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Richard Stallman et le logiciel libre
19 octobre 2011, par
Mis à jour : Mai 2013
Langue : français
Type : Texte
Autres articles (43)
-
MediaSPIP v0.2
21 juin 2013, parMediaSPIP 0.2 est la première version de MediaSPIP stable.
Sa date de sortie officielle est le 21 juin 2013 et est annoncée ici.
Le fichier zip ici présent contient uniquement les sources de MediaSPIP en version standalone.
Comme pour la version précédente, il est nécessaire d’installer manuellement l’ensemble des dépendances logicielles sur le serveur.
Si vous souhaitez utiliser cette archive pour une installation en mode ferme, il vous faudra également procéder à d’autres modifications (...) -
Mise à disposition des fichiers
14 avril 2011, parPar défaut, lors de son initialisation, MediaSPIP ne permet pas aux visiteurs de télécharger les fichiers qu’ils soient originaux ou le résultat de leur transformation ou encodage. Il permet uniquement de les visualiser.
Cependant, il est possible et facile d’autoriser les visiteurs à avoir accès à ces documents et ce sous différentes formes.
Tout cela se passe dans la page de configuration du squelette. Il vous faut aller dans l’espace d’administration du canal, et choisir dans la navigation (...) -
Installation en mode ferme
4 février 2011, parLe mode ferme permet d’héberger plusieurs sites de type MediaSPIP en n’installant qu’une seule fois son noyau fonctionnel.
C’est la méthode que nous utilisons sur cette même plateforme.
L’utilisation en mode ferme nécessite de connaïtre un peu le mécanisme de SPIP contrairement à la version standalone qui ne nécessite pas réellement de connaissances spécifique puisque l’espace privé habituel de SPIP n’est plus utilisé.
Dans un premier temps, vous devez avoir installé les mêmes fichiers que l’installation (...)
Sur d’autres sites (4105)
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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.
-
ffmpeg : Trying to access Ebur128Context->integrated_loudness but unsuccessful
12 avril 2019, par Sourabh Jain[FFMPEG] Trying to access Ebur128Context->integrated_loudness but unsuccessful
I am trying to run ebur128Filter on audio file . similar to be doing
[http://ffmpeg.org/doxygen/2.6/f__ebur128_8c_source.html#l00135]ffmpeg -i sample.wav -filter_complex ebur128=peak=true -f null -
result of which is :
[Parsed_ebur128_0 @ 0x7f9d38403ec0] Summary:
Integrated loudness:
I: -15.5 LUFS
Threshold: -25.6 LUFS
Loudness range:
LRA: 1.5 LU
Threshold: -35.5 LUFS
LRA low: -16.3 LUFS
LRA high: -14.8 LUFS
True peak:
Peak: -0.4 dBFS/*
* Copyright (c) 2010 Nicolas George
* Copyright (c) 2011 Stefano Sabatini
* Copyright (c) 2012 Clément Bœsch
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* API example for audio decoding and filtering
* @example filtering_audio.c
*/
#include
#include <libavcodec></libavcodec>avcodec.h>
#include <libavformat></libavformat>avformat.h>
#include <libavfilter></libavfilter>buffersink.h>
#include <libavfilter></libavfilter>buffersrc.h>
#include <libavutil></libavutil>opt.h>
#define MAX_CHANNELS 63
static const char *filter_descr = "ebur128=peak=true";
static AVFormatContext *fmt_ctx;
static AVCodecContext *dec_ctx;
AVFilterContext *buffersink_ctx;
AVFilterContext *buffersrc_ctx;
AVFilterGraph *filter_graph;
static int audio_stream_index = -1;
struct rect { int x, y, w, h; };
struct hist_entry {
int count; ///< how many times the corresponding value occurred
double energy; ///< E = 10^((L + 0.691) / 10)
double loudness; ///< L = -0.691 + 10 * log10(E)
};
struct integrator {
double *cache[MAX_CHANNELS]; ///< window of filtered samples (N ms)
int cache_pos; ///< focus on the last added bin in the cache array
double sum[MAX_CHANNELS]; ///< sum of the last N ms filtered samples (cache content)
int filled; ///< 1 if the cache is completely filled, 0 otherwise
double rel_threshold; ///< relative threshold
double sum_kept_powers; ///< sum of the powers (weighted sums) above absolute threshold
int nb_kept_powers; ///< number of sum above absolute threshold
struct hist_entry *histogram; ///< histogram of the powers, used to compute LRA and I
};
typedef struct EBUR128Context {
const AVClass *class; ///< AVClass context for log and options purpose
/* peak metering */
int peak_mode; ///< enabled peak modes
double *true_peaks; ///< true peaks per channel
double *sample_peaks; ///< sample peaks per channel
double *true_peaks_per_frame; ///< true peaks in a frame per channel
#if CONFIG_SWRESAMPLE
SwrContext *swr_ctx; ///< over-sampling context for true peak metering
double *swr_buf; ///< resampled audio data for true peak metering
int swr_linesize;
#endif
/* video */
int do_video; ///< 1 if video output enabled, 0 otherwise
int w, h; ///< size of the video output
struct rect text; ///< rectangle for the LU legend on the left
struct rect graph; ///< rectangle for the main graph in the center
struct rect gauge; ///< rectangle for the gauge on the right
AVFrame *outpicref; ///< output picture reference, updated regularly
int meter; ///< select a EBU mode between +9 and +18
int scale_range; ///< the range of LU values according to the meter
int y_zero_lu; ///< the y value (pixel position) for 0 LU
int y_opt_max; ///< the y value (pixel position) for 1 LU
int y_opt_min; ///< the y value (pixel position) for -1 LU
int *y_line_ref; ///< y reference values for drawing the LU lines in the graph and the gauge
/* audio */
int nb_channels; ///< number of channels in the input
double *ch_weighting; ///< channel weighting mapping
int sample_count; ///< sample count used for refresh frequency, reset at refresh
/* Filter caches.
