* 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 srt
* 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 'lavfi.freezedetect.freeze
* 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 filter
* 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?
--
LAST SHIP HOME
Winner of the German Ocean Film Award 2019
--->> https:/
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 20
peg/commit/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.
--
LAST SHIP HOME
Die Weltumsegelung der Peter von Danzig
Ein Film von Michael Weber und Chris
kB other streams:0kB global headers:0kB
muxing overhead: 0.968647%
TIA
Christian
--
LAST SHIP HOME
Die Weltumsegelung der Peter von Danzig
Ein Film von Michael Weber und Christian Ebert
--->> https://lastshiphome.de
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fares with -map 0
though.
See: https://trac.ffmpeg.org/ticket/5492
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Ein Film von Michael Weber und Christian Ebert
--->> https://lastshiphome.de
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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 wrote:
>>> So I looked back at your above -af and realised that the 1024 should
>>> actually be 2112 which is Apple’s chosen fixed encoding delay.
>>> https
* Mark Burton on Monday, May 22, 2017 at 15:22:34 +0100
> On 15 Apr 2017, at 09:22, Christian Ebert wrote:
>> Somewhat counterintuitive, but you never know:
>>
>> -filter:a aresample=async=1:first_pts=0,asetpts=PTS-STARTPTS+1024
>>
>> combined with the -t inca
* 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 whic
* 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 turned out that out-o
* Mark Burton on Friday, April 14, 2017 at 22:44:52 +0100
>> On 14 Apr 2017, at 22:22, Christian Ebert 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 try:
>>&g
* 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 :Operat
* William Caulfield on Tuesday, July 19, 2016 at 10:24:12 -0700
> On Tue, Jul 19, 2016 at 6:53 AM, Christian Ebert wrote:
>> * 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 -
* 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 streams mapped was rec
* 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 "expr:gte(t,n_forced*
* 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 i
* 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.
--
\black\
* 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 imag
* 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.
> [...]
>> Is
* 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 d
* 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 pa
* 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
* MrNice on Monday, August 03, 2015 at 12:06:20 +0100
> Obviously I don't have SDL output:
>
> ./ffmpeg -formats | grep SDL
ffmpeg -devices
narrows it down better for this purpose imho.
--
\black\trash movie _SAME TIME SAME PLACE_
New York, in the summer of 2001
--->> http
* 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.
>
* Tim Greiser on Tuesday, July 07, 2015 at 13:39:41 -0600
> I have been using -vf "transpose=1" and similar filters for some time with
> good results to rotate mp4 files that have been taken on mobile devices. In
> my latest build of ffmpeg, the transpose filter no longer seems to have any
> effect
* Werner Robitza on Tuesday, May 12, 2015 at 15:10:07 +0200
> On Tue, May 12, 2015 at 2:45 PM, Christian Ebert 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
>>>
* 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 tha
* 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
>> wrote:
>>> Does anyone have any suggestions for faster vp9 encoding?
>>
>> I don't have the ffmpeg technical chops to analyze your com
* 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]
duration=254
* 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, th
* 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
* 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,
* 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 h
* Claudiu Rad on Tuesday, September 16, 2014 at 22:37:20 +0300
> i am trying to generate a HLS stream from a standard MP4 file. the
> muxer generates a pair of warnings for each output .ts file.
> however the result seems alright.
>
> why is this? shouldn't this be fixed?
I think that's
https://t
* Carl Eugen Hoyos on Tuesday, August 19, 2014 at 13:51:54 +
> You are right, the following works fine here:
> $ ffmpeg -f webm_dash_manifest -i file.webm
>
> For which file does it fail for you?
All.
> How was it created?
Those I created with ffmpeg, but same for e.g.:
http://video.webmfil
* Carl Eugen Hoyos on Tuesday, August 19, 2014 at 13:54:08 +
> Christian Ebert gmx.net> writes:
>> Or it is this hard to reproduce regression:
>> https://trac.ffmpeg.org/ticket/3861
>
> Is your ticket reproducible without the segment
> muxer? If yes, you have (unf
* Moritz Barsnick on Tuesday, August 19, 2014 at 09:00:49 +0200
> On Wed, Aug 13, 2014 at 12:46:26 +0200, Ibrahim Tachijian wrote:
>> Using the generated "dump.ts" I am trying to do the following command but
>> failing.
>>
>> "./ffmpeg -i dump.ts -acodec copy -vcodec copy -f mpegts dump2.ts"
>
>
* Carl Eugen Hoyos on Monday, August 18, 2014 at 22:02:00 +
> Christian Ebert gmx.net> writes:
>
>> $ ffmpeg -f webm_dash_manifest -i test1.webm
>
> Why do you believe that a file with a webm
> suffix is actually a (xml) manifest?
> And what's wrong with FFm
Hi,
Has anyone got the webm_dash_manifest muxer to work? I just get
this:
$ ffmpeg -f webm_dash_manifest -i test1.webm \
> -f webm_dash_manifest -i test1.webm \
> -f webm_dash_manifest -i test2.webm \
> -f webm_dash_manifest -i test3.webm \
> -map 0:a -map 0:v -map 1:v -map 2:v \
> -c copy -f web
* 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
> earlie
* 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 wron
* Lou on Sunday, July 27, 2014 at 13:51:25 -0800
> On Sun, 27 Jul 2014 19:23:45 +0100
> Christian Ebert wrote:
>> * Carl Eugen Hoyos on Monday, July 21, 2014 at 15:01:08 +
>>>
>>> FFmpeg contains a native opus decoder, libopus is not
>>> needed.
>&g
* Robert Krüger on Sunday, July 27, 2014 at 21:40:50 +0200
> congratulations on the new website! Well done.
+1
--
\black\trash movie _SAME TIME SAME PLACE_
New York, in the summer of 2001
--->> http://www.blacktrash.org/underdogma/stsp.php
___
* Carl Eugen Hoyos on Monday, July 21, 2014 at 15:01:08 +
> Reindl Harald 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 is
* Carl Eugen Hoyos on Saturday, July 26, 2014 at 21:14:14 +
> Luke Davis 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...
>
* Werner Robitza on Wednesday, July 23, 2014 at 15:27:12 +0200
> On Wed, Jul 23, 2014 at 2:46 PM, Christian Ebert wrote:
>>
>> I guess you just want to copy (-c copy both video and audio), not
>> re-encode (-c:v libx264)
>
> I do want to re-encode (for various rea
* Werner Robitza on Wednesday, July 23, 2014 at 13:53:51 +0200
> On Wed, Jul 23, 2014 at 12:42 PM, Carl Eugen Hoyos 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 tmp/tmpPass2
* 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 i
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