BWE allocation strategy

webrtc/modules/pacing/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 96 matching lines @@
 
   // 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.
   void InsertPacket(RtpPacketSender::Priority priority,
                     uint32_t ssrc,
                     uint16_t sequence_number,
                     int64_t capture_time_ms,
                     size_t bytes,
                     bool retransmission) override;
 
+  // Currently audio traffic is not accounted by pacer and passed through.
+  // With the introduction of audio BWE audio traffic will be accounted for
+  // the pacer budget calculation. The audio traffic still will be injected
+  // at high priority.
+  void SetAccountForAudioPackets(bool account_for_audio) override;

[nisse-webrtc, 2017/09/05 11:03:46]

Unclear to me how this change is related to the BWE strategy?

It's preferable to split out independent changes to their own cls, to ease
review and testing, but I don't insist that you do that here.

[alexnarest, 2017/09/05 13:37:46]

The strategy allows us to enable audio BWE for video calls and as soon as it
will be enabled it will make sense to account for audio packets. 

+
   // Returns the time since the oldest queued packet was enqueued.
   virtual int64_t QueueInMs() const;
 
   virtual size_t QueueSizePackets() const;
 
   // Returns the time when the first packet was sent, or -1 if no packet is
   // sent.
   virtual int64_t FirstSentPacketTimeMs() const;
 
   // Returns the number of milliseconds it will take to send the current
@@ skipping 64 matching lines @@
 
   int64_t time_last_update_us_ GUARDED_BY(critsect_);
   int64_t first_sent_packet_ms_ GUARDED_BY(critsect_);
 
   std::unique_ptr<paced_sender::PacketQueue> packets_ GUARDED_BY(critsect_);
   uint64_t packet_counter_;
   ProcessThread* process_thread_ = nullptr;
 
   float pacing_factor_ GUARDED_BY(critsect_);
   int64_t queue_time_limit GUARDED_BY(critsect_);
+  bool account_for_audio_ GUARDED_BY(critsect_);
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_PACING_PACED_SENDER_H_