Recherche avancée

Médias (1)

Mot : - Tags -/berlin

Autres articles (67)

  • Publier sur MédiaSpip

    13 juin 2013

    Puis-je poster des contenus à partir d’une tablette Ipad ?
    Oui, si votre Médiaspip installé est à la version 0.2 ou supérieure. Contacter au besoin l’administrateur de votre MédiaSpip pour le savoir

  • HTML5 audio and video support

    13 avril 2011, par

    MediaSPIP uses HTML5 video and audio tags to play multimedia files, taking advantage of the latest W3C innovations supported by modern browsers.
    The MediaSPIP player used has been created specifically for MediaSPIP and can be easily adapted to fit in with a specific theme.
    For older browsers the Flowplayer flash fallback is used.
    MediaSPIP allows for media playback on major mobile platforms with the above (...)

  • Support audio et vidéo HTML5

    10 avril 2011

    MediaSPIP utilise les balises HTML5 video et audio pour la lecture de documents multimedia en profitant des dernières innovations du W3C supportées par les navigateurs modernes.
    Pour les navigateurs plus anciens, le lecteur flash Flowplayer est utilisé.
    Le lecteur HTML5 utilisé a été spécifiquement créé pour MediaSPIP : il est complètement modifiable graphiquement pour correspondre à un thème choisi.
    Ces technologies permettent de distribuer vidéo et son à la fois sur des ordinateurs conventionnels (...)

Sur d’autres sites (7548)

  • Nginx rtmp module - on_publish fires multiple time instead of once

    29 juillet 2017, par Stephen Wright

    This is copy and pasted from the bug report I created on the rtmp-module by Arut, I am not completely sure if it is a bug or me not understanding how the module works, I have read the whole directives of module as from https://github.com/arut/nginx-rtmp-module/wiki/Directives

    Proper explanation, if code is not displayed properly I will edit and fix

    Hi, been using the module and finding it very very good !

    Think I have found a issue though, although it may be me misunderstanding the directives.

    Essentially I wish to fire a script (/usr/local/bin/make_thumbnail.sh) which creates a thumbnail automatically from a stream (using ffmpeg), the idea is to have this done for every stream as soon as it is published in order to create a function a bit like twitch tv where the streamer will not have to specify any thumbnail image, authenticated users simply start a stream (which will later be authenticated but is not yet) The script does also write data into the database however this stage works fine and I don’t believe the issue is related, if I comment out these lines then the thumbnail creation still works and my issue continues.

    Initially this was done using the "exec" command as I believe I mis-read the documentation and I believe the exec command doesn’t work for my problem as ". When publishing stops the process is terminated." does this mean it will continually execute until stream stops ?

    I have started using the exec_publish command to try and fix this issue however the same issue seems to occur. The entire script repeats approximately every 15-17 seconds, a new thumbnail is created and a new database entry is create with all the correct information.

    Below is the nginx.conf line. Please ignore if indentation is incorrect couldn’t see a way to indent blocks of code and it’s late here, assume all code is indented correctly unless you believe that could be the issue in which case I will post it indented as early as I can.

    application live {
    
allow play all;
    
live on;
    
record all;
    
record_path /var/stream/video_recordings/;
    
record_unique on; 
    
hls on;
    
hls_nested on;
    
hls_path /var/stream/HLS/live;
    
hls_fragment 10s;
    

    
#on publish create thumbnail using first second of stream and save in 
    
/var/stream/video_recordings/thumbnails
    
exec_publish usr/local/bin/make_thumbnail.sh $name;

    The rest can be pasted or attached if needed but is working nginx config for rtmp + website

    The most simple version of the make_thumbnail..sh is pasted below, I have omitted the variables that I have used for database entryys obviously but as the script works without fail from terminal I believe this to be an nginx issue (if I run the command manually under the nginx user e.g. sudo -u nginx /usr/local/bin/make_thumbnail.sh with a name the same as any running stream, it works and only executes once as would expect, all permissions in script are ok and tested.

    make_thumbnail.sh

    #!/bin/bash
    

    
TIME=$(date +%s)
    
NAME=$1 
    
echo "time: "
    
FILENAME=${TIME}_${NAME}
    

    
ffmpeg -i rtmp://192.168.0.98:1935/live/$1 -vframes 1 -s 150x150 -ss 10 -
    
strftime 1 /var/stream/video_recordings/thumbnails/"$FILENAME.jpg";
    

    
#Writes path to video into database
    
mysql --user=$DB_USER --password=$DB_PASSWD $DB_NAME << EOF
    
INSERT INTO $TABLE3 (thumbnailfile) VALUES ('$FILENAME');
    
set @last_id_in_thumbnails = LAST_INSERT_ID();
    
