I want to take snapshots periodically of a RTMP live video stream.
I can see the rtmp video stream using VLC. This is the rtmp url:
rtmp://antena3fms35livefs.fplive.net/antena3fms35live-live/stream-antena3_1
I'm using the following command to capture the snapshots, according to the official FFmpeg site here:
ffmpeg -i rtmp://antena3fms35livefs.fplive.net/antena3fms35live-live/stream-antena3_1 -f image2 -vf fps=fps=1 out%d.png
The command produces the following output:
ffmpeg version N-64667-gd595361 Copyright (c) 2000-2014 the FFmpeg developers
built on Jul 14 2014 22:09:48 with gcc 4.8.3 (GCC)
configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzl
libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amr
enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --ena
libavutil 52. 92.100 / 52. 92.100
libavcodec 55. 69.100 / 55. 69.100
libavformat 55. 47.100 / 55. 47.100
libavdevice 55. 13.102 / 55. 13.102
libavfilter 4. 10.100 / 4. 10.100
libswscale 2. 6.100 / 2. 6.100
libswresample 0. 19.100 / 0. 19.100
libpostproc 52. 3.100 / 52. 3.100
HandShake: client signature does not match!
Closing connection: NetStream.Play.StreamNotFound
rtmp://antena3fms35livefs.fplive.net/antena3fms35live-live/stream-antena3_1: Unknown error occurred
I've tried it with another rmtp streams, but I'm still getting the exact same error.
What could be the problem?
Thank you!
I just tried your command and it worked fine for me. Maybe it is something about your FFMPEG installation? I am using version 2.4 on a Mac (tessus build).
I know other/older versions used "librtmp" for rtmp connections, which required some extra options behind the stream URL. See ffmpeg docs here:
ffmpeg documentation on librtmp
And librtmp docs here:
librtmp documentation
For an unprotected live stream, you may want to try quoting the stream URL and appending " live=1" within the quotes:
ffmpeg -i "rtmp://antena3fms35livefs.fplive.net/antena3fms35live-live/stream-antena3_1 live=1" -f image2 -vf fps=fps=1 out%d.png
Related
The quality of JPGs extracted by ffmpeg from an mp4 is much poorer than pause frame from video player (vlc). I am looking for ffmpeg cmd option to improve output quality.
Using following cmd :
/home/tools/bin/ffmpeg -i Merkurtransit_20191111_crf20_8fps_crop.mp4 Merkurtransit_20191111_crf20_8fps_crop_%04d.jpg -hide_banner
The ffmpeg cmd is from instructions found here :
https://www.bugcodemaster.com/article/extract-images-frame-frame-video-file-using-ffmpeg
A comparing screen copy is here:
http://skywatcher.space/download/vlc_player_vs_ffmpeg_bug.png
A few items of note. I created the mp4 myself from high res png (actually originally from 16bit tiff) using ffmpeg :
/home/tools/bin/ffmpeg -framerate 8.0 -i ./AS_P10_RS6_png_reg/Merkurtransit_20191111_%03d.png -vf "crop=760:560:20:40" -pix_fmt yuv420p -crf 20 -r 24 -y ./Video/Merkurtransit_20191111_crf20_8fps_crop.mp4
The crf 20 is pretty high quality, close to 100% and the recovered frame should be close to original. The video player pause frame shows adequate quality. (though I can't say if it is on a key frame or not)
ffmpeg version info:
home/tools/bin/ffmpeg -v
ffmpeg version N-80251-g0c7fa15 Copyright (c) 2000-2016 the FFmpeg developers
built with gcc 4.8 (Ubuntu 4.8.4-2ubuntu1~14.04.3)
configuration: --prefix=/home/tools/ffmpeg_build --pkg-config-flags=--static --extra-cflags=-I/home/tools/ffmpeg_build/include --extra-ldflags=-L/home/tools/ffmpeg_build/lib --bindir=/home/tools/bin --enable-gpl --enable-libass --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libopus --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-nonfree
libavutil 55. 24.100 / 55. 24.100
libavcodec 57. 45.100 / 57. 45.100
libavformat 57. 37.101 / 57. 37.101
libavdevice 57. 0.101 / 57. 0.101
libavfilter 6. 46.101 / 6. 46.101
libswscale 4. 1.100 / 4. 1.100
libswresample 2. 0.101 / 2. 0.101
libpostproc 54. 0.100 / 54. 0.100
FFmpeg is primarily a video convertor and JPEG output is the result of a MJPEG encoder generating a single image. When no rate control parameters are set, a default bitrate of 200 kbps is selected.
