Re: [FFmpeg-user] Why FFMPEG?

[Simon Brown]

>
>
> 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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Re: [FFmpeg-user] When downloading HLS video with FFMPEG it does not download the audio

[Simon Brown]

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 <
> thiago.frank...@portalser.org> escreveu:
>
> > I have a problem, when I try to download an HLS video with FFMPEG, it
> > downloads the video track, but it doesn't find the AUDIO. When running
> the
> > HLS .m3u8 file in a player, it plays normally, audio and video, but when
> > trying to download, it shows an error message saying that the audio
> cannot
> > be found.
> >
> > https://teste-etv.espiritismo.tv/437602.m3u8
> >
> > *I'm running the following command:*
> >
> > ffmpeg.exe -i "https://teste-etv.espiritismo.tv/437602.m3u8; -codec:a
> > libmp3lame -b:a 96k teste-hls.mp3
> >
> > *OR*
> >
> > ffmpeg -i "https://appsetv.b-cdn.net/hls/437602/437602.m3u8; c copy
> -absf
> > aac_adtstoasc teste-hls.mp3
> >
> > *Displays the following error when trying to download the audio:*
> >
> > Consider increasing the value for the 'analyzeduration' and 'probesize'
> options
> > Input #0, hls, from 'https://appsetv.b-cdn.net/hls/437602/437602.m3u8':
> >   Duration: 00:01:41.00, start: 2.08, bitrate: 0 kb/s
> >   Program 0
> > Metadata:
> >   variant_bitrate : 2509173
> > Stream #0:0: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p,
> 1920x1080 [SAR 1:1 DAR 16:9], 24 fps, 24 tbr, 90k tbn, 48 tbc
> > Metadata:
> >   variant_bitrate : 2509173
> > Stream #0:1: Audio: aac ([15][0][0][0] / 0x000F), 0 channels, fltp
> > Metadata:
> >   variant_bitrate : 2509173
> >   Program 1
> > Metadata:
> >   variant_bitrate : 1205957
> > Stream #0:2: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p,
> 1280x720 [SAR 1:1 DAR 16:9], 24 fps, 24 tbr, 90k tbn, 48 tbc
> > Metadata:
> >   variant_bitrate : 1205957
> > Stream #0:3: Audio: aac ([15][0][0][0] / 0x000F), 0 channels, fltp
> > Metadata:
> >   variant_bitrate : 1205957
> >   Program 2
> > Metadata:
> >   variant_bitrate : 1165600
> > Stream #0:4: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p,
> 960x540 [SAR 1:1 DAR 16:9], 24 fps, 24 tbr, 90k tbn, 48 tbc
> > Metadata:
> >   variant_bitrate : 1165600
> > Stream #0:5: Audio: aac ([15][0][0][0] / 0x000F), 0 channels, fltp
> > Metadata:
> >   variant_bitrate : 1165600
> >   Program 3
> > Metadata:
> >   variant_bitrate : 656245
> > Stream #0:6: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p,
> 640x360 [SAR 1:1 DAR 16:9], 24 fps, 24 tbr, 90k tbn, 48 tbc
> > Metadata:
> >   variant_bitrate : 656245
> > Stream #0:7: Audio: aac ([15][0][0][0] / 0x000F), 0 channels, fltp
> > Metadata:
> >   variant_bitrate : 656245
>
> Probaby not much use, but I tried:
ffmpeg.exe -i "https://teste-etv.espiritismo.tv/437602.m3u8; -map 0:a:0 -f
mp3 teste.mp3
and that said converting aac to mp3 but then found no data.

Your output above shows 4 programs, each with h264 video and aac audio, but
each audio stream says "0 channels".
I think that is your first problem at least.  ffplay won't play audio, VLC
wouldn't play anything.

Regards,
Simon
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Re: [FFmpeg-user] Preserving AAC LC status when converting to fragmented MP4

[Simon Brown]

>
>
>
> 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 AAC
> LC.  However, the codec string given by mp4info is mp4a.67.  If I want to
> send it to a web page it has to be AAC LC, but the typically accepted codec
> string is mp4a.40.2 which is also supposed to map to AAC LC.  So now I am
> wondering how I would encode that in the ESDS or similar so that a) mp4info
> is happy that it's mp4a.40.2 and that the chrome video decoder is happy
> with the media type (because mp4a.67 codec string although in theory
> supported gives me "Unrecognised media codec: mp4a.67" in Chrome.
>
>
> Another update:
I've updated movenc.c now, having reverted isom.c to original.
This (as I'm only ever dealing with AAC_LC) hacks the decoder specific info
len to force an extra 10 bytes if the track->par->codec_id =
AV_CODEC_ID_AAC.
And then later on where decoder specific info is inserted from vos_len and
vos_data (I couldn't find where these get set - hence me doing it this way)
I force these to insert the extra info required to make it think it's
mp4a.40.2.
This was done by comparing the esds packet from a faulty encoding and from
the correct encoding.
The correct encoding for mp4a.40.2 adds a descriptor of 5, length 5, and
then the bytes: 0x11 0x90 0x56 0xe5 0x00
mp4info accepts this is mp4a.40.2 and so does Chrome.

Thank you again for pointing me in the right direction, and apologies that
I've hacked ffmpeg purely for my purposes without finding a "proper"
solution.

Cheers,
Simon
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Re: [FFmpeg-user] Preserving AAC LC status when converting to fragmented MP4

[Simon Brown]

>
>
>
> 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
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 AAC
LC.  However, the codec string given by mp4info is mp4a.67.  If I want to
send it to a web page it has to be AAC LC, but the typically accepted codec
string is mp4a.40.2 which is also supposed to map to AAC LC.  So now I am
wondering how I would encode that in the ESDS or similar so that a) mp4info
is happy that it's mp4a.40.2 and that the chrome video decoder is happy
with the media type (because mp4a.67 codec string although in theory
supported gives me "Unrecognised media codec: mp4a.67" in Chrome.

Regards,
Simon
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Re: [FFmpeg-user] Preserving AAC LC status when converting to fragmented MP4

[Simon Brown]

>
>
> 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, 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.  But
> if
> > the input source is AAC LC why would it change the output type to AAC?
> >
> > ffmpeg.exe -f mpegts -fflags +nobuffer+nofillin -probesize 500 -i
> > soc_udp_rx_02.ts -c:a copy -bsf:a aac_adtstoasc -c:v copy -f mp4
> > -frag_duration 8 -movflags +empty_moov+default_base_moof -metadata
> > title="media source exentions" testaudio.mp4
>
> This shouldn't change anything in the AAC stream, unless the bitstream
> filter is capable of ruining it.
>
> What does it say about the original? (Nothing, presumably, because it's
> MPEG-TS.) What does ffmpeg say about the output file? Do you have any
> other tool which can use to check?
>
> Cheers,
> Moritz
>
> Hi Moritz,
Thank you for your reply.  If it is the bitstream filter could you point me
to the relevant source file that is responsible for this filter?

mp4info doesn't deal with TS, as you surmise.  FFMpeg reports the output
file as AAC (not LC).  I'm not sure VLC gives that detail.  But given that
the ffmpeg processor and the mp4info program both seem to concur when there
is or isn't AAC-LC I'm not sure there is much value in trying a third
tool.

I will download the latest prebuilt binaries for Windows and give it a try
with those.  I'd still be interested in the source file responsible as the
version on the embedded system I'm using is not the most recent build, and
rebuilding for that isn't something I want to do at this stage.

Regards,
Simon
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[FFmpeg-user] Preserving AAC LC status when converting to fragmented MP4

[Simon Brown]

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.  But if
the input source is AAC LC why would it change the output type to AAC?

ffmpeg.exe -f mpegts -fflags +nobuffer+nofillin -probesize 500 -i
soc_udp_rx_02.ts -c:a copy -bsf:a aac_adtstoasc -c:v copy -f mp4
-frag_duration 8 -movflags +empty_moov+default_base_moof -metadata
title="media source exentions" testaudio.mp4
ffmpeg version git-2020-06-19-2f59946 Copyright (c) 2000-2020 the FFmpeg
developers
  built with gcc 9.3.1 (GCC) 20200523
  configuration: --enable-gpl --enable-version3 --enable-sdl2
--enable-fontconfig --enable-gnutls --enable-iconv --enable-libass
--enable-libdav1d --enable-libbluray --enable-libfreetype
--enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb
--enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy
--enable-libsoxr --enable-libsrt --enable-libtheora --enable-libtwolame
--enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264
--enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma
--enable-zlib --enable-gmp --enable-libvidstab --enable-libvmaf
--enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa
--enable-libspeex --enable-libxvid --enable-libaom --disable-w32threads
--enable-libmfx --enable-ffnvcodec --enable-cuda-llvm --enable-cuvid
--enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2
--enable-avisynth --enable-libopenmpt --enable-amf
  libavutil  56. 55.100 / 56. 55.100
  libavcodec 58. 93.100 / 58. 93.100
  libavformat58. 47.100 / 58. 47.100
  libavdevice58. 11.100 / 58. 11.100
  libavfilter 7. 86.100 /  7. 86.100
  libswscale  5.  8.100 /  5.  8.100
  libswresample   3.  8.100 /  3.  8.100
  libpostproc55.  8.100 / 55.  8.100
[h264 @ 01967ec1ec00] non-existing PPS 0 referenced
Last message repeated 1 times
[h264 @ 01967ec1ec00] decode_slice_header error
[h264 @ 01967ec1ec00] no frame!
[h264 @ 01967ec1ec00] non-existing PPS 0 referenced
Last message repeated 1 times
[h264 @ 01967ec1ec00] decode_slice_header error
[h264 @ 01967ec1ec00] no frame!
[mpegts @ 01967ebfed00] start time for stream 0 is not set in
estimate_timings_from_pts
[mpegts @ 01967ebfed00] start time for stream 1 is not set in
estimate_timings_from_pts
[mpegts @ 01967ebfed00] Packet corrupt (stream = 0, dts = 590952620).
[mpegts @ 01967ebfed00] Packet corrupt (stream = 0, dts = 590952620).
[mpegts @ 01967ebfed00] Packet corrupt (stream = 0, dts = 590952620).
[mpegts @ 01967ebfed00] stream 0 : no TS found at start of file,
duration not set
[mpegts @ 01967ebfed00] stream 1 : no TS found at start of file,
duration not set
Input #0, mpegts, from 'soc_udp_rx_02.ts':
  Duration: N/A, bitrate: N/A
  Program 1
Stream #0:0[0x1100]: Video: h264 (Main) ([27][0][0][0] / 0x001B),
yuv420p(tv, progressive), 1280x720 [SAR 1:1 DAR 16:9], 50 fps, 50 tbr, 90k
tbn, 100 tbc
Stream #0:1[0x1110]: Audio: aac (LC) ([15][0][0][0] / 0x000F), 48000
Hz, stereo, fltp, 128 kb/s
File 'testaudio.mp4' already exists. Overwrite? [y/N] y
Output #0, mp4, to 'testaudio.mp4':
  Metadata:
title   : media source exentions
encoder : Lavf58.47.100
Stream #0:0: Video: h264 (Main) (avc1 / 0x31637661), yuv420p(tv,
progressive), 1280x720 [SAR 1:1 DAR 16:9], q=2-31, 50 fps, 50 tbr, 90k tbn,
90k tbc
Stream #0:1: Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo,
fltp, 128 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #0:1 -> #0:1 (copy)
Press [q] to stop, [?] for help
[mpegts @ 01967ebfed00] Packet corrupt (stream = 0, dts = 590952620).
frame=  902 fps=0.0 q=-1.0 Lsize=1803kB time=00:00:18.09 bitrate=
816.3kbits/s speed= 816x
video:1469kB audio:277kB subtitle:0kB other streams:0kB global headers:0kB
muxing overhead: 3.293837%

