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

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 29 matching lines @@
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/video/call_stats.h"
 #include "webrtc/video/video_receive_stream.h"
 #include "webrtc/video/video_send_stream.h"
 #include "webrtc/video/vie_remb.h"
 #include "webrtc/voice_engine/include/voe_codec.h"
 
 namespace webrtc {
 
 const int Call::Config::kDefaultStartBitrateBps = 300000;
+// Indicates to the bandwidth estimator that there is no upper cap on the
+// estimated bitrate.
+static const int kBitrateUnlimited = -1;
 
 namespace internal {
 
 class Call : public webrtc::Call, public PacketReceiver,
              public BitrateObserver {
  public:
   explicit Call(const Call::Config& config);
   virtual ~Call();
 
   PacketReceiver* Receiver() override;
@@ skipping 144 matching lines @@
       first_packet_sent_ms_(-1),
       estimated_send_bitrate_sum_kbits_(0),
       pacer_bitrate_sum_kbits_(0),
       num_bitrate_updates_(0),
       remb_(clock_),
       congestion_controller_(new CongestionController(clock_, this, &remb_)) {
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   RTC_DCHECK_GE(config.bitrate_config.min_bitrate_bps, 0);
   RTC_DCHECK_GE(config.bitrate_config.start_bitrate_bps,
                 config.bitrate_config.min_bitrate_bps);
-  if (config.bitrate_config.max_bitrate_bps != -1) {
-    RTC_DCHECK_GE(config.bitrate_config.max_bitrate_bps,
+  if (config.bitrate_config.max_bitrate_bps) {
+    RTC_DCHECK_GT(*config.bitrate_config.max_bitrate_bps, 0);
+    RTC_DCHECK_GE(*config.bitrate_config.max_bitrate_bps,
                   config.bitrate_config.start_bitrate_bps);
   }
   if (config.audio_state.get()) {
     ScopedVoEInterface<VoECodec> voe_codec(voice_engine());
     event_log_ = voe_codec->GetEventLog();
   }
 
   Trace::CreateTrace();
   call_stats_->RegisterStatsObserver(congestion_controller_.get());
 
+  // The congestion controller uses -1 to represent unlimited bitrate.
+  // TODO(skvlad): Remove this conversion when congestion controller code
+  // is updated to use rtc::Optional.
+  int bwe_max_bitrate =
+      config_.bitrate_config.max_bitrate_bps.value_or(kBitrateUnlimited);
   congestion_controller_->SetBweBitrates(
       config_.bitrate_config.min_bitrate_bps,
-      config_.bitrate_config.start_bitrate_bps,
-      config_.bitrate_config.max_bitrate_bps);
+      config_.bitrate_config.start_bitrate_bps, bwe_max_bitrate);
   congestion_controller_->GetBitrateController()->SetEventLog(event_log_);
 
   module_process_thread_->Start();
   module_process_thread_->RegisterModule(call_stats_.get());
   module_process_thread_->RegisterModule(congestion_controller_.get());
   pacer_thread_->RegisterModule(congestion_controller_->pacer());
   pacer_thread_->RegisterModule(
       congestion_controller_->GetRemoteBitrateEstimator(true));
   pacer_thread_->Start();
 }
@@ skipping 290 matching lines @@
   stats.pacer_delay_ms = congestion_controller_->GetPacerQueuingDelayMs();
   stats.rtt_ms = call_stats_->rtcp_rtt_stats()->LastProcessedRtt();
   return stats;
 }
 
 void Call::SetBitrateConfig(
     const webrtc::Call::Config::BitrateConfig& bitrate_config) {
   TRACE_EVENT0("webrtc", "Call::SetBitrateConfig");
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   RTC_DCHECK_GE(bitrate_config.min_bitrate_bps, 0);
-  if (bitrate_config.max_bitrate_bps != -1)
-    RTC_DCHECK_GT(bitrate_config.max_bitrate_bps, 0);
+  if (bitrate_config.max_bitrate_bps) {
+    RTC_DCHECK_GT(*bitrate_config.max_bitrate_bps, 0);
+  }
   if (config_.bitrate_config.min_bitrate_bps ==
           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;
+  // The congestion controller uses -1 to represent unlimited bitrate.
+  // TODO(skvlad): Remove this conversion when congestion controller code
+  // is updated to use rtc::Optional.
+  int bwe_max_bitrate =
+      bitrate_config.max_bitrate_bps.value_or(kBitrateUnlimited);
   congestion_controller_->SetBweBitrates(bitrate_config.min_bitrate_bps,
                                          bitrate_config.start_bitrate_bps,
-                                         bitrate_config.max_bitrate_bps);
+                                         bwe_max_bitrate);
 }
 
 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;
@@ skipping 232 matching lines @@
   // thread. Then this check can be enabled.
   // RTC_DCHECK(!configuration_thread_checker_.CalledOnValidThread());
   if (RtpHeaderParser::IsRtcp(packet, length))
     return DeliverRtcp(media_type, packet, length);
 
   return DeliverRtp(media_type, packet, length, packet_time);
 }
 
 }  // namespace internal
 }  // namespace webrtc