Index: webrtc/ortc/rtptransport_unittest.cc |
diff --git a/webrtc/ortc/rtptransport_unittest.cc b/webrtc/ortc/rtptransport_unittest.cc |
new file mode 100644 |
index 0000000000000000000000000000000000000000..6d162301beee8057e758697e208c20ca57e55136 |
--- /dev/null |
+++ b/webrtc/ortc/rtptransport_unittest.cc |
@@ -0,0 +1,227 @@ |
+/* |
+ * Copyright 2017 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 <memory> |
+ |
+#include "webrtc/base/gunit.h" |
+#include "webrtc/media/base/fakemediaengine.h" |
+#include "webrtc/ortc/ortcfactory.h" |
+#include "webrtc/ortc/testrtpparameters.h" |
+#include "webrtc/p2p/base/fakepackettransport.h" |
+ |
+namespace webrtc { |
+ |
+// This test uses fake packet transports and a fake media engine, in order to |
+// test the RtpTransport at only an API level. Any end-to-end test should go in |
+// ortcfactory_integrationtest.cc instead. |
+class RtpTransportTest : public testing::Test { |
+ public: |
+ RtpTransportTest() { |
+ fake_media_engine_ = new cricket::FakeMediaEngine(); |
+ // Note: This doesn't need to use fake network classes, since it uses |
+ // FakePacketTransports. |
+ auto result = OrtcFactory::Create( |
+ nullptr, nullptr, nullptr, nullptr, nullptr, |
+ std::unique_ptr<cricket::MediaEngineInterface>(fake_media_engine_)); |
+ ortc_factory_ = result.MoveValue(); |
+ } |
+ |
+ protected: |
+ // Owned by |ortc_factory_|. |
+ cricket::FakeMediaEngine* fake_media_engine_; |
+ std::unique_ptr<OrtcFactoryInterface> ortc_factory_; |
+}; |
+ |
+// Test GetRtpPacketTransport and GetRtcpPacketTransport, with and without RTCP |
+// muxing. |
+TEST_F(RtpTransportTest, GetPacketTransports) { |
+ rtc::FakePacketTransport rtp("rtp"); |
+ rtc::FakePacketTransport rtcp("rtcp"); |
+ // With muxed RTCP. |
+ RtcpParameters rtcp_parameters; |
+ rtcp_parameters.mux = true; |
+ auto result = ortc_factory_->CreateRtpTransport(rtcp_parameters, &rtp, |
+ nullptr, nullptr); |
+ ASSERT_TRUE(result.ok()); |
+ EXPECT_EQ(&rtp, result.value()->GetRtpPacketTransport()); |
+ EXPECT_EQ(nullptr, result.value()->GetRtcpPacketTransport()); |
+ result.MoveValue().reset(); |
+ // With non-muxed RTCP. |
+ rtcp_parameters.mux = false; |
+ result = |
+ ortc_factory_->CreateRtpTransport(rtcp_parameters, &rtp, &rtcp, nullptr); |
+ ASSERT_TRUE(result.ok()); |
+ EXPECT_EQ(&rtp, result.value()->GetRtpPacketTransport()); |
+ EXPECT_EQ(&rtcp, result.value()->GetRtcpPacketTransport()); |
+} |
+ |
+// If an RtpTransport starts out un-muxed and then starts muxing, the RTCP |
+// packet transport should be forgotten and GetRtcpPacketTransport should |
+// return null. |
+TEST_F(RtpTransportTest, EnablingRtcpMuxingUnsetsRtcpTransport) { |
+ rtc::FakePacketTransport rtp("rtp"); |
+ rtc::FakePacketTransport rtcp("rtcp"); |
+ |
+ // Create non-muxed. |
+ RtcpParameters rtcp_parameters; |
+ rtcp_parameters.mux = false; |
+ auto result = |
+ ortc_factory_->CreateRtpTransport(rtcp_parameters, &rtp, &rtcp, nullptr); |
+ ASSERT_TRUE(result.ok()); |
+ auto rtp_transport = result.MoveValue(); |
+ |
+ // Enable muxing. |
+ rtcp_parameters.mux = true; |
+ EXPECT_TRUE(rtp_transport->SetRtcpParameters(rtcp_parameters).ok()); |
+ EXPECT_EQ(nullptr, rtp_transport->GetRtcpPacketTransport()); |
+} |
+ |
+TEST_F(RtpTransportTest, GetAndSetRtcpParameters) { |
