Recherche avancée

Médias (17)

Mot : - Tags -/wired

Autres articles (106)

  • MediaSPIP version 0.1 Beta

    16 avril 2011, par

    MediaSPIP 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 (...)

  • MediaSPIP 0.1 Beta version

    25 avril 2011, par

    MediaSPIP 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 (...)

  • Amélioration de la version de base

    13 septembre 2013

    Jolie sélection multiple
    Le plugin Chosen permet d’améliorer l’ergonomie des champs de sélection multiple. Voir les deux images suivantes pour comparer.
    Il suffit pour cela d’activer le plugin Chosen (Configuration générale du site > Gestion des plugins), puis de configurer le plugin (Les squelettes > Chosen) en activant l’utilisation de Chosen dans le site public et en spécifiant les éléments de formulaires à améliorer, par exemple select[multiple] pour les listes à sélection multiple (...)

Sur d’autres sites (9865)

  • CMD Batch Variable Won't Save FFprobe Output

    15 juillet 2017, par Matt McManis

    I have an CMD Batch Script that will convert a folder of mp4 videos to webm.

    You will need :

    • FFmpeg/FFprobe installed and set in Environment Variables to run from CMD.
    • A folder with an mp4 for FFprobe to parse.

    To make it easy, this is only the first part of the script, showing the Video Bitrate variable.

    Here is a full script, just replace the paths.
    https://pastebin.com/raw/3ng77Exz

    How the Script works :

    • Loops through all videos in folder
    • Has FFprobe parse the Video’s Bitrate and save it to %V and
      %vBitrate%.
    • Has FFmpeg use %V. Such as -b:v %V will become the parsed value
      -b:v 9401k.
    • Converts each video from mp4 to webm using the parsed Bitrate

    Problem

    I can’t get FFprobe’s Output to save to the variable. I’ve come up with a workaround, having it first save the bitrate value to a temp file, then import that to the %vBitrate% variable.

    Example : (%V > tmp_vBitrate) & SET /p vBitrate= < tmp_vBitrate.


    Works

    Temp File Variable

    cd "C:\Users\Matt\Videos\" && for %f in (*.mp4) do ffprobe -i "C:\Users\Matt\Desktop\Test\%~f" -select_streams v:0 -show_entries stream=bit_rate -v quiet -of csv="p=0" & for /f "tokens=*" %V in ("ffprobe -i "%~f" -select_streams v:0 -show_entries stream=bit_rate -v quiet -of csv=p=0") do (echo ) & (%V > tmp_vBitrate) & SET /p vBitrate= < tmp_vBitrate & del tmp_vBitrate & for /F %V in ('echo %vBitrate%') do (echo %V)

    Does Not Work

    Memory Variable

    cd "C:\Users\Matt\Videos\" && for %f in (*.mp4) do ffprobe -i "C:\Users\Matt\Desktop\Test\%~f" -select_streams v:0 -show_entries stream=bit_rate -v quiet -of csv="p=0" & for /f "tokens=*" %V in ("ffprobe -i "%~f" -select_streams v:0 -show_entries stream=bit_rate -v quiet -of csv=p=0") do (echo ) & SET vBitrate=%V & for /F %V in ('echo %vBitrate%') do (echo %V)

    Testing It

    Run the first command. When it is finished, type echo %vBitrate% in CMD and press Enter. You’ll see the bitrate of the last mp4 file parsed.

    Do the same for the second command and you’ll see it doesn’t work.


    Solution

    I would like to get rid of the Temp File Variable and get the second command to work.

    (%V > tmp_vBitrate) & SET /p vBitrate= < tmp_vBitrate to just SET vBitrate=%V.

    Maybe this whole thing can be simplified ? Am I using the variables wrong ?

  • How to save synthesized audio during streaming distribution

    7 avril 2022, par its-ogawa

    I would like to stream a composite of microphone audio and digital sound sources by ffmpeg and save the delivery to an m3u8 file.

    


    Below are the commands I have actually tried.

    


    ffmpeg -rtbufsize 100M -f dshow -i video=<my webcam="webcam">:audio=<my microphone="microphone"> -re -stream_loop -1 -i <my sound="sound" source="source"> -filter_complex "[0]aformat=sample_fmts=fltp:sample_rates=44100:channel_layouts=stereo,adelay=2100|2100,volume@voice=volume=10dB[voice],[1]aformat=sample_fmts=fltp:sample_rates=44100:channel_layouts=stereo,azmq,volume@bgm=volume=0.2[bgm],[voice][bgm]amerge=inputs=2[out]" -map 0:v -map [out]:a -f mpegts -flush_packets 0 udp://XXX.XXX.XXX.XXX:XXX?pkt_size=1316 -f hls -hls_time 5 hls.m3u8&#xA;</my></my></my>

    &#xA;

    I have played an m3u8 created this way, but I cannot hear the digital sound source that I am supposed to have synthesized.&#xA;Even though the synthesized audio was audible when streamed.

