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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
Bjorn Terelius5c2f1f02019-01-16 17:45:05 +010013#include <map>
kwiberg4a206a92016-03-31 10:24:26 -070014#include <memory>
Bjorn Terelius5c2f1f02019-01-16 17:45:05 +010015#include <string>
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000016#include <vector>
17
Elad Alond8d32482019-02-18 23:45:57 +010018#include "absl/types/optional.h"
Danil Chapovalov99b71df2018-10-26 15:57:48 +020019#include "api/test/video/function_video_decoder_factory.h"
20#include "api/test/video/function_video_encoder_factory.h"
Jiawei Ouc2ebe212018-11-08 10:02:56 -080021#include "api/video/video_bitrate_allocator_factory.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020022#include "call/call.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020023#include "logging/rtc_event_log/rtc_event_log.h"
Artem Titov3faa8322018-03-07 14:44:00 +010024#include "modules/audio_device/include/test_audio_device.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020025#include "test/encoder_settings.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020026#include "test/fake_decoder.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020027#include "test/fake_videorenderer.h"
Ilya Nikolaevskiyb0588e62018-08-27 14:12:27 +020028#include "test/fake_vp8_encoder.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020029#include "test/frame_generator_capturer.h"
30#include "test/rtp_rtcp_observer.h"
31#include "test/single_threaded_task_queue.h"
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000032
33namespace webrtc {
34namespace test {
35
36class BaseTest;
37
38class CallTest : public ::testing::Test {
39 public:
40 CallTest();
Stefan Holmer9fea80f2016-01-07 17:43:18 +010041 virtual ~CallTest();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000042
Ilya Nikolaevskiy255d1cd2017-12-21 18:02:59 +010043 static constexpr size_t kNumSsrcs = 6;
44 static const int kNumSimulcastStreams = 3;
perkjfa10b552016-10-02 23:45:26 -070045 static const int kDefaultWidth = 320;
46 static const int kDefaultHeight = 180;
47 static const int kDefaultFramerate = 30;
Peter Boström5811a392015-12-10 13:02:50 +010048 static const int kDefaultTimeoutMs;
49 static const int kLongTimeoutMs;
Ilya Nikolaevskiy465a5d92018-03-16 11:12:06 +010050 enum classPayloadTypes : uint8_t {
51 kSendRtxPayloadType = 98,
52 kRtxRedPayloadType = 99,
53 kVideoSendPayloadType = 100,
54 kAudioSendPayloadType = 103,
55 kRedPayloadType = 118,
56 kUlpfecPayloadType = 119,
57 kFlexfecPayloadType = 120,
58 kPayloadTypeH264 = 122,
59 kPayloadTypeVP8 = 123,
60 kPayloadTypeVP9 = 124,
61 kFakeVideoSendPayloadType = 125,
62 };
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +000063 static const uint32_t kSendRtxSsrcs[kNumSsrcs];
Stefan Holmer9fea80f2016-01-07 17:43:18 +010064 static const uint32_t kVideoSendSsrcs[kNumSsrcs];
65 static const uint32_t kAudioSendSsrc;
brandtr841de6a2016-11-15 07:10:52 -080066 static const uint32_t kFlexfecSendSsrc;
Stefan Holmer9fea80f2016-01-07 17:43:18 +010067 static const uint32_t kReceiverLocalVideoSsrc;
68 static const uint32_t kReceiverLocalAudioSsrc;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000069 static const int kNackRtpHistoryMs;
minyue20c84cc2017-04-10 16:57:57 -070070 static const std::map<uint8_t, MediaType> payload_type_map_;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000071
72 protected:
Elad Alond8d32482019-02-18 23:45:57 +010073 void RegisterRtpExtension(const RtpExtension& extension);
74
Fredrik Solenberg8f5787a2018-01-11 13:52:30 +010075 // RunBaseTest overwrites the audio_state of the send and receive Call configs
76 // to simplify test code.