For a better quality output, use
ffmpeg -i in.mp4 -q:v 1 -qmin 1 -qmax 1 out%d.jpg
The quantizer is clamped to exactly 1.
I am using ffmpeg to convert mkv movies to mp4, like so:
$ ffmpeg -i source.mkv -c:v copy -c:a aac destination.mp4
By doing this, the audio stream gets converted from the original DTS:
Stream #0:1: Audio: dts (DTS), 48000 Hz, 5.1(side), fltp, 1536 kb/s (default)
To AAC:
Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, 5.1, fltp, 394 kb/s (default)
The resulting file plays just fine, except that the spoken audio (what I am assuming would be sent to the central channel in a 5.1 configuration) sounds clearly shifted to the right, when listening through the MacBook's built-in speakers or my stereo headphones. Note that music and other sound effects appear unaffected, properly balanced. Also note that I have been able to reproduce this behavior with a variety of source files.
Here's ffmpeg's banner:
ffmpeg version 4.0.2 Copyright (c) 2000-2018 the FFmpeg developers
built with Apple LLVM version 9.1.0 (clang-902.0.39.2)
configuration: --prefix=/usr/local/Cellar/ffmpeg/4.0.2 --enable-shared --enable-pthreads --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-gpl --enable-libmp3lame --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-libxvid --enable-opencl --enable-videotoolbox --disable-lzma
libavutil 56. 14.100 / 56. 14.100
libavcodec 58. 18.100 / 58. 18.100
libavformat 58. 12.100 / 58. 12.100
libavdevice 58. 3.100 / 58. 3.100
libavfilter 7. 16.100 / 7. 16.100
libavresample 4. 0. 0 / 4. 0. 0
libswscale 5. 1.100 / 5. 1.100
libswresample 3. 1.100 / 3. 1.100
libpostproc 55. 1.100 / 55. 1.100
An FFmpeg commit, aacenc: support extended channel layouts using PCEs, in Nov 2017 added support for many more channel layouts than specified in the MPEG standard for AAC. Unfortunately, it seems it has broken encoding for layouts that worked fine before.
The 3.4 release series is the last before the said commit and should be used for multichannel AAC encoding if you encounter an error with more recent builds.
There is an open bug report at https://trac.ffmpeg.org/ticket/7273. You may post a comment in there to showcase your example.
I wanted to use mplayer (a command line tool) to play a SoundCloud url that I fetched with there official API. The url looks like this
https://cf-media.sndcdn.com/qop35iYKlHhs.128.mp3?Policy=eyJTdGF0ZW1lbnQiOlt7IlJlc291cmNlIjoiKjovL2NmLW1lZGlhLnNuZGNkbi5jb20vcW9wMzVpWUtsSGhzLjEyOC5tcDMiLCJDb25kaXRpb24iOnsiRGF0ZUxlc3NUaGFuIjp7IkFXUzpFcG9jaFRpbWUiOjE0Njg3NTM4NjB9fX1dfQ__&Signature=OZ4IawmjdU6hEN8KB8e~t3QTy4lBFVBbIDg-a6tuPkb65e0mPskkCFFASkTWbyy5lsaT9IrURan6y70sZhLXPbng1IfkTsGdX1dO938NwVYnKs-BS7IL4TiVFxTBXoKJgmCmUD0qmmGmYqm3YdGZQrP~Sj~mw9~fmtwdHQu0rhl3O-dKsgk497JAR6pMorQs7BSs0XIosV1Mmv2DMD6iifquCWV9Ezq4ekneQ1gfSjVmzjjnKvsxjPpgmU~5DTdewwzlNClVxdzHSQONWM7c0YMGlBcVz97NviaeZOQSCAW8QZS59WULXJBJ9OmEPctWtpe3O0mo-GYjwEkbVGYl9A__&Key-Pair-Id=APKAJAGZ7VMH2PFPW6UQ
I use the command "mplayer 'url'" to play it but I always get the output:
libavformat version 57.25.100 (internal)
https protocol not found, recompile FFmpeg with openssl, gnutls,
or securetransport enabled.
