Greetings!
Have mercy ...
I have one input file that contains a single video track and two
audio tracks. I discard the unwanted audio
track with the following command:
ffmpeg -i input.mp4 -map 0:v -map 0:a:1 -c copy tmp.mp4
From an input file containing one subtitle track, I add the
I'm sorry this mail is going to be rather long. However, I would
like to explain my opinion on this matter in detail.
I would also like to kindly ask everybody to calm down and refrain
from further ad-hominem attacks, because these are not going to make
things better for anybody.
Thank you for
rueger/lesspain/samples/software/compressor/compressor_dvd_mpeg2_480_24p.m2v':
> >Duration: N/A, bitrate: N/A
> >Stream #0:0: Video: mpeg2video (Main), yuv420p(tv, smpte170m,
> > progressive), 720x480 [SAR 32:27 DAR 16:9], 30 fps, 59.94 tbr, 1200k tbn
> >
/0/0 buffer size: 1835008 vbv_delay:
N/A
At least one output file must be specified
Thanks,
Robert
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the hand offs and embedded players to the source.
On Tue, 12 Apr 2022, 08:21 Bo Berglund, wrote:
> On Tue, 12 Apr 2022 03:46:52 +1200, Robert Jeffares <
> jeffares.rob...@gmail.com>
> wrote:
>
> >A script using curl to download the site to a file then grep m3u8 will
&g
job which does this but use youtube-dl
Hope this answers your question
Regards
Robert
On Tue, 12 Apr 2022, 02:37 Rob Hallam, wrote:
> On Mon, 11 Apr 2022 at 09:42, Bo Berglund wrote:
> > I have tried other linix oriented lists like Ubuntu users but got no
> replies.
>
>
Greetings
I encountered some RPL acorn replay video files of a compression type 7
that ffmpeg doesn't have support for.
I submitted a bug report here: https://trac.ffmpeg.org/ticket/9655
It includes sample files and a link to the file format.
I'd love to see support added to ffmpeg for this comp
k we can rescue much from what you have.
regards
Robert
On 1/01/22 8:57 pm, Stephen Liu via ffmpeg-user wrote:
Hi Robert,
Thanks for your advice.
Those are old VHS and V8 cassette tapes. I haven't played them for long time
since the VHS player and Video camera broken down, leaving
play as smoothly as possible.
VHS never copies that well V8 is alignment [azimuth] critical and be
aware the technology of that time is not as good as what we have today.
Would be interested in seeing a clip not a still. Can you upload 30 sec
or so to a dropbox service?
regards
Robert
On 1
; pick the one(s) you want;
extract in some usable format; place somewhere convenient.
= clutter,chaff,confusion
I vote youtube-dl. It does what it says and it runs on Ubuntu. That said
the other program yt-dlp may be better.
Bo should be downloading by now.
lunchtime!
Nice warm day
regar
ts of producers who make a living from creating
content.
News; especially publicly funded news; should be unencumbered, but it is
polite to seek approval.
I looked at the page source and had a scout around for links. It looks
java based. The video files will be on an internal server.
DIO -ss $HEADNUM -to
$TAILNUM audio-$(date +'%H')_hours.wav
This is far more accurate then my previous attempts have delivered
My thanks to all who contributed to this code.
Robert
--
Robert Jeffares
Communication Consultants
64 Warner Park Avenue
Laingholm
Auckland
New Zealand 06
i/doc-files/G729WaveFileFormat.html
suggests you need to find a converter to make the g729 into something
user friendly.
regards
Robert
On 25/05/21 8:18 am, pink panther wrote:
I have a 325kb audio wave file recorded in g729. I am trying to trim it to
1 minute. I have tried almost every comm
:// URIs cannot work with dynamic RTP payload format (97
I tried including a sdp file but clearly the code ignores it.
I will work with what I have and hopefully it will get sorted sometime.
regards
Robert
On 22/03/21 9:51 pm, Moritz Barsnick wrote:
Hi Robert,
On Mon, Mar 22, 2021 at 18:08
I am getting...
