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Autres articles (105)
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Encoding and processing into web-friendly formats
13 avril 2011, parMediaSPIP automatically converts uploaded files to internet-compatible formats.
Video files are encoded in MP4, Ogv and WebM (supported by HTML5) and MP4 (supported by Flash).
Audio files are encoded in MP3 and Ogg (supported by HTML5) and MP3 (supported by Flash).
Where possible, text is analyzed in order to retrieve the data needed for search engine detection, and then exported as a series of image files.
All uploaded files are stored online in their original format, so you can (...) -
Contribute to a better visual interface
13 avril 2011MediaSPIP is based on a system of themes and templates. Templates define the placement of information on the page, and can be adapted to a wide range of uses. Themes define the overall graphic appearance of the site.
Anyone can submit a new graphic theme or template and make it available to the MediaSPIP community. -
Création définitive du canal
12 mars 2010, parLorsque votre demande est validée, vous pouvez alors procéder à la création proprement dite du canal. Chaque canal est un site à part entière placé sous votre responsabilité. Les administrateurs de la plateforme n’y ont aucun accès.
A la validation, vous recevez un email vous invitant donc à créer votre canal.
Pour ce faire il vous suffit de vous rendre à son adresse, dans notre exemple "http://votre_sous_domaine.mediaspip.net".
A ce moment là un mot de passe vous est demandé, il vous suffit d’y (...)
Sur d’autres sites (4510)
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Problem : FFmpeg and C++ extract and save frame
26 mai 2021, par Simba_cl25I am doing a project where I must do the following : extract frames (along with the associated metadata - KLV) from a video given a period (for the moment every 10 seconds).
I have followed codes found on internet but I get an error that I can find a solution to.


extern "C" {
#include <libavcodec></libavcodec>avcodec.h>
#include <libavformat></libavformat>avformat.h>
#include <libavdevice></libavdevice>avdevice.h>
#include <libswscale></libswscale>swscale.h>
#include <libavfilter></libavfilter>avfilter.h>
#include <libswresample></libswresample>swresample.h>
#include <libavutil></libavutil>avutil.h>
#include <libavutil></libavutil>imgutils.h> 
}

static void SaveFrame(AVFrame *pFrame, int width, int height, int iFrame);


int main(int argc, const char * argv[])
{
 AVFormatContext *pFormatCtx;
 int i, videoStream;
 AVCodecContext *pCodecCtx = NULL;
 const AVCodec *pCodec = NULL;
 AVFrame *pFrame;
 AVFrame *pFrameRGB;
 AVPacket *packet = av_packet_alloc();
 AVStream *pStream;
 int numBytes;
 int64_t Duration;
 uint8_t *buffer;
 bool frameFinished = false;

 // Open video file - check for errors
 pFormatCtx = 0;
 if (avformat_open_input(&pFormatCtx, "00Video\\VideoR.ts", 0, 0) != 0)
 return -1; 

 if (avformat_find_stream_info(pFormatCtx, 0) < 0)
 return -1;

 if (av_find_best_stream(pFormatCtx, AVMEDIA_TYPE_VIDEO, -1, -1, 0, 0) <0)
 return -1;

 av_dump_format(pFormatCtx, 0, "00Video\\VideoR.ts", false);


 // Find the first video stream
 videoStream = -1;
 for (i = 0; i < pFormatCtx->nb_streams; i++)
 if (pFormatCtx->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_VIDEO)
 {
 videoStream = i;
 break;
 }
 if (videoStream == -1)
 return -1; 
 


 // Find the decoder for the video stream
 pCodec = avcodec_find_decoder(pFormatCtx->streams[videoStream]->codecpar->codec_id);
 pCodecCtx = avcodec_alloc_context3(pCodec);

 if (pCodec == NULL)
 {
 fprintf(stderr, "Codec not found\n");
 return -1; 
 }


 if (avcodec_open2(pCodecCtx, pCodec, NULL) < 0)
 {
 fprintf(stderr, "Could not open codec\n");
 return -1; 
 }

 // Hack to correct wrong frame rates that seem to be generated by some codecs
 if (pCodecCtx->time_base.num > 1000 && pCodecCtx->time_base.den == 1)
 pCodecCtx->time_base.den = 1000;




 // Allocate video frame - original frame 
 pFrame = av_frame_alloc();

 if (!pFrame) {
 fprintf(stderr, "Could not allocate video frame\n");
 return -1;
 }