For although there it says aac (LC) if I run mp4info on the resultant file
I get this:

mp4info testaudio.mp4
File:
  major brand:  iso5
  minor version:200
  compatible brand: iso5
  compatible brand: iso6
  compatible brand: mp41
  fast start:   yes

Movie:
  duration:   0 ms
  time scale: 1000
  fragments:  yes

Found 2 Tracks
Track 1:
  flags:3 ENABLED IN-MOVIE
  id:   1
  type: Video
  duration: 0 ms
  language: und
  media:
sample count: 0
timescale:9
duration: 0 (media timescale units)
duration: 0 (ms)
bitrate (computed): 664.930 Kbps
sample count with fragments: 902
duration with fragments: 1628337
duration with fragments: 18093 (ms)
  display width:  1280.00
  display height: 720.00
  Sample Description 0
Coding:  avc1 (H.264)
Width:   1280
Height:  720
Depth:   

Re: [FFmpeg-user] Compiling FFMpeg

[Simon Brown]

>
>
>
> 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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[FFmpeg-user] Compiling FFMpeg

[Simon Brown]

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
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[FFmpeg-user] FFmpeg on Raspberry pi

[Simon Brown]

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 CPU and GPU.  So I have tried this command:
ffmpeg -c:v h264_v4l2m2m -i Omniseq4.ts -f rawvideo dump.raw
with output:
ffmpeg version 4.1.6-1~deb10u1+rpt1 Copyright (c) 2000-2020 the FFmpeg
developers
  built with gcc 8 (Raspbian 8.3.0-6+rpi1)
  configuration: --prefix=/usr --extra-version='1~deb10u1+rpt1'
--toolchain=hardened --incdir=/usr/include/arm-linux-gnueabihf --enable-gpl
--disable-stripping --enable-avresample --disable-filter=resample
--enable-avisynth --enable-gnutls --enable-ladspa --enable-libaom
--enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca
--enable-libcdio --enable-libcodec2 --enable-libflite
--enable-libfontconfig --enable-libfreetype --enable-libfribidi
--enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame
--enable-libmysofa --enable-libopenjpeg --enable-libopenmpt
--enable-libopus --enable-libpulse --enable-librsvg --enable-librubberband
--enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex
--enable-libssh --enable-libtheora --enable-libtwolame --enable-libvidstab
--enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp
--enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq
--enable-libzvbi --enable-lv2 --enable-omx --enable-openal --enable-opengl
--enable-sdl2 --enable-omx-rpi --enable-mmal --enable-neon --enable-rpi
--enable-libdc1394 --enable-libdrm --enable-libiec61883
--enable-chromaprint --enable-frei0r --enable-libx264 --enable-shared
--libdir=/usr/lib/arm-linux-gnueabihf --cpu=arm1176jzf-s --arch=arm
  WARNING: library configuration mismatch
  avutil  configuration: --prefix=/usr --extra-version='1~deb10u1+rpt1'
--toolchain=hardened --incdir=/usr/include/arm-linux-gnueabihf --enable-gpl
--disable-stripping --enable-avresample --disable-filter=resample
--enable-avisynth --enable-gnutls --enable-ladspa --enable-libaom
--enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca
--enable-libcdio --enable-libcodec2 --enable-libflite
--enable-libfontconfig --enable-libfreetype --enable-libfribidi
--enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame
--enable-libmysofa --enable-libopenjpeg --enable-libopenmpt
--enable-libopus --enable-libpulse --enable-librsvg --enable-librubberband
--enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex
--enable-libssh --enable-libtheora --enable-libtwolame --enable-libvidstab
--enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp
--enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq
--enable-libzvbi --enable-lv2 --enable-omx --enable-openal --enable-opengl
--enable-sdl2 --enable-omx-rpi --enable-mmal --enable-neon --enable-rpi
--enable-libdc1394 --enable-libdrm --enable-libiec61883
--enable-chromaprint --enable-frei0r --enable-libx264
--libdir=/usr/lib/arm-linux-gnueabihf/neon/vfp --cpu=cortex-a7
--arch=armv6t2 --disable-thumb --enable-shared --disable-doc
--disable-programs
  avcodec configuration: --prefix=/usr --extra-version='1~deb10u1+rpt1'
--toolchain=hardened --incdir=/usr/include/arm-linux-gnueabihf --enable-gpl
--disable-stripping --enable-avresample --disable-filter=resample
--enable-avisynth --enable-gnutls --enable-ladspa --enable-libaom
--enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca
--enable-libcdio --enable-libcodec2 --enable-libflite
--enable-libfontconfig --enable-libfreetype --enable-libfribidi
--enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame
--enable-libmysofa --enable-libopenjpeg --enable-libopenmpt
--enable-libopus --enable-libpulse --enable-librsvg --enable-librubberband
--enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex
--enable-libssh --enable-libtheora --enable-libtwolame --enable-libvidstab
--enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp
--enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq
--enable-libzvbi --enable-lv2 --enable-omx --enable-openal --enable-opengl
--enable-sdl2 --enable-omx-rpi --enable-mmal --enable-neon --enable-rpi
--enable-libdc1394 --enable-libdrm --enable-libiec61883
--enable-chromaprint --enable-frei0r --enable-libx264
--libdir=/usr/lib/arm-linux-gnueabihf/neon/vfp --cpu=cortex-a7
--arch=armv6t2 --disable-thumb --enable-shared --disable-doc
--disable-programs
  avformatconfiguration: --prefix=/usr --extra-version='1~deb10u1+rpt1'
--toolchain=hardened --incdir=/usr/include/arm-linux-gnueabihf --enable-gpl
--disable-stripping --enable-avresample --disable-filter=resample
--enable-avisynth --enable-gnutls --enable-ladspa --enable-libaom
--enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca
--enable-libcdio --enable-libcodec2 

[FFmpeg-user] How do I decode once and pass frames to multiple destinations

[Simon Brown]

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 different medium and
c) scale and re-encode that video to restream out of the unit.

I can do all 3 of these things individually, and therefore I can do all 3
of these things by running 3 instances of ffmpeg, but at that point my
system is decoding the stream 3 times and I really don't want to spend the
CPU power doing that.

So is it possible to just decode the stream once, and then send it to HDMI,
scaler, etc. ?

Regards,
Simon
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Re: [FFmpeg-user] Uninstalling ffmpeg

[Simon Brown]

> 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 idiot?
> That's just too, too, too delicious.
>
> And you have to hide behind an anonymous email as well, no signature.
What are you afraid of?

Simon
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Re: [FFmpeg-user] Uninstalling ffmpeg

[Simon Brown]

> 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 everything
he writes.  But this petty sniping at his English is also ridiculous.  Have
you considered that English might not be his first language?  Have you
considered that he could be dyslexic?  I am quite happy to be grammar
police when it comes to adverts I see on Facebook, but in a 'forum' where
people are voicing opinions it's not really required.

I think this should all stop - it's like you're all a bunch of 5 year olds!

rant over.
Regards,
Simon
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[FFmpeg-user] Compiling FFMpeg with MSVC 2015 tools.

[Simon Brown]

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
--enable-shared --disable-static
configure finishes with "WARNING: pkg-config not found, library detection
may fail."
I run "make".
I get two xyz.mak:n: *** missing separator errors.
These are solved by inserting a tab between $( and eval in the relevant
files.
Then run "make" again and I get:
make: *** No rule to make target 'libavdevice/avdevice.dll', needed by
'all-yes'. Stop.

Any suggestions gratefully received.
Regards,
Simon
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[FFmpeg-user] Are pipe's slow?

[Simon Brown]

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 can only get
15 seconds of video at a time.  If I run ffmpeg on its own creating a file
on the disk, then it can run without buffer overrun errors.  Equally, if I
run myTestOnDemandRTSPServer from a file then it can play it without
issue.  It seems to be the combination of the two that causes the problem.

So is there something that can be optimised with using pipes?  Or would it
be better to change the input of myTestOnDemandRTSPServer to use a UDP
source for the stream?