[rtp @ 0x55e52ffe8c00] AAC with no global headers is currently not
supported.
SDP:
with
ffmpeg -i http://localhost:8000/wireless.aacp -acodec copy -vn -flags
+global_header -f rtp rtp://192.168.1.100:7001
is there support for aac coming?
regards
Robert
On Mon, Sep 14, 2020 at 9:53 AM Robert Krüger
wrote:
>
>
> On Sat, Sep 12, 2020 at 3:43 PM Carl Eugen Hoyos
> wrote:
>
>> Am Mi., 9. Sept. 2020 um 17:18 Uhr schrieb Robert Krüger
>> :
>> >
>> > Hi,
>> >
>> > I have a secur
On Sat, Sep 12, 2020 at 3:43 PM Carl Eugen Hoyos wrote:
> Am Mi., 9. Sept. 2020 um 17:18 Uhr schrieb Robert Krüger
> :
> >
> > Hi,
> >
> > I have a security camera file that gives me decoding errors which are
> very
> > likely the reason for several ma
On Sat, Sep 12, 2020 at 3:48 PM Carl Eugen Hoyos wrote:
> Am Fr., 11. Sept. 2020 um 15:48 Uhr schrieb Robert Krüger
> :
>
> > I'm playing around with rav1e and noticed in my first test that only one
> > core is used of the 8 (16 virtual) I have. I tried with and wi
On Fri, Sep 11, 2020 at 9:47 AM Carl Eugen Hoyos wrote:
>
>
> > Am 09.09.2020 um 17:18 schrieb Robert Krüger >:
> >
> > I have a security camera file that gives me decoding errors which are
> very
> > likely the reason for several macroblocks being broken in
like x264, x265, kvazaar multithread fine with the same built.
So it's not a general problem with multithreading.
Working on OS X Catalina.
Thanks for any hints,
Robert
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ticket with the file?
Thanks,
Robert
Command line and output:
~/src/FFmpeg/ffmpeg -i file.h264 -c:v mpeg4 -q 1 file-transcoded.mov
ffmpeg version N-98974-g2a19232c19 Copyright (c) 2000-2020 the FFmpeg
developers
built with Apple clang version 11.0.0 (clang-1100.0.33.17)
configuration
n_sets "id=0,streams=0,2 id=1,streams=4 id=2,streams=1,3
id=3,streams=5" \
-f dash output.mpd
Thanks for your time.
Robert Simmons JR MSc. ● as Founder, CEO and CTO ● Xsiom ● +1 720 520 5676
● Denver, Colorado, USA
https://www.linkedin.com/in/robert-simmons-a3985240
d work.
Thanks,
Robert
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Thank you.
I'm trying to convert an mp3 file to ogg file to send through to a WhatApp
contact through their API. The Whatsapp API only accepts " audio ogg-file in
opus codec". I convert it using
ffmpeg -i audio.mp3 -c:a libopus newfilename.ogg
newfilename.ogg does not play in whatsapp when I
Thank you. I'm new and still learning. :-)
I need to convert with the following settings:
Bitrate of 19.1kbps
1 channel
Sample rate of 16.0khz
How would that look in:
ffmpeg -i audio.mp3 -c:a libopus newfilename.ogg
Is the Writing library " Lavf58.28.102" the same as "WhatsApp"?
Thanks for al
Hi There,
I need to convert an MP3 to OGG (opus codec). I need to execute this (below)
in PHP.
ffmpeg -i filename.mp3 c:a libopus newfilename.ogg
Is this correct?
Can you please let me know.
Thanks,
Rob
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e like I'm an
idiot. There is no such thing as a stupid question if it is asked from a
genuine point of ignorance.
On 3/20/2019 7:15 PM, Carl Eugen Hoyos wrote:
2019-03-21 0:12 GMT+01:00, Robert Aronson :
Honestly, I don't appreciate your condescending attitude.
Then you are unfort
mmand line outside of VirtualDub.
Thank you for your help thus far.