 // Allocate an AVFrame structure
 pFrameRGB = av_frame_alloc();

 if (pFrameRGB == NULL)
 {
 fprintf(stderr, "Could not allocate video RGB frame\n");
 return -1;
 }

 
 Duration = av_rescale_q(vstrm->duration, vstrm->time_base, { 1,1000 });
 
 numBytes = av_image_get_buffer_size(pCodecCtx->pix_fmt, pCodecCtx->width, pCodecCtx->height, 0);
 buffer = (uint8_t*)av_malloc(numBytes * sizeof(uint8_t));


 av_image_fill_arrays(pFrameRGB->data, pFrameRGB->linesize, buffer, AV_PIX_FMT_RGB24, pCodecCtx->width, pCodecCtx->height, 1);
 
 
 
 
 i = 0;
 while (av_read_frame(pFormatCtx, packet) >= 0)
 {
 // Is this a packet from the video stream?
 if (packet->stream_index == videoStream)
 {
 int ret;
 ret = avcodec_send_packet(pCodecCtx, packet);
 if (ret < 0 || ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
 fprintf(stderr, "Error sending a packet for decoding\n");
 //break;
 }
 while (ret >= 0) {
 ret = avcodec_receive_frame(pCodecCtx, pFrame);
 if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
 return -1;
 else if (ret < 0) {
 fprintf(stderr, "Error during decoding\n");
 return -1;
 frameFinished = true;
 }


 // Did we get a video frame?
 if (frameFinished)
 {
 static struct SwsContext *img_convert_ctx;

 
 if (img_convert_ctx == NULL) {
 int w = pCodecCtx->width;
 int h = pCodecCtx->height;
 img_convert_ctx = sws_getContext(w, h,
 pCodecCtx->pix_fmt,
 w, h, AV_PIX_FMT_RGB24, SWS_FAST_BILINEAR,
 NULL, NULL, NULL);

 if (img_convert_ctx == NULL) {
 fprintf(stderr, "Cannot initialize the conversion context!\n");
 exit(1);
 }
 }

 int ret = sws_scale(img_convert_ctx, pFrame->data, pFrame->linesize, 0,
 pCodecCtx->height, pFrameRGB->data, pFrameRGB->linesize);


 // Save the frame to disk
 if (i <= Duration)
 {
 SaveFrame(pFrameRGB, pCodecCtx->width, pCodecCtx->height, i);
 i += 10*1000;
 }
 }
 }

 
 }

 // Free the packet that was allocated by av_read_frame
 av_packet_unref(packet);
 }

 

 // Free the RGB image
 free(buffer);
 av_free(pFrameRGB);

 // Free the YUV frame
 av_free(pFrame);

 // Close the codec
 avcodec_close(pCodecCtx);

 // Close the video file
 avformat_close_input(&pFormatCtx);
 return 0;
 
}




static void SaveFrame(AVFrame *pFrame, int width, int height, int iFrame)
{
 FILE *pFile;
 char szFilename[32];
 int y;


 // Open file
 sprintf(szFilename, "Im\\frame%d.png", iFrame);
 pFile = fopen(szFilename, "wb");
 if (pFile == NULL)
 return;

 // Write header
 fprintf(pFile, "P6\n%d %d\n255\n", width, height);
 // Write pixel data
 for (y = 0; y < height; y++)
 fwrite(pFrame->data[0] + y * pFrame->linesize[0], 1, width*3, pFile);

 // Close file
 fclose(pFile);
}


The error I get is :


[swscaler @ 03055A80] bad dst image pointers


I think is because


numBytes = av_image_get_buffer_size(pCodecCtx->pix_fmt, pCodecCtx->width, pCodecCtx->height, 0);


Returns a negative value ( -22) but I don't know why.


Thanks,


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Pion WebRTC Audio stream cutting out while video works
19 août 2021, par TwometerI am trying to send an MP4 video through Pion WebRTC to the browser.


Using FFmpeg, I split it into an Opus OGG stream and an Annex-B H.264 video stream. While the video works fine, the audio keeps cutting in and out. It plays fine for a few seconds, then stops for a second, and continues.


