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Autres articles (56)
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La sauvegarde automatique de canaux SPIP
1er avril 2010, parDans le cadre de la mise en place d’une plateforme ouverte, il est important pour les hébergeurs de pouvoir disposer de sauvegardes assez régulières pour parer à tout problème éventuel.
Pour réaliser cette tâche on se base sur deux plugins SPIP : Saveauto qui permet une sauvegarde régulière de la base de donnée sous la forme d’un dump mysql (utilisable dans phpmyadmin) mes_fichiers_2 qui permet de réaliser une archive au format zip des données importantes du site (les documents, les éléments (...) -
Script d’installation automatique de MediaSPIP
25 avril 2011, parAfin de palier aux difficultés d’installation dues principalement aux dépendances logicielles coté serveur, un script d’installation "tout en un" en bash a été créé afin de faciliter cette étape sur un serveur doté d’une distribution Linux compatible.
Vous devez bénéficier d’un accès SSH à votre serveur et d’un compte "root" afin de l’utiliser, ce qui permettra d’installer les dépendances. Contactez votre hébergeur si vous ne disposez pas de cela.
La documentation de l’utilisation du script d’installation (...) -
MediaSPIP Core : La Configuration
9 novembre 2010, parMediaSPIP Core fournit par défaut trois pages différentes de configuration (ces pages utilisent le plugin de configuration CFG pour fonctionner) : une page spécifique à la configuration générale du squelettes ; une page spécifique à la configuration de la page d’accueil du site ; une page spécifique à la configuration des secteurs ;
Il fournit également une page supplémentaire qui n’apparait que lorsque certains plugins sont activés permettant de contrôler l’affichage et les fonctionnalités spécifiques (...)
Sur d’autres sites (6862)
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How to fix ffmpeg's official tutorials03 bug that sound does't work well ? [on hold]
31 janvier 2019, par xiaodaiI want to make a player with ffmpeg and sdl. The tutorial I used is this though I have resampled the audio from decode stream, the sound still plays with loud noise.
I have no ideas to fix it anymore.
I used the following :
- the latest ffmpeg and sdl1
- Visual Studio 2010
// tutorial03.c
// A pedagogical video player that will stream through every video frame as fast as it can
// and play audio (out of sync).
//
// This tutorial was written by Stephen Dranger (dranger@gmail.com).
//
// Code based on FFplay, Copyright (c) 2003 Fabrice Bellard,
// and a tutorial by Martin Bohme (boehme@inb.uni-luebeckREMOVETHIS.de)
// Tested on Gentoo, CVS version 5/01/07 compiled with GCC 4.1.1
//
// Use the Makefile to build all examples.
//
// Run using
// tutorial03 myvideofile.mpg
//
// to play the stream on your screen.
extern "C"{
#include <libavcodec></libavcodec>avcodec.h>
#include <libavformat></libavformat>avformat.h>
#include <libswscale></libswscale>swscale.h>
#include <libavutil></libavutil>channel_layout.h>
#include <libavutil></libavutil>common.h>
#include <libavutil></libavutil>frame.h>
#include <libavutil></libavutil>samplefmt.h>
#include "libswresample/swresample.h"
#include <sdl></sdl>SDL.h>
#include <sdl></sdl>SDL_thread.h>
};
#ifdef __WIN32__
#undef main /* Prevents SDL from overriding main() */
#endif
#include
#define SDL_AUDIO_BUFFER_SIZE 1024
#define MAX_AUDIO_FRAME_SIZE 192000
struct SwrContext *audio_swrCtx;
FILE *pFile=fopen("output.pcm", "wb");
FILE *pFile_stream=fopen("output_stream.pcm","wb");
int audio_len;
typedef struct PacketQueue {
AVPacketList *first_pkt, *last_pkt;
int nb_packets;
int size;
SDL_mutex *mutex;
SDL_cond *cond;
} PacketQueue;
PacketQueue audioq;
int quit = 0;
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));
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);
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;
