1 /* 2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 #ifndef WEBRTC_TEST_CALL_TEST_H_ 11 #define WEBRTC_TEST_CALL_TEST_H_ 12 13 #include <vector> 14 15 #include "webrtc/call.h" 16 #include "webrtc/call/transport_adapter.h" 17 #include "webrtc/system_wrappers/include/scoped_vector.h" 18 #include "webrtc/test/fake_audio_device.h" 19 #include "webrtc/test/fake_decoder.h" 20 #include "webrtc/test/fake_encoder.h" 21 #include "webrtc/test/frame_generator_capturer.h" 22 #include "webrtc/test/rtp_rtcp_observer.h" 23 24 namespace webrtc { 25 26 class VoEBase; 27 class VoECodec; 28 class VoENetwork; 29 30 namespace test { 31 32 class BaseTest; 33 34 class CallTest : public ::testing::Test { 35 public: 36 CallTest(); 37 virtual ~CallTest(); 38 39 static const size_t kNumSsrcs = 3; 40 41 static const int kDefaultTimeoutMs; 42 static const int kLongTimeoutMs; 43 static const uint8_t kVideoSendPayloadType; 44 static const uint8_t kSendRtxPayloadType; 45 static const uint8_t kFakeVideoSendPayloadType; 46 static const uint8_t kRedPayloadType; 47 static const uint8_t kRtxRedPayloadType; 48 static const uint8_t kUlpfecPayloadType; 49 static const uint8_t kAudioSendPayloadType; 50 static const uint32_t kSendRtxSsrcs[kNumSsrcs]; 51 static const uint32_t kVideoSendSsrcs[kNumSsrcs]; 52 static const uint32_t kAudioSendSsrc; 53 static const uint32_t kReceiverLocalVideoSsrc; 54 static const uint32_t kReceiverLocalAudioSsrc; 55 static const int kNackRtpHistoryMs; 56 57 protected: 58 // RunBaseTest overwrites the audio_state and the voice_engine of the send and 59 // receive Call configs to simplify test code and avoid having old VoiceEngine 60 // APIs in the tests. 61 void RunBaseTest(BaseTest* test); 62 63 void CreateCalls(const Call::Config& sender_config, 64 const Call::Config& receiver_config); 65 void CreateSenderCall(const Call::Config& config); 66 void CreateReceiverCall(const Call::Config& config); 67 void DestroyCalls(); 68 69 void CreateSendConfig(size_t num_video_streams, 70 size_t num_audio_streams, 71 Transport* send_transport); 72 void CreateMatchingReceiveConfigs(Transport* rtcp_send_transport); 73 74 void CreateFrameGeneratorCapturer(); 75 void CreateFakeAudioDevices(); 76 77 void CreateVideoStreams(); 78 void CreateAudioStreams(); 79 void Start(); 80 void Stop(); 81 void DestroyStreams(); 82 83 Clock* const clock_; 84 85 rtc::scoped_ptr<Call> sender_call_; 86 rtc::scoped_ptr<PacketTransport> send_transport_; 87 VideoSendStream::Config video_send_config_; 88 VideoEncoderConfig video_encoder_config_; 89 VideoSendStream* video_send_stream_; 90 AudioSendStream::Config audio_send_config_; 91 AudioSendStream* audio_send_stream_; 92 93 rtc::scoped_ptr<Call> receiver_call_; 94 rtc::scoped_ptr<PacketTransport> receive_transport_; 95 std::vector<VideoReceiveStream::Config> video_receive_configs_; 96 std::vector<VideoReceiveStream*> video_receive_streams_; 97 std::vector<AudioReceiveStream::Config> audio_receive_configs_; 98 std::vector<AudioReceiveStream*> audio_receive_streams_; 99 100 rtc::scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_; 101 test::FakeEncoder fake_encoder_; 102 ScopedVector<VideoDecoder> allocated_decoders_; 103 size_t num_video_streams_; 104 size_t num_audio_streams_; 105 106 private: 107 // TODO(holmer): Remove once VoiceEngine is fully refactored to the new API. 108 // These methods are used to set up legacy voice engines and channels which is 109 // necessary while voice engine is being refactored to the new stream API. 110 struct VoiceEngineState { 111 VoiceEngineState() 112 : voice_engine(nullptr), 113 base(nullptr), 114 network(nullptr), 115 codec(nullptr), 116 channel_id(-1), 117 transport_adapter(nullptr) {} 118 119 VoiceEngine* voice_engine; 120 VoEBase* base; 121 VoENetwork* network; 122 VoECodec* codec; 123 int channel_id; 124 rtc::scoped_ptr<internal::TransportAdapter> transport_adapter; 125 }; 126 127 void CreateVoiceEngines(); 128 void SetupVoiceEngineTransports(PacketTransport* send_transport, 129 PacketTransport* recv_transport); 130 void DestroyVoiceEngines(); 131 132 VoiceEngineState voe_send_; 133 VoiceEngineState voe_recv_; 134 135 // The audio devices must outlive the voice engines. 136 rtc::scoped_ptr<test::FakeAudioDevice> fake_send_audio_device_; 137 rtc::scoped_ptr<test::FakeAudioDevice> fake_recv_audio_device_; 138 }; 139 140 class BaseTest : public RtpRtcpObserver { 141 public: 142 explicit BaseTest(unsigned int timeout_ms); 143 virtual ~BaseTest(); 144 145 virtual void PerformTest() = 0; 146 virtual bool ShouldCreateReceivers() const = 0; 147 148 virtual size_t GetNumVideoStreams() const; 149 virtual size_t GetNumAudioStreams() const; 150 151 virtual Call::Config GetSenderCallConfig(); 152 virtual Call::Config GetReceiverCallConfig(); 153 virtual void OnCallsCreated(Call* sender_call, Call* receiver_call); 154 155 virtual test::PacketTransport* CreateSendTransport(Call* sender_call); 156 virtual test::PacketTransport* CreateReceiveTransport(); 157 158 virtual void ModifyVideoConfigs( 159 VideoSendStream::Config* send_config, 160 std::vector<VideoReceiveStream::Config>* receive_configs, 161 VideoEncoderConfig* encoder_config); 162 virtual void OnVideoStreamsCreated( 163 VideoSendStream* send_stream, 164 const std::vector<VideoReceiveStream*>& receive_streams); 165 166 virtual void ModifyAudioConfigs( 167 AudioSendStream::Config* send_config, 168 std::vector<AudioReceiveStream::Config>* receive_configs); 169 virtual void OnAudioStreamsCreated( 170 AudioSendStream* send_stream, 171 const std::vector<AudioReceiveStream*>& receive_streams); 172 173 virtual void OnFrameGeneratorCapturerCreated( 174 FrameGeneratorCapturer* frame_generator_capturer); 175 }; 176 177 class SendTest : public BaseTest { 178 public: 179 explicit SendTest(unsigned int timeout_ms); 180 181 bool ShouldCreateReceivers() const override; 182 }; 183 184 class EndToEndTest : public BaseTest { 185 public: 186 explicit EndToEndTest(unsigned int timeout_ms); 187 188 bool ShouldCreateReceivers() const override; 189 }; 190 191 } // namespace test 192 } // namespace webrtc 193 194 #endif // WEBRTC_TEST_CALL_TEST_H_ 195
