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Update bug reporting and contributing docs

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test.webrtc.org is gone and webrtc-internals got some updates which make
it more clear which dump is used

BUG=None

No-Try: true
Change-Id: I040e54398ced78148345804a4ab4922f67de133d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312360
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Christoffer Jansson <jansson@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40463}
This commit is contained in:
Philipp Hancke 2023-07-20 10:44:19 +02:00 committed by WebRTC LUCI CQ
parent f09fba81be
commit 15f0fabfb3
2 changed files with 13 additions and 9 deletions
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21
docs/bug-reporting.md
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@ -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
<https://test.webrtc.org>. 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

1
docs/native-code/development/contributing.md
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@ -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: