Disable pacer disabling.

webrtc/modules/pacing/include/paced_sender.h

 /*
  *  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.
  */
 
@@ skipping 53 matching lines @@
   static const size_t kMinProbePacketSize = 200;
 
   PacedSender(Clock* clock,
               Callback* callback,
               int bitrate_kbps,
               int max_bitrate_kbps,
               int min_bitrate_kbps);
 
   virtual ~PacedSender();
 
-  // Enable/disable pacing.
-  void SetStatus(bool enable);
-
-  bool Enabled() const;
-
   // Temporarily pause all sending.
   void Pause();
 
   // Resume sending packets.
   void Resume();
 
   // Enable bitrate probing. Enabled by default, mostly here to simplify
   // testing. Must be called before any packets are being sent to have an
   // effect.
   void SetProbingEnabled(bool enabled);
 
   // Set target bitrates for the pacer.
   // We will pace out bursts of packets at a bitrate of |max_bitrate_kbps|.
   // |bitrate_kbps| is our estimate of what we are allowed to send on average.
   // Padding packets will be utilized to reach |min_bitrate| unless enough media
   // packets are available.
   void UpdateBitrate(int bitrate_kbps,
                      int max_bitrate_kbps,
                      int min_bitrate_kbps);
 
   // Returns true if we send the packet now, else it will add the packet
   // information to the queue and call TimeToSendPacket when it's time to send.
-  bool SendPacket(RtpPacketSender::Priority priority,
-                  uint32_t ssrc,
-                  uint16_t sequence_number,
-                  int64_t capture_time_ms,
-                  size_t bytes,
-                  bool retransmission) override;
+  void InsertPacket(RtpPacketSender::Priority priority,
+                    uint32_t ssrc,
+                    uint16_t sequence_number,
+                    int64_t capture_time_ms,
+                    size_t bytes,
+                    bool retransmission) override;
 
   // Returns the time since the oldest queued packet was enqueued.
   virtual int64_t QueueInMs() const;
 
   virtual size_t QueueSizePackets() const;
 
   // Returns the number of milliseconds it will take to send the current
   // packets in the queue, given the current size and bitrate, ignoring prio.
   virtual int64_t ExpectedQueueTimeMs() const;
 
@@ skipping 10 matching lines @@
       EXCLUSIVE_LOCKS_REQUIRED(critsect_);
 
   bool SendPacket(const paced_sender::Packet& packet)
       EXCLUSIVE_LOCKS_REQUIRED(critsect_);
   void SendPadding(size_t padding_needed) EXCLUSIVE_LOCKS_REQUIRED(critsect_);
 
   Clock* const clock_;
   Callback* const callback_;
 
   rtc::scoped_ptr<CriticalSectionWrapper> critsect_;
-  bool enabled_ GUARDED_BY(critsect_);
   bool paused_ GUARDED_BY(critsect_);
   bool probing_enabled_;
   // This is the media budget, keeping track of how many bits of media
   // we can pace out during the current interval.
   rtc::scoped_ptr<paced_sender::IntervalBudget> media_budget_
       GUARDED_BY(critsect_);
   // This is the padding budget, keeping track of how many bits of padding we're
   // allowed to send out during the current interval. This budget will be
   // utilized when there's no media to send.
   rtc::scoped_ptr<paced_sender::IntervalBudget> padding_budget_
       GUARDED_BY(critsect_);
 
   rtc::scoped_ptr<BitrateProber> prober_ GUARDED_BY(critsect_);
   int bitrate_bps_ GUARDED_BY(critsect_);
 
   int64_t time_last_update_us_ GUARDED_BY(critsect_);
 
   rtc::scoped_ptr<paced_sender::PacketQueue> packets_ GUARDED_BY(critsect_);
   uint64_t packet_counter_;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_PACING_INCLUDE_PACED_SENDER_H_