* R J on Tuesday, October 05, 2021 at 16:05:16 +0530:
I want to convert a .srt file into ttml file.
I tried using below command.
ffmpeg -i my_srt.srt -y srt-to-ttml.xml -v verbose
How is ffmpeg supposed to know to which format you intend to
convert?
Either
ffmpeg -i my_srt.srt -f ttml
* Bo Berglund on Friday, July 09, 2021 at 08:14:28 +0200:
wrote:
It did. See the line I highlighted above. All you need to do now is
something like filter the output of ffmpeg like:
ffmpeg -i bluearrow.mp4 -vf "freezedetect=n=0.01:d=5" -map 0:v:0 -f null - |
grep -q
* Mark Filipak (ffmpeg) on Monday, February 15, 2021 at 21:02:33 -0500:
On 02/15/2021 06:29 PM, Mike Soultanian wrote:
-bigsnip-
You can - easily - let the audio stream start with 0 but the problem
is that the result will not play in-sync, see the setps documentation.
What is 'setps'? A
* Peter B. on Thursday, October 01, 2020 at 14:16:02 +0200:
On 29.09.20 09:56, Christian Ebert wrote:
How about doing quick diagnosis with ffprobe before you start,
something like:
ffprobe -v error \
-print_format default=noprint_wrappers=1:nokey=1 \
-select_streams V -show_entries stream
Hi Peter,
* Peter B. on Monday, September 28, 2020 at 20:42:56 +0200:
...but I still have to find out which files are interpreted as
"yuvj420p" by ffmpeg - and then fish them out and treat them
with a separate command, since I have batches with and without
color_range set, therefore "-pix_fmt
* Paul B Mahol on Tuesday, September 01, 2020 at 10:26:15 +0200:
On 9/1/20, Christian Ebert wrote:
man ffmpeg-codecs recommends apl0 as prores_ks -vendor:
http://ffmpeg.org/ffmpeg-all.html#Private-Options-for-prores_002dks
whereas the WIKI says ap10:
https://trac.ffmpeg.org/wiki/Encode/VFX
Hi,
man ffmpeg-codecs recommends apl0 as prores_ks -vendor:
http://ffmpeg.org/ffmpeg-all.html#Private-Options-for-prores_002dks
whereas the WIKI says ap10:
https://trac.ffmpeg.org/wiki/Encode/VFX#Prores
Which one is it?
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LAST SHIP HOME
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--->>
Hi,
I'm trying to downsample 4 channel (unknown layout) pcm to
stereo. However, aresample's in_channel_layout option has no
effect:
$ ffmpeg -report -guess_layout_max 0 -i 4ac.wav -filter:a
aresample=in_channel_layout=4.0:out_channel_layout=stereo -c:a pcm_s16le -y
out.wav
ffmpeg started on
/301cee61fa61c55b1c178ebfbc590872e8b033e6
as an attempt to fix an existing bug: https://trac.ffmpeg.org/ticket/4184
Indeed, I kind of guessed that this is meant to warn about
duplicate -vf or -af.
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Ein Film von Michael Weber und Christian Ebert
ltumsegelung der Peter von Danzig
Ein Film von Michael Weber und Christian Ebert
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though.
See: https://trac.ffmpeg.org/ticket/5492
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* Mark Burton on Wednesday, May 24, 2017 at 13:20:34 +0100
> On 23 May 2017, at 11:20, Christian Ebert <blacktr...@gmx.net> wrote:
>>> So I looked back at your above -af and realised that the 1024 should
>>> actually be 2112 which is Apple’s chosen fixe
* Mark Burton on Monday, May 22, 2017 at 15:22:34 +0100
> On 15 Apr 2017, at 09:22, Christian Ebert <blacktr...@gmx.net> wrote:
>> Somewhat counterintuitive, but you never know:
>>
>> -filter:a aresample=async=1:first_pts=0,asetpts=PTS-STARTPTS+1024
>>
&g
* Marton Balint on Saturday, April 15, 2017 at 12:25:58 +0200
> On Sat, 15 Apr 2017, Christian Ebert wrote:
>> * Marton Balint on Saturday, April 15, 2017 at 07:55:22 +0200
>>> Last time I checked (a year ago or so), ffmpeg created a correct .mov
>>> edit list
* Marton Balint on Saturday, April 15, 2017 at 07:55:22 +0200
> Last time I checked (a year ago or so), ffmpeg created a correct .mov
> edit list to signal the audio priming.
>
> https://developer.apple.com/library/mac/documentation/QuickTime/QTFF/QTFFAppenG/QTFFAppenG.html
>
> Here is patch
* Carl Eugen Hoyos on Saturday, April 15, 2017 at 00:44:43 +0200
> 2017-04-14 23:44 GMT+02:00 Mark Burton :
>> I find it hard having to accept an encode will always play out of
>> sync on certain players.
>
> Could you elaborate a little?
