Update bug reporting and contributing docs
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}
diff --git a/docs/bug-reporting.md b/docs/bug-reporting.md
index c21186a..7948cda 100644
--- a/docs/bug-reporting.md
+++ b/docs/bug-reporting.md
@@ -22,9 +22,10 @@
* 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 @@
* 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 @@
* 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 762e9a9..9189481 100644
--- a/docs/native-code/development/contributing.md
+++ b/docs/native-code/development/contributing.md
@@ -38,6 +38,7 @@
[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:
