[FFmpeg-user] hls with subtitles vtt.m3u8 missing last segment and #EXT-X-ENLIST

2015-11-05 Thread Adrian . 

I am segmenting for hls with subtitles.
All the segments for video and for webvtt subtitles are created,
however, the vtt m3u8 file consistently omits the last segment entry
and the #EXT-X-ENLIST.

I am wondering I am doing something incorrectly.
here is my command and output.

I appreciate any feedback y'all can provide.

Adrian

a@cloudy:~/test# ffmpeg -i vid.mp4 -i sub.vtt -c copy -bsf:v 
h264_mp4toannexb  out.m3u8
ffmpeg version N-76474-g973c3db Copyright (c) 2000-2015 the FFmpeg 
developers

  built with gcc 5.2.1 (Debian 5.2.1-22) 20151010
  configuration: --enable-shared --enable-libx264 --enable-nonfree 
--enable-gpl --enable-libzvbi --enable-version3 --enable-libtheora 
--enable-libvorbis --enable-openssl --enable-libcaca --enable-libfaac 
--enable-libopus --enable-libquvi --enable-libssh --enable-libopenjpeg 
--enable-libmp3lame --enable-librtmp --enable-libvpx --enable-libwebp 
--enable-sdl

  libavutil  55.  5.100 / 55.  5.100
  libavcodec 57. 14.100 / 57. 14.100
  libavformat57. 14.100 / 57. 14.100
  libavdevice57.  0.100 / 57.  0.100
  libavfilter 6. 14.101 /  6. 14.101
  libswscale  4.  0.100 /  4.  0.100
  libswresample   2.  0.100 /  2.  0.100
  libpostproc54.  0.100 / 54.  0.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'vid.mp4':
  Metadata:
major_brand : isom
minor_version   : 512
compatible_brands: isomiso2avc1mp41
encoder : Lavf57.14.100
  Duration: 00:00:21.02, start: 0.021333, bitrate:  kb/s
Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 
1440x1080 [SAR 4:3 DAR 16:9], 2126 kb/s, 29.97 fps, 29.97 tbr, 11988 
tbn, 59.94 tbc (default)

Metadata:
  handler_name: VideoHandler
Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, 
mono, fltp, 87 kb/s (default)

Metadata:
  handler_name: SoundHandler
Input #1, webvtt, from 'sub.vtt':
  Duration: N/A, bitrate: N/A
Stream #1:0: Subtitle: webvtt
Output #0, hls, to 'out.m3u8':
  Metadata:
major_brand : isom
minor_version   : 512
compatible_brands: isomiso2avc1mp41
encoder : Lavf57.14.100
Stream #0:0(und): Video: h264 (avc1 / 0x31637661), yuv420p, 
1440x1080 [SAR 4:3 DAR 16:9], q=2-31, 2126 kb/s, 29.97 fps, 29.97 tbr, 
90k tbn, 29.97 tbc (default)

Metadata:
  handler_name: VideoHandler
Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 48000 Hz, mono, 
87 kb/s (default)

Metadata:
  handler_name: SoundHandler
Stream #0:2: Subtitle: webvtt
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #0:1 -> #0:1 (copy)
  Stream #1:0 -> #0:2 (copy)
Press [q] to stop, [?] for help
frame=  630 fps=0.0 q=-1.0 Lsize=N/A time=00:00:21.03 bitrate=N/A
video:5456kB audio:224kB subtitle:0kB other streams:0kB global 
headers:0kB muxing overhead: unknown


a@cloudy:~/test# ls
out0.ts  out0.vtt  out1.ts  out1.vtt  out2.ts  out2.vtt  out.m3u8 
out_vtt.m3u8  sub.vtt  vid.mp4


a@cloudy:~/test# tail *.m3u8
==> out.m3u8 <==
#EXT-X-VERSION:3
#EXT-X-TARGETDURATION:9
#EXT-X-MEDIA-SEQUENCE:0
#EXTINF:8.341678,
out0.ts
#EXTINF:8.341667,
out1.ts
#EXTINF:4.270944,
out2.ts
#EXT-X-ENLIST

==> out_vtt.m3u8 <==
#EXTM3U
#EXT-X-VERSION:3
#EXT-X-TARGETDURATION:9
#EXT-X-MEDIA-SEQUENCE:0
#EXTINF:8.341678,
out0.vtt
#EXTINF:8.341667,
out1.vtt
a@cloudy:~/test#
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Re: [FFmpeg-user] re-wrap BluRay M2TS as MP4

2015-11-06 Thread Adrian . 

ffmpeg -i m2.ts -c copy out.mp4

On 11/06/15 11:29, John Pilgrim wrote:

Hi all,
What would the correct ffmpeg invocation be to re-wrap an h264 M2TS file (from 
a non-commercial BluRay disc) into an MP4 file **without** recompressing the 
h264 data?
Similarly, what would the correct ffmpeg invocation be to re-wrap an mpeg2video 
M2TS file (from a non-commercial BluRay disc) into an MP2 file **without** 
recompressing the mpeg2video data?
Lastly, what would the correct ffmpeg invocation be to re-wrap an mpeg2 VOB 
file (from a non-commercial DVD) into an MP2 file **without** recompressing the 
mpeg2 data?
Thank you so much for your help!
John
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Re: [FFmpeg-user] Multicast to HLS, step by step

2015-10-17 Thread Adrian . 

It's my understanding that hls uses the http protocol, probably best
to drop the multicast udp, and install a webserver.

ffmpeg -i vid.mp4 output.m3u8

will segment the mp4 and even write you a m3u8 file.

I usually set the list size to 0, so that I can have all my segments in 
the one m3u8,I am a lazy, lazy man.

I also enable caching with  -hls_allow_cache 1

ffmpeg  -i vid.mp4 -hls_list_size 0 -hls_allow_cache 1  output.m3u8

What I love best about ffmpeg and hls,
is how easy it is to add subtitles.
Here, I copy the codecs to avoid long transcoding times.


ffmpeg  -i vid.264  -i vid.aac  -i web.vtt \
-hls_list_size 0 -hls_allow_cache 1 -c copy output.m3u8


Pick one of the above, and place the ts files and m3u8 in your 
webservers directory so that it can be served via http.

