blob: cf15aacf114a6d7ac0fba35329d2585191f52938 [file] [log] [blame]
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001/*
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 */
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020010#ifndef TEST_CALL_TEST_H_
11#define TEST_CALL_TEST_H_
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000012
kwiberg4a206a92016-03-31 10:24:26 -070013#include <memory>
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000014#include <vector>
15
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020016#include "call/call.h"
17#include "call/rtp_transport_controller_send.h"
18#include "logging/rtc_event_log/rtc_event_log.h"
19#include "test/encoder_settings.h"
20#include "test/fake_audio_device.h"
21#include "test/fake_decoder.h"
22#include "test/fake_encoder.h"
23#include "test/fake_videorenderer.h"
24#include "test/frame_generator_capturer.h"
25#include "test/rtp_rtcp_observer.h"
26#include "test/single_threaded_task_queue.h"
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000027
28namespace webrtc {
Stefan Holmer9fea80f2016-01-07 17:43:18 +010029
30class VoEBase;
Stefan Holmer9fea80f2016-01-07 17:43:18 +010031
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000032namespace test {
33
34class BaseTest;
35
36class CallTest : public ::testing::Test {
37 public:
38 CallTest();
Stefan Holmer9fea80f2016-01-07 17:43:18 +010039 virtual ~CallTest();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000040
41 static const size_t kNumSsrcs = 3;
perkjfa10b552016-10-02 23:45:26 -070042 static const int kDefaultWidth = 320;
43 static const int kDefaultHeight = 180;
44 static const int kDefaultFramerate = 30;
Peter Boström5811a392015-12-10 13:02:50 +010045 static const int kDefaultTimeoutMs;
46 static const int kLongTimeoutMs;
Stefan Holmer9fea80f2016-01-07 17:43:18 +010047 static const uint8_t kVideoSendPayloadType;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000048 static const uint8_t kSendRtxPayloadType;
Stefan Holmer9fea80f2016-01-07 17:43:18 +010049 static const uint8_t kFakeVideoSendPayloadType;
stefan@webrtc.org01581da2014-09-04 06:48:14 +000050 static const uint8_t kRedPayloadType;
Shao Changbine62202f2015-04-21 20:24:50 +080051 static const uint8_t kRtxRedPayloadType;
stefan@webrtc.org01581da2014-09-04 06:48:14 +000052 static const uint8_t kUlpfecPayloadType;
brandtr841de6a2016-11-15 07:10:52 -080053 static const uint8_t kFlexfecPayloadType;
Stefan Holmer9fea80f2016-01-07 17:43:18 +010054 static const uint8_t kAudioSendPayloadType;
ilnik863f03b2017-07-11 02:38:36 -070055 static const uint8_t kPayloadTypeH264;
56 static const uint8_t kPayloadTypeVP8;
57 static const uint8_t kPayloadTypeVP9;
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +000058 static const uint32_t kSendRtxSsrcs[kNumSsrcs];
Stefan Holmer9fea80f2016-01-07 17:43:18 +010059 static const uint32_t kVideoSendSsrcs[kNumSsrcs];
60 static const uint32_t kAudioSendSsrc;
brandtr841de6a2016-11-15 07:10:52 -080061 static const uint32_t kFlexfecSendSsrc;
Stefan Holmer9fea80f2016-01-07 17:43:18 +010062 static const uint32_t kReceiverLocalVideoSsrc;
63 static const uint32_t kReceiverLocalAudioSsrc;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000064 static const int kNackRtpHistoryMs;
sprangd2702ef2017-07-10 08:41:10 -070065 static const uint8_t kDefaultKeepalivePayloadType;
minyue20c84cc2017-04-10 16:57:57 -070066 static const std::map<uint8_t, MediaType> payload_type_map_;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000067
68 protected:
Stefan Holmer9fea80f2016-01-07 17:43:18 +010069 // RunBaseTest overwrites the audio_state and the voice_engine of the send and
70 // receive Call configs to simplify test code and avoid having old VoiceEngine
71 // APIs in the tests.
