Add experiment on weak ping delay during call set up time

webrtc/p2p/base/p2ptransportchannel.cc

 /*
  *  Copyright 2004 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/p2p/base/p2ptransportchannel.h"
 
 #include <algorithm>
 #include <set>
 #include "webrtc/p2p/base/common.h"
 #include "webrtc/p2p/base/relayport.h"  // For RELAY_PORT_TYPE.
 #include "webrtc/p2p/base/stunport.h"  // For STUN_PORT_TYPE.
 #include "webrtc/base/common.h"
 #include "webrtc/base/crc32.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/stringencode.h"
+#include "webrtc/system_wrappers/interface/field_trial.h"
 
 namespace {
 
 // messages for queuing up work for ourselves
 enum { MSG_SORT = 1, MSG_CHECK_AND_PING };
 
 // When the socket is unwritable, we will use 10 Kbps (ignoring IP+UDP headers)
 // for pinging.  When the socket is writable, we will use only 1 Kbps because
 // we don't want to degrade the quality on a modem.  These numbers should work
 // well on a 28.8K modem, which is the slowest connection on which the voice
@@ skipping 181 matching lines @@
       best_connection_(NULL),
       pending_best_connection_(NULL),
       sort_dirty_(false),
       was_writable_(false),
       remote_ice_mode_(ICEMODE_FULL),
       ice_role_(ICEROLE_UNKNOWN),
       tiebreaker_(0),
       remote_candidate_generation_(0),
       gathering_state_(kIceGatheringNew),
       check_receiving_delay_(MIN_CHECK_RECEIVING_DELAY * 5),
-      receiving_timeout_(MIN_CHECK_RECEIVING_DELAY * 50) {}
+      receiving_timeout_(MIN_CHECK_RECEIVING_DELAY * 50) {
+  uint32_t weak_ping_delay = ::strtoul(
+      webrtc::field_trial::FindFullName("WebRTC-WeakPingDelay").c_str(),

[pthatcher1, 2015/10/16 21:30:31]

While we call it "weak ping delay" internally, the name is a bit ugly.  I think
we may want a nicer one for the trial name.  How about
"WebRTC-StunInterPacketDelay"?

+      nullptr, 10);
+  weak_ping_delay_ = weak_ping_delay ? weak_ping_delay : WEAK_PING_DELAY;

[pthatcher1, 2015/10/16 21:30:31]

I think this would be more readable as:

if (weak_ping_delay) {
  weak_ping_delay_ = weak_ping_delay;
}

+}
 
 P2PTransportChannel::~P2PTransportChannel() {
   ASSERT(worker_thread_ == rtc::Thread::Current());
 
   for (size_t i = 0; i < allocator_sessions_.size(); ++i)
     delete allocator_sessions_[i];
 }
 
 // Add the allocator session to our list so that we know which sessions
 // are still active.
@@ skipping 915 matching lines @@
   SortConnections();
 }
 
 // Handle queued up check-and-ping request
 void P2PTransportChannel::OnCheckAndPing() {
   // Make sure the states of the connections are up-to-date (since this affects
   // which ones are pingable).
   UpdateConnectionStates();
   // When the best connection is either not receiving or not writable,
   // switch to weak ping delay.
-  int ping_delay = weak() ? WEAK_PING_DELAY : STRONG_PING_DELAY;
+  int ping_delay = weak() ? weak_ping_delay_ : STRONG_PING_DELAY;
   if (rtc::Time() >= last_ping_sent_ms_ + ping_delay) {
     Connection* conn = FindNextPingableConnection();
     if (conn) {
       PingConnection(conn);
     }
   }
   int check_delay = std::min(ping_delay, check_receiving_delay_);
   thread()->PostDelayed(check_delay, this, MSG_CHECK_AND_PING);
 }
 
@@ skipping 186 matching lines @@
   }
 }
 
 void P2PTransportChannel::OnReadyToSend(Connection* connection) {
   if (connection == best_connection_ && writable()) {
     SignalReadyToSend(this);
   }
 }
 
 }  // namespace cricket