I am trying to implement a webrtc-based chat room. And I encountered the following problems in laptop. Without connecting to any other peer, just use getUserMedia(), I can get local video stream.
When I unmuted the <video>, echo happened.
Then I wear headphones, and found there is a continued noise. And I can hear my voice clearly.
I tried to turn down the volume, but it doesn't work.
Thanks in advance.
Make sure that you are muting the local <video> element if you have it in the DOM:
<video id="vid1" autoplay="true" muted="muted"></video>
See this post on the discuss-webrtc mailing list for more details and the WebRTC samples.
Do the followings:
1) In localVideo do the this:
localVideo.volume = 0;
localVideo.muted = 0;
2) Do same for the remoteVideo also:
remoteVideo.volume = 0;
remoteVideo.muted = 0;
To resolve noise related issue, you should set autoplay=false for localstream.
Related
I am trying to make myself Digital audio to Analog audio converter
I have STM32F769i Discovery0: https://www.st.com/en/evaluation-tools/32f769idiscovery.html
Which has SPDIFRX and SPDIFTX ports
I found a fearly good starting point here: http://www.tjaekel.com/DiscoveryF7Audio/index.html
Also the guy posted here: https://www.openstm32.org/forumthread921
But the guy used STM32746G Discovery: https://www.st.com/en/evaluation-tools/32f746gdiscovery.html
Instead
So I went and tried to just get his SPDIF audio portion working on my board
My Project can be found here (I hope it compiles, with CubeIDE you never know what will happen :)): https://www.mediafire.com/file/n0s2z9p6nn735qg/SPDIF_Example.zip/file
I have no idea what I am doing wrong, but for some reason SPDIF_RX_IRQHandler (in stm32f7xx_it.c) is never called (LED never turns green, yea my debugging tehniques are primitive, but will explain why later)
So because of that HAL_SPDIFRX_ReceiveDataFlow_IT (in spdifrx.c) always returns HAL_TIMEOUNT, and of course no audio is ever played on the speakers
I am not sure what I am doing wrong
When I start MCU I call BSP_SPDIF_Init() (defined in spdifrx.c) in main.c right after I take care of the clock
if (BSP_SPDIFRX_Init() != HAL_OK)
{
Error_Handler();
}
And it appears it initializes all right, because I get HAL_OK back
Maybe I am not inializing GPIO properly from HAL_MspInit in stm32f7xx_hal_msp.c inproperly
I am realy out of ideas, what I am doing wrong, because the analog side of the audio does init, I can hear that as pop pop from the speakers when I power up my MCU, its just that SPDIF side has problems
I am is my setup crocked?
I am using this component radio as my SPDIF transmitter (Hama DIT2000M): https://de.hama.com/webresources/article-documents/00054/man/00054821man_en.pdf
It says it has SPDIF Audio out (it says its digital over coaxial)
I know its optical side is working fine because on my component receiver it plays just fine (it reports as 48khz Stereo)
Is my cable to long? I am using this cable: https://i.imgur.com/JqAxePF.jpg
(not sure who made it)
Now why do I debug with blinking leds, because where my test subject is (my Hama receiver), there is no computer so…. Blinking leds it is, I would like to avoid aditional libraries and have a minimum working example, because you never know what problems they could bring so that's why LCD is not used right now
I hope someone has any advice, eather how to get any data in to SPDIF port (because right now for some reason, I don't get anything) or what I am doing wrong for my audio not to play, the usage of STM32F769i Discovery0 instead of STM32746G Discovery is probably not responsible
I hope that this is a proper place for this king of questions, because I did ask a question regarding SPDIF on the STM forum: https://community.st.com/s/feed/0D53W00001z0RaqSAE
But didn't get any usefull advice there
Now SPDIF realy does not have much examples, there is only a polling example which does work (with the same cable), there is no interupt example, my interupt example (you can read the post on the STM forum post I linked) is not working as well (interupts are probably not broken right?)
So yea, I am lost a bit not sure what to do, and who to ask, so I tried here
PS: I know stackvoverflow does not like links to code, but I believe something is wrong with my project (interupts don't fire for some reason), and its realy hard to put this all into the question
Thanks for Anwsering and Best Regards
I managed to solve this, I guess I did not initialize SPDIF GPIO properly
after setting this
GPIO_InitStructure.Pin = GPIO_PIN_7;
GPIO_InitStructure.Mode = GPIO_MODE_AF_PP;
GPIO_InitStructure.Pull = GPIO_NOPULL;
GPIO_InitStructure.Speed = GPIO_SPEED_FAST;
GPIO_InitStructure.Alternate = DISCOVERY_SPDIFRX_AF;
HAL_GPIO_Init(GPIOD, &GPIO_InitStructure);
to this
GPIO_InitStructure.Pin = GPIO_PIN_12;
GPIO_InitStructure.Mode = GPIO_MODE_AF_PP;
GPIO_InitStructure.Pull = GPIO_NOPULL;
GPIO_InitStructure.Speed = GPIO_SPEED_FAST;
GPIO_InitStructure.Alternate = GPIO_AF7_SPDIFRX;
HAL_GPIO_Init(GPIOG, &GPIO_InitStructure);
interupts started to fire
Good day to you all,
I have a problem, maybe someone can provide some helpful ideas on how to implement or if it's even possible:
I want to record RTSP stream from an IP-cam and I would like to add some text information and logo into the recording so it might be viewed when played back.