stefane74eef12016-01-08 06:47:13 -080072 void RunBaseTest(BaseTest* test);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000073
74 void CreateCalls(const Call::Config& sender_config,
75 const Call::Config& receiver_config);
76 void CreateSenderCall(const Call::Config& config);
77 void CreateReceiverCall(const Call::Config& config);
Fredrik Solenberg4f4ec0a2015-10-22 10:49:27 +020078 void DestroyCalls();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000079
Stefan Holmer9fea80f2016-01-07 17:43:18 +010080 void CreateSendConfig(size_t num_video_streams,
81 size_t num_audio_streams,
brandtr841de6a2016-11-15 07:10:52 -080082 size_t num_flexfec_streams,
Stefan Holmer9fea80f2016-01-07 17:43:18 +010083 Transport* send_transport);
ilnika014cc52017-03-07 04:21:04 -080084
pbos2d566682015-09-28 09:59:31 -070085 void CreateMatchingReceiveConfigs(Transport* rtcp_send_transport);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000086
perkjfa10b552016-10-02 23:45:26 -070087 void CreateFrameGeneratorCapturerWithDrift(Clock* drift_clock,
88 float speed,
89 int framerate,
90 int width,
91 int height);
92 void CreateFrameGeneratorCapturer(int framerate, int width, int height);
oprypin92220ff2017-03-23 03:40:03 -070093 void CreateFakeAudioDevices(
94 std::unique_ptr<FakeAudioDevice::Capturer> capturer,
95 std::unique_ptr<FakeAudioDevice::Renderer> renderer);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000096
Stefan Holmer9fea80f2016-01-07 17:43:18 +010097 void CreateVideoStreams();
98 void CreateAudioStreams();
brandtr841de6a2016-11-15 07:10:52 -080099 void CreateFlexfecStreams();
eladalonc0d481a2017-08-02 07:39:07 -0700100
101 void AssociateFlexfecStreamsWithVideoStreams();
102 void DissociateFlexfecStreamsFromVideoStreams();
103
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000104 void Start();
105 void Stop();
106 void DestroyStreams();
Perba7dc722016-04-19 15:01:23 +0200107 void SetFakeVideoCaptureRotation(VideoRotation rotation);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000108
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +0000109 Clock* const clock_;
110
philipel4fb651d2017-04-10 03:54:05 -0700111 std::unique_ptr<webrtc::RtcEventLog> event_log_;
kwibergbfefb032016-05-01 14:53:46 -0700112 std::unique_ptr<Call> sender_call_;
sprangdb2a9fc2017-08-09 06:42:32 -0700113 RtpTransportControllerSend* sender_call_transport_controller_;
kwibergbfefb032016-05-01 14:53:46 -0700114 std::unique_ptr<PacketTransport> send_transport_;
stefanff483612015-12-21 03:14:00 -0800115 VideoSendStream::Config video_send_config_;
116 VideoEncoderConfig video_encoder_config_;
117 VideoSendStream* video_send_stream_;
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100118 AudioSendStream::Config audio_send_config_;
119 AudioSendStream* audio_send_stream_;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000120
kwibergbfefb032016-05-01 14:53:46 -0700121 std::unique_ptr<Call> receiver_call_;
122 std::unique_ptr<PacketTransport> receive_transport_;
stefanff483612015-12-21 03:14:00 -0800123 std::vector<VideoReceiveStream::Config> video_receive_configs_;
124 std::vector<VideoReceiveStream*> video_receive_streams_;
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100125 std::vector<AudioReceiveStream::Config> audio_receive_configs_;
126 std::vector<AudioReceiveStream*> audio_receive_streams_;
brandtr841de6a2016-11-15 07:10:52 -0800127 std::vector<FlexfecReceiveStream::Config> flexfec_receive_configs_;
128 std::vector<FlexfecReceiveStream*> flexfec_receive_streams_;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000129
kwibergbfefb032016-05-01 14:53:46 -0700130 std::unique_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000131 test::FakeEncoder fake_encoder_;
kwiberg4a206a92016-03-31 10:24:26 -0700132 std::vector<std::unique_ptr<VideoDecoder>> allocated_decoders_;
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100133 size_t num_video_streams_;
134 size_t num_audio_streams_;
brandtr841de6a2016-11-15 07:10:52 -0800135 size_t num_flexfec_streams_;
Rasmus Brandt31027342017-09-29 13:48:12 +0000136 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
137 rtc::scoped_refptr<AudioEncoderFactory> encoder_factory_;
sakal55d932b2016-09-30 06:19:08 -0700138 test::FakeVideoRenderer fake_renderer_;
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100139
eladalon413ee9a2017-08-22 04:02:52 -0700140 SingleThreadedTaskQueueForTesting task_queue_;
141
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100142 private:
143 // TODO(holmer): Remove once VoiceEngine is fully refactored to the new API.
144 // These methods are used to set up legacy voice engines and channels which is
145 // necessary while voice engine is being refactored to the new stream API.
