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Keeping control of your media in your hands
13 avril 2011, parThe vocabulary used on this site and around MediaSPIP in general, aims to avoid reference to Web 2.0 and the companies that profit from media-sharing.
While using MediaSPIP, you are invited to avoid using words like "Brand", "Cloud" and "Market".
MediaSPIP is designed to facilitate the sharing of creative media online, while allowing authors to retain complete control of their work.
MediaSPIP aims to be accessible to as many people as possible and development is based on expanding the (...) -
Personnaliser en ajoutant son logo, sa bannière ou son image de fond
5 septembre 2013, parCertains thèmes prennent en compte trois éléments de personnalisation : l’ajout d’un logo ; l’ajout d’une bannière l’ajout d’une image de fond ;
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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 (...)
Sur d’autres sites (6667)
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FFMpeg compilation error "Undefined symbols for architecture x86_64"
23 décembre 2011, par Saptarshi BiswasI am trying to compile ffmpeg (0.9.0.git) with x264(0.120) like this
./configure --enable-gpl --enable-libmp3lame \
--enable-static --disable-shared --enable-libx264 \
--enable-pthreads --disable-doc --enable-avfilter \
--disable-ffplay --disable-ffserver --disable-ffprobe
makeand I encounter the following error :
LD ffmpeg_g
Undefined symbols for architecture x86_64:
"_MM_FIX_0_707106781", referenced from:
_filter in libavfilter.a(vf_fspp.o)
"_MM_FIX_0_541196100", referenced from:
_filter in libavfilter.a(vf_fspp.o)
ld: symbol(s) not found for architecture x86_64
collect2: ld returned 1 exit status
make: *** [ffmpeg_g] Error 1Any idea what the fix might be ?
Update : Might be unrelated but this discussion https://trac.macports.org/ticket/30093 says the issue was ffmpeg picking up llvm instead of gcc-4.2 but then this had already fixed in 0.7.4
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Linking to libx264 library from c code Ubuntu
30 janvier 2012, par Martin ConaghanI'm trying to write a small C application which uses the x264 API, and I'm having problems compiling the code with a link to the x264 libaray.
In the /project/ directory there are two sub-folders :
/project/mycode/ and
/project/x264-snapshot-20120120-2245.I have installed x264 in the latter subdirectory using ./configure and then 'make'. As such the library I think I want to link to is /project/x264-snapshot-20120120-2245/libx264.a
In /project/mycode/ I have a single source code file (prototype.c), which has the following imports :
#include
#include
#include "../x264-snapshot-20120120-2245/x264_config.h"
#include "../x264-snapshot-20120120-2245/x264.h"As expected, if I try to compile without linking to the x264 library, I get an error :
/project/mycode: gcc -o prototype prototype.c
/tmp/cc5NwRTp.o: In function `main':
prototype.c:(.text+0x6c): undefined reference to `x264_param_default_preset'
prototype.c:(.text+0xf6): undefined reference to `x264_param_apply_profile'
collect2: ld returned 1 exit statusSo I try to link the library I mentioned above, but it isn't found :
/project/mycode: gcc -o prototype prototype.c -I../x264-snapshot-20120120-2245/ -llibx264.a
/usr/bin/ld: cannot find -llibx264.a
collect2: ld returned 1 exit statusI've tried a few variations, like :
gcc -o prototype prototype.c -I../x264-snapshot-20120120-2245/ -l ../x264-snapshot-20120120-2245/libx264.a
gcc -o prototype prototype.c -I../x264-snapshot-20120120-2245/ -llibx264
gcc -I ../x264-snapshot-20120120-2245/ -llibx264.a -o prototype prototype.cAs is probably obvious by now, I'm fairly new to this, so I'm hoping there is an easy solution
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How to concat mp4 files using libffmpeg in c program ?
