Add PeerConnectionInterface::UpdateCallBitrate with call tests.

webrtc/call/call.h

Index: webrtc/call/call.h
diff --git a/webrtc/call/call.h b/webrtc/call/call.h
index ec73bf7714926b2a6b04210f3c66a881f9acbedf..563bd9f8d90c58a97f11fc733006110fcea2dc58 100644
--- a/webrtc/call/call.h
+++ b/webrtc/call/call.h
@@ -10,9 +10,11 @@
 #ifndef WEBRTC_CALL_CALL_H_
 #define WEBRTC_CALL_CALL_H_
 
+#include <memory>
 #include <string>
 #include <vector>
 
+#include "webrtc/api/rtcerror.h"
 #include "webrtc/base/networkroute.h"
 #include "webrtc/base/platform_file.h"
 #include "webrtc/base/socket.h"
@@ -20,6 +22,7 @@
 #include "webrtc/call/audio_send_stream.h"
 #include "webrtc/call/audio_state.h"
 #include "webrtc/call/flexfec_receive_stream.h"
+#include "webrtc/call/rtp_transport_controller_send.h"
 #include "webrtc/common_types.h"
 #include "webrtc/video_receive_stream.h"
 #include "webrtc/video_send_stream.h"
@@ -75,6 +78,14 @@ class Call {
       int max_bitrate_bps = -1;
     } bitrate_config;
 
+    // TODO(zstein): Consider using PeerConnectionInterface::BitrateParameters
+    // instead (and move BitrateParameters to its own file in api/).
+    struct BitrateConfigMask {
+      rtc::Optional<int> min_bitrate_bps;
+      rtc::Optional<int> start_bitrate_bps;
+      rtc::Optional<int> max_bitrate_bps;
+    };
+
     // AudioState which is possibly shared between multiple calls.
     // TODO(solenberg): Change this to a shared_ptr once we can use C++11.
     rtc::scoped_refptr<AudioState> audio_state;
@@ -100,6 +111,10 @@ class Call {
 
   static Call* Create(const Call::Config& config);
 
+  static Call* Create(

[Taylor_Brandstetter, 2017/04/26 15:46:11]

Could you add a comment that this Create method is provided so that
"transport_send" can be mocked/faked for testing?

[Zach Stein, 2017/05/04 22:32:43]

Done.

+      const Call::Config& config,
+      std::unique_ptr<RtpTransportControllerSendInterface> transport_send);
+
   virtual AudioSendStream* CreateAudioSendStream(
       const AudioSendStream::Config& config) = 0;
   virtual void DestroyAudioSendStream(AudioSendStream* send_stream) = 0;
@@ -136,14 +151,20 @@ class Call {
   // pacing delay, etc.
   virtual Stats GetStats() const = 0;
 
-  // TODO(pbos): Like BitrateConfig above this is currently per-stream instead
-  // of maximum for entire Call. This should be fixed along with the above.
-  // Specifying a start bitrate (>0) will currently reset the current bitrate
-  // estimate. This is due to how the 'x-google-start-bitrate' flag is currently
+  // The min and start values will only be used if they are not set by
+  // SetBitrateConfigMask. The minimum max set by the two calls will be used.
+  // Specifying a start bitrate (>0) will reset the current bitrate estimate.
+  // This is due to how the 'x-google-start-bitrate' flag is currently
   // implemented.
   virtual void SetBitrateConfig(
       const Config::BitrateConfig& bitrate_config) = 0;
 
+  // The min and start values set here are preferred to values set by
+  // SetBitrateConfig. The minimum of the max set by the two calls will be used.
+  // Assumes 0 <= min <= start <= max holds for set parameters.
+  virtual RTCError SetBitrateConfigMask(
+      const Config::BitrateConfigMask& bitrate_mask) = 0;
+
   // TODO(skvlad): When the unbundled case with multiple streams for the same
   // media type going over different networks is supported, track the state
   // for each stream separately. Right now it's global per media type.