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Autres articles (112)
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Personnaliser en ajoutant son logo, sa bannière ou son image de fond
5 septembre 2013, parCertains thèmes prennent en compte trois éléments de personnalisation : l’ajout d’un logo ; l’ajout d’une bannière l’ajout d’une image de fond ;
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Script d’installation automatique de MediaSPIP
25 avril 2011, parAfin de palier aux difficultés d’installation dues principalement aux dépendances logicielles coté serveur, un script d’installation "tout en un" en bash a été créé afin de faciliter cette étape sur un serveur doté d’une distribution Linux compatible.
Vous devez bénéficier d’un accès SSH à votre serveur et d’un compte "root" afin de l’utiliser, ce qui permettra d’installer les dépendances. Contactez votre hébergeur si vous ne disposez pas de cela.
La documentation de l’utilisation du script d’installation (...) -
La sauvegarde automatique de canaux SPIP
1er avril 2010, parDans le cadre de la mise en place d’une plateforme ouverte, il est important pour les hébergeurs de pouvoir disposer de sauvegardes assez régulières pour parer à tout problème éventuel.
Pour réaliser cette tâche on se base sur deux plugins SPIP : Saveauto qui permet une sauvegarde régulière de la base de donnée sous la forme d’un dump mysql (utilisable dans phpmyadmin) mes_fichiers_2 qui permet de réaliser une archive au format zip des données importantes du site (les documents, les éléments (...)
Sur d’autres sites (11061)
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Unknown issue with Discord.js and ytdl, completely skips playing audio
21 novembre 2017, par Gman0064One of the commands I have for my Discord bot is to play a predefined music clip in the current user’s voice channel. The bot can connect, but rather than playing the song, it instantaneously leaves. I’ve tried using both
connection.playStreamas well asconnection.playFile, and both seem to return the same (lack of) output. Am I missing some sort of dependency or is my code just written incorrectly ? Any help would be greatly appreciated !const Discord = require('discord.js');
const ytdl = require('ytdl-core');
const client = new Discord.Client();
const streamOptions = { seek: 0, volume: 1};
client.on('ready', () => {
console.log('Login Success');
});
client.on('message', message => {
if (message.content === '$vaporwave') {
if (!message.guild) return;
if(message.member.voiceChannel) {
message.member.voiceChannel.join().then(connection => {
console.log("joined channel");
//const stream = ytdl('https://www.youtube.com/watch?v=cU8HrO7XuiE', { filter : 'audioonly' });
const dispatcher = connection.playFile('./mcp420.mp3');
//const dispatcher = connection.playStream(stream, streamOptions);
dispatcher.on("end", end => {
console.log("left channel");
message.member.voiceChannel.leave();
});
}).catch(err => console.log(err));
}
}
});- NPM v4.6.1
- Node.js v8.9.1
- FFMPEG v3.2.8-1
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How can I dynamically update metadata in audio output with libav so that updates appear in MPV ?
9 mars, par TeddyMy 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 &
#=> <output>
$ mpv <output>
</output></output>

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


I’m using FFmpeg version 7.1.


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


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


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

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

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


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.


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


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.


/* proxy_add_metadata.c */
#include 

#include <libavformat></libavformat>avformat.h>
#include <libavcodec></libavcodec>avcodec.h>

int main() {
 int _err;

 /* MP3 input */
 AVFormatContext *input_ctx = avformat_alloc_context();
 _err = avformat_open_input(&input_ctx, "pipe:0", NULL, NULL);
 _err = avformat_find_stream_info(input_ctx, NULL);

 AVFormatContext *output_ctx;
 _err = avformat_alloc_output_context2(&output_ctx, NULL, "matroska", "pipe:1");

 AVStream *input_stream = input_ctx->streams[0];
 AVStream *output_stream = avformat_new_stream(output_ctx, NULL);

 _err = avcodec_parameters_copy(output_stream->codecpar, input_stream->codecpar);
 _err = avio_open(&output_ctx->pb, "pipe:1", AVIO_FLAG_WRITE);

 _err = avformat_write_header(output_ctx, NULL);

 AVPacket *packet = av_packet_alloc();

 /* Set up packet side data. */
 /*
 AVDictionary *packet_side_data_dict;
 av_dict_set(&packet_side_data_dict, "title", "Title 1", 0);

 size_t packet_side_data_size = 0;
 uint8_t *packet_side_data = av_packet_pack_dictionary(
 packet_side_data_dict,
 &packet_side_data_size
 );
 av_dict_free(&packet_side_data_dict);
 */

 while (1) {
 _err = av_read_frame(input_ctx, packet);
 if (_err < 0) {
 break;
 }

 /* Can metadata updates be made here? */

 /* Option 1: Attempt to write metadata to the container. */
 /*
 _err = av_dict_set(&output_ctx->metadata, "title", "Title 1", 0);
 if (_err < 0) {
 fprintf(stderr, "error: can't set metadata title in stream: %s\n", av_err2str(_err));
 break;
 }
 */

 /* Option 2: Attempt to write metadata to packet side data. */
 /*
 _err = av_packet_add_side_data(
 packet,
 AV_PKT_DATA_METADATA_UPDATE,
 packet_side_data,
 packet_side_data_size
 );
 if (_err < 0) {
 fprintf(stderr, "error: can't add side data to packet: %s\n", av_err2str(_err));
 break;
 }
 */

 AVStream *input_stream = input_ctx->streams[packet->stream_index];
 AVStream *output_stream = output_ctx->streams[packet->stream_index];

 av_packet_rescale_ts(packet, input_stream->time_base, output_stream->time_base);
 packet->pos = -1;

 _err = av_interleaved_write_frame(output_ctx, packet);
 if (_err < 0) {
 fprintf(stderr, "error: packet write: %s\n", av_err2str(_err));
 break;
 }
 }

 av_write_trailer(output_ctx);

 av_packet_free_side_data(packet);
 av_packet_free(&packet);

 avio_closep(&output_ctx->pb);
 avformat_free_context(output_ctx);

 avformat_close_input(&input_ctx);

 return 0;
}


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


$ < sample.mp3 ./proxy_add_metadata | mpv -


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ffmpeg produced .wav reads only zeros with scipy.io.wavfile
8 janvier 2015, par question_markHi 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.wavThe .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/sIf 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/sBut 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 inputand
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>
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