Looking which protocols have been installed tells me that I have https input and output support (ffmpeg -protocols):
ffmpeg version 3.1.1 Copyright (c) 2000-2016 the FFmpeg developers
built with Apple LLVM version 7.3.0 (clang-703.0.31)
configuration: --prefix=/usr/local/Cellar/ffmpeg/3.1.1 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-opencl --enable-libx264 --enable-libmp3lame --enable-libxvid --enable-libtheora --enable-libvorbis --enable-libvpx --enable-librtmp --enable-libfaac --enable-ffplay --enable-libspeex --enable-openssl --enable-libopus --disable-lzma --enable-nonfree --enable-vda
libavutil 55. 28.100 / 55. 28.100
libavcodec 57. 48.101 / 57. 48.101
libavformat 57. 41.100 / 57. 41.100
libavdevice 57. 0.101 / 57. 0.101
libavfilter 6. 47.100 / 6. 47.100
libavresample 3. 0. 0 / 3. 0. 0
libswscale 4. 1.100 / 4. 1.100
libswresample 2. 1.100 / 2. 1.100
libpostproc 54. 0.100 / 54. 0.100
Supported file protocols:
Input:
async
cache
concat
crypto
data
file
ftp
gopher
hls
http
httpproxy
https
mmsh
mmst
pipe
rtp
srtp
subfile
tcp
tls
udp
udplite
unix
rtmp
rtmpe
rtmps
rtmpt
rtmpte
Output:
crypto
file
ftp
gopher
http
httpproxy
https
icecast
md5
pipe
rtp
srtp
tcp
tls
udp
udplite
unix
rtmp
rtmpe
rtmps
rtmpt
rtmpte
I am kind of new to the streaming world... so, please forgive me if I ask kind of dumb question.
I am trying to stream my pre-recorded file over RTSP through FFserver.
My config file is :
RTSPPort 8544
<Feed feed2.ffm>
File /home/xyz/tmp/feed2.ffm
FileMaxSize 200K
ACL allow 127.0.0.1
</Feed>
<Stream test.sdp>
Feed feed2.ffm
Format rtsp
VideoFrameRate 15
VideoSize 352x240
VideoBitRate 256
VideoBufferSize 40
VideoGopSize 30
AudioBitRate 64
StartSendOnKey
</Stream>
After starting the server, it give the below log:
$ ./ffserver -f doc/ffserver.conf
ffserver version 0.11.1 Copyright (c) 2000-2012 the FFmpeg developers
built on Sep 17 2012 19:46:38 with gcc 4.1.2 20080704 (Red Hat 4.1.2-52)
configuration: --enable-gpl --enable-libmp3lame --enable-libtheora --enable-libvo-aacenc
-enable-libvorbis --enable-libvpx --enable-libx264 --enable-version3
libavutil 51. 54.100 / 51. 54.100
libavcodec 54. 23.100 / 54. 23.100
libavformat 54. 6.100 / 54. 6.100
libavdevice 54. 0.100 / 54. 0.100
libavfilter 2. 77.100 / 2. 77.100
libswscale 2. 1.100 / 2. 1.100
libswresample 0. 15.100 / 0. 15.100
libpostproc 52. 0.100 / 52. 0.100
Wed Sep 19 17:03:32 2012 FFserver started.
