Chromium Code Reviews| Index: chrome/app/generated_resources.grd |
| diff --git a/chrome/app/generated_resources.grd b/chrome/app/generated_resources.grd |
| index 0af3e6333f24cd2eb516b07bd93bc3d90de57db6..dbecfaaf7d976173ee91b55fd4e96e84ccce9a3e 100644 |
| --- a/chrome/app/generated_resources.grd |
| +++ b/chrome/app/generated_resources.grd |
| @@ -5502,6 +5502,12 @@ Keep your key file in a safe place. You will need it to create new versions of y |
| <message name="IDS_FLAGS_WEBRTC_HW_H264_ENCODING_DESCRIPTION" desc="Description of chrome:flags option to turn on WebRTC hardware h264 video encoding support."> |
| Support in WebRTC for encoding h264 video streams using platform hardware. |
| </message> |
| + <message name="IDS_FLAGS_WEBRTC_SRTP_AES_GCM_NAME" desc="Name of chrome:flags option to enable GCM cipher suites for WebRTC"> |
|
grt (UTC plus 2)
2017/03/23 10:59:18
translateable="false" here and below
joachim
2017/03/23 20:07:10
Done.
|
| + Negotiation with GCM cipher suites for SRTP in WebRTC |
| + </message> |
| + <message name="IDS_FLAGS_WEBRTC_SRTP_AES_GCM_DESCRIPTION" desc="Description of chrome:flags option to enable GCM cipher suites for WebRTC"> |
| + When enabled, WebRTC will try to negotiate GCM cipher suites for SRTP. |
| + </message> |
| <message name="IDS_FLAGS_WEBRTC_STUN_ORIGIN_NAME" desc="Name of chrome:flags option to turn on Origin header for WebRTC STUN messages"> |
| WebRTC Stun origin header |
| </message> |