Recherche avancée
Médias (1)
-
Rennes Emotion Map 2010-11
19 octobre 2011, par
Mis à jour : Juillet 2013
Langue : français
Type : Texte
Autres articles (58)
-
Encodage et transformation en formats lisibles sur Internet
10 avril 2011MediaSPIP transforme et ré-encode les documents mis en ligne afin de les rendre lisibles sur Internet et automatiquement utilisables sans intervention du créateur de contenu.
Les vidéos sont automatiquement encodées dans les formats supportés par HTML5 : MP4, Ogv et WebM. La version "MP4" est également utilisée pour le lecteur flash de secours nécessaire aux anciens navigateurs.
Les documents audios sont également ré-encodés dans les deux formats utilisables par HTML5 :MP3 et Ogg. La version "MP3" (...) -
Les autorisations surchargées par les plugins
27 avril 2010, parMediaspip core
autoriser_auteur_modifier() afin que les visiteurs soient capables de modifier leurs informations sur la page d’auteurs -
ANNEXE : Les plugins utilisés spécifiquement pour la ferme
5 mars 2010, parLe site central/maître de la ferme a besoin d’utiliser plusieurs plugins supplémentaires vis à vis des canaux pour son bon fonctionnement. le plugin Gestion de la mutualisation ; le plugin inscription3 pour gérer les inscriptions et les demandes de création d’instance de mutualisation dès l’inscription des utilisateurs ; le plugin verifier qui fournit une API de vérification des champs (utilisé par inscription3) ; le plugin champs extras v2 nécessité par inscription3 (...)
Sur d’autres sites (8831)
-
Decode audio and video and process both flows — ffmpeg, sdl, opencv
24 février 2012, par EricMy goal is to proceed on audio and video of mpeg-2 file independently, and to keep synchronicity on both flows. Duration of video is about 1 or 2 minutes maximum.
-
First, following this post "opencv for reading videos (and do the process),ffmpeg for audio , and SDL used to play both" sounds perfect. I have done some modification on the code considering recent ffmpeg naming changes. Compilation with cmake on 64-bits machine is fine. I get an error "Unsupported codec [3]" when opening codec.
The code is following. -
Second, I looking for code dealing with synchronicity on both flows.
#include "opencv/highgui.h"
#include "opencv/cv.h"
#ifndef INT64_C
#define INT64_C(c) (c ## LL)
#define UINT64_C(c) (c ## ULL)
#endif
extern "C"{
#include <sdl></sdl>SDL.h>
#include <sdl></sdl>SDL_thread.h>
#include <libavcodec></libavcodec>avcodec.h>
#include <libavformat></libavformat>avformat.h>
}
#include <iostream>
#include
#include
using namespace cv;
#define SDL_AUDIO_BUFFER_SIZE 1024
typedef struct PacketQueue
{
AVPacketList *first_pkt, *last_pkt;
int nb_packets;
int size;
SDL_mutex *mutex;
SDL_cond *cond;
} PacketQueue;
PacketQueue audioq;
int audioStream = -1;
int videoStream = -1;
int quit = 0;
SDL_Surface* screen = NULL;
SDL_Surface* surface = NULL;
AVFormatContext* pFormatCtx = NULL;
AVCodecContext* aCodecCtx = NULL;
AVCodecContext* pCodecCtx = NULL;
void show_frame(IplImage* img){
if (!screen){
screen = SDL_SetVideoMode(img->width, img->height, 0, 0);
if (!screen){
fprintf(stderr, "SDL: could not set video mode - exiting\n");
exit(1);
}
}
// Assuming IplImage packed as BGR 24bits
SDL_Surface* surface = SDL_CreateRGBSurfaceFrom((void*)img->imageData,
img->width,
img->height,
img->depth * img->nChannels,
img->widthStep,
0xff0000, 0x00ff00, 0x0000ff, 0
);
SDL_BlitSurface(surface, 0, screen, 0);
SDL_Flip(screen);
}
void packet_queue_init(PacketQueue *q){
memset(q, 0, sizeof(PacketQueue));
q->mutex = SDL_CreateMutex();
q->cond = SDL_CreateCond();
}
int packet_queue_put(PacketQueue *q, AVPacket *pkt){
AVPacketList *pkt1;
if (av_dup_packet(pkt) < 0){
return -1;
}
pkt1 = (AVPacketList*) av_malloc(sizeof(AVPacketList));
//pkt1 = (AVPacketList*) malloc(sizeof(AVPacketList));
if (!pkt1) return -1;
pkt1->pkt = *pkt;
pkt1->next = NULL;
SDL_LockMutex(q->mutex);
if (!q->last_pkt)
q->first_pkt = pkt1;
else
q->last_pkt->next = pkt1;
q->last_pkt = pkt1;
q->nb_packets++;
q->size += pkt1->pkt.size;
SDL_CondSignal(q->cond);
SDL_UnlockMutex(q->mutex);
return 0;
}
static int packet_queue_get(PacketQueue *q, AVPacket *pkt, int block){
AVPacketList *pkt1;
int ret;
SDL_LockMutex(q->mutex);
for (;;){
if( quit){
ret = -1;
break;
}
pkt1 = q->first_pkt;
if (pkt1){
q->first_pkt = pkt1->next;
if (!q->first_pkt)
q->last_pkt = NULL;
q->nb_packets--;
q->size -= pkt1->pkt.size;
*pkt = pkt1->pkt;
av_free(pkt1);
//free(pkt1);
ret = 1;
break;
}
else if (!block){
ret = 0;
break;
}
else{
