I had a look and I think it's just that AVI is not
able to store the sync information. It's a really
outdated container format, and when I used to use
it I seem to recall all sorts of problems with AV sync.
Did you try the command line I posted?
It worked fine here...
Sadly I just noticed
On 07/09/14 15:01, Carl Eugen Hoyos wrote:
Elliott Balsley elliottbalsley at gmail.com writes:
I want to take a video at 60fps and convert it to 24fps,
without adding or dropping frames.
Use the input option -r.
which is ignored if using stream copy...
or was in my testing:-
ffmpeg -r
On Mon, Sep 08, 2014 at 00:42:21 +0100, hh280 . wrote:
ERROR: libfdk_aac not found
You don't seem to have libfdk_aac installed correctly.
Did the libfdk_aac section of the instructions go okay?
https://trac.ffmpeg.org/wiki/CompilationGuide/Centos#CompilationInstallation
Include the
Hi Jay,
On Sun, Sep 07, 2014 at 12:01:25 -0400, Jay Muthialu wrote:
I went through all the patches specified in the link but I am not
sure how to apply the patches. Could you please guide me on how to
apply the patches and also what would be command I would have to use
to take audio input ?
Hi Jay,
[oops, sent too early]
On Sun, Sep 07, 2014 at 12:01:25 -0400, Jay Muthialu wrote:
I went through all the patches specified in the link but I am not
sure how to apply the patches. Could you please guide me on how to
apply the patches and also what would be command I would have to use
tim nicholson nichot20-at-yahoo.com at ffmpeg.org writes:
I want to take a video at 60fps and convert it
to 24fps, without adding or dropping frames.
Use the input option -r.
which is ignored if using stream copy...
Use avi as intermediate container.
(The OP originally didn't say
My expectation is that ffmpeg should start streaming the data.
On other end if I send request from vlc or ffplay then this player should play
the streaming data.
Regards
Ajay
-Original Message-
From: ffmpeg-user-boun...@ffmpeg.org [mailto:ffmpeg-user-boun...@ffmpeg.org] On
Behalf Of
On 9/8/2014 11:53 AM, Ajay Parashar wrote:
My expectation is that ffmpeg should start streaming the data.
On other end if I send request from vlc or ffplay then this player should play
the streaming data.
I don't think so. This is what ffserver is for.
--
Claudiu
Hi,
I made it work but it doesn't work the way I expect. tee is used for
delivering the same file to different destinations, right? We use it to prevent
FFMPEG encode the same file twice. But when I try to use two tee's it needs to
transcode again.
Here is the command and output which DOES
On 08/09/14 09:19, Carl Eugen Hoyos wrote:
tim nicholson nichot20-at-yahoo.com at ffmpeg.org writes:
I want to take a video at 60fps and convert it
to 24fps, without adding or dropping frames.
Use the input option -r.
which is ignored if using stream copy...
Use avi as intermediate
tim nicholson nichot20-at-yahoo.com at ffmpeg.org writes:
h264 in avi is not going to be understood by
much other than ffmpeg...
Works fine with vanilla WMP.
(And I wonder which other application should be
relevant for avi.)
Carl Eugen
___
Yes FFmpeg can feed static file to ffserver.
Is it possible that FFmpeg will listen multicast data packets from VLC and
simultaneously feed these packets to ffserver?
Regards
Ajay
The information contained in this e-mail and any accompanying documents may
Hello, I have Blackmagic Intensity Pro video card and I want to stream video
from that card to multicast using ffmpeg. But my problem is that I can get
only video from that card, no audio.
Ok, time for some commands.
List of my dshow devices:
-=-=-=-=-=-=-
ffmpeg -list_devices true -f
Hello everyone.
I write because i have a problem with cut mp4 and ffmpeg.
I need make precise cuts without re-encode the original video.
FFmpeg makes cuts without trouble but as long as the cut is started on a
keyframe.
To solve this we are doing the next one.
We analyzed the video with
Hello,
I am trying to stream webm from ffserver. I am able to stream using rtsp
from ffserver.
But am stuck trying to stream in webm format.
The feed to the server is from an android device using the javacv libraries.
I have compiled ffmpeg with ogg and vorbis libraries.
I get the following
On 9/8/2014 6:34 PM, Arunava wrote:
I am setting the framesize as 176x144, the framerate as 30 and the
avcodec as H264.
Sorry, I made a typo in the mail. I am setting the avcodec as VP9 using
the javacv api.