Regards,

Simon
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[FFmpeg-user] Specify input options

[Simon Brown]

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).

Regards,
Simon
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Re: [FFmpeg-user] Virtual camera

[Simon Brown]

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
> > directshow as an input device.
>
> On Linux, the v4l2loopback device can be used to do this, I have had
> success with this. I have also read of akvcam, but not tested it.
>
> The latter is a project by webcamoid, which is also implemented for Mac
> and Windows. Perhaps it's worth looking what technology they use on
> those two platforms, as those were the ones you named. Under Windows
> it's probably a DirectShow filter.
>
> If you use webcamoid directly, it's no longer ffmpeg though:
> https://github.com/webcamoid/webcamoid/wiki/Virtual-camera-support
>
> Cheers,
> Moritz
>
> Thanks for all the suggestions, tried webcamoid earlier today and the lag
was enormous (15 seconds?) so not sure that's a solution.
I'll test out some of the other suggestions and go from there.  If I find a
workable solution I'll post back here.

Cheers,
Simon
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[FFmpeg-user] Virtual camera

[Simon Brown]

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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Re: [FFmpeg-user] Needed: 10 minute, p24 test video

[Simon Brown]

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 non-copyrighted material.
>
> Any ideas?
> ___
>
> I believe Sintel the full movie is over 10 minutes and is non-copyright.
I also believe it to be p24 but if it's not you could always reencode it as
such.

Regards,
Simon
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Re: [FFmpeg-user] FFMpeg and H.323

[Simon Brown]

>
>
> 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
> > this yet.
>
> I’m not surprised, H.323 covers infrastructure at a scope that is on a
> different level than ffmpeg, or any other single application for that
> matter.
>
> Since it’s not a single standard I don’t really know what to say it
> supports, but it stipulates all endpoint (terminal) equipment be capable of
> both G.711 as a minimum, and H.261 if it has video capability. Any
> additional codec support is H.245 negotiated by connecting equipment. H.264
> is commonly implemented, as well as speex (which I think you mean when you
> say opus) but neither capability is required.
>
> Can you tell us more about the situation where you need to encode AV
> streams usable in a H.323 system out of band? There isn’t really a
> “packaging” step to speak of, and If you are creating a software based
> implementation the most ffmpeg is going to be of help to you is RTP. H.323
> is more of a protocol than format.
>
> Speaking generally, I guess you could say ffmpeg can produce a stream that
> conforms to H.323, (by encoding mu-law/a-law and optionally H.261 and using
> RTP) but anything else is going to depend on (all) the equipment
> facilitating session communication.
>
> Regards,
> Ted Park
>
>
> Hi Ted,
Many thanks for your quick reply.
I thought H.323 was a packaging a bit like HLS might be, or Fragmented
MP4.  The hope is to be able to integrate a camera system generating H.264
into Zoom and other web-conferencing systems which require H.323 to work.

So what you're saying is I'd need to generate my own communications handler
that manages the H.323 traffic, and passing the H.264 stream to that
handler to pass on to the endpoint?

Cheers,
Simon
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[FFmpeg-user] FFMpeg and H.323

[Simon Brown]

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
this yet.

Any pointers gratefully received.

Regards,
Simon
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Re: [FFmpeg-user] Using FFMpeg to produce LL-HLS

[Simon Brown]

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 message as before?
> You can force the codec tag to be anything with -tag:v $TAG but I doubt
> that is relevant here.
>
> Regards,
> Ted Park
>

No, I'm not going to re-encode - I cannot afford the CPU power.
Yes, that is the only error.  I have done this:
C:\xampp\htdocs>\ffmpeg-20200216-8578433-win64-static\bin\ffmpeg.exe
-fflags igndts+nofillin -i udp://127.0.0.1:9034 -map 0 -c:v copy -an -f
h264 udp://127.0.0.1:9036
in one DOS prompt, and this:
C:\xampp\htdocs>\ffmpeg-20200216-8578433-win64-static\bin\ffmpeg.exe -i
udp://127.0.0.1:9036 -codec copy -an -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 mp4
-ldash 1 -strict experimental -f dash manifest.mpd
in the second DOS prompt.  This gives me a stream that I can view in a
browser, but latency is 8s which is not acceptable.  We need it <1s if at
all possible.

So clearly if I send an h264 stream at the dash muxer it is happy, yet if I
just use the ts with -map 0 -c:v copy -an it doesn't work and gives the
earlier error.
There are other side effects in that it starts complaining about
non-monotonous DTS but that goes away after a while:
ffmpeg version git-2020-02-16-8578433 Copyright (c) 2000-2020 the FFmpeg
developers
  built with gcc 9.2.1 (GCC) 20200122
  configuration: --enable-gpl --enable-version3 --enable-sdl2
--enable-fontconfig --enable-gnutls --enable-iconv --enable-libass
--enable-libdav1d --enable-libbluray --enable-libfreetype
--enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb
--enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy
--enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx
--enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265
--enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp
--enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc
--enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom
--enable-libmfx --enable-ffnvcodec --enable-cuvid --enable-d3d11va
--enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth
--enable-libopenmpt --enable-amf
  libavutil  56. 41.100 / 56. 41.100
  libavcodec 58. 70.100 / 58. 70.100
  libavformat58. 38.101 / 58. 38.101
  libavdevice58.  9.103 / 58.  9.103
  libavfilter 7. 76.100 /  7. 76.100
  libswscale  5.  6.100 /  5.  6.100
  libswresample   3.  6.100 /  3.  6.100
  libpostproc55.  6.100 / 55.  6.100
[mov,mp4,m4a,3gp,3g2,mj2 @ 01b04824bec0] DTS discontinuity in stream 0:
packet 3 with DTS 41, packet 4 with DTS 80
[mov,mp4,m4a,3gp,3g2,mj2 @ 01b04824bec0] DTS discontinuity in stream 0:
packet 5 with DTS 81, packet 6 with DTS 120
[mov,mp4,m4a,3gp,3g2,mj2 @ 01b04824bec0] DTS discontinuity in stream 0:
packet 7 with DTS 121, packet 8 with DTS 160
[mov,mp4,m4a,3gp,3g2,mj2 @ 01b04824bec0] DTS discontinuity in stream 0:
packet 9 with DTS 161, packet 10 with DTS 200
[mov,mp4,m4a,3gp,3g2,mj2 @ 01b04824bec0] DTS discontinuity in stream 0:
packet 11 with DTS 201, packet 12 with DTS 240
[mov,mp4,m4a,3gp,3g2,mj2 @ 01b04824bec0] DTS discontinuity in stream 0:
packet 13 with DTS 241, packet 14 with DTS 980
[mov,mp4,m4a,3gp,3g2,mj2 @ 01b04824bec0] DTS discontinuity in stream 0:
packet 15 with DTS 981, packet 16 with DTS 1020
[mov,mp4,m4a,3gp,3g2,mj2 @ 01b04824bec0] DTS discontinuity in stream 0:
packet 17 with DTS 1021, packet 18 with DTS 1060
[mov,mp4,m4a,3gp,3g2,mj2 @ 01b04824bec0] DTS discontinuity in stream 0:
packet 19 with DTS 1061, packet 20 with DTS 1100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'udp://127.0.0.1:9036?listen':
  Metadata:
major_brand : isml
minor_version   : 512
compatible_brands: ismlpiff
encoder : Lavf58.38.101
  Duration: 00:00:02.70, start: 0.00, bitrate: N/A
Stream #0:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p(tv),
1920x1080 [SAR 1:1 DAR 16:9], 1874 kb/s, 37.04 fps, 50 tbr, 1k tbn, 50
tbc (default)
Metadata:
  handler_name: VideoHandler
[dash @ 01b04942f040] Opening 'init-stream0.m4s' for writing
Output #0, dash, to 'manifest.mpd':
  Metadata:
major_brand : isml
minor_version   : 512
compatible_brands: ismlpiff
encoder : Lavf58.38.101
Stream #0:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p(tv),
1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 1874 kb/s, 37.04 fps, 50 tbr, 12800
tbn, 25 tbc (default)
Metadata:
  handler_name: VideoHandler
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
Press [q] to stop, [?] 

Re: [FFmpeg-user] Using FFMpeg to produce LL-HLS

[Simon Brown]

