BWE allocation strategy

webrtc/modules/pacing/paced_sender.cc

 /*
  *  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 238 matching lines @@
       probing_send_failure_(false),
       estimated_bitrate_bps_(0),
       min_send_bitrate_kbps_(0u),
       max_padding_bitrate_kbps_(0u),
       pacing_bitrate_kbps_(0),
       time_last_update_us_(clock->TimeInMicroseconds()),
       first_sent_packet_ms_(-1),
       packets_(new paced_sender::PacketQueue(clock)),
       packet_counter_(0),
       pacing_factor_(kDefaultPaceMultiplier),
-      queue_time_limit(kMaxQueueLengthMs) {
+      queue_time_limit(kMaxQueueLengthMs),
+      account_for_audio_(false) {
   UpdateBudgetWithElapsedTime(kMinPacketLimitMs);
 }
 
 PacedSender::~PacedSender() {}
 
 void PacedSender::CreateProbeCluster(int bitrate_bps) {
   rtc::CritScope cs(&critsect_);
   prober_->CreateProbeCluster(bitrate_bps, clock_->TimeInMilliseconds());
 }
 
@@ skipping 70 matching lines @@
   prober_->OnIncomingPacket(bytes);
 
   if (capture_time_ms < 0)
     capture_time_ms = now_ms;
 
   packets_->Push(paced_sender::Packet(priority, ssrc, sequence_number,
                                       capture_time_ms, now_ms, bytes,
                                       retransmission, packet_counter_++));
 }
 
+void PacedSender::SetAccountForAudioPackets(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);
   return static_cast<int64_t>(packets_->SizeInBytes() * 8 /
                               pacing_bitrate_kbps_);
 }
 
 rtc::Optional<int64_t> PacedSender::GetApplicationLimitedRegionStartTime()
     const {
   rtc::CritScope cs(&critsect_);
@@ skipping 145 matching lines @@
     return false;
   }
 
   critsect_.Leave();
   const bool success = packet_sender_->TimeToSendPacket(
       packet.ssrc, packet.sequence_number, packet.capture_time_ms,
       packet.retransmission, pacing_info);
   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.
       // TODO(eladalon): TimeToSendPacket() can also return |true| in some
       // situations where nothing actually ended up being sent to the network,
       // and we probably don't want to update the budget in such cases.
       // https://bugs.chromium.org/p/webrtc/issues/detail?id=8052
       UpdateBudgetWithBytesSent(packet.bytes);
     }
   }
 
   return success;
@@ skipping 27 matching lines @@
   rtc::CritScope cs(&critsect_);
   pacing_factor_ = pacing_factor;
 }
 
 void PacedSender::SetQueueTimeLimit(int limit_ms) {
   rtc::CritScope cs(&critsect_);
   queue_time_limit = limit_ms;
 }
 
 }  // namespace webrtc