Fix the Chromium crash when creating an answer without a remote description.

webrtc/pc/peerconnection.cc

 /*
  *  Copyright 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 1769 matching lines @@
   if (rtc_options.offer_to_receive_audio != RTCOfferAnswerOptions::kUndefined) {
     recv_audio = (rtc_options.offer_to_receive_audio > 0);
   }
   if (rtc_options.offer_to_receive_video != RTCOfferAnswerOptions::kUndefined) {
     recv_video = (rtc_options.offer_to_receive_video > 0);
   }
 
   rtc::Optional<size_t> audio_index;
   rtc::Optional<size_t> video_index;
   rtc::Optional<size_t> data_index;
-  // There should be a pending remote description that's an offer...
-  RTC_DCHECK(session_->remote_description());
-  RTC_DCHECK(session_->remote_description()->type() ==
-             SessionDescriptionInterface::kOffer);
-  // Generate m= sections that match those in the offer.
-  // Note that mediasession.cc will handle intersection our preferred direction
-  // with the offered direction.
-  GenerateMediaDescriptionOptions(
-      session_->remote_description(),
-      cricket::RtpTransceiverDirection(send_audio, recv_audio),
-      cricket::RtpTransceiverDirection(send_video, recv_video), &audio_index,
-      &video_index, &data_index, session_options);
+  if (session_->remote_description()) {
+    // The pending remote description should be an offer.
+    RTC_DCHECK(session_->remote_description()->type() ==
+               SessionDescriptionInterface::kOffer);
+    // Generate m= sections that match those in the offer.
+    // Note that mediasession.cc will handle intersection our preferred
+    // direction with the offered direction.
+    GenerateMediaDescriptionOptions(
+        session_->remote_description(),
+        cricket::RtpTransceiverDirection(send_audio, recv_audio),
+        cricket::RtpTransceiverDirection(send_video, recv_video), &audio_index,
+        &video_index, &data_index, session_options);
+  }
 
   cricket::MediaDescriptionOptions* audio_media_description_options =
       !audio_index ? nullptr
                    : &session_options->media_description_options[*audio_index];
   cricket::MediaDescriptionOptions* video_media_description_options =
       !video_index ? nullptr
                    : &session_options->media_description_options[*video_index];
   cricket::MediaDescriptionOptions* data_media_description_options =
       !data_index ? nullptr
                   : &session_options->media_description_options[*data_index];
@@ skipping 711 matching lines @@
   return event_log_->StartLogging(file, max_size_bytes);
 }
 
 void PeerConnection::StopRtcEventLog_w() {
   if (event_log_) {
     event_log_->StopLogging();
   }
 }
 
 }  // namespace webrtc