diff --git a/docs/bug-reporting.md b/docs/bug-reporting.md index c21186a9b7..7948cda8b7 100644 --- a/docs/bug-reporting.md +++ b/docs/bug-reporting.md @@ -22,9 +22,10 @@ Anyone with a [Google account][1] can file bugs in the Chrome and WebRTC tracker * Identify which bug tracker to use: * If you're hitting a problem in Chrome, file the bug using the - [the Chromium issue wizard](https://chromiumbugs.appspot.com/?token=0) + [the Chromium issue wizard](https://crbug.com/new) Choose "Web Developer" and "API", then fill out the form. For the component choose * Blink>GetUserMedia for camera/microphone issues + * Blink>GetDisplayMedia for screen capture issues * Blink>MediaRecording for issues with the MediaRecorder API * Blink>WebRTC for issues with the RTCPeerConnection API This ensures the right people will look at your bug. @@ -51,10 +52,10 @@ Anyone with a [Google account][1] can file bugs in the Chrome and WebRTC tracker * Camera and microphone model and version (if applicable) - * For Chrome audio and video device issues, please run the tests at - . After the tests finish running, click the bug - icon at the top, download the report, and attach the report to the issue - tracker. + * Try reproducing with the minimal samples at + https://webrtc.github.io/samples/src/content/getusermedia/audio/ + and + https://webrtc.github.io/samples/src/content/getusermedia/gum/ * Web site URL @@ -76,17 +77,19 @@ Anyone with a [Google account][1] can file bugs in the Chrome and WebRTC tracker * For **connectivity** issues on Chrome, ensure **chrome://webrtc-internals** is open in another tab before starting the call and while the call is in progress, - * expand the **Create Dump** section, + * expand the **Create a WebRTC-Internals dump** section, - * click the **Download the PeerConnection updates and stats data** button. + * click the **Download the webrtc-internals dump** button. You will be prompted to save the dump to your local machine. Please - attach that dump to the bug report. + attach that dump to the bug report. You can inspect the dump and + remove any information you consider personally identifiable such as + IP addresses. * For **audio quality** issues on Chrome, while the call is in progress, * please open **chrome://webrtc-internals** in another tab, - * expand the **Create Dump** section, + * expand the **Create a WebRTC-Internals dump** section, * fill in the **Enable diagnostic audio recordings** checkbox. You will be prompted to save the recording to your local machine. After ending the diff --git a/docs/native-code/development/contributing.md b/docs/native-code/development/contributing.md index 762e9a9c36..918948166e 100644 --- a/docs/native-code/development/contributing.md +++ b/docs/native-code/development/contributing.md @@ -38,6 +38,7 @@ You will not have to repeat the above. After all that, you’re ready to upload: [AUTHORS]: https://webrtc.googlesource.com/src/+/refs/heads/main/AUTHORS [new-password]: https://webrtc.googlesource.com/new-password [discuss-webrtc]: https://groups.google.com/forum/#!forum/discuss-webrtc +[Chromium recommendations for code reviews]: https://chromium.googlesource.com/chromium/src/+/main/docs/cl_tips.md ### Uploading your First Patch Now that you have your account set up, you can do the actual upload: