Don't send audio packets if the network is down.

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 370 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 = pacing_bitrate_kbps_;
-  // 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 =
@@ skipping 28 matching lines @@
       packets_->FinalizePop(packet);
       if (is_probing && bytes_sent > recommended_probe_size)
         break;
     } else {
       // Send failed, put it back into the queue.
       packets_->CancelPop(packet);
       break;
     }
   }
 
-  // TODO(holmer): Remove the paused_ check when issue 5307 has been fixed.
   if (packets_->Empty() && !paused_) {
     // We can not send padding unless a normal packet has first been sent. If we
     // do, timestamps get messed up.
     if (packet_counter_ > 0) {
       int padding_needed =
           static_cast<int>(is_probing ? (recommended_probe_size - bytes_sent)
                                       : padding_budget_->bytes_remaining());
 
       if (padding_needed > 0)
         bytes_sent += SendPadding(padding_needed, probe_cluster_id);
     }
   }
   if (is_probing && bytes_sent > 0)
     prober_->ProbeSent(clock_->TimeInMilliseconds(), bytes_sent);
   alr_detector_->OnBytesSent(bytes_sent, now_us / 1000);
 }
 
 bool PacedSender::SendPacket(const paced_sender::Packet& packet,
                              int probe_cluster_id) {
-  // 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 (packet.priority != kHighPriority) {
-    if (paused_)
-      return false;
-    if (media_budget_->bytes_remaining() == 0 &&
-        probe_cluster_id == PacketInfo::kNotAProbe) {
-      return false;
-    }
+  if (paused_)
+    return false;
+  if (media_budget_->bytes_remaining() == 0 &&
+      probe_cluster_id == PacketInfo::kNotAProbe) {
+    return false;
   }
   critsect_->Leave();
   const bool success = packet_sender_->TimeToSendPacket(
       packet.ssrc, packet.sequence_number, packet.capture_time_ms,
       packet.retransmission, probe_cluster_id);
   critsect_->Enter();
 
   if (success) {
     // TODO(holmer): High priority packets should only be accounted for if we
     // are allocating bandwidth for audio.
@@ skipping 21 matching lines @@
 void PacedSender::UpdateBudgetWithElapsedTime(int64_t delta_time_ms) {
   media_budget_->IncreaseBudget(delta_time_ms);
   padding_budget_->IncreaseBudget(delta_time_ms);
 }
 
 void PacedSender::UpdateBudgetWithBytesSent(size_t bytes_sent) {
   media_budget_->UseBudget(bytes_sent);
   padding_budget_->UseBudget(bytes_sent);
 }
 }  // namespace webrtc