On Sat, Apr 9, 2022 at 5:22 PM morgan holly via ffmpeg-user <
ffmpeg-user@ffmpeg.org> wrote:
> this is part question and part warning. When I run the cmd below FFMPEG
> automatically knocks my 24bit source audio down to 8bits(!). I was able to
> change that by adding 'aformat=sample_fmts=s32'
On Mon, Apr 4, 2022 at 6:53 PM Carl Zwanzig wrote:
> On 3/31/2022 1:19 PM, Paul B Mahol wrote:
> > Provocations are not tolerated here.
>
> Really? What changed?
>
>
If you keep provocative behavior, then you will be banned.
z!
> _
On Wed, Mar 30, 2022 at 9:12 PM Wolfgang Hugemann wrote:
> I would rather use FLACs original frontend (than ffmpeg), which also
> provides a nice Windows GUI interface. FLAC conversion is also included
> in a variety of software products, i.e. Exact Audio Copy, which I used
> to rip all my music
Hello,
I would like to inform you about the new tool, located here:
https://github.com/richardpl/lavfi-preview
It is GUI filtergraph editor and foremost ffmpeg's libavfilter filtergraph
preview tool.
It is in early stage, but already functional.
Regards
Better use ebur128 scanner filter, it is much faster, after it reports
values, just apply volume filter.
loudnorm dynamic processing is not perfect.
___
ffmpeg-user mailing list
ffmpeg-user@ffmpeg.org
https://ffmpeg.org/mailman/listinfo/ffmpeg-user
To
On Fri, Mar 18, 2022 at 12:25 AM Michael Koch
wrote:
> Question 1:
> I have two streams with RGB24 pixel formats.
> Is it possible to do the following calculation pixel-wise and write the
> result to a grayscale stream?
> output = abs(R1-R2) + abs(G1-G2) + abs(B1-B2)
>
> Question 2:
> Three
On Thu, Mar 10, 2022 at 10:10 AM Reindl Harald
wrote:
>
>
> Am 10.03.22 um 09:54 schrieb Paul B Mahol:
> > On Thu, Mar 10, 2022 at 9:46 AM Reindl Harald
> > wrote:
> >> your "FFmpeg is right tool for this jobs. You just do not
> >> know how to use
On Thu, Mar 10, 2022 at 9:46 AM Reindl Harald
wrote:
>
>
> Am 10.03.22 um 09:42 schrieb Reindl Harald:
> > Am 10.03.22 um 09:41 schrieb Paul B Mahol:
> >> On Thu, Mar 10, 2022 at 9:38 AM Reindl Harald
> >> wrote:
> >>
> >>>
> >>&g
On Thu, Mar 10, 2022 at 9:38 AM Reindl Harald
wrote:
>
>
> Am 10.03.22 um 09:36 schrieb Paul B Mahol:
> >> we have 2022 and volume normalization is done via anlazye and meta-tags
> >> where tools exists FOR R128 normalization
> >>
> >
> > FFmpe
On Thu, Mar 10, 2022 at 9:26 AM Reindl Harald
wrote:
>
>
> Am 10.03.22 um 08:54 schrieb Paul B Mahol:
>
> >> Perhaps you need a different tool. If you're actually just dealing with
> >> music files you should snag foobar2000 and use it for this job
> ins
On Thu, Mar 10, 2022 at 8:14 AM Clayton Macleod
wrote:
> On Wed, Mar 9, 2022 at 8:26 PM CMG DiGiTaL wrote:
>
> >
> > OK mark, very nice to hear about your audio experience!...I love it, it
> > makes me feel free and alive!
> >
> > With this internet audio standardization movement, I'm paying
On 3/8/22, morgan holly via ffmpeg-user wrote:
>
>
>> On Mar 8, 2022, at 12:07 PM, Paul B Mahol wrote:
>>
>> On 3/8/22, morgan holly via ffmpeg-user > <mailto:ffmpeg-user@ffmpeg.org>> wrote:
>>> Hi,
>>>
>>> I’m trying to rem
On 3/8/22, llee040 wrote:
>
>
>> On Mar 8, 2022, at 11:32 AM, morgan holly via ffmpeg-user
>> wrote:
>>
>> Hi,
>>
>> I’m trying to remove telecine from a prores mov file using the pullup
>> filter:
>>
>> Source file:
>> Video: prores (LT), top coded first (swapped)), 720x480, 29.97 fps, 29.97
>>
On 3/8/22, CMG DiGiTaL wrote:
> Yeah, this is what I suspected. You've got source files that are below
>>
>
> Okay Clayton, I get the concept!