INSERT INTO $TABLE (created_at, updated_at, thumnailID) VALUES 
    
(NOW(),NOW(),@last_id_in_thumbnails);
    
SET @last_id_in_livestreams = LAST_INSERT_ID();
    
INSERT INTO $TABLE2 (created_at, updated_at, filename,liveID) VALUES 
    
(NOW(),NOW(),'$FILENAME',@last_id_in_livestreams); 
    
EOF

    I have not got the nginx rtmp logs installed, I can obviously do this however some of the logs appear in the nginx error.log, strangely the latest stream I tried did not update in the access log, however I think this is because I did not attempt to connect to it via any method. I don’t fully understand the error.log, in my stupidity I decided to use nginx with which I am quite inexperienced and I am finding it very difficult to troubleshoot this issue, it appears to me that as part of the RTMP protocol or my streaming software (OBS) is either directly pinging the rtmp stream or is being pinged by the server to ensure the connection is still there. And this ping is

    I have left a stream running from approx 4 minutes without interacting with the server, streaming software, computer running the stream, I have ensured the internet connection is constant as my first though was the connection dropped, however on inspecting the database the executing is done always after at least 11 seconds however usually this is 16, I can’t seem to figure out how to select the closest dates from the database however there has been at least a few 17 second differences (potentially when
    I am unsure if this is an issue or if it is intended behavior but I do require this to finish a university degree, I’m not asking for answers but if it is a legitimate issue then I would be happy to spend as much time I can commit to it if some insight into what is causing it, or if there is a workaround I believe it should be documented somewhere, I have googled into making any exec commands run only once on publish

    I can’t seem to pinpoint where in the log the issue is happening however think it is something to do with the below exceprts I would attach the file but can’t seem to select all lines after the timestamp upon starting a stream

    2017/07/26 18:17:35 [info] 1451#0: *2229 exec: starting managed child 
    
'ffmpeg', client: 192.168.0.78, server: 0.0.0.0:1935
    
2017/07/26 18:17:35 [info] 1451#0: *2412 client connected '192.168.0.98'
    
2017/07/26 18:17:35 [info] 1451#0: *2412 connect: app='live' args='' 
    
flashver='LNX 9,0,124,2' swf_url='' tc_url='rtmp://192.168.0.98:1935/live' 
    
page_url='' acodecs=4071 vcodecs=252 object_encoding=0, client: 
    
192.168.0.98, server: 0.0.0.0:1935
    
2017/07/26 18:17:35 [info] 1451#0: *2412 createStream, client: 192.168.0.98, 
    
server: 0.0.0.0:1935
    
2017/07/26 18:17:35 [info] 1451#0: *2412 play: name='newname' args='' 
    
start=-2000 duration=0 reset=0 silent=0, client: 192.168.0.98, server:  
    
0.0.0.0:1935
    
2017/07/26 18:17:36 [info] 1451#0: *2410 recv() failed (104: Connection  
    
reset by peer), client: 192.168.0.98, server: 0.0.0.0:1935
    
2017/07/26 18:17:36 [info] 1451#0: *2410 disconnect, client: 192.168.0.98, 
    
server: 0.0.0.0:1935
    
2017/07/26 18:17:36 [info] 1451#0: *2410 deleteStream, client: 192.168.0.98, 
    
server: 0.0.0.0:1935
    
2017/07/26 18:17:36 [notice] 1451#0: signal 17 (SIGCHLD) received
    
2017/07/26 18:17:36 [notice] 1451#0: unknown process 10487 exited with code 
    
0
    
2017/07/26 18:17:36 [info] 1451#0: *2229 exec: child 10487 exited; ignoring, 
    
client: 192.168.0.78, server: 0.0.0.0:1935
    

    
ver: 0.0.0.0:1935
    
2017/07/26 18:17:41 [info] 1451#0: *2229 exec: starting managed child 
    
'usr/local/bin/make_thumbnail.sh', client: 192.168.0.78, server:  
    
0.0.0.0:1935
    
2017/07/26 18:17:41 [info] 1451#0: *2413 client connected '192.168.0.98'
    
2017/07/26 18:17:41 [info] 1451#0: *2413 connect: app='live' args='' 
    
flashver='LNX 9,0,124,2' swf_url='' tc_url='rtmp://192.168.0.98:1935/live' 
    
page_url='' acodecs=4071 vcodecs=252 object_encoding=0, client: 
    
192.168.0.98, 
    
server: 0.0.0.0:1935
    
2017/07/26 18:17:41 [info] 1451#0: *2413 createStream, client: 192.168.0.98, 
    
server: 0.0.0.0:1935
    
2017/07/26 18:17:41 [info] 1451#0: *2413 play: name='newname' args='' 
    
start=-2000 duration=0 reset=0 silent=0, client: 192.168.0.98, server: 
    
