Fix the issue of setting local description with datachannel only and max-bundle.

webrtc/api/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 1625 matching lines @@
   if (rtc_options.offer_to_receive_audio == RTCOfferAnswerOptions::kUndefined) {
     session_options->recv_audio =
         session_options->HasSendMediaStream(cricket::MEDIA_TYPE_AUDIO) ||
         !remote_audio_tracks_.empty();
   }
   if (rtc_options.offer_to_receive_video == RTCOfferAnswerOptions::kUndefined) {
     session_options->recv_video =
         session_options->HasSendMediaStream(cricket::MEDIA_TYPE_VIDEO) ||
         !remote_video_tracks_.empty();
   }
-  session_options->bundle_enabled =
-      session_options->bundle_enabled &&
-      (session_options->has_audio() || session_options->has_video() ||
-       session_options->has_data());
 
   // Intentionally unset the data channel type for RTP data channel with the
   // second condition. Otherwise the RTP data channels would be successfully
   // negotiated by default and the unit tests in WebRtcDataBrowserTest will fail
   // when building with chromium. We want to leave RTP data channels broken, so
   // people won't try to use them.
   if (HasDataChannels() && session_->data_channel_type() != cricket::DCT_RTP) {
     session_options->data_channel_type = session_->data_channel_type();
   }
 
+  session_options->bundle_enabled =
+      session_options->bundle_enabled &&
+      (session_options->has_audio() || session_options->has_video() ||
+       session_options->has_data());
+
   session_options->rtcp_cname = rtcp_cname_;
   session_options->crypto_options = factory_->options().crypto_options;
   return true;
 }
 
 void PeerConnection::InitializeOptionsForAnswer(
     cricket::MediaSessionOptions* session_options) {
   session_options->recv_audio = false;
   session_options->recv_video = false;
   session_options->enable_ice_renomination = enable_ice_renomination_;
 }
 
 void PeerConnection::FinishOptionsForAnswer(
     cricket::MediaSessionOptions* session_options) {
   // TODO(deadbeef): Once we have transceivers, enumerate them here instead of
   // ContentInfos.
   if (session_->remote_description()) {
     // Initialize the transport_options map.
     for (const cricket::ContentInfo& content :
          session_->remote_description()->description()->contents()) {
       session_options->transport_options[content.name] =
           cricket::TransportOptions();
     }
   }
   AddSendStreams(session_options, senders_, rtp_data_channels_);
-  session_options->bundle_enabled =
-      session_options->bundle_enabled &&
-      (session_options->has_audio() || session_options->has_video() ||
-       session_options->has_data());
-
   // RTP data channel is handled in MediaSessionOptions::AddStream. SCTP streams
   // are not signaled in the SDP so does not go through that path and must be
   // handled here.
   // Intentionally unset the data channel type for RTP data channel. Otherwise
   // the RTP data channels would be successfully negotiated by default and the
   // unit tests in WebRtcDataBrowserTest will fail when building with chromium.
   // We want to leave RTP data channels broken, so people won't try to use them.
   if (session_->data_channel_type() != cricket::DCT_RTP) {
     session_options->data_channel_type = session_->data_channel_type();
   }
+  session_options->bundle_enabled =
+      session_options->bundle_enabled &&
+      (session_options->has_audio() || session_options->has_video() ||
+       session_options->has_data());
+
   session_options->crypto_options = factory_->options().crypto_options;
 }
 
 bool PeerConnection::GetOptionsForAnswer(
     const MediaConstraintsInterface* constraints,
     cricket::MediaSessionOptions* session_options) {
   InitializeOptionsForAnswer(session_options);
   if (!ParseConstraintsForAnswer(constraints, session_options)) {
     return false;
   }
@@ skipping 631 matching lines @@
 
 bool PeerConnection::StartRtcEventLog_w(rtc::PlatformFile file,
                                         int64_t max_size_bytes) {
   return event_log_->StartLogging(file, max_size_bytes);
 }
 
 void PeerConnection::StopRtcEventLog_w() {
   event_log_->StopLogging();
 }
 }  // namespace webrtc