If you add subtitles, include the .vtt and vtt.m3u8 files as well.



On 10/17/15 11:38, Boris Krajnc wrote:

Hello,



I am network specialist but I am very curious and I would like to know how
to deal with video.

I read a lot about of ffmpeg and how strong it is. I hear that there is no
big deal to set up system to stream HLS, unfortunately I just hear..nobody
show me how to do ...



OK, I spent a lot of hours to start streaming HLS with ffmpeg but no
success. I search in tutorials and I try step by
step..http://ffmpeg.gusari.org/viewtopic.php?f=12=20 still no success



What I have and what I would like to do.I have server with Ubuntu where I
install ffmpeg,  with 2 NIC (eth0 with public IP and eth1 with multicast
stream). I have multicast stream on udp://239.0.0.10:1234 whith mpeg2 video.
As I understand I have to convert mpeg2 first  than play stream with
ffserver. I would like to stream in two diferent bit rate



Is there anybody prepare to share step by step how to do that?



Boris

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Re: [FFmpeg-user] Exactly one WebVTT stream is needed

2015-11-21 Thread adrian 

I was going to tell you that your formats were all wrong,
and that you needed h264 and aac
but I checked and hls has no limit on formats.

However, just because you can, doesn't mean you should,
hls implies h264 video, aac audio and 608 or webvtt captions.

May I ask why you would do it your way?
I am just curious.


- Reply message -
From: "Cloudclimber" 
To: 
Subject: [FFmpeg-user] Exactly one WebVTT stream is needed
Date: Fri, Nov 20, 2015 4:50 PM

Hi there,

this command works without Problems in ffmpeg 2.7.2:

|./ffmpeg -i udp://@239.100.1.1:1234 -map 0 -probesize 100
-analyzeduration 100 -c copy -copy_unknown -f hls -hls_time 10 -hls_wrap
10 -hls_list_size 10 /var/www/test1_.m3u8

but since |ffmpeg 2.8.0 i got the Output below, even with ffmpeg 2.8.2 .
Sure i could exclude subtitle Streams, but i want to have dvb_teletext
in the stream also, so this is no solution.

Anyone has an idea how to map all streams, inclusive subtitle streams
with ffmpeg 2.8.2?



OUTPUT:

root@ffmpegserver:~/ffmpeg-2.8.2-64bit-static# ./ffmpeg -i
udp://@239.100.1.1:1234  -map 0 -probesize 100
-analyzeduration 100 -c copy -copy_unknown -f hls -hls_time 10 -hls_wrap
10 -hls_list_size 10 /var/www/test1_.m3u8 -loglevel debug
ffmpeg version 2.8.2-statichttp://johnvansickle.com/ffmpeg/  Copyright
(c) 2000-2015 the FFmpeg developers
built with gcc 5.2.1 (Debian 5.2.1-23) 20151028
configuration: --enable-gpl --enable-version3 --disable-shared
--disable-debug --enable-runtime-cpudetect --enable-libmp3lame
--enable-libx264 --enable-libx265 --enable-libwebp --enable-libspeex
--enable-libvorbis --enable-libvpx --enable-libfreetype
--enable-fontconfig --enable-libxvid --enable-libopencore-amrnb
--enable-libopencore-amrwb --enable-libtheora --enable-libvo-aacenc
--enable-libvo-amrwbenc --enable-gray --enable-libopenjpeg
--enable-libopus --enable-libass --enable-gnutls --enable-libvidstab
--enable-libsoxr --enable-frei0r --enable-libfribidi
--disable-indev=sndio --disable-outdev=sndio --cc=gcc
libavutil 54. 31.100 / 54. 31.100
libavcodec 56. 60.100 / 56. 60.100
libavformat 56. 40.101 / 56. 40.101
libavdevice 56. 4.100 / 56. 4.100
libavfilter 5. 40.101 / 5. 40.101
libswscale 3. 1.101 / 3. 1.101
libswresample 1. 2.101 / 1. 2.101
libpostproc 53. 3.100 / 53. 3.100
Splitting the commandline.
Reading option '-i' ... matched as input file with argument
'udp://@239.100.1.1:1234'.
Reading option '-map' ... matched as option 'map' (set input stream
mapping) with argument '0'.
Reading option '-probesize' ... matched as AVOption 'probesize' with
argument '100'.
Reading option '-analyzeduration' ... matched as AVOption
'analyzeduration' with argument '100'.
Reading option '-c' ... matched as option 'c' (codec name) with argument
'copy'.
Reading option '-copy_unknown' ... matched as option 'copy_unknown'
(Copy unknown stream types) with argument '1'.
Reading option '-f' ... matched as option 'f' (force format) with
argument 'hls'.
Reading option '-hls_time' ... matched as AVOption 'hls_time' with
argument '10'.
Reading option '-hls_wrap' ... matched as AVOption 'hls_wrap' with
argument '10'.
Reading option '-hls_list_size' ... matched as AVOption 'hls_list_size'
with argument '10'.
Reading option '/var/www/test1_.m3u8' ... matched as output file.
Reading option '-loglevel' ... matched as option 'loglevel' (set logging
level) with argument 'debug'.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option copy_unknown (Copy unknown stream types) with argument 1.
Applying option loglevel (set logging level) with argument debug.
Successfully parsed a group of options.
Parsing a group of options: input file udp://@239.100.1.1:1234
.
Successfully parsed a group of options.
Opening an input file: udp://@239.100.1.1:1234 .
[udp @ 0x3c048c0] end receive buffer size reported is 131072
[mpegts @ 0x3c04080] Format mpegts probed with size=2048 and score=100
[mpegts @ 0x3c04080] stream=0 stream_type=3 pid=90 prog_reg_desc=
[mpegts @ 0x3c04080] stream=1 stream_type=6 pid=92 prog_reg_desc=
[mpegts @ 0x3c04080] stream=2 stream_type=2 pid=a8 prog_reg_desc=
[mpegts @ 0x3c04080] stream=3 stream_type=5 pid=585 prog_reg_desc=
[mpegts @ 0x3c04080] Before avformat_find_stream_info() pos: 0 bytes
read:215824 seeks:0
[mpegts @ 0x3c04080] parser not found for codec dvb_teletext, packets or
times may be invalid.
[mpegts @ 0x3c04080] parser not found for codec none, packets or times
may be invalid.
[mpegts @ 0x3c04080] parser not found for codec dvb_teletext, packets or
times may be invalid.
[mpeg2video @ 0x3c2a080] Invalid frame dimensions 0x0.
Last message repeated 4 times
[mpegts @ 0x3c04080] max_analyze_duration 500 reached at 500
microseconds st:2
[mpegts @ 0x3c04080] Could not find codec parameters for stream 3
(Unknown: none ([5][0][0][0] / 0x0005)): unknown codec
Consider increasing the value for the 'analyzeduration' and 'probesize'
options
[mpegts @