>
> As I'm doing a batch for myself, if necessary, normalize the LUFS of my
> files, I was
> left with this doubt as to why I can't do it in linear mode
On 3/8/22, morgan holly via ffmpeg-user wrote:
> Hi,
>
> I’m trying to remove telecine from a prores mov file using the pullup
> filter:
>
> Source file:
> Video: prores (LT), top coded first (swapped)), 720x480, 29.97 fps, 29.97
> tbr, 30k tbn, 30k tbc (default)
>
> Command:
> ffmpeg -i
On 3/7/22, CMG DiGiTaL wrote:
> hi,
>
> About loudnorm normalization: Normalization in Dual-pass mode.
>
> Rules for normalization to be linear:
>
> 1 - Values entry:
> LUFS_TARGET (user entry)
> LRA (get input file - Pass 1 loudnorm)
where is measured_tp ?
> TP_TARGET
On 3/4/22, Bo Berglund wrote:
> On Sat, 9 Oct 2021 03:23:52 +0300, Anatoly
> wrote:
>
>>On Thu, 07 Oct 2021 10:27:51 +0200
>>Bo Berglund wrote:
>>
>>> On Wed, 1 Sep 2021 16:30:00 +0300, anat...@kazanfieldhockey.ru wrote:
>>>
>>> >Right, you need m3u8.
>>> >F12->Network->Enter "m3u8" in the
On 3/4/22, Sebastian Wallkötter wrote:
> Hi!
>
> I'm trying to understand how ffmpeg goes about demosaicing and which Bayer
> formats are available. Currently, I'm looking at the bayer_rggb8 format, and
> what is confusing me is that the format info reports that the format has
> less than 8 bits
eam
> > “[Parsed_movie_0 @ 0x561d883e3800] Failed to avformat_open_input 'https'”.
> > Is there another way to do this command to support streaming the movie or
> > amovie – if the latter is applicable?
> >
>
> It is only matter of properly escaping ':' with on she
gt; Is there another way to do this command to support streaming the movie or
> amovie – if the latter is applicable?
>
It is only matter of properly escaping ':' with on shell, in windows
it is similar. You can scale before split, it will be faster, when scaling
to same size.
>
> Thank y
On Tue, Mar 1, 2022 at 11:26 PM Bryce Newman
wrote:
> Hello,
>
> Using FFmpeg 4.4.1_3 or 5.0 the command below causes FFmpeg to use 6.5GB
> of RAM.
> The input file amix-audio.mov is only 348MB. I have also seen the same
> large memory footprint using join.
> Using either join or adelay+amix
On Mon, Feb 22, 2021 at 6:52 PM Jonathan Baecker wrote:
> Hello!
>
> I'm trying to nomalize an audio file with FFmpeg. I'm using the loudnorm
> filter. The source loudness is -23 LUFS and I want to make it -17 LUFS.
> As far as I know, loudnorm has 2 modes of normalizing audio: linear and
>
On Mon, Feb 14, 2022 at 1:50 PM Remko Tronçon wrote:
> Hi,
>
> When I run an audio file through the 'loudnorm' filter, I get the
> following:
> > ffmpeg -i -af loudnorm=print_format=json -f null -
> ...
> "input_i" : "-13.87",
> "input_tp" : "0.06",
> "input_lra" : "8.50",
>
peak
one than threshold could get lower...
>
> Thanks again for your help, really appreciate it.
>
> All the best,
>
> Jools
>
>
> On Sun, 2022-02-20 at 13:01 +0100, Paul B Mahol wrote:
> > On Sun, Feb 20, 2022 at 12:33 PM Julian Pilfold-Bagwell <
> > ju
han just filtering out only when background
noise is present and of certain signal to noise ratio.
so threshold argument can be changed to actual dB of your background noise
typically captured.