0.0.0.0:1935
    
2017/07/26 18:17:43 [info] 1451#0: *2229 exec: starting managed child 
    
'ffmpeg', 
    
client: 192.168.0.78, server: 0.0.0.0:1935
    
2017/07/26 18:17:43 [info] 1451#0: *2414 client connected '192.168.0.98'
    
2017/07/26 18:17:43 [info] 1451#0: *2414 connect: app='live' args='' 
    
flashver='LNX 9,0,124,2' swf_url='' tc_url='rtmp://192.168.0.98:1935/live' 
    
page_url='' acodecs=4071 vcodecs=252 object_encoding=0, client: 
    
192.168.0.98, 
    
server: 0.0.0.0:1935
    
@

  • When generating a video from images and adding audio the video isn't playable

    1er août 2017, par randommman

    So I have a bunch of images in a folder, with the following structure :

    image-0.png
    
image-1.png
    
image-2.png

    Sometimes the folder can have just 1 image which will be saved as :

    image-0.png

    My code to generate the video and to add the audio

    // create the file
    
using (WaveFileReader wf = new WaveFileReader(Audio))
    
{
    
    // the files
    
    OldVideo = "old.avi";
    
    NewVideo = "video.avi";
    
    Audio = "sound.wav";
    

    
    // Get the time of the audio and divide by images
    
    time = wf.TotalTime.TotalSeconds;
    
    mimtime = time / imagescount;
    

    
    ffmpegPath = "ffmpeg.exe";
    
    ffmpegParams = " -r 1/" + mimtime + " -i " + ImagesFolder + "image-%d.png -t " + time + " -y -vf scale=1280:-2 " + OldVideo;
    
    ffmaudioParams = " -i " + OldVideo + " -i " + Audio + " -c copy -shortest " + NewVideo;
    
}
    

    

    
using (Process ffmpeg = new Process())
    
{
    
    //Generate video
    
    ffmpegstartInfo = new ProcessStartInfo();
    
    ffmpegstartInfo.FileName = ffmpegPath;
    
    ffmpegstartInfo.Arguments = ffmpegParams;
    
    ffmpegstartInfo.RedirectStandardOutput = true;
    
    ffmpegstartInfo.RedirectStandardError = true;
    
    ffmpegstartInfo.UseShellExecute = false;
    
    ffmpegstartInfo.CreateNoWindow = true;
    

    
    ffmpeg.StartInfo.FileName = ffmpegPath;
    
    ffmpeg.StartInfo.Arguments = ffmpegParams;
    
    ffmpeg.StartInfo = ffmpegstartInfo;
    
    ffmpeg.Start();
    
    ffmpeg.WaitForExit(30000);
    

    

    
    /*ffmpeg.StartInfo.FileName = "cmd.exe";
    
    ffmpeg.StartInfo.Arguments = "/k " + ffmpegPath + " " + ffmpegParams;
    
    ffmpeg.Start();
    
    ffmpeg.WaitForExit(30000);*/
    
}
    

    
using (Process ffmaudio = new Process())
    
{
    
    //Add audio to video
    
    ffmaudiostartInfo = new ProcessStartInfo();
    
    ffmaudiostartInfo.FileName = ffmpegPath;
    
    ffmaudiostartInfo.Arguments = ffmaudioParams;
    
    ffmaudiostartInfo.RedirectStandardOutput = true;
    
    ffmaudiostartInfo.RedirectStandardError = true;
    
    ffmaudiostartInfo.UseShellExecute = false;
    
    ffmaudiostartInfo.CreateNoWindow = true;
    

    
    ffmaudio.StartInfo.FileName = ffmpegPath;
    
    ffmaudio.StartInfo.Arguments = ffmaudioParams;
    
    ffmaudio.StartInfo = ffmaudiostartInfo;
    
    ffmaudio.Start();
    
    ffmaudio.WaitForExit(30000);
    

    
    /*ffmaudio.StartInfo.FileName = "cmd.exe";
    
    ffmaudio.StartInfo.Arguments = "/k " + ffmpegPath + " " + ffmaudioParams;
    
    ffmaudio.Start();
    
    ffmaudio.WaitForExit(30000);*/
    
}

    So, what my code does, with the wavfilereader it gets the total length of my audio file, and then splits that time up by the number of images I have, so I can generate a video with the images split equally. Then I generate the video to a file called ’old.avi’, which is then used with my wav file to generate a new file called video.avi (which I am trying to have the generated video and audio together)

    My problem is when I generate the video it becomes uplayable. It claims to be (for example) 1 minute 50 seconds long, but once I click it, it won’t play. It just becomes unplayable, as you can see below

    http://i.imgur.com/tsck2NX.gifv

    I am using C# with process and Winforms, which opens the cmd line to run ffmpeg, with the 2 commands called above with the named variables below :

    ffmpegParams;
    
ffmaudioParams;

    Which has all the commands I am using

    -r 1/ (audio length /divided by total images) so they're equal
    
-t (to limit to the total time which is the audio's total seconds)
    
... I am using more as you can see in my code above

    How can I fix my problem, that I have described ?

