webrtc/call/call.cc - Issue 2685673003: Define RtpTransportControllerSendInterface.

Unified Diff: webrtc/call/call.cc

Issue 2685673003: Define RtpTransportControllerSendInterface. (Closed)

Patch Set: Comment improvements. Created 3 years, 9 months ago

Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.

Jump to:

View side-by-side diff with in-line comments

Index: webrtc/call/call.cc

diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc

index bb83b7201671488289d4df52344cee29b5ebc931..8319346459eece7906d2f2321e9e3883d25d6ae0 100644

--- a/webrtc/call/call.cc

+++ b/webrtc/call/call.cc

@@ -33,6 +33,7 @@

#include "webrtc/call/bitrate_allocator.h"

#include "webrtc/call/call.h"

#include "webrtc/call/flexfec_receive_stream_impl.h"

+#include "webrtc/call/rtp_transport_controller.h"

#include "webrtc/config.h"

#include "webrtc/logging/rtc_event_log/rtc_event_log.h"

#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"

@@ -87,6 +88,45 @@ bool UseSendSideBwe(const FlexfecReceiveStream::Config& config) {

return UseSendSideBwe(config.rtp_header_extensions, config.transport_cc);

}

+class RtpTransportController : public RtpTransportControllerSendInterface {

+ public:

+ RtpTransportController(Clock* clock, webrtc::RtcEventLog* event_log);

+ void InitCongestionControl(CongestionController::Observer* observer);

+ VieRemb* remb() override { return &remb_; }

+ PacketRouter* packet_router() override { return &packet_router_; }

+ CongestionController* congestion_controller() override {

+ return congestion_controller_.get();

+ }

+ TransportFeedbackObserver* transport_feedback_observer() override {

+ return congestion_controller_.get();

+ }

+ RtpPacketSender* packet_sender() override {

+ return congestion_controller_->pacer();

+ }

+ private:

+ Clock* const clock_;

+ webrtc::RtcEventLog* const event_log_;

+ VieRemb remb_;

+ PacketRouter packet_router_;

+ // Construction delayed until InitCongestionControl, since the

+ // CongestionController wants its observer as a construction time

+ // argument, and setting it later seems no-trivial.

+ std::unique_ptr<CongestionController> congestion_controller_;

+};

+RtpTransportController::RtpTransportController(Clock* clock,

+ webrtc::RtcEventLog* event_log)

+ : clock_(clock), event_log_(event_log), remb_(clock) {}

+void RtpTransportController::InitCongestionControl(

+ CongestionController::Observer* observer) {

+ // Must be called only once.

+ RTC_CHECK(!congestion_controller_);

+ congestion_controller_.reset(new CongestionController(

+ clock_, observer, &remb_, event_log_, &packet_router_));

} // namespace

namespace internal {

@@ -97,7 +137,8 @@ class Call : public webrtc::Call,

public CongestionController::Observer,

public BitrateAllocator::LimitObserver {

public:

- explicit Call(const Call::Config& config);

+ Call(const Call::Config& config,

+ std::unique_ptr<RtpTransportController> transport);

virtual ~Call();

// Implements webrtc::Call.

@@ -271,11 +312,7 @@ class Call : public webrtc::Call,

std::map<std::string, rtc::NetworkRoute> network_routes_;

- VieRemb remb_;

- PacketRouter packet_router_;

- // TODO(nisse): Could be a direct member, except for constness

- // issues with GetRemoteBitrateEstimator (and maybe others).

- const std::unique_ptr<CongestionController> congestion_controller_;

+ std::unique_ptr<RtpTransportController> transport_;

const std::unique_ptr<SendDelayStats> video_send_delay_stats_;

const int64_t start_ms_;

// TODO(perkj): |worker_queue_| is supposed to replace

@@ -301,12 +338,16 @@ std::string Call::Stats::ToString(int64_t time_ms) const {

}

Call* Call::Create(const Call::Config& config) {

- return new internal::Call(config);

+ return new internal::Call(

+ config,

+ std::unique_ptr<RtpTransportController>(new RtpTransportController(

+ Clock::GetRealTimeClock(), config.event_log)));

}

namespace internal {

-Call::Call(const Call::Config& config)

+Call::Call(const Call::Config& config,

+ std::unique_ptr<RtpTransportController> transport)

: clock_(Clock::GetRealTimeClock()),

num_cpu_cores_(CpuInfo::DetectNumberOfCores()),

module_process_thread_(ProcessThread::Create("ModuleProcessThread")),

@@ -328,12 +369,7 @@ Call::Call(const Call::Config& config)

configured_max_padding_bitrate_bps_(0),

estimated_send_bitrate_kbps_counter_(clock_, nullptr, true),

pacer_bitrate_kbps_counter_(clock_, nullptr, true),

- remb_(clock_),

- congestion_controller_(new CongestionController(clock_,

- this,

- &remb_,

- event_log_,

- &packet_router_)),

+ transport_(std::move(transport)),

video_send_delay_stats_(new SendDelayStats(clock_)),

start_ms_(clock_->TimeInMilliseconds()),

worker_queue_("call_worker_queue") {

@@ -347,26 +383,28 @@ Call::Call(const Call::Config& config)

config.bitrate_config.start_bitrate_bps);

}

Trace::CreateTrace();

- call_stats_->RegisterStatsObserver(congestion_controller_.get());

- congestion_controller_->SignalNetworkState(kNetworkDown);

- congestion_controller_->SetBweBitrates(

+ transport_->InitCongestionControl(this);

+ transport_->congestion_controller()->SignalNetworkState(kNetworkDown);

+ transport_->congestion_controller()->SetBweBitrates(

config_.bitrate_config.min_bitrate_bps,

config_.bitrate_config.start_bitrate_bps,

config_.bitrate_config.max_bitrate_bps);

+ call_stats_->RegisterStatsObserver(transport_->congestion_controller());

module_process_thread_->Start();

module_process_thread_->RegisterModule(call_stats_.get(), RTC_FROM_HERE);

- module_process_thread_->RegisterModule(congestion_controller_.get(),

+ module_process_thread_->RegisterModule(transport_->congestion_controller(),

RTC_FROM_HERE);

- pacer_thread_->RegisterModule(congestion_controller_->pacer(), RTC_FROM_HERE);

+ pacer_thread_->RegisterModule(transport_->congestion_controller()->pacer(),

+ RTC_FROM_HERE);

pacer_thread_->RegisterModule(

- congestion_controller_->GetRemoteBitrateEstimator(true), RTC_FROM_HERE);

+ transport_->congestion_controller()->GetRemoteBitrateEstimator(true),

+ RTC_FROM_HERE);

pacer_thread_->Start();

}

Call::~Call() {

- RTC_DCHECK(!remb_.InUse());

+ RTC_DCHECK(!transport_->remb()->InUse());

RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());

RTC_CHECK(audio_send_ssrcs_.empty());

@@ -377,13 +415,13 @@ Call::~Call() {

RTC_CHECK(video_receive_streams_.empty());

pacer_thread_->Stop();

- pacer_thread_->DeRegisterModule(congestion_controller_->pacer());

+ pacer_thread_->DeRegisterModule(transport_->congestion_controller()->pacer());

pacer_thread_->DeRegisterModule(

- congestion_controller_->GetRemoteBitrateEstimator(true));

- module_process_thread_->DeRegisterModule(congestion_controller_.get());

+ transport_->congestion_controller()->GetRemoteBitrateEstimator(true));

+ module_process_thread_->DeRegisterModule(transport_->congestion_controller());

module_process_thread_->DeRegisterModule(call_stats_.get());

module_process_thread_->Stop();

- call_stats_->DeregisterStatsObserver(congestion_controller_.get());

+ call_stats_->DeregisterStatsObserver(transport_->congestion_controller());

// Only update histograms after process threads have been shut down, so that

// they won't try to concurrently update stats.

