BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic.

This CL implements the main logic and IOS appRTC integration.

Unit tests and Android appRTC will be in separate CL.

Bug: webrtc:8243
Change-Id: If8e5195294046a47316e9fade1b0dfec211155e1
Reviewed-on: https://webrtc-review.googlesource.com/4860
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20329}

modules/rtp_rtcp/include/rtp_rtcp_defines.h

@@ -498,6 +498,14 @@ class RtpPacketSender {
                             int64_t capture_time_ms,
                             size_t bytes,
                             bool retransmission) = 0;
+
+  // 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.
+  // TODO(alexnarest): Make it pure virtual after rtp_sender_unittest will be
+  // updated to support it
+  virtual void SetAccountForAudioPackets(bool account_for_audio) {}
 };
 
 class TransportSequenceNumberAllocator {