Re: [FFmpeg-user] Playback of mp4 file created by concatenating 9 mp4 files

2019-09-13 Thread Adrian 
On September 13, 2019 1:46:41 PM EDT, JD  wrote:
>The resulting movie will not play.
>To wit:
>$ ffplay Cremation_History.mp4
>ffplay version 3.4.6 Copyright (c) 2003-2019 the FFmpeg developers
>   built with gcc 4.8.5 (GCC) 20150623 (Red Hat 4.8.5-36)
>   configuration: --prefix=/usr --bindir=/usr/bin 
>--datadir=/usr/share/ffmpeg --docdir=/usr/share/doc/ffmpeg 
>--incdir=/usr/include/ffmpeg --libdir=/usr/lib64
>--mandir=/usr/share/man 
>--arch=x86_64 --optflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 
>-fexceptions -fstack-protector-strong --param=ssp-buffer-size=4 
>-grecord-gcc-switches -m64 -mtune=generic'
>--extra-ldflags='-Wl,-z,relro 
>' --extra-cflags=' ' --enable-libopencore-amrnb 
>--enable-libopencore-amrwb --enable-libvo-amrwbenc --enable-version3 
>--enable-bzlib --disable-crystalhd --enable-fontconfig --enable-gcrypt 
>--enable-gnutls --enable-ladspa --enable-libass --enable-libbluray 
>--enable-libcdio --enable-libdrm --enable-indev=jack 
>--enable-libfreetype --enable-libfribidi --enable-libgsm 
>--enable-libmp3lame --enable-nvenc --enable-openal --enable-opencl 
>--enable-opengl --enable-libopenjpeg --enable-libopus 
>--disable-encoder=libopus --enable-libpulse --enable-librsvg 
>--enable-libsoxr --enable-libspeex --enable-libtheora
>--enable-libvorbis 
>--enable-libv4l2 --enable-libvidstab --enable-libx264 --enable-libx265 
>--enable-libxvid --enable-libzvbi --enable-avfilter --enable-avresample
>
>--enable-postproc --enable-pthreads --disable-static --enable-shared 
>--enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 
>--enable-libmfx --enable-runtime-cpudetect
>   libavutil  55. 78.100 / 55. 78.100
>   libavcodec 57.107.100 / 57.107.100
>   libavformat    57. 83.100 / 57. 83.100
>   libavdevice    57. 10.100 / 57. 10.100
>   libavfilter 6.107.100 /  6.107.100
>   libavresample   3.  7.  0 /  3.  7.  0
>   libswscale  4.  8.100 /  4.  8.100
>   libswresample   2.  9.100 /  2.  9.100
>   libpostproc    54.  7.100 / 54.  7.100
>[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f30cc000920] moov atom not found0/0
>Cremation_History.mp4: Invalid data found when processing input
>
>=
>This is how the concatenation was done:
>
>$ /bin/ffmpeg -f concat -i flist  -s 1920x1080 -vf 
>scale=1920x1080,setdar=16/9,setdar=16/9 -b:a 320k -b:v 4000k 
>Cremation_History.mp4
>ffmpeg version 3.4.6 Copyright (c) 2000-2019 the FFmpeg developers
>   built with gcc 4.8.5 (GCC) 20150623 (Red Hat 4.8.5-36)
>   configuration: --prefix=/usr --bindir=/usr/bin 
>--datadir=/usr/share/ffmpeg --docdir=/usr/share/doc/ffmpeg 
>--incdir=/usr/include/ffmpeg --libdir=/usr/lib64
>--mandir=/usr/share/man 
>--arch=x86_64 --optflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 
>-fexceptions -fstack-protector-strong --param=ssp-buffer-size=4 
>-grecord-gcc-switches -m64 -mtune=generic'
>--extra-ldflags='-Wl,-z,relro 
>' --extra-cflags=' ' --enable-libopencore-amrnb 
>--enable-libopencore-amrwb --enable-libvo-amrwbenc --enable-version3 
>--enable-bzlib --disable-crystalhd --enable-fontconfig --enable-gcrypt 
>--enable-gnutls --enable-ladspa --enable-libass --enable-libbluray 
>--enable-libcdio --enable-libdrm --enable-indev=jack 
>--enable-libfreetype --enable-libfribidi --enable-libgsm 
>--enable-libmp3lame --enable-nvenc --enable-openal --enable-opencl 
>--enable-opengl --enable-libopenjpeg --enable-libopus 
>--disable-encoder=libopus --enable-libpulse --enable-librsvg 
>--enable-libsoxr --enable-libspeex --enable-libtheora
>--enable-libvorbis 
>--enable-libv4l2 --enable-libvidstab --enable-libx264 --enable-libx265 
>--enable-libxvid --enable-libzvbi --enable-avfilter --enable-avresample
>
>--enable-postproc --enable-pthreads --disable-static --enable-shared 
>--enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 
>--enable-libmfx --enable-runtime-cpudetect
>   libavutil  55. 78.100 / 55. 78.100
>   libavcodec 57.107.100 / 57.107.100
>   libavformat    57. 83.100 / 57. 83.100
>   libavdevice    57. 10.100 / 57. 10.100
>   libavfilter 6.107.100 /  6.107.100
>   libavresample   3.  7.  0 /  3.  7.  0
>   libswscale  4.  8.100 /  4.  8.100
>   libswresample   2.  9.100 /  2.  9.100
>   libpostproc    54.  7.100 / 54.  7.100
>[mov,mp4,m4a,3gp,3g2,mj2 @ 0x1859a80] Auto-inserting h264_mp4toannexb 
>bitstream filter
>Input #0, concat, from 'flist':
>   Duration: N/A, start: 0.00, bitrate: 402 kb/s
>     Stream #0:0(und): Video: h264 (Constrained Baseline) (avc1 / 
>0x31637661), yuv420p(tv, bt709), 640x360 [SAR 1:1 DAR 16:9], 306 kb/s, 
>30 fps, 30 tbr, 15360 tbn, 60 tbc
>     Metadata:
>   creation_time   : 2019-08-29T19:28:20.00Z
>   handler_name    :
>     Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, 
>stereo, fltp, 96 kb/s
>     Metadata:
>   creation_time   : 2019-08-29T19:28:20.00Z
>   handler_name    :
>File 'Cremation_History.mp4' already exists. Overwrite ? [y/N] y
>Stream

