Rename ChannelGroup to CongestionController and move to webrtc/call/.

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.
  */
 
 #include <string.h>
 
 #include <map>
 #include <vector>
 
 #include "webrtc/audio/audio_receive_stream.h"
 #include "webrtc/audio/audio_send_stream.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/call.h"
+#include "webrtc/call/congestion_controller.h"
 #include "webrtc/call/rtc_event_log.h"
 #include "webrtc/common.h"
 #include "webrtc/config.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/utility/interface/process_thread.h"
 #include "webrtc/system_wrappers/interface/cpu_info.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/interface/logging.h"
 #include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
@@ skipping 55 matching lines @@
                             const uint8_t* packet,
                             size_t length,
                             const PacketTime& packet_time);
 
   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<CallStats> call_stats_;
-  const rtc::scoped_ptr<ChannelGroup> channel_group_;
+  const rtc::scoped_ptr<CongestionController> congestion_controller_;
   Call::Config config_;
   rtc::ThreadChecker configuration_thread_checker_;
 
   // Needs to be held while write-locking |receive_crit_| or |send_crit_|. This
   // ensures that we have a consistent network state signalled to all senders
   // and receivers.
   rtc::CriticalSection network_enabled_crit_;
   bool network_enabled_ GUARDED_BY(network_enabled_crit_);
 
   rtc::scoped_ptr<RWLockWrapper> receive_crit_;
@@ skipping 24 matching lines @@
 Call* Call::Create(const Call::Config& config) {
   return new internal::Call(config);
 }
 
 namespace internal {
 
 Call::Call(const Call::Config& config)
     : num_cpu_cores_(CpuInfo::DetectNumberOfCores()),
       module_process_thread_(ProcessThread::Create("ModuleProcessThread")),
       call_stats_(new CallStats()),
-      channel_group_(new ChannelGroup(module_process_thread_.get(),
-                     call_stats_.get())),
+      congestion_controller_(new CongestionController(
+          module_process_thread_.get(), call_stats_.get())),
       config_(config),
       network_enabled_(true),
       receive_crit_(RWLockWrapper::CreateRWLock()),
       send_crit_(RWLockWrapper::CreateRWLock()),
       event_log_(nullptr) {
   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,
                   config.bitrate_config.start_bitrate_bps);
   }
   if (config.voice_engine) {
     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();
   module_process_thread_->RegisterModule(call_stats_.get());
 
-  channel_group_->SetBweBitrates(config_.bitrate_config.min_bitrate_bps,
-                                 config_.bitrate_config.start_bitrate_bps,
-                                 config_.bitrate_config.max_bitrate_bps);
+  congestion_controller_->SetBweBitrates(
+      config_.bitrate_config.min_bitrate_bps,
+      config_.bitrate_config.start_bitrate_bps,
+      config_.bitrate_config.max_bitrate_bps);
 }
 
 Call::~Call() {
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   RTC_CHECK(audio_send_ssrcs_.empty());
   RTC_CHECK(video_send_ssrcs_.empty());
   RTC_CHECK(video_send_streams_.empty());
   RTC_CHECK(audio_receive_ssrcs_.empty());
   RTC_CHECK(video_receive_ssrcs_.empty());
   RTC_CHECK(video_receive_streams_.empty());
@@ skipping 44 matching lines @@
     RTC_DCHECK(num_deleted == 1);
   }
   delete audio_send_stream;
 }
 
 webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream(
     const webrtc::AudioReceiveStream::Config& config) {
   TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream");
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   AudioReceiveStream* receive_stream = new AudioReceiveStream(
-      channel_group_->GetRemoteBitrateEstimator(false), config);
+      congestion_controller_->GetRemoteBitrateEstimator(false), config);
   {
     WriteLockScoped write_lock(*receive_crit_);
     RTC_DCHECK(audio_receive_ssrcs_.find(config.rtp.remote_ssrc) ==
                audio_receive_ssrcs_.end());
     audio_receive_ssrcs_[config.rtp.remote_ssrc] = receive_stream;
     ConfigureSync(config.sync_group);
   }
   return receive_stream;
 }
 
@@ skipping 21 matching lines @@
 }
 
 webrtc::VideoSendStream* Call::CreateVideoSendStream(
     const webrtc::VideoSendStream::Config& config,
     const VideoEncoderConfig& encoder_config) {
   TRACE_EVENT0("webrtc", "Call::CreateVideoSendStream");
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
 
   // TODO(mflodman): Base the start bitrate on a current bandwidth estimate, if
   // the call has already started.
-  VideoSendStream* send_stream = new VideoSendStream(num_cpu_cores_,
-      module_process_thread_.get(), call_stats_.get(), channel_group_.get(),
-      config, encoder_config, suspended_video_send_ssrcs_);
+  VideoSendStream* send_stream = new VideoSendStream(
+      num_cpu_cores_, module_process_thread_.get(), call_stats_.get(),
+      congestion_controller_.get(), config, encoder_config,
+      suspended_video_send_ssrcs_);
 
