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Sur d’autres sites (8226)

  • How can I dynamically update metadata in audio output with libav so that updates appear in MPV ?

    9 mars, par Teddy

    My ultimate goal is to proxy an internet radio station and programmatically add metadata to it during the stream that can be displayed and updated in MPV, the media player playing the audio.

    


    The metadata I would like to add is primarily the song title, but ideally additional information including artist, composer, and album.

    


    I envision running the proxy program like this :

    


    $ curl https://example.com/stream.mp3 | ./proxy_add_metadata | mpv -


    


    or maybe this :

    


    $ ./proxy_add_metadata &amp;&#xA;#=> <output>&#xA;$ mpv <output>&#xA;</output></output>

    &#xA;

    How could I make song metadata updates dynamically over time using libav ?

    &#xA;

    I’m using FFmpeg version 7.1.

    &#xA;

    I first tried changing the metadata dictionary shortly before writing a frame with a few different container formats :

    &#xA;

    av_dict_set(&amp;output_ctx->metadata, "title", "Title 1", 0);&#xA;/* [...] */&#xA;av_interleaved_write_frame(output_ctx, packet);&#xA;

    &#xA;

    Setting metadata with av_dict_set(&amp;output_ctx->metadata, "title", "Title 1", 0); only appears to work when done before writing the output header.

    &#xA;

    My next idea was to try setting metadata in AVPacket side data, but I’m unclear which container formats support this for the kind of metadata I’m working with.

    &#xA;

    I’m open to any output media container format or FFmpeg-originated network stream.

    &#xA;

    It’s unclear to me whether the metadata should be written to a separate stream within the media container or whether it should be written as side data in the output packets.

    &#xA;

    If what I’m trying to do is impossible, please explain why.

    &#xA;

    What I have so far reads audio from standard input and writes it to standard output. The input audio can be assumed to be in MP3 format. Non-working sections for metadata updates are commented out.

    &#xA;