Re: [FFmpeg-user] Can't write packet with unknown timestamp

2020-04-24 Thread Adrian 
They answered your question,  It is time to let it go.

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[FFmpeg-user] AV Foundation screen capture FPS

2014-11-04 Thread Adrian Perez 
I’ve had no success trying to screen record past 15 FPS.  Here’s my thread with 
the details: http://ffmpeg.gusari.org/viewtopic.php?f=11t=1783 
http://ffmpeg.gusari.org/viewtopic.php?f=11t=1783

I’d greatly appreciate any help with this, thanks.
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[FFmpeg-user] dvd2concat error with loopback mounted ISO

2014-12-03 Thread Adrian Meyer 
I am mounting an ISO through loopback and I get 5 messages from libdvdread 
through lsdvd and 1 from perl crashing on the eval:
libdvdread: Attempting to use device /dev/loop0 mounted on 
/media/raid/Video/divx/iso for CSS authentication
libdvdread: No VTS_TMAPT available - skipping.
libdvdread: No VTS_TMAPT available - skipping.
libdvdread: No VTS_TMAPT available - skipping.
Couldn't read enough bytes for title.
Bad name after device' at (eval 5) line 2.

I am not a perl programmer. How would I manage to ignore STDERR in this section 
of dvd2concat to for lsdvd only if it was succesfull? The -q on lsdvd does not 
suppress the output:
my $lsdvd = do {
  open( my $l, -|, lsdvd, -Op, -x, $path )
    or die You need to install lsdvd for this script to work.\n$lsdvd_message;
   local $/;
  $l;
};
my %lsdvd = eval $lsdvd;
die $@ if $@;
Thanks!

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Re: [FFmpeg-user] dvd2concat error with loopback mounted ISO

2014-12-04 Thread Adrian Meyer 
 = content'Undefined',
  %s = streamid'0x22',
    },
    {
  %s = ix4,
  %s = langcode'xx',
  %s = language'Unknown',
  %s = content'Undefined',
  %s = streamid'0x23',
    },
    {
  %s = ix5,
  %s = langcode'xx',
  %s = language'Unknown',
  %s = content'Undefined',
  %s = streamid'0x24',
    },
    {
  %s = ix6,
  %s = langcode'xx',
  %s = language'Unknown',
  %s = content'Undefined',
  %s = streamid'0x25',
    },
    {
  %s = ix7,
  %s = langcode'xx',
  %s = language'Unknown',
  %s = content'Undefined',
  %s = streamid'0x26',
    },
    {
  %s = ix8,
  %s = langcode'xx',
  %s = language'Unknown',
  %s = content'Undefined',
  %s = streamid'0x27',
    },
    {
  %s = ix9,
  %s = langcode'xx',
  %s = language'Unknown',
  %s = content'Undefined',
  %s = streamid'0x28',
    },
    {
  %s = ix10,
  %s = langcode'xx',
  %s = language'Unknown',
  %s = content'Undefined',
  %s = streamid'0x29',
    },
    {
  %s = ix11,
  %s = langcode'xx',
  %s = language'Unknown',
  %s = content'Undefined',
  %s = streamid'0x2a',
    },
    {
  %s = ix12,
  %s = langcode'xx',
  %s = language'Unknown',
  %s = content'Undefined',
  %s = streamid'0x2b',
    },
  ],
    },
  ],
  %s = longest_track1,
); 


  From: Nicolas George geo...@nsup.org
 To: FFmpeg user questions ffmpeg-user@ffmpeg.org 
 Sent: Wednesday, December 3, 2014 3:21 PM
 Subject: Re: [FFmpeg-user] dvd2concat error with loopback mounted ISO
   
Le tridi 13 frimaire, an CCXXIII, Adrian Meyer a écrit :
 I am mounting an ISO through loopback and I get 5 messages from libdvdread 
 through lsdvd and 1 from perl crashing on the eval:
 libdvdread: Attempting to use device /dev/loop0 mounted on 
 /media/raid/Video/divx/iso for CSS authentication
 libdvdread: No VTS_TMAPT available - skipping.
 libdvdread: No VTS_TMAPT available - skipping.
 libdvdread: No VTS_TMAPT available - skipping.
 Couldn't read enough bytes for title.
 Bad name after device' at (eval 5) line 2.

Can you show the output of lsdvd -Op -x /dev/loop0? Please make sure you
distinguish stdout and stderr, for example by redirecting each in a separate
file ( log_stdout 2 log_stderr)?



 I am not a perl programmer. How would I manage to ignore STDERR in this
 section of dvd2concat to for lsdvd only if it was succesfull? The -q on
 lsdvd does not suppress the output:

There is no need to ignore stderr, it is already written separately. You
would not see it on the terminal otherwise.

What you report looks like a bug in lsdvd (its output in -Op mode should be
valid perl under any circumstance), but I would need to see the output you
get, because I can not reproduce the problem with my version and the ISO
file I have laying around.

Regards,

-- 
  Nicolas George
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[FFmpeg-user] delay problems when using ffserver, ffmpeg for streaming of web cam and vlc for viewing.

2015-04-29 Thread Adrian Schwartz 
hello ffmpeg user group,
I'm having delay problem ( when using ffserver and ffmpeg)
I'm getting the error ffmpeg past duration too large when using the
ffmpeg for streaming.
 VLC is configured to work in minimum delay and when streaming from //
192.168.1.1:8090/test1.asf it shows 2 sec delay, when the communication
interface delay is aroung 15 msec.


Is there some recommendation on how to lower the delay?
Regards
Adrian

#
#Streaming command:
#

ffmpeg -f dshow  -rtbufsize 1M   -i video=Logitech
Webca:audio=Microphone (2- Webcam C160) -fflags nobuffer -tune
zerolatency http://192.168.1.1:8090/feed1.ffm


##ffserver config:



Port 8090
BindAddress 0.0.0.0
MaxClients 1000
MaxBandwidth 3
CustomLog /var/log/ffserver.log
NoDaemon

Feed feed1.ffm
File /tmp/feed1.ffm
FileMaxSize 1M
ACL allow 192.168.26.0 192.168.29.255
ACL allow 127.0.0.1
ACL allow 10.0.0.0 10.0.0.255
/Feed



Stream test1.asf
Feed feed1.ffm
Format asf
VideoFrameRate 15
VideoSize 352x240
VideoBitRate 256
VideoGopSize 30
StartSendOnKey
NoAudio
/Stream

Stream stat.html
Format status
ACL allow localhost
ACL allow 192.168.0.0 192.168.255.255

/Stream
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Re: [FFmpeg-user] delay problems when using ffserver, ffmpeg for streaming of web cam and vlc for viewing.

2015-04-30 Thread Adrian Schwartz 
I forgot to mention, that over Ethernet there is still a ~2 sec delay?

On Thu, Apr 30, 2015 at 9:00 AM, Adrian Schwartz schwart...@gmail.com
wrote:

 Hi Roger
 When I changed the network interface from wireless to Ethernet, I stopped
 seeing the past duration to large error.
 I do see dup=[number] at the end and drop=0, what does it means?


 On Wed, Apr 29, 2015 at 9:24 PM, Roger Pack rogerdpa...@gmail.com wrote:

 Do you get past duration too large when just saving to file?

 On 4/29/15, Adrian Schwartz schwart...@gmail.com wrote:
  hello ffmpeg user group,
  I'm having delay problem ( when using ffserver and ffmpeg)
  I'm getting the error ffmpeg past duration too large when using the
  ffmpeg for streaming.
   VLC is configured to work in minimum delay and when streaming from //
  192.168.1.1:8090/test1.asf it shows 2 sec delay, when the communication
  interface delay is aroung 15 msec.
 
 
  Is there some recommendation on how to lower the delay?
  Regards
  Adrian
 
  #
  #Streaming command:
  #
 
  ffmpeg -f dshow  -rtbufsize 1M   -i video=Logitech
  Webca:audio=Microphone (2- Webcam C160) -fflags nobuffer -tune
  zerolatency http://192.168.1.1:8090/feed1.ffm
 
  
  ##ffserver config:
  
 
 
  Port 8090
  BindAddress 0.0.0.0
  MaxClients 1000
  MaxBandwidth 3
  CustomLog /var/log/ffserver.log
  NoDaemon
 
  Feed feed1.ffm
  File /tmp/feed1.ffm
  FileMaxSize 1M
  ACL allow 192.168.26.0 192.168.29.255
  ACL allow 127.0.0.1
  ACL allow 10.0.0.0 10.0.0.255
  /Feed
 
 
 
  Stream test1.asf
  Feed feed1.ffm
  Format asf
  VideoFrameRate 15
  VideoSize 352x240
  VideoBitRate 256
  VideoGopSize 30
  StartSendOnKey
  NoAudio
  /Stream
 
  Stream stat.html
  Format status
  ACL allow localhost
  ACL allow 192.168.0.0 192.168.255.255
 
  /Stream
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[FFmpeg-user] ffserver crashes while retrieving a stream with missing feed

2015-06-04 Thread Adrian Schwartz 
Hello users
I'm using the ffserver (2.6.3 on ubuntu) with the configuration mentioned
below.
When the feeder is offline and I try to stream using rtp with vlc, the
server is crashing after a few play retries of the vlc player.

When I run the ffserver with loglevel 64 I saw that the before the crash
the server had indicated AVIOContext statistics , I'm not sure if thats
related.

How can I avoid this crash?

Regards
Adrian

HTTPPort 8090
RTSPPort 9990
BindAddress 0.0.0.0
MaxClients 1000
MaxBandwidth 3
CustomLog /var/log/ffserver.log

Feed feed1.ffm
File /tmp/feed1.ffm
FileMaxSize 100K
ACL allow 192.168.26.0 192.168.29.255
ACL allow 127.0.0.1
ACL allow 10.0.0.0 10.0.0.255
ACL allow 192.168.26.0 192.168.29.255
/Feed

Feed feed2.ffm
File /tmp/feed2.ffm
FileMaxSize 100K
ACL allow 192.168.26.0 192.168.29.255
ACL allow 127.0.0.1
ACL allow 10.0.0.0 10.0.0.255
ACL allow 192.168.26.0 192.168.29.255
/Feed

Feed feed3.ffm
File /tmp/feed3.ffm
FileMaxSize 100K
ACL allow 192.168.26.0 192.168.29.255
ACL allow 127.0.0.1
ACL allow 10.0.0.0 10.0.0.255
ACL allow 192.168.26.0 192.168.29.255
/Feed

Feed feed4.ffm
File /tmp/feed4.ffm
FileMaxSize 100K
ACL allow 192.168.26.0 192.168.29.255
ACL allow 127.0.0.1
ACL allow 10.0.0.0 10.0.0.255
ACL allow 192.168.26.0 192.168.29.255
/Feed


Stream live1.mp4
Feed feed1.ffm
Format rtp
#BitExact
#DctFastint
IdctSimple
VideoFrameRate 30
VideoSize 320x240
VideoBitRate 256
VideoGopSize 10
NoAudio

