0x73cf08003b00] Failed to avformat_open_input '-'
[AVFilterGraph @ 0x73cf080012c0] Error processing filtergraph: No such
file or directory
amovie=-: No such file or directory
Neither does using the "pipe:" protocol.
I can't find any other way to read
0
channel_layout:'stereo' nb_samples:1024
[lavfi @ 027a34babc80] All info found
Input #0, lavfi, from 'anullsrc':
Duration: N/A, start: 0.00, bitrate: 705 kb/s
Stream #0:0, 1, 1/44100: Audio: pcm_u8, 44100 Hz, stereo, u8, 705 kb/s
Succes
I get the same result using this command:
ffmpeg -rtsp_transport tcp -i
rtsp://onvif:bridge@localhost:554/live/41cf4f34-e137-4559-8278-47d912c64c5b
rtmp://a.rtmp.youtube.com/live2/xxx -loglevel trace
Result:
ffmpeg version N-91172-gebf85d3190 Copyright (c) 2000-2018 the FFmpeg developers
b
pts and put them into the container to avoid those messages?
Here are the logs (for first snippet): https://pastebin.com/xWq9U2sN
(The message from the pastebin-logs [h264 @ 005037c0] Stream #0: not
enough frames to estimate rate; consider increasing probesize I could fix with
MP_V2 from mkv with -i. Without success.
-Ursprüngliche Nachricht-
Von: ffmpeg-user Im Auftrag von Carl Eugen
Hoyos
Gesendet: Montag, 17. Dezember 2018 15:10
An: FFmpeg user questions
Betreff: Re: [FFmpeg-user] Audio converting and muxing error/warning messages
2018-12-17 10:04 GMT+01:00, Fel
Thank you. 02:24:29:53 is the correct duration.
The timestamp and pts problems are still present.
I tried to generate pts with -fflags +genpts. Without success.
I tried to generate new timestamps with -vsync drop.
Without success.
For the frame size is not set problem.
I get the frame rate:
f
> Please provide a command line you tested including complete,
> uncut console output.
Please have a look further up in this conversation.
> Don't use this functionality unless you have a very good reason.
My reason is to avoid those messages and generate a clean output.
_
Nobody has an idea?
I tried -fflags +genpts and -r on the raw h264 stream.
But that also didn’t work.
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I finally solved it.
I simply take the video-stream directly from one container to the other:
ffmpeg -i german_dd_to_alac__english_truehd_to_alac.mkv -map 0:0 -c copy
german_dd_to_alac__english_truehd_to_alac.m4v
And than do my thing:
ffmpeg^
-i german_dd_to_alac__english_truehd_to_alac.m4v -i
8 and force framerate
with -r. But that don't work. When I try to mux the raw h264 stream (ffmpeg
-r 24000/1001 -i video.264 -c copy test.mp4) I get this message: "pts has no
value time=00:02:38.19 bitrate=1312.4kbits/s speed= 154x" many times. Can
you help me?
Best regards,
Felix
Thank you for your answer.
I appended the command line and console output in my first mail. Please scroll
down.
The raw videostream is a cfr stream.
Every frame has a duration of 1/(24000/1001). So there are no timestamps or
anything like that.
I don’t understand the error message.
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I really don't understand this.
For the first command the metadata (handler, language) is stored correctly.
The second keeps the metadata from temp.mkv.
What am I doing wrong?
Works:
ffmpeg^
-i video.h264 -i audio1.dts -i audio2.dts -i sbtl1.srt -i sbtl2.srt^
-map 0:0 -map 1:0 -ma
Oh man.
Your request forcing me to redo the "working" command.
And I realized it also didn't work!
I thought it would work because I do it many times before so don't tested it.
But ...
I updated from v4.0.3 to v4.1.
With v4.0.3 everything works fine!
And I was searching for many hours.
I can
That works!
Thank you.
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Hello,
I'm trying to convert Dolby TrueHD embedded in mkv container to alac and mux
it into mp4(m4v)-container.
I get the following message: [out_0_0 @ 0045a0c0] 100 buffers queued
in out_0_0, something may be wrong.
This can be replicated using one of those Dolby Trailers:
https://thedig
Additional Information:
The message occurs with every output (alac, flac, wav, ...).
I don't understand the message.
Can you tell me what it means?
And something must be wrong or not?
I only have the problem with Dolby TrueHD.
DTS-HD MA, DTS-HD HR, DD, DD+ and so on are all working fine.
__
Thank you.
I opened issue #7703.
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According to the answer by Hendrik.
This message is never related to harm the output?
Just to make that clear.
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All right.
So I‘ll ignore the message in the future.
Maybe it should be hidden to not confuse the user.
Best
Felix
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Hello,
I'm trying to convert DTS-HD Master Audio track embedded in mkv container to
alac and mux it into mp4(m4v)-container.