    CMD line when generating the video :

    ffmpeg version N-86848-g03a9e6f Copyright (c) 2000-2017 the FFmpeg developers
    
  built with gcc 7.1.0 (GCC)
    
  configuration: --enable-gpl --enable-version3 --enable-cuda --enable-cuvid --enable-d3d11va --enable-dxva2 --enable-libmfx --enable-nvenc --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-zlib
    
  libavutil      55. 68.100 / 55. 68.100
    
  libavcodec     57.102.100 / 57.102.100
    
  libavformat    57. 76.100 / 57. 76.100
    
  libavdevice    57.  7.100 / 57.  7.100
    
  libavfilter     6. 95.100 /  6. 95.100
    
  libswscale      4.  7.101 /  4.  7.101
    
  libswresample   2.  8.100 /  2.  8.100
    
  libpostproc    54.  6.100 / 54.  6.100
    
Input #0, image2, from 'C:\Users\Laptop\Documents\program/images/image-%d.png':
    
  Duration: 00:00:00.04, start: 0.000000, bitrate: N/A
    
    Stream #0:0: Video: png, rgba(pc), 800x800, 25 tbr, 25 tbn, 25 tbc
    
Stream mapping:
    
  Stream #0:0 -> #0:0 (png (native) -> mpeg4 (native))
    
Press [q] to stop, [?] for help
    
[mpeg4 @ 00000000024f44a0] bitrate tolerance 4000000 too small for bitrate 200000, overriding
    
Output #0, avi, to 'C:\Users\Laptop\Documents\program/old.avi':
    
  Metadata:
    
    ISFT            : Lavf57.76.100
    
    Stream #0:0: Video: mpeg4 (FMP4 / 0x34504D46), yuv420p, 1280x1280, q=2-31, 200 kb/s, 0.01 fps, 0.01 tbn, 0.01 tbc
    
    Metadata:
    
      encoder         : Lavc57.102.100 mpeg4
    
    Side data:
    
      cpb: bitrate max/min/avg: 0/0/200000 buffer size: 0 vbv_delay: -1
    
frame=    1 fps=0.0 q=6.8 Lsize=      46kB time=00:01:37.57 bitrate=   3.9kbits/s speed=1.43e+003x
    
video:41kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 13.641467%

    When adding the audio to the video

    ffmpeg version N-86848-g03a9e6f Copyright (c) 2000-2017 the FFmpeg developers
    
  built with gcc 7.1.0 (GCC)
    
  configuration: --enable-gpl --enable-version3 --enable-cuda --enable-cuvid --enable-d3d11va --enable-dxva2 --enable-libmfx --enable-nvenc --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-zlib
    
  libavutil      55. 68.100 / 55. 68.100
    
  libavcodec     57.102.100 / 57.102.100
    
  libavformat    57. 76.100 / 57. 76.100
    
  libavdevice    57.  7.100 / 57.  7.100
    
  libavfilter     6. 95.100 /  6. 95.100
    
  libswscale      4.  7.101 /  4.  7.101
    
  libswresample   2.  8.100 /  2.  8.100
    
  libpostproc    54.  6.100 / 54.  6.100
    
Input #0, avi, from 'C:\Users\Laptop\Documents\program/old.avi':
    
  Metadata:
    
    encoder         : Lavf57.76.100
    
  Duration: 00:01:37.58, start: 0.000000, bitrate: 3 kb/s
    
    Stream #0:0: Video: mpeg4 (Simple Profile) (FMP4 / 0x34504D46), yuv420p, 1280x1280 [SAR 1:1 DAR 1:1], 0.01 fps, 0.01 tbr, 0.01 tbn, 40 tbc
    
Guessed Channel Layout for Input Stream #1.0 : mono
    
Input #1, wav, from 'C:\Users\Laptop\Documents\program/audio.wav':
    
  Duration: 00:01:37.57, bitrate: 352 kb/s
    
    Stream #1:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 22050 Hz, mono, s16, 352 kb/s
    
Output #0, avi, to 'C:\Users\Laptop\Documents\program/video.avi':
    
  Metadata:
    
