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The pirate bay depuis la Belgique
1er avril 2013, par
Mis à jour : Avril 2013
Langue : français
Type : Image
Autres articles (43)
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Publier sur MédiaSpip
13 juin 2013Puis-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 -
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 : (...) -
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 (...)
Sur d’autres sites (5172)
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JSmpeg is not playing audio from websocket stream
5 juin 2023, par NikI am trying to stream RTSP to web browser using ffmpeg through web socket relay written in node js taken from https://github.com/phoboslab/jsmpeg , and on the browser i am using JSMpeg to display the RTSP stream, the video is playing fine, but audio is not playing,


The ffmpeg command :


ffmpeg -rtsp_transport tcp -i rtsp://wowzaec2demo.streamlock.net/vod/mp4:BigBuckBunny_115k.mp4 
 -f mpegts -c:v mpeg1video -c:a mp2 http://127.0.0.1:8081/stream_from_ffmpeg/


The node js web socket relay :


// Use the websocket-relay to serve a raw MPEG-TS over WebSockets. You can use
// ffmpeg to feed the relay. ffmpeg -> websocket-relay -> browser
// Example:
// node websocket-relay yoursecret 8081 8082
// ffmpeg -i <some input="input"> -f mpegts http://localhost:8081/yoursecret

var fs = require('fs'),
 http = require('http'),
 WebSocket = require('ws');

if (process.argv.length < 3) {
 console.log(
 'Usage: \n' +
 'node websocket-relay.js <secret> [ ]'
 );
 process.exit();
}

var STREAM_SECRET = process.argv[2],
 STREAM_PORT = process.argv[3] || 8081,
 WEBSOCKET_PORT = process.argv[4] || 8082,
 RECORD_STREAM = false;

// Websocket Server
var socketServer = new WebSocket.Server({port: WEBSOCKET_PORT, perMessageDeflate: false});
socketServer.connectionCount = 0;
socketServer.on('connection', function(socket, upgradeReq) {
 socketServer.connectionCount++;
 console.log(
 'New WebSocket Connection: ',
 (upgradeReq || socket.upgradeReq).socket.remoteAddress,
 (upgradeReq || socket.upgradeReq).headers['user-agent'],
 '('+socketServer.connectionCount+' total)'
 );
 socket.on('close', function(code, message){
 socketServer.connectionCount--;
 console.log(
 'Disconnected WebSocket ('+socketServer.connectionCount+' total)'
 );
 });
});
socketServer.broadcast = function(data) {
 socketServer.clients.forEach(function each(client) {
 if (client.readyState === WebSocket.OPEN) {
 client.send(data);
 }
 });
};

// HTTP Server to accept incoming MPEG-TS Stream from ffmpeg
var streamServer = http.createServer( function(request, response) {
 var params = request.url.substr(1).split('/');

 if (params[0] !== STREAM_SECRET) {
 console.log(
 'Failed Stream Connection: '+ request.socket.remoteAddress + ':' +
 request.socket.remotePort + ' - wrong secret.'
 );
 response.end();
 }

 response.connection.setTimeout(0);
 console.log(
 'Stream Connected: ' +
 request.socket.remoteAddress + ':' +
 request.socket.remotePort
 );
 request.on('data', function(data){
 socketServer.broadcast(data);
 if (request.socket.recording) {
 request.socket.recording.write(data);
 }
 });
 request.on('end',function(){
 console.log('close');
 if (request.socket.recording) {
 request.socket.recording.close();
 }
 });

 // Record the stream to a local file?
 if (RECORD_STREAM) {
 var path = 'recordings/' + Date.now() + '.ts';
 request.socket.recording = fs.createWriteStream(path);
 }
})
// Keep the socket open for streaming
streamServer.headersTimeout = 0;
streamServer.listen(STREAM_PORT);

console.log('Listening for incoming MPEG-TS Stream on http://127.0.0.1:'+STREAM_PORT+'/<secret>');
console.log('Awaiting WebSocket connections on ws://127.0.0.1:'+WEBSOCKET_PORT+'/');
</secret></secret></some>

The front end code




 
 
 
 
 <code class="echappe-js"><script src='http://stackoverflow.com/feeds/tag/jsmpeg.min.js'></script>
 