* The mult by 3 in the following is for X[i], X[i-1] and X[i-2] */
double x[MAX_CHANNELS * 3]; ///< 3 input samples cache for each channel
double y[MAX_CHANNELS * 3]; ///< 3 pre-filter samples cache for each channel
double z[MAX_CHANNELS * 3]; ///< 3 RLB-filter samples cache for each channel
#define I400_BINS (48000 * 4 / 10)
#define I3000_BINS (48000 * 3)
struct integrator i400; ///< 400ms integrator, used for Momentary loudness (M), and Integrated loudness (I)
struct integrator i3000; ///< 3s integrator, used for Short term loudness (S), and Loudness Range (LRA)
/* I and LRA specific */
double integrated_loudness; ///< integrated loudness in LUFS (I)
double loudness_range; ///< loudness range in LU (LRA)
double lra_low, lra_high; ///< low and high LRA values
/* misc */
int loglevel; ///< log level for frame logging
int metadata; ///< whether or not to inject loudness results in frames
int dual_mono; ///< whether or not to treat single channel input files as dual-mono
double pan_law; ///< pan law value used to calculate dual-mono measurements
int target; ///< target level in LUFS used to set relative zero LU in visualization
int gauge_type; ///< whether gauge shows momentary or short
int scale; ///< display scale type of statistics
} EBUR128Context;
void dump_ebur128_context(void *priv);
static int open_input_file(const char *filename)
{
int ret;
AVCodec *dec;
if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
return ret;
}
if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
return ret;
}
/* select the audio stream */
ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_AUDIO, -1, -1, &dec, 0);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find an audio stream in the input file\n");
return ret;
}
audio_stream_index = ret;
/* create decoding context */
dec_ctx = avcodec_alloc_context3(dec);
if (!dec_ctx)
return AVERROR(ENOMEM);
avcodec_parameters_to_context(dec_ctx, fmt_ctx->streams[audio_stream_index]->codecpar);
/* init the audio decoder */
if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open audio decoder\n");
return ret;
}
return 0;
}
static int init_filters(const char *filters_descr)
{
char args[512];
int ret = 0;
const AVFilter *abuffersrc = avfilter_get_by_name("abuffer");
const AVFilter *abuffersink = avfilter_get_by_name("abuffersink");
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
static const enum AVSampleFormat out_sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
static const int64_t out_channel_layouts[] = { AV_CH_LAYOUT_MONO, -1 };
static const int out_sample_rates[] = { 8000, -1 };
const AVFilterLink *outlink;
AVRational time_base = fmt_ctx->streams[audio_stream_index]->time_base;
filter_graph = avfilter_graph_alloc();
if (!outputs || !inputs || !filter_graph) {
ret = AVERROR(ENOMEM);
goto end;
}
/* buffer audio source: the decoded frames from the decoder will be inserted here. */
if (!dec_ctx->channel_layout)
dec_ctx->channel_layout = av_get_default_channel_layout(dec_ctx->channels);
snprintf(args, sizeof(args),
"time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,
time_base.num, time_base.den, dec_ctx->sample_rate,
av_get_sample_fmt_name(dec_ctx->sample_fmt), dec_ctx->channel_layout);
ret = avfilter_graph_create_filter(&buffersrc_ctx, abuffersrc, "in",
args, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
goto end;
}
/* buffer audio sink: to terminate the filter chain. */
ret = avfilter_graph_create_filter(&buffersink_ctx, abuffersink, "out",
NULL, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
goto end;
}
ret = av_opt_set_int_list(buffersink_ctx, "sample_fmts", out_sample_fmts, -1,
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n");
goto end;
}
ret = av_opt_set_int_list(buffersink_ctx, "channel_layouts", out_channel_layouts, -1,
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
goto end;
}
ret = av_opt_set_int_list(buffersink_ctx, "sample_rates", out_sample_rates, -1,
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n");
goto end;
}
/*
* Set the endpoints for the filter graph. The filter_graph will
* be linked to the graph described by filters_descr.