    ISFT            : Lavf57.76.100
    
    Stream #0:0: Video: mpeg4 (Simple Profile) (FMP4 / 0x34504D46), yuv420p, 1280x1280 [SAR 1:1 DAR 1:1], q=2-31, 0.01 fps, 0.01 tbr, 0.01 tbn, 0.01 tbc
    
    Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 22050 Hz, mono, s16, 352 kb/s
    
Stream mapping:
    
  Stream #0:0 -> #0:0 (copy)
    
  Stream #1:0 -> #0:1 (copy)
    
Press [q] to stop, [?] for help
    
frame=    1 fps=0.0 q=-1.0 Lsize=      55kB time=00:01:37.57 bitrate=   4.6kbits/s speed=1.95e+005x
    
video:41kB audio:4kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 21.946869%

  • TCP connection refused with FFMPEG

    2 février 2017, par Samul

    OFFICIAL EDIT :

    I thank you so much for your help but I am still encountering problems.

    My ffserver.conf file is like this :

    # Port on which the server is listening. You must select a different
    
# port from your standard HTTP web server if it is running on the same
    
# computer.
    
HTTPPort 8090
    

    
# Address on which the server is bound. Only useful if you have
    
# several network interfaces.
    
HTTPBindAddress 0.0.0.0
    

    
# Number of simultaneous HTTP connections that can be handled. It has
    
# to be defined *before* the MaxClients parameter, since it defines the
    
# MaxClients maximum limit.
    
MaxHTTPConnections 2000
    

    
# Number of simultaneous requests that can be handled. Since FFServer
    
# is very fast, it is more likely that you will want to leave this high
    
# and use MaxBandwidth, below.
    
MaxClients 1000
    

    
# This the maximum amount of kbit/sec that you are prepared to
    
# consume when streaming to clients.
    
MaxBandwidth 1000
    

    
# Access log file (uses standard Apache log file format)
    
# '-' is the standard output.
    
CustomLog -
    

    
##################################################################
    
# Definition of the live feeds. Each live feed contains one video
    
# and/or audio sequence coming from an ffmpeg encoder or another
    
# ffserver. This sequence may be encoded simultaneously with several
    
# codecs at several resolutions.
    

    
<feed>
    

    
# You must use 'ffmpeg' to send a live feed to ffserver. In this
    
# example, you can type:
    
#
    
# ffmpeg http://localhost:8090/feed1.ffm
    

    
# ffserver can also do time shifting. It means that it can stream any
    
# previously recorded live stream. The request should contain:
    
# "http://xxxx?date=[YYYY-MM-DDT][[HH:]MM:]SS[.m...]".You must specify
    
# a path where the feed is stored on disk. You also specify the
    
# maximum size of the feed, where zero means unlimited. Default:
    
# File=/tmp/feed_name.ffm FileMaxSize=5M
    
File /tmp/feed1.ffm
    
FileMaxSize 200K
    

    
# You could specify
    
# ReadOnlyFile /saved/specialvideo.ffm
    
# This marks the file as readonly and it will not be deleted or updated.
    

    
# Specify launch in order to start ffmpeg automatically.
    
# First ffmpeg must be defined with an appropriate path if needed,
    
# after that options can follow, but avoid adding the http:// field
    
#Launch ffmpeg
    

    
# Only allow connections from localhost to the feed.
    
#ACL allow 127.0.0.1
    
#ACL allow 189.34.0.158
    
</feed>
    

    

