Fix two invalid DCHECKs related to audio BWE.

webrtc/audio/audio_send_stream.cc

 /*
  *  Copyright (c) 2015 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 228 matching lines @@
   // calls on the worker thread. We should move towards always using a network
   // thread. Then this check can be enabled.
   // RTC_DCHECK(!worker_thread_checker_.CalledOnValidThread());
   return channel_proxy_->ReceivedRTCPPacket(packet, length);
 }
 
 uint32_t AudioSendStream::OnBitrateUpdated(uint32_t bitrate_bps,
                                            uint8_t fraction_loss,
                                            int64_t rtt,
                                            int64_t probing_interval_ms) {
+  // A send stream may be allocated a bitrate of zero if the allocator decides
+  // to disable it. For now we ignore this decision and keep sending on min
+  // bitrate.
+  if (bitrate_bps == 0) {
+    bitrate_bps = config_.min_bitrate_bps;
+  }
   RTC_DCHECK_GE(bitrate_bps,
                 static_cast<uint32_t>(config_.min_bitrate_bps));
   // The bitrate allocator might allocate an higher than max configured bitrate
   // if there is room, to allow for, as example, extra FEC. Ignore that for now.
   const uint32_t max_bitrate_bps = config_.max_bitrate_bps;
   if (bitrate_bps > max_bitrate_bps)
     bitrate_bps = max_bitrate_bps;
 
   channel_proxy_->SetBitrate(bitrate_bps, probing_interval_ms);
 
@@ skipping 165 matching lines @@
         LOG(LS_WARNING) << "SetVADStatus() failed.";
         return false;
       }
     }
   }
   return true;
 }
 
 }  // namespace internal
 }  // namespace webrtc