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Carte de Schillerkiez
13 mai 2011, par
Mis à jour : Septembre 2011
Langue : English
Type : Texte
Autres articles (99)
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Automated installation script of MediaSPIP
25 avril 2011, parTo overcome the difficulties mainly due to the installation of server side software dependencies, an "all-in-one" installation script written in bash was created to facilitate this step on a server with a compatible Linux distribution.
You must have access to your server via SSH and a root account to use it, which will install the dependencies. Contact your provider if you do not have that.
The documentation of the use of this installation script is available here.
The code of this (...) -
Les statuts des instances de mutualisation
13 mars 2010, parPour des raisons de compatibilité générale du plugin de gestion de mutualisations avec les fonctions originales de SPIP, les statuts des instances sont les mêmes que pour tout autre objets (articles...), seuls leurs noms dans l’interface change quelque peu.
Les différents statuts possibles sont : prepa (demandé) qui correspond à une instance demandée par un utilisateur. Si le site a déjà été créé par le passé, il est passé en mode désactivé. publie (validé) qui correspond à une instance validée par un (...) -
Que fait exactement ce script ?
18 janvier 2011, parCe script est écrit en bash. Il est donc facilement utilisable sur n’importe quel serveur.
Il n’est compatible qu’avec une liste de distributions précises (voir Liste des distributions compatibles).
Installation de dépendances de MediaSPIP
Son rôle principal est d’installer l’ensemble des dépendances logicielles nécessaires coté serveur à savoir :
Les outils de base pour pouvoir installer le reste des dépendances Les outils de développements : build-essential (via APT depuis les dépôts officiels) ; (...)
Sur d’autres sites (11592)
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How to mux video and audio without using FFMPEG ?
20 janvier 2019, par Yekta SarıoğluI need to mux video(mp4) and audio(wav) into a single MP4 file on the Android platform. I don’t want to use FFMPEG due to its legal issues. I tried to do with MediaMuxer. And I used this solution as an example Muxer Example but I could not succeed and got an error failed to instantiate extractor where you set data source as a string path. I looked up for more answers but wherever I search on the Internet, all I found was FFMPEG based on solutions. I could not find any reliable answers. I hope you guys can help with it. Thanks in advance.
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nginx RTMP to HLS : FFMPG error when trying multiple bitrate output [closed]
28 mai 2014, par user3685074I’m currently trying to convert my RTMP Livestream into a HLS with 3 quality-settings.
I followed this guide
I’ve compiled my own FFMPEG and it’s working if I just convert 1 file.
It seems libx264 isn’t able to do multiple encodings at the same time ?I’m using these command :
exec /usr/local/bin/ffmpeg -i rtmp://localhost/src/$name
-c:a libfdk_aac -b:a 32k -c:v libx264 -b:v 128K -f flv rtmp://localhost/hls/$name_low
-c:a libfdk_aac -b:a 64k -c:v libx264 -b:v 256K -f flv rtmp://localhost/hls/$name_mid
-c:a libfdk_aac -b:a 128k -c:v libx264 -b:v 512K -f flv rtmp://localhost/hls/$name_hi 2>>/tmp/ffmpeg.log;this is the output :
ffmpeg version N-63519-g61917a1 Copyright (c) 2000-2014 the FFmpeg developers
