Implement RTP keepalive in native stack.

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
@@ skipping 14 matching lines @@
 #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/system_wrappers/include/sleep.h"
 #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"
 #include "webrtc/video_send_stream.h"
 
 namespace webrtc {
 
 enum VideoFormat { kGeneric, kVP8, };
 
 void ExpectEqualFramesVector(const std::vector<VideoFrame>& frames1,
@@ skipping 3348 matching lines @@
     Call* call_;
     rtc::CriticalSection crit_;
     uint32_t max_bitrate_bps_ GUARDED_BY(&crit_);
     bool first_packet_sent_ GUARDED_BY(&crit_);
     rtc::Event bitrate_changed_event_;
   } test;
 
   RunBaseTest(&test);
 }
 
+TEST_F(VideoSendStreamTest, SendsKeepAlive) {
+  const int kTimeoutMs = 50;  // Really short timeout for testing.
+  const int kPayloadType = 20;
+
+  class KeepaliveObserver : public test::SendTest {
+   public:
+    KeepaliveObserver() : 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 started. Stop it now.
+        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