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Video d’abeille en portrait
14 mai 2011, par
Mis à jour : Février 2012
Langue : français
Type : Video
Autres articles (63)
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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 (...) -
MediaSPIP version 0.1 Beta
16 avril 2011, parMediaSPIP 0.1 beta est la première version de MediaSPIP décrétée comme "utilisable".
Le fichier zip ici présent contient uniquement les sources de MediaSPIP en version standalone.
Pour avoir une installation fonctionnelle, 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 (...) -
Librairies et binaires spécifiques au traitement vidéo et sonore
31 janvier 2010, parLes logiciels et librairies suivantes sont utilisées par SPIPmotion d’une manière ou d’une autre.
Binaires obligatoires FFMpeg : encodeur principal, permet de transcoder presque tous les types de fichiers vidéo et sonores dans les formats lisibles sur Internet. CF ce tutoriel pour son installation ; Oggz-tools : outils d’inspection de fichiers ogg ; Mediainfo : récupération d’informations depuis la plupart des formats vidéos et sonores ;
Binaires complémentaires et facultatifs flvtool2 : (...)
Sur d’autres sites (7439)
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Date and segment comparison feature
31 octobre 2019, par Matomo Core Team — Analytics Tips, Development -
MP3 files created using FFmpeg are not starting playback in browser immediately. Is there any major difference between FFmpeg and AVCONV ?
23 janvier 2019, par AR5I am working on a website that streams music. We recently changed server from Debian (with avconv) to a Centos7 (with FFmpeg) server.
The mp3 files created on Debian server start playback on browser (I have tested Chrome and Firefox) start almost at the same time they start loading into the browser (I used Network tab on Developer Tools to monitor this)Now after the switch to Centos/FFmpeg server, the files being created on this new server are displaying a strange behavior. They only start playback after about 1MB is loaded into the browser.
I have used identical settings for converting original file into MP3 in both AVCONV and FFmpeg but the files created using FFmpeg are showing this issue. Is there something that might be causing such an issue ? Are there differences in terms of audio conversion between AVCONV and FFmpeg ?
I have already tried
I first found that the files created on old server (Debian/Avconv) were VBR (variable bitrate) and the ones created on new server were CBR (constant bitrate), so I tried switching to VBR but the issue still persisted.
I checked the mp3 files using MediaInfo app and there seems to be no difference between the files.
I also checked if both files were being served as 206 Partial Content and they both are indeed.
I am trying to create mp3 files using FFmpeg that work exactly like the ones created before using avconv
I am trying to make the streaming site work on the new server but the mp3 files created using FFmpeg are not playing back correctly as compared to the ones created on the old server. I am trying to figure out what I might be doing wrong ? or if there is a difference between avconv and FFmpeg that is causing this issue.
I am really stuck on this issue, any help will be really appreciated.
Edit
I don’t have access to old server anymore so I couldn’t retrieve the log output of avconv. The command that I was using was as follows :
avconv -y -i "/test/Track 01.mp3" -ac 2 -ar 44100 -acodec libmp3lame -b:a 128k "/test/Track 01 (converted).mp3"
Here is the command and log output from new server :
ffmpeg -y -i "/test/Track 01.mp3" -ac 2 -ar 44100 -acodec libmp3lame -b:a 128k "/test/Track 01 (converted).mp3"
ffmpeg version 2.8.15 Copyright (c) 2000-2018 the FFmpeg developers
built with gcc 4.8.5 (GCC) 20150623 (Red Hat 4.8.5-28)
configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --optflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector-strong --param=ssp-buffer-size=4 -grecord-gcc-switches -m64 -mtune=generic' --extra-ldflags='-Wl,-z,relro ' --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvo-amrwbenc --enable-version3 --enable-bzlib --disable-crystalhd --enable-gnutls --enable-ladspa --enable-libass --enable-libcdio --enable-libdc1394 --disable-indev=jack --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-openal --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libv4l2 --enable-libx264 --enable-libx265 --enable-libxvid --enable-x11grab --enable-avfilter --enable-avresample --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 --enable-runtime-cpudetect
libavutil 54. 31.100 / 54. 31.100
libavcodec 56. 60.100 / 56. 60.100
libavformat 56. 40.101 / 56. 40.101
libavdevice 56. 4.100 / 56. 4.100
libavfilter 5. 40.101 / 5. 40.101
libavresample 2. 1. 0 / 2. 1. 0
libswscale 3. 1.101 / 3. 1.101
libswresample 1. 2.101 / 1. 2.101
libpostproc 53. 3.100 / 53. 3.100
[mp3 @ 0xd60be0] Skipping 0 bytes of junk at 240044.