On 3/20/2019 6:44 PM, Moritz Barsnick wrote:
On Wed, Mar 20, 2019 at 12:47:31 -0400, Robert Aronson wrote:
I can't for the life of me get FFMPEG to work with VirtualDub.
Have you tried google?
The following line works f
-i - -c:v copy "%(tempvideofile)"
Output file:" %(outputname).h264
...it doesn't work at all. What can I do to fix this?
Thanks,
Robert Aronson
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On Tue, Dec 18, 2018 at 11:40 PM Carl Eugen Hoyos
wrote:
> 2018-12-18 21:49 GMT+01:00, Robert Krüger :
> > On Fri, Dec 14, 2018 at 5:28 PM Carl Eugen Hoyos
> wrote:
> >
> >> 2018-12-14 17:19 GMT+01:00, Robert Krüger :
> >> > On Fri, Dec 14, 2018
On Fri, Dec 14, 2018 at 5:31 PM Carl Zwanzig wrote:
> On 12/14/2018 8:19 AM, Robert Krüger wrote:
> > Just a comparison: This approach gives me the duration information for a
> > 180 minute file in 1.8 seconds while mediainfo does it in 0.25 seconds
> > (being exact while f
On Fri, Dec 14, 2018 at 5:28 PM Carl Eugen Hoyos wrote:
> 2018-12-14 17:19 GMT+01:00, Robert Krüger :
> > On Fri, Dec 14, 2018 at 3:03 AM Carl Eugen Hoyos
> wrote:
> >
> >> 2018-12-13 17:10 GMT+01:00, Robert Krüger :
> >>
> >> > is there a way to
On Fri, Dec 14, 2018 at 3:03 AM Carl Eugen Hoyos wrote:
> 2018-12-13 17:10 GMT+01:00, Robert Krüger :
>
> > is there a way to get the exact duration of a raw mpeg1 file using the
> > ffmpeg command line? For testing I created a 30 minute test file using
> > ffmpeg (
Hi Carl,
On Thu, Dec 13, 2018 at 5:31 PM Carl Zwanzig wrote:
> On 12/13/2018 8:10 AM, Robert Krüger wrote:
> > Are there command line parameters to make it compute the exact size or is
> > this a design or implementation limitation?
>
> IIRC (and I could be wrong), th
Hi again,
On Thu, Dec 13, 2018 at 6:01 PM Robert Krüger
wrote:
> Hi Carl,
>
> On Thu, Dec 13, 2018 at 5:31 PM Carl Zwanzig wrote:
>
>> On 12/13/2018 8:10 AM, Robert Krüger wrote:
>> > Are there command line parameters to make it compute the exact size or
shows the file's duration as 00:00:04.98 after displaying this warning:
[mpegvideo @ 0x7fe212806000] Estimating duration from bitrate, this may be
inaccurate
Are there command line parameters to make it compute the exact size or is
this a design or implementation limitation?
Thanks,
R
extracted.
In other words I need something like the famed getstream, which was just
demultiplexer/remultiplexer and nothing else without any deeper analysis of
the streams' content.
Is there any switch to force all streams being handled as unknown?
R
ion.
Many thanks!
Robert
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Ah, sorry.
ulimit -a from terminal:
core file size (blocks, -c) 0
data seg size (kbytes, -d) unlimited
scheduling priority (-e) 5
file size (blocks, -f) unlimited
pending signals (-i) 120594
max locked memory (kbytes, -l) 64
max me
Running it with -threads 1 in the terminal works but with PHP I get the same
error:
ffmpeg version 4.0.1-static https://johnvansickle.com/ffmpeg/ Copyright (c)
2000-2018 the FFmpeg developers
built with gcc 6.3.0 (Debian 6.3.0-18+deb9u1) 20170516
configuration: --enable-gpl --enable-version
Sorry for the delay in reply.
