| /* |
| * 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 "absl/memory/memory.h" |
| #include "api/audio_codecs/builtin_audio_decoder_factory.h" |
| #include "api/task_queue/default_task_queue_factory.h" |
| #include "api/test/fake_media_transport.h" |
| #include "api/test/mock_audio_mixer.h" |
| #include "audio/audio_receive_stream.h" |
| #include "audio/audio_send_stream.h" |
| #include "call/audio_state.h" |
| #include "call/call.h" |
| #include "logging/rtc_event_log/rtc_event_log.h" |
| #include "modules/audio_device/include/mock_audio_device.h" |
| #include "modules/audio_processing/include/mock_audio_processing.h" |
| #include "modules/pacing/mock/mock_paced_sender.h" |
| #include "modules/rtp_rtcp/include/rtp_rtcp.h" |
| #include "test/fake_encoder.h" |
| #include "test/gtest.h" |
| #include "test/mock_audio_decoder_factory.h" |
| #include "test/mock_transport.h" |
| |
| namespace { |
| |
| struct CallHelper { |
| CallHelper() { |
| task_queue_factory_ = webrtc::CreateDefaultTaskQueueFactory(); |
| webrtc::AudioState::Config audio_state_config; |
| audio_state_config.audio_mixer = |
| new rtc::RefCountedObject<webrtc::test::MockAudioMixer>(); |
| audio_state_config.audio_processing = |
| new rtc::RefCountedObject<webrtc::test::MockAudioProcessing>(); |
| audio_state_config.audio_device_module = |
| new rtc::RefCountedObject<webrtc::test::MockAudioDeviceModule>(); |
| webrtc::Call::Config config(&event_log_); |
| config.audio_state = webrtc::AudioState::Create(audio_state_config); |
| config.task_queue_factory = task_queue_factory_.get(); |
| call_.reset(webrtc::Call::Create(config)); |
| } |
| |
| webrtc::Call* operator->() { return call_.get(); } |
| |
| private: |
| webrtc::RtcEventLogNullImpl event_log_; |
| std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory_; |
| std::unique_ptr<webrtc::Call> call_; |
| }; |
| } // namespace |
| |
| namespace webrtc { |
| |
| TEST(CallTest, ConstructDestruct) { |
| CallHelper call; |
| } |
| |
| TEST(CallTest, CreateDestroy_AudioSendStream) { |
| CallHelper call; |
| MockTransport send_transport; |
| AudioSendStream::Config config(&send_transport, MediaTransportConfig()); |
| config.rtp.ssrc = 42; |
| AudioSendStream* stream = call->CreateAudioSendStream(config); |
| EXPECT_NE(stream, nullptr); |
| call->DestroyAudioSendStream(stream); |
| } |
| |
| TEST(CallTest, CreateDestroy_AudioReceiveStream) { |
| CallHelper call; |
| AudioReceiveStream::Config config; |
| MockTransport rtcp_send_transport; |
| config.rtp.remote_ssrc = 42; |
| config.rtcp_send_transport = &rtcp_send_transport; |
| config.decoder_factory = |
| new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>(); |
| AudioReceiveStream* stream = call->CreateAudioReceiveStream(config); |
| EXPECT_NE(stream, nullptr); |
| call->DestroyAudioReceiveStream(stream); |
| } |
| |
| TEST(CallTest, CreateDestroy_AudioSendStreams) { |
| CallHelper call; |
| MockTransport send_transport; |
| AudioSendStream::Config config(&send_transport, MediaTransportConfig()); |
| 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) { |
| CallHelper call; |
| AudioReceiveStream::Config config; |
| MockTransport rtcp_send_transport; |
| config.rtcp_send_transport = &rtcp_send_transport; |
| config.decoder_factory = |
| new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>(); |
| std::list<AudioReceiveStream*> streams; |
| for (int i = 0; i < 2; ++i) { |
| for (uint32_t ssrc = 0; ssrc < 1234567; ssrc += 34567) { |
| config.rtp.remote_ssrc = ssrc; |
| AudioReceiveStream* 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_AssociateAudioSendReceiveStreams_RecvFirst) { |
| CallHelper call; |
| AudioReceiveStream::Config recv_config; |
| MockTransport rtcp_send_transport; |
| recv_config.rtp.remote_ssrc = 42; |
| recv_config.rtp.local_ssrc = 777; |
| recv_config.rtcp_send_transport = &rtcp_send_transport; |
| recv_config.decoder_factory = |
| new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>(); |
| AudioReceiveStream* recv_stream = call->CreateAudioReceiveStream(recv_config); |
| EXPECT_NE(recv_stream, nullptr); |
| |
| MockTransport send_transport; |
| AudioSendStream::Config send_config(&send_transport, MediaTransportConfig()); |
| send_config.rtp.ssrc = 777; |
| AudioSendStream* send_stream = call->CreateAudioSendStream(send_config); |
| EXPECT_NE(send_stream, nullptr); |
| |