This is the FFmpeg command I use for audio :


ffmpeg -i demo.mp4 -c:a libopus -vn -page_duration 20000 demo.ogg


And this is my transmitter (shortened) :


var lastGranule uint64
for {
 pageData, pageHeader, err := ogg.ParseNextPage() // Uses Pion OggReader

 // Taken from the play-from-disk example
 sampleCount := float64(pageHeader.GranulePosition - lastGranule)
 lastGranule = pageHeader.GranulePosition
 sampleDuration := time.Duration((sampleCount/48000)*1000) * time.Millisecond

 err = audioTrack.WriteSample(media.Sample{Data: pageData, Duration: sampleDuration})
 util.HandleError(err)

 time.Sleep(sampleDuration)
}


I tried hardcoding the delay to 15ms, which fixes the issue that it's cutting out, but then it randomly plays way too fast or starts skipping. Since I had glitchy video before updating my FFmpeg command (add keyframes and remove b-frames), I assume this is also an encoder problem.


What could be the cause for this ?


Update : Using WebRTC logging in Chrome, I discovered the following log lines that occurred frequently :


[27216:21992:0809/141533.175:WARNING:rtcp_receiver.cc(452)] 30 RTCP blocks were skipped due to being malformed or of unrecognized/unsupported type, during the past 10 second period.


This is probably the reason for the cutouts, although I can't figure out why it receives malformed data.


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How to embed subtitles into an mp4 file using gstreamer
27 août 2021, par StephenMy Goal


I'm trying to embed subtitles into an mp4 file using the
mp4muxgstreamer element.

What I've tried


The pipeline I would like to use is :


GST_DEBUG=3 gst-launch-1.0 filesrc location=sample-nosub-avc.mp4 ! qtdemux ! queue ! video/x-h264 ! mp4mux name=mux reserved-moov-update-period=1000 ! filesink location=output.mp4 filesrc location=english.srt ! subparse ! queue ! text/x-raw,format=utf8 ! mux.subtitle_0


It just demuxes a sample mp4 file for the h.264 stream and then muxes it together with an srt subtitle file.


The error I get is :


Setting pipeline to PAUSED ...
0:00:00.009958915 1324869 0x5624a8c7a0a0 WARN basesrc gstbasesrc.c:3600:gst_base_src_start_complete:<filesrc0> pad not activated yet
Pipeline is PREROLLING ...
0:00:00.010128080 1324869 0x5624a8c53de0 WARN basesrc gstbasesrc.c:3072:gst_base_src_loop:<filesrc1> error: Internal data stream error.
0:00:00.010129102 1324869 0x5624a8c53e40 WARN qtdemux qtdemux_types.c:239:qtdemux_type_get: unknown QuickTime node type pasp
0:00:00.010140810 1324869 0x5624a8c53de0 WARN basesrc gstbasesrc.c:3072:gst_base_src_loop:<filesrc1> error: streaming stopped, reason not-negotiated (-4)
0:00:00.010172990 1324869 0x5624a8c53e40 WARN qtdemux qtdemux.c:3237:qtdemux_parse_trex:<qtdemux0> failed to find fragment defaults for stream 1
ERROR: from element /GstPipeline:pipeline0/GstFileSrc:filesrc1: Internal data stream error.
Additional debug info:
gstbasesrc.c(3072): gst_base_src_loop (): /GstPipeline:pipeline0/GstFileSrc:filesrc1:
streaming stopped, reason not-negotiated (-4)
ERROR: pipeline doesn't want to preroll.
Setting pipeline to NULL ...
Freeing pipeline ...
</qtdemux0></filesrc1></filesrc1></filesrc0>

My Thoughts


I believe the issue is not related to the above warning but rather
mp4mux's incompatibility with srt subtitles.

The reason I belive this is because, other debug logs hint at it, but also stealing the subititles from another mp4 file and muxing it back together does work.


gst-launch-1.0 filesrc location=sample-nosub-avc.mp4 ! qtdemux ! mp4mux name=mux ! filesink location=output.mp4 filesrc location=sample-with-subs.mp4 ! qtdemux name=demux demux.subtitle_1 ! text/x-raw,format=utf8 ! queue ! mux.subtitle_0


A major catch 22 I am having is that mp4 files don't typically support srt subtitles, but gstreamer's
subparseelement doesn't support parsing mp4 subtitle formats (tx3g, ttxt, etc.) so I'm not sure how I'm meant to put it all together.

I'm very sorry for the lengthy question but I've tried many things so it was difficult to condense it. Any hints or help is appreciated. Thank you.


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