___
ffmpeg-user mailing list
You can use this script :D (you must be root.)
https://github.com/csepulveda/Scripts/blob/master/install_ffmpeg.sh
Bye!.
--
César Sepúlveda
Jefe de Plataforma
Mediastream Chile
Email: csepulv...@mediastre.am
Teléfono: +56 2 24029750
2014-09-08 4:36 GMT-03:00 Moritz Barsnick
Le duodi 22 fructidor, an CCXXII, Moritz Barsnick a écrit :
This is OT, but that's a bad idea. It is totally sufficient - actually
important, if you ask me - to build everything as non-root, and use
your root priviledges only for make install (if actually needed).
Make /opt/ffmpeg writable
Oli Jeal oli at storment.co.uk writes:
Not sure if this is the place to ask but I have a
licensing questing regarding if I can use FFmpeg
or not.
You may only use FFmpeg if you agree with the
warranty clause that is part of both the LGPL
and the GPL.
Carl Eugen
It's almost impossible to get any real, substantive information about what is
and is not permitted under the various versions of the various licenses used by
a lot of free software. Ordinarily one would speak to whoever was in charge,
but with most projects there isn't really anyone in charge,
Am 08.09.2014 um 18:39 schrieb Phil Rhodes:
My personal understanding is that any time you send an executable to
someone you must include the source code. This is obviously inappropriate
and pretty silly most of the time, when the end user is non-technical,
but that's the rule. Obviously
As the title says, is this possible? When I create an MXF with “-vcodec
dnxhd”, it’s always OP1a, and therefore can’t be read by Avid Media Composer.
___
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ffmpeg-user@ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user
ffmpeg will not create this for you. You can convert it with bmxtranswrap, a
tool found in the bmx open source library.
-jeanne
- Original Message -
As the title says, is this possible? When I create an MXF with “-vcodec
dnxhd”, it’s always OP1a, and therefore can’t be read by Avid
I have many videos in various formats and sizes on a Mac.
I wish to recode them as modern mp4 to 300p high and as wide as required to
maintain correct aspect ratio, rounded to the nearest 2p. No sound, but all
other qualities the same.
Sorry if this is obvious. I can't even figure out the
Hi,
I can get a specific channel stream using this command:
szap -H -r CHANNEL 1 | ffmpeg -y -i /dev/dvb/adapter0/dvr0 -c copy -f
mpegts udp://10.10.201.14:2345
But I need to get ALL channels on a given transponder and after transcoding
them I need to upstream them to different locations. If I
On Mon, Sep 8, 2014 at 4:22 PM, Soner CAN son...@gmail.com wrote:
Hi,
I can get a specific channel stream using this command:
szap -H -r CHANNEL 1 | ffmpeg -y -i /dev/dvb/adapter0/dvr0 -c copy -f
mpegts udp://10.10.201.14:2345
But I need to get ALL channels on a given transponder and after
2014-09-08 23:24 GMT+03:00 Loadlinx limite...@gmail.com:
On Mon, Sep 8, 2014 at 4:22 PM, Soner CAN son...@gmail.com wrote:
Hi,
I can get a specific channel stream using this command:
szap -H -r CHANNEL 1 | ffmpeg -y -i /dev/dvb/adapter0/dvr0 -c copy -f
mpegts udp://10.10.201.14:2345
Oli Jeal oli at storment.co.uk writes:
I seem to remember reading that Youtube uses (Or
at least used to use) FFmpeg for encoding.
This is insofar not 100% correct as I am quite sure
there is no such site that does not use FFmpeg.
;-)
(And no, youtube does not use FFmpeg for encoding
but
On Mon, Sep 8, 2014, at 12:04 PM, Harold Goranson wrote:
I wish to recode them as modern mp4 to 300p high and as wide as required
to maintain correct aspect ratio, rounded to the nearest 2p. No sound,
but all other qualities the same.
You can use the scale filter:
-vf scale=300:-2
I am trying to convert .flv files to mp4 files. In some cases I get the sound
but in most I do not. I have tried different command line options based some
other posts on the internet, but none of them have worked.
My command line is...
ffmpeg -i in.flv -ar 22050 -c:v libx264 -strict -2 -c:a aac
I am using ARM processor (DM8168).
The total memory available is 256 MB.
Also, there are several other processes going on in the system like live
streaming of 4 streams.
In this scenario system does not always have enough memory to support 138 MB
allocation.
Thanks,
Tithi
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