Updated structures and functions for setting the max bitrate limit to take rtc::Optional<int>

webrtc/call.h

Index: webrtc/call.h
diff --git a/webrtc/call.h b/webrtc/call.h
index 313c5e58c16f49446c8d6fbcdaa92950179628a8..a195aee5d3a3219579dd89d724e6c10363170b4c 100644
--- a/webrtc/call.h
+++ b/webrtc/call.h
@@ -17,6 +17,7 @@
 #include "webrtc/audio_receive_stream.h"
 #include "webrtc/audio_send_stream.h"
 #include "webrtc/audio_state.h"
+#include "webrtc/base/optional.h"
 #include "webrtc/base/socket.h"
 #include "webrtc/video_receive_stream.h"
 #include "webrtc/video_send_stream.h"
@@ -77,7 +78,7 @@ class Call {
     struct BitrateConfig {
       int min_bitrate_bps = 0;
       int start_bitrate_bps = kDefaultStartBitrateBps;
-      int max_bitrate_bps = -1;
+      rtc::Optional<int> max_bitrate_bps;
     } bitrate_config;
 
     // AudioState which is possibly shared between multiple calls.