static AVFrame frame;
int len1, data_size = 0;
for(;;) {
while(audio_pkt_size > 0) {
int got_frame = 0;
len1 = avcodec_decode_audio4(aCodecCtx, &frame, &got_frame, &pkt);
if(len1 < 0) {
/* if error, skip frame */
audio_pkt_size = 0;
break;
}
audio_pkt_data += len1;
audio_pkt_size -= len1;
data_size = 0;
/*
au_convert_ctx = swr_alloc();
au_convert_ctx=swr_alloc_set_opts(au_convert_ctx,out_channel_layout, out_sample_fmt, out_sample_rate,
in_channel_layout,pCodecCtx->sample_fmt , pCodecCtx->sample_rate,0, NULL);
swr_init(au_convert_ctx);
swr_convert(au_convert_ctx,&out_buffer, MAX_AUDIO_FRAME_SIZE,(const uint8_t **)pFrame->data , pFrame->nb_samples);
*/
if( got_frame ) {
audio_swrCtx=swr_alloc();
audio_swrCtx=swr_alloc_set_opts(audio_swrCtx, // we're allocating a new context
AV_CH_LAYOUT_STEREO,//AV_CH_LAYOUT_STEREO, // out_ch_layout
AV_SAMPLE_FMT_S16, // out_sample_fmt
44100, // out_sample_rate
aCodecCtx->channel_layout, // in_ch_layout
aCodecCtx->sample_fmt, // in_sample_fmt
aCodecCtx->sample_rate, // in_sample_rate
0, // log_offset
NULL); // log_ctx
int ret=swr_init(audio_swrCtx);
int out_samples = av_rescale_rnd(swr_get_delay(audio_swrCtx, aCodecCtx->sample_rate) + 1024, 44100, aCodecCtx->sample_rate, AV_ROUND_UP);
ret=swr_convert(audio_swrCtx,&audio_buf, MAX_AUDIO_FRAME_SIZE,(const uint8_t **)frame.data ,frame.nb_samples);
data_size =
av_samples_get_buffer_size
(
&data_size,
av_get_channel_layout_nb_channels(AV_CH_LAYOUT_STEREO),
ret,
AV_SAMPLE_FMT_S16,
1
);
fwrite(audio_buf, 1, data_size, pFile);
//memcpy(audio_buf, frame.data[0], data_size);
swr_free(&audio_swrCtx);
}
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 /*audio_len,*/ audio_size;
static uint8_t audio_buf[(MAX_AUDIO_FRAME_SIZE * 3) / 2];
static unsigned int audio_buf_size = 0;
static unsigned int audio_buf_index = 0;
//SDL_memset(stream, 0, len);
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, audio_buf_size);
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;
}
audio_len = audio_buf_size - audio_buf_index;
if(audio_len > len) {
audio_len = len;
}
memcpy(stream, (uint8_t *)audio_buf , audio_len);
//SDL_MixAudio(stream,(uint8_t*)audio_buf,audio_len,SDL_MIX_MAXVOLUME);
fwrite(audio_buf, 1, audio_len, pFile_stream);
len -= audio_len;
stream += audio_len;
audio_buf_index += audio_len;
audio_len=len;
}
}
int main(int argc, char *argv[]) {
AVFormatContext *pFormatCtx = NULL;
int i, videoStream, audioStream;
AVCodecContext *pCodecCtx = NULL;
AVCodec *pCodec = NULL;
AVFrame *pFrame = NULL;
AVPacket packet;
int frameFinished;
//float aspect_ratio;
AVCodecContext *aCodecCtx = NULL;
AVCodec *aCodec = NULL;
SDL_Overlay *bmp = NULL;
SDL_Surface *screen = NULL;
SDL_Rect rect;
SDL_Event event;
SDL_AudioSpec wanted_spec, spec;
struct SwsContext *sws_ctx = NULL;
AVDictionary *videoOptionsDict = NULL;
AVDictionary *audioOptionsDict = NULL;
if(argc < 2) {
fprintf(stderr, "Usage: test <file>\n");
exit(1);
}
// Register all formats and codecs
av_register_all();
if(SDL_Init(SDL_INIT_VIDEO | SDL_INIT_AUDIO | SDL_INIT_TIMER)) {
fprintf(stderr, "Could not initialize SDL - %s\n", SDL_GetError());
exit(1);
}
// Open video file
if(avformat_open_input(&pFormatCtx, argv[1]/*"file.mov"*/, NULL, NULL) != 0) {
return -1; // Couldn't open file
}
// Retrieve stream information
if(avformat_find_stream_info(pFormatCtx, NULL) < 0) {
return -1; // Couldn't find stream information
}
// Dump information about file onto standard error
av_dump_format(pFormatCtx, 0, argv[1], 0);
// Find the first video stream
videoStream = -1;
audioStream = -1;
for(i = 0; 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_AUDIO &&
audioStream < 0) {
audioStream = i;
}
}
if(videoStream == -1) {
return -1; // Didn't find a video stream
}
if(audioStream == -1) {
return -1;
}
aCodecCtx = pFormatCtx->streams[audioStream]->codec;
// Set audio settings from codec info
wanted_spec.freq = 44100;
wanted_spec.format = AUDIO_S16SYS;