 
 
 
 
<script>&#xA; let url;&#xA; let player;&#xA; let canvas = document.getElementById("video-canvas");&#xA; let ipAddr = "127.0.0.1:8082";&#xA; window.onload = async() => {&#xA; url = `ws://${ipAddr}`;&#xA; player = new JSMpeg.Player(url, { canvas: canvas, });&#xA; };&#xA;&#xA; </script>




The above code works fine and plays the video, but no audio is playing
Things I tried :


Changed the audio context state inside the player object from suspended to running


player.audioOut.context.onstatechange = async () => {
 console.log("Event triggered by audio");

 if (player.audioOut.context === "suspended") {
 await player.audioOut.context.resume();
 }
}


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libavcodec : how to encode with h264 codec ,with mp4 container using controllable frame rate and bitrate(through c code)
26 mai 2016, par musimbateI am trying to record the screen of a pc and encode the recorded frames using h264 encoder
and wrap them into a mp4 container.I want to do this because this super user link http://superuser.com/questions/300897/what-is-a-codec-e-g-divx-and-how-does-it-differ-from-a-file-format-e-g-mp/300997#300997 suggests it allows good trade-off between size and quality of the output file.The application I am working on should allow users to record a few hours of video and have the minimum output file size with decent quality.
The code I have cooked up so far allows me to record and save .mpg(container) files with the mpeg1video encoder
Running :
ffmpeg -i test.mpg
on the output file gives the following output :
[mpegvideo @ 028c7400] Estimating duration from bitrate, this may be inaccurate
Input #0, mpegvideo, from 'test.mpg':
Duration: 00:00:00.29, bitrate: 104857 kb/s
Stream #0:0: Video: mpeg1video, yuv420p(tv), 1366x768 [SAR 1:1 DAR 683:384], 104857 kb/s, 25 fps, 25 tbr, 1200k tbn, 25 tbcI have these settings for my output :
const char * filename="test.mpg";
int codec_id= AV_CODEC_ID_MPEG1VIDEO;
AVCodec *codec11;
AVCodecContext *outContext= NULL;
int got_output;
FILE *f;
AVPacket pkt;
uint8_t endcode[] = { 0, 0, 1, 0xb7 };
/* put sample parameters */
outContext->bit_rate = 400000;
/* resolution must be a multiple of two */
outContext->width=pCodecCtx->width;
outContext->height=pCodecCtx->height;
/* frames per second */
outContext->time_base.num=1;
outContext->time_base.den=25;
/* emit one intra frame every ten frames
* check frame pict_type before passing frame
* to encoder, if frame->pict_type is AV_PICTURE_TYPE_I
* then gop_size is ignored and the output of encoder
* will always be I frame irrespective to gop_size
*/
outContext->gop_size = 10;
outContext->max_b_frames = 1;
outContext->pix_fmt = AV_PIX_FMT_YUV420P;When I change int codec_id= AV_CODEC_ID_MPEG1VIDEO to int codec_id= AV_CODEC_ID_H264 i get a file that does not play with vlc.
I have read that writing the
uint8_t endcode[] = { 0, 0, 1, 0xb7 };array at the end of your file when finished encoding makes your file a legitimate mpeg file.It is written like this :
fwrite(endcode, 1, sizeof(endcode), f);
fclose(f);in my code. Should I do the same thing when I change my encoder to AV_CODEC_ID_H264 ?
I am capturing using gdi input like this :
AVDictionary* options = NULL;
//Set some options
//grabbing frame rate
av_dict_set(&options,"framerate","30",0);
AVInputFormat *ifmt=av_find_input_format("gdigrab");
if(avformat_open_input(&pFormatCtx,"desktop",ifmt,&options)!=0){
printf("Couldn't open input stream.\n");
return -1;
}I want to be able to modify my grabbing rate to optimize for the outptut file size
but When I change it to 20 for example I get a video that plays so fast.How do
I get a video that plays with normal speed with frames captured at 20 fps or any
lower frame rate value ?While recording I get the following output on the standard error output :
[gdigrab @ 00cdb8e0] Capturing whole desktop as 1366x768x32 at (0,0)
