Add PeerConnectionInterface::UpdateCallBitrate.

webrtc/call/call.cc

 /*
  *  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.
  */
 
@@ skipping 162 matching lines @@
                                const uint8_t* packet,
                                size_t length,
                                const PacketTime& packet_time) override;
 
   // Implements RecoveredPacketReceiver.
   bool OnRecoveredPacket(const uint8_t* packet, size_t length) override;
 
   void SetBitrateConfig(
       const webrtc::Call::Config::BitrateConfig& bitrate_config) override;
 
+  void UpdateBitrateConfig(
+      const webrtc::Call::Config::BitrateConfigMask& bitrate_config) override;
+
   void SignalChannelNetworkState(MediaType media, NetworkState state) override;
 
   void OnTransportOverheadChanged(MediaType media,
                                   int transport_overhead_per_packet) override;
 
   void OnNetworkRouteChanged(const std::string& transport_name,
                              const rtc::NetworkRoute& network_route) override;
 
   void OnSentPacket(const rtc::SentPacket& sent_packet) override;
 
@@ skipping 722 matching lines @@
   // no point in remembering for the future.
   if (bitrate_config.start_bitrate_bps > 0)
     config_.bitrate_config.start_bitrate_bps = bitrate_config.start_bitrate_bps;
   config_.bitrate_config.max_bitrate_bps = bitrate_config.max_bitrate_bps;
   RTC_DCHECK_NE(bitrate_config.start_bitrate_bps, 0);
   transport_send_->send_side_cc()->SetBweBitrates(
       bitrate_config.min_bitrate_bps, bitrate_config.start_bitrate_bps,
       bitrate_config.max_bitrate_bps);
 }
 
+void Call::UpdateBitrateConfig(
+    const webrtc::Call::Config::BitrateConfigMask& mask) {
+  TRACE_EVENT0("webrtc", "Call::UpdateBitrateConfig");
+  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+
+  const Config::BitrateConfig& current_config = config_.bitrate_config;
+  Config::BitrateConfig new_config;
+  new_config.min_bitrate_bps =
+      mask.min_bitrate_bps.value_or(current_config.min_bitrate_bps);
+  new_config.start_bitrate_bps =
+      mask.start_bitrate_bps.value_or(current_config.start_bitrate_bps);
+  new_config.max_bitrate_bps =
+      mask.max_bitrate_bps.value_or(current_config.max_bitrate_bps);
+
+  SetBitrateConfig(new_config);
+}
+
 void Call::SignalChannelNetworkState(MediaType media, NetworkState state) {
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   switch (media) {
     case MediaType::AUDIO:
       audio_network_state_ = state;
       break;
     case MediaType::VIDEO:
       video_network_state_ = state;
       break;
     case MediaType::ANY:
@@ skipping 390 matching lines @@
       (use_send_side_bwe && header.extension.hasTransportSequenceNumber)) {
     receive_side_cc_.OnReceivedPacket(
         packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(),
         header);
   }
 }
 
 }  // namespace internal
 
 }  // namespace webrtc