jojobond / 3las Goto Github PK

Hi, thanks again for this great work!
Sometimes a client would be listening to an mp3 stream over WS perfectly fine (nice audio quality) and all of a sudden it starts playing with some background "static" and quality seems to drop. The source is fine, as I am listening to it on another client on a nother machine, where everything is fine. If I simply reload the player on the browser where I hear the bad quality.static, then it all goes back to nice quality.

This happens also when for example in Safari when we call "FormatReader.PurgeData();" after that call, static is heard on the audio playback. I can replicate this by simply adding a click handler to an HTML button on the playback page that calls "FormatReader.PurgeData();" from the click handler..

Hi,

is there a chance to avoid standby on iOS devices after a certain period independently from the users iOS settings? The stream always pauses when the screen goes dark.

I was thinking of providing the user with an additional audio stream that he/she can open directly in a native audio app when required such as VLC player. This way, the screen can go dark but audio will continue to play.

Maybe you can do it even with the HTML/JS-player but I could not find a way. DO you have an idea how to do this or recycle the server command for a suitable stream on a separate port?

#MP3
ffmpeg -y -f alsa -i hw:1 -rtbufsize 64 -probesize 64 \
-acodec libmp3lame -b:a 320k -ac 1 -reservoir 0 \
-f mp3 -write_xing 0 -id3v2_version 0 -fflags +nobuffer -flush_packets 1 - \
| node /home/3LAS/example/server/3las.stdinstreamer.js -port 9601 -type mpeg &

You have done an Awesome job! I was trying to think of how i was going to implement something like this. thanks a million. Now my question is, How would i send the ffmpeg output to 3LAS running on a remote server? you see i want to let the users use their own internet connection and not Wi-Fi to connect to the stream. but the stream should be sent from the recording computer to a remote server running 3LAS.
At the moment i'm using butt (broadcast using this tool) to record and stream the feed to an icecast server. the lag is ok for non-realtime application but for use with translation it is unacceptable. Your tool works great for this. but i am not sure how to pipe the output of ffmpeg to my google cloud vm instance. thanks again and any help on this matter would be greatly appreciated. Cheers.

Hello,

It's possible to use socket.IO instead of ws ?
It can solve two new issues (I opened here) with the client and the server part of 3LAS...

I want to add your project inside mine, a Low latency web mobile robot, For all I use socket.io but not for audio where I use HTML5 and ffserver to stream but I get 1 to 5 second latency... I need to stream the audio inside websocket tunel and add a WebAudio API player.

https://www.serveurperso.com/?page=robot

Hello,

I am fairly new to micro controllers but I have a project where I would like to live stream the audio signal of a microphone using a Raspberry Pico connected to the internet. For each connected microphone I can expect to have many thousand clients that need to hear the audio with latency below 1 second.
Do you think that it is achievable using 3LAS and does it work on Raspberry Pico for example ?

Client tested on Firefox version 55.0.3 and 53.0.3 (32 bits)

Log :

[17:44:26] Detected: Windows, Firefox

[17:44:26] Using MIME: audio/mpeg on port: 9601

[17:44:26] Init of HTMLPlayerControls succeeded

[17:44:26] Init of PCMAudioPlayer succeeded

[17:44:26] Init of AudioFormatReader succeeded

[17:44:27] Init of WebSocketClient failed: SecurityError: The operation is insecure.

Hi there!

First of all: Great tool, this is really high value. Many thanks for this!

Seems like I am doing something wrong:

• Nginx, NPM and npm ws installed
• Installed the server side somewhere on my Raspberry Pi home folder
• Installed the client side in the www-folder of my nginx installation
• Changed the 'SocketHost' variable in the script/3las.js to match the IP of the Raspberry Pi
• Ran sudo ffmpeg -y -f alsa -i hw:1 -rtbufsize 64 -probesize 64 -acodec pcm_s16le -ar 16000 -ac 1 -f s16le -fflags +nobuffer -packetsize 384 -flush_packets 1 - | node ./3las.stdinstreamer.js -port 9603 -type pcm -chunksize 384
• Opened site http://[IP]:8888/index.html and just see a black page with black text:
  Show/Hide event log We recommend viewing this page with Google Chrome. Download Google Chrome Ignore and continue

No audio playing, no player buttons shown. The console window does not give me any hint except for these messages for all the js files:
The script from “http://[IP]:8888/script/3las.helpers.js” was loaded even though its MIME type (“application/octet-stream”) is not a valid JavaScript MIME type.