    
##################################################################
    
# Now you can define each stream which will be generated from the
    
# original audio and video stream. Each format has a filename (here
    
# 'test1.mpg'). FFServer will send this stream when answering a
    
# request containing this filename.
    

    
<stream>
    

    
# coming from live feed 'feed1'
    
Feed feed1.ffm
    

    
# Format of the stream : you can choose among:
    
# mpeg       : MPEG-1 multiplexed video and audio
    
# mpegvideo  : only MPEG-1 video
    
# mp2        : MPEG-2 audio (use AudioCodec to select layer 2 and 3 codec)
    
# ogg        : Ogg format (Vorbis audio codec)
    
# rm         : RealNetworks-compatible stream. Multiplexed audio and video.
    
# ra         : RealNetworks-compatible stream. Audio only.
    
# mpjpeg     : Multipart JPEG (works with Netscape without any plugin)
    
# jpeg       : Generate a single JPEG image.
    
# asf        : ASF compatible streaming (Windows Media Player format).
    
# swf        : Macromedia Flash compatible stream
    
# avi        : AVI format (MPEG-4 video, MPEG audio sound)
    
Format mpeg
    

    
# Bitrate for the audio stream. Codecs usually support only a few
    
# different bitrates.
    
AudioBitRate 32
    

    
# Number of audio channels: 1 = mono, 2 = stereo
    
AudioChannels 1
    

    
# Sampling frequency for audio. When using low bitrates, you should
    
# lower this frequency to 22050 or 11025. The supported frequencies
    
# depend on the selected audio codec.
    
AudioSampleRate 44100
    

    
# Bitrate for the video stream
    
VideoBitRate 64
    

    
# Ratecontrol buffer size
    
VideoBufferSize 40
    

    
# Number of frames per second
    
VideoFrameRate 3
    

    
# Size of the video frame: WxH (default: 160x128)
    
# The following abbreviations are defined: sqcif, qcif, cif, 4cif, qqvga,
    
# qvga, vga, svga, xga, uxga, qxga, sxga, qsxga, hsxga, wvga, wxga, wsxga,
    
# wuxga, woxga, wqsxga, wquxga, whsxga, whuxga, cga, ega, hd480, hd720,
    
# hd1080
    
VideoSize 160x128
    

    
# Transmit only intra frames (useful for low bitrates, but kills frame rate).
    
#VideoIntraOnly
    

    
# If non-intra only, an intra frame is transmitted every VideoGopSize
    
# frames. Video synchronization can only begin at an intra frame.
    
VideoGopSize 12
    

    
# More MPEG-4 parameters
    
# VideoHighQuality
    
# Video4MotionVector
    

    
# Choose your codecs:
    
#AudioCodec mp2
    
#VideoCodec mpeg1video
    

    
# Suppress audio
    
#NoAudio
    

    
# Suppress video
    
#NoVideo
    

    
#VideoQMin 3
    
#VideoQMax 31
    

    
# Set this to the number of seconds backwards in time to start. Note that
    
# most players will buffer 5-10 seconds of video, and also you need to allow
    
# for a keyframe to appear in the data stream.
    
#Preroll 15
    

    
# ACL:
    

    
# You can allow ranges of addresses (or single addresses)
    
#ACL ALLOW <first address="address"> <last address="address">
    

    
# You can deny ranges of addresses (or single addresses)
    
#ACL DENY <first address="address"> <last address="address">
    

    
# You can repeat the ACL allow/deny as often as you like. It is on a per
    
# stream basis. The first match defines the action. If there are no matches,
    
# then the default is the inverse of the last ACL statement.
    
#
    
# Thus 'ACL allow localhost' only allows access from localhost.
    
# 'ACL deny 1.0.0.0 1.255.255.255' would deny the whole of network 1 and
    
# allow everybody else.
    

    
</last></first></last></first></stream>
    

    

    
##################################################################
    
# Example streams
    

    

    
# Multipart JPEG
    

    
#<stream>
    
#Feed feed1.ffm
    
#Format mpjpeg
    
#VideoFrameRate 2
    
#VideoIntraOnly
    
#NoAudio
    
#Strict -1
    
#</stream>
    

    

    
# Single JPEG
    

    
#<stream>
    
#Feed feed1.ffm
    
#Format jpeg
    
#VideoFrameRate 2
    
#VideoIntraOnly
    
##VideoSize 352x240
    
#NoAudio
    
#Strict -1
    
#</stream>
    

    

    
# Flash
    

    
#<stream>
    
#Feed feed1.ffm
    
#Format swf
    
#VideoFrameRate 2
    
#VideoIntraOnly
    
#NoAudio
    
#</stream>
    

    

    
# ASF compatible
    

    
<stream>
    
Feed feed1.ffm
    
Format asf
    
VideoFrameRate 15
    
VideoSize 352x240
    
VideoBitRate 256
    
VideoBufferSize 40
    
VideoGopSize 30
    
AudioBitRate 64
    
StartSendOnKey
    
</stream>
    

    

    
# MP3 audio
    

    
#<stream>
    
#Feed feed1.ffm
    
#Format mp2
    
#AudioCodec mp3
    
#AudioBitRate 64
    
#AudioChannels 1
    
#AudioSampleRate 44100
    
#NoVideo
    
#</stream>
    

    

    
# Ogg Vorbis audio
    

    
#<stream>
    
#Feed feed1.ffm
    
#Metadata title "Stream title"
    
#AudioBitRate 64
    
#AudioChannels 2
    
#AudioSampleRate 44100
    
#NoVideo
    
#</stream>
    

    

    
# Real with audio only at 32 kbits
    

    
#<stream>
    
#Feed feed1.ffm
    
#Format rm
    
#AudioBitRate 32
    
#NoVideo
    
#NoAudio
    
#</stream>
    

    

    
# Real with audio and video at 64 kbits
    

    
#<stream>
    
#Feed feed1.ffm
    
#Format rm
    
#AudioBitRate 32
    
#VideoBitRate 128
    
#VideoFrameRate 25
    
#VideoGopSize 25
    
#NoAudio
    
#</stream>
    

    