I ran ffmpeg with ulimit -a but it comes back with unrecognized option 'a'
It does that with both PHP and terminal.
ffmpeg version 4.0.1-static https://johnvansickle.com/ffmpeg/ Copyright (c)
2000-2018 the FFmpeg developers
built with gcc 6.3.0 (Debian 6.3.0-18+de
nginx-rtmp-module does not support rtp
On Thu, Jun 7, 2018 at 12:48 PM, Mitja Pirih wrote:
> On 07. 06. 2018 08:52, Robert Clove wrote:
> > Hi,
> >
> > My setup is like this:
> >
> > It contains components:
> > Web Cam: which send out the RTP packet
Thank you so much for your help, Moritz!
On Thu, Jun 7, 2018 at 7:12 AM Moritz Barsnick wrote:
> On Wed, Jun 06, 2018 at 15:52:48 -0600, Robert Frazier wrote:
>
> > *ffmpeg -video_size 1280x720 -pixel_format uyvy422 -framerate 60 -vsync
> > > drop -f dshow -rtbufsize 15
Thanks for the link Mitja.
Just wanted to know:
I wanted to use in one of mine another project, my rtp stream is encrypted
with SRTP keys, how will it handle decryption process, any clue?
On Thu, Jun 7, 2018 at 12:48 PM, Mitja Pirih wrote:
> On 07. 06. 2018 08:52, Robert Clove wrote:
>
Hi,
My setup is like this:
It contains components:
Web Cam: which send out the RTP packets when i get a stream request
Agent server: Which receives the stream from the Webcam and forwards the
same to ffmpeg server.
ffmpeg server: this where i need to run the ffmpeg as window service,
receives the
an't read two streams from the DeckLink card simultaneously (though I feel
like I've read I can). Another concern is introduced here too: one of my
streams does not run with frame rate set to 60fps, I need to set it to
59.94fps to work, otherwise it is a black screen.
Would I need to split thes
5 tbn, 35 tbc
>
> From the output, we can konw that the codec of video in flame.avi is not
> h264. So, you can not use -vcodec copy .
>
> try this:
> ffmpeg -i flame.avi -c:v libx264 -f h264 stream1.h264
>
>
>
> 发件人: ffmpeg-user 代表 Robert Clo
Moritz Barsnick wrote:
> On Fri, Jun 01, 2018 at 11:33:26 +0530, Robert Clove wrote:
> > Can someone point me to a libavformat file for which i can use the
>
> What is a libavformat file? You probably mean an AVI file?
>
> > following command ffmpeg -i abc.avi -an -vcode
Can someone point me to a libavformat file for which i can use the
following command ffmpeg -i abc.avi -an -vcodec copy stream.264
On Fri, Jun 1, 2018 at 10:55 AM, Robert Clove wrote:
> Hi,
>
> Can someone point me to a libavformat file
>
> OR
>
> I have a decrypted PCAP f
Hi,
Can someone point me to a libavformat file
OR
I have a decrypted PCAP file, how can i stream it to rtsp://. on
windows platform, what tools can be used for the same.
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Hi,
Can i use FFMPEG to extract the H.264 content from RTP stream.?
How can i achieve that?
Regards
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Hi,
i am using this on windows, i think we tcpreplay on Linux platform.
On Wed, May 30, 2018 at 11:28 AM, Daniel Kucera wrote:
> Hello,
>
> you can use tpcreplay to play the pcap in reatime and capture it with
> ffmpeg. For example like this:
>
> # tcpreplay -i lo -l 0 lkv.pcap
> # route add 23
Hi,
I have pcap(decrypted rtp stream) files of 3-4 video stream.
Can someone please guide me on how can i play it and make ffmpeg capture
the live stream and join/stitch them to get a single output file.
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p 1:1 \
-program title=ProgOne:st=0:st=1 -program title=ProgTwo:st=2:st=3 \
-f mpegts mpts.ts
And also i wanted the end video to be into H.264 format.
