Wire up send-side bandwidth estimation.

webrtc/video_engine/vie_channel_group.cc

 /*
  *  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/video_engine/vie_channel_group.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/interface/rtp_rtcp.h"
 #include "webrtc/modules/utility/interface/process_thread.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/interface/logging.h"
 #include "webrtc/video_engine/call_stats.h"
 #include "webrtc/video_engine/encoder_state_feedback.h"
 #include "webrtc/video_engine/payload_router.h"
 #include "webrtc/video_engine/vie_channel.h"
 #include "webrtc/video_engine/vie_encoder.h"
@@ skipping 122 matching lines @@
                              PacedSender::kDefaultPaceMultiplier *
                                  BitrateController::kDefaultStartBitrateKbps,
                              0)),
       process_thread_(process_thread),
       pacer_thread_(ProcessThread::Create("PacerThread")),
       // Constructed last as this object calls the provided callback on
       // construction.
       bitrate_controller_(
           BitrateController::CreateBitrateController(Clock::GetRealTimeClock(),
                                                      this)) {
-  remote_bitrate_estimator_.reset(new WrappingBitrateEstimator(
-      remb_.get(), Clock::GetRealTimeClock()));
+  Clock* clock = Clock::GetRealTimeClock();
+  remote_bitrate_estimator_.reset(
+      new WrappingBitrateEstimator(remb_.get(), clock));
+  remote_estimator_proxy_.reset(
+      new RemoteEstimatorProxy(clock, packet_router_.get()));

[stefan-webrtc, 2015/09/18 10:57:23]

Can't these be instantiated in the initializer list?

[sprang_webrtc, 2015/09/21 10:44:10]

Done.

 
   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(call_stats_.get());
   process_thread->RegisterModule(bitrate_controller_.get());
 }
 
 ChannelGroup::~ChannelGroup() {
   pacer_thread_->Stop();
   pacer_thread_->DeRegisterModule(pacer_.get());
   process_thread_->DeRegisterModule(bitrate_controller_.get());
   process_thread_->DeRegisterModule(call_stats_.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(channel_map_.empty());
   RTC_DCHECK(!remb_->InUse());
   RTC_DCHECK(vie_encoder_map_.empty());
 }
 
 bool ChannelGroup::CreateSendChannel(int channel_id,
                                      int engine_id,
                                      Transport* transport,
                                      int number_of_cores,
-                                     const std::vector<uint32_t>& ssrcs) {
+                                     const VideoSendStream::Config& config) {
+  TransportFeedbackObserver* transport_feedback_observer = nullptr;
+  bool transport_seq_enabled = false;
+  for (const RtpExtension& extension : config.rtp.extensions) {
+    if (extension.name == RtpExtension::kTransportSequenceNumber) {
+      transport_seq_enabled = true;
+      break;
+    }
+  }
+  if (transport_seq_enabled) {
+    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(),
+              RemoteBitrateEstimator::kDefaultMinBitrateBps));
+      call_stats_->RegisterStatsObserver(transport_feedback_adapter_.get());
+    }
+    transport_feedback_observer = transport_feedback_adapter_.get();
+  }
+
+  const std::vector<uint32_t>& ssrcs = config.rtp.ssrcs;
   RTC_DCHECK(!ssrcs.empty());
   rtc::scoped_ptr<ViEEncoder> vie_encoder(
       new ViEEncoder(channel_id, number_of_cores, *process_thread_,
                      pacer_.get(), bitrate_allocator_.get()));
   if (!vie_encoder->Init()) {
     return false;
   }
   ViEEncoder* encoder = vie_encoder.get();
   if (!CreateChannel(channel_id, engine_id, transport, number_of_cores,
-                     vie_encoder.release(), ssrcs.size(), true)) {
+                     vie_encoder.release(), ssrcs.size(), true,
+                     remote_bitrate_estimator_.get(),
+                     transport_feedback_observer)) {
     return false;
   }
   ViEChannel* channel = channel_map_[channel_id];
   // Connect the encoder with the send packet router, to enable sending.
   encoder->StartThreadsAndSetSharedMembers(channel->send_payload_router(),
                                            channel->vcm_protection_callback());
 
   encoder_state_feedback_->AddEncoder(ssrcs, encoder);
   std::vector<uint32_t> first_ssrc(1, ssrcs[0]);
   encoder->SetSsrcs(first_ssrc);
   return true;
 }
 
