Uniform volume when ringing on speakerphone vs video

// FREEBIE

Signal/src/call/CallAudioService.swift

@@ -188,7 +188,10 @@ struct AudioSource: Hashable {
             // SoloAmbient plays through speaker, but respects silent switch
             setAudioSession(category: AVAudioSessionCategorySoloAmbient,
                             mode: AVAudioSessionModeDefault)
-        } else if call.hasLocalVideo {
+        } else if call.state == .connected, call.hasLocalVideo {
+            // Because ModeVideoChat affects gain, we don't want to apply it until the call is connected.
+            // otherwise sounds like ringing will be extra loud for video vs. speakerphone
+
             // Apple Docs say that setting mode to AVAudioSessionModeVideoChat has the
             // side effect of setting options: .allowBluetooth, when I remove the (seemingly unnecessary)
             // option, and inspect AVAudioSession.sharedInstance.categoryOptions == 0. And availableInputs
@@ -216,9 +219,14 @@ struct AudioSource: Hashable {
                 try session.setPreferredInput(call.audioSource?.portDescription)
             }
 
-            if call.isSpeakerphoneEnabled {
+            if call.isSpeakerphoneEnabled || (call.hasLocalVideo && call.state != .connected)  {
+                // We want consistent ringer-volume between speaker-phone and video chat.
+                // But because using VideoChat mode has noticeably higher output gain, we treat
+                // video chat like speakerphone mode until the call is connected.
+                Logger.verbose("\(TAG) enabling speakerphone overrideOutputAudioPort(.speaker)")
                 try session.overrideOutputAudioPort(.speaker)
             } else {
+                Logger.verbose("\(TAG) disabling spearkerphone overrideOutputAudioPort(.none) ")
                 try session.overrideOutputAudioPort(.none)
             }
         } catch {

Signal/src/call/CallAudioSession.swift

@@ -11,8 +11,7 @@ import WebRTC
  * permission requested (and recording banner) before they even know they have an incoming call.
  *
  * By using the `useManualAudio` and `isAudioEnabled` attributes of the RTCAudioSession we can delay recording until
- * it makes sense. However, the headers for RTCAudioSession are not exported by default, so we've vendored the header
- * into our project. See "Libraries/WebRTC"
+ * it makes sense.
  */
 class CallAudioSession {