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Support de tous types de médias
10 avril 2011Contrairement à beaucoup de logiciels et autres plate-formes modernes de partage de documents, MediaSPIP a l’ambition de gérer un maximum de formats de documents différents qu’ils soient de type : images (png, gif, jpg, bmp et autres...) ; audio (MP3, Ogg, Wav et autres...) ; vidéo (Avi, MP4, Ogv, mpg, mov, wmv et autres...) ; contenu textuel, code ou autres (open office, microsoft office (tableur, présentation), web (html, css), LaTeX, Google Earth) (...)
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Les tâches Cron régulières de la ferme
1er décembre 2010, parLa gestion de la ferme passe par l’exécution à intervalle régulier de plusieurs tâches répétitives dites Cron.
Le super Cron (gestion_mutu_super_cron)
Cette tâche, planifiée chaque minute, a pour simple effet d’appeler le Cron de l’ensemble des instances de la mutualisation régulièrement. Couplée avec un Cron système sur le site central de la mutualisation, cela permet de simplement générer des visites régulières sur les différents sites et éviter que les tâches des sites peu visités soient trop (...) -
Les statuts des instances de mutualisation
13 mars 2010, parPour des raisons de compatibilité générale du plugin de gestion de mutualisations avec les fonctions originales de SPIP, les statuts des instances sont les mêmes que pour tout autre objets (articles...), seuls leurs noms dans l’interface change quelque peu.
Les différents statuts possibles sont : prepa (demandé) qui correspond à une instance demandée par un utilisateur. Si le site a déjà été créé par le passé, il est passé en mode désactivé. publie (validé) qui correspond à une instance validée par un (...)
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RTMP Broadcast packet body structure for Twitch
22 mai 2018, par DobbyI’m currently working on a project similar to OBS, where I’m capturing screen data, encoding it with the x264 library, and then broadcasting it to a twitch server.
Currently, the servers are accepting the data, but no video is being played - it buffers for a moment, then returns an error code "2000 : network error"
Like OBS Classic, I’m dividing each NAL provided by x264 by its type, and then making changes to each
int frame_size = x264_encoder_encode(encoder, &nals, &num_nals, &pic_in, &pic_out);
//sort the NAL's into their types and make necessary adjustments
int timeOffset = int(pic_out.i_pts - pic_out.i_dts);
timeOffset = htonl(timeOffset);//host to network translation, ensure the bytes are in the right format
BYTE *timeOffsetAddr = ((BYTE*)&timeOffset) + 1;
videoSection sect;
bool foundFrame = false;
uint8_t * spsPayload = NULL;
int spsSize = 0;
for (int i = 0; i/std::cout << "VideoEncoder: EncodedImages Size: " << encodedImages->size() << std::endl;
x264_nal_t &nal = nals[i];
//std::cout << "NAL is:" << nal.i_type << std::endl;
//need to account for pps/sps, seems to always be the first frame sent
if (nal.i_type == NAL_SPS) {
spsSize = nal.i_payload;
spsPayload = (uint8_t*)malloc(spsSize);
memcpy(spsPayload, nal.p_payload, spsSize);
} else if (nal.i_type == NAL_PPS){
//pps always happens after sps
if (spsPayload == NULL) {
std::cout << "VideoEncoder: critical error, sps not set" << std::endl;
}
uint8_t * payload = (uint8_t*)malloc(nal.i_payload + spsSize);
memcpy(payload, spsPayload, spsSize);
memcpy(payload, nal.p_payload + spsSize, nal.i_payload);
sect = { nal.i_payload + spsSize, payload, nal.i_type };
encodedImages->push(sect);
} else if (nal.i_type == NAL_SEI || nal.i_type == NAL_FILLER) {
//these need some bytes at the start removed
BYTE *skip = nal.p_payload;
while (*(skip++) != 0x1);
int skipBytes = (int)(skip - nal.p_payload);
int newPayloadSize = (nal.i_payload - skipBytes);
uint8_t * payload = (uint8_t*)malloc(newPayloadSize);
memcpy(payload, nal.p_payload + skipBytes, newPayloadSize);
sect = { newPayloadSize, payload, nal.i_type };
encodedImages->push(sect);
} else if (nal.i_type == NAL_SLICE_IDR || nal.i_type == NAL_SLICE) {
//these packets need an additional section at the start
BYTE *skip = nal.p_payload;
while (*(skip++) != 0x1);
int skipBytes = (int)(skip - nal.p_payload);
std::vector<byte> bodyData;
if (!foundFrame) {
if (nal.i_type == NAL_SLICE_IDR) { bodyData.push_back(0x17); } else { bodyData.push_back(0x27); } //add a 17 or a 27 as appropriate
bodyData.push_back(1);
bodyData.push_back(*timeOffsetAddr);
foundFrame = true;
}
