[Cast] RTT clean-up to the max!

media/cast/sender/frame_sender.cc

 // Copyright 2014 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "media/cast/sender/frame_sender.h"
 
 #include "base/debug/trace_event.h"
 
 namespace media {
 namespace cast {
 namespace {
 
 const int kMinSchedulingDelayMs = 1;
 const int kNumAggressiveReportsSentAtStart = 100;
 
 }  // namespace
 
 FrameSender::FrameSender(scoped_refptr<CastEnvironment> cast_environment,
                          bool is_audio,
                          CastTransportSender* const transport_sender,
                          base::TimeDelta rtcp_interval,
                          int rtp_timebase,
                          uint32 ssrc,
                          double max_frame_rate,
                          base::TimeDelta playout_delay,
                          CongestionControl* congestion_control)
     : cast_environment_(cast_environment),
       transport_sender_(transport_sender),
       ssrc_(ssrc),
-      rtt_available_(false),
       rtcp_interval_(rtcp_interval),
       max_frame_rate_(max_frame_rate),
       frames_in_encoder_(0),
       num_aggressive_rtcp_reports_sent_(0),
       last_sent_frame_id_(0),
       latest_acked_frame_id_(0),
       duplicate_ack_counter_(0),
       rtp_timebase_(rtp_timebase),
       congestion_control_(congestion_control),
       is_audio_(is_audio),
       weak_factory_(this) {
+  DCHECK(transport_sender_);
   DCHECK_GT(rtp_timebase_, 0);
+  DCHECK(congestion_control_);
   SetTargetPlayoutDelay(playout_delay);
   send_target_playout_delay_ = false;
   memset(frame_rtp_timestamps_, 0, sizeof(frame_rtp_timestamps_));
 }
 
 FrameSender::~FrameSender() {
 }
 
 void FrameSender::ScheduleNextRtcpReport() {
   DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
@@ skipping 28 matching lines @@
   const int64 rtp_delta = TimeDeltaToRtpDelta(time_delta, rtp_timebase_);
   const uint32 now_as_rtp_timestamp =
       GetRecordedRtpTimestamp(last_sent_frame_id_) +
           static_cast<uint32>(rtp_delta);
   transport_sender_->SendSenderReport(ssrc_, now, now_as_rtp_timestamp);
 
   if (schedule_future_reports)
     ScheduleNextRtcpReport();
 }
 
-void FrameSender::OnReceivedRtt(base::TimeDelta rtt,
-                                base::TimeDelta avg_rtt,
-                                base::TimeDelta min_rtt,
-                                base::TimeDelta max_rtt) {
-  rtt_available_ = true;
-  rtt_ = rtt;
-  avg_rtt_ = avg_rtt;
-  min_rtt_ = min_rtt;
-  max_rtt_ = max_rtt;
+void FrameSender::OnMeasuredRoundTripTime(base::TimeDelta rtt) {
+  DCHECK(rtt > base::TimeDelta());
+  current_round_trip_time_ = rtt;
 }
 
 void FrameSender::SetTargetPlayoutDelay(
     base::TimeDelta new_target_playout_delay) {
   target_playout_delay_ = new_target_playout_delay;
   max_unacked_frames_ =
       std::min(kMaxUnackedFrames,
                1 + static_cast<int>(target_playout_delay_ *
                                     max_frame_rate_ /
                                     base::TimeDelta::FromSeconds(1)));
@@ skipping 124 matching lines @@
   if (send_target_playout_delay_) {
     encoded_frame->new_playout_delay_ms =
         target_playout_delay_.InMilliseconds();
   }
   transport_sender_->InsertFrame(ssrc_, *encoded_frame);
 }
 
 void FrameSender::OnReceivedCastFeedback(const RtcpCastMessage& cast_feedback) {
   DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
 
-  base::TimeDelta rtt;
-  base::TimeDelta avg_rtt;
-  base::TimeDelta min_rtt;
-  base::TimeDelta max_rtt;
-  if (is_rtt_available()) {
-    rtt = rtt_;
-    avg_rtt = avg_rtt_;
-    min_rtt = min_rtt_;
-    max_rtt = max_rtt_;
+  const bool have_valid_rtt = current_round_trip_time_ > base::TimeDelta();
+  if (have_valid_rtt) {
+    congestion_control_->UpdateRtt(current_round_trip_time_);
 
-    congestion_control_->UpdateRtt(rtt);
-
-    // Don't use a RTT lower than our average.
-    rtt = std::max(rtt, avg_rtt);
-
-    // Having the RTT values implies the receiver sent back a receiver report
+    // Having the RTT value implies the receiver sent back a receiver report
     // based on it having received a report from here.  Therefore, ensure this
     // sender stops aggressively sending reports.
     if (num_aggressive_rtcp_reports_sent_ < kNumAggressiveReportsSentAtStart) {
       VLOG(1) << "No longer a need to send reports aggressively (sent "
               << num_aggressive_rtcp_reports_sent_ << ").";
       num_aggressive_rtcp_reports_sent_ = kNumAggressiveReportsSentAtStart;
       ScheduleNextRtcpReport();
     }
-  } else {
-    // We have no measured value use default.
-    rtt = base::TimeDelta::FromMilliseconds(kStartRttMs);
   }
 
   if (last_send_time_.is_null())
     return;  // Cannot get an ACK without having first sent a frame.
 
   if (cast_feedback.missing_frames_and_packets.empty()) {
     OnAck(cast_feedback.ack_frame_id);
 
     // We only count duplicate ACKs when we have sent newer frames.
     if (latest_acked_frame_id_ == cast_feedback.ack_frame_id &&
@@ skipping 62 matching lines @@
           << duration_in_flight.InMicroseconds() << " usec ("
           << (target_playout_delay_ > base::TimeDelta() ?
                   100 * duration_in_flight / target_playout_delay_ :
                   kint64max) << "%)";
   return frames_in_flight >= max_unacked_frames_ ||
       duration_in_flight >= target_playout_delay_;
 }
 
 }  // namespace cast
 }  // namespace media