| /* | 
 |  *  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 "call/call.h" | 
 |  | 
 | #include <cstdint> | 
 | #include <list> | 
 | #include <memory> | 
 | #include <string> | 
 | #include <utility> | 
 | #include <vector> | 
 |  | 
 | #include "absl/strings/string_view.h" | 
 | #include "api/adaptation/resource.h" | 
 | #include "api/environment/environment.h" | 
 | #include "api/environment/environment_factory.h" | 
 | #include "api/make_ref_counted.h" | 
 | #include "api/media_types.h" | 
 | #include "api/scoped_refptr.h" | 
 | #include "api/test/mock_audio_mixer.h" | 
 | #include "api/test/video/function_video_encoder_factory.h" | 
 | #include "api/units/timestamp.h" | 
 | #include "api/video/builtin_video_bitrate_allocator_factory.h" | 
 | #include "api/video_codecs/sdp_video_format.h" | 
 | #include "audio/audio_send_stream.h" | 
 | #include "call/adaptation/test/fake_resource.h" | 
 | #include "call/adaptation/test/mock_resource_listener.h" | 
 | #include "call/audio_receive_stream.h" | 
 | #include "call/audio_send_stream.h" | 
 | #include "call/audio_state.h" | 
 | #include "call/call_config.h" | 
 | #include "call/flexfec_receive_stream.h" | 
 | #include "call/video_send_stream.h" | 
 | #include "modules/audio_device/include/mock_audio_device.h" | 
 | #include "modules/audio_processing/include/mock_audio_processing.h" | 
 | #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" | 
 | #include "modules/rtp_rtcp/source/rtp_packet_received.h" | 
 | #include "test/fake_encoder.h" | 
 | #include "test/gmock.h" | 
 | #include "test/gtest.h" | 
 | #include "test/mock_audio_decoder_factory.h" | 
 | #include "test/mock_transport.h" | 
 | #include "test/run_loop.h" | 
 | #include "video/config/video_encoder_config.h" | 
 |  | 
 | namespace webrtc { | 
 | namespace { | 
 |  | 
 | using test::FakeEncoder; | 
 | using test::FunctionVideoEncoderFactory; | 
 | using test::MockAudioDeviceModule; | 
 | using test::MockAudioMixer; | 
 | using test::MockAudioProcessing; | 
 | using test::RunLoop; | 
 | using ::testing::_; | 
 | using ::testing::MockFunction; | 
 | using ::testing::NiceMock; | 
 | using ::testing::StrictMock; | 
 |  | 
 | struct CallHelper { | 
 |   explicit CallHelper(bool use_null_audio_processing) { | 
 |     AudioState::Config audio_state_config; | 
 |     audio_state_config.audio_mixer = make_ref_counted<MockAudioMixer>(); | 
 |     audio_state_config.audio_processing = | 
 |         use_null_audio_processing | 
 |             ? nullptr | 
 |             : make_ref_counted<NiceMock<MockAudioProcessing>>(); | 
 |     audio_state_config.audio_device_module = | 
 |         make_ref_counted<MockAudioDeviceModule>(); | 
 |     CallConfig config(CreateEnvironment()); | 
 |     config.audio_state = AudioState::Create(audio_state_config); | 
 |     call_ = Call::Create(std::move(config)); | 
 |   } | 
 |  | 
 |   Call* operator->() { return call_.get(); } | 
 |  | 
 |  private: | 
 |   RunLoop loop_; | 
 |   std::unique_ptr<Call> call_; | 
 | }; | 
 |  | 
 | scoped_refptr<Resource> FindResourceWhoseNameContains( | 
 |     const std::vector<scoped_refptr<Resource>>& resources, | 
 |     absl::string_view name_contains) { | 
 |   for (const auto& resource : resources) { | 
