Wire up RTP keep-alive in ortc api.

webrtc/ortc/rtptransportcontrolleradapter.cc

Index: webrtc/ortc/rtptransportcontrolleradapter.cc
diff --git a/webrtc/ortc/rtptransportcontrolleradapter.cc b/webrtc/ortc/rtptransportcontrolleradapter.cc
index 5e0b62112357ba849fec33d5e3428e1e5c235932..96ba8be155b803590f4dfb7ee3f895d9b5f6570f 100644
--- a/webrtc/ortc/rtptransportcontrolleradapter.cc
+++ b/webrtc/ortc/rtptransportcontrolleradapter.cc
@@ -92,13 +92,15 @@ END_PROXY_MAP()
 std::unique_ptr<RtpTransportControllerInterface>
 RtpTransportControllerAdapter::CreateProxied(
     const cricket::MediaConfig& config,
+    const RtpTransportParameters& parameters,
     cricket::ChannelManager* channel_manager,
     webrtc::RtcEventLog* event_log,
     rtc::Thread* signaling_thread,
     rtc::Thread* worker_thread) {
   std::unique_ptr<RtpTransportControllerAdapter> wrapped(
-      new RtpTransportControllerAdapter(config, channel_manager, event_log,
-                                        signaling_thread, worker_thread));
+      new RtpTransportControllerAdapter(config, parameters, channel_manager,
+                                        event_log, signaling_thread,
+                                        worker_thread));
   return RtpTransportControllerProxyWithInternal<
       RtpTransportControllerAdapter>::Create(signaling_thread, worker_thread,
                                              std::move(wrapped));
@@ -129,11 +131,10 @@ RtpTransportControllerAdapter::~RtpTransportControllerAdapter() {
 
 RTCErrorOr<std::unique_ptr<RtpTransportInterface>>
 RtpTransportControllerAdapter::CreateProxiedRtpTransport(
-    const RtcpParameters& rtcp_parameters,
+    const RtpTransportParameters& parameters,
     PacketTransportInterface* rtp,
     PacketTransportInterface* rtcp) {
-  auto result =
-      RtpTransportAdapter::CreateProxied(rtcp_parameters, rtp, rtcp, this);
+  auto result = RtpTransportAdapter::CreateProxied(parameters, rtp, rtcp, this);
   if (result.ok()) {
     transport_proxies_.push_back(result.value().get());
     transport_proxies_.back()->GetInternal()->SignalDestroyed.connect(
@@ -144,11 +145,11 @@ RtpTransportControllerAdapter::CreateProxiedRtpTransport(
 
 RTCErrorOr<std::unique_ptr<SrtpTransportInterface>>
 RtpTransportControllerAdapter::CreateProxiedSrtpTransport(
-    const RtcpParameters& rtcp_parameters,
+    const RtpTransportParameters& parameters,
     PacketTransportInterface* rtp,
     PacketTransportInterface* rtcp) {
   auto result =
-      RtpTransportAdapter::CreateSrtpProxied(rtcp_parameters, rtp, rtcp, this);
+      RtpTransportAdapter::CreateSrtpProxied(parameters, rtp, rtcp, this);
   if (result.ok()) {
     transport_proxies_.push_back(result.value().get());
     transport_proxies_.back()->GetInternal()->SignalDestroyed.connect(
@@ -219,12 +220,20 @@ RtpTransportControllerAdapter::GetTransports() const {
   return transport_proxies_;
 }
 
-RTCError RtpTransportControllerAdapter::SetRtcpParameters(
-    const RtcpParameters& parameters,
+RTCError RtpTransportControllerAdapter::SetRtpTransportAndParameters(
+    const RtpTransportParameters& parameters,
     RtpTransportInterface* inner_transport) {
+  if ((video_channel_ != nullptr || voice_channel_ != nullptr) &&
+      (parameters.keepalive != keepalive_)) {
+    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
+                         "Cannot change keep-alive settings after creating "
+                         "media streams.");

[Taylor Brandstetter, 2017/08/03 16:50:36]

nit: Someone using this API may interpret "media stream" as referring to a
MediaStream object, which isn't what this is talking about. "Stream" is a pretty
ambiguous term for us, since it has different meanings in the contexts of
WebRTC, SDP and RTP.

Can you rephrase this message in the terminology used by this API? Which I think
would be "RtpSender has started sending"?

+  }
+  RTC_DCHECK(call_->SetRtpKeepAliveConfig(parameters.keepalive));

[Taylor Brandstetter, 2017/08/03 01:18:03]

If this is in a DCHECK, it will only be executed in debug builds, right?

