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Configuration spécifique d’Apache
4 février 2011, parModules spécifiques
Pour la configuration d’Apache, il est conseillé d’activer certains modules non spécifiques à MediaSPIP, mais permettant d’améliorer les performances : mod_deflate et mod_headers pour compresser automatiquement via Apache les pages. Cf ce tutoriel ; mode_expires pour gérer correctement l’expiration des hits. Cf ce tutoriel ;
Il est également conseillé d’ajouter la prise en charge par apache du mime-type pour les fichiers WebM comme indiqué dans ce tutoriel.
Création d’un (...) -
HTML5 audio and video support
13 avril 2011, parMediaSPIP uses HTML5 video and audio tags to play multimedia files, taking advantage of the latest W3C innovations supported by modern browsers.
The MediaSPIP player used has been created specifically for MediaSPIP and can be easily adapted to fit in with a specific theme.
For older browsers the Flowplayer flash fallback is used.
MediaSPIP allows for media playback on major mobile platforms with the above (...) -
Use, discuss, criticize
13 avril 2011, parTalk to people directly involved in MediaSPIP’s development, or to people around you who could use MediaSPIP to share, enhance or develop their creative projects.
The bigger the community, the more MediaSPIP’s potential will be explored and the faster the software will evolve.
A discussion list is available for all exchanges between users.
Sur d’autres sites (5550)
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How to fix ffmpeg's offical tutorials03 bug that sound does't work well ?
31 janvier 2019, par xiaodaiI want to learn to make a player with ffmpeg and sdl. The tutorial I used is this.[http://dranger.com/ffmpeg/tutorial03.html] 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.
-
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.
-
How to add delay between input and outpout with ffmpeg
12 février 2019, par ManYouTrollI would like to add the delay so that it leaves for example 1 minutes after having received with this command :
ffmpeg -i rtmp://xx1.xx1.xx1.xx1/live -f flv rtmp://xx2.xx2.xx2.xx2/delay/pass
So a shift of 1 minute between the 2 video streams.
I hope you can help me, thank you in advance.