Roll protocol to r1271365

ChromeDevTools · Mar 12, 2024 · 08fa435 · 08fa435
1 parent d932e1e
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diff --git a/changelog.md b/changelog.md
@@ -1,7 +1,26 @@
 
 
+## Roll protocol to r1271365 — _2024-03-12T04:27:11.000Z_
+######  Diff: [`d932e1e...a93a67b`](https://github.com/ChromeDevTools/devtools-protocol/compare/d932e1e...a93a67b)
+
+```diff
+@@ browser_protocol.pdl:6356 @@ domain Network
+       number uploadThroughput
+       # Connection type if known.
+       optional ConnectionType connectionType
++      # WebRTC packet loss (percent, 0-100). 0 disables packet loss emulation, 100 drops all the packets.
++      experimental optional number packetLoss
++      # WebRTC packet queue length (packet). 0 removes any queue length limitations.
++      experimental optional integer packetQueueLength
++      # WebRTC packetReordering feature.
++      experimental optional boolean packetReordering
+
+   # Enables network tracking, network events will now be delivered to the client.
+   command enable
+```
+
 ## Roll protocol to r1269399 — _2024-03-07T04:25:11.000Z_
-######  Diff: [`61cfdc3...4828871`](https://github.com/ChromeDevTools/devtools-protocol/compare/61cfdc3...4828871)
+######  Diff: [`61cfdc3...d932e1e`](https://github.com/ChromeDevTools/devtools-protocol/compare/61cfdc3...d932e1e)
 
 ```diff
 @@ browser_protocol.pdl:8953 @@ domain Page

diff --git a/json/browser_protocol.json b/json/browser_protocol.json
@@ -13818,6 +13818,27 @@
                             "description": "Connection type if known.",
                             "optional": true,
                             "$ref": "ConnectionType"
+                        },
+                        {
+                            "name": "packetLoss",
+                            "description": "WebRTC packet loss (percent, 0-100). 0 disables packet loss emulation, 100 drops all the packets.",
+                            "experimental": true,
+                            "optional": true,
+                            "type": "number"
+                        },
+                        {
+                            "name": "packetQueueLength",
+                            "description": "WebRTC packet queue length (packet). 0 removes any queue length limitations.",
+                            "experimental": true,
+                            "optional": true,
+                            "type": "integer"
+                        },
+                        {
+                            "name": "packetReordering",
+                            "description": "WebRTC packetReordering feature.",
+                            "experimental": true,
+                            "optional": true,
+                            "type": "boolean"
                         }
                     ]
                 },

diff --git a/package.json b/package.json
@@ -1,6 +1,6 @@
 {
   "name": "devtools-protocol",
-  "version": "0.0.1269399",
+  "version": "0.0.1271365",
   "description": "The Chrome DevTools Protocol JSON",
   "repository": "https://github.com/ChromeDevTools/devtools-protocol",
   "author": "The Chromium Authors",

diff --git a/pdl/browser_protocol.pdl b/pdl/browser_protocol.pdl
@@ -6356,6 +6356,12 @@ domain Network
       number uploadThroughput
       # Connection type if known.
       optional ConnectionType connectionType
+      # WebRTC packet loss (percent, 0-100). 0 disables packet loss emulation, 100 drops all the packets.
+      experimental optional number packetLoss
+      # WebRTC packet queue length (packet). 0 removes any queue length limitations.
+      experimental optional integer packetQueueLength
+      # WebRTC packetReordering feature.
+      experimental optional boolean packetReordering
 
   # Enables network tracking, network events will now be delivered to the client.
   command enable

diff --git a/types/protocol.d.ts b/types/protocol.d.ts
@@ -11220,6 +11220,18 @@ export namespace Protocol {
              * Connection type if known.
              */
             connectionType?: ConnectionType;
+            /**
+             * WebRTC packet loss (percent, 0-100). 0 disables packet loss emulation, 100 drops all the packets.
+             */
+            packetLoss?: number;
+            /**
+             * WebRTC packet queue length (packet). 0 removes any queue length limitations.
+             */
+            packetQueueLength?: integer;
+            /**
+             * WebRTC packetReordering feature.
+             */
+            packetReordering?: boolean;
         }
 
         export interface EnableRequest {