Hi,
I can take an MP4 file and stream it out using FFMpeg as HLS. This works
well.
What I want is to take a stream that is coming into my computer and I want
to send it out as HLS.
The working command for the MP4 file is:
ffmpeg -i sintel.mp4 -f hls -hls_time 2 -hls_list_size 5 -vcodec copy
On 1 August 2017 at 22:59, tasos wrote:
> Hello.
> I get a video file with
>
> ffmpeg -y -i /dev/dvb/adapter0/dvr0 -c:v copy -c:a copy foo.avi
>>
> This file is displayed on vlc as a 50fps video file.
>
> Input #0, mpegts, from '/dev/dvb/adapter0/dvr0':
>>
I have an encoder which gives me the option of generating sliced I frames.
These frames have a slice of I frame inserted in each frame, but there is
never a full I frame. If I use FFMpeg to rebroadcast the stream to an HLS
stream, then it works well when I'm not using sliced I frames, but only
But OBS is free and should do the job without much effort
On 2 Jun 2017 17:59, "Ron Barnes" wrote:
Heh... Trying to create a video about online news reports and social media
sites and what not - and until I win the lottery, I have to stick with my
meager skills.
>
> Here's my command line:
>
> ffmpeg -f lavfi -i anullsrc -rtsp_transport tcp -thread_queue_size 512 -i
> rtsp://admin:passw...@xx.xx.xx.xxx/544/h264/ch1/main/av_stream -tune
> zerolatency -vcodec libx264 -preset slower -x264opts
> bitrate=2500:vbv-maxrate=2500:vbv-bufsize=166 -g 120 -pix_fmt
No. Type in the set command, press enter. Then type in the ffplay command.
On Sat, 5 May 2018, 13:49 swades, wrote:
> Hello Respected's I really apologize. i run this command but still not
> able
> to hear audio * set SDL_AUDIODRIVER=directsound*
>
>
I have a system that encodes video from a camera, and this can then be sent
to YouTube. I've written calls to the API to setup the video and test the
stream, then switch to live, and then stop. However, my encoder doesn't
support streaming as RTMP, so I use FFMpeg to convert the base H264 stream
On Mon, 25 Feb 2019 at 15:30, Bernd Butscheidt <
bbutscheidt-at-yahoo...@ffmpeg.org> wrote:
> Hello,
>
>
> I would like to try out this filter:
>
> https://ffmpeg.org/ffmpeg-filters.html#sofalizer
>
> But the website linked to ( http://www.sofacoustics.org/ ) which should
> provide the sofa-files
On Mon, 12 Aug 2019 at 13:47, Reindl Harald wrote:
>
>
> Am 12.08.19 um 14:30 schrieb Simon Brown:
> > I run the following configure script using the latest ffmpeg pulled from
> > github.
> >
> > ./configure --disable-decoders --disable-encoders --enable-deco
On Mon, 12 Aug 2019 at 14:30, Moritz Barsnick wrote:
>
> This looks like libcrypto.so isn't providing the symbols it is supposed
> to. My guess is that /usr/lib/libcrypto.so is broken. (Or is
> accidentally version 1.1.0, where the symbols were renamed, and which
> therefore wouldn't fit to
>
>
>
>
> Am 19.08.19 um 17:45 schrieb Simon Brown:
>
> on modern hardware TLS has no overhead at all after the handshake
> google for aes-ni
> ___
>
> Ah, but this is an ARM A9. Not sure if that's good enou
Hi,
further to my question the other day about getting FFMpeg built with
openssl, I was wondering what the overhead is on generating an rtmps stream
from a normal TS, rather than an rtmp stream. When I streamed to Facebook
live using FFmpeg and an rtmps stream the live video had significant
Hi,
I've built ffmpeg from the latest Git head and enabled the hardware
acceleration for decode and encode on the raspberry pi 4. If I run ffmpeg
-decoders | grep h264 it responds with:
h264
h264_v4l2m2m
h264_mmal
However, if I try and run ffmpeg on a stream and ask it to decode with
h264_mmal I
I have a transport stream arriving at my Raspberry Pi 4, and the h264_mmal
decoder picks it up and decodes it, however although I'm sending
1280x720p50 to it, the decoder reports that it's getting 1280x720p23.98,
ffmpeg -i udp://127.0.0.1:10020 -codec copy -f mpegts output.avi
ffmpeg version
Hi,
I would like to take an H264+AAC transport stream and send it to a web
page. The easiest way to do this is as WebRTC. Can FFmpeg convert an
H264+AAC transport stream into WebRTC? It seems to be able to convert to
most things, eg MP4, RTMP, etc. But I can't find a webrtc option.
