Set the start bitrate to the delay-based BWE.

webrtc/call/call.cc

 /*
  *  Copyright (c) 2013 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 289 matching lines @@
                                                       this,
                                                       &remb_,
                                                       event_log_,
                                                       &packet_router_)),
       video_send_delay_stats_(new SendDelayStats(clock_)),
       start_ms_(clock_->TimeInMilliseconds()),
       worker_queue_("call_worker_queue") {
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   RTC_DCHECK(config.event_log != nullptr);
   RTC_DCHECK_GE(config.bitrate_config.min_bitrate_bps, 0);
-  RTC_DCHECK_GE(config.bitrate_config.start_bitrate_bps,
+  RTC_DCHECK_GT(config.bitrate_config.start_bitrate_bps,

[terelius, 2017/01/24 16:05:52]

Why not DCHECK_GE? That seems more natural to me.

[stefan-webrtc, 2017/01/24 16:29:43]

It's just a bit unclear to me why you'd want to start with a bitrate of zero. I
could allow it, but seems like a risk for bugs where we'd potentially get stuck
if some encoder can't just produces nothing.

WDYT, should I change?

[terelius, 2017/01/25 16:41:27]

Ok, I see. Is there a reason for allowing min_bitrate_bps_ == 0 in that case?

[stefan-webrtc, 2017/01/27 09:27:05]

Good question. It will immediately get clamped here:
https://cs.chromium.org/chromium/src/third_party/webrtc/modules/congestion_co...

I don't want to change this rule in this CL as 0 is default in the call config.
Let's pick it up separately if we feel it's worth changing.

                 config.bitrate_config.min_bitrate_bps);
   if (config.bitrate_config.max_bitrate_bps != -1) {
     RTC_DCHECK_GE(config.bitrate_config.max_bitrate_bps,
                   config.bitrate_config.start_bitrate_bps);
   }
   Trace::CreateTrace();
   call_stats_->RegisterStatsObserver(congestion_controller_.get());
 
   congestion_controller_->SignalNetworkState(kNetworkDown);
   congestion_controller_->SetBweBitrates(
@@ skipping 488 matching lines @@
           bitrate_config.max_bitrate_bps) {
     // Nothing new to set, early abort to avoid encoder reconfigurations.
     return;
   }
   config_.bitrate_config.min_bitrate_bps = bitrate_config.min_bitrate_bps;
   // Start bitrate of -1 means we should keep the old bitrate, which there is
   // no point in remembering for the future.
   if (bitrate_config.start_bitrate_bps > 0)
     config_.bitrate_config.start_bitrate_bps = bitrate_config.start_bitrate_bps;
   config_.bitrate_config.max_bitrate_bps = bitrate_config.max_bitrate_bps;
+  RTC_DCHECK_NE(bitrate_config.start_bitrate_bps, 0);
   congestion_controller_->SetBweBitrates(bitrate_config.min_bitrate_bps,
                                          bitrate_config.start_bitrate_bps,
                                          bitrate_config.max_bitrate_bps);
 }
 
 void Call::SignalChannelNetworkState(MediaType media, NetworkState state) {
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   switch (media) {
     case MediaType::AUDIO:
       audio_network_state_ = state;
@@ skipping 76 matching lines @@
   if (kv->second != network_route) {
     kv->second = network_route;
     LOG(LS_INFO) << "Network route changed on transport " << transport_name
                  << ": new local network id " << network_route.local_network_id
                  << " new remote network id " << network_route.remote_network_id
                  << " Reset bitrates to min: "
                  << config_.bitrate_config.min_bitrate_bps
                  << " bps, start: " << config_.bitrate_config.start_bitrate_bps
                  << " bps,  max: " << config_.bitrate_config.start_bitrate_bps
                  << " bps.";
+    RTC_DCHECK_GT(config_.bitrate_config.start_bitrate_bps, 0);
     congestion_controller_->ResetBweAndBitrates(
         config_.bitrate_config.start_bitrate_bps,
         config_.bitrate_config.min_bitrate_bps,
         config_.bitrate_config.max_bitrate_bps);
   }
 }
 
 void Call::UpdateAggregateNetworkState() {
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
 
@@ skipping 281 matching lines @@
 
 void Call::NotifyBweOfReceivedPacket(const RtpPacketReceived& packet) {
   RTPHeader header;
   packet.GetHeader(&header);
   congestion_controller_->OnReceivedPacket(packet.arrival_time_ms(),
                                            packet.payload_size(), header);
 }
 
 }  // namespace internal
 }  // namespace webrtc