stefane74eef12016-01-08 06:47:13 -080077 void RunBaseTest(BaseTest* test);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000078
Sebastian Jansson8e6602f2018-07-13 10:43:20 +020079 void CreateCalls();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000080 void CreateCalls(const Call::Config& sender_config,
81 const Call::Config& receiver_config);
Sebastian Jansson8e6602f2018-07-13 10:43:20 +020082 void CreateSenderCall();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000083 void CreateSenderCall(const Call::Config& config);
84 void CreateReceiverCall(const Call::Config& config);
Fredrik Solenberg4f4ec0a2015-10-22 10:49:27 +020085 void DestroyCalls();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000086
Ilya Nikolaevskiy255d1cd2017-12-21 18:02:59 +010087 void CreateVideoSendConfig(VideoSendStream::Config* video_config,
88 size_t num_video_streams,
89 size_t num_used_ssrcs,
90 Transport* send_transport);
91 void CreateAudioAndFecSendConfigs(size_t num_audio_streams,
92 size_t num_flexfec_streams,
93 Transport* send_transport);
Sebastian Jansson3bd2c792018-07-13 13:29:03 +020094 void SetAudioConfig(const AudioSendStream::Config& config);
95
96 void SetSendFecConfig(std::vector<uint32_t> video_send_ssrcs);
97 void SetSendUlpFecConfig(VideoSendStream::Config* send_config);
98 void SetReceiveUlpFecConfig(VideoReceiveStream::Config* receive_config);
Stefan Holmer9fea80f2016-01-07 17:43:18 +010099 void CreateSendConfig(size_t num_video_streams,
100 size_t num_audio_streams,
brandtr841de6a2016-11-15 07:10:52 -0800101 size_t num_flexfec_streams,
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100102 Transport* send_transport);
ilnika014cc52017-03-07 04:21:04 -0800103
Sebastian Jansson3bd2c792018-07-13 13:29:03 +0200104 void CreateMatchingVideoReceiveConfigs(
Ilya Nikolaevskiy255d1cd2017-12-21 18:02:59 +0100105 const VideoSendStream::Config& video_send_config,
106 Transport* rtcp_send_transport);
Sebastian Jansson3bd2c792018-07-13 13:29:03 +0200107 void CreateMatchingVideoReceiveConfigs(
108 const VideoSendStream::Config& video_send_config,
109 Transport* rtcp_send_transport,
110 bool send_side_bwe,
Niels Möllercbcbc222018-09-28 09:07:24 +0200111 VideoDecoderFactory* decoder_factory,
Sebastian Jansson3bd2c792018-07-13 13:29:03 +0200112 absl::optional<size_t> decode_sub_stream,
113 bool receiver_reference_time_report,
114 int rtp_history_ms);
115 void AddMatchingVideoReceiveConfigs(
116 std::vector<VideoReceiveStream::Config>* receive_configs,
117 const VideoSendStream::Config& video_send_config,
118 Transport* rtcp_send_transport,
119 bool send_side_bwe,
Niels Möllercbcbc222018-09-28 09:07:24 +0200120 VideoDecoderFactory* decoder_factory,
Sebastian Jansson3bd2c792018-07-13 13:29:03 +0200121 absl::optional<size_t> decode_sub_stream,
122 bool receiver_reference_time_report,
123 int rtp_history_ms);
124
Ilya Nikolaevskiy255d1cd2017-12-21 18:02:59 +0100125 void CreateMatchingAudioAndFecConfigs(Transport* rtcp_send_transport);
Sebastian Jansson3bd2c792018-07-13 13:29:03 +0200126 void CreateMatchingAudioConfigs(Transport* transport, std::string sync_group);
127 static AudioReceiveStream::Config CreateMatchingAudioConfig(
128 const AudioSendStream::Config& send_config,
129 rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
130 Transport* transport,
131 std::string sync_group);
132 void CreateMatchingFecConfig(
133 Transport* transport,
134 const VideoSendStream::Config& video_send_config);
pbos2d566682015-09-28 09:59:31 -0700135 void CreateMatchingReceiveConfigs(Transport* rtcp_send_transport);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000136
perkjfa10b552016-10-02 23:45:26 -0700137 void CreateFrameGeneratorCapturerWithDrift(Clock* drift_clock,
138 float speed,
139 int framerate,
140 int width,
141 int height);
142 void CreateFrameGeneratorCapturer(int framerate, int width, int height);
oprypin92220ff2017-03-23 03:40:03 -0700143 void CreateFakeAudioDevices(
Artem Titov3faa8322018-03-07 14:44:00 +0100144 std::unique_ptr<TestAudioDeviceModule::Capturer> capturer,
145 std::unique_ptr<TestAudioDeviceModule::Renderer> renderer);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000146
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100147 void CreateVideoStreams();
Sebastian Janssonf33905d2018-07-13 09:49:00 +0200148 void CreateVideoSendStreams();
149 void CreateVideoSendStream(const VideoEncoderConfig& encoder_config);
