Make the audio channel communicate network state changes to the call.

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 361 matching lines @@
   int64_t elapsed_time_ms = (elapsed_time_us + 500) / 1000;
   return std::max<int64_t>(kMinPacketLimitMs - elapsed_time_ms, 0);
 }
 
 void PacedSender::Process() {
   int64_t now_us = clock_->TimeInMicroseconds();
   CriticalSectionScoped cs(critsect_.get());
   int64_t elapsed_time_ms = (now_us - time_last_update_us_ + 500) / 1000;
   time_last_update_us_ = now_us;
   int target_bitrate_kbps = max_bitrate_kbps_;
-  // TODO(holmer): Remove the !paused_ check when issue 5307 has been fixed.
+  // TODO(skvlad): Remove the !paused_ check when media channel state tracking
+  // is correctly implemented in call.cc
   if (!paused_ && elapsed_time_ms > 0) {
     size_t queue_size_bytes = packets_->SizeInBytes();
     if (queue_size_bytes > 0) {
       // Assuming equal size packets and input/output rate, the average packet
       // has avg_time_left_ms left to get queue_size_bytes out of the queue, if
       // time constraint shall be met. Determine bitrate needed for that.
       packets_->UpdateQueueTime(clock_->TimeInMilliseconds());
       int64_t avg_time_left_ms = std::max<int64_t>(
           1, kMaxQueueLengthMs - packets_->AverageQueueTimeMs());
       int min_bitrate_needed_kbps =
@@ skipping 21 matching lines @@
       packets_->FinalizePop(packet);
       if (prober_->IsProbing())
         return;
     } else {
       // Send failed, put it back into the queue.
       packets_->CancelPop(packet);
       return;
     }
   }
 
-  // TODO(holmer): Remove the paused_ check when issue 5307 has been fixed.
+  // TODO(skvlad): Remove the paused_ check when media channel state tracking
+  // is correctly implemented in call.cc
   if (paused_ || !packets_->Empty())
     return;
 
   size_t padding_needed;
   if (prober_->IsProbing()) {
     padding_needed = prober_->RecommendedPacketSize();
   } else {
     padding_needed = padding_budget_->bytes_remaining();
   }
 
   if (padding_needed > 0)
     SendPadding(static_cast<size_t>(padding_needed));
 }
 
 bool PacedSender::SendPacket(const paced_sender::Packet& packet) {
-  // TODO(holmer): Because of this bug issue 5307 we have to send audio
-  // packets even when the pacer is paused. Here we assume audio packets are
-  // always high priority and that they are the only high priority packets.
+  // TODO(skvlad): Even though issue 5307 is fixed, we're keeping the
+  // workaround to send audio packets when the pacer is paused until accurate
+  // tracking of the state of audio and video MediaChannels is implemented in
+  // call.cc.
+  // Here we assume audio packets are always high priority and that they are
+  // the only high priority packets.
   if (paused_ && packet.priority != kHighPriority)
     return false;
   critsect_->Leave();
   const bool success = callback_->TimeToSendPacket(packet.ssrc,
                                                    packet.sequence_number,
                                                    packet.capture_time_ms,
                                                    packet.retransmission);
   critsect_->Enter();
 
   // TODO(holmer): High priority packets should only be accounted for if we are
@@ skipping 18 matching lines @@
     media_budget_->UseBudget(bytes_sent);
     padding_budget_->UseBudget(bytes_sent);
   }
 }
 
 void PacedSender::UpdateBytesPerInterval(int64_t delta_time_ms) {
   media_budget_->IncreaseBudget(delta_time_ms);
   padding_budget_->IncreaseBudget(delta_time_ms);
 }
 }  // namespace webrtc