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SWFUpload Process
6 septembre 2011, par
Mis à jour : Septembre 2011
Langue : français
Type : Texte
Autres articles (73)
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Publier sur MédiaSpip
13 juin 2013Puis-je poster des contenus à partir d’une tablette Ipad ?
Oui, si votre Médiaspip installé est à la version 0.2 ou supérieure. Contacter au besoin l’administrateur de votre MédiaSpip pour le savoir -
Submit bugs and patches
13 avril 2011Unfortunately a software is never perfect.
If you think you have found a bug, report it using our ticket system. Please to help us to fix it by providing the following information : the browser you are using, including the exact version as precise an explanation as possible of the problem if possible, the steps taken resulting in the problem a link to the site / page in question
If you think you have solved the bug, fill in a ticket and attach to it a corrective patch.
You may also (...) -
MediaSPIP 0.1 Beta version
25 avril 2011, parMediaSPIP 0.1 beta is the first version of MediaSPIP proclaimed as "usable".
The zip file provided here only contains the sources of MediaSPIP in its standalone version.
To get a working installation, you must manually install all-software dependencies on the server.
If you want to use this archive for an installation in "farm mode", you will also need to proceed to other manual (...)
Sur d’autres sites (9137)
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Extracting frames from a video does not work correctly [closed]
13 avril 2024, par Al TilmidhUsing the libraries (libav) and (ffmpeg), I try to extract frames as
.jpgfiles from avideo.mp4, the problem is that my program crashes when I use theCV_8UC3parameter, but by changing this parameter toCV_8UC1, the extracted images end up without color (grayscale), I don't really know what I missed, here is a minimal code to reproduce the two situations :

#include <opencv2></opencv2>opencv.hpp>

extern "C"
{
#include <libavformat></libavformat>avformat.h>
#include <libavcodec></libavcodec>avcodec.h>
}

int main()
{
 AVFormatContext *formatContext = nullptr;

 if (avformat_open_input(&formatContext, "video.mp4", nullptr, nullptr) != 0)
 {
 return -1;
 }

 if (avformat_find_stream_info(formatContext, nullptr) < 0)
 {
 return -1;
 }

 AVPacket packet;
 const AVCodec *codec = nullptr;
 AVCodecContext *codecContext = nullptr;

 int videoStreamIndex = av_find_best_stream(formatContext, AVMEDIA_TYPE_VIDEO, -1, -1, &codec, 0);
 if (videoStreamIndex < 0)
 {
 return -1;
 }

 codecContext = avcodec_alloc_context3(codec);
 avcodec_parameters_to_context(codecContext, formatContext->streams[videoStreamIndex]->codecpar);

 if (avcodec_open2(codecContext, codec, nullptr) < 0)
 {
 return -1;
 }

 AVFrame *frame = av_frame_alloc();

 while (av_read_frame(formatContext, &packet) >= 0)
 {
 if (packet.stream_index == videoStreamIndex)
 {
 int response = avcodec_send_packet(codecContext, &packet);
 
 if (response < 0)
 {
 break;
 }

 while (response >= 0)
 {
 response = avcodec_receive_frame(codecContext, frame);
 if (response == AVERROR(EAGAIN))
 {
 // NO FRAMES
 break;
 }

 else if (response == AVERROR_EOF)
 {
 // END OF FILE
 break;
 }

 else if (response < 0)
 {
 break;
 }

 //cv::Mat frameMat(frame->height, frame->width, CV_8UC3, frame->data[0]); // CV_8UC3 → THE PROGRAM CRASHES
 cv::Mat frameMat(frame->height, frame->width, CV_8UC1, frame->data[0]); // CV_8UC1 → WORK BUT IMAGES ARE IN GRAYSCALE
 cv::imwrite("frame_" + std::to_string(frame->pts) + ".jpg", frameMat);
 }
 }

 av_packet_unref(&packet);
 }

 av_frame_free(&frame);
 avcodec_free_context(&codecContext);
 avformat_close_input(&formatContext);

