webrtc_m130/webrtc/call/congestion_controller.cc
Henrik Kjellander ff761fba82 modules: more interface -> include renames
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface

To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
                "use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)

BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc

R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1417683006 .

Cr-Commit-Position: refs/heads/master@{#10500}
2015-11-04 07:32:04 +00:00

305 lines
11 KiB
C++

/*
* Copyright (c) 2012 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 "webrtc/call/congestion_controller.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/common.h"
#include "webrtc/modules/pacing/include/paced_sender.h"
#include "webrtc/modules/pacing/include/packet_router.h"
#include "webrtc/modules/remote_bitrate_estimator/include/send_time_history.h"
#include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h"
#include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h"
#include "webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h"
#include "webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/utility/include/process_thread.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/logging.h"
#include "webrtc/video_engine/call_stats.h"
#include "webrtc/video_engine/payload_router.h"
#include "webrtc/video_engine/vie_encoder.h"
#include "webrtc/video_engine/vie_remb.h"
#include "webrtc/voice_engine/include/voe_video_sync.h"
namespace webrtc {
namespace {
static const uint32_t kTimeOffsetSwitchThreshold = 30;
class WrappingBitrateEstimator : public RemoteBitrateEstimator {
public:
WrappingBitrateEstimator(RemoteBitrateObserver* observer, Clock* clock)
: observer_(observer),
clock_(clock),
crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
rbe_(new RemoteBitrateEstimatorSingleStream(observer_, clock_)),
using_absolute_send_time_(false),
packets_since_absolute_send_time_(0),
min_bitrate_bps_(RemoteBitrateEstimator::kDefaultMinBitrateBps) {}
virtual ~WrappingBitrateEstimator() {}
void IncomingPacket(int64_t arrival_time_ms,
size_t payload_size,
const RTPHeader& header,
bool was_paced) override {
CriticalSectionScoped cs(crit_sect_.get());
PickEstimatorFromHeader(header);
rbe_->IncomingPacket(arrival_time_ms, payload_size, header, was_paced);
}
int32_t Process() override {
CriticalSectionScoped cs(crit_sect_.get());
return rbe_->Process();
}
int64_t TimeUntilNextProcess() override {
CriticalSectionScoped cs(crit_sect_.get());
return rbe_->TimeUntilNextProcess();
}
void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) override {
CriticalSectionScoped cs(crit_sect_.get());
rbe_->OnRttUpdate(avg_rtt_ms, max_rtt_ms);
}
void RemoveStream(unsigned int ssrc) override {
CriticalSectionScoped cs(crit_sect_.get());
rbe_->RemoveStream(ssrc);
}
bool LatestEstimate(std::vector<unsigned int>* ssrcs,
unsigned int* bitrate_bps) const override {
CriticalSectionScoped cs(crit_sect_.get());
return rbe_->LatestEstimate(ssrcs, bitrate_bps);
}
bool GetStats(ReceiveBandwidthEstimatorStats* output) const override {
CriticalSectionScoped cs(crit_sect_.get());
return rbe_->GetStats(output);
}
void SetMinBitrate(int min_bitrate_bps) {
CriticalSectionScoped cs(crit_sect_.get());
rbe_->SetMinBitrate(min_bitrate_bps);
min_bitrate_bps_ = min_bitrate_bps;
}
private:
void PickEstimatorFromHeader(const RTPHeader& header)
EXCLUSIVE_LOCKS_REQUIRED(crit_sect_.get()) {
if (header.extension.hasAbsoluteSendTime) {
// If we see AST in header, switch RBE strategy immediately.
if (!using_absolute_send_time_) {
LOG(LS_INFO) <<
"WrappingBitrateEstimator: Switching to absolute send time RBE.";
using_absolute_send_time_ = true;
PickEstimator();
}
packets_since_absolute_send_time_ = 0;
} else {
// When we don't see AST, wait for a few packets before going back to TOF.
if (using_absolute_send_time_) {
++packets_since_absolute_send_time_;
if (packets_since_absolute_send_time_ >= kTimeOffsetSwitchThreshold) {
LOG(LS_INFO) << "WrappingBitrateEstimator: Switching to transmission "
<< "time offset RBE.";
using_absolute_send_time_ = false;
PickEstimator();
}
}
}
}
// Instantiate RBE for Time Offset or Absolute Send Time extensions.
void PickEstimator() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_.get()) {
if (using_absolute_send_time_) {
rbe_.reset(new RemoteBitrateEstimatorAbsSendTime(observer_, clock_));
} else {
rbe_.reset(new RemoteBitrateEstimatorSingleStream(observer_, clock_));
}
rbe_->SetMinBitrate(min_bitrate_bps_);
}
RemoteBitrateObserver* observer_;
Clock* clock_;
rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
rtc::scoped_ptr<RemoteBitrateEstimator> rbe_;
bool using_absolute_send_time_;
uint32_t packets_since_absolute_send_time_;
int min_bitrate_bps_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WrappingBitrateEstimator);
};
} // namespace
CongestionController::CongestionController(ProcessThread* process_thread,
CallStats* call_stats)
: remb_(new VieRemb()),