>
> On Sun, 20 Feb 2022 at 11:12, Paul B Mahol wrote:
>
> > On Sun, Feb 20, 2022 a
On Sun, Feb 20, 2022 at 12:03 PM Julian Pilfold-Bagwell <
julian.pilfoldbagw...@gmail.com> wrote:
> Hi All,
>
>
> I'm trying to set up a tannoy system in our building using FFMPEG and
> Icecast. The server that is streaming is being fed from the 0db recording
> output from an analogue amplifier
On Sun, Feb 20, 2022 at 4:19 AM Simon Roberts <
si...@dancingcloudservices.com> wrote:
> Thank you Leo, Michael, Ronnie, your info was very helpful. Michael's link
> to the crossfading part worked very nicely, and with -loop 1 and -shortest,
> the result repeats the images until the audio runs
On Tue, Feb 8, 2022 at 11:07 AM Jim DeLaHunt
wrote:
> On 2022-02-08 00:19, Paul B Mahol wrote:
>
> > Please stop spamming this mailing list with unrelated stuff.
>
> Paul:
>
> Mark's stuff is not unrelated. Mark asked nicely if people were
> interested in his stuff. Mu
On Tue, Feb 8, 2022 at 3:51 AM Mark Filipak
wrote:
> Help!
>
> Command: ffmpeg -report
> -i 24'fps[24pps].vob
> -filter_complex "[0:a]asetpts=expr=N/TB*1001/24000[a]"
> -bsf setts=pts=N/TB*1001/24000:dts=N/TB*1001/24000
> -target ntsc-dvd -map 0:v -map
Please stop spamming this mailing list with unrelated stuff.
___
ffmpeg-user mailing list
ffmpeg-user@ffmpeg.org
https://ffmpeg.org/mailman/listinfo/ffmpeg-user
To unsubscribe, visit link above, or email
ffmpeg-user-requ...@ffmpeg.org with subject
On Thu, Feb 3, 2022 at 1:11 PM Reindl Harald wrote:
>
>
> Am 03.02.22 um 13:07 schrieb 桃源老師
> > I experienced funny ffmpeg compille error "ERROR: libvmaf >= 2.0.0 not
> found using pkg-config". The reason why I feel funny is that ffmpeg commit
> of
On Wed, Jan 12, 2022 at 5:00 PM Michael Koch
wrote:
> Hello,
>
> I have a projector with a 180° fisheye lens in a dome. Things are easy
> if the lens is at the center of the dome. Any input format can be
> transformed to fisheye output with the v360 filter. But things are
> getting complicated
On Fri, Nov 12, 2021 at 10:34 PM flp via ffmpeg-user
wrote:
> Hey,
>
> I'm trying to debug when my ZMQ message for a complex filter bound by
> zmq=bind_address arrives. I tried the -debug option and I see there that
> the message arrives but I only can tell from my video result it doesn't
> work
On Thu, Nov 4, 2021 at 9:41 AM Tom Vercauteren
wrote:
> > > Apologies for cross-posting a question I initially posted on
> stackoverflow
> > > (
> > > https://stackoverflow.com/q/69739665/17261462) but having had no
> response
> > > there I thought this mailing list may be a better place for it.
On Wed, Nov 3, 2021 at 11:57 PM Tom Vercauteren
wrote:
> Hello,
>
> Apologies for cross-posting a question I initially posted on stackoverflow
> (
> https://stackoverflow.com/q/69739665/17261462) but having had no response
> there I thought this mailing list may be a better place for it.
>
> I
On Wed, Nov 3, 2021 at 12:57 PM LianCheng Tan wrote:
> I have tried to do what you have suggested. Somehow, it still cause ffmpeg
> to crash at the exact moment the file is being renamed.
>
> By the way, I am running in Windows environment. Not sure, if in Linux
> will this issue occurs.
>
Top
On Sat, Oct 23, 2021 at 3:26 PM Android PowerUser
wrote:
> I try
>
> ffmpeg -y -i "/storage/emulated/0/Download/(M4A_128K).m4a"
> compand=0|0:1|1:-90/-900|-70/-70|-30/-9|0/-3:6:0:0:0
> "/storage/emulated/0/Download/(M4A_128K)_1.m4a"
>
> for dynamic range compression, but it call back: invalid
hat.
>
> thanks, roman
>
> -Original Message-
> From: ffmpeg-user On Behalf Of Paul B
> Mahol
> Sent: Sunday, 12 September 2021 20:56
> To: FFmpeg user questions
> Subject: Re: [FFmpeg-user] How can chromakey ignore shadows ?