Input #0, mp3, from '/test/Track 01.mp3':
Metadata:
album : Future Hndrxx Presents: The WIZRD
artist : Future
genre : Hip-Hop
title : Never Stop
track : 1
lyrics-eng : rgf.is
WEB SITE : rgf.is
TAGGINGTIME : rgf.is
WEB : rgf.is
date : 2019
encoder : Lavf56.40.101
Duration: 00:04:51.40, start: 0.025056, bitrate: 121 kb/s
Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16p, 114 kb/s
Metadata:
encoder : Lavc56.60
Stream #0:1: Video: png, rgb24(pc), 333x333 [SAR 1:1 DAR 1:1], 90k tbr, 90k tbn, 90k tbc
Metadata:
comment : Cover (front)
[mp3 @ 0xd66ec0] Frame rate very high for a muxer not efficiently supporting it.
Please consider specifying a lower framerate, a different muxer or -vsync 2
Output #0, mp3, to '/test/Track 01 (converted).mp3':
Metadata:
TALB : Future Hndrxx Presents: The WIZRD
TPE1 : Future
TCON : Hip-Hop
TIT2 : Never Stop
TRCK : 1
lyrics-eng : rgf.is
WEB SITE : rgf.is
TAGGINGTIME : rgf.is
WEB : rgf.is
TDRC : 2019
TSSE : Lavf56.40.101
Stream #0:0: Video: png, rgb24, 333x333 [SAR 1:1 DAR 1:1], q=2-31, 200 kb/s, 90k fps, 90k tbn, 90k tbc
Metadata:
comment : Cover (front)
encoder : Lavc56.60.100 png
Stream #0:1: Audio: mp3 (libmp3lame), 44100 Hz, stereo, s16p, 128 kb/s
Metadata:
encoder : Lavc56.60.100 libmp3lame
Stream mapping:
Stream #0:1 -> #0:0 (png (native) -> png (native))
Stream #0:0 -> #0:1 (mp3 (native) -> mp3 (libmp3lame))
Press [q] to stop, [?] for help
[libmp3lame @ 0xd9b0c0] Trying to remove 1152 samples, but the queue is emptys/s
frame= 1 fps=0.1 q=-0.0 Lsize= 4788kB time=00:04:51.39 bitrate= 134.6kbits/s
video:234kB audio:4553kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.014809%Samples of MP3 files
I have uploaded samples of mp3 files created using both avconv and FFmpeg. Please find these here : https://drive.google.com/drive/folders/1gRTmMM2iSK0VWQ4Zaf_iBNQe5laFJl08?usp=sharing
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ffmpeg avcodec_send_packet/avcodec_receive_frame memory leak
23 janvier 2019, par G HamlinI’m attempting to decode frames, but memory usage grows with every frame (more specifically, with every call to avcodec_send_packet) until finally the code crashes with a bad_alloc. Here’s the basic decode loop :
int rfret = 0;
while((rfret = av_read_frame(inctx.get(), &packet)) >= 0){
if (packet.stream_index == vstrm_idx) {
//std::cout << "Sending Packet" << std::endl;
int ret = avcodec_send_packet(ctx.get(), &packet);
if (ret < 0 || ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
std::cout << "avcodec_send_packet: " << ret << std::endl;
break;
}
while (ret >= 0) {
//std::cout << "Receiving Frame" << std::endl;
ret = avcodec_receive_frame(ctx.get(), fr);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
//std::cout << "avcodec_receive_frame: " << ret << std::endl;
av_frame_unref(fr);
// av_frame_free(&fr);
break;
}
std::cout << "frame: " << ctx->frame_number << std::endl;
// eventually do something with the frame here...
av_frame_unref(fr);
// av_frame_free(&fr);
}
}
else {
//std::cout << "Not Video" << std::endl;
}
av_packet_unref(&packet);
}Memory usage/leakage seems to scale with the resolution of the video I’m decoding. For example, for a 3840x2160 resolution video, the memory usage in windows task manager consistently jumps up by about 8mb (1 byte per pixel ??) for each received frame. Do I need to do something besides call av_frame_unref to release the memory ?