[sprang_webrtc, 2017/08/03 13:08:14]

Oops, right. Let's do a check instead then.

   do {
     if (inner_transport == inner_audio_transport_) {
-      CopyRtcpParametersToDescriptions(parameters, &local_audio_description_,
+      CopyRtcpParametersToDescriptions(parameters.rtcp,
+                                       &local_audio_description_,
                                        &remote_audio_description_);
       if (!voice_channel_->SetLocalContent(&local_audio_description_,
                                            cricket::CA_OFFER, nullptr)) {
@@ -235,7 +244,8 @@ RTCError RtpTransportControllerAdapter::SetRtcpParameters(
         break;
       }
     } else if (inner_transport == inner_video_transport_) {
-      CopyRtcpParametersToDescriptions(parameters, &local_video_description_,
+      CopyRtcpParametersToDescriptions(parameters.rtcp,
+                                       &local_video_description_,
                                        &remote_video_description_);
       if (!video_channel_->SetLocalContent(&local_video_description_,
                                            cricket::CA_OFFER, nullptr)) {
@@ -270,7 +280,7 @@ RTCError RtpTransportControllerAdapter::ValidateAndApplyAudioSenderParameters(
   }
 
   auto stream_params_result = MakeSendStreamParamsVec(
-      parameters.encodings, inner_audio_transport_->GetRtcpParameters().cname,
+      parameters.encodings, inner_audio_transport_->GetParameters().rtcp.cname,
       local_audio_description_);
   if (!stream_params_result.ok()) {
     return stream_params_result.MoveError();
@@ -359,7 +369,7 @@ RTCError RtpTransportControllerAdapter::ValidateAndApplyVideoSenderParameters(
   }
 
   auto stream_params_result = MakeSendStreamParamsVec(
-      parameters.encodings, inner_video_transport_->GetRtcpParameters().cname,
+      parameters.encodings, inner_video_transport_->GetParameters().rtcp.cname,
       local_video_description_);
   if (!stream_params_result.ok()) {
     return stream_params_result.MoveError();
@@ -582,6 +592,7 @@ RTCError RtpTransportControllerAdapter::ValidateAndApplyVideoReceiverParameters(
 
 RtpTransportControllerAdapter::RtpTransportControllerAdapter(
     const cricket::MediaConfig& config,
+    const RtpTransportParameters& params,
     cricket::ChannelManager* channel_manager,
     webrtc::RtcEventLog* event_log,
     rtc::Thread* signaling_thread,
@@ -589,6 +600,7 @@ RtpTransportControllerAdapter::RtpTransportControllerAdapter(
     : signaling_thread_(signaling_thread),
       worker_thread_(worker_thread),
       media_config_(config),
+      keepalive_(params.keepalive),
       channel_manager_(channel_manager),
       event_log_(event_log) {
   RTC_DCHECK_RUN_ON(signaling_thread_);
@@ -607,12 +619,13 @@ RtpTransportControllerAdapter::RtpTransportControllerAdapter(
 
   worker_thread_->Invoke<void>(
       RTC_FROM_HERE,
-      rtc::Bind(&RtpTransportControllerAdapter::Init_w, this));
+      rtc::Bind(&RtpTransportControllerAdapter::Init_w, this, params));
 }
 
 // TODO(nisse): Duplicates corresponding method in PeerConnection (used
 // to be in MediaController).
-void RtpTransportControllerAdapter::Init_w() {
+void RtpTransportControllerAdapter::Init_w(
+    const RtpTransportParameters& params) {
   RTC_DCHECK(worker_thread_->IsCurrent());
   RTC_DCHECK(!call_);
 
@@ -625,6 +638,7 @@ void RtpTransportControllerAdapter::Init_w() {
   call_config.bitrate_config.min_bitrate_bps = kMinBandwidthBps;
   call_config.bitrate_config.start_bitrate_bps = kStartBandwidthBps;
   call_config.bitrate_config.max_bitrate_bps = kMaxBandwidthBps;
+  call_config.keepalive_config = params.keepalive;
 
   call_.reset(webrtc::Call::Create(call_config));
 }
@@ -656,7 +670,7 @@ RTCError RtpTransportControllerAdapter::AttachAudioSender(
   // If setting new transport, extract its RTCP parameters and create voice
   // channel.
   if (!inner_audio_transport_) {
-    CopyRtcpParametersToDescriptions(inner_transport->GetRtcpParameters(),
+    CopyRtcpParametersToDescriptions(inner_transport->GetParameters().rtcp,
                                      &local_audio_description_,
                                      &remote_audio_description_);
     inner_audio_transport_ = inner_transport;
@@ -691,7 +705,7 @@ RTCError RtpTransportControllerAdapter::AttachVideoSender(
   // If setting new transport, extract its RTCP parameters and create video
   // channel.
   if (!inner_video_transport_) {
-    CopyRtcpParametersToDescriptions(inner_transport->GetRtcpParameters(),
+    CopyRtcpParametersToDescriptions(inner_transport->GetParameters().rtcp,
                                      &local_video_description_,
                                      &remote_video_description_);
     inner_video_transport_ = inner_transport;
@@ -726,7 +740,7 @@ RTCError RtpTransportControllerAdapter::AttachAudioReceiver(
   // If setting new transport, extract its RTCP parameters and create voice
   // channel.
   if (!inner_audio_transport_) {
-    CopyRtcpParametersToDescriptions(inner_transport->GetRtcpParameters(),
+    CopyRtcpParametersToDescriptions(inner_transport->GetParameters().rtcp,
                                      &local_audio_description_,
                                      &remote_audio_description_);
     inner_audio_transport_ = inner_transport;
@@ -761,7 +775,7 @@ RTCError RtpTransportControllerAdapter::AttachVideoReceiver(
   // If setting new transport, extract its RTCP parameters and create video
   // channel.
   if (!inner_video_transport_) {
-    CopyRtcpParametersToDescriptions(inner_transport->GetRtcpParameters(),
+    CopyRtcpParametersToDescriptions(inner_transport->GetParameters().rtcp,
                                      &local_video_description_,
                                      &remote_video_description_);
     inner_video_transport_ = inner_transport;