After the half I get the following messages:
[dca @ 02a7bb80] Failed to decode block code(s)
Error while decoding stream #0:1: Invalid data found when processing inp
I created the sample file: ffmpeg -ss 01:26:00 -i input.mkv -t 00:02:00 -map
0:1 -c copy output.mkv
The problem can easily be reproduced and occurs exactly at the chapter mark.
https://ufile.io/g1l40
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The file isn't broken.
It was accurately ripped from an official medium.
It is a special edition from a movie which regular duration is 01:27:12.
The extended duration is almost 3 hours.
So could it be that the audio stream from the regular movie has a DTS Core and
the audio stream from the exte
I did this:
ffmpeg -i input.mkv -map 0:1 -c copy -f rawvideo out1.dts
clean log
mkvextract tracks input.mkv 1:out2.dts
also no errors or warnings
out1.dts and out2.dts are bit-identical.
Then I tried to convert the raw dts-stream:
ffmpeg -i out1.dts -c:a alac alac.m4a
ffmpeg version N-93069-g6
After the error video and audio are a bit asynchronous.
I'll create a new rip and do further testing.
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ffmpeg-us
Sorry, this was a problem of the ripping software I used.
No problem anymore.
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Hi everyone!
I’m new to FFMPEG and look for help with saving video stream from a webcam in
.mpg format. What I have is a working code that will save the video stream in
.avi format in 1 hour junks:
ffmpeg -r 20 -f dshow -s 960x544 -i video="Microsoft LifeCam HD-5000" -codec
copy -f segment -se
Thanks a lot for your reply!
>> …but I’m having trouble saving in .mpg format. The reason
>> why I would like to save in .mpg format is that
>
>> (1) .avi files are very large
>
> This is wrong and indicates that you don't know what you
> are talking about. This is generally no problem (that'
>
>> Thus, is it possible to save in avi format, but use
>> a different codec that will generate smaller files?
>
> Yes, try "-vcodec mpeg4 -qscale 2" or
> "-vcodec mpeg1video -mbd 2 -qscale 2" or use h264.
>
> Carl Eugen
“-vcodec mpeg4 -qscale 2” reduces the file size by a factor of 50 and
2 -f segment -segment_time 3600 -reset_timestamps 1
-segment_start_number 1 -segment_format avi -r 25 c:\Users\Desktop\out%%03d.avi
Any idea what might be going on?
Leo
> On Aug 5, 2015, at 11:24 AM, Felix Baier wrote:
>
>
>>
>>> Thus, is it possible to save
> On Aug 28, 2015, at 12:10 PM, Carl Eugen Hoyos wrote:
>
> Felix Baier ffmpeg.org> writes:
>
>> I changed “-vcodec mpeg4 -qscale 2” to “-vcodec mpeg4 -q:v 2”
>> after I got a notification that the first version was
>> ambiguous.
>
> For
My audio recorder was in a different time zone, plus there was an
inaccuracy of a few minutes:
$ ffprobe -v quiet -print_format flat -show_format "ZOOM0004.WAV"
[…]
format.tags.date="2021-09-13"
format.tags.creation_time="19:21:33"
[…]
This needs to be adjusted by adding 7:56,
I tried now to change the creation time, but this actually removes it:
$ ffprobe ZOOM0004.WAV
[…]
Metadata:
encoded_by : ZOOM Field Recorder F1
date: 2021-09-13
creation_time : 19:21:33
time_reference : 3345264000
coding_his
Bouke writes:
> why would you want to change the creation time if you have a bwf
> timestamp?
Because I wasn’t aware. Thanks for pointing that out. Maybe I best use
[BWF MetaEdit][1] for that?
[1]: https://mediaarea.net/BWFMetaEdit
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I'm streaming to Wowza:
$ ffmpeg -re -i sintel_trailer-480p.mp4 -c copy -f flv \
rtmp://192.168.56.1/live/myStream
*How do I switch to another stream once playback has finished?*
Note that when playback of the first stream starts, it is not clear
what will be the next stream. So somethin
Say, I have a TV channel and, at night time, I want to broadcast a
[test pattern][1], then:
*How do I approach that? Should I convert the image into a video and
loop that? How?*
It's just a JPEG image. No sound needs to be streamed.
[2]: http://de.wikipedia.org/wiki/Testbild#mediaviewer/File:FuB
> http://www.wowza.com/...
Thanks for the links, but: *Can't I do that from ffmpeg?*
Thing is, I want to script switching of streams, and I want to have
the option to later use another server for broadcasting.