1er août 2013, par chichienI know ffmpeg command line is easy, but how to programmatically implement? I'm not good at this,here is some code from internet, it is used to convert .mp4 to .ts,and i made some changes,but the audio stream problem persists:
#include
#include
#include
#include
#include "libavformat/avformat.h"
#include "libavcodec/avcodec.h"
#include "libavutil/avutil.h"
#include "libavutil/rational.h"
#include "libavdevice/avdevice.h"
#include "libavutil/mathematics.h"
#include "libswscale/swscale.h"
static AVStream* add_output_stream(AVFormatContext* output_format_context, AVStream* input_stream)
{
AVCodecContext* input_codec_context = NULL;
AVCodecContext* output_codec_context = NULL;
AVStream* output_stream = NULL;
output_stream = av_new_stream(output_format_context, 0);
if (!output_stream)
{
printf("Call av_new_stream function failed\n");
return NULL;
}
input_codec_context = input_stream->codec;
output_codec_context = output_stream->codec;
output_codec_context->codec_id = input_codec_context->codec_id;
output_codec_context->codec_type = input_codec_context->codec_type;
output_codec_context->codec_tag = input_codec_context->codec_tag;
output_codec_context->bit_rate = input_codec_context->bit_rate;
output_codec_context->extradata = input_codec_context->extradata;
output_codec_context->extradata_size = input_codec_context->extradata_size;
if (av_q2d(input_codec_context->time_base) * input_codec_context->ticks_per_frame > av_q2d(input_stream->time_base) && av_q2d(input_stream->time_base) < 1.0 / 1000)
{
output_codec_context->time_base = input_codec_context->time_base;
output_codec_context->time_base.num *= input_codec_context->ticks_per_frame;
}
else
{
output_codec_context->time_base = input_stream->time_base;
}
switch (input_codec_context->codec_type)
{
case AVMEDIA_TYPE_AUDIO:
output_codec_context->channel_layout = input_codec_context->channel_layout;
output_codec_context->sample_rate = input_codec_context->sample_rate;
output_codec_context->channels = input_codec_context->channels;
output_codec_context->frame_size = input_codec_context->frame_size;
if ((input_codec_context->block_align == 1 && input_codec_context->codec_id == CODEC_ID_MP3) || input_codec_context->codec_id == CODEC_ID_AC3)
{
output_codec_context->block_align = 0;
}
else
{
output_codec_context->block_align = input_codec_context->block_align;
}
break;
case AVMEDIA_TYPE_VIDEO:
output_codec_context->pix_fmt = input_codec_context->pix_fmt;
output_codec_context->width = input_codec_context->width;
output_codec_context->height = input_codec_context->height;
output_codec_context->has_b_frames = input_codec_context->has_b_frames;
if (output_format_context->oformat->flags & AVFMT_GLOBALHEADER)
{
output_codec_context->flags |= CODEC_FLAG_GLOBAL_HEADER;
}
break;
default:
break;
}
return output_stream;
}
//[[** from ffmpeg.c
static void write_frame(AVFormatContext *s, AVPacket *pkt, AVCodecContext *avctx, AVBitStreamFilterContext *bsfc){
int ret;
while(bsfc){
AVPacket new_pkt= *pkt;
int a= av_bitstream_filter_filter(bsfc, avctx, NULL,
&new_pkt.data, &new_pkt.size,
pkt->data, pkt->size,
pkt->flags & AV_PKT_FLAG_KEY);
if(a>0){
av_free_packet(pkt);
new_pkt.destruct= av_destruct_packet;
} else if(a<0){
fprintf(stderr, "%s failed for stream %d, codec %s\n",
bsfc->filter->name, pkt->stream_index,
avctx->codec ? avctx->codec->name : "copy");
//print_error("", a);
//if (exit_on_error)
// ffmpeg_exit(1);
}
*pkt= new_pkt;
bsfc= bsfc->next;
}
ret= av_interleaved_write_frame(s, pkt);
if(ret < 0){
//print_error("av_interleaved_write_frame()", ret);
fprintf(stderr, "av_interleaved_write_frame(%d)\n", ret);
exit(1);
}
}
//]]**
int main(int argc, char* argv[])
{
const char* input;
const char* output;
const char* output_prefix = NULL;
char* segment_duration_check = 0;
const char* index = NULL;
char* tmp_index = NULL;
const char* http_prefix = NULL;
long max_tsfiles = NULL;
double prev_segment_time = 0;
double segment_duration = 0;
AVInputFormat* ifmt = NULL;
AVOutputFormat* ofmt = NULL;
AVFormatContext* ic = NULL;
AVFormatContext* oc = NULL;
AVStream* video_st = NULL;
AVStream* audio_st = NULL;
AVCodec* codec = NULL;
AVDictionary* pAVDictionary = NULL;
long frame_count = 0;
if (argc != 3) {
fprintf(stderr, "Usage: %s inputfile outputfile\n", argv[0]);
exit(1);
}
input = argv[1];
output = argv[2];
av_register_all();
char szError[256] = {0};
int nRet = avformat_open_input(&ic, input, ifmt, &pAVDictionary);
if (nRet != 0)
{
av_strerror(nRet, szError, 256);