@@ -501,9 +539,8 @@ webrtc::AudioSendStream* Call::CreateAudioSendStream(

RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());

event_log_->LogAudioSendStreamConfig(config);

AudioSendStream* send_stream = new AudioSendStream(

- config, config_.audio_state, &worker_queue_, &packet_router_,

- congestion_controller_.get(), bitrate_allocator_.get(), event_log_,

- call_stats_->rtcp_rtt_stats());

+ config, config_.audio_state, &worker_queue_, transport_.get(),

+ bitrate_allocator_.get(), event_log_, call_stats_->rtcp_rtt_stats());

{

WriteLockScoped write_lock(*send_crit_);

RTC_DCHECK(audio_send_ssrcs_.find(config.rtp.ssrc) ==

@@ -556,8 +593,7 @@ webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream(

RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());

event_log_->LogAudioReceiveStreamConfig(config);

AudioReceiveStream* receive_stream = new AudioReceiveStream(

- &packet_router_, config,

- config_.audio_state, event_log_);

+ transport_->packet_router(), config, config_.audio_state, event_log_);

{

WriteLockScoped write_lock(*receive_crit_);

RTC_DCHECK(audio_receive_ssrcs_.find(config.rtp.remote_ssrc) ==

@@ -591,7 +627,8 @@ void Call::DestroyAudioReceiveStream(

WriteLockScoped write_lock(*receive_crit_);

const AudioReceiveStream::Config& config = audio_receive_stream->config();

uint32_t ssrc = config.rtp.remote_ssrc;

- congestion_controller_->GetRemoteBitrateEstimator(UseSendSideBwe(config))

+ transport_->congestion_controller()

+ ->GetRemoteBitrateEstimator(UseSendSideBwe(config))

->RemoveStream(ssrc);

size_t num_deleted = audio_receive_ssrcs_.erase(ssrc);

RTC_DCHECK(num_deleted == 1);

@@ -623,10 +660,9 @@ webrtc::VideoSendStream* Call::CreateVideoSendStream(

std::vector<uint32_t> ssrcs = config.rtp.ssrcs;

VideoSendStream* send_stream = new VideoSendStream(

num_cpu_cores_, module_process_thread_.get(), &worker_queue_,

- call_stats_.get(), congestion_controller_.get(), &packet_router_,

- bitrate_allocator_.get(), video_send_delay_stats_.get(), &remb_,

- event_log_, std::move(config), std::move(encoder_config),

- suspended_video_send_ssrcs_);

+ call_stats_.get(), transport_.get(), bitrate_allocator_.get(),

+ video_send_delay_stats_.get(), event_log_, std::move(config),

+ std::move(encoder_config), suspended_video_send_ssrcs_);

{

WriteLockScoped write_lock(*send_crit_);

@@ -683,8 +719,9 @@ webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream(

RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());

VideoReceiveStream* receive_stream = new VideoReceiveStream(

- num_cpu_cores_, &packet_router_, std::move(configuration),

- module_process_thread_.get(), call_stats_.get(), &remb_);

+ num_cpu_cores_, transport_->packet_router(),

+ std::move(configuration), module_process_thread_.get(), call_stats_.get(),

+ transport_->remb());

const webrtc::VideoReceiveStream::Config& config = receive_stream->config();

ReceiveRtpConfig receive_config(config.rtp.extensions,

@@ -740,7 +777,8 @@ void Call::DestroyVideoReceiveStream(

}

const VideoReceiveStream::Config& config = receive_stream_impl->config();

- congestion_controller_->GetRemoteBitrateEstimator(UseSendSideBwe(config))

+ transport_->congestion_controller()

+ ->GetRemoteBitrateEstimator(UseSendSideBwe(config))

->RemoveStream(config.rtp.remote_ssrc);

UpdateAggregateNetworkState();

@@ -816,7 +854,8 @@ void Call::DestroyFlexfecReceiveStream(FlexfecReceiveStream* receive_stream) {

++media_it;

}

- congestion_controller_->GetRemoteBitrateEstimator(UseSendSideBwe(config))

+ transport_->congestion_controller()

+ ->GetRemoteBitrateEstimator(UseSendSideBwe(config))

->RemoveStream(ssrc);

flexfec_receive_streams_.erase(receive_stream_impl);

@@ -832,15 +871,18 @@ Call::Stats Call::GetStats() const {

Stats stats;

// Fetch available send/receive bitrates.

uint32_t send_bandwidth = 0;

- congestion_controller_->GetBitrateController()->AvailableBandwidth(

- &send_bandwidth);

+ transport_->congestion_controller()

+ ->GetBitrateController()

+ ->AvailableBandwidth(&send_bandwidth);

std::vector<unsigned int> ssrcs;

uint32_t recv_bandwidth = 0;

