webrtc_m130/webrtc/call/congestion_controller.cc
Peter Boström 7b971e728b Remove extra_options from VideoCodec.
Constructing default options is racy when initializing multiple VP8
encoders in parallel. This is only used for VP8 temporal layers. Adding
TemporalLayerFactory to VP8 codec specifics instead of generic options.

Removes the last webrtc::Config uses/includes from video code.

Also removes VideoCodec equality operators which are no longer in use.

BUG=webrtc:5410
R=stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1606613003 .

Cr-Commit-Position: refs/heads/master@{#11307}
2016-01-19 15:26:24 +00:00

294 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/logging.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
#include "webrtc/modules/pacing/paced_sender.h"
#include "webrtc/modules/pacing/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/video/call_stats.h"
#include "webrtc/video/payload_router.h"
#include "webrtc/video/vie_encoder.h"
#include "webrtc/video/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,
BitrateObserver* bitrate_observer)
: remb_(new VieRemb(Clock::GetRealTimeClock())),
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(),
bitrate_observer)),
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();
}
void CongestionController::UpdatePacerBitrate(int bitrate_kbps,
int max_bitrate_kbps,
int min_bitrate_kbps) {
pacer_->UpdateBitrate(bitrate_kbps, max_bitrate_kbps, min_bitrate_kbps);
}
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();
}
}
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