webrtc | webrtc学习整理 | Video Utils library

by feixiao JavaScript Version: Current License: MIT

X-Ray Key Features Code Snippets Community Discussions(10)Vulnerabilities Install Support

kandi X-RAY | webrtc Summary

webrtc is a JavaScript library typically used in Video, Video Utils applications. webrtc has no bugs, it has a Permissive License and it has low support. However webrtc has 1 vulnerabilities. You can download it from GitHub.

webrtc学习整理(业务和代码梳理)

Support

Quality

Security

License

Reuse

Support

webrtc has a low active ecosystem.

It has 102 star(s) with 56 fork(s). There are 10 watchers for this library.

It had no major release in the last 6 months.

webrtc has no issues reported. There are 1 open pull requests and 0 closed requests.

It has a neutral sentiment in the developer community.

The latest version of webrtc is current.

Quality

webrtc has 0 bugs and 0 code smells.

Security

webrtc has 1 vulnerability issues reported (0 critical, 0 high, 1 medium, 0 low).

webrtc code analysis shows 0 unresolved vulnerabilities.

There are 0 security hotspots that need review.

License

webrtc is licensed under the MIT License. This license is Permissive.

Permissive licenses have the least restrictions, and you can use them in most projects.

Reuse

webrtc releases are not available. You will need to build from source code and install.

webrtc saves you 115 person hours of effort in developing the same functionality from scratch.

It has 292 lines of code, 0 functions and 25 files.

It has low code complexity. Code complexity directly impacts maintainability of the code.

Top functions reviewed by kandi - BETA

kandi has reviewed webrtc and discovered the below as its top functions. This is intended to give you an instant insight into webrtc implemented functionality, and help decide if they suit your requirements.

process video data
Add video to track

Get all kandi verified functions for this library.

webrtc Key Features

No Key Features are available at this moment for webrtc.

webrtc Examples and Code Snippets

No Code Snippets are available at this moment for webrtc.

Community Discussions

Trending Discussions on webrtc

Xcode error 'building for iOS Simulator, but linking in dylib built for iOS .. for architecture arm64' from Apple Silicon M1 Mac

Quagga javascript barcode scanner - Uncaught TypeError: Quagga.init is not a function

WebRTC settings for Low Latency

WebRTC localConnection.setRemoteDescription(answer) pending for too long

Code doesn't even enter the onIceCandiate() while answering the SDP for webRTC in flutter

Android webrtc still running after app exited with back button

How to make WebRTC video streaming on local network working?

JSON type NSMutableDictionary cannot be converted to .sdp must not be null

Android Webrtc change stream to STREAM_MUSIC

How to notify the user about the offer call in flutter?

QUESTION

Xcode error 'building for iOS Simulator, but linking in dylib built for iOS .. for architecture arm64' from Apple Silicon M1 Mac

Asked 2021-Jun-14 at 09:55

I have an app which compiles and runs fine in older Macs with Intel processors in physical devices & iOS simulators.

The same app also compiles and runs fine from newer Apple Silicon Mac with M1 processor with physical iPhone devices, but, it refuse to be compiled for iOS simulator.

Without simulator support, debugging turn around time gets gets really long so I am trying to solve this issue. Not to mention Xcode preview feature isn't working either which is annoying.

The first error that I encountered without making any changes (but moved from Intel Mac to M1 Mac) is like below.

building for iOS Simulator, but linking in dylib built for iOS, file '/Users/andy/workspace/app/Pods/GoogleWebRTC/Frameworks/frameworks/WebRTC.framework/WebRTC' for architecture arm64

The Cocoapods library that I am using is GoogleWebRTC, and according to its doc, arm64 should be supported so I am baffled why above error is getting thrown. As I have said before, it compiles fine in real device which I believe is running on arm64.

According to the doc..

This pod contains the WebRTC iOS SDK in binary form. It is a dynamic library that contains the armv7, arm64 and x86_64 slices. Bitcode is not supported. Our currently provided API’s are Objective C only.

I searched online and it appears there appears to be 2 workarounds for this issue.

The first one is by adding arm64 to Excluded Architectures
The second option is to mark Build Active Architecture Only for Release build.

I don't exactly understand if above are necessary even when I am compiling my app on M1 Mac which is running under arm64 architecture, because the solution seems to be applicable only for for Intel Mac which does not support arm64 simulator, as for Intel Mac, simulators might have been running in x86_64, not with arm64, so solution #1 is not applicable in my case.

When I adapt the second change only, nothing really changes and the same error is thrown.

When I make both changes and tried building, I now get the following 2nd error during build. (not really 100% sure if I solved the 1st error / I might have introduced 2nd error in addition to 1st by adapting two changes)

Could not find module 'Lottie' for target 'x86_64-apple-ios-simulator'; found: arm64, arm64-apple-ios-simulator

The second library that I am using is lottie-ios and I am pulling this in with a swift package manager. I guess what is happening is that because I excluded arm64 in build setting for iOS simulator, Xcode is attempting to run my app in x86_64. However, library is not supported running in x86_64 for some reason, and is throwing an error. I don't have much insights into what dictates whether or not library can run in x86_64 or arm64 so I couldn't dig to investigate this issue.

