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Les formats acceptés
28 janvier 2010, parLes commandes suivantes permettent d’avoir des informations sur les formats et codecs gérés par l’installation local de ffmpeg :
ffmpeg -codecs ffmpeg -formats
Les format videos acceptés en entrée
Cette liste est non exhaustive, elle met en exergue les principaux formats utilisés : h264 : H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 m4v : raw MPEG-4 video format flv : Flash Video (FLV) / Sorenson Spark / Sorenson H.263 Theora wmv :
Les formats vidéos de sortie possibles
Dans un premier temps on (...) -
Ajouter notes et légendes aux images
7 février 2011, parPour pouvoir ajouter notes et légendes aux images, la première étape est d’installer le plugin "Légendes".
Une fois le plugin activé, vous pouvez le configurer dans l’espace de configuration afin de modifier les droits de création / modification et de suppression des notes. Par défaut seuls les administrateurs du site peuvent ajouter des notes aux images.
Modification lors de l’ajout d’un média
Lors de l’ajout d’un média de type "image" un nouveau bouton apparait au dessus de la prévisualisation (...) -
Organiser par catégorie
17 mai 2013, parDans MédiaSPIP, une rubrique a 2 noms : catégorie et rubrique.
Les différents documents stockés dans MédiaSPIP peuvent être rangés dans différentes catégories. On peut créer une catégorie en cliquant sur "publier une catégorie" dans le menu publier en haut à droite ( après authentification ). Une catégorie peut être rangée dans une autre catégorie aussi ce qui fait qu’on peut construire une arborescence de catégories.
Lors de la publication prochaine d’un document, la nouvelle catégorie créée sera proposée (...)
Sur d’autres sites (3898)
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Trouble syncing libavformat/ffmpeg with x264 and RTP
26 décembre 2012, par Jacob PeddicordI've been working on some streaming software that takes live feeds
from various kinds of cameras and streams over the network using
H.264. To accomplish this, I'm using the x264 encoder directly (with
the "zerolatency" preset) and feeding NALs as they are available to
libavformat to pack into RTP (ultimately RTSP). Ideally, this
application should be as real-time as possible. For the most part,
this has been working well.Unfortunately, however, there is some sort of synchronization issue :
any video playback on clients seems to show a few smooth frames,
followed by a short pause, then more frames ; repeat. Additionally,
there appears to be approximately a 4-second delay. This happens with
every video player I've tried : Totem, VLC, and basic gstreamer pipes.I've boiled it all down to a somewhat small test case :
#include
#include
#include
#include
#include <libavformat></libavformat>avformat.h>
#include <libswscale></libswscale>swscale.h>
#define WIDTH 640
#define HEIGHT 480
#define FPS 30
#define BITRATE 400000
#define RTP_ADDRESS "127.0.0.1"
#define RTP_PORT 49990
struct AVFormatContext* avctx;
struct x264_t* encoder;
struct SwsContext* imgctx;
uint8_t test = 0x80;
void create_sample_picture(x264_picture_t* picture)
{
// create a frame to store in
x264_picture_alloc(picture, X264_CSP_I420, WIDTH, HEIGHT);
// fake image generation
// disregard how wrong this is; just writing a quick test
int strides = WIDTH / 8;
uint8_t* data = malloc(WIDTH * HEIGHT * 3);
memset(data, test, WIDTH * HEIGHT * 3);
test = (test << 1) | (test >> (8 - 1));
// scale the image
sws_scale(imgctx, (const uint8_t* const*) &data, &strides, 0, HEIGHT,
picture->img.plane, picture->img.i_stride);
}
int encode_frame(x264_picture_t* picture, x264_nal_t** nals)
{
// encode a frame
x264_picture_t pic_out;
int num_nals;
int frame_size = x264_encoder_encode(encoder, nals, &num_nals, picture, &pic_out);
// ignore bad frames
if (frame_size < 0)
{
return frame_size;
}
return num_nals;
}
void stream_frame(uint8_t* payload, int size)
{
// initalize a packet
AVPacket p;
av_init_packet(&p);
p.data = payload;
p.size = size;
p.stream_index = 0;
p.flags = AV_PKT_FLAG_KEY;
p.pts = AV_NOPTS_VALUE;
p.dts = AV_NOPTS_VALUE;
// send it out
av_interleaved_write_frame(avctx, &p);
}
int main(int argc, char* argv[])
{
// initalize ffmpeg
av_register_all();
// set up image scaler
// (in-width, in-height, in-format, out-width, out-height, out-format, scaling-method, 0, 0, 0)
imgctx = sws_getContext(WIDTH, HEIGHT, PIX_FMT_MONOWHITE,
WIDTH, HEIGHT, PIX_FMT_YUV420P,
SWS_FAST_BILINEAR, NULL, NULL, NULL);
// set up encoder presets
x264_param_t param;
x264_param_default_preset(&param, "ultrafast", "zerolatency");
param.i_threads = 3;
param.i_width = WIDTH;
param.i_height = HEIGHT;
param.i_fps_num = FPS;
param.i_fps_den = 1;
param.i_keyint_max = FPS;
param.b_intra_refresh = 0;
param.rc.i_bitrate = BITRATE;
param.b_repeat_headers = 1; // whether to repeat headers or write just once
param.b_annexb = 1; // place start codes (1) or sizes (0)
// initalize
x264_param_apply_profile(&param, "high");
encoder = x264_encoder_open(&param);
// at this point, x264_encoder_headers can be used, but it has had no effect
// set up streaming context. a lot of error handling has been ommitted
// for brevity, but this should be pretty standard.