    /* proxy_add_metadata.c */&#xA;#include &#xA;&#xA;#include <libavformat></libavformat>avformat.h>&#xA;#include <libavcodec></libavcodec>avcodec.h>&#xA;&#xA;int main() {&#xA;    int _err;&#xA;&#xA;    /* MP3 input */&#xA;    AVFormatContext *input_ctx = avformat_alloc_context();&#xA;    _err = avformat_open_input(&amp;input_ctx, "pipe:0", NULL, NULL);&#xA;    _err = avformat_find_stream_info(input_ctx, NULL);&#xA;&#xA;    AVFormatContext *output_ctx;&#xA;    _err = avformat_alloc_output_context2(&amp;output_ctx, NULL, "matroska", "pipe:1");&#xA;&#xA;    AVStream *input_stream = input_ctx->streams[0];&#xA;    AVStream *output_stream = avformat_new_stream(output_ctx, NULL);&#xA;&#xA;    _err = avcodec_parameters_copy(output_stream->codecpar, input_stream->codecpar);&#xA;    _err = avio_open(&amp;output_ctx->pb, "pipe:1", AVIO_FLAG_WRITE);&#xA;&#xA;    _err = avformat_write_header(output_ctx, NULL);&#xA;&#xA;    AVPacket *packet = av_packet_alloc();&#xA;&#xA;    /* Set up packet side data. */&#xA;    /*&#xA;    AVDictionary *packet_side_data_dict;&#xA;    av_dict_set(&amp;packet_side_data_dict, "title", "Title 1", 0);&#xA;&#xA;    size_t packet_side_data_size = 0;&#xA;    uint8_t *packet_side_data = av_packet_pack_dictionary(&#xA;        packet_side_data_dict,&#xA;        &amp;packet_side_data_size&#xA;    );&#xA;    av_dict_free(&amp;packet_side_data_dict);&#xA;    */&#xA;&#xA;    while (1) {&#xA;        _err = av_read_frame(input_ctx, packet);&#xA;        if (_err &lt; 0) {&#xA;            break;&#xA;        }&#xA;&#xA;        /* Can metadata updates be made here? */&#xA;&#xA;        /* Option 1: Attempt to write metadata to the container. */&#xA;        /*&#xA;        _err = av_dict_set(&amp;output_ctx->metadata, "title", "Title 1", 0);&#xA;        if (_err &lt; 0) {&#xA;            fprintf(stderr, "error: can&#x27;t set metadata title in stream: %s\n", av_err2str(_err));&#xA;            break;&#xA;        }&#xA;        */&#xA;&#xA;        /* Option 2: Attempt to write metadata to packet side data. */&#xA;        /*&#xA;        _err = av_packet_add_side_data(&#xA;            packet,&#xA;            AV_PKT_DATA_METADATA_UPDATE,&#xA;            packet_side_data,&#xA;            packet_side_data_size&#xA;        );&#xA;        if (_err &lt; 0) {&#xA;            fprintf(stderr, "error: can&#x27;t add side data to packet: %s\n", av_err2str(_err));&#xA;            break;&#xA;        }&#xA;        */&#xA;&#xA;        AVStream *input_stream = input_ctx->streams[packet->stream_index];&#xA;        AVStream *output_stream = output_ctx->streams[packet->stream_index];&#xA;&#xA;        av_packet_rescale_ts(packet, input_stream->time_base, output_stream->time_base);&#xA;        packet->pos = -1;&#xA;&#xA;        _err = av_interleaved_write_frame(output_ctx, packet);&#xA;        if (_err &lt; 0) {&#xA;            fprintf(stderr, "error: packet write: %s\n", av_err2str(_err));&#xA;            break;&#xA;        }&#xA;    }&#xA;&#xA;    av_write_trailer(output_ctx);&#xA;&#xA;    av_packet_free_side_data(packet);&#xA;    av_packet_free(&amp;packet);&#xA;&#xA;    avio_closep(&amp;output_ctx->pb);&#xA;    avformat_free_context(output_ctx);&#xA;&#xA;    avformat_close_input(&amp;input_ctx);&#xA;&#xA;    return 0;&#xA;}&#xA;

    &#xA;

    cc \&#xA;    -Wall \&#xA;    -g \&#xA;    -I/.../ffmpeg7/include \&#xA;    -o proxy_add_metadata \&#xA;    proxy_add_metadata.c \&#xA;    -L/.../ffmpeg7/lib -lavformat -lavcodec&#xA;

    &#xA;

    $ &lt; sample.mp3 ./proxy_add_metadata | mpv -&#xA;

    &#xA;

  • ffmpeg produced .wav reads only zeros with scipy.io.wavfile

    8 janvier 2015, par question_mark

    Hi everyone and thanks for reading.

    I wanted to do some analysis on a song using Python’s scipy.io.wavfile. Since I only have the song as .mp3 I converted the file to .wav using ffmpeg the following way :

    ffmpeg -i test.mp3 test.wav

    The .wav file plays perfectly well with vlc player, but wavfile shows only zeroes when reading it :

    from scipy.io import wavfile as wf

    data = wf.read("test.wav")
    C:\Program Files\Anaconda\lib\site-packages\scipy\io\wavfile.py:42: WavFileWarning: Unknown wave file format
     warnings.warn("Unknown wave file format", WavFileWarning)

    data
    (44100, array([[0, 0],
           [0, 0],
           [0, 0],
           ...,
           [0, 0],
           [0, 0],
           [0, 0]], dtype=int16))

    I tried getting the data with Python’s built-in wave module before to the same effect (only zeros).
    I am using the 64bit version of ffmpeg (ffmpeg-20140218-git-61d5970-win64-static).