>
> On Sun, Sep 12, 2021 at 8:34
On Thu, Oct 14, 2021 at 1:18 PM Michael Koch
wrote:
> Am 14.10.2021 um 13:11 schrieb Paul B Mahol:
> > On Thu, Oct 14, 2021 at 1:08 PM Michael Koch <
> astroelectro...@t-online.de>
> > wrote:
> >
> >> Am 14.10.2021 um 13:01 schrieb Paul B Mahol:
> &g
On Thu, Oct 14, 2021 at 1:08 PM Michael Koch
wrote:
> Am 14.10.2021 um 13:01 schrieb Paul B Mahol:
> > On Thu, Oct 14, 2021 at 12:59 PM Michael Koch <
> astroelectro...@t-online.de>
> > wrote:
> >
> >> Am 14.10.2021 um 12:53 schrieb Paul B Mahol:
> &g
On Thu, Oct 14, 2021 at 12:59 PM Michael Koch
wrote:
> Am 14.10.2021 um 12:53 schrieb Paul B Mahol:
> > On Thu, Oct 14, 2021 at 12:29 PM Michael Koch <
> astroelectro...@t-online.de>
> > wrote:
> >
> >> Am 14.10.2021 um 12:07 schrieb Paul B Mahol:
>
On Thu, Oct 14, 2021 at 12:29 PM Michael Koch
wrote:
> Am 14.10.2021 um 12:07 schrieb Paul B Mahol:
> > Sorry but I'm not on windows, so I can not use your script.
>
> Then try the below (slightly improved) version. It would become much
> simpler with variables:
> ARRAY_H =
Sorry but I'm not on windows, so I can not use your script.
___
ffmpeg-user mailing list
ffmpeg-user@ffmpeg.org
https://ffmpeg.org/mailman/listinfo/ffmpeg-user
To unsubscribe, visit link above, or email
ffmpeg-user-requ...@ffmpeg.org with subject
On Wed, Oct 13, 2021 at 11:31 PM Sylvain Leforestier <
sylvain.leforest...@gmail.com> wrote:
> Hi every boby,
> I found pleasant to use ffmpeg to clean my records from old CD using Matlab
> :
> * audiowrite('t.wav',d/2,f,'bitspersample',32);*
>
>
> *[sta,ret]=system ('ffmpeg -i
On Wed, Oct 13, 2021 at 10:51 PM Michael Koch
wrote:
> Am 12.10.2021 um 19:29 schrieb Michael Koch:
> > Am 12.10.2021 um 18:52 schrieb Paul B Mahol:
> >> On Tue, Oct 12, 2021 at 10:55 AM Michael Koch
> >>
> >> wrote:
> >>
> >>> I have
On Tue, Oct 12, 2021 at 10:55 AM Michael Koch
wrote:
> I have a question about the "fftfilt" filter. What's the default value
> of the weight_U and weight_V options? I'm asking because I get an
> unexpected result.
>
> This command line creates my input image for testing:
>
> ffmpeg -f lavfi -i
On Mon, Oct 11, 2021 at 11:15 AM Michael Koch
wrote:
> Hello,
>
> I have a question about the shuffleplanes filter. In this example the
> pixel format is RGB24. So the first plane is red and the second plane is
> green. I want to swap the red and green planes.
>
> ffmpeg -f lavfi -i testsrc2
On Fri, Oct 1, 2021 at 10:11 AM Bo Berglund wrote:
> On Thu, 30 Sep 2021 14:35:27 +0200, Bo Berglund
> wrote:
>
> >I searched similar questions and found this regarding blackdetect filter
> on
> >stackoverflow:
> >
>
On Thu, Sep 30, 2021 at 7:06 PM Michael Koch
wrote:
> I like the sample images for the blend filter that Paul has added here:
> https://trac.ffmpeg.org/wiki/Blend
>
> Can you please also share the command line for making these images?
>
mpv
On Thu, Sep 30, 2021 at 11:58 AM Arif Driessen wrote:
> Consider this script:
>
> ---
> #!/bin/bash
>
> ffmpeg \
> -v error \
> -filter_complex \
> "
> color=blue,
> trim=0:3
> [blue];
>
> color=red,
> trim=0:5
> [red];
>
>
On Tue, Sep 28, 2021 at 10:11 PM Michael Koch
wrote:
> Am 21.09.2021 um 08:57 schrieb Paul B Mahol:
> >>> See new morpho filter, that gonna be in master soon.