 return 0;
}


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Playing RTSP in WPF application with low latency using FFMPEG / FFMediaElement (FFME)
22 mars 2019, par PabokaI’m trying to use FFMediaElement (FFME, WPF MediaElement replacement based on FFmpeg) component to play RSTP live video in my WPF application.
I have a good connection to my camera and I want to play it with minimum available latency.
I’ve reduced the latency by changing
ProbeSizeto its minimal value :private void Media_OnMediaInitializing(object Sender, MediaInitializingRoutedEventArgs e)
{
e.Configuration.GlobalOptions.ProbeSize = 32;
}But I still have about 1 second of latency since the very beginning of the stream. I mean, when I start playing, I have to wait for 1 second till the video appears and then I have 1s of latency.
I’ve also tried to change following parameters :
e.Configuration.GlobalOptions.EnableReducedBuffering = true;
e.Configuration.GlobalOptions.FlagNoBuffer = true;
e.Configuration.GlobalOptions.MaxAnalyzeDuration = TimeSpan.Zero;but it gave no result.
I measured time-interval between FFmpeg output lines (the number in the first column is the time, elapsed from the previous line, ms)
---- OpenCommand: Entered
39 FFInterop.Initialize: FFmpeg v4.0
0 EVENT START: MediaInitializing
0 EVENT DONE : MediaInitializing
379 EVENT START: MediaOpening
0 EVENT DONE : MediaOpening
0 COMP VIDEO: Start Offset: 0,000; Duration: N/A
41 SYNC-BUFFER: Started.
609 SYNC-BUFFER: Finished. Clock set to 1534932751,634
0 EVENT START: MediaOpened
0 EVENT DONE : MediaOpened
0 EVENT START: BufferingStarted
0 EVENT DONE : BufferingStarted
0 OpenCommand: Completed
0 V BLK: 1534932751,634 | CLK: 1534932751,634 | DFT: 0 | IX: 0 | PQ: 0,0k | TQ: 0,0k
0 Command Queue (1 commands): Before ProcessNext
0 Play - ID: 404 Canceled: False; Completed: False; Status: WaitingForActivation; State:
94 V BLK: 1534932751,675 | CLK: 1534932751,699 | DFT: 24 | IX: 1 | PQ: 0,0k | TQ: 0,0kSo, the most the process of "sync-buffering" takes the most of the time.
Is there any parameter of FFmpeg which allows reducing a size of the buffer ?
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Use ffmpeg libraries to convert stream formats
17 septembre 2021, par SyrinxI'm attempting to write a small program and link it to a minimal set of ffmpeg libraries, like libavformat and whatever other libraries I need.


I am looking for documentation to get me started, or maybe a quick fix to the example program I am using.


I know ffmpeg (the project) provides example programs to help developers get started. I'm using the transcoding example, as it's close to my end goal, but it exits during init with an error about an audio issue.


Here I am using the transcoding example program that come with ffmpeg v4.4, on Ubuntu 18.04. My input source has one video channel (h264) and one audio channel (pcm_mulaw).


$ LD_LIBRARY_PATH=../../dist/lib ./transcoding rtsp://ip-camera/stream out.flv
...
 Stream #0:0: Video: h264, yuv420p, 1280x720, q=2-31, 20 tbn
 Stream #0:1: Audio: pcm_mulaw, 8000 Hz, 0 channels, s16
auto_resampler_0 @ 0x55da787fa140] [SWR @ 0x55da787fa5c0] Rematrix is needed between mono and 0 channels but there is not enough information to do it
[auto_resampler_0 @ 0x55da787fa140] Failed to configure output pad on auto_resampler_0


In libswresample/swresample.c :


320 if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) {
321 av_log(s, AV_LOG_ERROR, "Rematrix is needed between %s and %s "
322 "but there is not enough information to do it\n", l1, l2);
323 ret = AVERROR(EINVAL);
324 goto fail;
325 }


I'd really like it if I could make the transcoding example program work (fix it, or maybe use it appropriately if I am misunderstanding something). But short of that, where should I look for documentation about using the ffmpeg libraries ?


I don't even care about the audio. If I can just disable the audio, I would be happy with that solution. I tried tracking the "-an" option to ffmpeg (the program) to see how it does that in source code, but the options handling is a mess and I can't distinguish the parts of the code that I need from all the noise.


ffmpeg has web pages like this that aren't useful at all. There is documentation in the source code that looks like it should be viewed as HTML, but I don't see it exported anywhere. "make doc" generates a very small set of man pages that are insufficient to get me started.


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