Input #0, gdigrab, from '(null)':
Duration: N/A, start: 1420718663.655713, bitrate: 1006131 kb/s
Stream #0:0: Video: bmp, bgra, 1366x768, 1006131 kb/s, 29.97 tbr, 1000k tbn, 29.97 tbc
[swscaler @ 00d24120] Warning: data is not aligned! This can lead to a speedloss
[mpeg1video @ 00cdd160] AVFrame.format is not set
[mpeg1video @ 00cdd160] AVFrame.width or height is not set
[mpeg1video @ 00cdd160] AVFrame.format is not set
[mpeg1video @ 00cdd160] AVFrame.width or height is not set
[mpeg1video @ 00cdd160] AVFrame.format is not setHow do I get rid of this error in my code ?
In summary :
1) How do I encode h264 video wrapped into mp4 container ?2) How do I capture at lower frame rates and still play
the encoded video at normal speed ?3) How do I set the format(and which format—depends on the codec ?)
and width and height info on the frames I write ?The code I am using in its entirety is shown below
extern "C"
{
#include "libavcodec/avcodec.h"
#include "libavformat/avformat.h"
#include "libswscale/swscale.h"
#include "libavdevice/avdevice.h"
#include <libavutil></libavutil>opt.h>
#include <libavutil></libavutil>channel_layout.h>
#include <libavutil></libavutil>common.h>
#include <libavutil></libavutil>imgutils.h>
#include <libavutil></libavutil>mathematics.h>
#include <libavutil></libavutil>samplefmt.h>
//SDL
#include "SDL.h"
#include "SDL_thread.h"
}
//Output YUV420P
#define OUTPUT_YUV420P 0
//'1' Use Dshow
//'0' Use GDIgrab
#define USE_DSHOW 0
int main(int argc, char* argv[])
{
//1.WE HAVE THE FORMAT CONTEXT
//THIS IS FROM THE DESKTOP GRAB STREAM.
AVFormatContext *pFormatCtx;
int i, videoindex;
AVCodecContext *pCodecCtx;
AVCodec *pCodec;
av_register_all();
avformat_network_init();
//ASSIGN STH TO THE FORMAT CONTEXT.
pFormatCtx = avformat_alloc_context();
//Register Device
avdevice_register_all();
//Windows
#ifdef _WIN32
#if USE_DSHOW
//Use dshow
//
//Need to Install screen-capture-recorder
//screen-capture-recorder
//Website: http://sourceforge.net/projects/screencapturer/
//
AVInputFormat *ifmt=av_find_input_format("dshow");
//if(avformat_open_input(&pFormatCtx,"video=screen-capture-recorder",ifmt,NULL)!=0){
if(avformat_open_input(&pFormatCtx,"video=UScreenCapture",ifmt,NULL)!=0){
printf("Couldn't open input stream.\n");
return -1;
}
#else
//Use gdigrab
AVDictionary* options = NULL;
//Set some options
//grabbing frame rate
av_dict_set(&options,"framerate","30",0);
//The distance from the left edge of the screen or desktop
//av_dict_set(&options,"offset_x","20",0);
//The distance from the top edge of the screen or desktop
//av_dict_set(&options,"offset_y","40",0);
//Video frame size. The default is to capture the full screen
//av_dict_set(&options,"video_size","640x480",0);
AVInputFormat *ifmt=av_find_input_format("gdigrab");
if(avformat_open_input(&pFormatCtx,"desktop",ifmt,&options)!=0){
printf("Couldn't open input stream.\n");
return -1;
}
#endif
#endif//FOR THE WIN32 THING.
if(avformat_find_stream_info(pFormatCtx,NULL)<0)
{
printf("Couldn't find stream information.\n");
return -1;
}
videoindex=-1;
for(i=0; inb_streams; i++)
if(pFormatCtx->streams[i]->codec->codec_type
==AVMEDIA_TYPE_VIDEO)
{
videoindex=i;
break;
}
if(videoindex==-1)
{
printf("Didn't find a video stream.\n");
return -1;
}
pCodecCtx=pFormatCtx->streams[videoindex]->codec;
pCodec=avcodec_find_decoder(pCodecCtx->codec_id);
if(pCodec==NULL)
{
printf("Codec not found.\n");
return -1;
}
if(avcodec_open2(pCodecCtx, pCodec,NULL)<0)
{
printf("Could not open codec.\n");
return -1;
}
//THIS IS WHERE YOU CONTROL THE FORMAT(THROUGH FRAMES).
AVFrame *pFrame;
pFrame=av_frame_alloc();
int ret, got_picture;
AVPacket *packet=(AVPacket *)av_malloc(sizeof(AVPacket));
//TRY TO INIT THE PACKET HERE
av_init_packet(packet);
//Output Information-----------------------------
printf("File Information---------------------\n");
av_dump_format(pFormatCtx,0,NULL,0);
printf("-------------------------------------------------\n");
