is it 
mandatory to get a video stream (and how to get / display it) ? Can't I 
get command with the frequency as input and the gain as output ? Of 
course, such a command would means I need to query the db output for 
each interesting frequency.






Thank you for your help,


Nicolas

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Re: [FFmpeg-user] Frequency response of an ffmpeg equalizer filter

[Paul B Mahol]

On 6/26/20, Nicolas VAMBAIRGUE  wrote:
> Hello,
>
>
> I'm starting to use the ffmpeg filters, using the documentation. It
> seems working but, I don't find the way to get the frequency response of
> a filter. For example, let's say I'm applying this filter :
>
> equalizer=f=7000:t=q:w=2:g=2,equalizer=f=1000:t=q:w=1:g=3,equalizer=f=80:t=q:w=2:g=-30
>
> in the following command :
>
> ffmpeg -re -i /home/dr_click/live.wav -acodec pcm_s16be -ar 44100 -ac 2
> -f rtp -af
> "equalizer=f=7000:t=q:w=2:g=2,equalizer=f=1000:t=q:w=1:g=3,equalizer=f=80:t=q:w=2:g=-30"
> rtp://127.0.0.1:1234
>
> It works, I'm hearing a distorted sound. And what I'd like to know, with
> such a filter, is the db gain for the 2000 Hz signal or for the 150Hz
> signal or any other frequency with a query like : give me your frequency
> and I'll tell you which gain is applied at this frequency with the
> defined filter.

Yes, with anequalizer filter.

>
>
> Thank you for your help,
>
>
> Nicolas
>
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[FFmpeg-user] Frequency response of an ffmpeg equalizer filter

[Nicolas VAMBAIRGUE]

Hello,


I'm starting to use the ffmpeg filters, using the documentation. It 
seems working but, I don't find the way to get the frequency response of 
a filter. For example, let's say I'm applying this filter :


equalizer=f=7000:t=q:w=2:g=2,equalizer=f=1000:t=q:w=1:g=3,equalizer=f=80:t=q:w=2:g=-30

in the following command :

ffmpeg -re -i /home/dr_click/live.wav -acodec pcm_s16be -ar 44100 -ac 2 
-f rtp -af 
"equalizer=f=7000:t=q:w=2:g=2,equalizer=f=1000:t=q:w=1:g=3,equalizer=f=80:t=q:w=2:g=-30" 
rtp://127.0.0.1:1234


It works, I'm hearing a distorted sound. And what I'd like to know, with 
such a filter, is the db gain for the 2000 Hz signal or for the 150Hz 
signal or any other frequency with a query like : give me your frequency 
and I'll tell you which gain is applied at this frequency with the 
defined filter.



Thank you for your help,


Nicolas

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Re: [FFmpeg-user] Fwd: OBS and OBS Portable Recording Into Same File Simultaneously

[David Bayles]

> showinfo operates upon decoded frames. This inspection will have to be
> done at the container level using a forensic tool which can identify
> start and end of block structures in Matroska.

Can you suggest a forensic tool or even a company/person that could help me 
recover this? I’d be willing to pay for this service/tool.

On Fri, Jun 26, 2020 at 5:23 AM Gyan Doshi  wrote:
>
>
> On 26-06-2020 02:08 pm, Moritz Barsnick wrote:
> > You could try to observe at which point ffmpeg detects the first errors
> > (the showinfo filter will indicate the byte position ffmpeg is
> > operating on), and from there remove blocks of 512, 1024, 2048 bytes,
> > hoping that after removal of some of them, you hit a proper block
> > again, and ffmpeg will no longer "lose sync" at that point, until again
> > a later point in the file.
>
> showinfo operates upon decoded frames. This inspection will have to be
> done at the container level using a forensic tool which can identify
> start and end of block structures in Matroska. This assumes that the two
> write operations didn't write on top of each other. One saving grace
> could be that OBS, like ffmpeg, flushes data to files in blocks of 256
> kB, although I don't know if this is the case.
>
> Gyan
>
>
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Re: [FFmpeg-user] FFMPEG endlessly increases memory usage over time with HLS packaging.