| internal::AudioReceiveStream* internal_recv_stream = |
| static_cast<internal::AudioReceiveStream*>(recv_stream); |
| EXPECT_EQ(send_stream, |
| internal_recv_stream->GetAssociatedSendStreamForTesting()); |
| |
| call->DestroyAudioSendStream(send_stream); |
| EXPECT_EQ(nullptr, internal_recv_stream->GetAssociatedSendStreamForTesting()); |
| |
| call->DestroyAudioReceiveStream(recv_stream); |
| } |
| |
| TEST(CallTest, CreateDestroy_AssociateAudioSendReceiveStreams_SendFirst) { |
| CallHelper call; |
| MockTransport send_transport; |
| AudioSendStream::Config send_config(&send_transport, MediaTransportConfig()); |
| send_config.rtp.ssrc = 777; |
| AudioSendStream* send_stream = call->CreateAudioSendStream(send_config); |
| EXPECT_NE(send_stream, nullptr); |
| |
| AudioReceiveStream::Config recv_config; |
| MockTransport rtcp_send_transport; |
| recv_config.rtp.remote_ssrc = 42; |
| recv_config.rtp.local_ssrc = 777; |
| recv_config.rtcp_send_transport = &rtcp_send_transport; |
| recv_config.decoder_factory = |
| new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>(); |
| AudioReceiveStream* recv_stream = call->CreateAudioReceiveStream(recv_config); |
| EXPECT_NE(recv_stream, nullptr); |
| |
| internal::AudioReceiveStream* internal_recv_stream = |
| static_cast<internal::AudioReceiveStream*>(recv_stream); |
| EXPECT_EQ(send_stream, |
| internal_recv_stream->GetAssociatedSendStreamForTesting()); |
| |
| call->DestroyAudioReceiveStream(recv_stream); |
| |
| call->DestroyAudioSendStream(send_stream); |
| } |
| |
| TEST(CallTest, CreateDestroy_FlexfecReceiveStream) { |
| CallHelper call; |
| MockTransport rtcp_send_transport; |
| FlexfecReceiveStream::Config config(&rtcp_send_transport); |
| config.payload_type = 118; |
| config.remote_ssrc = 38837212; |
| config.protected_media_ssrcs = {27273}; |
| |
| FlexfecReceiveStream* stream = call->CreateFlexfecReceiveStream(config); |
| EXPECT_NE(stream, nullptr); |
| call->DestroyFlexfecReceiveStream(stream); |
| } |
| |
| TEST(CallTest, CreateDestroy_FlexfecReceiveStreams) { |
| CallHelper call; |
| 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.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) { |
| CallHelper call; |
| 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.remote_ssrc = 838383; |
| stream = call->CreateFlexfecReceiveStream(config); |
| EXPECT_NE(stream, nullptr); |
| streams.push_back(stream); |
| |
| config.remote_ssrc = 424993; |
| stream = call->CreateFlexfecReceiveStream(config); |
| EXPECT_NE(stream, nullptr); |
| streams.push_back(stream); |
| |
| config.remote_ssrc = 99383; |
| stream = call->CreateFlexfecReceiveStream(config); |
| EXPECT_NE(stream, nullptr); |
| streams.push_back(stream); |
| |
| 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, RecreatingAudioStreamWithSameSsrcReusesRtpState) { |
| constexpr uint32_t kSSRC = 12345; |
| CallHelper call; |
| |
| auto create_stream_and_get_rtp_state = [&](uint32_t ssrc) { |
| MockTransport send_transport; |
| AudioSendStream::Config config(&send_transport, MediaTransportConfig()); |
| 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_ms, rtp_state2.capture_time_ms); |
| EXPECT_EQ(rtp_state1.last_timestamp_time_ms, |
| rtp_state2.last_timestamp_time_ms); |
| EXPECT_EQ(rtp_state1.media_has_been_sent, rtp_state2.media_has_been_sent); |
| } |
| |
| TEST(CallTest, RegisterMediaTransportBitrateCallbacksInCreateStream) { |
| CallHelper call; |
| MediaTransportSettings settings; |
| webrtc::FakeMediaTransport fake_media_transport(settings); |
| |
| EXPECT_EQ(0, fake_media_transport.target_rate_observers_size()); |
| // TODO(solenberg): This test shouldn't require a Transport, but currently |
| // RTCPSender requires one. |
| MockTransport send_transport; |
| AudioSendStream::Config config(&send_transport, |
| MediaTransportConfig(&fake_media_transport)); |
| |
| call->MediaTransportChange(&fake_media_transport); |
| AudioSendStream* stream = call->CreateAudioSendStream(config); |
| |
| // We get 2 subscribers: one subscriber from call.cc, and one from |
| // ChannelSend. |
| EXPECT_EQ(2, fake_media_transport.target_rate_observers_size()); |
| |
| call->DestroyAudioSendStream(stream); |
| EXPECT_EQ(1, fake_media_transport.target_rate_observers_size()); |
| |
| call->MediaTransportChange(nullptr); |
| EXPECT_EQ(0, fake_media_transport.target_rate_observers_size()); |
| } |
| |
| } // namespace webrtc |