Prepare the AudioSendStream to be hooked up to send-side BWE.

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 340 matching lines @@
   int64_t elapsed_time_us = clock_->TimeInMicroseconds() - time_last_update_us_;
   int64_t elapsed_time_ms = (elapsed_time_us + 500) / 1000;
   return std::max<int64_t>(kMinPacketLimitMs - elapsed_time_ms, 0);
 }
 
 int32_t 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;
-  if (paused_)
-    return 0;
   int target_bitrate_kbps = max_bitrate_kbps_;
-  if (elapsed_time_ms > 0) {
+  // TODO(holmer): Remove the !paused_ check when issue 5307 has been fixed.
+  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 =
           static_cast<int>(queue_size_bytes * 8 / avg_time_left_ms);
       if (min_bitrate_needed_kbps > target_bitrate_kbps)
         target_bitrate_kbps = min_bitrate_needed_kbps;
     }
 
     media_budget_->set_target_rate_kbps(target_bitrate_kbps);
 
     int64_t delta_time_ms = std::min(kMaxIntervalTimeMs, elapsed_time_ms);
     UpdateBytesPerInterval(delta_time_ms);
   }
   while (!packets_->Empty()) {
     if (media_budget_->bytes_remaining() == 0 && !prober_->IsProbing())
       return 0;
 
     // Since we need to release the lock in order to send, we first pop the
     // element from the priority queue but keep it in storage, so that we can
     // reinsert it if send fails.
     const paced_sender::Packet& packet = packets_->BeginPop();
-    if (SendPacket(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.
+    if ((!paused_ || packet.priority == kHighPriority) && SendPacket(packet)) {
       // Send succeeded, remove it from the queue.
       packets_->FinalizePop(packet);
       if (prober_->IsProbing())
         return 0;
     } else {
       // Send failed, put it back into the queue.
       packets_->CancelPop(packet);
       return 0;
     }
   }
 
-  if (!packets_->Empty())
+  // TODO(holmer): Remove the paused_ check when issue 5307 has been fixed.
+  if (paused_ || !packets_->Empty())
     return 0;
 
   size_t padding_needed;
   if (prober_->IsProbing()) {
     padding_needed = prober_->RecommendedPacketSize();
   } else {
     padding_needed = padding_budget_->bytes_remaining();
   }
 
   if (padding_needed > 0)
@@ skipping 29 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