Recherche avancée

Médias (0)

Mot : - Tags -/diogene

Aucun média correspondant à vos critères n’est disponible sur le site.

Autres articles (39)

  • Les autorisations surchargées par les plugins

    27 avril 2010, par

    Mediaspip core
    autoriser_auteur_modifier() afin que les visiteurs soient capables de modifier leurs informations sur la page d’auteurs

  • Des sites réalisés avec MediaSPIP

    2 mai 2011, par

    Cette page présente quelques-uns des sites fonctionnant sous MediaSPIP.
    Vous pouvez bien entendu ajouter le votre grâce au formulaire en bas de page.

  • Librairies et binaires spécifiques au traitement vidéo et sonore

    31 janvier 2010, par

    Les logiciels et librairies suivantes sont utilisées par SPIPmotion d’une manière ou d’une autre.
    Binaires obligatoires FFMpeg : encodeur principal, permet de transcoder presque tous les types de fichiers vidéo et sonores dans les formats lisibles sur Internet. CF ce tutoriel pour son installation ; Oggz-tools : outils d’inspection de fichiers ogg ; Mediainfo : récupération d’informations depuis la plupart des formats vidéos et sonores ;
    Binaires complémentaires et facultatifs flvtool2 : (...)

Sur d’autres sites (8633)

  • App Preview, how to add blank audio track in iMovie ? or ffmpeg ?

    2 mars 2017, par toast

    Apple (customer support) is telling me that I must have audio in my app preview.

    My question is, how do I add a blank audio track to my mp4 app preview file ?

    Or is there some other way I can get my app preview approved without having audio ?

    There are many reasons why I dont want audio in my app preview. 1) I dont want to add some weird music in there thats not copyrighted. 2) I’m not going to narrate my app.

  • 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 ?

  • How to link libavcodec, libavformat in automake ?

    15 juillet 2014, par user2212461

    I am using the libavcodec and libavformat libraries from ffmpeg in a C++ project. Linking -lavcodec -lavformat with the g++ compiler works fine, but I’m not sure what goes wrong when I try to use the same code compiled in an automake project.

    Working fine :

    g++ -o test -D__STDC_CONSTANT_MACROS -lavcodec -lavformat test.cpp

    Not working Makefile.am :

    binaryname_LDFLAGS= -lavcodec -lavformat

    Error :

    ....
    
/usr/lib/gcc/i686-linux-gnu/4.6/../../../i386-linux-gnu/libavformat.a(nut.o):function 
    
ff_nut_add_sp: error: undefined reference to 'av_tree_node_size'
    
/usr/lib/gcc/i686-linux-gnu/4.6/../../../i386-linux-gnu/libavformat.a(nut.o):function 
    
ff_nut_add_sp: error: undefined reference to 'av_tree_insert'
    
/usr/lib/gcc/i686-linux-gnu/4.6/../../../i386-linux-gnu/libavformat.a(nut.o):function 
    
ff_nut_free_sp: error: undefined reference to 'av_tree_enumerate'
    
/usr/lib/gcc/i686-linux-gnu/4.6/../../../i386-linux-gnu/libavformat.a(nut.o):function 
    
ff_nut_free_sp: error: undefined reference to 'av_tree_destroy'
    
/usr/lib/gcc/i686-linux-gnu/4.6/../../../i386-linux-gnu/libavformat.a(rtp.o):function 
    
ff_rtp_get_payload_type: error: undefined reference to 'av_opt_get_int'
    
...

    also not working :

    LDFLAGS=...-lavcodec -lavformat

    Error :

    src/dsp/audioDecoder.cpp:99: error: undefined reference to 'av_register_all'
    
src/dsp/audioDecoder.cpp:101: error: undefined reference to 'avcodec_find_decoder'
    
src/dsp/audioDecoder.cpp:109: error: undefined reference to 'avcodec_alloc_context'
    
src/dsp/audioDecoder.cpp:120: error: undefined reference to 'avcodec_open2'
    
src/dsp/audioDecoder.cpp:125: error: undefined reference to 'av_init_packet'
    
src/dsp/audioDecoder.cpp:188: error: undefined reference to 'avcodec_decode_audio3'

    In the second case, if I don’t set any linker, the include header cannot be found so the linker seems somehow recognized.

    make V=1 returns :

    make  all-am
    
make[1]: Go to '/path/to/trunk'
    
/bin/bash ./libtool --tag=CXX   --mode=link g++  -O2 -lrt -D__STDC_CONSTANT_MACROS   -o binaryname progsrc/binaryname/binaryname-binaryname.o    -lm  -lpthread -ldl -lmygeneratedlibrary
    
libtool: link: g++ -O2 -D__STDC_CONSTANT_MACROS -o binaryname progsrc/binaryname/binaryname-binaryname.o  -lm /path/to//trunk/.libs/lmygeneratedlibrary.a -lrt -lavutil -lavcodec -lavformat -lpthread -ldl
    
make[1]: Leave '/path/to/trunk'

    What am I doing wrong here ?

Navigation