- congestion_controller_->GetRemoteBitrateEstimator(false)->LatestEstimate(

- &ssrcs, &recv_bandwidth);

+ transport_->congestion_controller()

+ ->GetRemoteBitrateEstimator(false)

+ ->LatestEstimate(&ssrcs, &recv_bandwidth);

stats.send_bandwidth_bps = send_bandwidth;

stats.recv_bandwidth_bps = recv_bandwidth;

- stats.pacer_delay_ms = congestion_controller_->GetPacerQueuingDelayMs();

+ stats.pacer_delay_ms =

+ transport_->congestion_controller()->GetPacerQueuingDelayMs();

stats.rtt_ms = call_stats_->rtcp_rtt_stats()->LastProcessedRtt();

{

rtc::CritScope cs(&bitrate_crit_);

@@ -873,9 +915,9 @@ void Call::SetBitrateConfig(

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);

- congestion_controller_->SetBweBitrates(bitrate_config.min_bitrate_bps,

- bitrate_config.start_bitrate_bps,

- bitrate_config.max_bitrate_bps);

+ transport_->congestion_controller()->SetBweBitrates(

+ bitrate_config.min_bitrate_bps, bitrate_config.start_bitrate_bps,

+ bitrate_config.max_bitrate_bps);

}

void Call::SignalChannelNetworkState(MediaType media, NetworkState state) {

@@ -970,7 +1012,7 @@ void Call::OnNetworkRouteChanged(const std::string& transport_name,

<< " bps, max: " << config_.bitrate_config.start_bitrate_bps

<< " bps.";

RTC_DCHECK_GT(config_.bitrate_config.start_bitrate_bps, 0);

- congestion_controller_->ResetBweAndBitrates(

+ transport_->congestion_controller()->ResetBweAndBitrates(

config_.bitrate_config.start_bitrate_bps,

config_.bitrate_config.min_bitrate_bps,

config_.bitrate_config.max_bitrate_bps);

@@ -1006,7 +1048,7 @@ void Call::UpdateAggregateNetworkState() {

LOG(LS_INFO) << "UpdateAggregateNetworkState: aggregate_state="

<< (aggregate_state == kNetworkUp ? "up" : "down");

- congestion_controller_->SignalNetworkState(aggregate_state);

+ transport_->congestion_controller()->SignalNetworkState(aggregate_state);

}

void Call::OnSentPacket(const rtc::SentPacket& sent_packet) {

@@ -1014,7 +1056,7 @@ void Call::OnSentPacket(const rtc::SentPacket& sent_packet) {

first_packet_sent_ms_ = clock_->TimeInMilliseconds();

video_send_delay_stats_->OnSentPacket(sent_packet.packet_id,

clock_->TimeInMilliseconds());

- congestion_controller_->OnSentPacket(sent_packet);

+ transport_->congestion_controller()->OnSentPacket(sent_packet);

}

void Call::OnNetworkChanged(uint32_t target_bitrate_bps,

@@ -1065,7 +1107,7 @@ void Call::OnNetworkChanged(uint32_t target_bitrate_bps,

void Call::OnAllocationLimitsChanged(uint32_t min_send_bitrate_bps,

uint32_t max_padding_bitrate_bps) {

- congestion_controller_->SetAllocatedSendBitrateLimits(

+ transport_->congestion_controller()->SetAllocatedSendBitrateLimits(

min_send_bitrate_bps, max_padding_bitrate_bps);

rtc::CritScope lock(&bitrate_crit_);

min_allocated_send_bitrate_bps_ = min_send_bitrate_bps;

@@ -1278,11 +1320,12 @@ void Call::NotifyBweOfReceivedPacket(const RtpPacketReceived& packet,

// should be fixed to use the same MediaType as the production code.

if (media_type != MediaType::AUDIO ||

(use_send_side_bwe && header.extension.hasTransportSequenceNumber)) {

- congestion_controller_->OnReceivedPacket(

+ transport_->congestion_controller()->OnReceivedPacket(

packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(),

header);

}

} // namespace internal

} // namespace webrtc

« no previous file with comments | « webrtc/call/BUILD.gn ('k') | webrtc/call/rtp_transport_controller.h » ('j') | webrtc/call/rtp_transport_controller.h » ('J')