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MediaSPIP version 0.1 Beta
16 avril 2011, parMediaSPIP 0.1 beta est la première version de MediaSPIP décrétée comme "utilisable".
Le fichier zip ici présent contient uniquement les sources de MediaSPIP en version standalone.
Pour avoir une installation fonctionnelle, il est nécessaire d’installer manuellement l’ensemble des dépendances logicielles sur le serveur.
Si vous souhaitez utiliser cette archive pour une installation en mode ferme, il vous faudra également procéder à d’autres modifications (...) -
Personnaliser en ajoutant son logo, sa bannière ou son image de fond
5 septembre 2013, parCertains thèmes prennent en compte trois éléments de personnalisation : l’ajout d’un logo ; l’ajout d’une bannière l’ajout d’une image de fond ;
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Amélioration de la version de base
13 septembre 2013Jolie sélection multiple
Le plugin Chosen permet d’améliorer l’ergonomie des champs de sélection multiple. Voir les deux images suivantes pour comparer.
Il suffit pour cela d’activer le plugin Chosen (Configuration générale du site > Gestion des plugins), puis de configurer le plugin (Les squelettes > Chosen) en activant l’utilisation de Chosen dans le site public et en spécifiant les éléments de formulaires à améliorer, par exemple select[multiple] pour les listes à sélection multiple (...)
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ffmpeg channels don't work (PHP)
15 avril 2016, par Y.SaadI have a lot of channels with " ffmpeg " (4 channels) that start automatically, I create a code for show the first channel and after 5 seconds show the second .. Etc All things as right but I have a small problem the seconds channel doesn’t start automatically, I need to make stop to the first channel
Code for 4 channel work 100% but without function for show first channel and after 5 seconds show the second ... Etc
<?php
ffmpeg -i http://clay24.webhop.net:8000/live/mario/mario/13.ts -i http://clay24.webhop.net:8000/live/mario/mario/12.ts -i http://clay24.webhop.net:8000/live/mario/mario/10.ts -map 0 -c:a aac -b:a 64k -strict -2 -preset fast -crf 25 -vcodec libx264 -f flv rtmp://178.33.231.108:1989/mylive/1 -map 1 -c:a aac -b:a 64k -strict -2 -preset fast -crf 25 -vcodec libx264 -f flv rtmp://178.33.231.108:1989/mylive/21 -map 2 -c:a aac -b:a 64k -strict -2 -preset fast -crf 25 -vcodec libx264 -f flv rtmp://178.33.231.108:1989/mylive/24
?>the second code with functions
<?php
echo'
<code class="echappe-js"><script><br />
<br />
window.addEventListener("load", function() {<br />
<br />
var urls = iframes = [ "'; ffmpeg -i http://mygameravatar.zapto.org:43666/live/test1/test2/81.ts -c:a aac -b:a 64k -preset fast -crf 25 -vcodec libx264 -f flv rtmp://178.33.231.108:1989/mylive/2 <br />
echo '];<br />
<br />
var iframes = document.querySelectorAll("div");<br />
<br />
var n = 0;<br />
<br />
var interval = setInterval(function() {<br />
<br />
iframes[++n].src = urls[n - 1];<br />
iframes[n].style.display = "block";<br />
console.log(n);<br />
if (n === iframes.length -1) {<br />
clearInterval(interval);<br />
console.log("all iframes loaded")<br />
}<br />
<br />
}, 5000)<br />
<br />
})<br />
</script>’ ;ffmpeg -i http://mygameravatar.zapto.org:43666/live/test1/test2/297.ts -c:a aac -b:a 64k -preset fast -crf 25 -vcodec libx264 -f flv rtmp ://178.33.231.108:1989/mylive/1
echo ’
’ ;
?> -
How to stream audio from ffserver
22 mai 2019, par DoroI trying to stream 2 files -
1.mkvwithout audio (which streaming ok) and2.mkvwith audio encoded with Vorbis codec which i can’t stream. For encoding I usedffmpeg -i 2.mp4 -strict -2 -c:a vorbis ex.mkvAnd it playing ok with
ffplayServer log :
Fri May 17 00:49:08 2019 Opening feed file '1.mkv' for stream 'test1-rtsp'
Fri May 17 00:49:08 2019 [matroska,webm @ 0x200746c0]Unknown entry 0x55B0
Thu Dec 14 21:35:00 1950 [h264 @ 0x2007dcc0]gray chroma
Fri May 17 00:49:08 2019 [h264 @ 0x2007dcc0]error while decoding MB 18 1, bytestream 1989
Fri May 17 00:49:08 2019 [h264 @ 0x2007dcc0]concealing 432 DC, 432 AC, 432 MV errors in I frame
Fri May 17 00:49:08 2019 Opening feed file '2.mkv' for stream 'test2-rtsp'
Fri May 17 00:49:08 2019 [matroska,webm @ 0x200746c0]Unknown entry 0x55B0
Fri May 17 00:49:08 2019 FFserver started.
Fri May 17 00:49:25 2019 [matroska,webm @ 0x20080de0]Unknown entry 0x55B0
Fri May 17 00:49:25 2019 127.0.0.1:33582 - - "PLAY test2-rtsp/streamid=0 RTP/UDP"
Fri May 17 00:49:25 2019 127.0.0.1 - - [SETUP] "rtsp://127.0.0.1:7654/test2-rtsp/ RTSP/1.0" 200 2553Client log :
Bad packed header lengths (30,0,1250,2673)
[udp @ 00000236f6318500] 'circular_buffer_size' option was set but it is not supported on this build (pthread support is required)
[udp @ 00000236f63185c0] 'circular_buffer_size' option was set but it is not supported on this build (pthread support is required)
[udp @ 00000236f633dc40] 'circular_buffer_size' option was set but it is not supported on this build (pthread support is required)
[udp @ 00000236f634df00] 'circular_buffer_size' option was set but it is not supported on this build (pthread support is required)
[rtsp @ 00000236f63153c0] method SETUP failed: 503 Service Unavailable
rtsp://127.0.0.1:7654/test2-rtsp: Server returned 5XX Server Error replyConfigure ffserver file :
Port 8090
BindAddress 0.0.0.0
MaxHTTPConnections 2000
MaxClients 1000
MaxBandwidth 500000
CustomLog -
NoDaemon
RTSPPort 7654
RTSPBindAddress 0.0.0.0
<stream>
Format rtp
File "1.mkv"
</stream>
<stream>
Format rtp
Strict -2
AudioCodec vorbis
File "2.mkv"
</stream> -
why AVFrame pts value doesn't affect bitrate of frames ?
31 janvier 2021, par fsdfhdsjkhfjkdsI'm trying code realtime screen sharing I noticed H264 codec doesn't do constant time encode for every frame. That causes being not able to encode exact same amount frame rate with context->time_base. When we encode less frames per second than time_base bitrate of second becomes lower than what we set.


I modified libav's example encode code and put 1/1000 time base and supply it with only 10 frame. I increase frame->pts related with time_base but bitrates still stay at low.


For results I just change
context->time_baseto 1, 1000, 1, 10 etc

1/1000 time base (as sum 1989 bytes per second) :


encoded frame 0 (size=1169)
encoded frame 100 (size=95)
encoded frame 200 (size=92)
encoded frame 300 (size=102)
encoded frame 400 (size=90)
encoded frame 500 (size=90)
encoded frame 600 (size=90)
encoded frame 700 (size=83)
encoded frame 800 (size=95)
encoded frame 900 (size=83)


1/10 time base (as sum 95324 bytes per second) :


encoded frame 0 (size=14187)
encoded frame 1 (size=6053)
encoded frame 2 (size=8530)
encoded frame 3 (size=9277)
encoded frame 4 (size=9508)
encoded frame 5 (size=11163)
encoded frame 6 (size=9685)
encoded frame 7 (size=9346)
encoded frame 8 (size=7662)
encoded frame 9 (size=9913)


code :


#include 
#include 
#include <libavcodec></libavcodec>avcodec.h>

static void encode(AVCodecContext *context, AVFrame *frame, AVPacket *pkt, FILE *outfile){
 int ret = avcodec_send_frame(context, frame);
 assert(ret >= 0);
 while(ret >= 0){
 ret = avcodec_receive_packet(context, pkt);
 if(ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
 return;
 else if(ret < 0)
 assert(0);
 printf("encoded frame %lld (size=%d)\n", pkt->pts, pkt->size);
 fwrite(pkt->data, 1, pkt->size, outfile);
 av_packet_unref(pkt);
 }
}

int main(int argc, char **argv){
 if(argc <= 1){
 fprintf(stderr, "Usage: %s <output file="file">\n", argv[0]);
 exit(0);
 }
 av_log_set_level(AV_LOG_QUIET);
 const char *filename = argv[1];
 const AVCodec *codec = avcodec_find_encoder(AV_CODEC_ID_H264);
 assert(codec);
 AVCodecContext *context = avcodec_alloc_context3(codec);
 assert(context);
 AVFrame *frame = av_frame_alloc();
 assert(frame);
 AVPacket *pkt = av_packet_alloc();
 assert(pkt);
 context->bit_rate = 800000;
 context->width = 1280;
 context->height = 720;
 context->time_base = (AVRational){1, 1000};
 context->pix_fmt = AV_PIX_FMT_YUV420P;
 AVDictionary *dict = 0;
 assert(av_dict_set(&dict, "preset", "veryfast", 0) >= 0);
 assert(av_dict_set(&dict, "tune", "zerolatency", 0) >= 0);
 assert(avcodec_open2(context, codec, &dict) >= 0);
 FILE *f = fopen(filename, "wb");
 assert(f);
 frame->format = context->pix_fmt;
 frame->width = context->width;
 frame->height = context->height;
 assert(av_frame_get_buffer(frame, 0) >= 0);
 for(int i = 0; i < 10; i++){
 for(int y = 0; y < context->height; y++){
 for(int x = 0; x < context->width; x++){
 frame->data[0][y * frame->linesize[0] + x] = x + y + i * 3;
 }
 }
 for(int y = 0; y < context->height / 2; y++){
 for(int x = 0; x < context->width / 2; x++){
 frame->data[1][y * frame->linesize[1] + x] = 128 + y + i * 2;
 frame->data[2][y * frame->linesize[2] + x] = 64 + x + i * 5;
 }
 }
 frame->pts = i * (context->time_base.den / 10);
 encode(context, frame, pkt, f);
 }
 fclose(f);
 avcodec_free_context(&context);
 av_frame_free(&frame);
 av_packet_free(&pkt);
 return 0;
}
</output>

How we can keep right bitrate with different time_base than frame rate ?


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