> So far, for every ticket, it either
* Mark Burton on Friday, April 14, 2017 at 22:44:52 +0100
>> On 14 Apr 2017, at 22:22, Christian Ebert <blacktr...@gmx.net> wrote:
>>>> Also, when you run with -v verbose, you'll see a delay (depends
>>>> on audio codec), for you case it's probably 1024. Maybe
* Mark Burton on Friday, April 14, 2017 at 21:45:07 +0100
>> * Mark Burton on Friday, April 14, 2017 at 16:57:06 +0100
>>> Here is the basic command to reproduce. I have attached the uncut loglevel
>>> 99 console output for this command:
>>> ffmpeg -i SyncTest24p.mov -c:v libx264 -pix_fmt yuv420p
* Mark Burton on Friday, April 14, 2017 at 16:57:06 +0100
> I appreciate this is a tricky area and there appear to be different ways that
> some encoders create AAC streams with regards to the padding and remaining
> samples etc. I won’t pretend to fully understand all the factors, but I would
* Walter Ebert on Tuesday, November 08, 2016 at 09:14:51 +0100
>> On Mobile Firefox - via http://dailymotion.github.io/hls.js/demo/ - i tr
>> to load local m3u8 file but i receive the following error in the demo's
>> info area:
>>
>> Buffer Add Codec Error for video/mp4; codecs=avc1.f4001f
* William Caulfield on Tuesday, July 19, 2016 at 10:24:12 -0700
> On Tue, Jul 19, 2016 at 6:53 AM, Christian Ebert <blacktr...@gmx.net> wrote:
>> * Kieran O Leary on Sunday, July 17, 2016 at 18:14:59 +0100
>>> On 17 Jul 2016 7:05 p.m., "Kelly Haydon" <kellyhay
* Kieran O Leary on Sunday, July 17, 2016 at 18:14:59 +0100
> On 17 Jul 2016 7:05 p.m., "Kelly Haydon" wrote:
>> Hello - I'd like to use ffmpeg to remove the QT timecode which exists
>> in Stream #0:2 of the file expressed below. Basic re-encoding with only the
>> first two
* Louis Letourneau on Thursday, July 07, 2016 at 14:05:37 -0400
> I just tried, converting audio for cutting works fine:
> ffmpeg -y -ss 0.042000 -i 2708-1.mp4 -codec:a pcm_s32le -codec:v libx264
> -crf 23 -preset fast -pix_fmt yuv420p -flags +global_header
> -force_key_frames
* Louis Letourneau on Thursday, July 07, 2016 at 12:38:26 -0400
> When cutting a video, even when transcoding, I often have a small delay
> between the video and audio.
>
> When looking at the video with either vlc or ffplay, I see the first frame
> shown twice.(The first is probably a dup until
* juan carlos Rebate on Wednesday, June 01, 2016 at 04:38:12 +0200
> the problem esque presupposes that I want to play in ffplay, but I do not
> want to play in ffplay I want reproduvcirlo in a html5 player for
> android
Most Android devices cannot play opus audio, use vorbis instead.
--
* Christoph Gerstbauer on Thursday, April 07, 2016 at 10:12:14 +0200
>> Depending on the calculation it may be easier/more intuitive to
>> use framerate:
>>
>> ffmpeg -i input -filter:v framerate=1/10 out-%d.jpg
>>
>> will create one image every ten secs.
>
> I dont need every x seconds one
* Moritz Barsnick on Wednesday, April 06, 2016 at 23:13:25 +0200
> On Wed, Apr 06, 2016 at 21:02:17 +0200, Christoph Gerstbauer wrote:
>> I want to extract 8 thumbnails from every video I have which represents
>> a linear "timeline" over the complete length (100percent) of the video.
> [...]
>>
* MKNwebsolutions . on Thursday, November 12, 2015 at 13:05:56 -0500
> I'm converting a live stream (-i rtmp) into libvpx-vp9 output to
> -webm_chunk. This looks like it's working perfectly (minus a few must have
> future features i.e. auto delete chunks / segments). Next we generate the
> webm
* James Darnley on Tuesday, November 03, 2015 at 01:29:43 +0100
> libx264 definitely supports changing resolution between first and later
> passes.
Also when then second pass involves a change in profile and/or
level, meaning change of frame types?
> It also determines frame types in the first
* Ricardo Kleemann on Monday, August 03, 2015 at 10:57:01 -0700
Thanks everyone for the follow-ups. Does anyone know if SDL also works on
OSX,
yes
or what would be the equivalent?
For this specific purpose (ffmpeg output device) there is no
equivalent as far as I know. But I haven't looked
* Moritz Barsnick on Monday, August 03, 2015 at 09:48:20 +0200
On Sun, Aug 02, 2015 at 17:51:39 -0700, Ricardo Kleemann wrote:
Good point, how would I display ffmpeg on OS X? I’m not quite sure
what the output device would be?
The output device sdl is the first that comes to mind.