Not sure what I could be doing wrong. Is there any more information in some log files?

Many thanks!

I don't know why but my previous issue seems to have been deleted :(

Anyway posting back. I have noticed that when using mp3, I clearly hear a kind of very light noise (similar to hearing an old vinyl). It is very clear on PC but on Mac I don't hear it.

Can this be something related to the audio decoding on Windows ?

Hi,

How many latency do you get ? In mp3 encoding, I got something very "real time" with an iOS (iPhone 6).
With an HTC One, Galaxy S7 or cheap Android, I always have, something like 400/500ms latency.
Do you have any idea to decrease this latency ?

Thanks you a lot for your hard and good work,

Marc

Android works without any problems.
The current situation is as follows.
1.You are calling.
2.Start streaming using the button on the screen.
Current time get Web Audio Context
3.call end.
4. 3las play streaming sound.
I expect it to be streaming. But there is no sound.

Steps:

Run the server
Start Playing,
Refresh the page

The server then prints an error message to the console:

events.js
throw er; Unhandled 'error' event

Error: read ECONNRESET
at TCP.onStreamRead (internal/stream_base_commons.js:200:27)....

Essentially it is failing when the websocket disconnects after it is played. So I need to rerun the server after it disconnects. Ideally it would keep running. I would try and fix it myself but unsure how.

From what I've seen from other websocket uses in nodeJS...

const Websocket = require('ws');
const wss = new WebSocket.Server({ port: 9999});

wss.on('connection, function(ws){ 
ws.on('close', console.log('disconnected');) 
});

whereas in 3las.stdinstreamer.js:

this.Server.on('connection', this.OnServerConnection.bind(this));

I think something like this might handle the error when dealing with a specific client? Currently there is no code to deal with any disconnections. I don't know how the "this" would carry into the function.

Sorry if this doesn't make any sense!!

hi. JoJoBond.
I useful 3LAS.
really thank you very much.

I have a few questions.

1. Noise is generated when the microphone function is activated.Is there any way to remove the noise?
2. if remove impossible noise.. how can I do sound mute?

I know it is an absurd question. But I believe you will help.

Hello,

It's a pita to get working.
I installed ws and dependency :

        3LAS/
        node_modules/safe-buffer/
        node_modules/ultron/
        node_modules/ws/

Node version :

(root|/var/www) node -v
v8.4.0
(root|/var/www) npm -v
5.3.0

I succefuly configured the ffmpeg cmdline to my hardware (you must move the "-ac 1" argument to begin of the ffmpeg cmdline.)

ffmpeg -y -f alsa -ac 1 -i hw:2,0 -rtbufsize 64 -probesize 64 \
-acodec libmp3lame -ab 320k -reservoir 0 -f mp3 \
-fflags +nobuffer - \
| nodejs stdinstreamer.js -port 9601 -type mp3 -burstsize 1

Error log :

(root|/var/www/3LAS/server) ./robot.sh
ffmpeg version N-81539-gd2e7431 Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 4.9.2 (Debian 4.9.2-10)
  configuration: --enable-gpl --enable-postproc --enable-swscale --enable-avfilter --enable-libmp3lame --enable-libvorbis --enable-libtheora --enable-libx264 --enable-libspeex --enable-shared --enable-pthreads --enable-libopenjpeg --enable-libfaac --enable-nonfree --extra-libs=-lasound
  libavutil      55. 29.100 / 55. 29.100
  libavcodec     57. 54.102 / 57. 54.102
  libavformat    57. 48.102 / 57. 48.102
  libavdevice    57.  0.102 / 57.  0.102
  libavfilter     6. 59.100 /  6. 59.100
  libswscale      4.  1.100 /  4.  1.100
  libswresample   2.  1.100 /  2.  1.100
  libpostproc    54.  0.100 / 54.  0.100
Guessed Channel Layout for Input Stream #0.0 : mono
Input #0, alsa, from 'hw:2,0':
  Duration: N/A, start: 1503762482.264030, bitrate: 768 kb/s
    Stream #0:0: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s
[mp3 @ 0x2351c00] Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar instead.
Output #0, mp3, to 'pipe:':
  Metadata:
    TSSE            : Lavf57.48.102
    Stream #0:0: Audio: mp3 (libmp3lame), 48000 Hz, mono, s16p, 320 kb/s
    Metadata:
      encoder         : Lavc57.54.102 libmp3lame
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s16le (native) -> mp3 (libmp3lame))
Press [q] to stop, [?] for help

/var/www/node_modules/ws/index.js:9
const WebSocket = require('./lib/WebSocket');
^^^^^
SyntaxError: Use of const in strict mode.
    at Module._compile (module.js:439:25)
    at Object.Module._extensions..js (module.js:474:10)
    at Module.load (module.js:356:32)
    at Function.Module._load (module.js:312:12)
    at Module.require (module.js:364:17)
    at require (module.js:380:17)
    at Object.<anonymous> (/var/www/3LAS/server/stdinstreamer.js:7:6)
    at Module._compile (module.js:456:26)
    at Object.Module._extensions..js (module.js:474:10)
    at Module.load (module.js:356:32)

If I remove use strict from all scripts, I get the illegal use of reserved word "class" strange error:(

새 텍스트 문서.txt

Hey there,

first of thanks for providing such a cool tool. Have been playing around with it a lot and it works amazingly well, except on iOS 11 (i'm using 11.02 on an iPhone 7). I hit the play button on my dev iPhone but i'm not able to hear any audio. The event log seems fine, tested in both Chrome and Safari.

[21:51:27] Detected: iOS, Chrome
[21:51:27] Using MIME: audio/mpeg on port: 9601
[21:51:27] Init of HTMLPlayerControls succeeded
[21:51:27] Init of PCMAudioPlayer succeeded
[21:51:27] Init of AudioFormatReader succeeded
[21:51:29] Init of WebSocketClient succeeded
[21:51:29] Trying to connect to server.
[21:51:29] Established connection with server.

Any clue how to solve this issue? Any help is very much appreciated!

hi JoJoBond.

I'm use WAV Stream player and mpeg player.
so....
What mean is TotalBatchSampleSize, TotalBatchByteSize?

I have 3LAS running behind a nginx reverse proxy; most of the time it works superbly well... but at some point i start to get a few "kipped buffer because it was too old;" errors and then it all the sudden fills up with this errror and stops playing.

Reloading the stream manually works; but i was wondering if we could 'somehow' restart the stream automatically after a number of 'too old' buffers; i've tried adding " this.NextScheduleTime = 0.0;" after the Logger.Log but has no effect.

Any proper way of doing it ? Thanks !

Very interesting project, I wonder if it would be possible to have it use a single port (over one port for each stream) and like a 'mountpoint' where clients can stream from like /mount1, /mount2 etc ...

Maybe with nginx ? Any idea on how to achieve this ?

Thanks !

Hi there,
1st of all great project! works great! except if there are brief network packet losses for a client. Where would I look to add some sort of small buffer (1 second for example) to reduce choppy audio??

Hi JoJoBond.
I am always grateful.
I don't understand you source file in 3las.formatreader.js
` AudioFormatReader_WAV.prototype.ExtractIntSamples = function () {
// Extract sample data from buffer

    var intSampleArray = new Uint8Array(this.DataBuffer.buffer.slice(0, this.TotalBatchByteSize));
    // Remove samples from buffer
    this.DataBuffer = new Uint8Array(this.DataBuffer.buffer.slice(this.BatchBytes));
    return intSampleArray;
};

`

Why are the cut lengths different?
I think... 0, this.TotalBatchByteSize and.. this.BataBuffer.buffer.slice(this.TotalBatchByteSize)

I wait for your answer.

I'm trying to find a low latency audio system to send audio from a shortwave radio through raspberry PI (With external sound card) to a webbrowser (Chrome), and came across 3LAS. It looks interesting and Im about to test it. But I see that there has not been done any work on this project for a long time.
Is it a dead end to use this one? Maybe someone else have similar javascript based solution?

Concerning the audio link, only low bitrate is needed in my case. (for SSB on short wave, a max of approximately 3kb audio bandwith should be enough, thus ~8kbit sampling rate should be good.

Br