Wire up RTP keep-alive in ortc api.

webrtc/video/video_send_stream_tests.cc

 /*
  *  Copyright (c) 2013 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.
  */
 #include <algorithm>  // max
 #include <memory>
 #include <vector>
 
 #include "webrtc/call/call.h"
+#include "webrtc/call/rtp_transport_controller_send.h"
 #include "webrtc/common_video/include/frame_callback.h"
 #include "webrtc/common_video/include/video_frame.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
 #include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h"
 #include "webrtc/rtc_base/bind.h"
 #include "webrtc/rtc_base/checks.h"
@@ skipping 3347 matching lines @@
   RunBaseTest(&test);
 }
 
 TEST_F(VideoSendStreamTest, SendsKeepAlive) {
   const int kTimeoutMs = 50;  // Really short timeout for testing.
 
   class KeepaliveObserver : public test::SendTest {
    public:
     KeepaliveObserver() : SendTest(kDefaultTimeoutMs) {}
 
-    Call::Config GetSenderCallConfig() override {
-      Call::Config config = SendTest::GetSenderCallConfig();
-      config.keepalive_config.timeout_interval_ms = kTimeoutMs;
-      config.keepalive_config.payload_type =
-          CallTest::kDefaultKeepalivePayloadType;
-      return config;
+    void OnRtpTransportControllerSendCreated(
+        RtpTransportControllerSend* controller) override {
+      RtpKeepAliveConfig config;
+      config.timeout_interval_ms = kTimeoutMs;
+      config.payload_type = CallTest::kDefaultKeepalivePayloadType;
+      controller->SetKeepAliveConfig(config);
     }
 
    private:
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
       RTPHeader header;
       EXPECT_TRUE(parser_->Parse(packet, length, &header));
 
       if (header.payloadType != CallTest::kDefaultKeepalivePayloadType) {
         // The video stream has started. Stop it now.
         if (capturer_)
@@ skipping 14 matching lines @@
       capturer_ = frame_generator_capturer;
     }
 
     test::FrameGeneratorCapturer* capturer_ = nullptr;
   } test;
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc