[Cast] Limit frames in flight by duration, and not by number of frames.

media/cast/sender/frame_sender.cc

Index: media/cast/sender/frame_sender.cc
diff --git a/media/cast/sender/frame_sender.cc b/media/cast/sender/frame_sender.cc
index 300e8146ed8e0157b875a4edf9c0c869b4710d9c..946658e14fba5b339ab9ee14eb80839bca5581e7 100644
--- a/media/cast/sender/frame_sender.cc
+++ b/media/cast/sender/frame_sender.cc
@@ -4,6 +4,8 @@
 
 #include "media/cast/sender/frame_sender.h"
 
+#include <algorithm>
+
 #include "base/debug/trace_event.h"
 
 namespace media {
@@ -13,6 +15,11 @@ namespace {
 const int kMinSchedulingDelayMs = 1;
 const int kNumAggressiveReportsSentAtStart = 100;
 
+// These control the amount and granularity of history in the "max round trip
+// time" estimation logic.  40 buckets * 15000 millis --> 10 minutes total
+const int kNumRttBuckets = 40;
+const int kRttBucketRotationMs = 15000;
+
 }  // namespace
 
 FrameSender::FrameSender(scoped_refptr<CastEnvironment> cast_environment,
@@ -33,11 +40,12 @@ FrameSender::FrameSender(scoped_refptr<CastEnvironment> cast_environment,
       last_sent_frame_id_(0),
       latest_acked_frame_id_(0),
       duplicate_ack_counter_(0),
-      rtp_timebase_(rtp_timebase),
       congestion_control_(congestion_control),
+      rtp_timebase_(rtp_timebase),
       is_audio_(is_audio),
       weak_factory_(this) {
   DCHECK(transport_sender_);
+  DCHECK_GE(max_frame_rate_, 0.001);
   DCHECK_GT(rtp_timebase_, 0);
   DCHECK(congestion_control_);
   SetTargetPlayoutDelay(playout_delay);
@@ -91,6 +99,30 @@ void FrameSender::SendRtcpReport(bool schedule_future_reports) {
 void FrameSender::OnMeasuredRoundTripTime(base::TimeDelta rtt) {
   DCHECK(rtt > base::TimeDelta());
   current_round_trip_time_ = rtt;
+
+  // Rotate the "max round trip time" buckets, if needed to prune-out old

[hubbe, 2014/09/11 23:05:06]

This is a fun algorithm, but do we need it?
Would it cause problems to use the current rtt or some sort of average?

[miu, 2014/09/18 01:10:52]

Removed as discussed.

+  // history, and then update the current bucket and overall expected value.
+  const base::TimeTicks now = cast_environment_->Clock()->NowTicks();
+  if (last_max_rtt_bucket_rotation_.is_null() ||
+      (now - last_max_rtt_bucket_rotation_) >
+          base::TimeDelta::FromMilliseconds(kRttBucketRotationMs)) {
+    if (max_rtt_buckets_.size() == kNumRttBuckets)
+      max_rtt_buckets_.pop_front();
+    max_rtt_buckets_.push_back(base::TimeDelta());
+    last_max_rtt_bucket_rotation_ = now;
+  }
+  if (current_round_trip_time_ > max_rtt_buckets_.back()) {
+    max_rtt_buckets_.back() = current_round_trip_time_;
+    const base::TimeDelta& max_expected_rtt =
+        *(std::max_element(max_rtt_buckets_.begin(), max_rtt_buckets_.end()));
+    expected_max_one_way_trip_time_ = max_expected_rtt / 2;
+  }
+
+  VLOG(2)
+      << (is_audio_ ? "AUDIO[" : "VIDEO[") << ssrc_ << "] Last measured RTT is "
+      << current_round_trip_time_.InMicroseconds()
+      << " usec (expected max one way trip time is "
+      << expected_max_one_way_trip_time_.InMicroseconds() << " usec)";
 }
 
 void FrameSender::SetTargetPlayoutDelay(
@@ -162,11 +194,30 @@ RtpTimestamp FrameSender::GetRecordedRtpTimestamp(uint32 frame_id) const {
   return frame_rtp_timestamps_[frame_id % arraysize(frame_rtp_timestamps_)];
 }
 
+int FrameSender::GetUnackedFrameCount() const {
+  const int count =
+      static_cast<int32>(last_sent_frame_id_ - latest_acked_frame_id_);
+  DCHECK_GE(count, 0);
+  return count;
+}
+
+base::TimeDelta FrameSender::GetAllowedInFlightMediaDuration() const {
+  // The total amount allowed in-flight media should equal the amount that fits
+  // within the entire playout delay window, plus the maximum amount of time it
+  // could take to receive an ACK from the receiver.
+  return target_playout_delay_ +
+      std::min(expected_max_one_way_trip_time_, target_playout_delay_ / 2);
+}
+
 void FrameSender::SendEncodedFrame(
     int requested_bitrate_before_encode,
     scoped_ptr<EncodedFrame> encoded_frame) {
   DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
 
+  VLOG(2) << (is_audio_ ? "AUDIO[" : "VIDEO[") << ssrc_
+          << "] About to send another frame: last_sent="
+          << last_sent_frame_id_ << ", latest_acked=" << latest_acked_frame_id_;
+
   const uint32 frame_id = encoded_frame->frame_id;
 
   const bool is_first_frame_to_be_sent = last_send_time_.is_null();
@@ -297,32 +348,5 @@ void FrameSender::OnReceivedCastFeedback(const RtcpCastMessage& cast_feedback) {
   }
 }
 
-bool FrameSender::ShouldDropNextFrame(base::TimeTicks capture_time) const {
-  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
-  int frames_in_flight = 0;
-  base::TimeDelta duration_in_flight;
-  if (!last_send_time_.is_null()) {
-    frames_in_flight =
-        static_cast<int32>(last_sent_frame_id_ - latest_acked_frame_id_);
-    if (frames_in_flight > 0) {
-      const uint32 oldest_unacked_frame_id = latest_acked_frame_id_ + 1;
-      duration_in_flight =
-          capture_time - GetRecordedReferenceTime(oldest_unacked_frame_id);
-    }
-  }
-  frames_in_flight += GetNumberOfFramesInEncoder();
-  VLOG(2) << frames_in_flight
-          << " frames in flight; last sent: " << last_sent_frame_id_
-          << "; latest acked: " << latest_acked_frame_id_
-          << "; frames in encoder: " << GetNumberOfFramesInEncoder()
-          << "; duration in flight: "
-          << 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