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

webrtc/api/mediacontroller.cc

Index: webrtc/api/mediacontroller.cc
diff --git a/webrtc/api/mediacontroller.cc b/webrtc/api/mediacontroller.cc
index 71964d52a7e0b5b5bcbadb7ecbcb825eda1910b1..f50da951ee8fb9ce5694b4ae119df2c790eec95f 100644
--- a/webrtc/api/mediacontroller.cc
+++ b/webrtc/api/mediacontroller.cc
@@ -63,7 +63,8 @@ class MediaController : public webrtc::MediaControllerInterface,
     call_config_.audio_state = media_engine->GetAudioState();
     call_config_.bitrate_config.min_bitrate_bps = kMinBandwidthBps;
     call_config_.bitrate_config.start_bitrate_bps = kStartBandwidthBps;
-    call_config_.bitrate_config.max_bitrate_bps = kMaxBandwidthBps;
+    call_config_.bitrate_config.max_bitrate_bps =
+        rtc::Optional<int>(kMaxBandwidthBps);
   }

[pthatcher1, 2016/03/17 21:51:28]

Can this be unset?

If not, why have it be optional?

[skvlad, 2016/03/18 00:49:17]

This value will become unset if a video stream is added with no bitrate limit.
I'm not that familiar with the codebase, but the behavior seems to be like this:
1. The Call gets created, and the maximum bitrate limit for the BWE is set at 2
Mbps.
2. When a video stream gets created, the max bitrate from codec parameters - if
any - overrides this limit. If it is not set - which is probably the common case
- this will clear the max bitrate in the call.
3. If there is a b= parameter in the remote description, it will override the
bitrate limit from the codec.

This probably also means that for audio-only calls with no limit in SDP the
bandwidth will remain capped at 2 Mbps.

So in summary - it can and most likely will be unset after the call starts.

   void Close_w() {
     RTC_DCHECK(worker_thread_->IsCurrent());