>
>
>> 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 mp4 -ldash 1 -f dash manifest.mpd
> ffmpeg version git-2020-02-16-8578433 Copyright (c) 2000-2020 the FFmpeg
> developers
>   built with gcc 9.2.1 (GCC) 20200122
>   configuration: --enable-gpl --enable-version3 --enable-sdl2
> --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass
> --enable-libdav1d --enable-libbluray --enable-libfreetype
> --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb
> --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy
> --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx
> --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265
> --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp
> --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc
> --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom
> --enable-libmfx --enable-ffnvcodec --enable-cuvid --enable-d3d11va
> --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth
> --enable-libopenmpt --enable-amf
>   libavutil  56. 41.100 / 56. 41.100
>   libavcodec 58. 70.100 / 58. 70.100
>   libavformat58. 38.101 / 58. 38.101
>   libavdevice58.  9.103 / 58.  9.103
>   libavfilter 7. 76.100 /  7. 76.100
>   libswscale  5.  6.100 /  5.  6.100
>   libswresample   3.  6.100 /  3.  6.100
>   libpostproc55.  6.100 / 55.  6.100
> Input #0, mpegts, from 'udp://127.0.0.1:9034':
>   Duration: N/A, start: 82854.078300, bitrate: N/A
>   Program 1
> Stream #0:0[0x1100]: Video: h264 (Main) ([27][0][0][0] / 0x001B),
> yuv420p(tv, top first), 1920x1080 [SAR 1:1 DAR 16:9], 25 fps, 16.67 tbr,
> 90k tbn, 50 tbc
> Stream #0:1[0x1110]: Audio: aac (LC) ([15][0][0][0] / 0x000F), 48000
> Hz, stereo, fltp, 192 kb/s
> [dash @ 020ab5934040] Opening 'init-stream0.m4s' for writing
> [mp4 @ 020ab4fbd040] Tag [27][0][0][0] incompatible with output codec
> id '27' (avc1)
> Could not write header for output file #0 (incorrect codec parameters ?):
> Invalid data found when processing input
> Stream mapping:
>   Stream #0:0 -> #0:0 (copy)
>   Stream #0:1 -> #0:1 (copy)
> Last message repeated 1 times
>
> If instead I try to reencode, the stream information is similar, but
> different:
> [dash @ 0272b2bec140] Opening 'init-stream0.m4s' for writing
> Output #0, dash, to 'manifest.mpd':
>   Metadata:
> encoder : Lavf58.38.101
> Stream #0:0: Video: h264 (libx264), yuv420p, 1920x1080 [SAR 1:1 DAR
> 16:9], q=-1--1, 6000 kb/s, 16.67 fps, 45k tbn, 16.67 tbc
> Metadata:
>   encoder : Lavc58.70.100 libx264
> Side data:
>   cpb: bitrate max/min/avg: 0/0/600 buffer size: 0 vbv_delay: N/A
> [
> So h264 is ok, but Main/avc1 isn't.  But libx264 is ok.  Aren't these much
> the same thing?  What do I need to do to my encoded stream to make the dash
> muxer accept it?
>
> Ok, further tests reveal that it is the TS nature of it that FFmpeg is
complaining about.  If I run two separate ffmpeg commands, the first one
re-encoding with libx264 and outputting mpegts to udp://127.0.0.1:9036 and
the next receiving from udp:9036 and converting to -f dash it throws the
same error "incompatible with output codec id", yet if the encoding goes
straight to the -f dash it's happy.  So now the question becomes:

How do I strip the h264 stream from the ts to feed to -f dash?

Cheers,
Simon

>
>
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Re: [FFmpeg-user] Using FFMpeg to produce LL-HLS

[Simon Brown]

>
>
>
> > 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:
> The issue is apparently that the dash muxer requires to know the bitrate of
> the h264 stream.
>
> Carl Eugen
> ___
>
> Thank you Carl.  I have now rejigged the command line to try and use ldash
instead of lhls, but am getting the same error, though now it's no longer
complaining about not having a bit rate because that is set.

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 mp4 -ldash 1 -f dash manifest.mpd
ffmpeg version git-2020-02-16-8578433 Copyright (c) 2000-2020 the FFmpeg
developers
  built with gcc 9.2.1 (GCC) 20200122
  configuration: --enable-gpl --enable-version3 --enable-sdl2
--enable-fontconfig --enable-gnutls --enable-iconv --enable-libass
--enable-libdav1d --enable-libbluray --enable-libfreetype
--enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb
--enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy
--enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx
--enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265
--enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp
--enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc
--enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom
--enable-libmfx --enable-ffnvcodec --enable-cuvid --enable-d3d11va
--enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth
--enable-libopenmpt --enable-amf
  libavutil  56. 41.100 / 56. 41.100
  libavcodec 58. 70.100 / 58. 70.100
  libavformat58. 38.101 / 58. 38.101
  libavdevice58.  9.103 / 58.  9.103
  libavfilter 7. 76.100 /  7. 76.100
  libswscale  5.  6.100 /  5.  6.100
  libswresample   3.  6.100 /  3.  6.100
  libpostproc55.  6.100 / 55.  6.100
Input #0, mpegts, from 'udp://127.0.0.1:9034':
  Duration: N/A, start: 82854.078300, bitrate: N/A
  Program 1
Stream #0:0[0x1100]: Video: h264 (Main) ([27][0][0][0] / 0x001B),
yuv420p(tv, top first), 1920x1080 [SAR 1:1 DAR 16:9], 25 fps, 16.67 tbr,
90k tbn, 50 tbc
Stream #0:1[0x1110]: Audio: aac (LC) ([15][0][0][0] / 0x000F), 48000
Hz, stereo, fltp, 192 kb/s
[dash @ 020ab5934040] Opening 'init-stream0.m4s' for writing
[mp4 @ 020ab4fbd040] Tag [27][0][0][0] incompatible with output codec
id '27' (avc1)
Could not write header for output file #0 (incorrect codec parameters ?):
Invalid data found when processing input
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #0:1 -> #0:1 (copy)
Last message repeated 1 times

If instead I try to reencode, the stream information is similar, but
different:
[dash @ 0272b2bec140] Opening 'init-stream0.m4s' for writing
Output #0, dash, to 'manifest.mpd':
  Metadata:
encoder : Lavf58.38.101
Stream #0:0: Video: h264 (libx264), yuv420p, 1920x1080 [SAR 1:1 DAR
16:9], q=-1--1, 6000 kb/s, 16.67 fps, 45k tbn, 16.67 tbc
Metadata:
  encoder : Lavc58.70.100 libx264
Side data:
  cpb: bitrate max/min/avg: 0/0/600 buffer size: 0 vbv_delay: N/A
[
So h264 is ok, but Main/avc1 isn't.  But libx264 is ok.  Aren't these much
the same thing?  What do I need to do to my encoded stream to make the dash
muxer accept it?

Cheers,
Simon
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Re: [FFmpeg-user] Using FFMpeg to produce LL-HLS

[Simon Brown]

>
>
> 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://
> 127.0.0.1:9034 -codec copy -bsf:a aac_adtstoasc -flags +global_header -f
> dash -seg_duration 1 -frag_duration 0.1 -window_size 5 -extra_window_size 5
> -single_file 0 -lhls 1 -hls_playlist 1 -strict experimental -streaming 1
> -adaptation_sets "id=0,streams=v id=1,streams=a" c:\\xampp\htdocs\video.m3u8
> ffmpeg version git-2020-02-16-8578433 Copyright (c) 2000-2020 the FFmpeg
> developers
>   built with gcc 9.2.1 (GCC) 20200122
>   configuration: --enable-gpl --enable-version3 --enable-sdl2
> --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass
> --enable-libdav1d --enable-libbluray --enable-libfreetype
> --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb
> --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy
> --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx
> --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265
> --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp
> --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc
> --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom
> --enable-libmfx --enable-ffnvcodec --enable-cuvid --enable-d3d11va
> --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth
> --enable-libopenmpt --enable-amf
>   libavutil  56. 41.100 / 56. 41.100
>   libavcodec 58. 70.100 / 58. 70.100
>   libavformat58. 38.101 / 58. 38.101
>   libavdevice58.  9.103 / 58.  9.103
>   libavfilter 7. 76.100 /  7. 76.100
>   libswscale  5.  6.100 /  5.  6.100
>   libswresample   3.  6.100 /  3.  6.100
>   libpostproc55.  6.100 / 55.  6.100
> [h264 @ 018f6b0d1980] non-existing PPS 0 referenced
> Last message repeated 1 times
> [h264 @ 018f6b0d1980] decode_slice_header error
> [h264 @ 018f6b0d1980] no frame!
> [h264 @ 018f6b0d1980] non-existing PPS 0 referenced
> Last message repeated 2 times
> [h264 @ 018f6b0d1980] decode_slice_header error
> [h264 @ 018f6b0d1980] no frame!
> [h264 @ 018f6b0d1980] non-existing PPS 0 referenced
> [h264 @ 018f6b0d1980] decode_slice_header error
> [h264 @ 018f6b0d1980] no frame!
> [h264 @ 018f6b0d1980] non-existing PPS 0 referenced
> Last message repeated 2 times
> [h264 @ 018f6b0d1980] decode_slice_header error
> [h264 @ 018f6b0d1980] no frame!
> [h264 @ 018f6b0d1980] non-existing PPS 0 referenced
> [h264 @ 018f6b0d1980] decode_slice_header error
> [h264 @ 018f6b0d1980] no frame!
> [h264 @ 018f6b0d1980] non-existing PPS 0 referenced
> Last message repeated 2 times
> [h264 @ 018f6b0d1980] decode_slice_header error
> [h264 @ 018f6b0d1980] no frame!
> [h264 @ 018f6b0d1980] non-existing PPS 0 referenced
> [h264 @ 018f6b0d1980] decode_slice_header error
> [h264 @ 018f6b0d1980] no frame!
> [h264 @ 018f6b0d1980] non-existing PPS 0 referenced
> Last message repeated 2 times
> [h264 @ 018f6b0d1980] decode_slice_header error
> [h264 @ 018f6b0d1980] no frame!
> [h264 @ 018f6b0d1980] non-existing PPS 0 referenced
> [h264 @ 018f6b0d1980] decode_slice_header error
> [h264 @ 018f6b0d1980] no frame!
> [h264 @ 018f6b0d1980] non-existing PPS 0 referenced
> Last message repeated 1 times
> [h264 @ 018f6b0d1980] decode_slice_header error
> [h264 @ 018f6b0d1980] no frame!
> Input #0, mpegts, from 'udp://127.0.0.1:9034':
>   Duration: N/A, start: 34730.859244, bitrate: N/A
>   Program 1
> Stream #0:0[0x1100]: Video: h264 (Main) ([27][0][0][0] / 0x001B),
> yuv420p(tv, top first), 1920x1080 [SAR 1:1 DAR 16:9], 25 fps, 25 tbr, 90k
> tbn, 50 tbc
> Stream #0:1[0x1110]: Audio: aac (LC) ([15][0][0][0] / 0x000F), 48000
> Hz, stereo, fltp, 48 kb/s
> [dash @ 018f6e3eec80] No bit rate set for stream 0
> [dash @ 018f6e3eec80] Opening 'init-stream0.m4s' for writing
> [mp4 @ 018f6b0bac40] Could not find tag for codec h264 in stream #0,
> codec not currently supported in container
> Could not write header for output file #0 (incorrect codec parameters ?):
> Invalid argument
> Stream mapping:
>   Stream #0:0 -> #0:0 (copy)
>   Stream #0:1 -> #0:1 (copy)
> Last message repeated 1 times
>
> Thanks,
> Simon
>
Ok, some progress - I have a stream that plays, but only because I'm
re-encoding it.
Is the codec tag h264 different to what libx264 produces?  Is there a
reason it can't work with h264 codec tag?
Anyway, the command line I'm now using is this:
C:\xampp\htdocs>C:\ffmpeg-20200216-8578433-win64-static\bin\ffmpeg.exe -i
udp://127.0.0.1:9034 -c:v libx264 -b:v 5000k -preset ultrafast -an -flags
+global_header -flags +cgop -g 5 -f dash -seg_duration 1 -frag_duration 0.1

