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Médias (2)
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SPIP - plugins - embed code - Exemple
2 septembre 2013, par
Mis à jour : Septembre 2013
Langue : français
Type : Image
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Publier une image simplement
13 avril 2011, par ,
Mis à jour : Février 2012
Langue : français
Type : Video
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Personnaliser en ajoutant son logo, sa bannière ou son image de fond
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Websites made with MediaSPIP
2 mai 2011, parThis page lists some websites based on MediaSPIP.
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(Dés)Activation de fonctionnalités (plugins)
18 février 2011, parPour gérer l’ajout et la suppression de fonctionnalités supplémentaires (ou plugins), MediaSPIP utilise à partir de la version 0.2 SVP.
SVP permet l’activation facile de plugins depuis l’espace de configuration de MediaSPIP.
Pour y accéder, il suffit de se rendre dans l’espace de configuration puis de se rendre sur la page "Gestion des plugins".
MediaSPIP est fourni par défaut avec l’ensemble des plugins dits "compatibles", ils ont été testés et intégrés afin de fonctionner parfaitement avec chaque (...)
Sur d’autres sites (8888)
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How can I play libvorbis(ogg) streams using ffplay received with udp ?
31 mars 2017, par crismanffmpeg
ffmpeg -f dshow -i audio="virtual-audio-capturer" -codec:a libvorbis -b:a 128k -ac 2 -ar 48000 -f mpegts udp://127.0.0.1:1111
ffplay
ffplay -codec:a libvorbis -b:a 128k -ar 48000 -ac 2 udp://127.0.0.1:1111
not playing audio.
Other codecs play back working, example below.
aac working
ffmpeg -f dshow -i audio="virtual-audio-capturer" -codec:a aac -b:a 128k -ac 2 -ar 48000 -f mpegts udp://127.0.0.1:1111
ffplay -codec:a aac -b:a 128k -ar 48000 -ac 2 udp://127.0.0.1:1111opus codec working
ffmpeg -f dshow -i audio="virtual-audio-capturer" -codec:a libopus -b:a 128k -ac 2 -ar 48000 -f mpegts udp://127.0.0.1:1111
ffplay -codec:a libopus -b:a 128k -ar 48000 -ac 2 udp://127.0.0.1:1111mp3 working
ffmpeg -f dshow -i audio="virtual-audio-capturer" -codec:a mp3 -b:a 128k -ac 2 -ar 48000 -f mpegts udp://127.0.0.1:1111
ffplay -codec:a mp3 -b:a 128k -ar 48000 -ac 2 udp://127.0.0.1:1111
why libvorbis not working ?
transport success, but can not play, i think.
// not woring ffplay displayed log
ffplay.exe -loglevel debug -codec:a vorbis -b:a 128k -ar 48000 -ac 2 -sync audio -i udp://127.0.0.1:1111
ffplay version N-83781-g3016e91 Copyright (c) 2003-2017 the FFmpeg developers
built with gcc 6.3.0 (GCC)
configuration: --enable-gpl --enable-version3 --enable-cuda --enable-cuvid --enable-d3d11va --enable-dxva2 --enable-libmfx --enable-nvenc --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-zlib
libavutil 55. 48.100 / 55. 48.100
libavcodec 57. 82.102 / 57. 82.102
libavformat 57. 66.103 / 57. 66.103
libavdevice 57. 3.100 / 57. 3.100
libavfilter 6. 74.100 / 6. 74.100
libswscale 4. 3.101 / 4. 3.101
libswresample 2. 4.100 / 2. 4.100
libpostproc 54. 2.100 / 54. 2.100
[udp @ 0000000000b68700] No default whitelist set sq= 0B f=0/0
[udp @ 0000000000b68700] end receive buffer size reported is 65536
