Add PeerConnectionInterface::UpdateCallBitrate.

webrtc/call/call.h

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 #ifndef WEBRTC_CALL_CALL_H_
@@ skipping 57 matching lines @@
     static const int kDefaultStartBitrateBps;
 
     // Bitrate config used until valid bitrate estimates are calculated. Also
     // used to cap total bitrate used.
     struct BitrateConfig {
       int min_bitrate_bps = 0;
       int start_bitrate_bps = kDefaultStartBitrateBps;
       int max_bitrate_bps = -1;
     } bitrate_config;
 
+    // TODO(zstein): Consider using PeerConnectionInterface::BitrateParameters
+    // instead (and move BitrateParameters to base/ or api/).

[Taylor Brandstetter, 2017/04/12 01:33:42]

Just FYI, it would be api/ since it's used in the API. The long term plan is for
api/ to not depend on any other directories.

[Zach Stein, 2017/04/13 00:26:36]

Acknowledged.

+    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;
 
     // Audio Processing Module to be used in this call.
     // TODO(solenberg): Change this to a shared_ptr once we can use C++11.
     AudioProcessing* audio_processing = nullptr;
 
     // RtcEventLog to use for this call. Required.
     // Use webrtc::RtcEventLog::CreateNull() for a null implementation.
@@ skipping 41 matching lines @@
 
   // All received RTP and RTCP packets for the call should be inserted to this
   // PacketReceiver. The PacketReceiver pointer is valid as long as the
   // Call instance exists.
   virtual PacketReceiver* Receiver() = 0;
 
   // Returns the call statistics, such as estimated send and receive bandwidth,
   // 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
-  // implemented.
+  // Parameters set here are only used if they are not masked by
+  // SetBitrateConfigMask. 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 implemented.
   virtual void SetBitrateConfig(
       const Config::BitrateConfig& bitrate_config) = 0;
 
+  // Values set here are preferred to values set by SetBitrateConfig.
+  virtual void SetBitrateConfigMask(

[Taylor Brandstetter, 2017/04/12 01:33:42]

Just an idea/bikeshed: calling these methods something like
"SetSdpBitrateConfig" and "SetApplicationBitrateConfig" may make it more clear
what the two represent.

[Zach Stein, 2017/04/13 00:26:36]

It felt wrong to talk about SDP here, so I tried to go with more generic terms.
I don't feel too strongly though.

+      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.
   virtual void SignalChannelNetworkState(MediaType media,
                                          NetworkState state) = 0;
 
   virtual void OnTransportOverheadChanged(
       MediaType media,
       int transport_overhead_per_packet) = 0;
 
   virtual void OnNetworkRouteChanged(
       const std::string& transport_name,
       const rtc::NetworkRoute& network_route) = 0;
 
   virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0;
 
   virtual ~Call() {}
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_CALL_CALL_H_