Re: [FFmpeg-user] Using FFMpeg to produce LL-HLS

[Simon Brown]

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 -window_size 5 -extra_window_size 5
> > -single_file 0 -lhls 1 -hls_playlist 1 ^
> > -strict experimental -streaming 1 -adaptation_sets "id=0,streams=v
> > id=1,streams=a" c:\xampp\htdocs\video.m3u8
> >
> > Then report back.
> >
> > Btw I'd recommend re-encoding the content wherever possible. A fixed
> > GOP size combined with the -flags +cgop with an encoder such as
> > libx264 produces the best results.
> > The GOP size should ideally be a fraction of the segment length, eg 2
> > seconds. The codec neutral option -g 2 should set that for you.
>
> -g is in frames, so -g 50 for a 25 fps stream. Or you could use
> -force_key_frames
>
> Gyan
> ___
>
> 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://
127.0.0.1:9034 -codec copy -bsf:a aac_adtstoasc -flags +global_header -f
dash -seg_duration 1 -frag_duration 0.1 -window_size 5 -extra_window_size 5
-single_file 0 -lhls 1 -hls_playlist 1 -strict experimental -streaming 1
-adaptation_sets "id=0,streams=v id=1,streams=a" c:\\xampp\htdocs\video.m3u8
ffmpeg version git-2020-02-16-8578433 Copyright (c) 2000-2020 the FFmpeg
developers
  built with gcc 9.2.1 (GCC) 20200122
  configuration: --enable-gpl --enable-version3 --enable-sdl2
--enable-fontconfig --enable-gnutls --enable-iconv --enable-libass
--enable-libdav1d --enable-libbluray --enable-libfreetype
--enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb
--enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy
--enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx
--enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265
--enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp
--enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc
--enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom
--enable-libmfx --enable-ffnvcodec --enable-cuvid --enable-d3d11va
--enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth
--enable-libopenmpt --enable-amf
  libavutil  56. 41.100 / 56. 41.100
  libavcodec 58. 70.100 / 58. 70.100
  libavformat58. 38.101 / 58. 38.101
  libavdevice58.  9.103 / 58.  9.103
  libavfilter 7. 76.100 /  7. 76.100
  libswscale  5.  6.100 /  5.  6.100
  libswresample   3.  6.100 /  3.  6.100
  libpostproc55.  6.100 / 55.  6.100
[h264 @ 018f6b0d1980] non-existing PPS 0 referenced
Last message repeated 1 times
[h264 @ 018f6b0d1980] decode_slice_header error
[h264 @ 018f6b0d1980] no frame!
[h264 @ 018f6b0d1980] non-existing PPS 0 referenced
Last message repeated 2 times
[h264 @ 018f6b0d1980] decode_slice_header error
[h264 @ 018f6b0d1980] no frame!
[h264 @ 018f6b0d1980] non-existing PPS 0 referenced
[h264 @ 018f6b0d1980] decode_slice_header error
[h264 @ 018f6b0d1980] no frame!
[h264 @ 018f6b0d1980] non-existing PPS 0 referenced
Last message repeated 2 times
[h264 @ 018f6b0d1980] decode_slice_header error
[h264 @ 018f6b0d1980] no frame!
[h264 @ 018f6b0d1980] non-existing PPS 0 referenced
[h264 @ 018f6b0d1980] decode_slice_header error
[h264 @ 018f6b0d1980] no frame!
[h264 @ 018f6b0d1980] non-existing PPS 0 referenced
Last message repeated 2 times
[h264 @ 018f6b0d1980] decode_slice_header error
[h264 @ 018f6b0d1980] no frame!
[h264 @ 018f6b0d1980] non-existing PPS 0 referenced
[h264 @ 018f6b0d1980] decode_slice_header error
[h264 @ 018f6b0d1980] no frame!
[h264 @ 018f6b0d1980] non-existing PPS 0 referenced
Last message repeated 2 times
[h264 @ 018f6b0d1980] decode_slice_header error
[h264 @ 018f6b0d1980] no frame!
[h264 @ 018f6b0d1980] non-existing PPS 0 referenced
[h264 @ 018f6b0d1980] decode_slice_header error
[h264 @ 018f6b0d1980] no frame!
[h264 @ 018f6b0d1980] non-existing PPS 0 referenced
Last message repeated 1 times
[h264 @ 018f6b0d1980] decode_slice_header error
[h264 @ 018f6b0d1980] no frame!
Input #0, mpegts, from 'udp://127.0.0.1:9034':
  Duration: N/A, start: 34730.859244, bitrate: N/A
  Program 1
Stream #0:0[0x1100]: Video: h264 (Main) ([27][0][0][0] / 0x001B),
yuv420p(tv, top first), 1920x1080 [SAR 1:1 DAR 16:9], 25 fps, 25 tbr, 90k
tbn, 50 tbc
Stream #0:1[0x1110]: Audio: aac (LC) ([15][0][0][0] / 0x000F), 48000
Hz, stereo, fltp, 48 kb/s
[dash @ 018f6e3eec80] No bit rate set for stream 0
[dash @ 018f6e3eec80] Opening 'init-stream0.m4s' for 

Re: [FFmpeg-user] Using FFMpeg to produce LL-HLS

[Simon Brown]

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
> > 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 that produces it.
> >
> > Does FFMpeg support generating LL-HLS?
> >
> > Regards,
> > Simon
> >
>
> Yes, though at the moment you'll have to use the dash muxer.
>
> See ffmpeg -h muxer=dash
>
> For usage.
>
> >
> ___
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>
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> ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".


Thanks Dennis,
I've read through the options and looked at some examples and have
downloaded the latest FFMpeg build from zeranoe (nightly build).  That
produces this result:

C:\ffmpeg-20200216-8578433-win64-static\bin>ffmpeg.exe -i udp://
127.0.0.1:9034 -codec copy -f dash -seg_duration 1 -frag_duration 0.1
-window_size 5 -extra_window_size 5 -single_file 0 -lhls 1 -hls_playlist 1
-strict experimental -streaming 1 -adaptation_sets "id=0,streams=v
id=1,streams=a" c:\xampp\htdocs\video.m3u8
ffmpeg version git-2020-02-16-8578433 Copyright (c) 2000-2020 the FFmpeg
developers
  built with gcc 9.2.1 (GCC) 20200122
  configuration: --enable-gpl --enable-version3 --enable-sdl2
--enable-fontconfig --enable-gnutls --enable-iconv --enable-libass
--enable-libdav1d --enable-libbluray --enable-libfreetype
--enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb
--enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy
--enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx
--enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265
--enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp
--enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc
--enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom
--enable-libmfx --enable-ffnvcodec --enable-cuvid --enable-d3d11va
--enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth
--enable-libopenmpt --enable-amf
  libavutil  56. 41.100 / 56. 41.100
  libavcodec 58. 70.100 / 58. 70.100
  libavformat58. 38.101 / 58. 38.101
  libavdevice58.  9.103 / 58.  9.103
  libavfilter 7. 76.100 /  7. 76.100
  libswscale  5.  6.100 /  5.  6.100
  libswresample   3.  6.100 /  3.  6.100
  libpostproc55.  6.100 / 55.  6.100
Input #0, mpegts, from 'udp://127.0.0.1:9034':
  Duration: N/A, start: 68873.648600, bitrate: N/A
  Program 1
Stream #0:0[0x1100]: Video: h264 (Main) ([27][0][0][0] / 0x001B),
yuv420p(tv, progressive), 1920x1080 [SAR 1:1 DAR 16:9], 50 fps, 50 tbr, 90k
tbn, 100 tbc
Stream #0:1[0x1110]: Audio: aac (LC) ([15][0][0][0] / 0x000F), 48000
Hz, stereo, fltp, 192 kb/s
[dash @ 022f500c6a40] No bit rate set for stream 0
[dash @ 022f500c6a40] Opening 'init-stream0.m4s' for writing
[mp4 @ 022f4e21ac00] Could not find tag for codec h264 in stream #0,
codec not currently supported in container
Could not write header for output file #0 (incorrect codec parameters ?):
Invalid argument
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #0:1 -> #0:1 (copy)
Last message repeated 1 times

What am I doing wrong?  I thought h264 was supported in HLS/Dash?

Regards,
Simon
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[FFmpeg-user] Using FFMpeg to produce LL-HLS

[Simon Brown]

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 that produces it.

Does FFMpeg support generating LL-HLS?