#StartSendOnKey
#MaxTime 100
/Stream


Stream live2.mp4
Feed feed2.ffm
Format rtp
#BitExact
#DctFastint
IdctSimple
VideoFrameRate 30
VideoSize 320x240
VideoBitRate 256
VideoGopSize 10
NoAudio

#StartSendOnKey
#MaxTime 100
/Stream

Stream live3.mp4
Feed feed3.ffm
Format rtp
#BitExact
#DctFastint
IdctSimple
VideoFrameRate 30
VideoSize 320x240
VideoBitRate 256
VideoGopSize 10
NoAudio

#StartSendOnKey
#MaxTime 100
/Stream

Stream live4.mp4
Feed feed4.ffm
Format rtp
#BitExact
#DctFastint
IdctSimple
VideoFrameRate 30
VideoSize 320x240
VideoBitRate 256
VideoGopSize 10
NoAudio

#StartSendOnKey
#MaxTime 100
/Stream

#Stream stat.html
#Format status

# Only allow local people to get to the status
#ACL allow localhost
#ACL allow 192.168.0.0 192.168.255.255

#/Stream
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[FFmpeg-user] How to dynamically drop frames as required to keep transcoding real time?

2021-04-05 Thread Adrian Cable 
Hi,
I have a question that I thought should have an ‘easy’ answer – I have spent a 
lot of time searching and I cannot find it, so thought I would ask here.

My application sounds simple. I have an input RTSP stream, which I need to 
transcode and output as an RTP stream. Command line is very simple, something 
like:

./ffmpeg -i rtsp://… -vf scale=1024:-1 -c:v libx264 -f rtp rtp://…

Now, ffmpeg is running on a CPU-constrained platform, and it may be that, in 
order for the transcoding to ‘keep up’ with the input, the output framerate 
needs to be less than the input framerate, which is totally fine. The 
problem/question is: how do I get ffmpeg to drop frames as required to keep the 
pipeline running real time?

I can’t simply ‘guess’ a sustainable output framerate and set using -r XXX or 
-vf fps=fps=XXX, because the properties of the input can change at any time, so 
any XXX won’t be constant. Besides, if the ‘guess’ is too high for the CPU 
power I have available, the encoder won’t keep up with the input frames and so 
will ‘run away’ (i.e. the time difference between each input frame and its 
corresponding output frame will get bigger and bigger), which is the problem I 
am trying to solve. On the other hand, if the ‘guess’ is too low, I will end up 
with real-time output but I will also be potentially transcoding to a lower 
output frame rate than I have the CPU power to do, which I don’t want either.

What I need is some kind of dynamic framerate filter that doesn’t take a number 
at all (or maybe takes simply a maximum), and drops frames if and only if the 
encoder can’t keep up … so the pipeline produces as high a frame rate output as 
possible, while still being real time.

I cannot figure out how to do this. Surely this wish can’t be super uncommon as 
you would imagine it would be quite important in all sorts of streaming 
applications where unlimited transcoding power is not guaranteed to be 
available. Can anyone help?

-Adrian
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Re: [FFmpeg-user] How to dynamically drop frames as required to keep transcoding real time?

2021-04-07 Thread Adrian Cable 
Mark – sorry if I wasn’t clear in my original email. The problem I am 
describing is real, not theoretical. I have enormous respect for the time of 
everyone on this list, and would not have emailed if my problem could be solved 
with “-vsync cfr” or anything as straightforward and as well documented.

XXX *will not* be constant and *will not* be independent of the input frame 
rate. This is unfortunately the problem I am trying to solve.

XXX will not be constant because the CPU load on the device (due to other 
running processes) varies over time. If other processes are loading the CPU, 
there will be less CPU available for encoding, so XXX will be lower. If the CPU 
load on the device (due to other running processes) is low, XXX will be higher. 
And, because CPU load (due to other running processes) may change during the 
transcode, XXX will vary during the transcode.

XXX will not be independent of the input frame rate, because decoding the input 
frames consumes CPU and memory bandwidth, which leaves less available for the 
re-encoding part. If the input frame rate is 60 fps, for example, XXX will be 
lower than if the input frame rate were 5 fps.

-Adrian


From: ffmpeg-user  on behalf of Mark Filipak 
(ffmpeg) 
Date: Wednesday, April 7, 2021 at 6:43 AM
To: ffmpeg-user@ffmpeg.org 
Subject: Re: [FFmpeg-user] How to dynamically drop frames as required to keep 
transcoding real time?
On 2021-04-05 21:02, Adrian Cable wrote:
> Hi,
> I have a question that I thought should have an ‘easy’ answer – I have spent 
> a lot of time searching and I cannot find it, so thought I would ask here.
>
> My application sounds simple. I have an input RTSP stream, which I need to 
> transcode and output as an RTP stream. Command line is very simple, something 
> like:
>
> ./ffmpeg -i rtsp://… -vf scale=1024:-1 -c:v libx264 -f rtp rtp://…
>
> Now, ffmpeg is running on a CPU-constrained platform, and it may be that, in 
> order for the transcoding to ‘keep up’ with the input, the output framerate 
> needs to be less than the input framerate, which is totally fine. The 
> problem/question is: how do I get ffmpeg to drop frames as required to keep 
> the pipeline running real time?
>
> I can’t simply ‘guess’ a sustainable output framerate and set using -r XXX or 
> -vf fps=fps=XXX, because the properties of the input can change at any time, 
> so any XXX won’t be constant.

XXX *will* be constant and independent of the input frame rate.
"Convert the video to specified constant frame rate by duplicating or dropping 
frames as necessary."

I hope this helps,
Mark.
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Re: [FFmpeg-user] How to dynamically drop frames as required to keep transcoding real time?

2021-04-07 Thread Adrian Cable 
Mark – yes – you’re understanding exactly right! I was thinking specifically of 
the encoder ‘throttle’ being a frame rate control, but other forms of quality 
control that could be self-adjusted by the encoder to keep up with the input 
would also work for me.

I did look through the history here but couldn’t find anything quite 
applicable. I do appreciate you taking the time to understand my issue! 
Hopefully someone who has direct experience of how to achieve this will read 
this as a result and be able to help.

