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HTML5 audio and video support
13 avril 2011, parMediaSPIP 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 2011MediaSPIP 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 (...) -
Librairies et binaires spécifiques au traitement vidéo et sonore
31 janvier 2010, parLes 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 (8545)
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Nginx RTMP Pull to HLS Streaming
5 avril 2023, par Nathaniel AndersonI've followed this guide on setting up RTMP to HLS streaming - https://web.archive.org/web/20221205201139/https://docs.peer5.com/guides/setting-up-hls-live-streaming-server-using-nginx/


RTMP streaming works just fine but for some reason I can't get HLS to link with the already existing RTMP server. I'm using OBS to stream to RTMP and it's set to be on x264 and as far as I know the default codec for audio is AAC so I'm not sure why it's not picking it up.


Current Nginx.conf


worker_processes auto;
events {
 worker_connections 1024;
}

# RTMP configuration
rtmp {
 server {
 listen 1935; # Listen on standard RTMP port
 chunk_size 4000;

# Define the Application
 application show {
 live on;
 pull rtmp://localhost:1935/stream/test;
 # Turn on HLS
 hls on;
 hls_path /mnt/hls/;
 hls_fragment 3;
 hls_playlist_length 60;
 # disable consuming the stream from nginx as rtmp
 deny play all;
 }

 # RTMP video on demand for mp4 files
 application vod {
 play /mnt/mp4s;
 }

 # RTMP stream using OBS
 application stream {
 live on;
 }

 }
}

http {
 sendfile off;
 tcp_nopush on;
 aio on;
 directio 512;
 default_type application/octet-stream;

 server {
 listen 8080;

 location / {
 # Disable cache
 add_header 'Cache-Control' 'no-cache';

 # CORS setup
 add_header 'Access-Control-Allow-Origin' '*' always;
 add_header 'Access-Control-Expose-Headers' 'Content-Length';

 # allow CORS preflight requests
 if ($request_method = 'OPTIONS') {
 add_header 'Access-Control-Allow-Origin' '*';
 add_header 'Access-Control-Max-Age' 1728000;
 add_header 'Content-Type' 'text/plain charset=UTF-8';
 add_header 'Content-Length' 0;
 return 204;
 }

 types {
 application/dash+xml mpd;
 application/vnd.apple.mpegurl m3u8;
 video/mp2t ts;
 }

 root /mnt/;
 }
 }
}



I've tried changing the rtmp link I have in that config too to both internal and external IP's since it can be reached. I tested watching the rtmp from another computer on the network to confirm it was functional. I wanted to avoid transcoding with ffmpeg since the server doesn't have that kind of power.


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lavc/h263dsp : R-V V {h,v}_loop_filter
19 mai 2024, par Rémi Denis-Courmontlavc/h263dsp : R-V V h,v_loop_filter
Since the horizontal and vertical filters are identical except for a
transposition, this uses a common subprocedure with an ad-hoc ABI.
To preserve return-address stack prediction, a link register has to be
used (c.f. the "Control Transfer Instructions" from the
RISC-V ISA Manual). The alternate/temporary link register T0 is used
here, so that the normal RA is preserved (something Arm cannot do !).To load the strength value based on `qscale`, the shortest possible
and PIC-compatible sequence is used : AUIPC ; ADD ; LBU. The classic
LLA ; ADD ; LBU sequence would add one more instruction since LLA is a
convenience alias for AUIPC ; ADDI. To ensure that this trick works,
relocation relaxation is disabled.To implement the two signed divisions by a power of two toward zero :
(x / (1 << SHIFT))
the code relies on the small range of integers involved, computing :
(x + (x >> (16 - SHIFT))) >> SHIFT
rather than the more general :
(x + ((x >> (16 - 1)) & ((1 << SHIFT) - 1))) >> SHIFT
Thus one ANDI instruction is avoided.T-Head C908 :
h263dsp.h_loop_filter_c : 228.2
h263dsp.h_loop_filter_rvv_i32 : 144.0
h263dsp.v_loop_filter_c : 242.7
h263dsp.v_loop_filter_rvv_i32 : 114.0
(C is probably worse in real use due to less predictible branches.) -
FFMPEG tee muxer giving "Output file #0 does not contain any stream"
31 août 2020, par Giorgi AptsiauriI am trying to create two streams : one is mpegts UDP stream another - rtmp to Twitch servers.


This command works :


ffmpeg -threads:v 2 -threads:a 16 -filter_threads 2 -thread_queue_size 16 -y \
 -f dshow -video_size 1920x1080 -pixel_format uyvy422 -framerate 25 -rtbufsize 500M -i video="Decklink Video Capture" \
 -f dshow -rtbufsize 100M -i audio="Decklink Audio Capture" \
 -preset ultrafast -c:v libx264 -tune zerolatency -b:v 900k -map 0:v:0 -f mpegts udp://127.0.0.1:5555 \ 
 -pix_fmt yuv420p -c:v libx264 -crf 20 -tune zerolatency -f flv rtmp://live-fra05.twitch.tv/app/stream_key



But it requires double the encoding CPU power.


So, following this, I rewrote the command like this :


ffmpeg -threads:v 2 -threads:a 16 -filter_threads 2 -thread_queue_size 16 -y \
 -f dshow -video_size 1920x1080 -pixel_format uyvy422 -framerate 25 -rtbufsize 500M -i video="Decklink Video Capture" \
 -f dshow -rtbufsize 100M -i audio="Decklink Audio Capture" \
 -preset ultrafast -c:v libx264 -tune zerolatency -b:v 900k \
 -f tee "[select=\'0:v:0\':f=mpegts]udp://127.0.0.1:5555|[select=\'0:v:0,1:a:0\':f=flv]rtmp://live-fra05.twitch.tv/app/stream_key"



By writing
-f tee "[select=\'0:v:0\':f=mpegts]udp://127.0.0.1:5555|[select=\'0:v:0,1:a:0\':f=flv]rtmp://live-fra05.twitch.tv/app/stream_key"
, I mean :

- 

- create UDP stream at udp ://127.0.0.1:5555 and only include video stream from "Decklink Video Capture"
- create RTMP stream where we include the same video stream as above and also the audio stream from "Decklink Audio Capture"






I get the error message :


Output file #0 does not contain any stream



How do I fix this ? I assume I made a mistake in the command.