Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage.

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 889 matching lines @@
 PeerConnectionInterface::IceGatheringState
 PeerConnection::ice_gathering_state() {
   return ice_gathering_state_;
 }
 
 rtc::scoped_refptr<DataChannelInterface>
 PeerConnection::CreateDataChannel(
     const std::string& label,
     const DataChannelInit* config) {
   TRACE_EVENT0("webrtc", "PeerConnection::CreateDataChannel");
+#ifdef HAVE_QUIC
+  if (session_->data_channel_type() == cricket::DCT_QUIC) {
+    // TODO(zhihuang): Handle case when config is NULL.
+    if (!config) {
+      LOG(LS_ERROR) << "Missing config for QUIC data channel.";
+      return nullptr;
+    }
+    // TODO(zhihuang): Allow unreliable or ordered QUIC data channels.
+    if (!config->reliable || config->ordered) {
+      LOG(LS_ERROR) << "QUIC data channel does not implement unreliable or "
+                       "ordered delivery.";
+      return nullptr;
+    }
+    return session_->quic_data_transport()->CreateDataChannel(label, config);
+  }
+#endif  // HAVE_QUIC
+
   bool first_datachannel = !HasDataChannels();
 
   std::unique_ptr<InternalDataChannelInit> internal_config;
   if (config) {
     internal_config.reset(new InternalDataChannelInit(*config));
   }
   rtc::scoped_refptr<DataChannelInterface> channel(
       InternalCreateDataChannel(label, internal_config.get()));
   if (!channel.get()) {
     return nullptr;
@@ skipping 691 matching lines @@
   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());
 
-  if (session_->data_channel_type() == cricket::DCT_SCTP && HasDataChannels()) {
-    session_options->data_channel_type = cricket::DCT_SCTP;
+  // 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->rtcp_cname = rtcp_cname_;
   session_options->crypto_options = factory_->options().crypto_options;
   return true;
 }
 
 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.
-  if (session_->data_channel_type() == cricket::DCT_SCTP) {
-    session_options->data_channel_type = cricket::DCT_SCTP;
+  // 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->crypto_options = factory_->options().crypto_options;
 }
 
 bool PeerConnection::GetOptionsForAnswer(
     const MediaConstraintsInterface* constraints,
     cricket::MediaSessionOptions* session_options) {
   session_options->recv_audio = false;
   session_options->recv_video = false;
   if (!ParseConstraintsForAnswer(constraints, session_options)) {
@@ skipping 384 matching lines @@
     RTC_DCHECK(channel->data_channel_type() == cricket::DCT_SCTP);
     sctp_data_channels_.push_back(channel);
     channel->SignalClosed.connect(this,
                                   &PeerConnection::OnSctpDataChannelClosed);
   }
 
   return channel;
 }
 
 bool PeerConnection::HasDataChannels() const {
+#ifdef HAVE_QUIC
+  return !rtp_data_channels_.empty() || !sctp_data_channels_.empty() ||
+         (session_->quic_data_transport() &&
+          session_->quic_data_transport()->HasDataChannels());
+#else
   return !rtp_data_channels_.empty() || !sctp_data_channels_.empty();
+#endif  // HAVE_QUIC
 }
 
 void PeerConnection::AllocateSctpSids(rtc::SSLRole role) {
   for (const auto& channel : sctp_data_channels_) {
     if (channel->id() < 0) {
       int sid;
       if (!sid_allocator_.AllocateSid(role, &sid)) {
         LOG(LS_ERROR) << "Failed to allocate SCTP sid.";
         continue;
       }
@@ skipping 223 matching lines @@
 
 bool PeerConnection::StartRtcEventLog_w(rtc::PlatformFile file,
                                         int64_t max_size_bytes) {
   return media_controller_->call_w()->StartEventLog(file, max_size_bytes);
 }
 
 void PeerConnection::StopRtcEventLog_w() {
   media_controller_->call_w()->StopEventLog();
 }
 }  // namespace webrtc