SDL_CondWait(q->cond, q->mutex);
}
}
SDL_UnlockMutex(q->mutex);
return ret;
}
int audio_decode_frame(AVCodecContext *aCodecCtx, uint8_t *audio_buf, int buf_size){
static AVPacket pkt;
static uint8_t *audio_pkt_data = NULL;
static int audio_pkt_size = 0;
int len1, data_size;
for (;;){
while (audio_pkt_size > 0){
data_size = buf_size;
len1 = avcodec_decode_audio3(aCodecCtx, (int16_t*)audio_buf, &data_size, &pkt);
if (len1 < 0){
// if error, skip frame
audio_pkt_size = 0;
break;
}
audio_pkt_data += len1;
audio_pkt_size -= len1;
if (data_size <= 0){
// No data yet, get more frames
continue;
}
// We have data, return it and come back for more later
return data_size;
}
if (pkt.data)
av_free_packet(&pkt);
if (quit) return -1;
if (packet_queue_get(&audioq, &pkt, 1) < 0) return -1;
audio_pkt_data = pkt.data;
audio_pkt_size = pkt.size;
}
}
void audio_callback(void *userdata, Uint8 *stream, int len){
AVCodecContext *aCodecCtx = (AVCodecContext *)userdata;
int len1, audio_size;
static uint8_t audio_buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2];
static unsigned int audio_buf_size = 0;
static unsigned int audio_buf_index = 0;
while (len > 0){
if (audio_buf_index >= audio_buf_size){
// We have already sent all our data; get more
audio_size = audio_decode_frame(aCodecCtx, audio_buf, sizeof(audio_buf));
if(audio_size < 0){
// If error, output silence
audio_buf_size = 1024; // arbitrary?
memset(audio_buf, 0, audio_buf_size);
}
else{
audio_buf_size = audio_size;
}
audio_buf_index = 0;
}
len1 = audio_buf_size - audio_buf_index;
if (len1 > len)
len1 = len;
memcpy(stream, (uint8_t *)audio_buf + audio_buf_index, len1);
len -= len1;
stream += len1;
audio_buf_index += len1;
}
}
void setup_ffmpeg(char* filename)
{
if (avformat_open_input(&pFormatCtx, filename, NULL, NULL) != 0){
fprintf(stderr, "FFmpeg failed to open file %s!\n", filename);
exit(-1);
}
if (av_find_stream_info(pFormatCtx) < 0){
fprintf(stderr, "FFmpeg failed to retrieve stream info!\n");
exit(-1);
}
// Dump information about file onto standard error
av_dump_format(pFormatCtx, 0, filename, 0);
// Find the first video stream
int i = 0;
for (i; i < pFormatCtx->nb_streams; i++){
if (pFormatCtx->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO && videoStream < 0){
videoStream = i;
}
if (pFormatCtx->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO && audioStream < 0){
audioStream = i;
}
}
if (videoStream == -1){
fprintf(stderr, "No video stream found in %s!\n", filename);
exit(-1);
}
if (audioStream == -1){
fprintf(stderr, "No audio stream found in %s!\n", filename);
exit(-1);
}
// Get a pointer to the codec context for the audio stream
aCodecCtx = pFormatCtx->streams[audioStream]->codec;
// Set audio settings from codec info
SDL_AudioSpec wanted_spec;
wanted_spec.freq = aCodecCtx->sample_rate;
wanted_spec.format = AUDIO_S16SYS;
wanted_spec.channels = aCodecCtx->channels;
wanted_spec.silence = 0;
wanted_spec.samples = SDL_AUDIO_BUFFER_SIZE;
wanted_spec.callback = audio_callback;
wanted_spec.userdata = aCodecCtx;
SDL_AudioSpec spec;
if (SDL_OpenAudio(&wanted_spec, &spec) < 0){
fprintf(stderr, "SDL_OpenAudio: %s\n", SDL_GetError());
exit(-1);
}
AVCodec* aCodec = avcodec_find_decoder(aCodecCtx->codec_id);
if (!aCodec){
fprintf(stderr, "Unsupported codec [1]!\n");
exit(-1);
}
avcodec_open(aCodecCtx, aCodec);
// audio_st = pFormatCtx->streams[index]
packet_queue_init(&audioq);
SDL_PauseAudio(0);
// Get a pointer to the codec context for the video stream
pCodecCtx = pFormatCtx->streams[videoStream]->codec;
// Find the decoder for the video stream
AVCodec* pCodec = avcodec_find_decoder(pCodecCtx->codec_id);
if (pCodec == NULL){
fprintf(stderr, "Unsupported codec [2]!\n");
exit(-1); // Codec not found
}
// Open codec
if (avcodec_open(pCodecCtx, pCodec) < 0){
fprintf(stderr, "Unsupported codec [3]!\n");
exit(-1); // Could not open codec
}
}
int main(int argc, char* argv[])
{
if (argc < 2){
std::cout << "Usage: " << argv[0] << " <video>" << std::endl;
return -1;
}
av_register_all();
// Init SDL
if (SDL_Init(SDL_INIT_VIDEO | SDL_INIT_AUDIO | SDL_INIT_TIMER))
{
fprintf(stderr, "Could not initialize SDL - %s\n", SDL_GetError());
return -1;
}
// Init ffmpeg and setup some SDL stuff related to Audio
setup_ffmpeg(argv[1]);
VideoCapture cap(argv[1]);
if (!cap.isOpened()){
std::cout << "Failed to load file!" << std::endl;
return -1;
}
AVPacket packet;
while (av_read_frame(pFormatCtx, &packet) >= 0)
{
if (packet.stream_index == videoStream)
{
// Actually this is were SYNC between audio/video would happen.