146 struct VoiceEngineState {
147 VoiceEngineState()
148 : voice_engine(nullptr),
149 base(nullptr),
mflodman3d7db262016-04-29 00:57:13 -0700150 channel_id(-1) {}
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100151
152 VoiceEngine* voice_engine;
153 VoEBase* base;
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100154 int channel_id;
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100155 };
156
157 void CreateVoiceEngines();
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100158 void DestroyVoiceEngines();
159
160 VoiceEngineState voe_send_;
161 VoiceEngineState voe_recv_;
peaha9cc40b2017-06-29 08:32:09 -0700162 rtc::scoped_refptr<AudioProcessing> apm_send_;
163 rtc::scoped_refptr<AudioProcessing> apm_recv_;
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100164
165 // The audio devices must outlive the voice engines.
kwibergbfefb032016-05-01 14:53:46 -0700166 std::unique_ptr<test::FakeAudioDevice> fake_send_audio_device_;
167 std::unique_ptr<test::FakeAudioDevice> fake_recv_audio_device_;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000168};
169
170class BaseTest : public RtpRtcpObserver {
171 public:
philipele828c962017-03-21 03:24:27 -0700172 BaseTest();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000173 explicit BaseTest(unsigned int timeout_ms);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000174 virtual ~BaseTest();
175
176 virtual void PerformTest() = 0;
177 virtual bool ShouldCreateReceivers() const = 0;
178
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100179 virtual size_t GetNumVideoStreams() const;
180 virtual size_t GetNumAudioStreams() const;
brandtr841de6a2016-11-15 07:10:52 -0800181 virtual size_t GetNumFlexfecStreams() const;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000182
oprypin92220ff2017-03-23 03:40:03 -0700183 virtual std::unique_ptr<FakeAudioDevice::Capturer> CreateCapturer();
184 virtual std::unique_ptr<FakeAudioDevice::Renderer> CreateRenderer();
185 virtual void OnFakeAudioDevicesCreated(FakeAudioDevice* send_audio_device,
186 FakeAudioDevice* recv_audio_device);
187
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000188 virtual Call::Config GetSenderCallConfig();
189 virtual Call::Config GetReceiverCallConfig();
sprangdb2a9fc2017-08-09 06:42:32 -0700190 virtual void OnRtpTransportControllerSendCreated(
191 RtpTransportControllerSend* controller);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000192 virtual void OnCallsCreated(Call* sender_call, Call* receiver_call);
stefane74eef12016-01-08 06:47:13 -0800193
eladalon413ee9a2017-08-22 04:02:52 -0700194 virtual test::PacketTransport* CreateSendTransport(
195 SingleThreadedTaskQueueForTesting* task_queue,
196 Call* sender_call);
197 virtual test::PacketTransport* CreateReceiveTransport(
198 SingleThreadedTaskQueueForTesting* task_queue);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000199
stefanff483612015-12-21 03:14:00 -0800200 virtual void ModifyVideoConfigs(
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000201 VideoSendStream::Config* send_config,
202 std::vector<VideoReceiveStream::Config>* receive_configs,
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +0000203 VideoEncoderConfig* encoder_config);
perkjfa10b552016-10-02 23:45:26 -0700204 virtual void ModifyVideoCaptureStartResolution(int* width,
205 int* heigt,
206 int* frame_rate);
stefanff483612015-12-21 03:14:00 -0800207 virtual void OnVideoStreamsCreated(
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000208 VideoSendStream* send_stream,
209 const std::vector<VideoReceiveStream*>& receive_streams);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000210
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100211 virtual void ModifyAudioConfigs(
212 AudioSendStream::Config* send_config,
213 std::vector<AudioReceiveStream::Config>* receive_configs);
214 virtual void OnAudioStreamsCreated(
215 AudioSendStream* send_stream,
216 const std::vector<AudioReceiveStream*>& receive_streams);
217
brandtr841de6a2016-11-15 07:10:52 -0800218 virtual void ModifyFlexfecConfigs(
219 std::vector<FlexfecReceiveStream::Config>* receive_configs);
220 virtual void OnFlexfecStreamsCreated(
221 const std::vector<FlexfecReceiveStream*>& receive_streams);
222
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000223 virtual void OnFrameGeneratorCapturerCreated(
224 FrameGeneratorCapturer* frame_generator_capturer);
skvlad11a9cbf2016-10-07 11:53:05 -0700225
Fredrik Solenberg73276ad2017-09-14 14:46:47 +0200226 virtual void OnStreamsStopped();
oprypin92220ff2017-03-23 03:40:03 -0700227
philipel4fb651d2017-04-10 03:54:05 -0700228 std::unique_ptr<webrtc::RtcEventLog> event_log_;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000229};
230
231class SendTest : public BaseTest {
232 public:
233 explicit SendTest(unsigned int timeout_ms);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000234
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000235 bool ShouldCreateReceivers() const override;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000236};
237
238class EndToEndTest : public BaseTest {
239 public:
philipele828c962017-03-21 03:24:27 -0700240 EndToEndTest();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000241 explicit EndToEndTest(unsigned int timeout_ms);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000242
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000243 bool ShouldCreateReceivers() const override;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000244};
245
246} // namespace test
247} // namespace webrtc
248
Mirko Bonadei92ea95e2017-09-15 06:47:31 +0200249#endif // TEST_CALL_TEST_H_