(more) complete code below
void AVFormatContextDeleter(AVFormatContext* ptr)
{
if (ptr) {
avformat_close_input(&ptr);
}
}
void AVCodecContextDeleter(AVCodecContext* ptr)
{
if (ptr) {
avcodec_free_context(&ptr);
}
}
typedef std::unique_ptr AVFormatContextPtr;
typedef std::unique_ptr AVCodecContextPtr;
AVCodecContextPtr createAvCodecContext(AVCodec *vcodec)
{
AVCodecContextPtr ctx(avcodec_alloc_context3(vcodec), AVCodecContextDeleter);
return ctx;
}
AVFormatContextPtr createFormatContext(const std::string& filename)
{
AVFormatContext* inctxPtr = nullptr;
int ret = avformat_open_input(&inctxPtr, filename.c_str(), nullptr, nullptr);
// int ret = avformat_open_input(&inctx, "D:/Videos/test.mp4", nullptr, nullptr);
if (ret != 0) {
inctxPtr = nullptr;
}
return AVFormatContextPtr(inctxPtr, AVFormatContextDeleter);
}
int testDecode()
{
// open input file context
AVFormatContextPtr inctx = createFormatContext("D:/Videos/Matt Chapman Hi Greg.MOV");
if (!inctx) {
// std::cerr << "fail to avforamt_open_input(\"" << infile << "\"): ret=" << ret;
return 1;
}
// retrieve input stream information
int ret = avformat_find_stream_info(inctx.get(), nullptr);
if (ret < 0) {
//std::cerr << "fail to avformat_find_stream_info: ret=" << ret;
return 2;
}
// find primary video stream
AVCodec* vcodec = nullptr;
const int vstrm_idx = av_find_best_stream(inctx.get(), AVMEDIA_TYPE_VIDEO, -1, -1, &vcodec, 0);
if (vstrm_idx < 0) {
//std::cerr << "fail to av_find_best_stream: vstrm_idx=" << vstrm_idx;
return 3;
}
AVCodecParameters* origin_par = inctx->streams[vstrm_idx]->codecpar;
if (vcodec == nullptr) { // is this even necessary?
vcodec = avcodec_find_decoder(origin_par->codec_id);
if (!vcodec) {
// Can't find decoder
return 4;
}
}
AVCodecContextPtr ctx = createAvCodecContext(vcodec);
if (!ctx) {
return 5;
}
ret = avcodec_parameters_to_context(ctx.get(), origin_par);
if (ret) {
return 6;
}
ret = avcodec_open2(ctx.get(), vcodec, nullptr);
if (ret < 0) {
return 7;
}
//print input video stream informataion
std::cout
//<< "infile: " << infile << "\n"
<< "format: " << inctx->iformat->name << "\n"
<< "vcodec: " << vcodec->name << "\n"
<< "size: " << origin_par->width << 'x' << origin_par->height << "\n"
<< "fps: " << av_q2d(ctx->framerate) << " [fps]\n"
<< "length: " << av_rescale_q(inctx->duration, ctx->time_base, {1,1000}) / 1000. << " [sec]\n"
<< "pixfmt: " << av_get_pix_fmt_name(ctx->pix_fmt) << "\n"
<< "frame: " << inctx->streams[vstrm_idx]->nb_frames << "\n"
<< std::flush;
AVPacket packet;
av_init_packet(&packet);
packet.data = nullptr;
packet.size = 0;
AVFrame *fr = av_frame_alloc();
if (!fr) {
return 8;
}
int rfret = 0;
while((rfret = av_read_frame(inctx.get(), &packet)) >= 0){
if (packet.stream_index == vstrm_idx) {
//std::cout << "Sending Packet" << std::endl;
int ret = avcodec_send_packet(ctx.get(), &packet);
if (ret < 0 || ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
std::cout << "avcodec_send_packet: " << ret << std::endl;
break;
}
while (ret >= 0) {
//std::cout << "Receiving Frame" << std::endl;
ret = avcodec_receive_frame(ctx.get(), fr);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
//std::cout << "avcodec_receive_frame: " << ret << std::endl;
av_frame_unref(fr);
// av_frame_free(&fr);
break;
}
std::cout << "frame: " << ctx->frame_number << std::endl;
// do something with the frame here...
av_frame_unref(fr);
// av_frame_free(&fr);
}
}
else {
//std::cout << "Not Video" << std::endl;
}
av_packet_unref(&packet);
}
std::cout << "RFRET = " << rfret << std::endl;
return 0;
}Update 1 : (1/21/2019) Compiling on a different machine and running with different video files I am not seeing the memory usage growing without bound. I’ll try to narrow down where the difference lies (compiler ?, ffmpeg version ?, or video encoding ?)
Update 2 : (1/21/2019) Ok, it looks like there is some interaction occurring between ffmpeg and Qt’s QCamera. In my application, I’m using Qt to manage the webcam, but decided to use ffmpeg libraries to handle decoding/encoding since Qt doesn’t have as comprehensive support for different codecs. If I have the camera turned on (through Qt), ffmpeg decoding memory consumption grows without bound. If the camera is off, ffmpeg behaves fine. I’ve tried this both with a physical camera (Logitech C920) and with a virtual camera using OBS-Virtualcam, with the same result. So far I’m baffled as to how the two systems are interacting...