// Right now I assume that every VIDEO packet contains an entire video frame, and that's not true. A video frame can be made by multiple packets!
// But for the time being, assume 1 video frame == 1 video packet,
// so instead of reading the frame through ffmpeg, I read it through OpenCV.
Mat frame;
cap >> frame; // get a new frame from camera
// do some processing on the frame, either as a Mat or as IplImage.
// For educational purposes, applying a lame grayscale conversion
IplImage ipl_frame = frame;
for (int i = 0; i < ipl_frame.width * ipl_frame.height * ipl_frame.nChannels; i += ipl_frame.nChannels)
{
ipl_frame.imageData[i] = (ipl_frame.imageData[i] + ipl_frame.imageData[i+1] + ipl_frame.imageData[i+2])/3; //B
ipl_frame.imageData[i+1] = (ipl_frame.imageData[i] + ipl_frame.imageData[i+1] + ipl_frame.imageData[i+2])/3; //G
ipl_frame.imageData[i+2] = (ipl_frame.imageData[i] + ipl_frame.imageData[i+1] + ipl_frame.imageData[i+2])/3; //R
}
// Display it on SDL window
show_frame(&ipl_frame);
av_free_packet(&packet);
}
else if (packet.stream_index == audioStream)
{
packet_queue_put(&audioq, &packet);
}
else
{
av_free_packet(&packet);
}
SDL_Event event;
SDL_PollEvent(&event);
switch (event.type)
{
case SDL_QUIT:
SDL_FreeSurface(surface);
SDL_Quit();
break;
default:
break;
}
}
// the camera will be deinitialized automatically in VideoCapture destructor
// Close the codec
avcodec_close(pCodecCtx);
// Close the video file
av_close_input_file(pFormatCtx);
return 0;
}
</video></iostream> -
-
lavc : drop encode() support for video.
23 février 2012, par Anton Khirnovlavc : drop encode() support for video.
-
Theatrical quality ffmpeg/x264 encoding of a high-motion 1080p video
2 décembre 2011, par IanI've been struggling with encoding videos using FFMPEG and x264. The output stutters when played back in Quicktime, while in VLC it shows a lot of compression artifacts at the same places Quicktime stutters. So it seems like Quicktime is stuttering because it's trying to suppress the corruption/artifacts.
The videos have a lot of random motion in them, including frames where 75% of the pixels will change at a random interval (the video is software generated so it's truly pseudo-random). The compression seems to be choking in these places where it's likely detecting a "scene cut" incorrectly. It also seems to choke at regular intervals where I guess it's doing a keyframe.
I've based my encoding preset off of the x264-hq preset that comes with FFMPEG. I've tried turning off scene cut detection, and playing with the
keyint/gandkeyint_minoptions. Settinggto 1 makes it work, but blows out the filesize. I've tried the lossless presets, but they won't playback at all in Quicktime. Oddly, I haven't had any problems when working with a lower-resolution test video (1440x810).Here's the preset I have right now, which works, but yields a file that's approximately 60% larger than the (non-working) hq preset yields. Is there any way to improve upon this ? The filesize doesn't matter much, I just want something that will playback anywhere and be very high quality.
coder=1 flags=+loop cmp=+chroma partitions=+parti8x8+parti4x4+partp8x8+partp4x4+partb8x8 me_method=umh subq=8 me_range=16 g=1 keyint_min=1 sc_threshold=0 i_qfactor=0.71 b_strategy=1crf=20 qcomp=0.6 qmin=20 qmax=51 qdiff=4 bf=16 refs=4 trellis=1 flags2=+dct8x8+wpred+bpyramid+mixed_refs wpredp=2
Here's the command :
ffmpeg \ -r 60 -i "frame-%06d.tiff" \ -vcodec libx264 -vpre my_preset \ -threads 0 \ -r 60 -an -f out.mp4
Syndiquer tout le site