On Wed, May 9, 2018 at 9:05 PM, Micael Silva wrote:
> On Wed, May 9, 2018 at 12:11 PM, Robert Clove
> wrote:
>
> > Hi,
> >
>
I have a single camera video and audio stream as input 0
Stream 0:0 Video
Stream 0:1 Audio
I am bringing in ADDITIONAL audio from a separate url as input 1
Stream 1:0 Audio
These are the errors I receive in the log file with some hesitation on
youtube live
[libmp3lame @ 0x29840c0] Queue input is
Hi,
Suppose i make a video call from cisco communicator for one user to another
user.
I want to record that video call into a single video file.
Can i do this from ffmpeg.
Any leads will be appreciated.
--Clove
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In avfilter it is very easy to override the default threading
implementation by just assigning to AVFilterGraph->execute.
However, this doesn't seem to be possible avcodec. You can't just simply
assign to AVCodecContext->execute(2) since ffmpeg internally always
initializes it's own threading.
Am
possibility to extract the correct timecode.
Is this a limitation or am I in fact missing something?
Thanks,
Robert
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Hi Erik,
On Fri, Dec 1, 2017 at 12:25 PM, Erik Dobberkau
wrote:
> Am Freitag, 1. Dezember 2017 schrieb Robert Krüger :
>
> > Hi Kieran,
> >
> > On Thu, Nov 30, 2017 at 5:39 PM, Kieran O Leary <
> kieran.o.le...@gmail.com
> > >
> > wrote:
> >
&g
Hi Kieran,
On Thu, Nov 30, 2017 at 5:39 PM, Kieran O Leary
wrote:
> On Thu, Nov 30, 2017 at 4:38 PM, Kieran O Leary
> wrote:
> > Hi,
> >
> > On Thu, Nov 30, 2017 at 3:46 PM, Robert Krüger
> > wrote:
> >> Hi,
> >>
> >> is there a
ne
Could not write header for output file #0 (incorrect codec parameters ?):
Operation not permitted
Error initializing output stream 0:0 --
Conversion failed!
Is this a conscious limitation of the muxer or am I missing a command line
option?
Thanks,
Robert
I have a scenario that I don't think can be solved by ffmpeg, but will ask
anyway.
I need to relay a live RTMP stream to a Flash Media Server, with the output
continuing uninterrupted even when the input stream drops.
If the input drops, rather than ffmpeg exiting, I'd like to send a black
screen
Hi,
On Sat, Jun 24, 2017 at 3:22 AM, Moritz Barsnick wrote:
> On Fri, Jun 23, 2017 at 20:59:50 +0200, Robert Krüger wrote:
> > Hi,
> >
> > I just did a clean check-out on master did a plain configure with no
> > options and the build failed. Heres the end of the outpu
nyone?
Thanks,
Robert
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from?
On 12/07/2016 07:22 PM, Robert Murphy wrote:
> Hi,
>
>
>
> I am trying to download the latest version of FFmpeg from the official
> website and when you click on the link you get a HTTP 403 Forbidden message.
> Is this a bug and when will th
Hi,
I am trying to download the latest version of FFmpeg from the official
website and when you click on the link you get a HTTP 403 Forbidden message.
Is this a bug and when will the link be restored? Thanks.
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If hls_list_size is enabled then the X-DATETIME tags will be wrong when
segments are started to be removed.
On Thu, Nov 10, 2016 at 4:10 PM, Steven Liu wrote:
> 2016-11-10 18:53 GMT+08:00 Robert Nagy :
>
> > If using hls_list_size together with program_date_time the
> > EXT-X
If using hls_list_size together with program_date_time the
EXT-X-PROGRAM-DATE-TIME
tags start to become wrong when segments are removed.
initial_prog_date_time should probably be incremented by the duration of
the segments removed.
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It would be nice if one could explicitly specify the initial_prog_date_time
instead of always using the current system clocks. Especially in cases
where you are transcoding.
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Hi Erik,
thanks for the hint. I know Exiftool but in this case I need the capability
within ffmpeg. At least that would make it a lot easier.
Robert
On Tue, Oct 25, 2016 at 8:25 PM, Erik Dobberkau
wrote:
> Am Dienstag, 25. Oktober 2016 schrieb Robert Krüger :
>
> > Hi,
> >
Hi,
is it possible to add EXIF metadata like title or description to a JPG or
PNG image using ffmpeg on the command line (or via api but that's probably
for the other list)?