+ rtc::FakePacketTransport rtp("rtp"); |
+ rtc::FakePacketTransport rtcp("rtcp"); |
+ // Start with non-muxed RTCP. |
+ RtcpParameters rtcp_parameters; |
+ rtcp_parameters.mux = false; |
+ rtcp_parameters.cname = "teST"; |
+ rtcp_parameters.reduced_size = false; |
+ auto result = |
+ ortc_factory_->CreateRtpTransport(rtcp_parameters, &rtp, &rtcp, nullptr); |
+ ASSERT_TRUE(result.ok()); |
+ auto transport = result.MoveValue(); |
+ EXPECT_EQ(rtcp_parameters, transport->GetRtcpParameters()); |
+ |
+ // Changing the CNAME is currently unsupported. |
+ rtcp_parameters.cname = "different"; |
+ EXPECT_EQ(RTCErrorType::UNSUPPORTED_OPERATION, |
+ transport->SetRtcpParameters(rtcp_parameters).type()); |
+ rtcp_parameters.cname = "teST"; |
+ |
+ // Enable RTCP muxing and reduced-size RTCP. |
+ rtcp_parameters.mux = true; |
+ rtcp_parameters.reduced_size = true; |
+ EXPECT_TRUE(transport->SetRtcpParameters(rtcp_parameters).ok()); |
+ EXPECT_EQ(rtcp_parameters, transport->GetRtcpParameters()); |
+ |
+ // Empty CNAME should result in the existing CNAME being used. |
+ rtcp_parameters.cname.clear(); |
+ EXPECT_TRUE(transport->SetRtcpParameters(rtcp_parameters).ok()); |
+ EXPECT_EQ("teST", transport->GetRtcpParameters().cname); |
+ |
+ // Disabling RTCP muxing after enabling shouldn't be allowed, since enabling |
+ // muxing should have made the RTP transport forget about the RTCP packet |
+ // transport initially passed into it. |
+ rtcp_parameters.mux = false; |
+ EXPECT_EQ(RTCErrorType::INVALID_STATE, |
+ transport->SetRtcpParameters(rtcp_parameters).type()); |
+} |
+ |
+// When Send or Receive is called on a sender or receiver, the RTCP parameters |
+// from the RtpTransport underneath the sender should be applied to the created |
+// media stream. The only relevant parameters (currently) are |cname| and |
+// |reduced_size|. |
+TEST_F(RtpTransportTest, SendAndReceiveApplyRtcpParametersToMediaEngine) { |
+ // First, create video transport with reduced-size RTCP. |
+ rtc::FakePacketTransport fake_packet_transport1("1"); |
+ RtcpParameters rtcp_parameters; |
+ rtcp_parameters.mux = true; |
+ rtcp_parameters.reduced_size = true; |
+ rtcp_parameters.cname = "foo"; |
+ auto rtp_transport_result = ortc_factory_->CreateRtpTransport( |
+ rtcp_parameters, &fake_packet_transport1, nullptr, nullptr); |
+ auto video_transport = rtp_transport_result.MoveValue(); |
+ |
+ // Create video sender and call Send, expecting parameters to be applied. |
+ auto sender_result = ortc_factory_->CreateRtpSender(cricket::MEDIA_TYPE_VIDEO, |
+ video_transport.get()); |
+ auto video_sender = sender_result.MoveValue(); |
+ EXPECT_TRUE(video_sender->Send(MakeMinimalVp8Parameters()).ok()); |
+ cricket::FakeVideoMediaChannel* fake_video_channel = |
+ fake_media_engine_->GetVideoChannel(0); |
+ ASSERT_NE(nullptr, fake_video_channel); |
+ EXPECT_TRUE(fake_video_channel->send_rtcp_parameters().reduced_size); |
+ ASSERT_EQ(1u, fake_video_channel->send_streams().size()); |
+ const cricket::StreamParams& video_send_stream = |
+ fake_video_channel->send_streams()[0]; |
+ EXPECT_EQ("foo", video_send_stream.cname); |
+ |
+ // Create video receiver and call Receive, expecting parameters to be applied |
+ // (minus |cname|, since that's the sent cname, not received). |
+ auto receiver_result = ortc_factory_->CreateRtpReceiver( |
