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Autres articles (104)

  • Les autorisations surchargées par les plugins

    27 avril 2010, par

    Mediaspip core
    autoriser_auteur_modifier() afin que les visiteurs soient capables de modifier leurs informations sur la page d’auteurs

  • Encoding and processing into web-friendly formats

    13 avril 2011, par

    MediaSPIP 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 (...)

  • Ajouter notes et légendes aux images

    7 février 2011, par

    Pour pouvoir ajouter notes et légendes aux images, la première étape est d’installer le plugin "Légendes".
    Une fois le plugin activé, vous pouvez le configurer dans l’espace de configuration afin de modifier les droits de création / modification et de suppression des notes. Par défaut seuls les administrateurs du site peuvent ajouter des notes aux images.
    Modification lors de l’ajout d’un média
    Lors de l’ajout d’un média de type "image" un nouveau bouton apparait au dessus de la prévisualisation (...)

Sur d’autres sites (12944)

  • Permission denied with JAVE on OSX

    18 octobre 2016, par clankill3r

    I try to convert video to audio using jave on OSX.

    http://www.sauronsoftware.it/projects/jave/manual.php

    As the documentation states

    ...You can even build it by yourself getting the code (and the
    documentation to build it) on the official ffmpeg site. Once you have
    obtained a ffmpeg executable suitable for your needs, you have to hook
    it in the JAVE library. That’s a plain operation.

    I builded ffmpeg, but running the code I get a Permission denied error.
    I even changed all file permissions to 777 recursive.
    Hope someone can help, the documentation and error messages are very unclear.

    public class Mp4ToSoundTest {


       public static void main(String[] args) {
           Mp4ToSoundTest a = new Mp4ToSoundTest();
           a.setup();
       }

       void setup() {

           File source = new File("/Users/doekewartena/Downloads/vids_future_proj/VID_20160523_180100.mp4");

           System.out.println(source.exists());

           FFMPEGLocator my_ffmpeg_locator = new FFMPEGLocator() {
               @Override
               protected String getFFMPEGExecutablePath() {
                   return "/Users/doekewartena/Downloads/ffmpeg-3.1.4";
               }
           };


           File target = new File("/Users/doekewartena/Downloads/vids_future_proj/VID_20160523_180100.mp3");
           AudioAttributes audio = new AudioAttributes();
           audio.setCodec("libmp3lame");
           audio.setBitRate(128000);
           audio.setChannels(2);
           audio.setSamplingRate(44100);
           EncodingAttributes attrs = new EncodingAttributes();
           attrs.setFormat("mp3");
           attrs.setAudioAttributes(audio);
           Encoder encoder = new Encoder(my_ffmpeg_locator);

           // The source file can't be decoded. It occurs when the source file container, the video stream format or the
           // audio stream format are not supported by the decoder. You can check for supported containers and plugged
           // decoders calling the encoder methods getSupportedDecodingFormats(), getAudioDecoders() and getVideoDecoders().
           try {
               String[] r = encoder.getSupportedDecodingFormats();
               System.out.println("a");
               for (String s : r) {
                   System.out.println(s);
               }
               System.out.println("b");

               System.out.println();
               System.out.println(encoder.getAudioDecoders());

           } catch (EncoderException e) {
               e.printStackTrace();
           }



           try {
               encoder.encode(source, target, attrs);
           } catch (EncoderException e) {
               e.printStackTrace();
           }
           System.out.println("done");

       }

    }
  • Cannot Play Video Output of Libavcodec (ffmpeg) Encoding Example

    29 octobre 2019, par user3707763

    From FFMPEG’s GitHub, I use the encode_video.c to generate a 1 second video. Here is the example in question : https://github.com/FFmpeg/FFmpeg/blob/master/doc/examples/encode_video.c

    I compile with : gcc -Wall -o ffencode encode_video.c -lavcodec -lavutil -lz -lm

    Clean compile, zero warnings.

    I test the program by running : ./ffencode video.mp4 libx264

    Lots of stats printed out (expected based on source code) as well as ffmpeg logs, but ultimately no errors or warnings.

    However, then the generated output video.mp4, can only be played by ffplay, and VLC Player (as well as Google Chrome) fail to play the video.