Regards,
On Tue, 18 Feb 2020 at 04:06, Gyan Doshi wrote:
>
>
> On 17-02-2020 08:38 pm, Dennis Mungai wrote:
> > Hey there,
> >
> > Try this instead:
> >
> > ffmpeg.exe -i udp://127.0.0.1:9034 -codec copy -bsf:a aac_adtstoasc
> > -flags +global_header -f dash ^
> > -seg_duration 1 -frag_duration 0.1
On Wed, 19 Feb 2020 at 15:19, Ted Park wrote:
>
> Hello,
>
> That’s sort of what a remux does in the first place.
> I take it you are going to re-encode after all, can you post the stdout
> and stderr output from running the commands? Is that the only error or are
> there more errors before that
>
>
>> C:\xampp\htdocs>\ffmpeg-20200216-8578433-win64-static\bin\ffmpeg.exe -i
> udp://127.0.0.1:9034 -codec copy -b:v 6000k -window_size 5
> -extra_window_size 5 -use_timeline 1 -seg_duration 1 -frag_duration 0.2
> -streaming 1 -adaptation_sets "id=0,streams=v id=1,streams=a"
> -dash_segment_type
I'm trying to reduce the latency of streaming video to a web page.
I have tried a few options:
1) WebRTC - great, but doesn't support interlaced video
2) fMP4 - couldn't get a working solution
3) HLS - great, plays interlaced video as well, but latency is appalling
4) LL-HLS - can't find anything
On Mon, 17 Feb 2020 at 12:06, Dennis Mungai wrote:
> On Mon, 17 Feb 2020, 14:50 Simon Brown, wrote:
>
> > I'm trying to reduce the latency of streaming video to a web page.
> > I have tried a few options:
> > 1) WebRTC - great, but doesn't support interlaced video
&g
>
>
> Thanks - I'm not re-encoding at the moment because I can't afford the CPU
> time. I'll tune the encoder that is generating the stream for a suitable
> GOP size.
>
> I've tried your options, Dennis and much the same result:
>
> C:\ffmpeg-20200216-8578433-win64-static\bin>ffmpeg.exe -i udp://
>
>
>
> > Am 18.02.2020 um 17:24 schrieb Simon Brown :
> >
> > Is the codec tag h264 different to what libx264 produces? Is there a
> > reason it can't work with h264 codec tag?
>
> Contrary to what is sometimes claimed you have to read errors from top:
On Tue, 21 Apr 2020 at 00:40, Mark Filipak <
markfilipak.windows+ffm...@gmail.com> wrote:
> To submit a trac ticket on the 'pp' filter, I need a p24 video that's over
> 10 minutes, I need to
> test using that video, and then upload the file and update the ticket to
> 'reopen'.
>
> It has to be
>
>
> Hi,
>
> > Is it possible for ffmpeg to produce a stream conforming to H.323? As I
> > understand it H.323 supports H.264 video and G.711 or OPUS audio. I have
> > an H.264 video stream, so would need to re-encode the audio, but then it
> > needs packaging as H.323 and I haven't found
Hi,
Is it possible for ffmpeg to produce a stream conforming to H.323? As I
understand it H.323 supports H.264 video and G.711 or OPUS audio. I have
an H.264 video stream, so would need to re-encode the audio, but then it
needs packaging as H.323 and I haven't found anything on the web that does
Using:
ffmpeg -i udp://:61120 -c:v copy -an -f h264 pipe:1 |
./myTestOnDemandRTSPServer
I have had some raw h264 video finally playing at the correct frame rate,
etc. However, in using ffmpeg to pipe it to myTestOnDemandRTSPServer I
find that ffmpeg runs out of buffer space before long and so I
Hi,
is it possible to stop FFmpeg from probing the input and just to tell it
exactly what it is getting (and obviously suffering the consequences if
it's different)? I can reduce probesize but I want essentially zero delay
through ffmpeg (no encoding, just repackaging a transport stream as MP4).