wanted_spec.channels = av_get_channel_layout_nb_channels(AV_CH_LAYOUT_STEREO);;
wanted_spec.silence = 0;
wanted_spec.samples = 1024;
wanted_spec.callback = audio_callback;
wanted_spec.userdata = aCodecCtx;
if(SDL_OpenAudio(&wanted_spec, &spec) < 0) {
fprintf(stderr, "SDL_OpenAudio: %s\n", SDL_GetError());
return -1;
}
aCodec = avcodec_find_decoder(aCodecCtx->codec_id);
if(!aCodec) {
fprintf(stderr, "Unsupported codec!\n");
return -1;
}
avcodec_open2(aCodecCtx, aCodec, &audioOptionsDict);
// 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
pCodec = avcodec_find_decoder(pCodecCtx->codec_id);
if(pCodec == NULL) {
fprintf(stderr, "Unsupported codec!\n");
return -1; // Codec not found
}
// Open codec
if(avcodec_open2(pCodecCtx, pCodec, &videoOptionsDict) < 0) {
return -1; // Could not open codec
}
// Allocate video frame
pFrame = av_frame_alloc();
// Make a screen to put our video
#ifndef __DARWIN__
screen = SDL_SetVideoMode(pCodecCtx->width, pCodecCtx->height, 0, 0);
#else
screen = SDL_SetVideoMode(pCodecCtx->width, pCodecCtx->height, 24, 0);
#endif
if(!screen) {
fprintf(stderr, "SDL: could not set video mode - exiting\n");
exit(1);
}
// Allocate a place to put our YUV image on that screen
bmp = SDL_CreateYUVOverlay(pCodecCtx->width,
pCodecCtx->height,
SDL_YV12_OVERLAY,
screen);
sws_ctx =
sws_getContext
(
pCodecCtx->width,
pCodecCtx->height,
pCodecCtx->pix_fmt,
pCodecCtx->width,
pCodecCtx->height,
PIX_FMT_YUV420P,
SWS_BILINEAR,
NULL,
NULL,
NULL
);
// Read frames and save first five frames to disk
i = 0;
while(av_read_frame(pFormatCtx, &packet) >= 0) {
// Is this a packet from the video stream?
if(packet.stream_index == videoStream) {
// Decode video frame
avcodec_decode_video2(pCodecCtx, pFrame, &frameFinished,
&packet);
// Did we get a video frame?
if(frameFinished) {
SDL_LockYUVOverlay(bmp);
AVPicture pict;
pict.data[0] = bmp->pixels[0];
pict.data[1] = bmp->pixels[2];
pict.data[2] = bmp->pixels[1];
pict.linesize[0] = bmp->pitches[0];
pict.linesize[1] = bmp->pitches[2];
pict.linesize[2] = bmp->pitches[1];
// Convert the image into YUV format that SDL uses
sws_scale
(
sws_ctx,
(uint8_t const * const *)pFrame->data,
pFrame->linesize,
0,
pCodecCtx->height,
pict.data,
pict.linesize
);
SDL_UnlockYUVOverlay(bmp);
rect.x = 0;
rect.y = 0;
rect.w = pCodecCtx->width;
rect.h = pCodecCtx->height;
SDL_DisplayYUVOverlay(bmp, &rect);
SDL_Delay(40);
av_free_packet(&packet);
}
} else if(packet.stream_index == audioStream) {
packet_queue_put(&audioq, &packet);
} else {
av_free_packet(&packet);
}
// Free the packet that was allocated by av_read_frame
SDL_PollEvent(&event);
switch(event.type) {
case SDL_QUIT:
quit = 1;
SDL_Quit();
exit(0);
break;
default:
break;
}
}
// Free the YUV frame
av_free(pFrame);
/*swr_free(&audio_swrCtx);*/
// Close the codec
avcodec_close(pCodecCtx);
fclose(pFile);
fclose(pFile_stream);
// Close the video file
avformat_close_input(&pFormatCtx);
return 0;
}
</file>I hope to play normally.
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Revision a00dad39bd : No arf right before real scene cut. To reduce pulsing we now allow an arf just
3 janvier 2014, par Paul WilkinsChanged Paths :
Modify /vp9/encoder/vp9_firstpass.c
No arf right before real scene cut.To reduce pulsing we now allow an arf just before forced key frames
and at the end of a clip or section (which may be stitched to
another clip or section). However, this does not make sense for
key frames arising from real scene cuts.Change from original patch reflects other recent changes in regard
to alignment of gf/arf and kf groups.Change-Id : I074a91d1207e9b3e28085af982f6718aa599775f
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Révision 21737 : Taille maximum des logos : inutile de definir les constante pour rien a chaque h...
29 octobre 2014, par cedric -+ permettre de ne definir que la largeur maxi ou que la hauteur maxi
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