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Rennes Emotion Map 2010-11
19 octobre 2011, par
Mis à jour : Juillet 2013
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Autres articles (25)
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Les autorisations surchargées par les plugins
27 avril 2010, parMediaspip core
autoriser_auteur_modifier() afin que les visiteurs soient capables de modifier leurs informations sur la page d’auteurs -
Publier sur MédiaSpip
13 juin 2013Puis-je poster des contenus à partir d’une tablette Ipad ?
Oui, si votre Médiaspip installé est à la version 0.2 ou supérieure. Contacter au besoin l’administrateur de votre MédiaSpip pour le savoir -
Supporting all media types
13 avril 2011, parUnlike most software and media-sharing platforms, MediaSPIP aims to manage as many different media types as possible. The following are just a few examples from an ever-expanding list of supported formats : images : png, gif, jpg, bmp and more audio : MP3, Ogg, Wav and more video : AVI, MP4, OGV, mpg, mov, wmv and more text, code and other data : OpenOffice, Microsoft Office (Word, PowerPoint, Excel), web (html, CSS), LaTeX, Google Earth and (...)
Sur d’autres sites (8390)
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FFMPEG SAR mismatch when concatenating videos
6 septembre 2020, par marcmanI'm trying to concatenate videos with different codecs using the
concatfilter. I'm also doing a bunch of processing to scale and pad the individual clips in this filter as well. However, I keep getting an error of this form :

[Parsed_concat_14 @ 0x556b918ed580] Input link in1:v0 parameters (size 960x1280, SAR 0:1) do not match the corresponding output link in0:v0 parameters (1280x720, SAR 1:1)


Here is my command :


ffmpeg \
 -y \
 -loglevel warning \
 -stats \
 -i videos/vid0.mp4 \
 -i videos/vid1.mp4 \
 -i videos/vid2.mov \
 -filter_complex \
"[0:v]setpts=PTS-STARTPTS, scale=1920:1080, setsar=1, drawtext=expansion=strftime: basetime=$(date +%s -d'2020-07-10 16:04:44')000000 : fontcolor=white : text='%^b %d, %Y%n%l\\:%M%p' : fontsize=36 : y=1080-4*lh : x=1920-text_w-2*max_glyph_w; \
[1:v]setpts=PTS-STARTPTS, scale=810:1080, pad=width=1920:height=1080:x=555:y=0:color=black, setsar=1, drawtext=expansion=strftime: basetime=$(date +%s -d'2020-08-20 21:12:27')000000 : fontcolor=white : text='%^b %d, %Y%n%l\\:%M%p' : fontsize=36 : y=1080-4*lh : x=1365-text_w-2*max_glyph_w; \
[2:v]setpts=PTS-STARTPTS, scale=607:1080, pad=width=1920:height=1080:x=656:y=0:color=black, setsar=1, drawtext=expansion=strftime: basetime=$(date +%s -d'2020-08-27 16:42:26')000000 : fontcolor=white : text='%^b %d, %Y%n%l\\:%M%p' : fontsize=36 : y=1080-4*lh : x=1263-text_w-2*max_glyph_w; \
[0:v][0:a][1:v][1:a][2:v][2:a]concat=n=3:v=1:a=1[v][a]" \
 -map "[v]" \
 -map "[a]" \
 videos.mp4


This gives the following error :


[Parsed_concat_14 @ 0x563ab9d840c0] Input link in1:v0 parameters (size 960x1280, SAR 0:1) do not match the corresponding output link in0:v0 parameters (1280x720, SAR 1:1)
[Parsed_concat_14 @ 0x563ab9d840c0] Input link in2:v0 parameters (size 1080x1920, SAR 0:1) do not match the corresponding output link in0:v0 parameters (1280x720, SAR 1:1)


The input videos have these resolutions :


- 

- vid0.mp4 : (1280x720)
- vid1.mp4 : (960x1280)
- vid2.mp4 : (1080x1920)








The output resolution of the concatenated output is to be 1920x1080.


I've added the
setsar=1command after all my scaling and padding operations, as per this question and answer. I've also triedsetdar=16/9as in this answer, but it made no difference.

What am I missing here ?


-
NodeJS - efficiently and correctly convert from raw PCM to WAV at scale (without FFMPEG ?)
13 juillet 2024, par Royi BernthalI have a stream of raw PCM buffers I need to convert to playable WAV buffers.