    Any help is appreciated :-)

    Edit : Included .wav header and tried forcing ffmpeg output format

    I guess the header information of the .wav file is included here :

    ffmpeg -i .\test.wav
    Guessed Channel Layout for  Input Stream #0.0 : stereo
    Input #0, wav, from '.\test.wav':
     Metadata:
       artist          : Joe Cocker
       copyright       : (C) 1987 Capitol Records, Inc.
       date            : 1987
       genre           : Pop
       title           : Unchain My Heart
       album           : Unchain My Heart
       track           : 1/10
       encoder         : Lavf55.33.100
     Duration: 00:05:04.33, bitrate: 1411 kb/s
     Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s

    If I try to specify the ffmpeg output format explicitly for the .mp3 conversion :

    ffmpeg -i .\test.mp3 -f s16le -ar 44100 -ac 2 test.wav
    Input #0, mp3, from '.\test.mp3':
     Metadata:
       title           : Unchain My Heart
       artist          : Joe Cocker
       album           : Unchain My Heart
       genre           : Pop
       composer        : Bobby Sharp
       track           : 1/10
       disc            : 1/1
       album_artist    : Joe Cocker
       copyright       : (C) 1987 Capitol Records, Inc.
       date            : 1987
     Duration: 00:05:04.35, start: 0.025056, bitrate: 240 kb/s
       Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16p, 235 kb/s
       Stream #0:1: Video: mjpeg, yuvj420p(pc), 600x600 [SAR 1:1 DAR 1:1], 90k tbr, 90k tbn, 90k tbc
       Metadata:
         title           :
         comment         : Cover (front)
    Output #0, s16le, to 'test.wav':
     Metadata:
       title           : Unchain My Heart
       artist          : Joe Cocker
       album           : Unchain My Heart
       genre           : Pop
       composer        : Bobby Sharp
       track           : 1/10
       disc            : 1/1
       album_artist    : Joe Cocker
       copyright       : (C) 1987 Capitol Records, Inc.
       date            : 1987
       encoder         : Lavf55.33.100
       Stream #0:0: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s
    Stream mapping:
     Stream #0:0 -> #0:0 (mp3 -> pcm_s16le)
    Press [q] to stop, [?] for help
    video:0kB audio:52425kB subtitle:0 data:0 global headers:0kB muxing overhead 0.000000%
    size=   52425kB time=00:05:04.32 bitrate=1411.2kbits/s

    But in this case (forced format), both ffmpeg and wavfile are not able to read the file :

    ffmpeg -i .\test.wav
    .\test.wav: Invalid data found when processing input

    and

    data = wf.read("test2.wav")
    ---------------------------------------------------------------------------
    ValueError                                Traceback (most recent call last)
    in <module>()
    ----> 1 data = wf.read("test2.wav")

    C:\Program Files\Anaconda\lib\site-packages\scipy\io\wavfile.pyc in read(filename, mmap)
       152
       153     try:
    --> 154         fsize = _read_riff_chunk(fid)
       155         noc = 1
       156         bits = 8

    C:\Program Files\Anaconda\lib\site-packages\scipy\io\wavfile.pyc in _read_riff_chunk(fid)
        98         _big_endian = True
        99     elif str1 != b'RIFF':
    --> 100         raise ValueError("Not a WAV file.")
       101     if _big_endian:
       102         fmt = '>I'

    ValueError: Not a WAV file.
    </module>
  • ffmpeg probesize and analyzeduration flags

    7 mai 2014, par Xerphiel

    Could someone explain to me in simple terms what these flags/options do ?

    Running "ffmpeg -h full | grep probesize/analyzeduration" yields the following documentation :

    -probesize         <int>        .D.... set probing size (from 32 to INT_MAX) (default 5e+06)
    -analyzeduration   <int>        .D.... specify how many microseconds are analyzed to probe the input (from 0 to INT_MAX) (default 5e+06)
    </int></int>

    This is beyond my understanding, so am hoping for a simpler explanation.

    All the discussions on this subject I’ve found via googling do not describe the basic function of the options.

    Any help is appreciated.

    Thank you.