> >>>
> >>> It is much faster than current filters and allow custom definition of
> >>
On Sun, Sep 26, 2021 at 1:55 PM Arif Driessen wrote:
> > should be working, if not report bug.
> Ah okay, I did some investigation, and indeed (as usual) I was wrong. The
> option does appear in the documentation for the OUTPUT device:
> https://ffmpeg.org/ffmpeg-devices.html#pulse-1
>
> Kind of
On Sun, Sep 26, 2021 at 1:02 PM Arif Driessen wrote:
> > Sorry, but loop filter does not work that way.
>
> size parameter of loop filter is by default 0, meaning it will loop
> nothing.
> For single images you must set this option to exactly 1.
>
> Hi Paul, unless I misunderstand, in my email I
On Fri, Sep 24, 2021 at 10:55 PM Arif Driessen wrote:
> > It is very hard to help when no info what exactly was tried is not
> provided.
> Well I tried every variation on what you said. that is:
>
> > One possible way is using loop video filter
> So I'm looking at this documentation:
>
On Fri, Sep 24, 2021 at 7:49 PM Arif Driessen wrote:
> > One possible way is using loop video filter, which will work with single
> images fine.
>
> Hi Paul, I've tried with this https://ffmpeg.org/ffmpeg-filters.html#loop
> and by setting the loop option on the `movie` filter. Both to no avail.
On Fri, Sep 24, 2021 at 7:31 PM Arif Driessen wrote:
> Hi everyone,
>
> So I'm still working on my filter_complex graph that xfades an image to the
> beginning of another video.
>
> And I managed to figure it out, it works for any input source, except for
> the two different .png files I have
gt; on the value specified, appropriate quality
> settings will
> be used.
> "--preset help" gives more info on these
>
> --help id3 ID3 tagging related options
>
> --longhelp full list of
g rate : 8 000 Hz
> Stream size: 356 KiB (100%)
>
Hm, what created such wav file? Because that adpcm variant should have
0x0017 twocc and not 0x1700 one.
>
> On Wed, Sep 22, 2021 at 3:17 PM Paul B Mahol wrote:
>
> > On Wed, Sep 22, 2021 at 1:22
stereo, s16p
> Metadata:
> encoder : Lavc58.134.100 libmp3lame
> size= 134kB time=00:00:45.58 bitrate= 24.2kbits/s speed= 134x
> video:0kB audio:134kB subtitle:0kB other streams:0kB global headers:0kB
> muxing overhead: 0.189990%
>
>
>
Is sound correct?
&g
On Wed, Sep 22, 2021 at 11:06 AM Arif Driessen wrote:
> Looking at your output it looks like it can't figure out what your
> input.wav is. Does your wav file work in other applications? Have you tried
> increasing the value for 'analyzeduration'?
> If you know the details you could manually
On Tue, Sep 21, 2021 at 8:31 AM Michael Koch
wrote:
> Am 20.09.2021 um 21:51 schrieb Paul B Mahol:
> > On Sat, Oct 17, 2015 at 3:58 PM Michael Koch <
> astroelectro...@t-online.de>
> > wrote:
> >
> >> > That looks like the job for the filter, see inf
On Sat, Oct 17, 2015 at 3:58 PM Michael Koch
wrote:
> > That looks like the job for the filter, see inflate, deflate, erosion
> and dilation filters.