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100150 void CreateAudioStreams();
brandtr841de6a2016-11-15 07:10:52 -0800151 void CreateFlexfecStreams();
eladalonc0d481a2017-08-02 07:39:07 -0700152
Sebastian Janssonf33905d2018-07-13 09:49:00 +0200153 void ConnectVideoSourcesToStreams();
154
eladalonc0d481a2017-08-02 07:39:07 -0700155 void AssociateFlexfecStreamsWithVideoStreams();
156 void DissociateFlexfecStreamsFromVideoStreams();
157
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000158 void Start();
Sebastian Janssonf33905d2018-07-13 09:49:00 +0200159 void StartVideoStreams();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000160 void Stop();
Sebastian Jansson3bd2c792018-07-13 13:29:03 +0200161 void StopVideoStreams();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000162 void DestroyStreams();
Sebastian Janssonf33905d2018-07-13 09:49:00 +0200163 void DestroyVideoSendStreams();
Perba7dc722016-04-19 15:01:23 +0200164 void SetFakeVideoCaptureRotation(VideoRotation rotation);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000165
Sebastian Janssonf33905d2018-07-13 09:49:00 +0200166 void SetVideoDegradation(DegradationPreference preference);
167
168 VideoSendStream::Config* GetVideoSendConfig();
169 void SetVideoSendConfig(const VideoSendStream::Config& config);
170 VideoEncoderConfig* GetVideoEncoderConfig();
171 void SetVideoEncoderConfig(const VideoEncoderConfig& config);
172 VideoSendStream* GetVideoSendStream();
Sebastian Jansson3bd2c792018-07-13 13:29:03 +0200173 FlexfecReceiveStream::Config* GetFlexFecConfig();
Sebastian Janssonf33905d2018-07-13 09:49:00 +0200174
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +0000175 Clock* const clock_;
176
Sebastian Jansson8e6602f2018-07-13 10:43:20 +0200177 std::unique_ptr<webrtc::RtcEventLog> send_event_log_;
178 std::unique_ptr<webrtc::RtcEventLog> recv_event_log_;
kwibergbfefb032016-05-01 14:53:46 -0700179 std::unique_ptr<Call> sender_call_;
180 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_;
Niels Möller1c931c42018-12-18 16:08:11 +0100197 std::vector<std::unique_ptr<rtc::VideoSourceInterface<VideoFrame>>>
198 video_sources_;
Sebastian Jansson3bd2c792018-07-13 13:29:03 +0200199 DegradationPreference degradation_preference_ =
200 DegradationPreference::MAINTAIN_FRAMERATE;
201
202 std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory_;
203
Niels Möller4db138e2018-04-19 09:04:13 +0200204 test::FunctionVideoEncoderFactory fake_encoder_factory_;
205 int fake_encoder_max_bitrate_ = -1;
Niels Möllercbcbc222018-09-28 09:07:24 +0200206 test::FunctionVideoDecoderFactory fake_decoder_factory_;
Jiawei Ouc2ebe212018-11-08 10:02:56 -0800207 std::unique_ptr<VideoBitrateAllocatorFactory> bitrate_allocator_factory_;
Sebastian Jansson3bd2c792018-07-13 13:29:03 +0200208 // Number of simulcast substreams.
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100209 size_t num_video_streams_;
210 size_t num_audio_streams_;
brandtr841de6a2016-11-15 07:10:52 -0800211 size_t num_flexfec_streams_;
Niels Möller2784a032018-03-28 14:16:04 +0200212 rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory_;
213 rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory_;
sakal55d932b2016-09-30 06:19:08 -0700214 test::FakeVideoRenderer fake_renderer_;
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100215
eladalon413ee9a2017-08-22 04:02:52 -0700216 SingleThreadedTaskQueueForTesting task_queue_;
217
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100218 private:
Elad Alond8d32482019-02-18 23:45:57 +0100219 absl::optional<RtpExtension> GetRtpExtensionByUri(
220 const std::string& uri) const;
221
222 void AddRtpExtensionByUri(const std::string& uri,
223 std::vector<RtpExtension>* extensions) const;
224
225 std::vector<RtpExtension> rtp_extensions_;
peaha9cc40b2017-06-29 08:32:09 -0700226 rtc::scoped_refptr<AudioProcessing> apm_send_;
227 rtc::scoped_refptr<AudioProcessing> apm_recv_;
Artem Titov3faa8322018-03-07 14:44:00 +0100228 rtc::scoped_refptr<TestAudioDeviceModule> fake_send_audio_device_;
229 rtc::scoped_refptr<TestAudioDeviceModule> fake_recv_audio_device_;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000230};
231
232class BaseTest : public RtpRtcpObserver {
233 public:
philipele828c962017-03-21 03:24:27 -0700234 BaseTest();
Sebastian Jansson72582242018-07-13 13:19:42 +0200235 explicit BaseTest(int timeout_ms);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000236 virtual ~BaseTest();