built on May 28 2014 18:06:42 with gcc 4.4.3 (Ubuntu 4.4.3-4ubuntu5.1)
configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libfaac --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-x11grab --enable-libvpx --enable-libmp3lame --enable-librtmp --enable-libspeex --enable-libfdk_aac
libavutil 52. 87.100 / 52. 87.100
libavcodec 55. 65.100 / 55. 65.100
libavformat 55. 41.100 / 55. 41.100
libavdevice 55. 13.101 / 55. 13.101
libavfilter 4. 5.100 / 4. 5.100
libswscale 2. 6.100 / 2. 6.100
libswresample 0. 19.100 / 0. 19.100
libpostproc 52. 3.100 / 52. 3.100
Metadata:
Server NGINX RTMP (github.com/arut/nginx-rtmp-module)
width 1280.00
height 720.00
displayWidth 1280.00
displayHeight 720.00
duration 0.00
framerate 25.00
fps 25.00
videodatarate 390.00
videocodecid 0.00
audiodatarate 27.00
audiocodecid 11.00
Input #0, flv, from 'rtmp://localhost/src/test':
Metadata:
Server : NGINX RTMP (github.com/arut/nginx-rtmp-module)
displayWidth : 1280
displayHeight : 720
fps : 25
profile :
level :
Duration: 00:00:00.00, start: 0.080000, bitrate: N/A
Stream #0:0: Video: h264 (High), yuv420p, 1280x720, 399 kb/s, 25 fps, 25 tbr, 1k tbn, 50 tbc
Stream #0:1: Audio: speex, 16000 Hz, mono, s16, 27 kb/s
[libx264 @ 0x5260380] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2
[libx264 @ 0x5260380] profile High, level 3.1
[libx264 @ 0x5260380] 264 - core 142 r2431 f23da7c - H.264/MPEG-4 AVC codec - Copyleft 2003-2014 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=24 lookahead_threads=4 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=abr mbtree=1 bitrate=128 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
[libx264 @ 0x525a920] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2
Output #0, flv, to 'rtmp://localhost/hls/test_low':
Metadata:
Server : NGINX RTMP (github.com/arut/nginx-rtmp-module)
displayWidth : 1280
displayHeight : 720
fps : 25
profile :
level :
Stream #0:0: Video: h264 (libx264), yuv420p, 1280x720, q=-1--1, 128 kb/s, 25 fps, 90k tbn, 25 tbc
Metadata:
encoder : Lavc55.65.100 libx264
Stream #0:1: Audio: aac (libfdk_aac), 16000 Hz, mono, s16, 32 kb/s
Metadata:
encoder : Lavc55.65.100 libfdk_aac
Output #1, flv, to 'rtmp://localhost/hls/test_mid':
Metadata:
Server : NGINX RTMP (github.com/arut/nginx-rtmp-module)
displayWidth : 1280
displayHeight : 720
fps : 25
profile :
level :
Stream #1:0: Video: h264, yuv420p, 1280x720, q=-1--1, 256 kb/s, 25 fps, 90k tbn, 25 tbc
Metadata:
encoder : Lavc55.65.100 libx264
Stream #1:1: Audio: aac, 16000 Hz, mono, s16
Metadata:
encoder : Lavc55.65.100 libfdk_aac
Output #2, flv, to 'rtmp://localhost/hls/test_hi':
Metadata:
Server : NGINX RTMP (github.com/arut/nginx-rtmp-module)
displayWidth : 1280
displayHeight : 720
fps : 25
profile :
level :
Stream #2:0: Video: h264, yuv420p, 1280x720, q=-1--1, 25 fps, 90k tbn, 25 tbc
Metadata:
encoder : Lavc55.65.100 libx264
Stream #2:1: Audio: aac, 16000 Hz, mono, s16
Metadata:
encoder : Lavc55.65.100 libfdk_aac
Stream mapping:
Stream #0:0 -> #0:0 (h264 -> libx264)
Stream #0:1 -> #0:1 (libspeex -> libfdk_aac)
Stream #0:0 -> #1:0 (h264 -> libx264)
Stream #0:1 -> #1:1 (libspeex -> libfdk_aac)
Stream #0:0 -> #2:0 (h264 -> libx264)
Stream #0:1 -> #2:1 (libspeex -> libfdk_aac)
Error while opening encoder for output stream #1:0 - maybe incorrect parameters such as bit_rate, rate, width or heightI hope you can help me and sorry for my bad english.
Greetz
Kevin -
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.
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