[mpegts @ 0000000000b68ac0] Format mpegts probed with size=2048 and score=50
[mpegts @ 0000000000b68ac0] stream=0 stream_type=6 pid=100 prog_reg_desc=
[mpegts @ 0000000000b68ac0] Before avformat_find_stream_info() pos: 0 bytes read:8208 seeks:0 nb_streams:1
[mpegts @ 0000000000b68ac0] probing stream 0 pp:2500
[mpegts @ 0000000000b68ac0] probing stream 0 pp:2499
[mpegts @ 0000000000b68ac0] probing stream 0 pp:2498
[mpegts @ 0000000000b68ac0] probing stream 0 pp:2497
[mpegts @ 0000000000b68ac0] probing stream 0 pp:2496
[mpegts @ 0000000000b68ac0] probing stream 0 pp:2495
[mpegts @ 0000000000b68ac0] probing stream 0 pp:2494= 0B f=0/0
[mpegts @ 0000000000b68ac0] Probe with size=10038, packets=7 detected mp3 with score=1
[mpegts @ 0000000000b68ac0] probing stream 0 pp:2493= 0B f=0/0
[mpegts @ 0000000000b68ac0] probing stream 0 pp:2492= 0B f=0/0
[mpegts @ 0000000000b68ac0] probing stream 0 pp:2491= 0B f=0/0
[mpegts @ 0000000000b68ac0] Probe with size=18321, packets=10 detected mp3 with score=1
[mpegts @ 0000000000b68ac0] probing stream 0 pp:2490= 0B f=0/0 -
ffprobe : fix printing packet side data information
25 mars 2017, par James Almer -
Copying audio stream via ffmpeg c api
28 mars 2017, par David BarishevI’m trying to replicate ffmpeg command
ffmpeg -i <input /> -vn -acodec copy <output></output>
,using the C api, in my c++ android project.There isn’t too much documentations on the subject, and some of the methods in the example are deprecated and i don’t know how to fix them.
Here is the code i have worked on, and understood,i have highlited some parts in with the comments :
extern "C"
int extract_audio(const char *in_filename,const char *out_filename){
// Library init
SSL_load_error_strings();
SSL_library_init();
av_register_all ();
avformat_network_init ();
//Logging init
av_log_set_callback(log_callback_android);
//Variable init
AVOutputFormat *ofmt = NULL;
AVFormatContext *ifmt_ctx = NULL, *ofmt_ctx = NULL;
AVPacket pkt;
int ret, i;
int stream_index = 0;
//Open input file
if ((ret = avformat_open_input(&ifmt_ctx, in_filename, 0, 0)) < 0) {
av_log(NULL,AV_LOG_FATAL, "Could not open input file '%s'", in_filename);
goto end;
}
if ((ret = avformat_find_stream_info(ifmt_ctx, 0)) < 0) {
av_log(NULL,AV_LOG_FATAL, "Failed to retrieve input stream information");
goto end;
}
//Print input file stream info
av_dump_format(ifmt_ctx, 0, in_filename, 0);
//Allocate new context for output file
avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, out_filename);
if (!ofmt_ctx) {
av_log(NULL,AV_LOG_FATAL, "Could not create output context");
ret = AVERROR_UNKNOWN;
goto end;
}
ofmt = ofmt_ctx->oformat;
// Iterate over all streams in input file
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
//Stream input
AVStream *in_stream = ifmt_ctx->streams[i];
//Filter non audio streams
if(in_stream->codecpar->codec_type == AVMEDIA_TYPE_AUDIO){
__android_log_print(ANDROID_LOG_INFO,APPNAME,"Audio stream found");
AVStream *out_stream = avformat_new_stream(ofmt_ctx, NULL);
if (!out_stream) {
__android_log_print(ANDROID_LOG_ERROR, APPNAME, "Failed allocating output stream\n");
ret = AVERROR_UNKNOWN;
goto end;
}