//put into the payload the bodyData followed by the nal payload
uint8_t * bodyDataPayload = (uint8_t*)malloc(bodyData.size());
memcpy(bodyDataPayload, bodyData.data(), bodyData.size() * sizeof(BYTE));
int newPayloadSize = (nal.i_payload - skipBytes);
uint8_t * payload = (uint8_t*)malloc(newPayloadSize + sizeof(bodyDataPayload));
memcpy(payload, bodyDataPayload, sizeof(bodyDataPayload));
memcpy(payload + sizeof(bodyDataPayload), nal.p_payload + skipBytes, newPayloadSize);
int totalSize = newPayloadSize + sizeof(bodyDataPayload);
sect = { totalSize, payload, nal.i_type };
encodedImages->push(sect);
} else {
std::cout << "VideoEncoder: Nal type did not match expected" << std::endl;
continue;
}
}
</byte>The NAL payload data is then put into a struct, VideoSection, in a queue buffer
//used to transfer encoded data
struct videoSection {
int frameSize;
uint8_t* payload;
int type;
};After which it is picked up by the broadcaster, a few more changes are made, and then I call rtmp_send()
videoSection sect = encodedImages->front();
encodedImages->pop();
//std::cout << "Broadcaster: Frame Size: " << sect.frameSize << std::endl;
//two methods of sending RTMP data, _sendpacket and _write. Using sendpacket for greater control
RTMPPacket * packet;
unsigned char* buf = (unsigned char*)sect.payload;
int type = buf[0]&0x1f; //I believe &0x1f sets a 32bit limit
int len = sect.frameSize;
long timeOffset = GetTickCount() - rtmp_start_time;
//assign space packet will need
packet = (RTMPPacket *)malloc(sizeof(RTMPPacket)+RTMP_MAX_HEADER_SIZE + len + 9);
memset(packet, 0, sizeof(RTMPPacket) + RTMP_MAX_HEADER_SIZE);
packet->m_body = (char *)packet + sizeof(RTMPPacket) + RTMP_MAX_HEADER_SIZE;
packet->m_nBodySize = len + 9;
//std::cout << "Broadcaster: Packet Size: " << sizeof(RTMPPacket) + RTMP_MAX_HEADER_SIZE + len + 9 << std::endl;
//std::cout << "Broadcaster: Packet Body Size: " << len + 9 << std::endl;
//set body to point to the packetbody
unsigned char *body = (unsigned char *)packet->m_body;
memset(body, 0, len + 9);
//NAL_SLICE_IDR represents keyframe
//first element determines packet type
body[0] = 0x27;//inter-frame h.264
if (sect.type == NAL_SLICE_IDR) {
body[0] = 0x17; //h.264 codec id
}
//-------------------------------------------------------------------------------
//this section taken from https://stackoverflow.com/questions/25031759/using-x264-and-librtmp-to-send-live-camera-frame-but-the-flash-cant-show
//in an effort to understand packet format. it does not resolve my previous issues formatting the data for twitch to play it
//sets body to be NAL unit
body[1] = 0x01;
body[2] = 0x00;
body[3] = 0x00;
body[4] = 0x00;
//>> is a shift right
//shift len to the right, and AND it
/*body[5] = (len >> 24) & 0xff;
body[6] = (len >> 16) & 0xff;
body[7] = (len >> 8) & 0xff;
body[8] = (len) & 0xff;*/
//end code sourced from https://stackoverflow.com/questions/25031759/using-x264-and-librtmp-to-send-live-camera-frame-but-the-flash-cant-show
//-------------------------------------------------------------------------------
//copy from buffer into rest of body
memcpy(&body[9], buf, len);
//DEBUG
//save individual packet body to a file with name rtmp[packetnum]
//determine why some packets do not have 0x27 or 0x17 at the start
//still happening, makes no sense given the above code
/*std::string fileLocation = "rtmp" + std::to_string(packCount++);
std::cout << fileLocation << std::endl;
const char * charConversion = fileLocation.c_str();
FILE* saveFile = NULL;
saveFile = fopen(charConversion, "w+b");//open as write and binary
if (!fwrite(body, len + 9, 1, saveFile)) {
std::cout << "VideoEncoder: Error while trying to write to file" << std::endl;
}
fclose(saveFile);*/
//END DEBUG
//other packet details
packet->m_hasAbsTimestamp = 0;
packet->m_packetType = RTMP_PACKET_TYPE_VIDEO;
if (rtmp != NULL) {
packet->m_nInfoField2 = rtmp->m_stream_id;
}
packet->m_nChannel = 0x04;
packet->m_headerType = RTMP_PACKET_SIZE_LARGE;
packet->m_nTimeStamp = timeOffset;
//send the packet
if (rtmp != NULL) {
RTMP_SendPacket(rtmp, packet, TRUE);
}I can see that Twitch is receiving the data in the inspector, at a steady 3kbps. so I’m sure something is wrong with how I’m adjusting the data before sending it. Can anyone advise me on what I’m doing wrong here ?
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ffmpeg - Determine what parameters
27 juin 2018, par AdmiralJonBI’ve got a video file that I’m trying to determine what parameters I can use to reproduce the encoding with ffmpeg.
