Wire up RTP keep-alive in ortc api.

webrtc/call/call_unittest.cc

 /*
  *  Copyright (c) 2015 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 <list>
 #include <map>
 #include <memory>
 #include <utility>
 
 #include "webrtc/api/test/mock_audio_mixer.h"
 #include "webrtc/call/audio_state.h"
 #include "webrtc/call/call.h"
 #include "webrtc/call/fake_rtp_transport_controller_send.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/modules/audio_device/include/mock_audio_device.h"
 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
 #include "webrtc/modules/congestion_controller/include/mock/mock_send_side_congestion_controller.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/rtc_base/ptr_util.h"
+#include "webrtc/test/fake_encoder.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/mock_audio_decoder_factory.h"
 #include "webrtc/test/mock_transport.h"
 #include "webrtc/test/mock_voice_engine.h"
 
 namespace {
 
 struct CallHelper {
   explicit CallHelper(
       rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory = nullptr)
@@ skipping 275 matching lines @@
   config.remote_ssrc = 5548;
   stream = call->CreateFlexfecReceiveStream(config);
   EXPECT_NE(stream, nullptr);
   streams.push_back(stream);
 
   for (auto s : streams) {
     call->DestroyFlexfecReceiveStream(s);
   }
 }
 
+TEST(CallTest, DontAllowKeepaliveReconfiguration) {
+  CallHelper call;
+  webrtc::RtpKeepAliveConfig keep_alive_config;
+
+  // Set default values, shouldn't be a problem.
+  EXPECT_TRUE(call->SetRtpKeepAliveConfig(keep_alive_config));
+
+  // Active keep-alive by setting a non-zero timeout.
+  keep_alive_config.timeout_interval_ms = 15000;
+  EXPECT_TRUE(call->SetRtpKeepAliveConfig(keep_alive_config));
+
+  // Create a minimal send stream.
+  MockTransport transport;
+  VideoSendStream::Config send_stream_config(&transport);
+  test::FakeEncoder fake_encoder(Clock::GetRealTimeClock());
+  send_stream_config.encoder_settings.payload_name = "VP8";
+  send_stream_config.encoder_settings.payload_type = 100;
+  send_stream_config.encoder_settings.encoder = &fake_encoder;
+  send_stream_config.rtp.ssrcs.push_back(1234);
+  VideoEncoderConfig encoder_config;
+  VideoSendStream* const stream = call->CreateVideoSendStream(
+      std::move(send_stream_config), encoder_config.Copy());
+
+  // Setting the keep-alive with the current values is OK (ignored).
+  EXPECT_TRUE(call->SetRtpKeepAliveConfig(keep_alive_config));
+
+  // Updating to a new interval is disallowed.
+  keep_alive_config.timeout_interval_ms = 1000;
+  EXPECT_FALSE(call->SetRtpKeepAliveConfig(keep_alive_config));
+
+  // Destroyg the video send stream, with active streams it's again OK to set
+  // a new keep-alive config.
+  call->DestroyVideoSendStream(stream);
+  EXPECT_TRUE(call->SetRtpKeepAliveConfig(keep_alive_config));
+}
+
 namespace {
 struct CallBitrateHelper {
   CallBitrateHelper() : CallBitrateHelper(Call::Config::BitrateConfig()) {}
 
   explicit CallBitrateHelper(const Call::Config::BitrateConfig& bitrate_config)
       : mock_cc_(Clock::GetRealTimeClock(), &event_log_, &packet_router_) {
     Call::Config config(&event_log_);
     config.bitrate_config = bitrate_config;
     call_.reset(
         Call::Create(config, rtc::MakeUnique<FakeRtpTransportControllerSend>(
@@ skipping 375 matching lines @@
   mask.min_bitrate_bps = rtc::Optional<int>(2000);
   EXPECT_CALL(call.mock_cc(), SetBweBitrates(1000, -1, 1000));
   call->SetBitrateConfigMask(mask);
 
   // Set min to 3000; the clamped value stays the same so nothing happens.
   mask.min_bitrate_bps = rtc::Optional<int>(3000);
   call->SetBitrateConfigMask(mask);
 }
 
 }  // namespace webrtc