-Adrian


From: ffmpeg-user  on behalf of Mark Filipak 
(ffmpeg) 
Date: Wednesday, April 7, 2021 at 7:19 PM
To: ffmpeg-user@ffmpeg.org 
Subject: Re: [FFmpeg-user] How to dynamically drop frames as required to keep 
transcoding real time?
On 2021-04-07 21:51, Adrian Cable wrote:
> Mark,
>
>> Sorry, I thought XXX referred to the output -- -vf fps=fps=XXX is *output* 
>> frame rate. To the best
>> of my knowledge, the input frame rate of VFR video can't be specified.
>
> I’m referring to XXX as the maximum *output* frame rate that the pipeline can 
> support at a given time, while keeping up with the input.

I think what you want is an encoder with a quality throttle that can be used to 
dial down the
quality (and hence, reduce encoding time) when it can't keep up with the input 
at the current
compression factor. I'm pretty sure that such an encoder control exists because 
what you want to do
is not at all uncommon. That is, if I understand what you want to do.

I don't know how to do it because I don't stream, but I've seen the issue 
addressed here previously.
Perhaps someone more knowledgeable will read this and help.
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Re: [FFmpeg-user] How to dynamically drop frames as required to keep transcoding real time?

2021-04-07 Thread Adrian Cable 
Mark,

> Sorry, I thought XXX referred to the output -- -vf fps=fps=XXX is *output* 
> frame rate. To the best
> of my knowledge, the input frame rate of VFR video can't be specified.

I’m referring to XXX as the maximum *output* frame rate that the pipeline can 
support at a given time, while keeping up with the input.

> If input FR rises above the CFR output, the 'fps' filter will drop input 
> frames in order to maintain
> CFR output.

Sure! But this isn’t what I’m looking for. If the actual output FR drops below 
the FR set (by -r XXX or -vf fps=fps=XXX etc.), the encoder will not drop 
frames. Instead the delay between input and output will just increase forever. 
I am looking for a way to make the encoder drop frames if the actual output FR 
drops below the FR set. For example: if I set -r 15 but (due to CPU limits) the 
encoder can only manage to produce 6 fps, right now it won’t drop frames, and 
as a result won’t keep up with the input. I want to make it drop frames if 
needed so it keeps up with the input.

-Adrian


From: ffmpeg-user  on behalf of Mark Filipak 
(ffmpeg) 
Date: Wednesday, April 7, 2021 at 2:49 PM
To: ffmpeg-user@ffmpeg.org 
Subject: Re: [FFmpeg-user] How to dynamically drop frames as required to keep 
transcoding real time?
On 2021-04-07 10:10, Adrian Cable wrote:
> Mark – sorry if I wasn’t clear in my original email. The problem I am 
> describing is real, not theoretical. I have enormous respect for the time of 
> everyone on this list, and would not have emailed if my problem could be 
> solved with “-vsync cfr” or anything as straightforward and as well 
> documented.
>
> XXX *will not* be constant and *will not* be independent of the input frame 
> rate. This is unfortunately the problem I am trying to solve.

Sorry, I thought XXX referred to the output -- -vf fps=fps=XXX is *output* 
frame rate. To the best
of my knowledge, the input frame rate of VFR video can't be specified.

> XXX will not be constant because the CPU load on the device (due to other 
> running processes) varies over time. If other processes are loading the CPU, 
> there will be less CPU available for encoding, so XXX will be lower. ...

If the input FR drops below the CFR output, the 'fps' filter will repeat input 
frames in order to
maintain CFR output.

>... If the CPU load on the device (due to other running processes) is low, XXX 
>will be higher. ...

If input FR rises above the CFR output, the 'fps' filter will drop input frames 
in order to maintain
CFR output.

>... And, because CPU load (due to other running processes) may change during 
>the transcode, XXX will vary during the transcode.

It seems to me that what you want is to somehow maintain picture rate 
regardless of frame rate. I'm
sorry if I misunderstand, but I don't think I can help you. Sorry. I suggest 
you look into variable
resolution encoding in order to maintain constant picture rate (but at lower 
resolution if/when
input FR exceeds output FR). I don't know how to do that.

Good hunting,
Mark.

> XXX will not be independent of the input frame rate, because decoding the 
> input frames consumes CPU and memory bandwidth, which leaves less available 
> for the re-encoding part. If the input frame rate is 60 fps, for example, XXX 
> will be lower than if the input frame rate were 5 fps.
>
> -Adrian
>
>
> From: ffmpeg-user  on behalf of Mark Filipak 
> (ffmpeg) 
> Date: Wednesday, April 7, 2021 at 6:43 AM
> To: ffmpeg-user@ffmpeg.org 
> Subject: Re: [FFmpeg-user] How to dynamically drop frames as required to keep 
> transcoding real time?
> On 2021-04-05 21:02, Adrian Cable wrote:
>> Hi,
>> I have a question that I thought should have an ‘easy’ answer – I have spent 
>> a lot of time searching and I cannot find it, so thought I would ask here.
>>
>> My application sounds simple. I have an input RTSP stream, which I need to 
>> transcode and output as an RTP stream. Command line is very simple, 
>> something like:
>>
>> ./ffmpeg -i rtsp://… -vf scale=1024:-1 -c:v libx264 -f rtp rtp://…
>>
>> Now, ffmpeg is running on a CPU-constrained platform, and it may be that, in 
>> order for the transcoding to ‘keep up’ with the input, the output framerate 
>> needs to be less than the input framerate, which is totally fine. The 
>> problem/question is: how do I get ffmpeg to drop frames as required to keep 
>> the pipeline running real time?
>>
>> I can’t simply ‘guess’ a sustainable output framerate and set using -r XXX 
>> or -vf fps=fps=XXX, because the properties of the input can change at any 
>> time, so any XXX won’t be constant.
>
> XXX *will* be constant and independent of the input frame rate.
> "Convert the video to specified constant frame rate by duplicating or 
> dropping frames as necessary.&

[FFmpeg-user] video concat without reencoding keeps audio in one order but not the other

2023-10-11 Thread Adrian Szatmari 
So I took a video and cut it two ways, one with reencoding and one
without reencoding, call them R and C (cut). Im trying to do
concatenation without reencoding, and I'm able to concat the video in
all orders , , , , but the sound disappears
for  and I have no idea why.

