Rename ChannelGroup to CongestionController and move to webrtc/call/.
BUG=webrtc:5079
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1419803002 .
Cr-Commit-Position: refs/heads/master@{#10358}
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index eff441c..b142453 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -21,6 +21,7 @@
#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"
@@ -96,7 +97,7 @@
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_;
@@ -141,8 +142,8 @@
: 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()),
@@ -168,9 +169,10 @@
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() {
@@ -235,7 +237,7 @@
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) ==
@@ -277,9 +279,10 @@
// 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.
@@ -338,8 +341,8 @@
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.
@@ -401,14 +404,15 @@
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.
@@ -439,9 +443,9 @@
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) {
@@ -450,7 +454,7 @@
// 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_) {
@@ -470,7 +474,7 @@
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) {