I have followed the instructions at:
https://trac.ffmpeg.org/wiki/CompilationGuide/MSVC
I have installed msys and yasm.
I have set up the build environment.
I clone ffmpeg with 'git clone git://source.ffmpeg.org/ffmpeg.git
I run
./configure --target-os=win64 --arch=x86_64 --toolchain=msvc
I was wondering if there is any way to use FFmpeg to create a virtual
camera for a PC/Mac from an H264 stream source. I know FFmpeg can use
directshow as an input device.
Thanks,
Simon
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On Fri, 10 Jul 2020 at 21:19, Moritz Barsnick wrote:
> On Thu, Jul 09, 2020 at 13:38:27 +0100, Simon Brown wrote:
> > I was wondering if there is any way to use FFmpeg to create a virtual
> > camera for a PC/Mac from an H264 stream source. I know FFmpeg can use
> > directsh
> Pro Tip: When insulting people for being stupid, always make sure you use
> language that avoids accidental self-owns.
>
> Your Truly,
>
> NotHarald
>
> For goodness sake, just quit guys. This is ridiculous. I don't like much
of what Reindl says, but that doesn't mean I have to reply to
I have used ffmpeg for a number of different things, including decoding
streams, transcoding format, encoding streams, etc. It's a wonderful
tool. Now I want to design something that can do three things at the same
time, eg
a) display a stream on a monitor,
b) send the raw video out on a
> Simon, of course English isn't his first language, and truth be told, no
> one cares he's the English language equivalent of a word salad in a
> hurricane of cow patties.
> But mobilizing your hapless grab bag of language skills as your weapon of
> choice to prove that everyone else is the
Hi,
I'm trying to decode video using the libav libraries. To test I'm running
ffmpeg with some options to see if it can run fast enough.
I was advised that the h264_v4l2m2m decoder would be faster on the
raspberry pi compared to the h264_mmal decoder because it needed less
memory copying between
I want to compile the latest git head on my Raspberry pi. The Pi comes
with a packaged version of ffmpeg with a huge config list of what is and
isn't enabled. Is there a simple way of using this config for the
./configure stage so that I don't have to type the whole lot in?
Cheers,
Simon
>
>
>
> where is the problem doing simply copy?
> ./configure
>
>
> None - thank you. I copied to file and then input from file to
./configure in the end. But it's the obvious answer. Thanks for replying.
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>
>
> On Fri, May 14, 2021 at 14:31:42 +0100, Simon Brown wrote:
> > Hi,
> > I have a mpeg2 transport stream with video as H264 and audio as AAC LC.
> If
> > I use the following command to convert it to fragmented MP4 by just
> copying
> > the encoded data,
Hi,
I have a mpeg2 transport stream with video as H264 and audio as AAC LC. If
I use the following command to convert it to fragmented MP4 by just copying
the encoded data, then the result is now AAC, and not AAC LC. If instead I
re-encode with AAC asking for profile:a aac_low then I get AAC LC.
>
>
>
> Ok, I've cheated somewhat, and taken my (much older) ffmpeg and
> editted isom.c so that rather than substituting 0x40 for AV_CODEC_ID_AAC it
> now substitutes 0x67 instead. If I look at the resulting output file with
> mp4info and the output of FFmpeg itself they both show that it's MP4
>
>
>
> So it seems to be a matter of TS demuxing passing along the codec
> extradata correctly.
>
> Regards,
> Tobias
>
> Thank you both for your replies.
Ok, I've cheated somewhat, and taken my (much older) ffmpeg and
editted isom.c so that rather than substituting 0x40 for AV_CODEC_ID_AAC it
>
>
> z!
> who doesn't speak German but can occasionally get the meaning (and google
> translate does an acceptable job)
>
>
> Interesting to know that someone with the surname Zwanzig doesn't speak
German. :-)
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On Tue, 3 Aug 2021 at 17:01, Thiago Franklin
wrote:
> can anybody help me? I've already turned the internet upside down and I
> can't find a solution. This problem only happens in some videos and others
> work normally.
>
> Em seg., 2 de ago. de 2021 23:03, Thiago Franklin <
>
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