Here’s the ffprobe of the particular stream in question.
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'video.mp4': Metadata:
major_brand : isom
minor_version : 512
compatible_brands: isomiso2avc1mp41
encoder : Lavf57.51.107
Duration: 00:05:02.84, start: 0.000000, bitrate: 4324 kb/s
Stream #0:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuvj420p(pc, bt709), 1920x1080 [SAR 1:1 DAR 16:9], 4323 kb/s, 25 fps, 25 tbr, 1200k tbn, 2400k tbc (default)
Metadata:
handler_name : VideoHandlerThe key part is that I’m noticing that the bitrate is quite low at 4324 kb/s, but this is actually an incredibly high quality video. To approach this sort of quality, I’ve only been able to make videos at 40000 kb/s (which is a huge increase in filesize). I also notice that it mentions yuvj420p which is an image format, but I have no idea what parameters with ffmpeg could produce that image format (if this at all would make a difference).
Would appreciate any help I can get.
Edit :
Here’s the output based on the comment below :
[STREAM]
index=0
codec_name=h264
codec_long_name=H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10
profile=Main
codec_type=video
codec_time_base=123187/3701760
codec_tag_string=avc1
codec_tag=0x31637661
width=1920
height=1080
coded_width=1920
coded_height=1080
has_b_frames=0
sample_aspect_ratio=1:1
display_aspect_ratio=16:9
pix_fmt=yuvj420p
level=41
color_range=pc
color_space=bt709
color_transfer=bt709
color_primaries=bt709
chroma_location=center
field_order=unknown
timecode=N/A
refs=1
is_avc=true
nal_length_size=4
id=N/A
r_frame_rate=15/1
avg_frame_rate=1850880/123187
time_base=1/15360
start_pts=0
start_time=0.000000
duration_ts=246374
duration=16.039974
bit_rate=13795677
max_bit_rate=N/A
bits_per_raw_sample=8
nb_frames=241
nb_read_frames=N/A
nb_read_packets=N/A
DISPOSITION:default=1
DISPOSITION:dub=0
DISPOSITION:original=0
DISPOSITION:comment=0
DISPOSITION:lyrics=0
DISPOSITION:karaoke=0
DISPOSITION:forced=0
DISPOSITION:hearing_impaired=0
DISPOSITION:visual_impaired=0
DISPOSITION:clean_effects=0
DISPOSITION:attached_pic=0
DISPOSITION:timed_thumbnails=0
TAG:language=und
TAG:handler_name=VideoHandler
[/STREAM]Edit 2 :
Here’s information from mediainfo
Video
ID : 1
Format : AVC
Format/Info : Advanced Video Codec
Format profile : Main@L4.1
Format settings, CABAC : Yes
Format settings, RefFrames : 1 frame
Format settings, GOP : M=1, N=32
Codec ID : avc1
Codec ID/Info : Advanced Video Coding
Duration : 5 min 2 s
Bit rate : 4 323 kb/s
Width : 1 920 pixels
Height : 1 080 pixels
Display aspect ratio : 16:9
Frame rate mode : Constant
Frame rate : 25.000 FPS
Color space : YUV
Chroma subsampling : 4:2:0
Bit depth : 8 bits
Scan type : Progressive
Bits/(Pixel*Frame) : 0.083
Stream size : 156 MiB (100%)
Color range : Full
Color primaries : BT.709
Transfer characteristics : BT.709
Matrix coefficients : BT.709 -
Solution for VB6 to broadcast Webcam
14 septembre 2018, par vantrung -cunconSorry, I know VB6 is decades ago, but I’m in a situation that I have to use VB6 to deliver live webcam stream beetween 2 PC in Server - Client Model program. Vb6-code holds the connection then I have no choice but to transfer all data via that connection.
I’ve tried weeks for this, uncountable approaches but went to nowhere.
My efforts focused on 3 major approaches :1/ Use ffmpeg to record live webcam as ".avi" file on hard disk, transfer parts of file to other end & play it. But I’ve stucked with a media-player that can play a "being written" avi file.
Windows Media Player control told me "file already in use..." & VLC Plugin can’t even be added to VB6 (axvlc.dll).
2/ Use ffmpeg to save live webcam as avi file, transfer each bit of that file to the other end, then in other end, extract 24 images / second from the avi to display continously in a picture box.
This approach is ok except that my hard disk get fulled of images in a time of wink and my program get very slow before hanging.3/ Use ffmpeg to stream the live webcam to a rtp-port like this :
ffmpeg -f dshow -i video="Lenovo EasyCamera" -vcodec mpeg2video -pix_fmt yuv422p -f rtp -an rtp://224.1.2.3:8191
I’ve successfully watch the stream in VLC, but VLC(axvlc.dll) refused to be integrated into ancient VB6. And more important, I don’t know how to redirect/reroute the rtp stream to other PC with VB6.
Any one please light me up ? (Any 3rd party component is welcomed)