>
> I just tested the dilation filter. It's a nice workaround if the
> downscaling factor is 3:1, as in my example. But if the downscaling
>
On Sat, Sep 18, 2021 at 6:07 PM Cecil Westerhof via ffmpeg-user <
ffmpeg-user@ffmpeg.org> wrote:
> I have several of the following (simplified):
> drawtext=
>enable = 'between(t, 105, 115)':
>text = 'speaker':
>y = main_h - (text_h * 4),
>
On Fri, Sep 17, 2021 at 11:39 PM Cecil Westerhof via ffmpeg-user <
ffmpeg-user@ffmpeg.org> wrote:
> Paul B Mahol writes:
>
> > On Fri, Sep 17, 2021 at 11:07 PM Cecil Westerhof via ffmpeg-user <
> > ffmpeg-user@ffmpeg.org> wrote:
> >
> >> Paul B Ma
On Fri, Sep 17, 2021 at 11:07 PM Cecil Westerhof via ffmpeg-user <
ffmpeg-user@ffmpeg.org> wrote:
> Paul B Mahol writes:
>
> > On Fri, Sep 17, 2021 at 10:33 PM Cecil Westerhof via ffmpeg-user <
> > ffmpeg-user@ffmpeg.org> wrote:
> >
> >> Paul B Ma
On Fri, Sep 17, 2021 at 10:33 PM Cecil Westerhof via ffmpeg-user <
ffmpeg-user@ffmpeg.org> wrote:
> Paul B Mahol writes:
>
> > On Fri, Sep 17, 2021 at 9:34 PM Cecil Westerhof via ffmpeg-user <
> > ffmpeg-user@ffmpeg.org> wrote:
> >
> >> Cecil Wester
On Fri, Sep 17, 2021 at 9:34 PM Cecil Westerhof via ffmpeg-user <
ffmpeg-user@ffmpeg.org> wrote:
> Cecil Westerhof via ffmpeg-user writes:
>
> When I put before the first chapter:
> [CHAPTER]
> TIMEBASE=1/1
> START=1
> END=2
> title=
>
>
I think chapter metadata needs own
On Tue, Sep 14, 2021 at 4:52 PM Richard Kimberly Heck
wrote:
> Hi, all,
>
> I've recently started experiencing problems when convering files with
> AC3 video. Basically, I end up with no sound. Here's an example. This
> file reports as:
>
> Input #0, mpeg, from 'vobfile.vob':
>Duration:
On Mon, Sep 13, 2021 at 11:06 AM Tim Lloyd wrote:
> Hi, Thanks for letting me join this mailing list. First post and my
> knowledge of ffmpeg is basic at best.
>
> I want to rip a radio stream and I can achieve this simply with
>
> /ffmpeg -i http://orion.shoutca.st:8726/stream -t 00:10:00
On Sun, Sep 12, 2021 at 8:34 PM smoothgroove wrote:
> Using chromakey=0x00ff00:0.24:0.14
>
> Is there any way to completely knock out the chromakey background, i.e.
> ignorning the lighting level, shadows and structure of the chroma
> background?
>
> In the examples:
>
> - The arm is casting a
Next time set option for overlay filter.
On Sat, Sep 11, 2021 at 5:47 PM Veronica & Stephen McGuckin <
vsmcguc...@outlook.com> wrote:
> Hello,
>
> I am trying to overlay pgs subtitles onto a 4K video. I have extracted 10
> minute to make sure I have the correct ffmpeg coding. The input file is
g -h outdev=pulse
should be working, if not report bug.
>
> On Thu, Sep 9, 2021 at 8:23 AM Paul B Mahol wrote:
>
> > On Wed, Sep 8, 2021 at 12:11 PM Arif Driessen wrote:
> >
> > > Yes, my initial suspicion is buffer size, but it appears you can't set
> >
On Wed, Sep 8, 2021 at 12:11 PM Arif Driessen wrote:
> Yes, my initial suspicion is buffer size, but it appears you can't set
> buffer size with the alsa device.
>
> It looks like there may be a compile-time constant MAX_BUFFER_SIZE but I
> don't want to have to compile ffmpeg.
>
> On Wed, Sep
On Mon, Sep 6, 2021 at 9:37 PM lejeczek via ffmpeg-user <
ffmpeg-user@ffmpeg.org> wrote:
>
>
> On 06/09/2021 20:11, Paul B Mahol wrote:
> >
> >
> > On Mon, Sep 6, 2021 at 9:05 PM lejeczek via ffmpeg-user
> > mailto:ffmpeg-user@ffmpeg.org>>
> >
On Mon, Sep 6, 2021 at 9:05 PM lejeczek via ffmpeg-user <
ffmpeg-user@ffmpeg.org> wrote:
> Hi guys.
>
> I have two mkv files which on their own playback fine but
> when I concatenate them then that second file's audio is
> pretty distorted.