//<<--FOR WRITING MPG FILES
//<<--START:PREPARE TO WRITE YOUR MPG FILE.
const char * filename="test.mpg";
int codec_id= AV_CODEC_ID_MPEG1VIDEO;
AVCodec *codec11;
AVCodecContext *outContext= NULL;
int got_output;
FILE *f;
AVPacket pkt;
uint8_t endcode[] = { 0, 0, 1, 0xb7 };
printf("Encode video file %s\n", filename);
/* find the mpeg1 video encoder */
codec11 = avcodec_find_encoder((AVCodecID)codec_id);
if (!codec11) {
fprintf(stderr, "Codec not found\n");
exit(1);
}
outContext = avcodec_alloc_context3(codec11);
if (!outContext) {
fprintf(stderr, "Could not allocate video codec context\n");
exit(1);
}
/* put sample parameters */
outContext->bit_rate = 400000;
/* resolution must be a multiple of two */
outContext->width=pCodecCtx->width;
outContext->height=pCodecCtx->height;
/* frames per second */
outContext->time_base.num=1;
outContext->time_base.den=25;
/* emit one intra frame every ten frames
* check frame pict_type before passing frame
* to encoder, if frame->pict_type is AV_PICTURE_TYPE_I
* then gop_size is ignored and the output of encoder
* will always be I frame irrespective to gop_size
*/
outContext->gop_size = 10;
outContext->max_b_frames = 1;
outContext->pix_fmt = AV_PIX_FMT_YUV420P;
if (codec_id == AV_CODEC_ID_H264)
av_opt_set(outContext->priv_data, "preset", "slow", 0);
/* open it */
if (avcodec_open2(outContext, codec11, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
exit(1);
}
f = fopen(filename, "wb");
if (!f) {
fprintf(stderr, "Could not open %s\n", filename);
exit(1);
}
AVFrame *outframe = av_frame_alloc();
int nbytes = avpicture_get_size(outContext->pix_fmt,
outContext->width,
outContext->height);
uint8_t* outbuffer = (uint8_t*)av_malloc(nbytes);
//ASSOCIATE THE FRAME TO THE ALLOCATED BUFFER.
avpicture_fill((AVPicture*)outframe, outbuffer,
AV_PIX_FMT_YUV420P,
outContext->width, outContext->height);
SwsContext* swsCtx_ ;
swsCtx_= sws_getContext(pCodecCtx->width,
pCodecCtx->height,
pCodecCtx->pix_fmt,
outContext->width, outContext->height,
outContext->pix_fmt,
SWS_BICUBIC, NULL, NULL, NULL);
//HERE WE START PULLING PACKETS FROM THE SPECIFIED FORMAT CONTEXT.
while(av_read_frame(pFormatCtx, packet)>=0)
{
if(packet->stream_index==videoindex)
{
ret= avcodec_decode_video2(pCodecCtx,
pFrame,
&got_picture,packet );
if(ret < 0)
{
printf("Decode Error.\n");
return -1;
}
if(got_picture)
{
sws_scale(swsCtx_, pFrame->data, pFrame->linesize,
0, pCodecCtx->height, outframe->data,
outframe->linesize);
av_init_packet(&pkt);
pkt.data = NULL; // packet data will be allocated by the encoder
pkt.size = 0;
ret = avcodec_encode_video2(outContext, &pkt, outframe, &got_output);
if (ret < 0) {
fprintf(stderr, "Error encoding frame\n");
exit(1);
}
if (got_output) {
printf("Write frame %3d (size=%5d)\n", i, pkt.size);
fwrite(pkt.data, 1, pkt.size, f);
av_free_packet(&pkt);
}
}
}
av_free_packet(packet);
}//THE LOOP TO PULL PACKETS FROM THE FORMAT CONTEXT ENDS HERE.
//
/* get the delayed frames */
for (got_output = 1; got_output; i++) {
//fflush(stdout);
ret = avcodec_encode_video2(outContext, &pkt, NULL, &got_output);
if (ret < 0) {
fprintf(stderr, "Error encoding frame\n");
exit(1);
}
if (got_output) {
printf("Write frame %3d (size=%5d)\n", i, pkt.size);
fwrite(pkt.data, 1, pkt.size, f);
av_free_packet(&pkt);
}
}
/* add sequence end code to have a real mpeg file */
fwrite(endcode, 1, sizeof(endcode), f);
fclose(f);
avcodec_close(outContext);
av_free(outContext);
//av_freep(&frame->data[0]);
//av_frame_free(&frame);
//THIS WAS ADDED LATER
av_free(outbuffer);
avcodec_close(pCodecCtx);
avformat_close_input(&pFormatCtx);
return 0;
}Thank you for your time.
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ffmpeg : Unrecognized option 'alpha_quality'
16 avril 2024, par László MondaI want to make transparent videos work in Safari, which doesn't support WebM for this purpose but only H265 with alpha transparency.