    &#xA;

    Perhaps you need to set up something like -map 0:v -map [out]:a when saving to the m3u8 file, in which case the Output with label &#x27;out&#x27; does not exist in any defined filter graph, or was already used elsewhere. message appears and does not work.

    &#xA;

  • convert pcm stream data to encoded aac data

    24 décembre 2019, par Kumar

    I tried to convert pulse-audio pcm stream data to aac encoded data using ffmpeg.
    But after encoding I get noise-full data, not the correct one. Here I post my code, anyone help me with some ideas.

    Initial configuration :

       av_register_all();

       int error;
       if ((error = avio_open(&amp;output_io_context,"out.aac",AVIO_FLAG_WRITE))&lt;0) {
               printf("could not open output file\n");
       }

       if (!(output_format_context = avformat_alloc_context())) {
               printf("output_format_context error\n");
       }

       output_format_context->pb = output_io_context;

       if(!(output_format_context->oformat = av_guess_format(NULL, "out.aac", NULL))) {
               printf("guess format error\n");
       }

       codec = avcodec_find_encoder(AV_CODEC_ID_AAC);
       if (codec == NULL) {
               printf("avcodec_find_encoder: ERROR\n");
       }

       if (!(stream = avformat_new_stream(output_format_context, NULL))) {
               printf("stream create error\n");
       }
       output_codec_context = avcodec_alloc_context3(codec);
       if(!output_codec_context) {
               printf("output_codec_context is null\n");
       }

       output_codec_context->channels       = CHANNELS;
       output_codec_context->channel_layout = av_get_default_channel_layout(CHANNELS);
       output_codec_context->sample_rate    = SAMPLE_RATE; //input_codec_context->sample_rate;
       output_codec_context->sample_fmt     = codec->sample_fmts[0];
       output_codec_context->bit_rate       = 48000; //OUTPUT_BIT_RATE;

       stream->time_base.den = SAMPLE_RATE;//input_codec_context->sample_rate;
       stream->time_base.num = 1;

       if(output_format_context->oformat->flags &amp; AVFMT_GLOBALHEADER)
               output_codec_context->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;

       if ((error = avcodec_open2(output_codec_context, codec, NULL)) &lt; 0) {
           printf("error");
       }

       error = avcodec_parameters_from_context(stream->codecpar, output_codec_context);

       if (write_output_file_header(output_format_context)) {
               printf("write header failure...\n");
       }

    Data encoding :

    AVFrame *output_frame;
    int frame_pos = 0, ctx_frame_size = output_codec_context->frame_size;
    int size = av_samples_get_buffer_size(NULL, CHANNELS,
                 output_codec_context->frame_size,output_codec_context->sample_fmt, 1);
    if((x =  avcodec_fill_audio_frame(output_frame, CHANNELS,
                  output_codec_context->sample_fmt, data, length, 1)) &lt; 0) {
         printf("avcodec_fill_audio_frame error : %s\n", av_err2str(x));
    }

    int data_written;
    if (encode_audio_frame(output_frame, output_format_context,
               output_codec_context, &amp;data_written)) {
           printf("encode_audio_frame error\n");
     }

     av_frame_free(&amp;output_frame);

    helper_function :

    int encode_audio_frame(AVFrame *frame,AVFormatContext *output_format_context,
                 AVCodecContext *output_codec_context, int *data_present)
    {
      AVPacket output_packet;
      int error;
      init_packet(&amp;output_packet);

      if (frame) {
         frame->pts = pts;
         pts += frame->nb_samples;
      }

      error = avcodec_send_frame(output_codec_context, frame);
      if (error == AVERROR_EOF) {
        error = 0;
        goto cleanup;
      } else if (error &lt; 0) {
         fprintf(stderr, "Could not send packet for encoding (error '%s')\n",
               av_err2str(error));
         return error;
      }

      error = avcodec_receive_packet(output_codec_context, &amp;output_packet);
      if (error == AVERROR(EAGAIN)) {
        error = 0;
        goto cleanup;
      } else if (error == AVERROR_EOF) {
         error = 0;
         goto cleanup;
      } else if (error &lt; 0) {
         fprintf(stderr, "Could not encode frame (error '%s')\n",
               av_err2str(error));
         goto cleanup;
      } else {
         *data_present = 1;
     }

     if (*data_present &amp;&amp;
         (error = av_write_frame(output_format_context, &amp;output_packet)) &lt; 0) {
         fprintf(stderr, "Could not write frame (error '%s')\n",
                av_err2str(error));
         goto cleanup;
     }

     cleanup:
         av_packet_unref(&amp;output_packet);
         return error;
    }
    • Do we need to fill AVFrame with sizeof(av_samples_get_buffer_size) or context->frame_size ?

    TYIA :) !!