为什么WebRTC丢弃音频 [英] Why Is WebRTC Dropping Audio

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问题描述

我正在制作一个视频会议应用程序,其中音频非常重要.我刚刚克服了一个主要的障碍(花了我2个半星期的时间,大约花了2个星期),切换了音频设备,但是现在当我要在2台计算机上对其进行测试并打开我的麦克风时,音频变得安静.然后,我打开了麦克风输入,在音量下降之前,该麦克风输入工作了大约相同的时间.我什至尝试将autoGainControl设置为false,但这似乎无济于事.有没有简单的方法可以做到这一点?

I am making a video conferencing app where audio is very important. I just overcame a major hurdle (took me about 2 weeks out of a 2 and a half weeks of working on it), switching audio devices, but now when I go to test it on 2 computers and turn up my microphone the audio goes quiet. I then turned up the mic input, which worked for about the same amount of time before the volume dropped. I even tried setting autoGainControl to false, but that didn't seem to do anything. Is there an easy way to do this?

我尝试过:noiseSuppression: false,(显然是autoGainControl),echoCancellation: falsechannelCount: 2latency: 0,但所有这些均无效.这是getUserMedia的当前代码:

I have tried: noiseSuppression: false, (obviously autoGainControl), echoCancellation: false, channelCount: 2, latency: 0, all of which did not work. Here is the current code for getUserMedia:

                audio: {
                    echoCancellation: echoCancellationMASTER,
                    noiseSuppression: noiseSuppressionMASTER,
                    latency: 0,
                    sampleRate: 48000,
                    sampleSize: 24,
                    autoGainControl: false
                    /*autoGainControl: false,
                    channelCount: 2,
                    latency: 0,
                    volume: 3.0*/
                }

请注意,注释掉的代码是我也尝试过的代码,但当前未使用.

单击此处获取github代码

感谢任何想法,谢谢!!

Any ideas are appreciated, thanks!!

推荐答案

首先,转到chrome://webrtc-internals以确保按照期望的方式正确应用了约束.

Firstly, go to chrome://webrtc-internals to ensure your constraints are getting applied correctly, in the way that you expect them to be.

接下来,在每个耳机上使用一组耳机以消除反馈的可能性.这将有助于确认问题,我怀疑这是某处的某些回声消除算法.一旦确认是这种情况...

Next, use a set of headphones on each to eliminate the possibility of feedback. This will help confirm the problem, which I suspect is some echo cancellation algorithm somewhere. Once you've confirmed that's the case...

即使将echoCancellation设置为false,并且将noiseSuppression设置为false许多设备仍然具有其自己的增强"功能.用于减少反馈.我怀疑这就是您所听到的.

Even though you set echoCancellation to false, and noiseSuppression to false, many devices still have their own "enhancements" for feedback reduction. I suspect this is what you're hearing.

如果是这种情况,那么作为一名Web开发人员,您实际上无能为力.它位于与浏览器完全不同的一层.在将麦克风用作通讯"功能时,我经常会看到联想的Realtek驱动程序出现此问题.设备,但在其他计算机上也有相同的问题.

If this is the case, then there really isn't anything you can do about it as a web developer. It's at a whole different layer apart from the browser. I often see this problem with Lenovo's Realtek drivers when using the microphone as a "communications" device, but have had the same issue on other computers.

这篇关于为什么WebRTC丢弃音频的文章就介绍到这了,希望我们推荐的答案对大家有所帮助,也希望大家多多支持IT屋!

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