 |     if (resource->Name().find(std::string(name_contains)) != std::string::npos) | 
 |       return resource; | 
 |   } | 
 |   return nullptr; | 
 | } | 
 |  | 
 | }  // namespace | 
 |  | 
 | TEST(CallTest, ConstructDestruct) { | 
 |   for (bool use_null_audio_processing : {false, true}) { | 
 |     CallHelper call(use_null_audio_processing); | 
 |   } | 
 | } | 
 |  | 
 | TEST(CallTest, CreateDestroy_AudioSendStream) { | 
 |   for (bool use_null_audio_processing : {false, true}) { | 
 |     CallHelper call(use_null_audio_processing); | 
 |     MockTransport send_transport; | 
 |     AudioSendStream::Config config(&send_transport); | 
 |     config.rtp.ssrc = 42; | 
 |     AudioSendStream* stream = call->CreateAudioSendStream(config); | 
 |     EXPECT_NE(stream, nullptr); | 
 |     call->DestroyAudioSendStream(stream); | 
 |   } | 
 | } | 
 |  | 
 | TEST(CallTest, CreateDestroy_AudioReceiveStream) { | 
 |   for (bool use_null_audio_processing : {false, true}) { | 
 |     CallHelper call(use_null_audio_processing); | 
 |     AudioReceiveStreamInterface::Config config; | 
 |     MockTransport rtcp_send_transport; | 
 |     config.rtp.remote_ssrc = 42; | 
 |     config.rtcp_send_transport = &rtcp_send_transport; | 
 |     config.decoder_factory = make_ref_counted<MockAudioDecoderFactory>(); | 
 |     AudioReceiveStreamInterface* stream = | 
 |         call->CreateAudioReceiveStream(config); | 
 |     EXPECT_NE(stream, nullptr); | 
 |     call->DestroyAudioReceiveStream(stream); | 
 |   } | 
 | } | 
 |  | 
 | TEST(CallTest, CreateDestroy_AudioSendStreams) { | 
 |   for (bool use_null_audio_processing : {false, true}) { | 
 |     CallHelper call(use_null_audio_processing); | 
 |     MockTransport send_transport; | 
 |     AudioSendStream::Config config(&send_transport); | 
 |     std::list<AudioSendStream*> streams; | 
 |     for (int i = 0; i < 2; ++i) { | 
 |       for (uint32_t ssrc = 0; ssrc < 1234567; ssrc += 34567) { | 
 |         config.rtp.ssrc = ssrc; | 
 |         AudioSendStream* stream = call->CreateAudioSendStream(config); | 
 |         EXPECT_NE(stream, nullptr); | 
 |         if (ssrc & 1) { | 
 |           streams.push_back(stream); | 
 |         } else { | 
 |           streams.push_front(stream); | 
 |         } | 
 |       } | 
 |       for (auto s : streams) { | 
 |         call->DestroyAudioSendStream(s); | 
 |       } | 
 |       streams.clear(); | 
 |     } | 
 |   } | 
 | } | 
 |  | 
 | TEST(CallTest, CreateDestroy_AudioReceiveStreams) { | 
 |   for (bool use_null_audio_processing : {false, true}) { | 
 |     CallHelper call(use_null_audio_processing); | 
 |     AudioReceiveStreamInterface::Config config; | 
 |     MockTransport rtcp_send_transport; | 
 |     config.rtcp_send_transport = &rtcp_send_transport; | 
 |     config.decoder_factory = make_ref_counted<MockAudioDecoderFactory>(); | 
 |     std::list<AudioReceiveStreamInterface*> streams; | 
 |     for (int i = 0; i < 2; ++i) { | 
 |       for (uint32_t ssrc = 0; ssrc < 1234567; ssrc += 34567) { | 
 |         config.rtp.remote_ssrc = ssrc; | 
 |         AudioReceiveStreamInterface* stream = | 
 |             call->CreateAudioReceiveStream(config); | 
 |         EXPECT_NE(stream, nullptr); | 
 |         if (ssrc & 1) { | 
 |           streams.push_back(stream); | 