I did not find anything regarding that in the docs.
Thanks in advance,
R
On Thu, Oct 20, 2016 at 10:49 AM, Carl Eugen Hoyos
wrote:
> 2016-10-20 10:41 GMT+02:00 Robert Krüger :
>
> > So would a patch be accepted that implements reading of this
> > additional metadata?
>
> (Sorry if this was not directed to me...)
>
> I am not maintainer
On Wed, Oct 19, 2016 at 11:35 PM, Carl Eugen Hoyos
wrote:
> 2016-10-19 19:31 GMT+02:00 Robert Krüger :
>
> > Are those meta atoms parsed in the mov demuxer?
>
> Yes, but most content (except "hdlr") is ignored.
>
>
Thanks for the clarification!
So would a
On Wed, Oct 19, 2016 at 5:39 PM, Moritz Barsnick wrote:
> On Wed, Oct 19, 2016 at 12:32:00 +0200, Robert Krüger wrote:
> > Hi,
> >
> > I am currently looking at files from Sony A7S cameras (h264 in mp4) and
> > looking at those files in a hex/text editor I can see that
:00:37.00Z
[/FORMAT]
but as you can see, the metadata is not exported.
Any hints would be appreciated.
Thanks,
Robert
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Hi Tobias,
On Mon, Sep 26, 2016 at 3:58 PM, Tobias Rapp wrote:
> On 26.09.2016 13:10, Robert Krüger wrote:
>
>> Hi,
>>
>> On Mon, Sep 26, 2016 at 12:42 PM, Carl Eugen Hoyos
>> wrote:
>>
>> 2016-09-26 12:17 GMT+02:00 Robert Krüger :
>>>
>
On Mon, Sep 26, 2016 at 2:04 PM, Carl Eugen Hoyos
wrote:
> 2016-09-26 13:10 GMT+02:00 Robert Krüger :
>
> > I meant something like writing a 4,6 or 8-channel wav that just
> > really contains 4,6,8 mono channels.
>
Not sure if I understand this sentence.
>
> I meant
Hi,
On Mon, Sep 26, 2016 at 12:42 PM, Carl Eugen Hoyos
wrote:
> 2016-09-26 12:17 GMT+02:00 Robert Krüger :
>
> > I would like to know if it is possible to create a wave file that
> contains
> > n channels of audio which do not have a channel layout like 5.1 or
> stereo,
Hi,
I would like to know if it is possible to create a wave file that contains
n channels of audio which do not have a channel layout like 5.1 or stereo,
much like a file used in broadcast for just storing n signals in one wav
file.
Thanks for any hints on this,
Robert
Hi Kieran,
On Sun, Aug 21, 2016 at 11:09 AM, Kieran O Leary
wrote:
> Hi,
>
> On Sun, Aug 21, 2016 at 8:52 AM, Robert Krüger
> wrote:
> > Hi,
> >
> > is there a systematic way to find out supported pixel formats for a codec
> > (or filter) on the command lin
Hi,
is there a systematic way to find out supported pixel formats for a codec
(or filter) on the command line? I only found ffmpeg -pix_fmts but that is
the global list of all known pixel formats.
Thanks,
Robert
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using the patch myself (thanks for the patch, btw.), just
asking the obvious question. Was this discussed as a patch and rejected? At
least having it as a muxer or command line option seems to be a better
solution than not having it at all in master as there are obviously use
cases for it out there.
Thanks,
Robert
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On Fri, Aug 5, 2016 at 6:28 PM, Paul B Mahol wrote:
> On 8/5/16, Robert Krüger wrote:
> > Hi,
> >
> > I would like to create a file that has a "discrete" channel layout, e.g.
> > have an audio track with 8 channels that are not automatically set to 7.1
&g
On Fri, Aug 5, 2016 at 3:52 PM, Carl Eugen Hoyos wrote:
> Hi!