*/
/*
* The buffer source output must be connected to the input pad of
* the first filter described by filters_descr; since the first
* filter input label is not specified, it is set to "in" by
* default.
*/
outputs->name = av_strdup("in");
outputs->filter_ctx = buffersrc_ctx;
outputs->pad_idx = 0;
outputs->next = NULL;
/*
* The buffer sink input must be connected to the output pad of
* the last filter described by filters_descr; since the last
* filter output label is not specified, it is set to "out" by
* default.
*/
inputs->name = av_strdup("out");
inputs->filter_ctx = buffersink_ctx;
inputs->pad_idx = 0;
inputs->next = NULL;
if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
&inputs, &outputs, NULL)) < 0)
goto end;
if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
goto end;
/* Print summary of the sink buffer
* Note: args buffer is reused to store channel layout string */
outlink = buffersink_ctx->inputs[0];
av_get_channel_layout_string(args, sizeof(args), -1, outlink->channel_layout);
av_log(NULL, AV_LOG_INFO, "Output: srate:%dHz fmt:%s chlayout:%s\n",
(int)outlink->sample_rate,
(char *)av_x_if_null(av_get_sample_fmt_name(outlink->format), "?"),
args);
end:
avfilter_inout_free(&inputs);
avfilter_inout_free(&outputs);
return ret;
}
static void print_frame(const AVFrame *frame)
{
// const int n = frame->nb_samples * av_get_channel_layout_nb_channels(frame->channel_layout);
// const uint16_t *p = (uint16_t*)frame->data[0];
// const uint16_t *p_end = p + n;
//
// while (p < p_end) {
// fputc(*p & 0xff, stdout);
// fputc(*p>>8 & 0xff, stdout);
// p++;
// }
// fflush(stdout);
}
int main(int argc, char **argv)
{
av_log_set_level(AV_LOG_DEBUG);
int ret;
AVPacket packet;
AVFrame *frame = av_frame_alloc();
AVFrame *filt_frame = av_frame_alloc();
if (!frame || !filt_frame) {
perror("Could not allocate frame");
exit(1);
}
if ((ret = open_input_file(argv[1])) < 0)
goto end;
if ((ret = init_filters(filter_descr)) < 0)
goto end;
/* read all packets */
while (1) {
if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
break;
if (packet.stream_index == audio_stream_index) {
ret = avcodec_send_packet(dec_ctx, &packet);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while sending a packet to the decoder\n");
break;
}
while (ret >= 0) {
ret = avcodec_receive_frame(dec_ctx, frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
break;
} else if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while receiving a frame from the decoder\n");
goto end;
}
if (ret >= 0) {
/* push the audio data from decoded frame into the filtergraph */
if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n");
break;
}
/* pull filtered audio from the filtergraph */
while (1) {
ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
break;
if (ret < 0)
goto end;
print_frame(filt_frame);
av_frame_unref(filt_frame);
}
av_frame_unref(frame);
}
}
}
av_packet_unref(&packet);
}
if(filter_graph->nb_filters){
av_log(filter_graph, AV_LOG_INFO, "hello : %d \n",
filter_graph->nb_filters);
int i;
for (int i = 0; i < filter_graph->nb_filters; i++){
av_log(filter_graph, AV_LOG_INFO, "name : %s \n",
filter_graph->filters[i]->name);
}
}
av_log(filter_graph, AV_LOG_INFO, "name : %s \n",
filter_graph->filters[2]->name);
void* priv = filter_graph->filters[2]->priv;
dump_ebur128_context(&priv);
end:
avfilter_graph_free(&filter_graph);
avcodec_free_context(&dec_ctx);
avformat_close_input(&fmt_ctx);
av_frame_free(&frame);
av_frame_free(&filt_frame);
if (ret < 0 && ret != AVERROR_EOF) {
fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
exit(1);
}
exit(0);
}
void dump_ebur128_context(void *priv){
EBUR128Context *ebur128 = priv;
av_log(ebur128, AV_LOG_INFO, "integrated_loudness : %5.1f \n",
ebur128->integrated_loudness);
av_log(ebur128, AV_LOG_INFO, "lra_low : %5.1f \n",
ebur128->lra_low);
av_log(ebur128, AV_LOG_INFO, "lra_high : %5.1f \n",
ebur128->lra_high);
}program fails while accessing integrated loudness in dump_ebur128_context.can someone guide me about , how I should proceed in here.
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