printf(szError);
printf("\n");
printf("Call avformat_open_input function failed!\n");
return 0;
}
if (av_find_stream_info(ic) < 0)
{
printf("Call av_find_stream_info function failed!\n");
return 0;
}
ofmt = av_guess_format("mpegts", NULL, NULL);
if (!ofmt)
{
printf("Call av_guess_format function failed!\n");
return 0;
}
oc = avformat_alloc_context();
if (!oc)
{
printf("Call av_guess_format function failed!\n");
return 0;
}
oc->oformat = ofmt;
int video_index = -1, audio_index = -1;
for (unsigned int i = 0; i < ic->nb_streams && (video_index < 0 || audio_index < 0); i++)
{
switch (ic->streams[i]->codec->codec_type)
{
case AVMEDIA_TYPE_VIDEO:
video_index = i;
ic->streams[i]->discard = AVDISCARD_NONE;
video_st = add_output_stream(oc, ic->streams[i]);
break;
case AVMEDIA_TYPE_AUDIO:
audio_index = i;
ic->streams[i]->discard = AVDISCARD_NONE;
audio_st = add_output_stream(oc, ic->streams[i]);
break;
default:
ic->streams[i]->discard = AVDISCARD_ALL;
break;
}
}
codec = avcodec_find_decoder(video_st->codec->codec_id);
if (codec == NULL)
{
printf("Call avcodec_find_decoder function failed!\n");
return 0;
}
if (avcodec_open(video_st->codec, codec) < 0)
{
printf("Call avcodec_open function failed !\n");
return 0;
}
if (avio_open(&oc->pb, output, AVIO_FLAG_WRITE) < 0)
{
return 0;
}
if (avformat_write_header(oc, &pAVDictionary))
{
printf("Call avformat_write_header function failed.\n");
return 0;
}
//[[++
AVBitStreamFilterContext *bsfc = av_bitstream_filter_init("h264_mp4toannexb");
//AVBitStreamFilterContext *absfc = av_bitstream_filter_init("aac_adtstoasc");
if (!bsfc) {
fprintf(stderr, "bsf init error!\n");
return -1;
}
//]]++
int decode_done = 0;
do
{
double segment_time = 0;
AVPacket packet;
decode_done = av_read_frame(ic, &packet);
if (decode_done < 0)
break;
if (av_dup_packet(&packet) < 0)
{
printf("Call av_dup_packet function failed\n");
av_free_packet(&packet);
break;
}
//[[**
if (packet.stream_index == audio_index) {
segment_time = (double)audio_st->pts.val * audio_st->time_base.num / audio_st->time_base.den;
nRet = av_interleaved_write_frame(oc, &packet);
} else if (packet.stream_index == video_index) {
if (packet.flags & AV_PKT_FLAG_KEY) {
segment_time = (double)video_st->pts.val * video_st->time_base.num / video_st->time_base.den;
} else {
segment_time = prev_segment_time;
}
//nRet = av_interleaved_write_frame(oc, &packet);
write_frame(oc, &packet, video_st->codec, bsfc);
}
//]]**
if (nRet < 0)
{
printf("Call av_interleaved_write_frame function failed: %d\n", nRet);
}
else if (nRet > 0)
{
printf("End of stream requested\n");
av_free_packet(&packet);
break;
}
av_free_packet(&packet);
frame_count++;
}while(!decode_done);
av_write_trailer(oc);
printf("frame_count = %d\n", frame_count);
av_bitstream_filter_close(bsfc);
avcodec_close(video_st->codec);
for(unsigned int k = 0; k < oc->nb_streams; k++)
{
av_freep(&oc->streams[k]->codec);
av_freep(&oc->streams[k]);
}
av_free(oc);
//getchar();
return 0;
}Compile this code, to got an executable file named
muxts
, and then :$ ./muxts vid1.mp4 vid1.ts
No error message printed,but the audio stream was unsynchronized and noise。Check the .ts file using ffmpeg :
$ ffmpeg -i vid1.ts
ffmpeg version 0.8.14-tessus, Copyright (c) 2000-2013 the FFmpeg developers
built on Jul 29 2013 17:05:18 with llvm_gcc 4.2.1 (Based on Apple Inc. build 5658) (LLVM build 2336.1.00)
configuration: --prefix=/usr/local --arch=x86_64 --as=yasm --extra-version=tessus --enable-gpl --enable-nonfree --enable-version3 --disable-ffplay --enable-libvorbis --enable-libmp3lame --enable-libx264 --enable-libxvid --enable-bzlib --enable-zlib --enable-postproc --enable-filters --enable-runtime-cpudetect --enable-debug=3 --disable-optimizations
libavutil 51. 9. 1 / 51. 9. 1
libavcodec 53. 8. 0 / 53. 8. 0
libavformat 53. 5. 0 / 53. 5. 0
libavdevice 53. 1. 1 / 53. 1. 1
libavfilter 2. 23. 0 / 2. 23. 0
libswscale 2. 0. 0 / 2. 0. 0
libpostproc 51. 2. 0 / 51. 2. 0
Seems stream 0 codec frame rate differs from container frame rate: 180000.00 (180000/1) -> 90000.00 (180000/2)
Input #0, mpegts, from 'vid1.ts':
Duration: 00:00:03.75, start: 0.000000, bitrate: 3656 kb/s
Program 1
Metadata:
service_name : Service01
service_provider: FFmpeg
Stream #0.0[0x100]: Video: h264 (Baseline), yuv420p, 640x480, 90k tbr, 90k tbn, 180k tbc
Stream #0.1[0x101]: Audio: aac, 48000 Hz, mono, s16, 190 kb/s
At least one output file must be specifiedWhat should i do ?
If this issue fixed , how can i concat multi .ts files into single .mp4 file ?