Fix suspend below min bitrate in new API by making it possible to set min bitrate at the receive-si…

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 67 matching lines @@
   void SignalNetworkState(NetworkState state) override;
 
  private:
   DeliveryStatus DeliverRtcp(MediaType media_type, const uint8_t* packet,
                              size_t length);
   DeliveryStatus DeliverRtp(MediaType media_type,
                             const uint8_t* packet,
                             size_t length,
                             const PacketTime& packet_time);
 
-  void SetBitrateControllerConfig(
-      const webrtc::Call::Config::BitrateConfig& bitrate_config);
-
   void ConfigureSync(const std::string& sync_group)
       EXCLUSIVE_LOCKS_REQUIRED(receive_crit_);
 
   const int num_cpu_cores_;
   const rtc::scoped_ptr<ProcessThread> module_process_thread_;
   const rtc::scoped_ptr<ChannelGroup> channel_group_;
   volatile int next_channel_id_;
   Call::Config config_;
 
   // Needs to be held while write-locking |receive_crit_| or |send_crit_|. This
@@ skipping 51 matching lines @@
     VoECodec* voe_codec = VoECodec::GetInterface(config.voice_engine);
     if (voe_codec) {
       event_log_ = voe_codec->GetEventLog();
       voe_codec->Release();
     }
   }
 
   Trace::CreateTrace();
   module_process_thread_->Start();
 
-  SetBitrateControllerConfig(config_.bitrate_config);
+  channel_group_->SetBweBitrates(config_.bitrate_config.min_bitrate_bps,
+                                 config_.bitrate_config.start_bitrate_bps,
+                                 config_.bitrate_config.max_bitrate_bps);
 }
 
 Call::~Call() {
   RTC_CHECK_EQ(0u, video_send_ssrcs_.size());
   RTC_CHECK_EQ(0u, video_send_streams_.size());
   RTC_CHECK_EQ(0u, audio_receive_ssrcs_.size());
   RTC_CHECK_EQ(0u, video_receive_ssrcs_.size());
   RTC_CHECK_EQ(0u, video_receive_streams_.size());
 
   module_process_thread_->Stop();
@@ skipping 207 matching lines @@
           bitrate_config.min_bitrate_bps &&
       (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) {
     // Nothing new to set, early abort to avoid encoder reconfigurations.
     return;
   }
   config_.bitrate_config = bitrate_config;
-  SetBitrateControllerConfig(bitrate_config);
-}
-
-void Call::SetBitrateControllerConfig(
-    const webrtc::Call::Config::BitrateConfig& bitrate_config) {
-  BitrateController* bitrate_controller =
-      channel_group_->GetBitrateController();
-  if (bitrate_config.start_bitrate_bps > 0)
-    bitrate_controller->SetStartBitrate(bitrate_config.start_bitrate_bps);
-  bitrate_controller->SetMinMaxBitrate(bitrate_config.min_bitrate_bps,
-                                       bitrate_config.max_bitrate_bps);
+  channel_group_->SetBweBitrates(bitrate_config.min_bitrate_bps,
+                                 bitrate_config.start_bitrate_bps,
+                                 bitrate_config.max_bitrate_bps);
 }
 
 void Call::SignalNetworkState(NetworkState state) {
   // Take crit for entire function, it needs to be held while updating streams
   // to guarantee a consistent state across streams.
   rtc::CritScope lock(&network_enabled_crit_);
   network_enabled_ = state == kNetworkUp;
   {
     ReadLockScoped write_lock(*send_crit_);
     for (auto& kv : video_send_ssrcs_) {
@@ skipping 129 matching lines @@
     size_t length,
     const PacketTime& packet_time) {
   if (RtpHeaderParser::IsRtcp(packet, length))
     return DeliverRtcp(media_type, packet, length);
 
   return DeliverRtp(media_type, packet, length, packet_time);
 }
 
 }  // namespace internal
 }  // namespace webrtc