bitrate_allocator_(new BitrateAllocator()),
packet_router_(new PacketRouter()),
pacer_(new PacedSender(Clock::GetRealTimeClock(),
packet_router_.get(),
BitrateController::kDefaultStartBitrateKbps,
PacedSender::kDefaultPaceMultiplier *
BitrateController::kDefaultStartBitrateKbps,
0)),
remote_bitrate_estimator_(
new WrappingBitrateEstimator(remb_.get(), Clock::GetRealTimeClock())),
remote_estimator_proxy_(
new RemoteEstimatorProxy(Clock::GetRealTimeClock(),
packet_router_.get())),
process_thread_(process_thread),
call_stats_(call_stats),
pacer_thread_(ProcessThread::Create("PacerThread")),
// Constructed last as this object calls the provided callback on
// construction.
bitrate_controller_(
BitrateController::CreateBitrateController(Clock::GetRealTimeClock(),
this)),
min_bitrate_bps_(RemoteBitrateEstimator::kDefaultMinBitrateBps) {
call_stats_->RegisterStatsObserver(remote_bitrate_estimator_.get());
pacer_thread_->RegisterModule(pacer_.get());
pacer_thread_->Start();
process_thread->RegisterModule(remote_estimator_proxy_.get());
process_thread->RegisterModule(remote_bitrate_estimator_.get());
process_thread->RegisterModule(bitrate_controller_.get());
}
CongestionController::~CongestionController() {
pacer_thread_->Stop();
pacer_thread_->DeRegisterModule(pacer_.get());
process_thread_->DeRegisterModule(bitrate_controller_.get());
process_thread_->DeRegisterModule(remote_bitrate_estimator_.get());
process_thread_->DeRegisterModule(remote_estimator_proxy_.get());
call_stats_->DeregisterStatsObserver(remote_bitrate_estimator_.get());
if (transport_feedback_adapter_.get())
call_stats_->DeregisterStatsObserver(transport_feedback_adapter_.get());
RTC_DCHECK(!remb_->InUse());
RTC_DCHECK(encoders_.empty());
}
void CongestionController::AddEncoder(ViEEncoder* encoder) {
rtc::CritScope lock(&encoder_crit_);
encoders_.push_back(encoder);
}
void CongestionController::RemoveEncoder(ViEEncoder* encoder) {
rtc::CritScope lock(&encoder_crit_);
for (auto it = encoders_.begin(); it != encoders_.end(); ++it) {
if (*it == encoder) {
encoders_.erase(it);
return;
}
}
}
void CongestionController::SetBweBitrates(int min_bitrate_bps,
int start_bitrate_bps,
int max_bitrate_bps) {
if (start_bitrate_bps > 0)
bitrate_controller_->SetStartBitrate(start_bitrate_bps);
bitrate_controller_->SetMinMaxBitrate(min_bitrate_bps, max_bitrate_bps);
if (remote_bitrate_estimator_.get())
remote_bitrate_estimator_->SetMinBitrate(min_bitrate_bps);
if (transport_feedback_adapter_.get())
transport_feedback_adapter_->GetBitrateEstimator()->SetMinBitrate(
min_bitrate_bps);
min_bitrate_bps_ = min_bitrate_bps;
}
BitrateController* CongestionController::GetBitrateController() const {
return bitrate_controller_.get();
}
RemoteBitrateEstimator* CongestionController::GetRemoteBitrateEstimator(
bool send_side_bwe) const {
if (send_side_bwe)
return remote_estimator_proxy_.get();
else
return remote_bitrate_estimator_.get();
}
TransportFeedbackObserver*
CongestionController::GetTransportFeedbackObserver() {
if (transport_feedback_adapter_.get() == nullptr) {
transport_feedback_adapter_.reset(new TransportFeedbackAdapter(
bitrate_controller_->CreateRtcpBandwidthObserver(),
Clock::GetRealTimeClock(), process_thread_));
transport_feedback_adapter_->SetBitrateEstimator(
new RemoteBitrateEstimatorAbsSendTime(
transport_feedback_adapter_.get(), Clock::GetRealTimeClock()));
transport_feedback_adapter_->GetBitrateEstimator()->SetMinBitrate(
min_bitrate_bps_);
call_stats_->RegisterStatsObserver(transport_feedback_adapter_.get());
}
return transport_feedback_adapter_.get();
}
int64_t CongestionController::GetPacerQueuingDelayMs() const {
return pacer_->QueueInMs();
}
// TODO(mflodman): Move out of this class.
void CongestionController::SetChannelRembStatus(bool sender,
bool receiver,
RtpRtcp* rtp_module) {
rtp_module->SetREMBStatus(sender || receiver);
if (sender) {
remb_->AddRembSender(rtp_module);
} else {
remb_->RemoveRembSender(rtp_module);
}
if (receiver) {
remb_->AddReceiveChannel(rtp_module);
} else {
remb_->RemoveReceiveChannel(rtp_module);
}
}
void CongestionController::SignalNetworkState(NetworkState state) {
if (state == kNetworkUp) {
pacer_->Resume();
} else {
pacer_->Pause();
}
}
// TODO(mflodman): Move this logic out from CongestionController.
void CongestionController::OnNetworkChanged(uint32_t target_bitrate_bps,
uint8_t fraction_loss,
int64_t rtt) {
bitrate_allocator_->OnNetworkChanged(target_bitrate_bps, fraction_loss, rtt);
int pad_up_to_bitrate_bps = 0;
{
rtc::CritScope lock(&encoder_crit_);
for (const auto& encoder : encoders_)
pad_up_to_bitrate_bps += encoder->GetPaddingNeededBps();
}
pacer_->UpdateBitrate(
target_bitrate_bps / 1000,
PacedSender::kDefaultPaceMultiplier * target_bitrate_bps / 1000,
pad_up_to_bitrate_bps / 1000);
}
void CongestionController::OnSentPacket(const rtc::SentPacket& sent_packet) {
if (transport_feedback_adapter_) {
transport_feedback_adapter_->OnSentPacket(sent_packet.packet_id,
sent_packet.send_time_ms);
}
}
} // namespace webrtc