 |         } else { | 
 |           streams.push_front(stream); | 
 |         } | 
 |       } | 
 |       for (auto s : streams) { | 
 |         call->DestroyAudioReceiveStream(s); | 
 |       } | 
 |       streams.clear(); | 
 |     } | 
 |   } | 
 | } | 
 |  | 
 | TEST(CallTest, CreateDestroy_FlexfecReceiveStream) { | 
 |   for (bool use_null_audio_processing : {false, true}) { | 
 |     CallHelper call(use_null_audio_processing); | 
 |     MockTransport rtcp_send_transport; | 
 |     FlexfecReceiveStream::Config config(&rtcp_send_transport); | 
 |     config.payload_type = 118; | 
 |     config.rtp.remote_ssrc = 38837212; | 
 |     config.protected_media_ssrcs = {27273}; | 
 |  | 
 |     FlexfecReceiveStream* stream = call->CreateFlexfecReceiveStream(config); | 
 |     EXPECT_NE(stream, nullptr); | 
 |     call->DestroyFlexfecReceiveStream(stream); | 
 |   } | 
 | } | 
 |  | 
 | TEST(CallTest, CreateDestroy_FlexfecReceiveStreams) { | 
 |   for (bool use_null_audio_processing : {false, true}) { | 
 |     CallHelper call(use_null_audio_processing); | 
 |     MockTransport rtcp_send_transport; | 
 |     FlexfecReceiveStream::Config config(&rtcp_send_transport); | 
 |     config.payload_type = 118; | 
 |     std::list<FlexfecReceiveStream*> streams; | 
 |  | 
 |     for (int i = 0; i < 2; ++i) { | 
 |       for (uint32_t ssrc = 0; ssrc < 1234567; ssrc += 34567) { | 
 |         config.rtp.remote_ssrc = ssrc; | 
 |         config.protected_media_ssrcs = {ssrc + 1}; | 
 |         FlexfecReceiveStream* stream = call->CreateFlexfecReceiveStream(config); | 
 |         EXPECT_NE(stream, nullptr); | 
 |         if (ssrc & 1) { | 
 |           streams.push_back(stream); | 
 |         } else { | 
 |           streams.push_front(stream); | 
 |         } | 
 |       } | 
 |       for (auto s : streams) { | 
 |         call->DestroyFlexfecReceiveStream(s); | 
 |       } | 
 |       streams.clear(); | 
 |     } | 
 |   } | 
 | } | 
 |  | 
 | TEST(CallTest, MultipleFlexfecReceiveStreamsProtectingSingleVideoStream) { | 
 |   for (bool use_null_audio_processing : {false, true}) { | 
 |     CallHelper call(use_null_audio_processing); | 
 |     MockTransport rtcp_send_transport; | 
 |     FlexfecReceiveStream::Config config(&rtcp_send_transport); | 
 |     config.payload_type = 118; | 
 |     config.protected_media_ssrcs = {1324234}; | 
 |     FlexfecReceiveStream* stream; | 
 |     std::list<FlexfecReceiveStream*> streams; | 
 |  | 
 |     config.rtp.remote_ssrc = 838383; | 
 |     stream = call->CreateFlexfecReceiveStream(config); | 
 |     EXPECT_NE(stream, nullptr); | 
 |     streams.push_back(stream); | 
 |  | 
 |     config.rtp.remote_ssrc = 424993; | 
 |     stream = call->CreateFlexfecReceiveStream(config); | 
 |     EXPECT_NE(stream, nullptr); | 
 |     streams.push_back(stream); | 
 |  | 
 |     config.rtp.remote_ssrc = 99383; | 
 |     stream = call->CreateFlexfecReceiveStream(config); | 
 |     EXPECT_NE(stream, nullptr); | 
 |     streams.push_back(stream); | 
 |  | 
 |     config.rtp.remote_ssrc = 5548; | 
 |     stream = call->CreateFlexfecReceiveStream(config); | 
 |     EXPECT_NE(stream, nullptr); | 
 |     streams.push_back(stream); | 
 |  | 
 |     for (auto s : streams) { | 