+ cricket::MEDIA_TYPE_VIDEO, video_transport.get()); |
+ auto video_receiver = receiver_result.MoveValue(); |
+ EXPECT_TRUE( |
+ video_receiver->Receive(MakeMinimalVp8ParametersWithSsrc(0xdeadbeef)) |
+ .ok()); |
+ EXPECT_TRUE(fake_video_channel->recv_rtcp_parameters().reduced_size); |
+ |
+ // Create audio transport with non-reduced size RTCP. |
+ rtc::FakePacketTransport fake_packet_transport2("2"); |
+ rtcp_parameters.reduced_size = false; |
+ rtcp_parameters.cname = "bar"; |
+ rtp_transport_result = ortc_factory_->CreateRtpTransport( |
+ rtcp_parameters, &fake_packet_transport2, nullptr, nullptr); |
+ auto audio_transport = rtp_transport_result.MoveValue(); |
+ |
+ // Create audio sender and call Send, expecting parameters to be applied. |
+ sender_result = ortc_factory_->CreateRtpSender(cricket::MEDIA_TYPE_AUDIO, |
+ audio_transport.get()); |
+ auto audio_sender = sender_result.MoveValue(); |
+ EXPECT_TRUE(audio_sender->Send(MakeMinimalIsacParameters()).ok()); |
+ |
+ cricket::FakeVoiceMediaChannel* fake_voice_channel = |
+ fake_media_engine_->GetVoiceChannel(0); |
+ ASSERT_NE(nullptr, fake_voice_channel); |
+ EXPECT_FALSE(fake_voice_channel->send_rtcp_parameters().reduced_size); |
+ ASSERT_EQ(1u, fake_voice_channel->send_streams().size()); |
+ const cricket::StreamParams& audio_send_stream = |
+ fake_voice_channel->send_streams()[0]; |
+ EXPECT_EQ("bar", audio_send_stream.cname); |
+ |
+ // Create audio receiver and call Receive, expecting parameters to be applied |
+ // (minus |cname|, since that's the sent cname, not received). |
+ receiver_result = ortc_factory_->CreateRtpReceiver(cricket::MEDIA_TYPE_AUDIO, |
+ audio_transport.get()); |
+ auto audio_receiver = receiver_result.MoveValue(); |
+ EXPECT_TRUE( |
+ audio_receiver->Receive(MakeMinimalOpusParametersWithSsrc(0xbaadf00d)) |
+ .ok()); |
+ EXPECT_FALSE(fake_voice_channel->recv_rtcp_parameters().reduced_size); |
+} |
+ |
+// When SetRtcpParameters is called, the modified parameters should be applied |
+// to the media engine. |
+// TODO(deadbeef): Once the implementation supports changing the CNAME, |
+// test that here. |
+TEST_F(RtpTransportTest, SetRtcpParametersAppliesParametersToMediaEngine) { |
+ rtc::FakePacketTransport fake_packet_transport("fake"); |
+ RtcpParameters rtcp_parameters; |
+ rtcp_parameters.mux = true; |
+ rtcp_parameters.reduced_size = false; |
+ auto rtp_transport_result = ortc_factory_->CreateRtpTransport( |
+ rtcp_parameters, &fake_packet_transport, nullptr, nullptr); |
+ auto rtp_transport = rtp_transport_result.MoveValue(); |
+ |
+ // Create video sender and call Send, applying an initial set of parameters. |
+ auto sender_result = ortc_factory_->CreateRtpSender(cricket::MEDIA_TYPE_VIDEO, |
+ rtp_transport.get()); |
+ auto sender = sender_result.MoveValue(); |
+ EXPECT_TRUE(sender->Send(MakeMinimalVp8Parameters()).ok()); |
+ |
+ // Modify parameters and expect them to be changed at the media engine level. |
+ rtcp_parameters.reduced_size = true; |
+ EXPECT_TRUE(rtp_transport->SetRtcpParameters(rtcp_parameters).ok()); |
+ |
+ cricket::FakeVideoMediaChannel* fake_video_channel = |
+ fake_media_engine_->GetVideoChannel(0); |
+ ASSERT_NE(nullptr, fake_video_channel); |
+ EXPECT_TRUE(fake_video_channel->send_rtcp_parameters().reduced_size); |
+} |
+ |
+} // namespace webrtc |