Remove overhead from video bitrate.

webrtc/video/video_send_stream.cc

 /*
  *  Copyright (c) 2013 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/video_send_stream.h"
@@ skipping 12 matching lines @@
 #include "webrtc/base/weak_ptr.h"
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
 #include "webrtc/modules/congestion_controller/include/congestion_controller.h"
 #include "webrtc/modules/pacing/packet_router.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/modules/video_coding/utility/ivf_file_writer.h"
 #include "webrtc/video/call_stats.h"
 #include "webrtc/video/vie_remb.h"
 #include "webrtc/video_send_stream.h"
+#include "webrtc/system_wrappers/include/field_trial.h"
 
 namespace webrtc {
 
 static const int kMinSendSidePacketHistorySize = 600;
 namespace {
 
 std::vector<RtpRtcp*> CreateRtpRtcpModules(
     Transport* outgoing_transport,
     RtcpIntraFrameObserver* intra_frame_callback,
     RtcpBandwidthObserver* bandwidth_callback,
     TransportFeedbackObserver* transport_feedback_callback,
     RtcpRttStats* rtt_stats,
     RtpPacketSender* paced_sender,
     TransportSequenceNumberAllocator* transport_sequence_number_allocator,
     FlexfecSender* flexfec_sender,
     SendStatisticsProxy* stats_proxy,
     SendDelayStats* send_delay_stats,
     RtcEventLog* event_log,
     RateLimiter* retransmission_rate_limiter,
+    OverheadObserver* overhead_observer,
     size_t num_modules) {
   RTC_DCHECK_GT(num_modules, 0u);
   RtpRtcp::Configuration configuration;
   ReceiveStatistics* null_receive_statistics = configuration.receive_statistics;
   configuration.audio = false;
   configuration.receiver_only = false;
   configuration.flexfec_sender = flexfec_sender;
   configuration.receive_statistics = null_receive_statistics;
   configuration.outgoing_transport = outgoing_transport;
   configuration.intra_frame_callback = intra_frame_callback;
   configuration.bandwidth_callback = bandwidth_callback;
   configuration.transport_feedback_callback = transport_feedback_callback;
   configuration.rtt_stats = rtt_stats;
   configuration.rtcp_packet_type_counter_observer = stats_proxy;
   configuration.paced_sender = paced_sender;
   configuration.transport_sequence_number_allocator =
       transport_sequence_number_allocator;
   configuration.send_bitrate_observer = stats_proxy;
   configuration.send_frame_count_observer = stats_proxy;
   configuration.send_side_delay_observer = stats_proxy;
   configuration.send_packet_observer = send_delay_stats;
   configuration.event_log = event_log;
   configuration.retransmission_rate_limiter = retransmission_rate_limiter;
-
+  configuration.overhead_observer = overhead_observer;
   std::vector<RtpRtcp*> modules;
   for (size_t i = 0; i < num_modules; ++i) {
     RtpRtcp* rtp_rtcp = RtpRtcp::CreateRtpRtcp(configuration);
     rtp_rtcp->SetSendingStatus(false);
     rtp_rtcp->SetSendingMediaStatus(false);
     rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
     modules.push_back(rtp_rtcp);
   }
   return modules;
 }
@@ skipping 191 matching lines @@
 namespace internal {
 
 // VideoSendStreamImpl implements internal::VideoSendStream.
 // It is created and destroyed on |worker_queue|. The intent is to decrease the
 // need for locking and to ensure methods are called in sequence.
 // Public methods except |DeliverRtcp| must be called on |worker_queue|.
 // DeliverRtcp is called on the libjingle worker thread or a network thread.
 // An encoder may deliver frames through the EncodedImageCallback on an
 // arbitrary thread.
 class VideoSendStreamImpl : public webrtc::BitrateAllocatorObserver,
+                            public webrtc::OverheadObserver,
                             public webrtc::VCMProtectionCallback,
                             public ViEEncoder::EncoderSink {
  public:
   VideoSendStreamImpl(SendStatisticsProxy* stats_proxy,
                       rtc::TaskQueue* worker_queue,
                       CallStats* call_stats,
                       CongestionController* congestion_controller,
                       BitrateAllocator* bitrate_allocator,
                       SendDelayStats* send_delay_stats,
                       VieRemb* remb,
@@ skipping 33 matching lines @@
                             uint8_t fraction_loss,
                             int64_t rtt) override;
 
   // Implements webrtc::VCMProtectionCallback.
   int ProtectionRequest(const FecProtectionParams* delta_params,
                         const FecProtectionParams* key_params,
                         uint32_t* sent_video_rate_bps,
                         uint32_t* sent_nack_rate_bps,
                         uint32_t* sent_fec_rate_bps) override;
 
+  // Implements OverheadObserver.
+  void OnOverheadChanged(size_t overhead_bytes_per_packet) override;
+
   void OnEncoderConfigurationChanged(std::vector<VideoStream> streams,
                                      int min_transmit_bitrate_bps) override;
 
   // Implements EncodedImageCallback. The implementation routes encoded frames
   // to the |payload_router_| and |config.pre_encode_callback| if set.
   // Called on an arbitrary encoder callback thread.
   EncodedImageCallback::Result OnEncodedImage(
       const EncodedImage& encoded_image,
       const CodecSpecificInfo* codec_specific_info,
       const RTPFragmentationHeader* fragmentation) override;
@@ skipping 41 matching lines @@
   const std::vector<RtpRtcp*> rtp_rtcp_modules_;
   PayloadRouter payload_router_;
 
   // |weak_ptr_| to our self. This is used since we can not call
   // |weak_ptr_factory_.GetWeakPtr| from multiple sequences but it is ok to copy
   // an existing WeakPtr.
   rtc::WeakPtr<VideoSendStreamImpl> weak_ptr_;
   // |weak_ptr_factory_| must be declared last to make sure all WeakPtr's are
   // invalidated before any other members are destroyed.
   rtc::WeakPtrFactory<VideoSendStreamImpl> weak_ptr_factory_;
+
+  rtc::CriticalSection overhead_per_packet_crit_;
+  size_t overhead_per_packet_ GUARDED_BY(overhead_per_packet_crit_);

[minyue-webrtc, 2016/11/24 10:09:49]

I think better to make the unit clearer:  _bytes_per_packet

+  size_t transport_overhead_per_packet_;
 };
 
 // TODO(tommi): See if there's a more elegant way to create a task that creates
 // an object on the correct task queue.
 class VideoSendStream::ConstructionTask : public rtc::QueuedTask {
  public:
   ConstructionTask(std::unique_ptr<VideoSendStreamImpl>* send_stream,
                    rtc::Event* done_event,
                    SendStatisticsProxy* stats_proxy,
                    ViEEncoder* vie_encoder,
@@ skipping 316 matching lines @@
           bandwidth_observer_.get(),
           congestion_controller_->GetTransportFeedbackObserver(),
           call_stats_->rtcp_rtt_stats(),
           congestion_controller_->pacer(),
           congestion_controller_->packet_router(),
           flexfec_sender_.get(),
           stats_proxy_,
           send_delay_stats,
           event_log,
           congestion_controller_->GetRetransmissionRateLimiter(),
+          this,
           config_->rtp.ssrcs.size())),
       payload_router_(rtp_rtcp_modules_,
                       config_->encoder_settings.payload_type),
-      weak_ptr_factory_(this) {
+      weak_ptr_factory_(this),
+      overhead_per_packet_(0) {
   RTC_DCHECK_RUN_ON(worker_queue_);
   LOG(LS_INFO) << "VideoSendStreamInternal: " << config_->ToString();
   weak_ptr_ = weak_ptr_factory_.GetWeakPtr();
   module_process_thread_checker_.DetachFromThread();
 
   RTC_DCHECK(!config_->rtp.ssrcs.empty());
   RTC_DCHECK(call_stats_);
   RTC_DCHECK(congestion_controller_);
   RTC_DCHECK(remb_);
 
@@ skipping 16 matching lines @@
   }
 
   remb_->AddRembSender(rtp_rtcp_modules_[0]);
   rtp_rtcp_modules_[0]->SetREMBStatus(true);
 
   ConfigureProtection();
   ConfigureSsrcs();
 