Regards,
Simon
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[FFmpeg-user] h264_mmal decoder doesn't detect correct frame rate

[Simon Brown]

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 N-95725-g45f03cdd20 Copyright (c) 2000-2019 the FFmpeg
developers
  built with gcc 8 (Raspbian 8.3.0-6+rpi1)
  configuration: --prefix=/usr --arch=armel --target-os=linux --enable-gpl
--enable-omx --enable-omx-rpi --enable-nonfree --enable-mmal
--enable-decoder=h264_mmal --disable-decoder=h264 --enable-shared
  WARNING: library configuration mismatch
  postprocconfiguration: --arch=armel --target-os=linux --enable-gpl
--enable-mmal --enable-omx --enable-omx-rpi --enable-nonfree
--enable-decoder=h264_mmal --enable-encoder=h264_omx --enable-libx265
--enable-shared
  libavutil  56. 35.101 / 56. 35.101
  libavcodec 58. 62.100 / 58. 62.100
  libavformat58. 35.100 / 58. 35.100
  libavdevice58.  9.100 / 58.  9.100
  libavfilter 7. 66.100 /  7. 66.100
  libswscale  5.  6.100 /  5.  6.100
  libswresample   3.  6.100 /  3.  6.100
  libpostproc55.  6.100 / 55.  6.100
[h264_mmal @ 0x1100e30] non-existing PPS 0 referenced
Last message repeated 42 times
[h264_mmal @ 0x1100e30] Changing output format.
Input #0, mpegts, from 'udp://127.0.0.1:10020':
  Duration: N/A, start: 80175.851422, bitrate: N/A
  Program 1
Stream #0:0[0x1100]: Video: h264 (Main) ([27][0][0][0] / 0x001B),
yuv420p(progressive), 1280x720 [SAR 1:1 DAR 16:9], 23.98 fps, 50 tbr, 90k
tbn, 23.98 tbc
Stream #0:1[0x1110]: Audio: aac (LC) ([15][0][0][0] / 0x000F), 48000
Hz, stereo, fltp, 192 kb/s
Output #0, mpegts, to 'output.avi':
  Metadata:
encoder : Lavf58.35.100
Stream #0:0: Video: h264 (Main) ([27][0][0][0] / 0x001B),
yuv420p(progressive), 1280x720 [SAR 1:1 DAR 16:9], q=2-31, 23.98 fps, 50
tbr, 90k tbn, 90k tbc
Stream #0:1: Audio: aac (LC) ([15][0][0][0] / 0x000F), 48000 Hz,
stereo, fltp, 192 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #0:1 -> #0:1 (copy)
Press [q] to stop, [?] for help
frame=   99 fps=0.0 q=-1.0 size= 256kB time=00:00:02.85 bitrate=
733.6kbits/
frame=  125 fps=123 q=-1.0 size= 256kB time=00:00:03.37 bitrate=
622.2kbits/
frame=  150 fps= 98 q=-1.0 size= 256kB time=00:00:03.88 bitrate=
540.1kbits/
frame=  176 fps= 86 q=-1.0 size= 256kB time=00:00:04.39 bitrate=
477.2kbits/
frame=  202 fps= 79 q=-1.0 size= 256kB time=00:00:04.90 bitrate=
427.4kbits/
frame=  227 fps= 74 q=-1.0 size= 512kB time=00:00:05.41 bitrate=
774.0kbits/
frame=  253 fps= 71 q=-1.0 size= 512kB time=00:00:05.93 bitrate=
707.2kbits/
frame=  278 fps= 68 q=-1.0 size= 512kB time=00:00:06.44 bitrate=
651.0kbits/
frame=  304 fps= 66 q=-1.0 size= 512kB time=00:00:06.95 bitrate=
603.1kbits/
frame=  329 fps= 64 q=-1.0 size= 768kB time=00:00:07.46 bitrate=
842.6kbits/
frame=  354 fps= 63 q=-1.0 size= 768kB time=00:00:07.95 bitrate=
790.6kbits/
frame=  380 fps= 62 q=-1.0 size= 768kB time=00:00:08.46 bitrate=
742.8kbits/
frame=  405 fps= 61 q=-1.0 size= 768kB time=00:00:08.98 bitrate=
700.5kbits/
frame=  431 fps= 60 q=-1.0 size= 768kB time=00:00:09.47 bitrate=
663.8kbits/
frame=  455 fps= 60 q=-1.0 size=1024kB time=00:00:09.98 bitrate=
840.2kbits/
frame=  481 fps= 59 q=-1.0 size=1024kB time=00:00:10.49 bitrate=
799.2kbits/
frame=  506 fps= 59 q=-1.0 size=1024kB time=00:00:11.00 bitrate=
762.0kbits/
frame=  531 fps= 58 q=-1.0 size=1024kB time=00:00:11.49 bitrate=
729.5kbits/
frame=  557 fps= 58 q=-1.0 size=1280kB time=00:00:12.01 bitrate=
873.0kbits/
frame=  567 fps= 57 q=-1.0 Lsize=1319kB time=00:00:12.20 bitrate=
885.8kbits/s speed=1.24x
video:887kB audio:286kB subtitle:0kB other streams:0kB global headers:0kB
muxing overhead: 12.458385%

If I play the file with VLC it reports (as does every other tool) that the
frame rate is 1280x720 50fps.

Is there a way of forcing h264_mmal to pick up the frame rate correctly?

Regards,
Simon
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[FFmpeg-user] Using h264_mmal decoder on Raspberry Pi 4

[Simon Brown]

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 get "Did not get output frame from MMAL." errors.

Command and console output:
pi@raspberrypi:~ $ sudo ffmpeg -c:v h264_mmal -i udp://@:10020 -f avi -c:v
rawvideo output.avi
ffmpeg version N-95607-gb414cff630 Copyright (c) 2000-2019 the FFmpeg
developers
  built with gcc 8 (Raspbian 8.3.0-6+rpi1)
  configuration: --prefix=/usr --enable-mmal --enable-omx --enable-omx-rpi
--enable-decoder=h264_mmal --enable-encoder=h264_omx --enable-libx265
--enable-shared --enable-gpl
  libavutil  56. 35.101 / 56. 35.101
  libavcodec 58. 60.100 / 58. 60.100
  libavformat58. 33.100 / 58. 33.100
  libavdevice58.  9.100 / 58.  9.100
  libavfilter 7. 66.100 /  7. 66.100
  libswscale  5.  6.100 /  5.  6.100
  libswresample   3.  6.100 /  3.  6.100
  libpostproc55.  6.100 / 55.  6.100
[h264 @ 0x2a75b0] non-existing PPS 0 referenced
Last message repeated 1 times
[h264 @ 0x2a75b0] decode_slice_header error
[h264 @ 0x2a75b0] no frame!
[h264 @ 0x2a75b0] non-existing PPS 0 referenced
Last message repeated 1 times
[h264 @ 0x2a75b0] decode_slice_header error
[h264 @ 0x2a75b0] no frame!
[h264 @ 0x2a75b0] non-existing PPS 0 referenced
Last message repeated 1 times
[h264 @ 0x2a75b0] decode_slice_header error
[h264 @ 0x2a75b0] no frame!
[h264 @ 0x2a75b0] Missing reference picture, default is 65536
Input #0, mpegts, from 'udp://@:10020':
  Duration: N/A, start: 60833.199278, bitrate: N/A
  Program 1
Stream #0:0[0x1100]: Video: h264 (Main) ([27][0][0][0] / 0x001B),
yuv420p(tv, progressive), 1280x720 [SAR 1:1 DAR 16:9], 50 fps, 50 tbr, 90k
tbn, 100 tbc
Stream #0:1[0x1110]: Audio: aac (LC) ([15][0][0][0] / 0x000F), 48000
Hz, stereo, fltp, 192 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (h264 (h264_mmal) -> rawvideo (native))
  Stream #0:1 -> #0:1 (aac (native) -> ac3 (native))
Press [q] to stop, [?] for help
[h264_mmal @ 0x2c1940] MMAL error 9 on control port
[h264_mmal @ 0x2c1940] Did not get output frame from MMAL.
Error while decoding stream #0:0: Unknown error occurred
[h264_mmal @ 0x2c1940] Did not get output frame from MMAL.
Error while decoding stream #0:0: Unknown error occurred
[h264_mmal @ 0x2c1940] Did not get output frame from MMAL.
Error while decoding stream #0:0: Unknown error occurred
[h264_mmal @ 0x2c1940] Did not get output frame from MMAL.
Error while decoding stream #0:0: Unknown error occurred
[h264_mmal @ 0x2c1940] Did not get output frame from MMAL.
Error while decoding stream #0:0: Unknown error occurred
frame=0 fps=0.0 q=0.0 size=   0kB time=-577014:32:22.77 bitrate=
 -0.0kb[h264_mmal @ 0x2c1940] Did not get output frame from MMAL.
Error while decoding stream #0:0: Unknown error occurred
[h264_mmal @ 0x2c1940] Did not get output frame from MMAL.
Error while decoding stream #0:0: Unknown error occurred
[h264_mmal @ 0x2c1940] Did not get output frame from MMAL.
Error while decoding stream #0:0: Unknown error occurred
[h264_mmal @ 0x2c1940] Did not get output frame from MMAL.
Error while decoding stream #0:0: Unknown error occurred
[h264_mmal @ 0x2c1940] Did not get output frame from MMAL.
Error while decoding stream #0:0: Unknown error occurred
frame=0 fps=0.0 q=0.0 size=   0kB time=-577014:32:22.77 bitrate=
 -0.0kb[h264_mmal @ 0x2c1940] Did not get output frame from MMAL.
Error while decoding stream #0:0: Unknown error occurred
[h264_mmal @ 0x2c1940] Did not get output frame from MMAL.
Error while decoding stream #0:0: Unknown error occurred
[h264_mmal @ 0x2c1940] Did not get output frame from MMAL.
Error while decoding stream #0:0: Unknown error occurred
[h264_mmal @ 0x2c1940] Did not get output frame from MMAL.
Error while decoding stream #0:0: Unknown error occurred
[h264_mmal @ 0x2c1940] Did not get output frame from MMAL.
Error while decoding stream #0:0: Unknown error occurred
Finishing stream 0:0 without any data written to it.
Output #0, avi, to 'output.avi':
  Metadata:
ISFT: Lavf58.33.100
Stream #0:0: Video: rawvideo (I420 / 0x30323449), yuv420p, 1280x720
[SAR 1:1 DAR 16:9], q=2-31, 552960 kb/s, 50 fps, 50 tbn, 50 tbc
Metadata:
  encoder : Lavc58.60.100 rawvideo
Stream #0:1: Audio: ac3 ([0] [0][0] / 0x2000), 48000 Hz, stereo, fltp,
192 kb/s
Metadata:
  encoder : Lavc58.60.100 ac3
frame=0 fps=0.0 q=0.0 Lsize=  96kB time=00:00:00.60
bitrate=1296.2kbits/s speed=0.394x
video:0kB audio:14kB subtitle:0kB other streams:0kB global headers:0kB
muxing overhead: 575.109619%
Exiting normally, received signal 2.

Any help gratefully received,
Cheers
Simon
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To 

[FFmpeg-user] FFMpeg and WebRTC

[Simon Brown]

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,
Simon
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Re: [FFmpeg-user] OpenSSL overhead

[Simon Brown]

>
>
>
>
> 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 enough, but I'll check.
Thanks,
Simon
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[FFmpeg-user] OpenSSL overhead

[Simon Brown]

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
breakup on it, yet the ffmpeg output showed no warnings or errors about
broken macro blocks or concealing errors, etc.

My command line is:
ffmpeg -re -i udp://@:65224 -codec copy -bsf:a aac_adtstoasc -f flv -fflags
nobuffer rtmps://facebookliveurl

This was running ffmpeg as compiled from the git head last week.  The
stream coming in on the udp port is a local stream - ie no network issues.

Cheers,
Simon
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Re: [FFmpeg-user] Configure says it can't find openssl

[Simon Brown]

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 libssl.so 1.0.1.)
>
> Actually, with 1.0.1, you shouldn't be getting this far, because those
> quoted lines are the check for OPENSSL_init_ssl we see here:
>
> > /tmp/ffconf.wZNnjr13/test.o:test.c:function check_OPENSSL_init_ssl:
> > error: undefined reference to 'OPENSSL_init_ssl'
> > /tmp/ffconf.wZNnjr13/test.o:test.c:function check_OPENSSL_init_ssl:
> > error: undefined reference to 'OPENSSL_init_ssl'
> > collect2: error: ld returned 1 exit status
>
> This check should fail in compilation (OPENSSL_init_ssl not being
> defined in the headers), not in linking.
>
> The subsequent test for SSL_library_init on the other hand should
> succeed with 1.0.1, but it fails in your log (in compilation).
>
> This hints at that you, Simon, may have the openssl headers for 1.1.x,
> but the libraries for 1.0.x (or even mixed libraries). It looks a bit
> broken, sorry. Did you compile openssl yourself? Did you first install
> 1.0.x, then 1.1.x, or vice versa, and failed to clean up inbetween?
>
> Moritz
>
Hi Moritz,
Many thanks for your reply, and you are right - mixed versions of openssl.
I've cleared out all openssl1.0.1 and rebuilt and installed openssl1.1.1c
and configure now works, (but only with ffmpeg git head, not with 3.15).
I will see if I can run everything else I want with the latest git-head and
go from there.

Regards,
Simon
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Re: [FFmpeg-user] Configure says it can't find openssl

[Simon Brown]

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-decoder=h264
> > --enable-decoder=vc1 --enable-decoder=aac --disable-ffplay
> > --disable-ffprobe --enable-openssl
> >
> > It's on an embedded system, hence disabling most of the features.
> > It returns with
> > ERROR: openssl not found
> >
> > If you think configure made a mistake, make sure you are using the latest
> > version from Git.  If the latest version fails, report the problem to the
> > ffmpeg-user@ffmpeg.org mailing list or IRC #ffmpeg on irc.freenode.net.
> > Include the log file "ffbuild/config.log" produced by configure as this
> > will help
> > solve the problem.
> >
> > The log file is attached.
> >
> > I have made sure that openssl is installed
> > opkg list-installed | grep openssl
> > openssl - 1.0.1p-r0.0
> > openssl-conf - 1.0.1p-r0.0
> > openssl-dbg - 1.0.1p-r0.0
> > openssl-dev - 1.0.1p-r0.0
>
> you don't say which version of ffmpeg you try to build and your openssl
> as well as your compiler are old
>
> BEGIN /tmp/ffconf.wZNnjr13/test.c
> 1   #include 
> 2   #include 
> 3   long check_OPENSSL_init_ssl(void) { return (long)
> OPENSSL_init_ssl; }
> 4   int main(void) { int ret = 0;
> 5ret |= ((intptr_t)check_OPENSSL_init_ssl) & 0x;
> 6   return ret; }
> END /tmp/ffconf.wZNnjr13/test.c
> gcc -D_ISOC99_SOURCE -D_FILE_OFFSET_BITS=64 -D_LARGEFILE_SOURCE
> -D_POSIX_C_SOURCE=200112 -D_XOPEN_SOURCE=600 -march=armv7-a -std=c11
> -fomit-frame-pointer -marm -pthread -c -o /tmp/ffconf.wZNnjr13/test.o
> /tmp/ffconf.wZNnjr13/test.c
> gcc -march=armv7-a -Wl,--as-needed -Wl,-z,noexecstack -o
> /tmp/ffconf.wZNnjr13/test /tmp/ffconf.wZNnjr13/test.o -lssl -lcrypto
> /usr/lib/gcc/arm-angstrom-linux-gnueabi/4.9.3/../../../libssl.so: error:
> undefined reference to 'sk_free'
> /usr/lib/gcc/arm-angstrom-linux-gnueabi/4.9.3/../../../libssl.so: error:
> undefined reference to 'sk_push'
> /usr/lib/gcc/arm-angstrom-linux-gnueabi/4.9.3/../../../libssl.so: error:
> undefined reference to 'sk_new_null'
> /usr/lib/gcc/arm-angstrom-linux-gnueabi/4.9.3/../../../libssl.so: error:
> undefined reference to 'sk_delete'
> /usr/lib/gcc/arm-angstrom-linux-gnueabi/4.9.3/../../../libssl.so: error:
> undefined reference to 'sk_num'
> /usr/lib/gcc/arm-angstrom-linux-gnueabi/4.9.3/../../../libssl.so: error:
> undefined reference to 'sk_value'
> /usr/lib/gcc/arm-angstrom-linux-gnueabi/4.9.3/../../../libssl.so: error:
> undefined reference to 'sk_find'
> /usr/lib/gcc/arm-angstrom-linux-gnueabi/4.9.3/../../../libssl.so: error:
> undefined reference to 'sk_dup'
> /usr/lib/gcc/arm-angstrom-linux-gnueabi/4.9.3/../../../libssl.so: error:
> undefined reference to 'EVP_MD_CTX_init'
> /usr/lib/gcc/arm-angstrom-linux-gnueabi/4.9.3/../../../libssl.so: error:
> undefined reference to 'CRYPTO_add_lock'
> /usr/lib/gcc/arm-angstrom-linux-gnueabi/4.9.3/../../../libssl.so: error:
> undefined reference to 'EVP_MD_CTX_cleanup'
> /usr/lib/gcc/arm-angstrom-linux-gnueabi/4.9.3/../../../libssl.so: error:
> undefined reference to 'sk_set_cmp_func'
> /usr/lib/gcc/arm-angstrom-linux-gnueabi/4.9.3/../../../libssl.so: error:
> undefined reference to 'OpenSSLDie'
> /usr/lib/gcc/arm-angstrom-linux-gnueabi/4.9.3/../../../libssl.so: error:
> undefined reference to 'EVP_CIPHER_CTX_init'
> /usr/lib/gcc/arm-angstrom-linux-gnueabi/4.9.3/../../../libssl.so: error:
> undefined reference to 'BUF_strdup'
> /usr/lib/gcc/arm-angstrom-linux-gnueabi/4.9.3/../../../libssl.so: error:
> undefined reference to 'BUF_strndup'
> /usr/lib/gcc/arm-angstrom-linux-gnueabi/4.9.3/../../../libssl.so: error:
> undefined reference to 'HMAC_CTX_init'
> /usr/lib/gcc/arm-angstrom-linux-gnueabi/4.9.3/../../../libssl.so: error:
> undefined reference to 'sk_pop_free'
> /usr/lib/gcc/arm-angstrom-linux-gnueabi/4.9.3/../../../libssl.so: error:
> undefined reference to 'sk_shift'
> /usr/lib/gcc/arm-angstrom-linux-gnueabi/4.9.3/../../../libssl.so: error:
> undefined reference to 'EVP_CIPHER_CTX_cleanup'
> /usr/lib/gcc/arm-angstrom-linux-gnueabi/4.9.3/../../../libssl.so: error:
> undefined reference to 'HMAC_CTX_cleanup'
> /usr/lib/gcc/arm-angstrom-linux-gnueabi/4.9.3/../../../libssl.so: error:
> undefined reference to 'sk_new'
> /usr/lib/gcc/arm-angstrom-linux-gnueabi/4.9.3/../../../libssl.so: error:
> undefined reference to 'EVP_MD_CTX_create'
> /usr/lib/gcc/arm-angstrom-linux-gnueabi/4.9.3/../../../libssl.so: error:
> undefined re

Re: [FFmpeg-user] source for audio filter sofalizer

[Simon Brown]

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 needed seems to be down? I only get a
>
>
>
> Forbidden
>
> You don't have permission to access / on this server.
> Apache/2.4.29 (Ubuntu) Server at www.sofacoustics.org Port 80
>
>
>
> Are there recommended alternative resources or does the documentation
> needs an update?
>
> Kind regards
> Bernd B.
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Is this more what you're looking for:
https://github.com/sofacoustics
?

Simon
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[FFmpeg-user] Control start and stop time precisely

[Simon Brown]

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
to an RTMP stream.  This works well.

However, by the time the stream hits YouTube it's a couple of seconds
(variable) delayed from when it left the camera - and I believe this is
down to FFmpeg buffering.  This means that when the user, looking at the
camera picture, wants to start and stop the stream isn't when the stream
will actually start and stop on YouTube.  Is there any way of a) reducing
the buffering within FFmpeg or b) controlling the buffering so that the
delay from the incoming live stream is predictable?

The only condition I have is that we cannot re-encode the stream as that
will take too much CPU time.

