BWE allocation strategy

webrtc/modules/pacing/paced_sender.cc

Index: webrtc/modules/pacing/paced_sender.cc
diff --git a/webrtc/modules/pacing/paced_sender.cc b/webrtc/modules/pacing/paced_sender.cc
index 7c24301b95f0d830af5effde988a9a182ac36d48..11d6596dfa67e734fa25e3fd12c8b98da3a82f7f 100644
--- a/webrtc/modules/pacing/paced_sender.cc
+++ b/webrtc/modules/pacing/paced_sender.cc
@@ -264,7 +264,8 @@ PacedSender::PacedSender(const Clock* clock,
       time_last_update_us_(clock->TimeInMicroseconds()),
       first_sent_packet_ms_(-1),
       packets_(new paced_sender::PacketQueue(clock)),
-      packet_counter_(0) {
+      packet_counter_(0),
+      account_for_audio_(false) {
   UpdateBudgetWithElapsedTime(kMinPacketLimitMs);
 }
 
@@ -351,6 +352,11 @@ void PacedSender::InsertPacket(RtpPacketSender::Priority priority,
                                       retransmission, packet_counter_++));
 }
 
+void PacedSender::AccountForAudioPackets(bool account_for_audio) {
+  rtc::CritScope cs(&critsect_);
+  account_for_audio_ = account_for_audio;
+}
+
 int64_t PacedSender::ExpectedQueueTimeMs() const {
   rtc::CritScope cs(&critsect_);
   RTC_DCHECK_GT(pacing_bitrate_kbps_, 0);
@@ -502,9 +508,7 @@ bool PacedSender::SendPacket(const paced_sender::Packet& packet,
   critsect_.Enter();
 
   if (success) {
-    // TODO(holmer): High priority packets should only be accounted for if we
-    // are allocating bandwidth for audio.
-    if (packet.priority != kHighPriority) {
+    if (packet.priority != kHighPriority || account_for_audio_) {
       // Update media bytes sent.
       UpdateBudgetWithBytesSent(packet.bytes);
     }