 |       call->DestroyFlexfecReceiveStream(s); | 
 |     } | 
 |   } | 
 | } | 
 |  | 
 | TEST(CallTest, | 
 |      DeliverRtpPacketOfTypeAudioTriggerOnUndemuxablePacketHandlerIfNotDemuxed) { | 
 |   CallHelper call(/*use_null_audio_processing=*/false); | 
 |   MockFunction<bool(const RtpPacketReceived& parsed_packet)> | 
 |       un_demuxable_packet_handler; | 
 |  | 
 |   RtpPacketReceived packet; | 
 |   packet.set_arrival_time(Timestamp::Millis(1)); | 
 |   EXPECT_CALL(un_demuxable_packet_handler, Call); | 
 |   call->Receiver()->DeliverRtpPacket( | 
 |       MediaType::AUDIO, packet, un_demuxable_packet_handler.AsStdFunction()); | 
 | } | 
 |  | 
 | TEST(CallTest, | 
 |      DeliverRtpPacketOfTypeVideoTriggerOnUndemuxablePacketHandlerIfNotDemuxed) { | 
 |   CallHelper call(/*use_null_audio_processing=*/false); | 
 |   MockFunction<bool(const RtpPacketReceived& parsed_packet)> | 
 |       un_demuxable_packet_handler; | 
 |  | 
 |   RtpPacketReceived packet; | 
 |   packet.set_arrival_time(Timestamp::Millis(1)); | 
 |   EXPECT_CALL(un_demuxable_packet_handler, Call); | 
 |   call->Receiver()->DeliverRtpPacket( | 
 |       MediaType::VIDEO, packet, un_demuxable_packet_handler.AsStdFunction()); | 
 | } | 
 |  | 
 | TEST(CallTest, | 
 |      DeliverRtpPacketOfTypeAnyDoesNotTriggerOnUndemuxablePacketHandler) { | 
 |   CallHelper call(/*use_null_audio_processing=*/false); | 
 |   MockFunction<bool(const RtpPacketReceived& parsed_packet)> | 
 |       un_demuxable_packet_handler; | 
 |  | 
 |   RtpPacketReceived packet; | 
 |   packet.set_arrival_time(Timestamp::Millis(1)); | 
 |   EXPECT_CALL(un_demuxable_packet_handler, Call).Times(0); | 
 |   call->Receiver()->DeliverRtpPacket( | 
 |       MediaType::ANY, packet, un_demuxable_packet_handler.AsStdFunction()); | 
 | } | 
 |  | 
 | TEST(CallTest, RecreatingAudioStreamWithSameSsrcReusesRtpState) { | 
 |   constexpr uint32_t kSSRC = 12345; | 
 |   for (bool use_null_audio_processing : {false, true}) { | 
 |     CallHelper call(use_null_audio_processing); | 
 |  | 
 |     auto create_stream_and_get_rtp_state = [&](uint32_t ssrc) { | 
 |       MockTransport send_transport; | 
 |       AudioSendStream::Config config(&send_transport); | 
 |       config.rtp.ssrc = ssrc; | 
 |       AudioSendStream* stream = call->CreateAudioSendStream(config); | 
 |       const RtpState rtp_state = | 
 |           static_cast<internal::AudioSendStream*>(stream)->GetRtpState(); | 
 |       call->DestroyAudioSendStream(stream); | 
 |       return rtp_state; | 
 |     }; | 
 |  | 
 |     const RtpState rtp_state1 = create_stream_and_get_rtp_state(kSSRC); | 
 |     const RtpState rtp_state2 = create_stream_and_get_rtp_state(kSSRC); | 
 |  | 
 |     EXPECT_EQ(rtp_state1.sequence_number, rtp_state2.sequence_number); | 
 |     EXPECT_EQ(rtp_state1.start_timestamp, rtp_state2.start_timestamp); | 
 |     EXPECT_EQ(rtp_state1.timestamp, rtp_state2.timestamp); | 
 |     EXPECT_EQ(rtp_state1.capture_time, rtp_state2.capture_time); | 
 |     EXPECT_EQ(rtp_state1.last_timestamp_time, rtp_state2.last_timestamp_time); | 
 |   } | 
 | } | 
 |  | 
 | TEST(CallTest, AddAdaptationResourceAfterCreatingVideoSendStream) { | 
 |   CallHelper call(true); | 
 |   // Create a VideoSendStream. | 
 |   FunctionVideoEncoderFactory fake_encoder_factory( | 