[Alessandro Molon]

Actually I have several machines in production (dual Xeon 5620 - 32GB - Nvidia 
P2000) that are running 14 ffmpeg process each, all decoding Full HD mpegts and 
transcoding them in 4 different profiles each (HD/SD/SDh/SDq) and to be honest 
they work like a sharm and are also very stable.

Based on this experience I actually expected, just to repacketize pre-encoded 
content in HLS, to run at least 40/45 (even more) process per machine

Let me go ahead with my experiments, I'll be more than happy to produce the 
results.

Alex

-Original Message-
From: ffmpeg-user  On Behalf Of Carl Eugen Hoyos
Sent: 25 June 2020 22:19
To: FFmpeg user questions 
Subject: Re: [FFmpeg-user] FFMPEG endlessly increases memory usage over time 
with HLS packaging.

Am Do., 25. Juni 2020 um 20:39 Uhr schrieb Alessandro Molon
:

> Let's see what happens if I run 20 processes for few days...

On most hardware, you cannot run 20 (real-life) FFmpeg processes for a few days 
but if there is constantly increasing memory footprint or a memory leak, one 
process should be sufficient to show it.

Please find out what top-posting means and avoid it here, Carl Eugen 
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Re: [FFmpeg-user] Fwd: OBS and OBS Portable Recording Into Same File Simultaneously

[Gyan Doshi]

On 26-06-2020 02:08 pm, Moritz Barsnick wrote:

You could try to observe at which point ffmpeg detects the first errors
(the showinfo filter will indicate the byte position ffmpeg is
operating on), and from there remove blocks of 512, 1024, 2048 bytes,
hoping that after removal of some of them, you hit a proper block
again, and ffmpeg will no longer "lose sync" at that point, until again
a later point in the file.


showinfo operates upon decoded frames. This inspection will have to be 
done at the container level using a forensic tool which can identify 
start and end of block structures in Matroska. This assumes that the two 
write operations didn't write on top of each other. One saving grace 
could be that OBS, like ffmpeg, flushes data to files in blocks of 256 
kB, although I don't know if this is the case.


Gyan


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Re: [FFmpeg-user] Fwd: OBS and OBS Portable Recording Into Same File Simultaneously

[Moritz Barsnick]

On Wed, Jun 24, 2020 at 10:59:51 -0400, David Bayles wrote:
> In OBS and OBS Portable, my video technician had the save location set to
> the same exact folder (this was not how I intended it to be, but he made a
> last minute change to accomadate what he suspected was a faulty hard
> drive). We also had a hotkey to start recording on both apps
> simultaneously. We left the auto naming scheme in place for both instances
> of OBS since they were supposed to be saving to two totally different
> external hard drives (we have since changed this). This perfect storm
> resulted in both recordings being written into a single file (see link
> above), instead of two files as intended. It is unplayabe in VLC.

Assuming that both programs appended to the file (and didn't overwrite
each other at the same file position), you would have double the data,
interleaved on operating system / file system level.

To properly disassemble at least one of the two halves, you would
probably need to operate of block level, understanding how large the
blocks were, and trying to fiddle out the ones from the one program and
the ones from the other, while making sense to a decoder.

I consider that ffmpeg gives you some results at all a coincidence - it
could just as well we confused at the first inconsistency. (I assume
Matroska allows for some of this recovery.

I do understand that you would like to use ffmpeg to re-assemble the
original video, especially as extracting audio looks somehwat
promising. Yet the video's block sizes will not correspond to the file
system's block sizes, and ffmpeg will keep encountering inconcistencies
to the point of giving up, unless you take away the conflicting
segments.

> My life would saved if there is a way to extract good video/audio from this
> corrupted file. Any help would be greatly appreciated, thanks in advance!

You could try to observe at which point ffmpeg detects the first errors
(the showinfo filter will indicate the byte position ffmpeg is
operating on), and from there remove blocks of 512, 1024, 2048 bytes,
hoping that after removal of some of them, you hit a proper block
again, and ffmpeg will no longer "lose sync" at that point, until again
a later point in the file.

Just a wild suggestion.

Oh, and if the two OBS overwrote each other's data, I'm pretty sure
you're lost.

Cheers,
Moritz
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