if ((ret = avcodec_parameters_copy(out_stream->codecpar, in_stream->codecpar)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Copying parameters for stream #%u failed\n", i);
return ret;
}
out_stream->time_base = in_stream->time_base;
}
}
//Dump output format
av_dump_format(ofmt_ctx, 0, out_filename, 1);
//Open output file
if (!(ofmt->flags & AVFMT_NOFILE)) {
ret = avio_open(&ofmt_ctx->pb, out_filename, AVIO_FLAG_WRITE);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Could not open output file '%s'", out_filename);
goto end;
}
}
//Writing output header
if ((ret = avformat_write_header(ofmt_ctx, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Error occurred when opening output file");
goto end;
}
while (1) {
AVStream *in_stream, *out_stream;
if ((ret = av_read_frame(ifmt_ctx, &pkt)) < 0)
break;
stream_index = pkt.stream_index;
/* remux this frame without reencoding */
av_packet_rescale_ts(&pkt,
ifmt_ctx->streams[stream_index]->time_base,
ofmt_ctx->streams[stream_index]->time_base);
if ((ret = ret = av_interleaved_write_frame(ofmt_ctx, &pkt)) < 0){
av_log(NULL,AV_LOG_FATAL,"Error muxing packet");
goto end;
}
av_packet_unref(&pkt);
}
av_write_trailer(ofmt_ctx);
end:
avformat_close_input(&ifmt_ctx);
/* close output */
if (ofmt_ctx && !(ofmt->flags & AVFMT_NOFILE))
avio_closep(&ofmt_ctx->pb);
avformat_free_context(ofmt_ctx);
if (ret < 0 && ret != AVERROR_EOF) {
av_log(NULL,AV_LOG_FATAL, "Error: %s", av_err2str(ret));
}
}This code produces the following output :
===Input Information===
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '<input />':
Metadata:
major_brand :
isom
minor_version :
512
compatible_brands:
isomiso2avc1mp41
encoder :
Lavf57.25.100
Duration:
00:00:58.82
, start:
0.000000
, bitrate:
7369 kb/s
Stream #0:0
(eng)
Audio: aac (mp4a / 0x6134706D), 44100 Hz, 2 channels, 160 kb/s
(default)
Metadata:
handler_name :
SoundHandler
Stream #0:1
(eng)
Video: h264 (avc1 / 0x31637661), none, 640x640, 7213 kb/s
30 fps,
30 tbr,
15360 tbn,
15360 tbc
(default)
Metadata:
handler_name :
VideoHandler
===Ouput Information===
Audio stream found
Output #0, adts, to 'out.aac':
Stream #0:0
Audio: aac (mp4a / 0x6134706D), 44100 Hz, 2 channels, 160 kb/s
==Packet Logging==
in: pts:-2048 pts_time:-0.0464399 dts:-2048 dts_time:-0.0464399 duration:1024 duration_time:0.02322 stream_index:0
out: pts:-2048 pts_time:-0.0464399 dts:-2048 dts_time:-0.0464399 duration:1024 duration_time:0.02322 stream_index:0
in: pts:-1024 pts_time:-0.02322 dts:-1024 dts_time:-0.02322 duration:1024 duration_time:0.02322 stream_index:0
out: pts:-1024 pts_time:-0.02322 dts:-1024 dts_time:-0.02322 duration:1024 duration_time:0.02322 stream_index:0
in: pts:0 pts_time:0 dts:0 dts_time:0 duration:1024 duration_time:0.02322 stream_index:0
out: pts:0 pts_time:0 dts:0 dts_time:0 duration:1024 duration_time:0.02322 stream_index:0
in: pts:0 pts_time:0 dts:0 dts_time:0 duration:512 duration_time:0.0333333 stream_index:1The snippit handlers file opening, context creation, stream codec copying fine, but crashes after a few loops in the packet muxing part, with signal error :
Signal: SIGSEGV (signal SIGSEGV: invalid address (fault address: 0x28))
What did i do wrong ? I wrote this code using some examples and got the code to the current state, but i’m lost now, any help would be greatly appreciated !