   // This needs to be taken before send_crit_ as both locks need to be held
   // while changing network state.
   rtc::CritScope lock(&network_enabled_crit_);
   WriteLockScoped write_lock(*send_crit_);
   for (uint32_t ssrc : config.rtp.ssrcs) {
     RTC_DCHECK(video_send_ssrcs_.find(ssrc) == video_send_ssrcs_.end());
     video_send_ssrcs_[ssrc] = send_stream;
   }
   video_send_streams_.insert(send_stream);
@@ skipping 38 matching lines @@
   }
 
   delete send_stream_impl;
 }
 
 webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream(
     const webrtc::VideoReceiveStream::Config& config) {
   TRACE_EVENT0("webrtc", "Call::CreateVideoReceiveStream");
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   VideoReceiveStream* receive_stream = new VideoReceiveStream(
-      num_cpu_cores_, channel_group_.get(), config, config_.voice_engine,
-      module_process_thread_.get(), call_stats_.get());
+      num_cpu_cores_, congestion_controller_.get(), config,
+      config_.voice_engine, module_process_thread_.get(), call_stats_.get());
 
   // This needs to be taken before receive_crit_ as both locks need to be held
   // while changing network state.
   rtc::CritScope lock(&network_enabled_crit_);
   WriteLockScoped write_lock(*receive_crit_);
   RTC_DCHECK(video_receive_ssrcs_.find(config.rtp.remote_ssrc) ==
              video_receive_ssrcs_.end());
   video_receive_ssrcs_[config.rtp.remote_ssrc] = receive_stream;
   // TODO(pbos): Configure different RTX payloads per receive payload.
   VideoReceiveStream::Config::Rtp::RtxMap::const_iterator it =
@@ skipping 41 matching lines @@
   delete receive_stream_impl;
 }
 
 Call::Stats Call::GetStats() const {
   // TODO(solenberg): Some test cases in EndToEndTest use this from a different
   // thread. Re-enable once that is fixed.
   // RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   Stats stats;
   // Fetch available send/receive bitrates.
   uint32_t send_bandwidth = 0;
-  channel_group_->GetBitrateController()->AvailableBandwidth(&send_bandwidth);
+  congestion_controller_->GetBitrateController()->AvailableBandwidth(
+      &send_bandwidth);
   std::vector<unsigned int> ssrcs;
   uint32_t recv_bandwidth = 0;
-  channel_group_->GetRemoteBitrateEstimator(false)->LatestEstimate(
+  congestion_controller_->GetRemoteBitrateEstimator(false)->LatestEstimate(
       &ssrcs, &recv_bandwidth);
   stats.send_bandwidth_bps = send_bandwidth;
   stats.recv_bandwidth_bps = recv_bandwidth;
-  stats.pacer_delay_ms = channel_group_->GetPacerQueuingDelayMs();
+  stats.pacer_delay_ms = congestion_controller_->GetPacerQueuingDelayMs();
   {
     ReadLockScoped read_lock(*send_crit_);
     // TODO(solenberg): Add audio send streams.
     for (const auto& kv : video_send_ssrcs_) {
       int rtt_ms = kv.second->GetRtt();
       if (rtt_ms > 0)
         stats.rtt_ms = rtt_ms;
     }
   }
   return stats;
@@ skipping 10 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;
-  channel_group_->SetBweBitrates(bitrate_config.min_bitrate_bps,
-                                 bitrate_config.start_bitrate_bps,
-                                 bitrate_config.max_bitrate_bps);
+  congestion_controller_->SetBweBitrates(bitrate_config.min_bitrate_bps,
+                                         bitrate_config.start_bitrate_bps,
+                                         bitrate_config.max_bitrate_bps);
 }
 
 void Call::SignalNetworkState(NetworkState state) {
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   // 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;
-  channel_group_->SignalNetworkState(state);
+  congestion_controller_->SignalNetworkState(state);
   {
     ReadLockScoped write_lock(*send_crit_);
     for (auto& kv : audio_send_ssrcs_) {
       kv.second->SignalNetworkState(state);
     }
     for (auto& kv : video_send_ssrcs_) {
       kv.second->SignalNetworkState(state);
     }
   }
   {
     ReadLockScoped write_lock(*receive_crit_);
     for (auto& kv : video_receive_ssrcs_) {
       kv.second->SignalNetworkState(state);
     }
   }
 }
 
 void Call::OnSentPacket(const rtc::SentPacket& sent_packet) {
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
-  channel_group_->OnSentPacket(sent_packet);
+  congestion_controller_->OnSentPacket(sent_packet);
 }
 
 void Call::ConfigureSync(const std::string& sync_group) {
   // Set sync only if there was no previous one.
   if (config_.voice_engine == nullptr || sync_group.empty())
     return;
 
   AudioReceiveStream* sync_audio_stream = nullptr;
   // Find existing audio stream.
   const auto it = sync_stream_mapping_.find(sync_group);
@@ skipping 114 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