-bool ChannelGroup::CreateReceiveChannel(int channel_id,
-                                        int engine_id,
-                                        Transport* transport,
-                                        int number_of_cores) {
+bool ChannelGroup::CreateReceiveChannel(
+    int channel_id,
+    int engine_id,
+    Transport* transport,
+    int number_of_cores,
+    const VideoReceiveStream::Config& config) {
+  bool send_side_bwe = false;
+  for (const RtpExtension& extension : config.rtp.extensions) {
+    if (extension.name == RtpExtension::kTransportSequenceNumber) {
+      send_side_bwe = true;
+      break;
+    }
+  }

[stefan-webrtc, 2015/09/18 10:57:23]

Maybe it would be good to DCHECK that we don't run some receive channels with
send_side_bwe = false and others with true?

[sprang_webrtc, 2015/09/21 10:44:10]

Right now a mix is supported. So the not very explicitly stated requirement is
that all (video) channels must be configured the same way wrt send-side bwe?

[stefan-webrtc, 2015/09/21 11:12:07]

Well, at least we could make that a requirement if we'd like to. It sounds like
asking for problems if we try to do both at the same time. On the other hand,
maybe there's a risk of getting a race when switching and therefore not worth
DCHECKing?

[sprang_webrtc, 2015/09/21 13:42:53]

Is suspect there might. It's also gonna be kinda messy to verify. Do we allow
mixing send-side bwe on the receive streams but not on the send streams?

[stefan-webrtc, 2015/09/21 14:06:30]

Yes, that may be so. Let's leave it as is for the moment at least. :)

+
+  RemoteBitrateEstimator* bitrate_estimator;
+  if (send_side_bwe) {
+    bitrate_estimator = remote_estimator_proxy_.get();
+  } else {
+    bitrate_estimator = remote_bitrate_estimator_.get();
+  }
   return CreateChannel(channel_id, engine_id, transport, number_of_cores,
-                       nullptr, 1, false);
+                       nullptr, 1, false, bitrate_estimator, nullptr);
 }
 
 bool ChannelGroup::CreateChannel(int channel_id,
                                  int engine_id,
                                  Transport* transport,
                                  int number_of_cores,
                                  ViEEncoder* vie_encoder,
                                  size_t max_rtp_streams,
-                                 bool sender) {
+                                 bool sender,
+                                 RemoteBitrateEstimator* bitrate_estimator,
+                                 TransportFeedbackObserver* feedback_observer) {
   rtc::scoped_ptr<ViEChannel> channel(new ViEChannel(
       channel_id, engine_id, number_of_cores, transport, process_thread_,
       encoder_state_feedback_->GetRtcpIntraFrameObserver(),
-      bitrate_controller_->CreateRtcpBandwidthObserver(), nullptr,
-      remote_bitrate_estimator_.get(), call_stats_->rtcp_rtt_stats(),
-      pacer_.get(), packet_router_.get(), max_rtp_streams, sender));
+      bitrate_controller_->CreateRtcpBandwidthObserver(), feedback_observer,
+      bitrate_estimator, call_stats_->rtcp_rtt_stats(), pacer_.get(),
+      packet_router_.get(), max_rtp_streams, sender));
   if (channel->Init() != 0) {
     return false;
   }
 
   // Register the channel to receive stats updates.
   call_stats_->RegisterStatsObserver(channel->GetStatsObserver());
 
   // Store the channel, add it to the channel group and save the vie_encoder.
   channel_map_[channel_id] = channel.release();
   if (vie_encoder) {
@@ skipping 116 matching lines @@
     rtc::CritScope lock(&encoder_map_crit_);
     for (const auto& encoder : vie_encoder_map_)
       pad_up_to_bitrate_bps += encoder.second->GetPaddingNeededBps();
   }
   pacer_->UpdateBitrate(
       target_bitrate_bps / 1000,
       PacedSender::kDefaultPaceMultiplier * target_bitrate_bps / 1000,
       pad_up_to_bitrate_bps / 1000);
 }
 }  // namespace webrtc