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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
Danil Chapovalov99b71df2018-10-26 15:57:48 +020016#include "api/test/video/function_video_decoder_factory.h"
17#include "api/test/video/function_video_encoder_factory.h"
Jiawei Ouc2ebe212018-11-08 10:02:56 -080018#include "api/video/video_bitrate_allocator_factory.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020019#include "call/call.h"
20#include "call/rtp_transport_controller_send.h"
21#include "logging/rtc_event_log/rtc_event_log.h"
Artem Titov3faa8322018-03-07 14:44:00 +010022#include "modules/audio_device/include/test_audio_device.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020023#include "test/encoder_settings.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020024#include "test/fake_decoder.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020025#include "test/fake_videorenderer.h"
Ilya Nikolaevskiyb0588e62018-08-27 14:12:27 +020026#include "test/fake_vp8_encoder.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020027#include "test/frame_generator_capturer.h"
28#include "test/rtp_rtcp_observer.h"
29#include "test/single_threaded_task_queue.h"
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000030
31namespace webrtc {
32namespace 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
Ilya Nikolaevskiy255d1cd2017-12-21 18:02:59 +010041 static constexpr size_t kNumSsrcs = 6;
42 static const int kNumSimulcastStreams = 3;
perkjfa10b552016-10-02 23:45:26 -070043 static const int kDefaultWidth = 320;
44 static const int kDefaultHeight = 180;
45 static const int kDefaultFramerate = 30;
Peter Boström5811a392015-12-10 13:02:50 +010046 static const int kDefaultTimeoutMs;
47 static const int kLongTimeoutMs;
Ilya Nikolaevskiy465a5d92018-03-16 11:12:06 +010048 enum classPayloadTypes : uint8_t {
49 kSendRtxPayloadType = 98,
50 kRtxRedPayloadType = 99,
51 kVideoSendPayloadType = 100,
52 kAudioSendPayloadType = 103,
53 kRedPayloadType = 118,
54 kUlpfecPayloadType = 119,
55 kFlexfecPayloadType = 120,
56 kPayloadTypeH264 = 122,
57 kPayloadTypeVP8 = 123,
58 kPayloadTypeVP9 = 124,
59 kFakeVideoSendPayloadType = 125,
60 };
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +000061 static const uint32_t kSendRtxSsrcs[kNumSsrcs];
Stefan Holmer9fea80f2016-01-07 17:43:18 +010062 static const uint32_t kVideoSendSsrcs[kNumSsrcs];
63 static const uint32_t kAudioSendSsrc;
brandtr841de6a2016-11-15 07:10:52 -080064 static const uint32_t kFlexfecSendSsrc;
Stefan Holmer9fea80f2016-01-07 17:43:18 +010065 static const uint32_t kReceiverLocalVideoSsrc;
66 static const uint32_t kReceiverLocalAudioSsrc;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000067 static const int kNackRtpHistoryMs;
sprangd2702ef2017-07-10 08:41:10 -070068 static const uint8_t kDefaultKeepalivePayloadType;
minyue20c84cc2017-04-10 16:57:57 -070069 static const std::map<uint8_t, MediaType> payload_type_map_;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000070
71 protected:
Fredrik Solenberg8f5787a2018-01-11 13:52:30 +010072 // RunBaseTest overwrites the audio_state of the send and receive Call configs
73 // to simplify test code.
stefane74eef12016-01-08 06:47:13 -080074 void RunBaseTest(BaseTest* test);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000075
Sebastian Jansson8e6602f2018-07-13 10:43:20 +020076 void CreateCalls();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000077 void CreateCalls(const Call::Config& sender_config,
78 const Call::Config& receiver_config);
Sebastian Jansson8e6602f2018-07-13 10:43:20 +020079 void CreateSenderCall();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000080 void CreateSenderCall(const Call::Config& config);
81 void CreateReceiverCall(const Call::Config& config);
Fredrik Solenberg4f4ec0a2015-10-22 10:49:27 +020082 void DestroyCalls();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000083
Ilya Nikolaevskiy255d1cd2017-12-21 18:02:59 +010084 void CreateVideoSendConfig(VideoSendStream::Config* video_config,
85 size_t num_video_streams,
86 size_t num_used_ssrcs,
87 Transport* send_transport);
88 void CreateAudioAndFecSendConfigs(size_t num_audio_streams,
89 size_t num_flexfec_streams,
90 Transport* send_transport);
Sebastian Jansson3bd2c792018-07-13 13:29:03 +020091 void SetAudioConfig(const AudioSendStream::Config& config);
92
93 void SetSendFecConfig(std::vector<uint32_t> video_send_ssrcs);
94 void SetSendUlpFecConfig(VideoSendStream::Config* send_config);
95 void SetReceiveUlpFecConfig(VideoReceiveStream::Config* receive_config);
Stefan Holmer9fea80f2016-01-07 17:43:18 +010096 void CreateSendConfig(size_t num_video_streams,
97 size_t num_audio_streams,
brandtr841de6a2016-11-15 07:10:52 -080098 size_t num_flexfec_streams,
Stefan Holmer9fea80f2016-01-07 17:43:18 +010099 Transport* send_transport);
ilnika014cc52017-03-07 04:21:04 -0800100
Sebastian Jansson3bd2c792018-07-13 13:29:03 +0200101 void CreateMatchingVideoReceiveConfigs(
Ilya Nikolaevskiy255d1cd2017-12-21 18:02:59 +0100102 const VideoSendStream::Config& video_send_config,
103 Transport* rtcp_send_transport);
Sebastian Jansson3bd2c792018-07-13 13:29:03 +0200104 void CreateMatchingVideoReceiveConfigs(
105 const VideoSendStream::Config& video_send_config,
106 Transport* rtcp_send_transport,
107 bool send_side_bwe,
Niels Möllercbcbc222018-09-28 09:07:24 +0200108 VideoDecoderFactory* decoder_factory,
Sebastian Jansson3bd2c792018-07-13 13:29:03 +0200109 absl::optional<size_t> decode_sub_stream,
110 bool receiver_reference_time_report,
111 int rtp_history_ms);
112 void AddMatchingVideoReceiveConfigs(
113 std::vector<VideoReceiveStream::Config>* receive_configs,
114 const VideoSendStream::Config& video_send_config,
115 Transport* rtcp_send_transport,
116 bool send_side_bwe,
Niels Möllercbcbc222018-09-28 09:07:24 +0200117 VideoDecoderFactory* decoder_factory,
Sebastian Jansson3bd2c792018-07-13 13:29:03 +0200118 absl::optional<size_t> decode_sub_stream,
119 bool receiver_reference_time_report,
120 int rtp_history_ms);
121
Ilya Nikolaevskiy255d1cd2017-12-21 18:02:59 +0100122 void CreateMatchingAudioAndFecConfigs(Transport* rtcp_send_transport);
Sebastian Jansson3bd2c792018-07-13 13:29:03 +0200123 void CreateMatchingAudioConfigs(Transport* transport, std::string sync_group);
124 static AudioReceiveStream::Config CreateMatchingAudioConfig(
125 const AudioSendStream::Config& send_config,
126 rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
127 Transport* transport,
128 std::string sync_group);
129 void CreateMatchingFecConfig(
130 Transport* transport,
131 const VideoSendStream::Config& video_send_config);
pbos2d566682015-09-28 09:59:31 -0700132 void CreateMatchingReceiveConfigs(Transport* rtcp_send_transport);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000133
perkjfa10b552016-10-02 23:45:26 -0700134 void CreateFrameGeneratorCapturerWithDrift(Clock* drift_clock,
135 float speed,
136 int framerate,
137 int width,
138 int height);
139 void CreateFrameGeneratorCapturer(int framerate, int width, int height);
oprypin92220ff2017-03-23 03:40:03 -0700140 void CreateFakeAudioDevices(
Artem Titov3faa8322018-03-07 14:44:00 +0100141 std::unique_ptr<TestAudioDeviceModule::Capturer> capturer,
142 std::unique_ptr<TestAudioDeviceModule::Renderer> renderer);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000143
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100144 void CreateVideoStreams();
Sebastian Janssonf33905d2018-07-13 09:49:00 +0200145 void CreateVideoSendStreams();
146 void CreateVideoSendStream(const VideoEncoderConfig& encoder_config);
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100147 void CreateAudioStreams();
brandtr841de6a2016-11-15 07:10:52 -0800148 void CreateFlexfecStreams();
eladalonc0d481a2017-08-02 07:39:07 -0700149
Sebastian Janssonf33905d2018-07-13 09:49:00 +0200150 void ConnectVideoSourcesToStreams();
151
eladalonc0d481a2017-08-02 07:39:07 -0700152 void AssociateFlexfecStreamsWithVideoStreams();
153 void DissociateFlexfecStreamsFromVideoStreams();
154
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000155 void Start();
Sebastian Janssonf33905d2018-07-13 09:49:00 +0200156 void StartVideoStreams();
157 void StartVideoCapture();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000158 void Stop();
Sebastian Jansson3bd2c792018-07-13 13:29:03 +0200159 void StopVideoCapture();
160 void StopVideoStreams();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000161 void DestroyStreams();
Sebastian Janssonf33905d2018-07-13 09:49:00 +0200162 void DestroyVideoSendStreams();
Perba7dc722016-04-19 15:01:23 +0200163 void SetFakeVideoCaptureRotation(VideoRotation rotation);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000164
Sebastian Janssonf33905d2018-07-13 09:49:00 +0200165 void SetVideoDegradation(DegradationPreference preference);
166
167 VideoSendStream::Config* GetVideoSendConfig();
168 void SetVideoSendConfig(const VideoSendStream::Config& config);
169 VideoEncoderConfig* GetVideoEncoderConfig();
170 void SetVideoEncoderConfig(const VideoEncoderConfig& config);
171 VideoSendStream* GetVideoSendStream();
Sebastian Jansson3bd2c792018-07-13 13:29:03 +0200172 FlexfecReceiveStream::Config* GetFlexFecConfig();
Sebastian Janssonf33905d2018-07-13 09:49:00 +0200173
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +0000174 Clock* const clock_;
175
Sebastian Jansson8e6602f2018-07-13 10:43:20 +0200176 std::unique_ptr<webrtc::RtcEventLog> send_event_log_;
177 std::unique_ptr<webrtc::RtcEventLog> recv_event_log_;
kwibergbfefb032016-05-01 14:53:46 -0700178 std::unique_ptr<Call> sender_call_;
sprangdb2a9fc2017-08-09 06:42:32 -0700179 RtpTransportControllerSend* sender_call_transport_controller_;
kwibergbfefb032016-05-01 14:53:46 -0700180 std::unique_ptr<PacketTransport> send_transport_;
Sebastian Jansson3bd2c792018-07-13 13:29:03 +0200181 std::vector<VideoSendStream::Config> video_send_configs_;
182 std::vector<VideoEncoderConfig> video_encoder_configs_;
183 std::vector<VideoSendStream*> video_send_streams_;
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100184 AudioSendStream::Config audio_send_config_;
185 AudioSendStream* audio_send_stream_;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000186
kwibergbfefb032016-05-01 14:53:46 -0700187 std::unique_ptr<Call> receiver_call_;
188 std::unique_ptr<PacketTransport> receive_transport_;
stefanff483612015-12-21 03:14:00 -0800189 std::vector<VideoReceiveStream::Config> video_receive_configs_;
190 std::vector<VideoReceiveStream*> video_receive_streams_;
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100191 std::vector<AudioReceiveStream::Config> audio_receive_configs_;
192 std::vector<AudioReceiveStream*> audio_receive_streams_;
brandtr841de6a2016-11-15 07:10:52 -0800193 std::vector<FlexfecReceiveStream::Config> flexfec_receive_configs_;
194 std::vector<FlexfecReceiveStream*> flexfec_receive_streams_;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000195
Sebastian Jansson3bd2c792018-07-13 13:29:03 +0200196 test::FrameGeneratorCapturer* frame_generator_capturer_;
197 std::vector<rtc::VideoSourceInterface<VideoFrame>*> video_sources_;
Sebastian Janssonf1f363f2018-08-13 14:24:58 +0200198 std::vector<std::unique_ptr<TestVideoCapturer>> video_capturers_;
Sebastian Jansson3bd2c792018-07-13 13:29:03 +0200199 DegradationPreference degradation_preference_ =
200 DegradationPreference::MAINTAIN_FRAMERATE;
201
202 std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory_;
Sebastian Jansson50eb4c42018-08-03 13:25:17 +0200203 std::unique_ptr<NetworkControllerFactoryInterface>
204 bbr_network_controller_factory_;
Sebastian Jansson3bd2c792018-07-13 13:29:03 +0200205
Niels Möller4db138e2018-04-19 09:04:13 +0200206 test::FunctionVideoEncoderFactory fake_encoder_factory_;
207 int fake_encoder_max_bitrate_ = -1;
Niels Möllercbcbc222018-09-28 09:07:24 +0200208 test::FunctionVideoDecoderFactory fake_decoder_factory_;
Jiawei Ouc2ebe212018-11-08 10:02:56 -0800209 std::unique_ptr<VideoBitrateAllocatorFactory> bitrate_allocator_factory_;
Sebastian Jansson3bd2c792018-07-13 13:29:03 +0200210 // Number of simulcast substreams.