 |       [](const Environment& env, const SdpVideoFormat& /* format */) { | 
 |         return std::make_unique<FakeEncoder>(env); | 
 |       }); | 
 |   auto bitrate_allocator_factory = CreateBuiltinVideoBitrateAllocatorFactory(); | 
 |   MockTransport send_transport; | 
 |   VideoSendStream::Config config(&send_transport); | 
 |   config.rtp.payload_type = 110; | 
 |   config.rtp.ssrcs = {42}; | 
 |   config.encoder_settings.encoder_factory = &fake_encoder_factory; | 
 |   config.encoder_settings.bitrate_allocator_factory = | 
 |       bitrate_allocator_factory.get(); | 
 |   VideoEncoderConfig encoder_config; | 
 |   encoder_config.max_bitrate_bps = 1337; | 
 |   VideoSendStream* stream1 = | 
 |       call->CreateVideoSendStream(config.Copy(), encoder_config.Copy()); | 
 |   EXPECT_NE(stream1, nullptr); | 
 |   config.rtp.ssrcs = {43}; | 
 |   VideoSendStream* stream2 = | 
 |       call->CreateVideoSendStream(config.Copy(), encoder_config.Copy()); | 
 |   EXPECT_NE(stream2, nullptr); | 
 |   // Add a fake resource. | 
 |   auto fake_resource = FakeResource::Create("FakeResource"); | 
 |   call->AddAdaptationResource(fake_resource); | 
 |   // An adapter resource mirroring the `fake_resource` should now be present on | 
 |   // both streams. | 
 |   auto injected_resource1 = FindResourceWhoseNameContains( | 
 |       stream1->GetAdaptationResources(), fake_resource->Name()); | 
 |   EXPECT_TRUE(injected_resource1); | 
 |   auto injected_resource2 = FindResourceWhoseNameContains( | 
 |       stream2->GetAdaptationResources(), fake_resource->Name()); | 
 |   EXPECT_TRUE(injected_resource2); | 
 |   // Overwrite the real resource listeners with mock ones to verify the signal | 
 |   // gets through. | 
 |   injected_resource1->SetResourceListener(nullptr); | 
 |   StrictMock<MockResourceListener> resource_listener1; | 
 |   EXPECT_CALL(resource_listener1, OnResourceUsageStateMeasured(_, _)) | 
 |       .Times(1) | 
 |       .WillOnce([injected_resource1](scoped_refptr<Resource> resource, | 
 |                                      ResourceUsageState usage_state) { | 
 |         EXPECT_EQ(injected_resource1, resource); | 
 |         EXPECT_EQ(ResourceUsageState::kOveruse, usage_state); | 
 |       }); | 
 |   injected_resource1->SetResourceListener(&resource_listener1); | 
 |   injected_resource2->SetResourceListener(nullptr); | 
 |   StrictMock<MockResourceListener> resource_listener2; | 
 |   EXPECT_CALL(resource_listener2, OnResourceUsageStateMeasured(_, _)) | 
 |       .Times(1) | 
 |       .WillOnce([injected_resource2](scoped_refptr<Resource> resource, | 
 |                                      ResourceUsageState usage_state) { | 
 |         EXPECT_EQ(injected_resource2, resource); | 
 |         EXPECT_EQ(ResourceUsageState::kOveruse, usage_state); | 
 |       }); | 
 |   injected_resource2->SetResourceListener(&resource_listener2); | 
 |   // The kOveruse signal should get to our resource listeners. | 
 |   fake_resource->SetUsageState(ResourceUsageState::kOveruse); | 
 |   call->DestroyVideoSendStream(stream1); | 
 |   call->DestroyVideoSendStream(stream2); | 
 | } | 
 |  | 
 | TEST(CallTest, AddAdaptationResourceBeforeCreatingVideoSendStream) { | 
 |   CallHelper call(true); | 