avctx = avformat_alloc_context();
struct AVOutputFormat* fmt = av_guess_format("rtp", NULL, NULL);
avctx->oformat = fmt;
snprintf(avctx->filename, sizeof(avctx->filename), "rtp://%s:%d", RTP_ADDRESS, RTP_PORT);
if (url_fopen(&avctx->pb, avctx->filename, URL_WRONLY) < 0)
{
perror("url_fopen failed");
return 1;
}
struct AVStream* stream = av_new_stream(avctx, 1);
// initalize codec
AVCodecContext* c = stream->codec;
c->codec_id = CODEC_ID_H264;
c->codec_type = AVMEDIA_TYPE_VIDEO;
c->flags = CODEC_FLAG_GLOBAL_HEADER;
c->width = WIDTH;
c->height = HEIGHT;
c->time_base.den = FPS;
c->time_base.num = 1;
c->gop_size = FPS;
c->bit_rate = BITRATE;
avctx->flags = AVFMT_FLAG_RTP_HINT;
// write the header
av_write_header(avctx);
// make some frames
for (int frame = 0; frame < 10000; frame++)
{
// create a sample moving frame
x264_picture_t* pic = (x264_picture_t*) malloc(sizeof(x264_picture_t));
create_sample_picture(pic);
// encode the frame
x264_nal_t* nals;
int num_nals = encode_frame(pic, &nals);
if (num_nals < 0)
printf("invalid frame size: %d\n", num_nals);
// send out NALs
for (int i = 0; i < num_nals; i++)
{
stream_frame(nals[i].p_payload, nals[i].i_payload);
}
// free up resources
x264_picture_clean(pic);
free(pic);
// stream at approx 30 fps
printf("frame %d\n", frame);
usleep(33333);
}
return 0;
}This test shows black lines on a white background that
should move smoothly to the left. It has been written for ffmpeg 0.6.5
but the problem can be reproduced on 0.8 and 0.10 (from what I've tested so far). I've taken some shortcuts in error handling to make this example as short as
possible while still showing the problem, so please excuse some of the
nasty code. I should also note that while an SDP is not used here, I
have tried using that already with similar results. The test can be
compiled with :gcc -g -std=gnu99 streamtest.c -lswscale -lavformat -lx264 -lm -lpthread -o streamtest
It can be played with gtreamer directly :
gst-launch udpsrc port=49990 ! application/x-rtp,payload=96,clock-rate=90000 ! rtph264depay ! decodebin ! xvimagesink
You should immediately notice the stuttering. One common "fix" I've
seen all over the Internet is to add sync=false to the pipeline :gst-launch udpsrc port=49990 ! application/x-rtp,payload=96,clock-rate=90000 ! rtph264depay ! decodebin ! xvimagesink sync=false
This causes playback to be smooth (and near-realtime), but is a
non-solution and only works with gstreamer. I'd like to fix the
problem at the source. I've been able to stream with near-identical
parameters using raw ffmpeg and haven't had any issues :ffmpeg -re -i sample.mp4 -vcodec libx264 -vpre ultrafast -vpre baseline -b 400000 -an -f rtp rtp://127.0.0.1:49990 -an
So clearly I'm doing something wrong. But what is it ?
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closes #3
4 août 2012, par lipkacloses #3
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Update README.md
6 août 2012Update README.md
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