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100211 size_t num_video_streams_;
212 size_t num_audio_streams_;
brandtr841de6a2016-11-15 07:10:52 -0800213 size_t num_flexfec_streams_;
Niels Möller2784a032018-03-28 14:16:04 +0200214 rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory_;
215 rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory_;
sakal55d932b2016-09-30 06:19:08 -0700216 test::FakeVideoRenderer fake_renderer_;
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100217
eladalon413ee9a2017-08-22 04:02:52 -0700218 SingleThreadedTaskQueueForTesting task_queue_;
219
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100220 private:
peaha9cc40b2017-06-29 08:32:09 -0700221 rtc::scoped_refptr<AudioProcessing> apm_send_;
222 rtc::scoped_refptr<AudioProcessing> apm_recv_;
Artem Titov3faa8322018-03-07 14:44:00 +0100223 rtc::scoped_refptr<TestAudioDeviceModule> fake_send_audio_device_;
224 rtc::scoped_refptr<TestAudioDeviceModule> fake_recv_audio_device_;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000225};
226
227class BaseTest : public RtpRtcpObserver {
228 public:
philipele828c962017-03-21 03:24:27 -0700229 BaseTest();
Sebastian Jansson72582242018-07-13 13:19:42 +0200230 explicit BaseTest(int timeout_ms);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000231 virtual ~BaseTest();
232
233 virtual void PerformTest() = 0;
234 virtual bool ShouldCreateReceivers() const = 0;
235
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100236 virtual size_t GetNumVideoStreams() const;
237 virtual size_t GetNumAudioStreams() const;
brandtr841de6a2016-11-15 07:10:52 -0800238 virtual size_t GetNumFlexfecStreams() const;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000239
Artem Titov3faa8322018-03-07 14:44:00 +0100240 virtual std::unique_ptr<TestAudioDeviceModule::Capturer> CreateCapturer();
241 virtual std::unique_ptr<TestAudioDeviceModule::Renderer> CreateRenderer();
242 virtual void OnFakeAudioDevicesCreated(
243 TestAudioDeviceModule* send_audio_device,
244 TestAudioDeviceModule* recv_audio_device);
oprypin92220ff2017-03-23 03:40:03 -0700245
Sebastian Jansson72582242018-07-13 13:19:42 +0200246 virtual void ModifySenderCallConfig(Call::Config* config);
247 virtual void ModifyReceiverCallConfig(Call::Config* config);
248
sprangdb2a9fc2017-08-09 06:42:32 -0700249 virtual void OnRtpTransportControllerSendCreated(
250 RtpTransportControllerSend* controller);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000251 virtual void OnCallsCreated(Call* sender_call, Call* receiver_call);
stefane74eef12016-01-08 06:47:13 -0800252
eladalon413ee9a2017-08-22 04:02:52 -0700253 virtual test::PacketTransport* CreateSendTransport(
254 SingleThreadedTaskQueueForTesting* task_queue,
255 Call* sender_call);
256 virtual test::PacketTransport* CreateReceiveTransport(
257 SingleThreadedTaskQueueForTesting* task_queue);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000258
stefanff483612015-12-21 03:14:00 -0800259 virtual void ModifyVideoConfigs(
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000260 VideoSendStream::Config* send_config,
261 std::vector<VideoReceiveStream::Config>* receive_configs,
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +0000262 VideoEncoderConfig* encoder_config);
perkjfa10b552016-10-02 23:45:26 -0700263 virtual void ModifyVideoCaptureStartResolution(int* width,
264 int* heigt,
265 int* frame_rate);
Åsa Perssoncb7eddb2018-11-05 14:11:44 +0100266 virtual void ModifyVideoDegradationPreference(
267 DegradationPreference* degradation_preference);
268
stefanff483612015-12-21 03:14:00 -0800269 virtual void OnVideoStreamsCreated(
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000270 VideoSendStream* send_stream,
271 const std::vector<VideoReceiveStream*>& receive_streams);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000272
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100273 virtual void ModifyAudioConfigs(
274 AudioSendStream::Config* send_config,
275 std::vector<AudioReceiveStream::Config>* receive_configs);
276 virtual void OnAudioStreamsCreated(
277 AudioSendStream* send_stream,
278 const std::vector<AudioReceiveStream*>& receive_streams);
279
brandtr841de6a2016-11-15 07:10:52 -0800280 virtual void ModifyFlexfecConfigs(
281 std::vector<FlexfecReceiveStream::Config>* receive_configs);
282 virtual void OnFlexfecStreamsCreated(
283 const std::vector<FlexfecReceiveStream*>& receive_streams);
284
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000285 virtual void OnFrameGeneratorCapturerCreated(
286 FrameGeneratorCapturer* frame_generator_capturer);
skvlad11a9cbf2016-10-07 11:53:05 -0700287
Fredrik Solenberg73276ad2017-09-14 14:46:47 +0200288 virtual void OnStreamsStopped();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000289};
290
291class SendTest : public BaseTest {
292 public:
Sebastian Jansson72582242018-07-13 13:19:42 +0200293 explicit SendTest(int timeout_ms);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000294
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000295 bool ShouldCreateReceivers() const override;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000296};
297
298class EndToEndTest : public BaseTest {
299 public:
philipele828c962017-03-21 03:24:27 -0700300 EndToEndTest();
Sebastian Jansson72582242018-07-13 13:19:42 +0200301 explicit EndToEndTest(int timeout_ms);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000302
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000303 bool ShouldCreateReceivers() const override;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000304};
305
306} // namespace test
307} // namespace webrtc
308
Mirko Bonadei92ea95e2017-09-15 06:47:31 +0200309#endif // TEST_CALL_TEST_H_