    Playing it via vlc command line actually prints :

    [00007ffd3550fec0] main libvlc: Running vlc with the default interface. Use 'cvlc' to use vlc without interface.
    TagLib: MP4: Invalid atom size
    TagLib: MP4: Invalid atom size
    TagLib: MP4: Invalid atom size

    Looking at ffprobe output, the bitrate and duration fields are empty :

    Input #0, h264, from 'video.mp4':
     Duration: N/A, bitrate: N/A
       Stream #0:0: Video: h264 (High), yuv420p(progressive), 352x288, 25 fps, 25 tbr, 1200k tbn, 50 tbc

    I am using ffmpeg 4.1 with the following configuration :

    ffprobe version 4.1 Copyright (c) 2007-2018 the FFmpeg developers
     built with Apple LLVM version 10.0.0 (clang-1000.11.45.5)
     configuration: --prefix=/usr/local/Cellar/ffmpeg/4.1 --enable-shared --enable-pthreads --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-ffplay --enable-gpl --enable-libmp3lame --enable-libopus --enable-libsnappy --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-libxvid --enable-lzma --enable-opencl --enable-videotoolbox
     libavutil      56. 22.100 / 56. 22.100
     libavcodec     58. 35.100 / 58. 35.100
     libavformat    58. 20.100 / 58. 20.100
     libavdevice    58.  5.100 / 58.  5.100
     libavfilter     7. 40.101 /  7. 40.101
     libavresample   4.  0.  0 /  4.  0.  0
     libswscale      5.  3.100 /  5.  3.100
     libswresample   3.  3.100 /  3.  3.100
     libpostproc    55.  3.100 / 55.  3.100

    Any ideas how to fix this ? It is pretty surprising to see an API’s official example to be lacking such basic information.

  • How to make audio sound batter ? (C + FFMpeg audio generation example)

    31 janvier 2014, par Spender

    So I found this grate C FFMpeg official example which I simplified :

    #include
    #include
    #include

    #ifdef HAVE_AV_CONFIG_H
    #undef HAVE_AV_CONFIG_H
    #endif

    #include "libavcodec/avcodec.h"
    #include "libavutil/mathematics.h"

    #define INBUF_SIZE 4096
    #define AUDIO_INBUF_SIZE 20480
    #define AUDIO_REFILL_THRESH 4096

    /*
    * Audio encoding example
    */
    static void audio_encode_example(const char *filename)
    {
       AVCodec *codec;
       AVCodecContext *c= NULL;
       int frame_size, i, j, out_size, outbuf_size;
       FILE *f;
       short *samples;
       float t, tincr;
       uint8_t *outbuf;

       printf("Audio encoding\n");

       /* find the MP2 encoder */
       codec = avcodec_find_encoder(CODEC_ID_MP2);
       if (!codec) {
           fprintf(stderr, "codec not found\n");
           exit(1);
       }

       c= avcodec_alloc_context();

       /* put sample parameters */
       c->bit_rate = 64000;
       c->sample_rate = 44100;
       c->channels = 2;

       /* open it */
       if (avcodec_open(c, codec) < 0) {
           fprintf(stderr, "could not open codec\n");
           exit(1);
       }

       /* the codec gives us the frame size, in samples */
       frame_size = c->frame_size;
       samples = malloc(frame_size * 2 * c->channels);
       outbuf_size = 10000;
       outbuf = malloc(outbuf_size);

       f = fopen(filename, "wb");
       if (!f) {
           fprintf(stderr, "could not open %s\n", filename);
           exit(1);
       }

       /* encode a single tone sound */
       t = 0;
       tincr = 2 * M_PI * 440.0 / c->sample_rate;
       for(i=0;i<200;i++) {
           for(j=0;j* encode the samples */
           out_size = avcodec_encode_audio(c, outbuf, outbuf_size, samples);
           fwrite(outbuf, 1, out_size, f);
       }
       fclose(f);
       free(outbuf);
       free(samples);

       avcodec_close(c);
       av_free(c);
    }

    int main(int argc, char **argv)
    {

       /* must be called before using avcodec lib */
       avcodec_init();

       /* register all the codecs */
       avcodec_register_all();

       audio_encode_example("test.mp2");

       return 0;
    }

    How should it sound like ? May be I don't get something but it sounds awful =( how to make audio generation sound batter/ more interesting/ melodical in a wary shourt way (no special functions just how to change this code to make it sound batter) ?