237
238 virtual void PerformTest() = 0;
239 virtual bool ShouldCreateReceivers() const = 0;
240
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100241 virtual size_t GetNumVideoStreams() const;
242 virtual size_t GetNumAudioStreams() const;
brandtr841de6a2016-11-15 07:10:52 -0800243 virtual size_t GetNumFlexfecStreams() const;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000244
Artem Titov3faa8322018-03-07 14:44:00 +0100245 virtual std::unique_ptr<TestAudioDeviceModule::Capturer> CreateCapturer();
246 virtual std::unique_ptr<TestAudioDeviceModule::Renderer> CreateRenderer();
247 virtual void OnFakeAudioDevicesCreated(
248 TestAudioDeviceModule* send_audio_device,
249 TestAudioDeviceModule* recv_audio_device);
oprypin92220ff2017-03-23 03:40:03 -0700250
Niels Möllerde8e6e62018-11-13 15:10:33 +0100251 virtual void ModifySenderBitrateConfig(BitrateConstraints* bitrate_config);
252 virtual void ModifyReceiverBitrateConfig(BitrateConstraints* bitrate_config);
Sebastian Jansson72582242018-07-13 13:19:42 +0200253
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000254 virtual void OnCallsCreated(Call* sender_call, Call* receiver_call);
stefane74eef12016-01-08 06:47:13 -0800255
eladalon413ee9a2017-08-22 04:02:52 -0700256 virtual test::PacketTransport* CreateSendTransport(
257 SingleThreadedTaskQueueForTesting* task_queue,
258 Call* sender_call);
259 virtual test::PacketTransport* CreateReceiveTransport(
260 SingleThreadedTaskQueueForTesting* task_queue);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000261
stefanff483612015-12-21 03:14:00 -0800262 virtual void ModifyVideoConfigs(
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000263 VideoSendStream::Config* send_config,
264 std::vector<VideoReceiveStream::Config>* receive_configs,
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +0000265 VideoEncoderConfig* encoder_config);
perkjfa10b552016-10-02 23:45:26 -0700266 virtual void ModifyVideoCaptureStartResolution(int* width,
267 int* heigt,
268 int* frame_rate);
Åsa Perssoncb7eddb2018-11-05 14:11:44 +0100269 virtual void ModifyVideoDegradationPreference(
270 DegradationPreference* degradation_preference);
271
stefanff483612015-12-21 03:14:00 -0800272 virtual void OnVideoStreamsCreated(
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000273 VideoSendStream* send_stream,
274 const std::vector<VideoReceiveStream*>& receive_streams);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000275
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100276 virtual void ModifyAudioConfigs(
277 AudioSendStream::Config* send_config,
278 std::vector<AudioReceiveStream::Config>* receive_configs);
279 virtual void OnAudioStreamsCreated(
280 AudioSendStream* send_stream,
281 const std::vector<AudioReceiveStream*>& receive_streams);
282
brandtr841de6a2016-11-15 07:10:52 -0800283 virtual void ModifyFlexfecConfigs(
284 std::vector<FlexfecReceiveStream::Config>* receive_configs);
285 virtual void OnFlexfecStreamsCreated(
286 const std::vector<FlexfecReceiveStream*>& receive_streams);
287
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000288 virtual void OnFrameGeneratorCapturerCreated(
289 FrameGeneratorCapturer* frame_generator_capturer);
skvlad11a9cbf2016-10-07 11:53:05 -0700290
Fredrik Solenberg73276ad2017-09-14 14:46:47 +0200291 virtual void OnStreamsStopped();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000292};
293
294class SendTest : public BaseTest {
295 public:
Sebastian Jansson72582242018-07-13 13:19:42 +0200296 explicit SendTest(int timeout_ms);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000297
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000298 bool ShouldCreateReceivers() const override;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000299};
300
301class EndToEndTest : public BaseTest {
302 public:
philipele828c962017-03-21 03:24:27 -0700303 EndToEndTest();
Sebastian Jansson72582242018-07-13 13:19:42 +0200304 explicit EndToEndTest(int timeout_ms);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000305
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000306 bool ShouldCreateReceivers() const override;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000307};
308
309} // namespace test
310} // namespace webrtc
311
Mirko Bonadei92ea95e2017-09-15 06:47:31 +0200312#endif // TEST_CALL_TEST_H_