$
* Nicolas George on Tuesday, May 05, 2015 at 17:28:33 +0200
As an additional note, the second solution if by far preferable, because
forcing the frame type too frequently ruins x264's bit allocation
algorithms.
As per this thread elsewhere there are different opinions on
that. Others say that
* Werner Robitza on Tuesday, May 12, 2015 at 15:10:07 +0200
On Tue, May 12, 2015 at 2:45 PM, Christian Ebert blacktr...@gmx.net wrote:
* Nicolas George on Tuesday, May 05, 2015 at 17:28:33 +0200
As an additional note, the second solution if by far preferable, because
forcing the frame type too
* tim nicholson on Wednesday, April 08, 2015 at 08:03:52 +0100
On 07/04/15 15:18, Matt Zagrabelny wrote:
On Mon, Apr 6, 2015 at 11:10 PM, Andrew Sinclair ajsincl...@gmail.com
wrote:
Does anyone have any suggestions for faster vp9 encoding?
I don't have the ffmpeg technical chops to analyze
* Claudiu Rad-Lohanel on Monday, February 16, 2015 at 20:05:50 +0200
Can't this actually be related to the player (maybe not implementing
HLS standard 100% precisely)?
Nope:
$ ffprobe -v warning -show_entries format=duration
http://media.blacktrash.org/ccc_trailer1.m3u8
[FORMAT]
* Claudiu Rad-Lohanel on Monday, February 16, 2015 at 12:31:58 +0200
On 2/16/2015 10:46 AM, Moritz Barsnick wrote:
For one, we might identify that your problem has been fixed already.
Here's a (fixed) ticket concerning a similar matter:
https://trac.ffmpeg.org/ticket/2857
No, that didn't
* Wesley Wen on Thursday, December 18, 2014 at 15:43:22 +
I'm transcoding one MPEG2-TS file to MP4, but I noticed the start PTSs of
video and audio of the generated MP4 file are different from the source.
The first video frame starts at 0, while the first audio PTS is negative. I
would
* Moritz Barsnick on Friday, December 05, 2014 at 10:37:14 +0100
Could we know how ffmpeg determine audio channel number from MP4 file? MP4
container indicates it's channel count is 2, but ffprobe shows mono as
expected.
ffprobe probably looks at the actual AAC stream?
Here's a hint:
* Petr Tresnak on Friday, December 05, 2014 at 10:34:50 +
the sound start is cut and audio is ahead even more with your command line.
ffmpeg -i mjpeg.avi -qscale 2 -strict -2 -vcodec mpeg4 -acodec aac out.mp4
encoder : Lavf56.14.100
Duration: 00:00:14.39, start: 0.092880,
* Michael Connolly on Monday, August 18, 2014 at 00:40:53 -0700
I'm new-ish to encoding with FFMPEG and am hoping you guys can
assist on a confusing issue.
I'm transcoding from ProRes with PCM audio to H.264 with AAC
audio. After transcoding, my audio track is advanced (appears
earlier in
* Mark Bogdanoff on Tuesday, July 29, 2014 at 06:36:36 -0700
Have you run Apple's mediastreamvalidator on your streams?
https://developer.apple.com/library/ios/technotes/tn2235/_index.html
Except that these 'HTTP Live Streaming Tools' are not available
anymore.
(I'd be happy to be proven
* Lou on Sunday, July 27, 2014 at 13:51:25 -0800
On Sun, 27 Jul 2014 19:23:45 +0100
Christian Ebert blacktr...@gmx.net wrote:
* Carl Eugen Hoyos on Monday, July 21, 2014 at 15:01:08 +
FFmpeg contains a native opus decoder, libopus is not
needed.
Then the fine new website is not up
* Carl Eugen Hoyos on Saturday, July 26, 2014 at 21:14:14 +
Luke Davis l1 at newanswertech.com writes:
I believe somebody (Carl) said this was fixed in the
latest version, but in a compilation of a version
obtained an hour ago from latest, it is still happening
when segmenting...
* Carl Eugen Hoyos on Monday, July 21, 2014 at 15:01:08 +
Reindl Harald h.reindl at thelounge.net writes:
at least for opus you need to enable the external
library --enable-libopus is your friend
FFmpeg contains a native opus decoder, libopus is not
needed.
Then the fine new website
* Werner Robitza on Wednesday, July 23, 2014 at 13:53:51 +0200
On Wed, Jul 23, 2014 at 12:42 PM, Carl Eugen Hoyos ceho...@ag.or.at wrote:
Why are you using a bitstream filter when re-encoding?
The normal usecase for a bitstream filter is remuxing.
If I don't use it, e.g.
ffmpeg -y -i
* Sam Marrocco on Thursday, July 10, 2014 at 12:31:02 -0400
I have an application that uses ffmpeg to perform file conversions including
applying the filter -vf colormatrix=bt601:bt709. I would like to also apply a
gamma changing filter such as -vf mp=eq2=1:1:0:1:2:1:1:1.
The problem is
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