Create a private API for controlling WebRTC's AEC3

chrome/browser/extensions/api/webrtc_audio_private/webrtc_audio_private_api.h

Index: chrome/browser/extensions/api/webrtc_audio_private/webrtc_audio_private_api.h
diff --git a/chrome/browser/extensions/api/webrtc_audio_private/webrtc_audio_private_api.h b/chrome/browser/extensions/api/webrtc_audio_private/webrtc_audio_private_api.h
index fda70d7faee0dddb63440905076de37d2109bda4..279c20238f61962b88b35a2567ee16454a755a6c 100644
--- a/chrome/browser/extensions/api/webrtc_audio_private/webrtc_audio_private_api.h
+++ b/chrome/browser/extensions/api/webrtc_audio_private/webrtc_audio_private_api.h
@@ -101,6 +101,13 @@ class WebrtcAudioPrivateFunction : public ChromeAsyncExtensionFunction {
   // |InitDeviceIDSalt()|.
   std::string device_id_salt() const;
 
+  // Returns the RenderProcessHost associated with the given |request|
+  // authorized by the |security_origin|. Returns null if unauthorized or
+  // the RPH does not exist.
+  content::RenderProcessHost* GetRenderProcessHostFromRequest(
+      const api::webrtc_audio_private::RequestInfo& request,
+      const std::string& security_origin);
+
  private:
   std::string device_id_salt_;
 
@@ -227,6 +234,21 @@ class WebrtcAudioPrivateGetAssociatedSinkFunction
   media::AudioDeviceDescriptions source_devices_;
 };
 
+class WebrtcAudioPrivateSetAudioExperimentsFunction
+    : public WebrtcAudioPrivateFunction {
+ public:
+  WebrtcAudioPrivateSetAudioExperimentsFunction();
+
+ protected:
+  ~WebrtcAudioPrivateSetAudioExperimentsFunction() override;
+
+ private:
+  DECLARE_EXTENSION_FUNCTION("webrtcAudioPrivate.setAudioExperiments",
+                             WEBRTC_AUDIO_PRIVATE_SET_AUDIO_EXPERIMENTS);
+
+  bool RunAsync() override;
+};
+
 }  // namespace extensions
 
 #endif  // CHROME_BROWSER_EXTENSIONS_API_WEBRTC_AUDIO_PRIVATE_WEBRTC_AUDIO_PRIVATE_API_H_