Current command line is:
ffmpeg -i udp://@xyz -codec copy -bsf:a aac_adtstoasc -f flv
rtmp://youtube.url

Many thanks,
Simon
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Re: [FFmpeg-user] No audio while plying ffplay

[Simon Brown]

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*
>
> https://postimg.cc/image/y2mo4c0ej/
>
> please look
>
>
>
> --
> Sent from: http://www.ffmpeg-archive.org/
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Re: [FFmpeg-user] Bitrate won't change

[Simon Brown]

>
> 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 yuv420p+
> -c:v copy -c:a aac -strict experimental -b:a 128k  -f flv rtmp://
> a.rtmp.youtube.com/live2/my_youtube_key
>
>
> Unless I'm much mistaken you have -c:v copy which implies it's not going
to change the video bit rate at all - it's just copying the video stream.
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Re: [FFmpeg-user] ffmpeg reported fps different than actual file's

[Simon Brown]

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':
>>   Duration: N/A, start: 7972.129344, bitrate: N/A
>>   Program 1
>> Stream #0:0[0x6e]: Video: h264 (High) ([27][0][0][0] / 0x001B),
>> yuv420p(tv, bt709, top first), 1920x1080 [SAR 1:1 DAR 16:9], 25 fps, 25
>> tbr, 90k tbn, 50 tbc
>> Stream #0:1[0x78]: Audio: mp2 ([3][0][0][0] / 0x0003), 48000 Hz,
>> stereo, s16p, 128 kb/s
>> Stream #0:2[0x82]: Audio: mp3 ([3][0][0][0] / 0x0003), 0 channels,
>> s16p
>>
> 25 fps,25 tbr.
> So my question is,is this correct?
> Thanks!
>
> My guess is that VLC is misreporting it, and it's 50 fields per second,
interlaced, and so FFMpeg is reporting it as 25 frames per second correctly.

I have seen something similar myself between the two.
Regards,
Simon
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Re: [FFmpeg-user] Screen Capture - Windows 10 - making progress

[Simon Brown]

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.



-Original Message-
From: ffmpeg-user [mailto:ffmpeg-user-boun...@ffmpeg.org] On Behalf Of
Moritz Barsnick
Sent: Friday, June 2, 2017 12:24 PM
To: FFmpeg user discussions 
Subject: Re: [FFmpeg-user] Screen Capture - Windows 10 - making progress

On Fri, Jun 02, 2017 at 11:06:32 -0400, Ron Barnes wrote:
> Second... The captured video seems to be the full browser including
> the headers, not just the video inside the TAB. Would anyone know of a
> way to just capture the video inside the TAB?

Recording a playing vdeo from screen is something I would only ever do as a
last resort. (Are you circumventing DRM? ;-)) Have you ever had a look at
youtube-dl? Or if it's a live stream: Tried getting hold of the stream URL?
(youtube-dl isn't successful with all sites though. And I wouldn't touch
Fox "News" with a 10 foot pole. :-P)

Moritz
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[FFmpeg-user] Transmitting Sliced-I frames with ffmpeg

[Simon Brown]

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
transmits audio when there are sliced I frames in the incoming stream.  Is
there any way of persuading it to accept sliced I frames?

I know I'm using an old version of ffmpeg, but it's built for an embedded
processor, and I've used different options and modified some examples for a
different application.  The result is that I REALLY don't want to have to
rebuild with the latest code, but if someone can assure me that the latest
build DOES support sliced I frame I will do that - I don't want to take a
gamble though.

 ~/ffmpeg/ffmpeg -i udp://127.0.0.1:65111 -bsf:v h264_mp4toannexb -hls_time
0.4 -hls_list_size 20 -hls_flags delete_segments+split_by_time -codec copy
browser.m3u8
ffmpeg version N-81696-gd38dff8e Copyright (c) 2000-2016 the FFmpeg
developers
  built with gcc 4.9.3 (Linaro GCC 4.9-2014.11) 20141031 (prerelease)
  configuration: --disable-decoders --enable-decoder=h264
--enable-decoder=vc1 --enable-decoder=aac --disable-ffplay
--disable-ffprobe --disable-ffserver --enable-neon
  libavutil  55. 29.100 / 55. 29.100
  libavcodec 57. 57.100 / 57. 57.100
  libavformat57. 49.100 / 57. 49.100
  libavdevice57.  0.102 / 57.  0.102
  libavfilter 6. 62.100 /  6. 62.100
  libswscale  4.  1.100 /  4.  1.100
  libswresample   2.  1.100 /  2.  1.100
[h264 @ 0xbc4a30] non-existing PPS 0 referenced
   Last message repeated 1 times
[h264 @ 0xbc4a30] decode_slice_header error
[h264 @ 0xbc4a30] non-existing PPS 0 referenced
[h264 @ 0xbc4a30] decode_slice_header error
[h264 @ 0xbc4a30] non-existing PPS 0 referenced
[h264 @ 0xbc4a30] decode_slice_header error
[h264 @ 0xbc4a30] no frame!
[h264 @ 0xbc4a30] non-existing PPS 0 referenced
Last message repeated 1 times
[h264 @ 0xbc4a30] decode_slice_header error
[h264 @ 0xbc4a30] non-existing PPS 0 referenced
[h264 @ 0xbc4a30] decode_slice_header error
[h264 @ 0xbc4a30] non-existing PPS 0 referenced
[h264 @ 0xbc4a30] decode_slice_header error
[h264 @ 0xbc4a30] no frame!
Input #0, mpegts, from 'udp://127.0.0.1:65111':
  Duration: N/A, start: 10874.683144, bitrate: N/A
  Program 1
Metadata:
  service_name: PROGRAM 001
  service_provider: SVP NETWORK
Stream #0:0[0x1100]: Video: h264 (Main) ([27][0][0][0] / 0x001B),
yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 60 fps, 60 tbr, 90k tbn, 120 tbc
Stream #0:1[0x1110]: Audio: aac (LC) ([15][0][0][0] / 0x000F), 48000
Hz, stereo, fltp, 109 kb/s
[hls @ 0xc2f540] Using AVStream.codec to pass codec parameters to muxers is
deprecated, use AVStream.codecpar instead.
Last message repeated 1 times
Output #0, hls, to '/home/root/studio_web/myapp/public/images/browser.m3u8':
  Metadata:
encoder : Lavf57.49.100
Stream #0:0: Video: h264 (Main) ([27][0][0][0] / 0x001B), yuv420p,
1280x720 [SAR 1:1 DAR 16:9], q=2-31, 60 fps, 60 tbr, 90k tbn, 60 tbc
Stream #0:1: Audio: aac (LC) ([15][0][0][0] / 0x000F), 48000 Hz,
stereo, 109 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #0:1 -> #0:1 (copy)

So it clearly sees the H264 stream, but never copies any frames.

Cheers,
Simon
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[FFmpeg-user] Sending a UDP stream out via HLS

[Simon Brown]

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
-acodec copy sintel.m3u8

and it works just fine.
if I change it to:
ffmpeg -i udp://@65111 -f hls -hls_time 2 -hls_list_size 5 -vcodec copy
-acodec copy browser.m3u8

I get a browser.m3u8 file and browser0.ts
However, browser0.ts doesn't change to browser1.ts at any stage and just
keeps growing.  The status information as it's doing this shows speed as
1.0x (or slightly above), but I'd expect that given that the stream coming
in is in real time.  If I press 'q' to quit it then tells me that it copied
0kb of video but eg 162kb of audio.  And the resultant browser0.ts does
indeed just have audio packets in it (and some TS control packets).

What am I doing wrong?

Output from ffmpeg is:
Input #0, mpegts, from 'udp://@:65111':
  Duration: N/A, start: 16818.048833, bitrate: N/A
  Program 1
Stream #0:0[0x1100]: Video: h264 (Main), 1 reference frame
([27][0][0][0] / 0x001B), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 59.94 fps,
60 tbr, 90k tbn, 119.88 tbc
Stream #0:1[0x1110]: Audio: aac (LC) ([15][0][0][0] / 0x000F), 48000
Hz, stereo, fltp, 109 kb/s
[hls @ 0x11623c0] Using AVStream.codec to pass codec parameters to muxers
is deprecated, use AVStream.codecpar instead.
Last message repeated 1 times
[mpegts @ 0xc27c10] muxrate VBR, pcr every 5 pkts, sdt every 2147483647,
pat/pmt every 2147483647 pkts
Output #0, hls, to 'browser.m3u8':
  Metadata:
encoder : Lavf57.49.100
Stream #0:0: Video: h264 (Main), 1 reference frame ([27][0][0][0] /
0x001B), yuv420p, 1280x720 (0x0) [SAR 1:1 DAR 16:9], q=2-31, 59.94 fps, 60
tbr, 90k tbn, 59.94 tbc
Stream #0:1: Audio: aac (LC) ([15][0][0][0] / 0x000F), 48000 Hz,
stereo, 109 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #0:1 -> #0:1 (copy)
Press [q] to stop, [?] for help
[hls @ 0x11623c0] EXT-X-MEDIA-SEQUENCE:0=00:00:23.63 bitrate=N/A speed=1.27x
frame=0 fps=0.0 q=-1.0 Lsize=N/A time=00:00:23.91 bitrate=N/A
speed=1.27x
video:0kB audio:336kB subtitle:0kB other streams:0kB global headers:0kB
muxing overhead: unknown
Input file #0 (udp://@:65111):
  Input stream #0:0 (video): 1432 packets read (14115198 bytes);
  Input stream #0:1 (audio): 1121 packets read (344521 bytes);
  Total: 2553 packets (14459719 bytes) demuxed
Output file #0 (browser.m3u8):
  Output stream #0:0 (video): 0 packets muxed (0 bytes);
  Output stream #0:1 (audio): 1121 packets muxed (344521 bytes);
  Total: 1121 packets (344521 bytes) muxed

Any clues gratefully received,
Regards,
Simon
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