    
##################################################################
    
# A stream coming from a file: you only need to set the input
    
# filename and optionally a new format. Supported conversions:
    
#    AVI -> ASF
    

    
#<stream>
    
#File "/usr/local/httpd/htdocs/tlive.rm"
    
#NoAudio
    
#</stream>
    

    
#<stream>
    
#File "/usr/local/httpd/htdocs/test.asf"
    
#NoAudio
    
#Metadata author "Me"
    
#Metadata copyright "Super MegaCorp"
    
#Metadata title "Test stream from disk"
    
#Metadata comment "Test comment"
    
#</stream>
    

    

    
##################################################################
    
# RTSP examples
    
#
    
# You can access this stream with the RTSP URL:
    
#   rtsp://localhost:5454/test1-rtsp.mpg
    
#
    
# A non-standard RTSP redirector is also created. Its URL is:
    
#   http://localhost:8090/test1-rtsp.rtsp
    

    
#<stream>
    
#Format rtp
    
#File "/usr/local/httpd/htdocs/test1.mpg"
    
#</stream>
    

    

    
# Transcode an incoming live feed to another live feed,
    
# using libx264 and video presets
    

    
#<stream>
    
#Format rtp
    
#Feed feed1.ffm
    
#VideoCodec libx264
    
#VideoFrameRate 24
    
#VideoBitRate 100
    
#VideoSize 480x272
    
#AVPresetVideo default
    
#AVPresetVideo baseline
    
#AVOptionVideo flags +global_header
    
#
    
#AudioCodec libfaac
    
#AudioBitRate 32
    
#AudioChannels 2
    
#AudioSampleRate 22050
    
#AVOptionAudio flags +global_header
    
#</stream>
    

    
##################################################################
    
# SDP/multicast examples
    
#
    
# If you want to send your stream in multicast, you must set the
    
# multicast address with MulticastAddress. The port and the TTL can
    
# also be set.
    
#
    
# An SDP file is automatically generated by ffserver by adding the
    
# 'sdp' extension to the stream name (here
    
# http://localhost:8090/test1-sdp.sdp). You should usually give this
    
# file to your player to play the stream.
    
#
    
# The 'NoLoop' option can be used to avoid looping when the stream is
    
# terminated.
    

    
#<stream>
    
#Format rtp
    
#File "/usr/local/httpd/htdocs/test1.mpg"
    
#MulticastAddress 224.124.0.1
    
#MulticastPort 5000
    
#MulticastTTL 16
    
#NoLoop
    
#</stream>
    

    

    
##################################################################
    
# Special streams
    

    
# Server status
    

    
<stream>
    
Format status
    

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

    
#FaviconURL http://pond1.gladstonefamily.net:8080/favicon.ico
    
</stream>
    

    

    
# Redirect index.html to the appropriate site
    

    
<redirect>
    
URL http://www.ffmpeg.org/
    
</redirect>

    I started my server and executed :

    ffserver -d -f /usr/share/doc/ffmpeg-2.6.8/ffserver.conf

    No error message and everything looks fine.

    After that I execute this (in your answer, I think you forgot the port number) :

    ffmpeg -i "rtsp://200.180.90.95:554/onvif1" -r 25 -s 640x480 -c:v libx264 -flags +global_header -f flv "http://45.79.207.38:8090/feed1.ffm"

    Then I get this log :