> In case it might matter, I see as 'concat' works:
>
On Sun, Sep 5, 2021 at 3:43 PM Veronica & Stephen McGuckin <
vsmcguc...@outlook.com> wrote:
> Hello,
>
> I am trying to overlay pgs subtitles onto a 4K video. I have extracted 10
> minute to make sure I have the correct ffmpeg coding. The input file is 5.6
> gigabytes with a bit rate of 78.6
On Fri, Sep 3, 2021 at 8:54 PM Michael Koch
wrote:
> Am 03.09.2021 um 20:01 schrieb Reino Wijnsma:
> > On 2021-09-03T16:25:50+0200, Michael Koch
> wrote:
> >> But if I write all in the FFmpeg command line, then it doesn't work:
> >>
> >> ffmpeg -loop 1 -i 1.png -loop 1 -i 2.png -lavfi
On Fri, Sep 3, 2021 at 3:19 PM Michael Koch
wrote:
> Am 03.09.2021 um 15:09 schrieb Paul B Mahol:
> > On Fri, Sep 3, 2021 at 3:07 PM Michael Koch >
> > wrote:
> >
> >> Am 03.09.2021 um 14:21 schrieb Michael Koch:
> >>> I have two video stream
On Fri, Sep 3, 2021 at 3:07 PM Michael Koch
wrote:
> Am 03.09.2021 um 14:21 schrieb Michael Koch:
> > I have two video streams and want to toggle between them in 1 second
> > intervals. I did try this command:
> >
> > ffmpeg -i in1.mp4 -i in2.mp4 -lavfi streamselect=map='mod(t,2)' -t 30
> >
On Wed, Sep 1, 2021 at 9:26 AM Michael Koch
wrote:
> Hello,
>
> sometimes it's an iterative process to find the best parameters for a
> filter, especially when the filter changes brightness, contrast or colors:
> 1) Edit the filter parameters in the FFmpeg command line
> 2) Run FFmpeg
> 3) Open
On Wed, Aug 25, 2021 at 11:22 AM Michael Koch
wrote:
> Hi,
>
> is it possible to convert the waveform of an entire audio file (or a
> segment of it) to a picture?
> The "showwavespic" filter has the problem that for each sample it draws
> a line from +level to -level. For example, if the input
On Fri, Aug 20, 2021 at 9:47 AM Alex R wrote:
> Hi everyone,
>
> I am attempting to leverage ffmpeg in a project that involves recording
> short audio clips. So far I have gotten some mixed results and I'd like to
> tap into your collective knowledge to ensure my approach is sound.
>
> Context:
On Thu, Jul 29, 2021 at 7:13 PM wrote:
> Hi Paul,
>
> thanks again for answering.
>
> > Please show how you use sidechaincompress filter.
>
> Basically we did what is mentioned here:
>
>
> https://superuser.com/questions/1093607/ffmpeg-sidechaincompress-give-more-importance-to-the-sidechain
>
>
On Wed, Jul 28, 2021 at 10:11 PM wrote:
> Hi Paul,
> thanks for your answer.
>
> > 3 loudnorm calls and 2 dynaudnorm calls?
> > for ducking there is sidechaincompress filter
>
> We tried sidechain of course, but will try this again.
> Our problem is, that the resulting audio from the
On Tue, Jul 27, 2021 at 12:36 PM wrote:
> Hi everyone,
>
> we are using ffmpeg to create hls streams in multiple resolutions and also
> languages from incoming rtmp streams.
> When encoding without the filter_complex for ducking we get a minimal
> latency of
> around 8 seconds, but the moment we
On Sat, Jun 19, 2021 at 8:50 PM wrote:
> Correction - adding the pix_fmt worked "Parcially"...
> When I checked it further I noticed that in the final video, the first part
> (before the transition) had the audio, but the second part had only video
> without the audio.
> I'm probably missing
On Sat, Jun 19, 2021 at 3:58 PM wrote:
> Yes, adding this parameter worked. Thanks for your suggestion, it allows me
> to generate the transitions I need...
> But I'm curious to understand why this parameter is required. All samples I
> was able to find for transitions, nobody included it.
>
use "-joint_stereo 0" option for ffmpeg.
Note that placing order is important.
On Sat, Apr 17, 2021 at 8:06 PM Alan Corey wrote:
> New to the list, I've used ffmpeg for years on video files. But I got
> from a torrent a bunch of classical wma files which not much in Linux
> can deal with. So
You can not reset all dts to 0, that does not make any sense.