@ffmpeg.wasmcan convert an individual buffer in the stream well, but it's limited to executing 1 command at a time, so it won't work in a real-life streaming scenario (streams x concurrent users). We can use it as a reference to a conversion that outputs a good playable wav.

import { FFmpeg, createFFmpeg, fetchFile } from '@ffmpeg.wasm/main';

async pcmToWavFFMPEG(buffer: Buffer) {
 // bitDepth - PCM signed 16-bit little-endian
 const options = { sampleRate: '24k', channels: '1', bitDepth: 's16le' };

 this.ffmpeg.FS('writeFile', 'input.pcm', await fetchFile(buffer));

 await this.ffmpeg.run(
 '-f',
 options.bitDepth,
 '-ar',
 options.sampleRate,
 '-ac',
 options.channels,
 '-i',
 'input.pcm',
 'output.wav',
 );

 const wavBuffer = this.ffmpeg.FS('readFile', 'output.wav');

 this.ffmpeg.FS('unlink', `input.pcm`);
 this.ffmpeg.FS('unlink', `output.wav`);

 return Buffer.from(wavBuffer);
 }


In order to get over the command execution limit, I've tried
fluent-ffmpeg. I couldn't find a way to convert a single buffer, so I'm just passing the whole readable stream so that ffmpeg can convert all of its buffers to wav. The buffers I'm getting in on('data') aren't playable wav. The same is true for concatting the accumulated buffers in on('complete') - the result is not a playable wav.

import ffmpeg from 'fluent-ffmpeg';
import internal from 'stream';

async pcmToWavFluentFFMPEG(
 readable: internal.Readable,
 callback: (chunk: Buffer) => void,
 ) {
 const options = { sampleRate: 24000, channels: 1, bitDepth: 's16le' };

 ffmpeg(readable)
 .inputFormat(options.bitDepth)
 .audioFrequency(options.sampleRate)
 .audioChannels(options.channels)
 .outputFormat('wav')
 .pipe()
 .on('data', callback);
 }


I've also tried using
node-wavto convert each buffer individually. It manages to convert everything to playable wavs that sound close to the desired result, however for some reason they're extremely loud and sound a bit weird.

import wav from 'node-wav';

pcmToWavBad(buffer: Buffer) {
 const pcmData = new Int16Array(
 buffer.buffer,
 buffer.byteOffset,
 buffer.byteLength / Int16Array.BYTES_PER_ELEMENT,
 );

 const channelData = [pcmData]; // assuming mono channel

 return wav.encode(channelData, { sampleRate: 24000, bitDepth: 16 });
 }


I've also tried wrapping the PCM as a WAV with
wavefilewithout any actual conversion (which is redundant as PCM is contained as is in WAV), but it results in white noise :

import { WaveFile } from 'wavefile';

pcmToWav(buffer: Buffer) {
 const wav = new WaveFile();

 wav.fromScratch(1, 24000, '16', buffer); // 's16le' is invalid

 return Buffer.from(wav.toBuffer());
 }


-
Creating a continuous stream for RTMP in Node.js
9 mars 2023, par hankthetank27I'm working on an app that downloads audio from youtube that will be streamed sequentially, similar to radio broadcast. I'm having trouble getting the individual tracks to stream continuously. My idea was to write the tracks sequentially into a readable stream that then is read by FFMPEG. Here is the code I'm having trouble with...


import ytdl from "ytdl-core";
import ffmpeg from 'fluent-ffmpeg';
import { Readable } from "node:stream"

export async function startAudioStream(): Promise<void>{

 const mainStream = await createStreamedQueue()

 ffmpeg(mainStream)
 .inputOptions([
 '-re'
 ])
 .outputOption([
 // '-c:v libx264',
 // '-preset veryfast',
 // '-tune zerolatency',
 '-c:a aac',
 '-ar 44100',
 ])
 .save('rtmp://localhost/live/main.flv');
};

async function createStreamedQueue(): Promise<readable>{

 function createStream(): Readable{
 const stream = new Readable({
 read(){},
 highWaterMark: 1024 * 512,
 });
 stream._destroy = () => { stream.destroyed = true };
 return stream;
 }

 const mainStream = createStream();

 const ref1 = 'https://youtu.be/lLCEUpIg8rE'
 const ref2 = 'https://youtu.be/bRdyzdXJ0KA';

 function queueSong(src: string, stream: Readable): Promise<void>{
 return new Promise<void>((resolve, reject) => {
 ytdl(src, {
 filter: 'audioonly',
 quality: 'highestaudio'
 })
 .on('data', (data) => {
 stream.push(data);
 })
 .on('end', () => {
 resolve();
 })
 .on('error', (err) => {
 console.error('Error downloading file from YouTube.', err);
 reject(err);
 })
 })
 }
 
 await queueSong(ref1, mainStream);
 // console.log('after firsrt: ', mainStream)
 await queueSong(ref2, mainStream);
 // console.log('after second: ', mainStream)
 return mainStream;
};
</void></void></readable></void>

To start,
startAudioStreamis called to initiate a readable stream of audio to an RTMP server via FFMPEG. That is working fine. The part I'm having trouble with is "queuing" the tracks into the stream that's being fed into FFMPEG. Right now, I have a "main" stream that each songs data is being pushed into, as you can see inqueueSong. At the end of ytdl stream, the promise is resolved, allowing for the next song to be queued and its data to be pushed intomainStream. The issue that I'm experiencing is that the audio fromref1is only every played.

I can see in the logs that
mainStreamdoes grow in length after each call toqueueSong, but still will only stream the audio from the first track. My initial thought was that there is a terminating character at the of the last data chunk thats being written to the steam for each song ? But maybe im getting screwed up on how streams work...

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