Implement RTP keepalive in native stack.

webrtc/video/video_send_stream_tests.cc

Index: webrtc/video/video_send_stream_tests.cc
diff --git a/webrtc/video/video_send_stream_tests.cc b/webrtc/video/video_send_stream_tests.cc
index 37a0249e377e1066a5cc8f3266be39b2ec329457..7cb0d9bf53814aebeda50586d684c43e4d52ae69 100644
--- a/webrtc/video/video_send_stream_tests.cc
+++ b/webrtc/video/video_send_stream_tests.cc
@@ -32,13 +32,14 @@
 #include "webrtc/test/call_test.h"
 #include "webrtc/test/configurable_frame_size_encoder.h"
 #include "webrtc/test/fake_texture_frame.h"
+#include "webrtc/test/field_trial.h"
 #include "webrtc/test/frame_generator.h"
+#include "webrtc/test/frame_generator_capturer.h"
 #include "webrtc/test/frame_utils.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/null_transport.h"
 #include "webrtc/test/rtcp_packet_parser.h"
 #include "webrtc/test/testsupport/perf_test.h"
-#include "webrtc/test/field_trial.h"
 
 #include "webrtc/video/send_statistics_proxy.h"
 #include "webrtc/video/transport_adapter.h"
@@ -3407,4 +3408,51 @@ TEST_F(VideoSendStreamTest, RemoveOverheadFromBandwidth) {
   RunBaseTest(&test);
 }
 
+TEST_F(VideoSendStreamTest, SendsKeepAlive) {
+  const int kTimeoutMs = 50;  // Really short timeout for testing.
+  const int kPayloadType = 20;
+
+  class CNameObserver : public test::SendTest {

[åsapersson, 2017/06/30 15:09:13]

rename class

[sprang_webrtc, 2017/06/30 16:21:00]

Done.

+   public:
+    CNameObserver() : SendTest(kDefaultTimeoutMs) {}
+
+   private:
+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
+      RTPHeader header;
+      EXPECT_TRUE(parser_->Parse(packet, length, &header));
+
+      if (header.payloadType != kPayloadType) {
+        // The video stream has start. Stop it now. But only once.

[åsapersson, 2017/06/30 15:09:13]

But only once?

[sprang_webrtc, 2017/06/30 16:21:00]

Missed cleanup. Removed.

+        if (capturer_)
+          capturer_->Stop();
+      } else {
+        observation_complete_.Set();
+      }
+
+      return SEND_PACKET;
+    }
+
+    void ModifyVideoConfigs(
+        VideoSendStream::Config* send_config,
+        std::vector<VideoReceiveStream::Config>* receive_configs,
+        VideoEncoderConfig* encoder_config) override {
+      send_config->rtp.keep_alive.timeout_interval_ms = kTimeoutMs;
+      send_config->rtp.keep_alive.payload_type = kPayloadType;
+    }
+
+    void PerformTest() override {
+      EXPECT_TRUE(Wait()) << "Timed out while waiting for keep-alive packet.";
+    }
+
+    void OnFrameGeneratorCapturerCreated(
+        test::FrameGeneratorCapturer* frame_generator_capturer) override {
+      capturer_ = frame_generator_capturer;
+    }
+
+    test::FrameGeneratorCapturer* capturer_ = nullptr;
+  } test;
+
+  RunBaseTest(&test);
+}
+
 }  // namespace webrtc