My weak conclusion is that GoogleWebRTC cannot be compiled to run in iOS simulator with arm64 for some reason (unlike what its doc says), and lottie-ios cannot be compiled to run in iOS simulator with x86_64. So I cannot use them both in this case.

Q1. I want to know what kind of changes I can make to resolve this issue...

The app compiles and runs perfectly in both device & simulator when compiled from Intel Mac. The app compiles and runs fine in device when compiled from Apple Silicon Mac. It is just that app refuse to be compiled and run in iOS simulator from Apple Silicon Mac, and I cannot seem to figure out why.

Q2. If there is no solution available, I want to understand why this is happening in the first place.

I really wish not to buy old Intel Mac again just to make things work in simulator.

...

ANSWER

Answered 2021-Mar-27 at 20:15

Answering my own question in a hope to help others who are having similar problems. (and until a good answer is added from another user)

I found out that GoogleWebRTC actually requires its source to be compiled with x64 based on its source depo.

For builds targeting iOS devices, this should be set to either "arm" or "arm64", depending on the architecture of the device. For builds to run in the simulator, this should be set to "x64".

https://webrtc.github.io/webrtc-org/native-code/ios/

This must be why I was getting the following error.

building for iOS Simulator, but linking in dylib built for iOS, file '/Users/andy/workspace/app/Pods/GoogleWebRTC/Frameworks/frameworks/WebRTC.framework/WebRTC' for architecture arm64

Please correct me if I am wrong, but by default, it seems that Xcode running in Apple M1 silicon seems to launch iOS simulator with arm arch type. Since my app did run fine on simulators in Intel Mac, I did the following as a workaround for now.

Quit Xcode.
Go to Finder and open Application Folder.
Right click on Xcode application, select Get Info
In the "Xcode Info Window" check on Open using Rosetta.
Open Xcode and try running again.

That was all I needed to do to make my app, which relies on a library that is not yet fully supported on arm simulator, work again. (I believe launching Xcode in Rosetta mode runs simulator in x86 as well..?? which explains why things are working after making the above change)

A lot of online sources (often posted before M1 Mac launch on Nov/2020) talks about "add arm64 to Excluded Architectures", but that solution seems to be only applicable to Intel Mac, and not M1 Mac, as I did not need to make that change to make things work again.

Of course, running Xcode in Rosetta mode is not a permanent solution, and Xcode slows down lil bit, but it is an interim solution that gets things going in case one of libraries that you are using is not runnable in arm64 simulator.. yet.

Source https://stackoverflow.com/questions/65978359

QUESTION

Quagga javascript barcode scanner - Uncaught TypeError: Quagga.init is not a function

Asked 2021-Jun-12 at 16:55

I'm trying to write some vanilla javascript code to do barcode scanning from my website, however I can't even get past the first step using the Quagga javascript library. My code is currently this:

...

ANSWER

Answered 2021-Jun-12 at 16:55

Turns out I had to use https://cdnjs.cloudflare.com/ajax/libs/quagga/0.12.1/quagga.min.js instead.

Source https://stackoverflow.com/questions/67937543

QUESTION

WebRTC settings for Low Latency

Asked 2021-Jun-11 at 14:42

i searched already in Stack Overflow, but i was not able to get the Answer i searched for. I am currently developing a Remote Control App with WebRTC.

I played around with the WebRTC Settings. Like Resolution, Bitrate, Codec. But after a bit of trying, my experience was that it works best when i leave the default Settings.

I want to ask what the best Settings are for the lowest Latency possible. The Quality is not really important. The Resolution could also be changed.

i have the following Settings in Mind:

...

ANSWER

Answered 2021-Jun-11 at 14:42

WebRTC is optimized for low latency by itself, because it's targeted for conferencing applications, so - yes - you could just use default settings. WebRTC will automatically decrease quality in favor of lowest latency - you don't need to worry about it.

Here, however, are few pointers from my experience:

VP8 codec has lower latency than H264.
Framerate should be 25-30 fps, not lower (if you try 10-15 fps then you can see higher latency).
Use moderate frame sizes and bitrates (like 800x600 or 640x480 and 800-1000 kbps), because a. Encoding large frame sizes like HD takes a lot of CPU and may overload it, resulting in increasing latency; b. High bitrate can slow things down if your bandwidth is not sufficient.

Source https://stackoverflow.com/questions/67936948

QUESTION

WebRTC localConnection.setRemoteDescription(answer) pending for too long

Asked 2021-Jun-11 at 08:00

I am trying to implement a simple messaging mechanism between my browser (peer 1) and another browser (peer 2) on a different network. I am using Google's public STUN servers for learning.

Peer 1 does the following first:

...

ANSWER

Answered 2021-Feb-15 at 02:22

Here is a complete example of what you are trying to accomplish. Notice that it also has code for a Google STUN server, but it is remarked out: https://owebio.github.io/serverless-webrtc-chat/.

That page uses two iframes:

Create: https://owebio.github.io/serverless-webrtc-chat/noserv.create.html

Join: https://owebio.github.io/serverless-webrtc-chat/noserv.join.html.

This should get you started.

Also, two libraries built on WebTorrent exist that can aid in discovering and connecting to peers using only the browser: Bugout, P2PT.

Source https://stackoverflow.com/questions/65852106

QUESTION

Code doesn't even enter the onIceCandiate() while answering the SDP for webRTC in flutter

Asked 2021-Jun-10 at 03:24

The code flow doesn't even enter the onIceCandidate function while answering the SDP for webRTC connection. The webRTC is used for Voice calling for VOIP in android and I have also setted up TURN server with viagene website.

...

ANSWER

Answered 2021-Jun-10 at 03:24

So, I can clearly see that there you haven't set any local description for the remote user who is going to answer this call.

Source https://stackoverflow.com/questions/67905857

QUESTION

Android webrtc still running after app exited with back button

Asked 2021-Jun-10 at 01:59

I'm new to Webrtc, I'm using the AWS Webrtc demo with Android Nav Component. When I exited the app with the back button, I can see that Webrtc is still running or I can see the following log:

...

ANSWER

Answered 2021-Jun-09 at 11:13

This is the way you should destroy your WebRTC session on onDestroy() or onStop().

Source https://stackoverflow.com/questions/67863676

QUESTION

How to make WebRTC video streaming on local network working?

Asked 2021-Jun-06 at 16:49

I'm trying to establish peer connection between two clients via WebRTC and then stream the video from camera through the connection. The problem is, there's no video shown on the remote side, although I can clearly see the remotePc.ontrack event was fired. Also no error was thrown. I do NOT want to use the icecandidates mechanism (and it should NOT be needed), because the result application will only be used on a local network (the signaling server will only exchange the SDPs for the clients). Why is my example not working?

...

ANSWER

Answered 2021-Jun-06 at 16:49

ICE candidates are needed, as they tell you the local addresses where the clients will connect to each other.

You won't need STUN servers though.

Source https://stackoverflow.com/questions/67859207

QUESTION

JSON type NSMutableDictionary cannot be converted to .sdp must not be null

Asked 2021-Jun-05 at 06:38

Am new to webRTC and am trying to create a react native app with video calling functionality using this tutorial here as an example to follow https://dipanshkhandelwal.medium.com/video-calling-using-firebase-and-webrtc-14cc2d4afceb

However i keep getting this error on iOS and on android the app just closes once i try to join a call. The error i get on iOS says:

...

ANSWER

Answered 2021-Jun-05 at 06:38

I guess you are trying to use firebase as a signalling medium and want to use react-native-webrtc for the video calling.

Here is the sample code I have for the same solution with the latest libraries and react-native version.

Firebase Installation React Native.

Just set up ios and android using this above link and then use the below code for reference.

Source https://stackoverflow.com/questions/67671361

QUESTION

Android Webrtc change stream to STREAM_MUSIC

Asked 2021-Jun-02 at 07:37

I have created a WebRTC session from one device to another, the device should be able to control the volume for music stream, but WebRTC is originally designed to stream voice_call so is using the voice_call channel and using the call volume control is not good behavior for non-call app.

I tried to change STREAM_VOICE_CALL to STREAM_MUSIC in WebRTC source WebRtcAudioTrack to use the stream music volume but the only change was android is detecting it as music but volume change with call volume.

...

ANSWER

Answered 2021-Jun-02 at 07:37

I found the solution to this. You have to change the opensls player for this to happen

change this from here

Source https://stackoverflow.com/questions/67508050

QUESTION

How to notify the user about the offer call in flutter?

Asked 2021-Jun-02 at 06:18

My apps needs a VOIP call functionality and I use webRTC to achieve it. In webRTC how the reciever knows about the incoming call?

All my users will register in Django and flutter as a frontend. If I use FCM how can specify the exact user to send notifcation. Some articles suggest to use UID, email and such things if I have been authenticated with the firebase I might know about the UID but I use my own server How to make this possible?

In case if we use email to send a notification i.e., will firebase send the notification to the particular user?

...

ANSWER

Answered 2021-Jun-02 at 06:18

Use firebase cloud messaging. If you are using Django or whatever server for the backend you just have to get the user's fcm token while user registers from the app. And in your database store the user's email with that token. So whenever you want to send a notification to a specific user you can trigger by their respective fcm token.

Use below code to get user's token in flutterfire.

Source https://stackoverflow.com/questions/67799363

Community Discussions, Code Snippets contain sources that include Stack Exchange Network

Vulnerabilities

No vulnerabilities reported

Install webrtc

You can download it from GitHub.

Support

For any new features, suggestions and bugs create an issue on GitHub. If you have any questions check and ask questions on community page Stack Overflow .

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