 |   // Add a fake resource. | 
 |   auto fake_resource = FakeResource::Create("FakeResource"); | 
 |   call->AddAdaptationResource(fake_resource); | 
 |   // Create a VideoSendStream. | 
 |   FunctionVideoEncoderFactory fake_encoder_factory( | 
 |       [](const Environment& env, const SdpVideoFormat& /* format */) { | 
 |         return std::make_unique<FakeEncoder>(env); | 
 |       }); | 
 |   auto bitrate_allocator_factory = CreateBuiltinVideoBitrateAllocatorFactory(); | 
 |   MockTransport send_transport; | 
 |   VideoSendStream::Config config(&send_transport); | 
 |   config.rtp.payload_type = 110; | 
 |   config.rtp.ssrcs = {42}; | 
 |   config.encoder_settings.encoder_factory = &fake_encoder_factory; | 
 |   config.encoder_settings.bitrate_allocator_factory = | 
 |       bitrate_allocator_factory.get(); | 
 |   VideoEncoderConfig encoder_config; | 
 |   encoder_config.max_bitrate_bps = 1337; | 
 |   VideoSendStream* stream1 = | 
 |       call->CreateVideoSendStream(config.Copy(), encoder_config.Copy()); | 
 |   EXPECT_NE(stream1, nullptr); | 
 |   config.rtp.ssrcs = {43}; | 
 |   VideoSendStream* stream2 = | 
 |       call->CreateVideoSendStream(config.Copy(), encoder_config.Copy()); | 
 |   EXPECT_NE(stream2, nullptr); | 
 |   // An adapter resource mirroring the `fake_resource` should be present on both | 
 |   // streams. | 
 |   auto injected_resource1 = FindResourceWhoseNameContains( | 
 |       stream1->GetAdaptationResources(), fake_resource->Name()); | 
 |   EXPECT_TRUE(injected_resource1); | 
 |   auto injected_resource2 = FindResourceWhoseNameContains( | 
 |       stream2->GetAdaptationResources(), fake_resource->Name()); | 
 |   EXPECT_TRUE(injected_resource2); | 
 |   // Overwrite the real resource listeners with mock ones to verify the signal | 
 |   // gets through. | 
 |   injected_resource1->SetResourceListener(nullptr); | 
 |   StrictMock<MockResourceListener> resource_listener1; | 
 |   EXPECT_CALL(resource_listener1, OnResourceUsageStateMeasured(_, _)) | 
 |       .Times(1) | 
 |       .WillOnce([injected_resource1](scoped_refptr<Resource> resource, | 
 |                                      ResourceUsageState usage_state) { | 
 |         EXPECT_EQ(injected_resource1, resource); | 
 |         EXPECT_EQ(ResourceUsageState::kUnderuse, usage_state); | 
 |       }); | 
 |   injected_resource1->SetResourceListener(&resource_listener1); | 
 |   injected_resource2->SetResourceListener(nullptr); | 
 |   StrictMock<MockResourceListener> resource_listener2; | 
 |   EXPECT_CALL(resource_listener2, OnResourceUsageStateMeasured(_, _)) | 
 |       .Times(1) | 
 |       .WillOnce([injected_resource2](scoped_refptr<Resource> resource, | 
 |                                      ResourceUsageState usage_state) { | 
 |         EXPECT_EQ(injected_resource2, resource); | 
 |         EXPECT_EQ(ResourceUsageState::kUnderuse, usage_state); | 
 |       }); | 
 |   injected_resource2->SetResourceListener(&resource_listener2); | 
 |   // The kUnderuse signal should get to our resource listeners. | 
 |   fake_resource->SetUsageState(ResourceUsageState::kUnderuse); | 
 |   call->DestroyVideoSendStream(stream1); | 
 |   call->DestroyVideoSendStream(stream2); | 
 | } | 
 |  | 
 | }  // namespace webrtc |