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On Sat, Oct 25, 2014 at 8:56 PM, DopeLabs wrote:
> ffmpeg -i http://shoutcasthost.com:8000 -re -loop 1 -i
> /path/to/image.jpg -c:a libfdk_aac -b:a 192k -c:v h264 -preset
> ultrafast -pix_fmt yuvj420p -bsf h264_mp4toannexb -f mpegts
> udp://127.0.0.1:1?pkt_size=1316
Thanks! I modified that a
Thanks Luke! Hm, I thought it was standard feature. But being not, I
guess I'll first look at Wowza which seems to be scriptable with
ActionScript. Fortunately, I've used ActionScript before: Many years
ago, I wrote a streaming video player, based on AMS.
Anyhow, there must be a smooth solution.
I’ve been thinking: For gapless playback, it’s probably best to have two
video streams running, and then switch between these with some kind of
mixer. As I’ve just found out, software implementations of mixers for
video are called [Software vision mixers][1].
[1]: http://en.wikipedia.org/wiki/Soft
On Sun, Oct 26, 2014 at 3:00 PM, Carl Eugen Hoyos
wrote:
> "-re" should be unneeded, "-bsf aac_adtstoasc" makes no sense if there
> is no audio input.
Thanks! I’m just getting started, currently investigating what setup I
need.
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In the meantime I discovered [FreeJ][1], which should be able to do
what I want. However, it's limited to Ogg Theora, so would require
live reencoding before sending the stream to the broadcasting
software.
Also, I asked on one of the GStreamer mailing lists if that tool could
be a solution. There
0:0 (copy)
Press [q] to stop, [?] for help
[matroska,webm @ 023c9800] Auto-inserting h264_mp4toannexb bitstream
filter
video:29430401kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB
muxing overhead: 0.013187%
Part1:
General
Complete name :
C:\Us
Concat filter is unfortunately not possible.
I cannot lossy reencode the stream.
Is there another solution?
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ff
head: 0.001046%
I also tried -t as an input option.
Same problem.
Best
Felix
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> What is the point of doing this calculation?
I need to trim 1.001 seconds
> It’s probably because flac is mostly a stream format, and you’re doing a
> stream copy. Re-encoding should fix all of this, it’s relatively inexpensive,
> > and it’s lossless after all, but if you really don’t want thi
eaders:0kB
muxing overhead: unknown
à 01:33:00.70 = 5580.7 seconds
Why is there a difference of 0.166708 seconds?
Best
Felix
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Now I'm confused
From my calculation:
Audio length: 5580.7 seconds
Video length: 133799/(24000/1001) = 5580.533292 seconds
From the container:
Audio length: 5580.704 --> that is what ffmpeg tell me when I reencode the
stream
Video length: 5580.575 --> Adobe Premiere Pro showed me 133799 frames,
To calculate by reencode is enough for me.
But is it possible to get a thousandth of a second?
0.00 is not precise enough for me.
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(5177/(24000/1001))-(2928/(24000/1001)) -codec flac audio_cut.flac
ffmpeg.exe -i audio.flac -ss 122.122 -t 93.80204167 -codec flac
audio_cut.flac
And from the documentation I should use -ss as an output option to be
accurate. But what is with -t? As input or output?
Best
I use -ss and not -t.
But thanks for the info.
I can bypass the problem with -codec flac.
Then it is accurate.
But it is strange anyway:
_ffmpeg.exe -i _audio.flac -t 10 -codec copy _audio_2.flac
—> it shows me 9.98
_ffmpeg.exe -i audio_2.flac -f null -
—> but encode is 10.08
_ffmpeg.exe -i _au
So what is the accurate way to cut audio by video frame?
I tried filters.
But they don‘t seem to be accurate for audio.
https://superuser.com/questions/866144/cutting-videos-at-exact-frames-with-ffmpeg-select-filter
So is the only way is to use -ss and -to and hope that timestamps are correct?
Audio and video stream have the same start_time
ffprobe -show_entries stream=codec_type,start_time -v 0 -of compact=p=1:nk=0
THE_GIRL_WITH_THE_DRAGON_TATTOO.Title1.mkv
stream|codec_type=video|start_time=0.00
stream|codec_type=audio|start_time=0.00
So is this the best way to cut frame 29
I want to make my own dub of a movie:
I check which frames are identical with Adobe Premiere Pro.
The UNCUT-movie has 133799 frames in Premiere but mkv-container shows
133800. So I add one frame for ffmpeg commands (+1).
Now these are the scenes I want to insert/replace (starting with
UNCUT-movie)
annels but ffmpeg does with ALAC?
Best
Felix
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Thank you for your fast reply.
I replaced -ac 8 with -filter:a "aformat=channel_layouts=7.1(wide)".
ffmpeg -i test.mkv -map 0:0 -map 0:1 -c:v copy -c:a alac -filter:a
"aformat=channel_layouts=7.1(wide)" test.m4v
ffmpeg version git-2019-12-06-b66a800 Copyright (c) 2000-2019 the FFmpeg
developers
That works.
Thank you.
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