According to this post, I used Shutter Encoder, but only some of its versions work for this purpose on Mac.


Instead of using Shutter Encoder on Mac, I want to use ffmpeg on my Linux PC. Shutter Encoder uses the following command in the background :


ffmpeg -threads 0 -hwaccel none -i input.mov -c:v hevc_videotoolbox -alpha_quality 1 -b:v 1000k -profile:v main -level 5.2 -map v:0 -an -pix_fmt yuva420p -sws_flags bicubic -tag:v hvc1 -metadata creation_time=2024-04-14T14:53:08.734684Z -y output.mp4

which yields the following output on my PC :


ffmpeg version 4.4.2-0ubuntu0.22.04.1 Copyright (c) 2000-2021 the FFmpeg developers
 built with gcc 11 (Ubuntu 11.2.0-19ubuntu1)
 configuration: --prefix=/usr --extra-version=0ubuntu0.22.04.1 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --arch=amd64 --enable-gpl --disable-stripping --enable-gnutls --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libcodec2 --enable-libdav1d --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librabbitmq --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzimg --enable-libzmq --enable-libzvbi --enable-lv2 --enable-omx --enable-openal --enable-opencl --enable-opengl --enable-sdl2 --enable-pocketsphinx --enable-librsvg --enable-libmfx --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libx264 --enable-shared
 WARNING: library configuration mismatch
 avcodec configuration: --prefix=/usr --extra-version=0ubuntu0.22.04.1 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --arch=amd64 --enable-gpl --disable-stripping --enable-gnutls --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libcodec2 --enable-libdav1d --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librabbitmq --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzimg --enable-libzmq --enable-libzvbi --enable-lv2 --enable-omx --enable-openal --enable-opencl --enable-opengl --enable-sdl2 --enable-pocketsphinx --enable-librsvg --enable-libmfx --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libx264 --enable-shared --enable-version3 --disable-doc --disable-programs --enable-libaribb24 --enable-libopencore_amrnb --enable-libopencore_amrwb --enable-libtesseract --enable-libvo_amrwbenc --enable-libsmbclient
 libavutil 56. 70.100 / 56. 70.100
 libavcodec 58.134.100 / 58.134.100
 libavformat 58. 76.100 / 58. 76.100
 libavdevice 58. 13.100 / 58. 13.100
 libavfilter 7.110.100 / 7.110.100
 libswscale 5. 9.100 / 5. 9.100
 libswresample 3. 9.100 / 3. 9.100
 libpostproc 55. 9.100 / 55. 9.100
Unrecognized option 'alpha_quality'.
Error splitting the argument list: Option not found


When googling for "Unrecognized option 'alpha_quality'.", there are no results, which I find very odd.


What's going on, and how can I make ffmpeg work for this purpose without Shutter Encoder ?


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