| Index: webrtc/api/webrtcsession.cc
|
| diff --git a/webrtc/api/webrtcsession.cc b/webrtc/api/webrtcsession.cc
|
| index 35a5fab4c6ec08850f13ce8a11fe015ae6e61609..7497e2c7e0a9f879f6590fe22f7e041f053d1938 100644
|
| --- a/webrtc/api/webrtcsession.cc
|
| +++ b/webrtc/api/webrtcsession.cc
|
| @@ -24,7 +24,6 @@
|
| #include "webrtc/api/webrtcsessiondescriptionfactory.h"
|
| #include "webrtc/audio_sink.h"
|
| #include "webrtc/base/basictypes.h"
|
| -#include "webrtc/base/bind.h"
|
| #include "webrtc/base/checks.h"
|
| #include "webrtc/base/helpers.h"
|
| #include "webrtc/base/logging.h"
|
| @@ -38,10 +37,6 @@
|
| #include "webrtc/pc/channel.h"
|
| #include "webrtc/pc/channelmanager.h"
|
| #include "webrtc/pc/mediasession.h"
|
| -
|
| -#ifdef HAVE_QUIC
|
| -#include "webrtc/p2p/quic/quictransportchannel.h"
|
| -#endif // HAVE_QUIC
|
|
|
| using cricket::ContentInfo;
|
| using cricket::ContentInfos;
|
| @@ -465,8 +460,7 @@
|
| rtc::Thread* signaling_thread,
|
| cricket::PortAllocator* port_allocator,
|
| std::unique_ptr<cricket::TransportController> transport_controller)
|
| - : network_thread_(network_thread),
|
| - worker_thread_(worker_thread),
|
| + : worker_thread_(worker_thread),
|
| signaling_thread_(signaling_thread),
|
| // RFC 3264: The numeric value of the session id and version in the
|
| // o line MUST be representable with a "64 bit signed integer".
|
| @@ -511,11 +505,6 @@
|
| SignalDataChannelDestroyed();
|
| channel_manager_->DestroyDataChannel(data_channel_.release());
|
| }
|
| -#ifdef HAVE_QUIC
|
| - if (quic_data_transport_) {
|
| - quic_data_transport_.reset();
|
| - }
|
| -#endif
|
| SignalDestroyed();
|
|
|
| LOG(LS_INFO) << "Session: " << id() << " is destroyed.";
|
| @@ -556,21 +545,7 @@
|
| // PeerConnectionFactoryInterface::Options.
|
| if (rtc_configuration.enable_rtp_data_channel) {
|
| data_channel_type_ = cricket::DCT_RTP;
|
| - }
|
| -#ifdef HAVE_QUIC
|
| - else if (rtc_configuration.enable_quic) {
|
| - // Use QUIC instead of DTLS when |enable_quic| is true.
|
| - data_channel_type_ = cricket::DCT_QUIC;
|
| - transport_controller_->use_quic();
|
| - if (dtls_enabled_) {
|
| - LOG(LS_INFO) << "Using QUIC instead of DTLS";
|
| - }
|
| - quic_data_transport_.reset(
|
| - new QuicDataTransport(signaling_thread(), worker_thread(),
|
| - network_thread(), transport_controller_.get()));
|
| - }
|
| -#endif // HAVE_QUIC
|
| - else {
|
| + } else {
|
| // DTLS has to be enabled to use SCTP.
|
| if (!options.disable_sctp_data_channels && dtls_enabled_) {
|
| data_channel_type_ = cricket::DCT_SCTP;
|
| @@ -1059,15 +1034,6 @@
|
| }
|
| const std::string& transport_name = *first_content_name;
|
| cricket::BaseChannel* first_channel = GetChannel(transport_name);
|
| -
|
| -#ifdef HAVE_QUIC
|
| - if (quic_data_transport_ &&
|
| - bundle.HasContentName(quic_data_transport_->content_name()) &&
|
| - quic_data_transport_->transport_name() != transport_name) {
|
| - LOG(LS_ERROR) << "Unable to BUNDLE " << quic_data_transport_->content_name()
|
| - << " on " << transport_name << "with QUIC.";
|
| - }
|
| -#endif
|
|
|
| auto maybe_set_transport = [this, bundle, transport_name,
|
| first_channel](cricket::BaseChannel* ch) {
|
| @@ -1577,17 +1543,9 @@
|
|
|
| const cricket::ContentInfo* data_info =
|
| cricket::GetFirstDataContent(desc);
|
| - if (!data_info || data_info->rejected) {
|
| - if (data_channel_) {
|
| - SignalDataChannelDestroyed();
|
| - channel_manager_->DestroyDataChannel(data_channel_.release());
|
| - }
|
| -#ifdef HAVE_QUIC
|
| - // Clean up the existing QuicDataTransport and its QuicTransportChannels.
|
| - if (quic_data_transport_) {
|
| - quic_data_transport_.reset();
|
| - }
|
| -#endif
|
| + if ((!data_info || data_info->rejected) && data_channel_) {
|
| + SignalDataChannelDestroyed();
|
| + channel_manager_->DestroyDataChannel(data_channel_.release());
|
| }
|
| }
|
|
|
| @@ -1701,15 +1659,6 @@
|
|
|
| bool WebRtcSession::CreateDataChannel(const cricket::ContentInfo* content,
|
| const std::string* bundle_transport) {
|
| -#ifdef HAVE_QUIC
|
| - if (data_channel_type_ == cricket::DCT_QUIC) {
|
| - RTC_DCHECK(transport_controller_->quic());
|
| - const std::string transport_name =
|
| - bundle_transport ? *bundle_transport : content->name;
|
| - quic_data_transport_->SetTransport(transport_name);
|
| - return true;
|
| - }
|
| -#endif // HAVE_QUIC
|
| bool sctp = (data_channel_type_ == cricket::DCT_SCTP);
|
| bool require_rtcp_mux =
|
| rtcp_mux_policy_ == PeerConnectionInterface::kRtcpMuxPolicyRequire;
|
| @@ -1893,7 +1842,7 @@
|
| const IceCandidateInterface* candidate,
|
| const SessionDescriptionInterface* remote_desc,
|
| bool* valid) {
|
| - *valid = true;
|
| + *valid = true;;
|
|
|
| const SessionDescriptionInterface* current_remote_desc =
|
| remote_desc ? remote_desc : remote_desc_.get();
|
| @@ -1916,12 +1865,13 @@
|
|
|
| cricket::ContentInfo content =
|
| current_remote_desc->description()->contents()[mediacontent_index];
|
| -
|
| - const std::string transport_name = GetTransportName(content.name);
|
| - if (transport_name.empty()) {
|
| - return false;
|
| - }
|
| - return transport_controller_->ReadyForRemoteCandidates(transport_name);
|
| + cricket::BaseChannel* channel = GetChannel(content.name);
|
| + if (!channel) {
|
| + return false;
|
| + }
|
| +
|
| + return transport_controller_->ReadyForRemoteCandidates(
|
| + channel->transport_name());
|
| }
|
|
|
| void WebRtcSession::OnTransportControllerGatheringState(
|
| @@ -2058,19 +2008,4 @@
|
| media_controller_->call_w()->OnSentPacket(sent_packet);
|
| }
|
|
|
| -const std::string WebRtcSession::GetTransportName(
|
| - const std::string& content_name) {
|
| - cricket::BaseChannel* channel = GetChannel(content_name);
|
| - if (!channel) {
|
| -#ifdef HAVE_QUIC
|
| - if (data_channel_type_ == cricket::DCT_QUIC && quic_data_transport_ &&
|
| - content_name == quic_data_transport_->transport_name()) {
|
| - return quic_data_transport_->transport_name();
|
| - }
|
| -#endif
|
| - // Return an empty string if failed to retrieve the transport name.
|
| - return "";
|
| - }
|
| - return channel->transport_name();
|
| -}
|
| } // namespace webrtc
|
|
|