To do so I have first created one MediaPlayer element to connect to the IP-cam, duplicate onto the display, and recast via UDP.
using (var stream01_view = new Media(libVLC, "rtsp://192.168.10.214:5554",FromType.FromLocation))
{
stream01_view.AddOption(
":sout=#duplicate{" +
"dst=display{noaudio}," +
"dst=std{access=udp,mux=ts,dst=:1234}");
stream01_view.AddOption(":sout-keep");
player.Play(stream01_view);
}
The second stream connects to the local UDP cast and outputs to file
using (var stream01_record = new Media(libVLC, "udp://#:1234", FromType.FromLocation))
{
stream01_record.AddOption(":sout=#transcode{sfilter=marq}:file{mux=ts,dst=VideoMarqLogo.mp4}");
stream01_record.AddOption(":sout-keep");
recorder.Play(stream01_record);
}
Calling class MediaPlayer methods SetMarqueeInt and SetMarqueeString don't give the expected result.
Thanks mfkl for pointing to the right direction.
So the thing that does the trick is:
stream01_record.AddOption(":sout=#transcode{ vcodec=h264, scale=0.75, " +
"sfilter=marq{file='marq.txt',position=9}," +
"vfilter=logo{file='logo.png',position=6}}" +
":file{mux=ts,dst=VideoMarqLogo.mp4}");
A bit of a warning though, this piece of code is CPU intensive.
I'm left wondering if there could be a way to do this using GPU encoding.
I've developed a custom receiver which works very nicely, however the rendering of the graphic on the album art image looks weird. I've also tried the "styled media receiver", just to be sure that I haven't screwed anything up - same result.
Anyone else seen this (check the image below)? The channel logo is a transparent PNG.
Screenshot of bug
let shadowRootElement = document.querySelector( 'cast-media-player' ).shadowRoot;
let style = document.createElement( 'style' );
style.innerHTML = '#castMetadataImage { background-image: none !important; }';
shadowRootElement.appendChild( style )
Run this right after the player has been initizied and it should fix the issue.
Please share your code. It's difficult to know without any code to review. Did you follow the official guide on https://developers.google.com/cast/docs/caf_receiver/customize_ui for receivers applications? Also you need to check your sender application.
The issue seems to be related to the cssText property - removing this, fixes the rendering issue:
var fixDoubleLogoBug = function(){
try {
document.querySelector("#player")
.shadowRoot
.querySelector("#castMetadataImage").style.cssText = "";
}catch(e){
console.warn("Could not remove logo");
}
}
I am currently trying to write code for actionscript that will play a sound. Normally this wouldn't be a problem, however I need to play the sound with the stipulation that it doesn't begin playing until after the sound is finished loading. Is there a way to make sure the sound is loaded before I play it?
You can do the following:
var sound:Sound = new Sound();
sound.addEventListener(Event.COMPLETE, onSoundLoaded);
sound.load(new URLRequest("mySound.mp3"));
function onSoundLoaded(event:Event):void
{
var loadedSound:Sound = event.target as Sound;
loadedSound.play();
}
This is taken from this adobe tutorial.
Good luck coding!
I would like to create a simple add-on that would play a different MP3 recording every time the user double clicks a word in a webpage he is visiting and selects a special option from the context menu.
The MP3 files are located on a remote server. Normally I would use JavaScript+Flash to play the MP3 file. In a Firefox add-on, however, I'm unable to load external scripts for some reason (playing the sound works fine if it's the webpage itself that loads the scripts, but of course I need it to work with every website and not just the ones that include the script).
So what's the easiest way to play a remote MP3 file in a Firefox add-on using JavaScript?
This may not entirely solve your question, as I don't BELIEVE it plays MP3s, but I'm not certain.
Firefox has nsISound, which I KNOW can play remote WAV files, as I've tested and proved it.
You may want to test it for yourself and see if it leads you a little closer!
var ios = Components.classes['#mozilla.org/network/io-service;1'].getService(Components.interfaces.nsIIOService);
var sound = ios.newURI("http://www.yoursite.com/snds/haha.wav", null, null);
var player = Components.classes["#mozilla.org/sound;1"].createInstance(Components.interfaces.nsISound);
player.play(sound);
Good luck, I hope this at least gets you close!
I know this is an old question, but if someone needs a way to do it:
let player = document.createElement("audio");
player.src = browser.runtime.getURL(SOUND_URL);
player.play();
There is one caveat: the user must have allowed autoplay on the website.
Here is a working code....
var sound = Components.classes["#mozilla.org/sound;1"].createInstance(Components.interfaces.nsISound);
var soundUri = Components.classes['#mozilla.org/network/standard-url;1'].createInstance(Components.interfaces.nsIURI);
soundUri.spec = "chrome://secchat/content/RING.WAV";
sound.play(soundUri);
var window = require('sdk/window/utils').getMostRecentBrowserWindow();
var audio = ('http://example.com/audio.mp3');
audio.play();