      libavutil      54. 20.100 / 54. 20.100
    
  libavcodec     56. 26.100 / 56. 26.100
    
  libavformat    56. 25.101 / 56. 25.101
    
  libavdevice    56.  4.100 / 56.  4.100
    
  libavfilter     5. 11.102 /  5. 11.102
    
  libavresample   2.  1.  0 /  2.  1.  0
    
  libswscale      3.  1.101 /  3.  1.101
    
  libswresample   1.  1.100 /  1.  1.100
    
  libpostproc    53.  3.100 / 53.  3.100
    
[h264 @ 0x1a23580] RTP: missed 1 packets
    
[pcm_alaw @ 0x1a24360] RTP: missed 2 packets
    
[h264 @ 0x1a23580] RTP: missed 1 packets
    
Invalid UE golomb code
    
[h264 @ 0x1a23580] cbp too large (3199971767) at 76 33
    
[h264 @ 0x1a23580] error while decoding MB 76 33
    
[h264 @ 0x1a23580] Cannot use next picture in error concealment
    
[h264 @ 0x1a23580] concealing 933 DC, 933 AC, 933 MV errors in P frame
    
[h264 @ 0x1a23580] RTP: missed 1 packets
    
[h264 @ 0x1a23580] cbp too large (62) at 50 24
    
[h264 @ 0x1a23580] error while decoding MB 50 24
    
[h264 @ 0x1a23580] Cannot use next picture in error concealment
    
[h264 @ 0x1a23580] concealing 1679 DC, 1679 AC, 1679 MV errors in P frame
    
[h264 @ 0x1a23580] RTP: missed 2 packets
    
[h264 @ 0x1a23580] Cannot use next picture in error concealment
    
[h264 @ 0x1a23580] concealing 1965 DC, 1965 AC, 1965 MV errors in P frame
    
[pcm_alaw @ 0x1a24360] RTP: missed 1 packets
    
    Last message repeated 1 times
    
[h264 @ 0x1a23580] RTP: missed 3 packets
    
[h264 @ 0x1a23580] mb_type 49 in P slice too large at 74 25
    
[h264 @ 0x1a23580] error while decoding MB 74 25
    
[h264 @ 0x1a23580] Cannot use next picture in error concealment
    
[h264 @ 0x1a23580] concealing 1575 DC, 1575 AC, 1575 MV errors in P frame
    
[h264 @ 0x1a23580] RTP: missed 2 packets
    
[h264 @ 0x1a23580] P sub_mb_type 29 out of range at 30 26
    
[h264 @ 0x1a23580] error while decoding MB 30 26
    
[h264 @ 0x1a23580] Cannot use next picture in error concealment
    
[h264 @ 0x1a23580] concealing 1539 DC, 1539 AC, 1539 MV errors in P frame
    
[h264 @ 0x1a23580] RTP: missed 1 packets
    
[h264 @ 0x1a23580] out of range intra chroma pred mode at 72 29
    
[h264 @ 0x1a23580] error while decoding MB 72 29
    
[h264 @ 0x1a23580] Cannot use next picture in error concealment
    
[h264 @ 0x1a23580] concealing 1257 DC, 1257 AC, 1257 MV errors in P frame
    
[h264 @ 0x1a23580] RTP: missed 3 packets
    
[h264 @ 0x1a23580] negative number of zero coeffs at 48 5
    
[h264 @ 0x1a23580] error while decoding MB 48 5
    
[h264 @ 0x1a23580] Cannot use next picture in error concealment
    
[h264 @ 0x1a23580] concealing 3201 DC, 3201 AC, 3201 MV errors in P frame
    
[pcm_alaw @ 0x1a24360] RTP: missed 1 packets
    
[rtsp @ 0x1a20ee0] decoding for stream 0 failed
    
Guessed Channel Layout for  Input Stream #0.1 : mono
    
Input #0, rtsp, from 'rtsp://200.180.90.95:554/onvif1':
    
  Metadata:
    
    title           : H.264 Video, RtspServer_0.0.0.2
    
  Duration: N/A, start: 0.000000, bitrate: N/A
    
    Stream #0:0: Video: h264 (Baseline), yuv420p, 1280x720, 90k tbr, 90k tbn, 180k tbc
    
    Stream #0:1: Audio: pcm_alaw, 8000 Hz, 1 channels, s16, 64 kb/s
    
[libx264 @ 0x1b728a0] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX AVX2 FMA3 LZCNT BMI2
    
[libx264 @ 0x1b728a0] profile High, level 3.0
    
[libx264 @ 0x1b728a0] 264 - core 142 r2495 6a301b6 - H.264/MPEG-4 AVC codec - Copyleft 2003-2014 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=1 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
    
[flv @ 0x1a66300] FLV does not support sample rate 8000, choose from (44100, 22050, 11025)
    
[flv @ 0x1a66300] Audio codec mp3 not compatible with flv
    
Output #0, flv, to 'http://45.79.207.38:8090/feed1.ffm':
    
  Metadata:
    
    title           : H.264 Video, RtspServer_0.0.0.2
    
    encoder         : Lavf56.25.101
    
    Stream #0:0: Video: h264 (libx264) ([7][0][0][0] / 0x0007), yuv420p, 640x480, q=-1--1, 25 fps, 1k tbn, 25 tbc
    
    Metadata:
    
      encoder         : Lavc56.26.100 libx264
    
    Stream #0:1: Audio: mp3 (libmp3lame) ([2][0][0][0] / 0x0002), 8000 Hz, mono, s16p
    
    Metadata:
    
      encoder         : Lavc56.26.100 libmp3lame
    
Stream mapping:
    
  Stream #0:0 -> #0:0 (h264 (native) -> h264 (libx264))
    
  Stream #0:1 -> #0:1 (pcm_alaw (native) -> mp3 (libmp3lame))
    
Could not write header for output file #0 (incorrect codec parameters ?): Function not implemented

    I am doing this in a clean install of CENTOS, no customization. Could you please helpe me ?

Navigation