On Wed, Apr 14, 2021 at 11:30 AM JD wrote:
> On Tue, Apr 13, 2021 at 11:47 AM Paul B Mahol wrote:
>
> > Please try setts bitstream filter.
> >
> > It is specifically designed for such videos.
> &g
Please try setts bitstream filter.
It is specifically designed for such videos.
Actually I tested setts bitstream filter with broken youtube files.
On Tue, Apr 13, 2021 at 12:18 AM Carl Eugen Hoyos
wrote:
> Am Mo., 12. Apr. 2021 um 23:46 Uhr schrieb JD :
> >
> > Noted.
> >
> > >While there,
On Tue, Apr 13, 2021 at 7:42 AM Emanuel Berg wrote:
> Paul B Mahol wrote:
>
> >>> One problem with just af=dynaudnorm is that when you
> >>> backtrack to play a sequence again the volume drops so
> >>> much you don't hear anything, almost, until it picks
On Tue, Apr 13, 2021 at 2:07 AM Emanuel Berg via ffmpeg-user <
ffmpeg-user@ffmpeg.org> wrote:
> > This
> >
> > af=dynaudnorm
> >
> > works tho to some degree [...]
> >
> > One problem with just af=dynaudnorm is that when you
> > backtrack to play a sequence again the volume drops so much
> >
On Wed, Mar 31, 2021 at 7:40 PM Ulf Zibis wrote:
> Hi,
>
> here is the original: http://www.jugkoeln.de/Videos/Mak/Mak.mpg
> transcoded to mp4: http://www.jugkoeln.de/Videos/Mak/Mak.mp4
> On the clapping hands, you see typical interlacing patterns.
>
> deinterlaced with bwdif:
On Tue, Mar 23, 2021 at 4:58 PM Peter B. wrote:
> Hi Paul!
>
> On 22.03.21 20:15, Paul B Mahol wrote:
> > It is warning for lazy devs.
> >
> > If for any reason it errors in muxing stage, you may use fresh setts
> > bitstream filter to fix it.
>
> Thanks f
On Mon, Mar 22, 2021 at 1:05 PM Peter B. wrote:
> Hi everyone!
>
> I'm using FFmpeg's segment muxer to split MPEG-TS recordings.
>
> It works as expected, but I get the following deprecation warning:
>
> --
> [segment @ 0x55bfa3c540c0] Timestamps are unset in a packet for stream
On Mon, Mar 22, 2021 at 8:45 AM Mark Filipak (ffmpeg)
wrote:
> On 2021-03-22 03:28, Paul B Mahol wrote:
> > On Mon, Mar 22, 2021 at 8:17 AM Mark Filipak (ffmpeg) <
> markfili...@bog.us>
> > wrote:
> >
> >> On 2021-03-22 03:00, Paul B Mahol wrote:
>
On Mon, Mar 22, 2021 at 8:17 AM Mark Filipak (ffmpeg)
wrote:
> On 2021-03-22 03:00, Paul B Mahol wrote:
> > On Mon, Mar 22, 2021 at 6:43 AM Mark Filipak (ffmpeg) <
> markfili...@bog.us>
> > wrote:
> >
> >> Hi All,
> >>
> >> A little h
On Mon, Mar 22, 2021 at 6:43 AM Mark Filipak (ffmpeg)
wrote:
> Hi All,
>
> A little help, please.
>
> Using the latest git source, using '-filter_complex_script', I get a
> mystery error.
>
> I have checked the filter_complex_script: 23fps.ffmpeg.filter.txt, for
> errors. Especially I've
>
On Sun, Mar 14, 2021 at 11:18 PM Mark Filipak (ffmpeg)
wrote:
> On 2021-03-14 10:47, Mark Filipak (ffmpeg) wrote:
> > I hoped that "marked as interlaced" [1] meant that
> >
> >
> 'select=expr=not(eq(interlace_type\,TOPFIRST)+eq(interlace_type\,BOTTOMFIRST))'
> [2]
>
> Have I made myself
patch welcome
On Tue, Mar 9, 2021 at 4:27 PM Peter White wrote:
> Hi all,
>
> as a follow-up to this discussion:
> https://ffmpeg.org/pipermail/ffmpeg-user/2021-March/052358.html
>
> I would like to ask, if frame counting with ffprobe can be made faster.
> The way it done now is obviously by
301 - 400 of 1282 matches
Mail list logo