   // TODO(pbos): Should we set CNAME on all RTP modules?
   rtp_rtcp_modules_.front()->SetCNAME(config_->rtp.c_name.c_str());
-  // 28 to match packet overhead in ModuleRtpRtcpImpl.
-  static const size_t kRtpPacketSizeOverhead = 28;
-  RTC_DCHECK_LE(config_->rtp.max_packet_size, 0xFFFFu + kRtpPacketSizeOverhead);
-  const uint16_t mtu = static_cast<uint16_t>(config_->rtp.max_packet_size +
-                                             kRtpPacketSizeOverhead);
+
   for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
     rtp_rtcp->RegisterRtcpStatisticsCallback(stats_proxy_);
     rtp_rtcp->RegisterSendChannelRtpStatisticsCallback(stats_proxy_);
-    rtp_rtcp->SetMaxTransferUnit(mtu);
+    rtp_rtcp->SetMaxTransferUnit(config_->rtp.max_packet_size);
     rtp_rtcp->RegisterVideoSendPayload(
         config_->encoder_settings.payload_type,
         config_->encoder_settings.payload_name.c_str());
   }
 
   RTC_DCHECK(config_->encoder_settings.encoder);
   RTC_DCHECK_GE(config_->encoder_settings.payload_type, 0);
   RTC_DCHECK_LE(config_->encoder_settings.payload_type, 127);
 
   vie_encoder_->SetStartBitrate(bitrate_allocator_->GetStartBitrate(this));
@@ skipping 349 matching lines @@
                                                 : RtcpMode::kOff);
   }
 }
 
 uint32_t VideoSendStreamImpl::OnBitrateUpdated(uint32_t bitrate_bps,
                                                uint8_t fraction_loss,
                                                int64_t rtt) {
   RTC_DCHECK_RUN_ON(worker_queue_);
   RTC_DCHECK(payload_router_.active())
       << "VideoSendStream::Start has not been called.";
+
+  if (webrtc::field_trial::FindFullName("WebRTC-SendSideBwe-WithOverhead") ==
+      "Enabled") {
+    // Substract overhead from bitrate.
+    rtc::CritScope lock(&overhead_per_packet_crit_);
+    int packets_per_second = std::ceil(
+        bitrate_bps /
+        (8 * (config_->rtp.max_packet_size + transport_overhead_per_packet_)));
+    int overhead_bps = 8 * overhead_per_packet_ * packets_per_second;
+    bitrate_bps -= overhead_bps;

[minyue-webrtc, 2016/11/24 10:09:49]

clamp by zero, can become really large otherwise

[michaelt, 2016/11/24 14:38:53]

Done.

+  }
+
   // Get the encoder target rate. It is the estimated network rate -
   // protection overhead.
   encoder_target_rate_bps_ = protection_bitrate_calculator_.SetTargetRates(
       bitrate_bps, stats_proxy_->GetSendFrameRate(), fraction_loss, rtt);
   uint32_t protection_bitrate = bitrate_bps - encoder_target_rate_bps_;
 
   encoder_target_rate_bps_ =
       std::min(encoder_max_bitrate_bps_, encoder_target_rate_bps_);
   vie_encoder_->OnBitrateUpdated(encoder_target_rate_bps_, fraction_loss, rtt);
   stats_proxy_->OnSetEncoderTargetRate(encoder_target_rate_bps_);
@@ skipping 39 matching lines @@
     rtp_rtcp->SetFecParameters(*delta_params, *key_params);
     rtp_rtcp->BitrateSent(&not_used, &module_video_rate, &module_fec_rate,
                           &module_nack_rate);
     *sent_video_rate_bps += module_video_rate;
     *sent_nack_rate_bps += module_nack_rate;
     *sent_fec_rate_bps += module_fec_rate;
   }
   return 0;
 }
 
+void VideoSendStreamImpl::OnOverheadChanged(size_t overhead_bytes_per_packet) {
+  rtc::CritScope lock(&overhead_per_packet_crit_);
+  overhead_per_packet_ = overhead_bytes_per_packet;
+}
+
 void VideoSendStreamImpl::SetTransportOverhead(
     int transport_overhead_per_packet) {

[minyue-webrtc, 2016/11/24 10:09:49]

I think better to make the unit clearer:  _bytes_per_packet

[michaelt, 2016/11/24 14:38:53]

Done.

-  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
+  transport_overhead_per_packet_ = transport_overhead_per_packet;
+  RTC_DCHECK_LE(config_->rtp.max_packet_size,
+                0xFFFFu + transport_overhead_per_packet);
+  const uint16_t mtu = static_cast<uint16_t>(config_->rtp.max_packet_size +
+                                             transport_overhead_per_packet);
+  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
     rtp_rtcp->SetTransportOverhead(transport_overhead_per_packet);
+    rtp_rtcp->SetMaxTransferUnit(mtu);
+  }
 }
 
 }  // namespace internal
 }  // namespace webrtc