I know I can concatenate with reencoding, but my question is
specifically if someone knows why the sound drops in  and maybe
how to fix it. Here is some code in python



import os
import json
import subprocess

video_file = 'yourpath/vid.mp4'
# get the time between frames

ffprobe_command = f"ffprobe -v error -select_streams v:0 -show_entries
stream=width,height,r_frame_rate,duration,nb_frames,codec_name -of
json {video_file}"
result = subprocess.run(ffprobe_command, shell=True,
stdout=subprocess.PIPE, stderr=subprocess.PIPE, text=True)

# Parse the JSON output from ffprobe
json_data = json.loads(result.stdout)

# Extract video information from the JSON data
video_info = {
'length': int(json_data['streams'][0]['nb_frames']),
'frames': int(json_data['streams'][0]['nb_frames']),
'duration_seconds': float(json_data['streams'][0]['duration']),
'fps': eval(json_data['streams'][0]['r_frame_rate']),
'codec': json_data['streams'][0]['codec_name'],
'time_between_frames': 1 / eval(json_data['streams'][0]['r_frame_rate'])
}
# get the time between frames
delta = video_info['time_between_frames']

directory, filename_with_extension = os.path.split(video_file)
filename, extension = os.path.splitext(filename_with_extension)
tag = "_reencode"
new_filename = f"{filename}{tag}{extension}"
reencode_file = directory + "/" + new_filename

tag = "_justcut"
new_filename = f"{filename}{tag}{extension}"
justcut_file = directory + "/" + new_filename

tag = "_concat"
new_filename = f"{filename}{tag}{extension}"
concat_file = directory + "/" + new_filename

start_time = 0.0 + 108 * delta
end_time = start_time + 180 * delta
end_frame = round(end_time / delta)
start_frame = round(start_time / delta)

# Reencode
cmd = [
"ffmpeg",
"-i", video_file,
"-ss", str(start_time),
"-to", str(end_time),
# "-c:a", "copy",
"-c:v", "libx264",
"-bf", str(0), # no B frames
"-crf", str(18),  # new
"-preset", "slow",  # new
# "-g", str(60), #forces key_frames
# "-force_key_frames expr:gte(t, n_forced * GOP_LEN_IN_SECONDS)"
# "-c:v", "mpeg4", #"copy", #"mpeg4"
# "-q:v", "2",
"-c:a", "copy",
# "video_track_timescale", str(90)+"K",
reencode_file
]
subprocess.run(cmd, check=True)

# Just Cut
cmd = [
'ffmpeg',
'-i', video_file,
'-c', 'copy',
'-ss', str(start_time),
'-to', str(end_time),
justcut_file
]
subprocess.run(cmd, check=True)

# Concat without reencoding
def concatenate_videos_without_reencoding(video1, video2, output_file):
try:
# Create a text file listing the videos to concatenate
with open('input.txt', 'w') as f:
f.write(f"file '{video1}'\n")
f.write(f"file '{video2}'\n")



As you can see I tried a bunch of stuff but I have no clue why the
audio is not working. I've tried to force the two input videos to be
as similar as possible in term of specs and keyframes and fps etc.
Nothing seems to work, kinda frustrating.

I've also tried this with both mpeg4 and h264 and different
framerates, but still get this behavior.
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[FFmpeg-user] Can be done with FFMPEG?

2021-04-29 Thread Adrian Perello Marin 
Hello I have used the FFMPEG and “manually” Works perfect, but I don’t know if 
can be done from a CSV get some info to process some tasks:

1 Insert a JPG frontend (allways the same)
2 Create a Blank page with a border and include text (Different size and 
formats, Title, Subtitle, etc.)
3 Insert a JPG backend (allways the same)
4 Saves for YouTube and upload
5 Saves for RRSS less quality

Actually I can do and test (1 and 3) the option 2 try with 
drawtext=textfile=FILE.TXT but the format for the text show very close ☹
The 4 I’m searching if can be done and how, the 5th option I see can be done 
but not how….

Perhaps I need a profesional that can Help me to do it, in that case contact me 
to see how can we work together, this is for a  non-profit organisation from 
Spain. Thanks ¡!


Enviado desde Correo para 
Windows 10

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Re: [FFmpeg-user] Credit Notes (bulk creation)

2021-04-28 Thread Adrian Perello Marin 
Great, I found how to do it thanks ¡!!

https://vimeo.com/445276018
https://ottverse.com/ffmpeg-drawtext-filter-dynamic-overlays-timecode-scrolling-text-credits/

Enviado desde Correo<https://go.microsoft.com/fwlink/?LinkId=550986> para 
Windows 10

De: Carl Zwanzig<mailto:c...@tuunq.com>
Enviado: miércoles, 28 de abril de 2021 21:20
Para: ffmpeg-user@ffmpeg.org<mailto:ffmpeg-user@ffmpeg.org>
Asunto: Re: [FFmpeg-user] Credit Notes (bulk creation)

On 4/28/2021 9:26 AM, Adrian Perello Marin wrote:
> Is it possible to do this by command line so that it reads the content of
> the CSV (video name and credits text) and processes them in a "more or
> less" automatic way?
As Nicolas says, not directly, however it would be a relatively simple
script to do this (read the csv, for each line break out the filename/text
into variables, put them into an ffmpeg command, execute that command).

There are a fair number of web pages that will give a path to drawing the
text and showing it at the right time- search "ffmpeg drawtext examples" as
a start.

z!
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[FFmpeg-user] Credit Notes (bulk creation)

2021-04-28 Thread Adrian Perello Marin 
Hi, I am not sure if with this tool I can do the following:
I receive about 100 videos, with a CSV file containing the credits of the 
video; and I have to convert the text in each of the videos to play at the 
beginning or at the end of the video. Is it possible to do this by command line 
so that it reads the content of the CSV (video name and credits text) and 
processes them in a "more or less" automatic way?

Translated with www.DeepL.com/Translator (free version)

Enviado desde Correo para 
Windows 10

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