>
> > Am 05.08.2016 um 10:36 schrieb Robert Krüger >:
> >
> > I would like to create a file that has a "discrete" channel layout, e.g.
> > have an audio track with 8 channels that are no
t (checked using
Quicktime 7 which displays the assignment of each channel and compared that
to files created by cameras or other software (e.g. Adobe Media Encoder)
that have this property.
Thanks,
Robert
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I'm unable to get ffplay/ffmpeg to properly wait and reconnect to a http
stream.
1. ffplay -reconnect 1 -i
http://localhost:8000/file/cir7q2zxh00013l5hh0src1fh.mxf
2. Kill file server
3. Start file server
4. ffplay does not recover
I've also tried with a H264 stream with similar results. Am I doi
On Mon, Mar 21, 2016 at 6:32 PM, Robert Krüger wrote:
>
>
> On Mon, Mar 21, 2016 at 5:31 PM, Carl Eugen Hoyos
> wrote:
>
>> Robert Krüger lesspain.de> writes:
>>
>> > major_brand : mp42
>>
>> > Stream #0:1(und): Audio: pcm_s16
On Mon, Mar 21, 2016 at 5:31 PM, Carl Eugen Hoyos wrote:
> Robert Krüger lesspain.de> writes:
>
> > major_brand : mp42
>
> > Stream #0:1(und): Audio: pcm_s16be (twos / 0x736F7774)
>
> If this is not invalid (I don't know) than the issue remain
On Mon, Mar 21, 2016 at 4:59 PM, Moritz Barsnick wrote:
> On Mon, Mar 21, 2016 at 15:59:04 +0100, Robert Krüger wrote:
>
> > the mp4 muxer currently does not accept pcm in mp4. However, there are
> > several cameras out there, that produce such files. Would it be possible
>
Could not write header for output file #0 (incorrect codec parameters ?):
Invalid argument
Best regards,
Robert
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When streaming a non-seekable http response (without Content-Length in
response) that is cancelled on server side, ffmpeg doesn't seem to provide
any warning and it looks to the user like everything worked fine.
Is there any container or options that could at least print a warning if
the connectio
I'm not suggesting to throw away the frames.
Keep the frames but set an appropriate start time. That way the "trimmed"
frames would be decoded but not displayed.
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Is it possible to achieve frame accurate trimming of clips without
transcoding?
e.g. if I have a 50 fps file with a fixed gop of 50 frames and I'd like to
trim the first 24 frames without transcoding. Shouldn't ffmpeg simply
generate a remuxed file with a startTime of 0.48s and a duration adjusted
>Not if there are filters that alter the timestamps, nor if the source file
> duration is not accurate enough or completely unknown.
True, but if that is not the case then it would? In theory it should
be possible to make a conservative guess.
Would be nice to be able to at least specify it manua
And why is that an issue? It could just pipe out the duration at the start
of the file since it knows the duration from the source file... or is it
possible to manually specify the duration so that it is written?
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When I set "pipe:1" as the output the resulting file always ends up without
a duration in it's metadata/header (ffprobe doesn't find it).
I'm primarily interested in outputting webm, but I have the same issues
with nut and mkv.
Is there a way around this?
_
Is there a way to always merge all audio streams into a single one? Without
knowing in advance how many there are?
I've tried to use the amerge filter but it requires the number of audio
streams provided manually...
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connect to camera protocol RTSP, my link :
rtsp://root:root@10.0.0.1:554/mpeg4/1/media.amp My camera model is AXIS
215 PTZ. Best Regards, Robert Pyrc
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-activation_bytes -i Volume1.aax -vn -c:a copy -v debug
Voume1.mp4
will work.
> On 14 Jan 2016, at 09:22, Moritz Barsnick wrote:
>
> On Thu, Jan 14, 2016 at 08:33:41 +0000, Robert Chalmers wrote:
>> [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f968400] [aax
Sent from the FFmpeg-users mailing list archive at Nabble.com.
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Robert Chalmers
rob...@chalmers.com.au Quantum Radio: http://tinyurl.com
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