And now from my VLC client I am trying to type the URL: rtsp://xxx.xxx.xxx.xxx:8554/test.sdp
But, what happens is that, there is no response on the ffserver.
I have no clue what might be the problem. Thanks in advance.
You don't have anything to stream.
you need to start
ffmpeg -i <source> http://localhost:8090/feed2.ffm
IF you enable Port 8090 for http,
with the directive (in your config file)
Port 8090
This has been asked before, but badly tagged, so I can't find it.
If anybody finds it, please link it here. As starting an empty server seems to be common.
Matthias is right. Currently you are not streaming anything.
And the given ffmpeg-command should work but you might want to consider, that maybe the feed section is contra productive.
If the video-file is already stored on the server, you don't need the feed (since the video itself will serve as one).
<Stream test.sdp>
File "path_to_your_file" #instead of the Feed
...
</Stream>
If the video-file is on a different computer, you have to stream it to the server first (see Matthias' answer).
Edit:
Also you need the feed if you want to mess around with the stream before streaming.
How can I keep the flow (protocol rtsp, codec h264) in file (container mp4)? That is, on inputting an endless stream (with CCTV camera), and the output files in mp4 format size of 5-10 minutes of recording time.
OS: debian, ubuntu
Software: vlc, ffmpeg (avconv)
Currently this scheme is used:
cvlc rtsp://admin:admin#10.1.1.1:554/ch1-s1 --sout=file/ts:stream.ts
ffmpeg -i stream.ts -vcodec copy -f mp4 stream.mp4
But it can not record video continuously (between restarts vlc loses about 10 seconds of live video).
See this question and answer on Server Fault. In short, switch tools. avconv will do what you want. (ffmpeg has become avconv.)
The feature you are looking for is called segmentation. Your command line would look something like this:
avconv -i rtsp://10.2.2.19/live/ch01_0 -c copy -map 0 -f segment -segment_time 300 -segment_format mp4 "capture-%03d.mp4"
Alexander Garden solution works for ffmpep using the version below. Replace avconv with ffmpeg.
./ffmpeg -i rtsp://10.2.2.19/live/ch01_0 -c copy -map 0 -f segment -segment_time 300 -segment_format mp4 "capture-%03d.mp4"
I'm including this header because of the FFmpeg confusion over versions, the ubuntu schism and rapid development.
ffmpeg version N-80023-gd55568d Copyright (c) 2000-2016 the FFmpeg developers
built with gcc 4.8 (Ubuntu 4.8.4-2ubuntu1~14.04.1)
configuration: --prefix=/home/rhinchley/q10/ffmpeg_build --pkg-config-flags=--static --extra-cflags=-I/home/rhinchley/q10/ffmpeg_build/include --extra-ldflags=-L/home/rhinchley/q10/ffmpeg_build/lib --bindir=/home/rhinchley/q10/bin --enable-gpl --enable-libass --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libopus --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-nonfree
libavutil 55. 24.100 / 55. 24.100
libavcodec 57. 42.100 / 57. 42.100
libavformat 57. 36.100 / 57. 36.100
libavdevice 57. 0.101 / 57. 0.101
libavfilter 6. 45.100 / 6. 45.100
libswscale 4. 1.100 / 4. 1.100
libswresample 2. 0.101 / 2. 0.101
libpostproc 54. 0.100 / 54. 0.100
Team work: Split the video source and have two processes alternate recording the time frame. You'll want to test how variable the startup time is, and how variable it is. You might want to set the processes priority to realtime to reduce start time variance. There will be some overlap but that sound like it might be ok for your application from what I infer. Example:
p1: sRRRRRRRRRwwwwwwwwsRRRRRRRRRwwwwwwwwsRRRRRRRRR...
p2: wwwwwwwwwsRRRRRRRRRwwwwwwwwsRRRRRRRRRwwwwwwwww...
time -->
s: startup
R: running
w: wait