Change order of tear down/create of default audio stream, to avoid starting/stopping audio card pla…

webrtc/media/engine/webrtcvoiceengine.cc

 /*
  *  Copyright (c) 2004 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 2481 matching lines @@
   }
 
   // Create a default receive stream for this unsignalled and previously not
   // received ssrc. If there already is a default receive stream, delete it.
   // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=5208
   uint32_t ssrc = 0;
   if (!GetRtpSsrc(packet->cdata(), packet->size(), &ssrc)) {
     return;
   }
 
-  if (default_recv_ssrc_ != -1) {
-    LOG(LS_INFO) << "Removing default receive stream with ssrc "
-                 << default_recv_ssrc_;
-    RTC_DCHECK_NE(ssrc, default_recv_ssrc_);
-    RemoveRecvStream(default_recv_ssrc_);
-    default_recv_ssrc_ = -1;
-  }
-
   StreamParams sp;
   sp.ssrcs.push_back(ssrc);
   LOG(LS_INFO) << "Creating default receive stream for SSRC=" << ssrc << ".";
   if (!AddRecvStream(sp)) {
     LOG(LS_WARNING) << "Could not create default receive stream.";
     return;
   }
+  if (default_recv_ssrc_ != -1) {
+    LOG(LS_INFO) << "Removing default receive stream with ssrc "
+                 << default_recv_ssrc_;
+    RTC_DCHECK_NE(ssrc, default_recv_ssrc_);
+    RemoveRecvStream(default_recv_ssrc_);
+  }
   default_recv_ssrc_ = ssrc;
+
   SetOutputVolume(default_recv_ssrc_, default_recv_volume_);
   if (default_sink_) {
     std::unique_ptr<webrtc::AudioSinkInterface> proxy_sink(
         new ProxySink(default_sink_.get()));
     SetRawAudioSink(default_recv_ssrc_, std::move(proxy_sink));
   }
   delivery_result = call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO,
                                                      packet->cdata(),
                                                      packet->size(),
                                                      webrtc_packet_time);
@@ skipping 192 matching lines @@
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   const auto it = send_streams_.find(ssrc);
   if (it != send_streams_.end()) {
     return it->second->channel();
   }
   return -1;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE