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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
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"
Artem Titov3faa8322018-03-07 14:44:00 +010019#include "modules/audio_device/include/test_audio_device.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020020#include "test/encoder_settings.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020021#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 {
29namespace test {
30
31class BaseTest;
32
33class CallTest : public ::testing::Test {
34 public:
35 CallTest();
Stefan Holmer9fea80f2016-01-07 17:43:18 +010036 virtual ~CallTest();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000037
Ilya Nikolaevskiy255d1cd2017-12-21 18:02:59 +010038 static constexpr size_t kNumSsrcs = 6;
39 static const int kNumSimulcastStreams = 3;
perkjfa10b552016-10-02 23:45:26 -070040 static const int kDefaultWidth = 320;
41 static const int kDefaultHeight = 180;
42 static const int kDefaultFramerate = 30;
Peter Boström5811a392015-12-10 13:02:50 +010043 static const int kDefaultTimeoutMs;
44 static const int kLongTimeoutMs;
Stefan Holmer9fea80f2016-01-07 17:43:18 +010045 static const uint8_t kVideoSendPayloadType;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000046 static const uint8_t kSendRtxPayloadType;
Stefan Holmer9fea80f2016-01-07 17:43:18 +010047 static const uint8_t kFakeVideoSendPayloadType;
stefan@webrtc.org01581da2014-09-04 06:48:14 +000048 static const uint8_t kRedPayloadType;
Shao Changbine62202f2015-04-21 20:24:50 +080049 static const uint8_t kRtxRedPayloadType;
stefan@webrtc.org01581da2014-09-04 06:48:14 +000050 static const uint8_t kUlpfecPayloadType;
brandtr841de6a2016-11-15 07:10:52 -080051 static const uint8_t kFlexfecPayloadType;
Stefan Holmer9fea80f2016-01-07 17:43:18 +010052 static const uint8_t kAudioSendPayloadType;
ilnik863f03b2017-07-11 02:38:36 -070053 static const uint8_t kPayloadTypeH264;
54 static const uint8_t kPayloadTypeVP8;
55 static const uint8_t kPayloadTypeVP9;
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +000056 static const uint32_t kSendRtxSsrcs[kNumSsrcs];
Stefan Holmer9fea80f2016-01-07 17:43:18 +010057 static const uint32_t kVideoSendSsrcs[kNumSsrcs];
58 static const uint32_t kAudioSendSsrc;
brandtr841de6a2016-11-15 07:10:52 -080059 static const uint32_t kFlexfecSendSsrc;
Stefan Holmer9fea80f2016-01-07 17:43:18 +010060 static const uint32_t kReceiverLocalVideoSsrc;
61 static const uint32_t kReceiverLocalAudioSsrc;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000062 static const int kNackRtpHistoryMs;
sprangd2702ef2017-07-10 08:41:10 -070063 static const uint8_t kDefaultKeepalivePayloadType;
minyue20c84cc2017-04-10 16:57:57 -070064 static const std::map<uint8_t, MediaType> payload_type_map_;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000065
66 protected:
Fredrik Solenberg8f5787a2018-01-11 13:52:30 +010067 // RunBaseTest overwrites the audio_state of the send and receive Call configs
68 // to simplify test code.
stefane74eef12016-01-08 06:47:13 -080069 void RunBaseTest(BaseTest* test);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000070
71 void CreateCalls(const Call::Config& sender_config,
72 const Call::Config& receiver_config);
73 void CreateSenderCall(const Call::Config& config);
74 void CreateReceiverCall(const Call::Config& config);
Fredrik Solenberg4f4ec0a2015-10-22 10:49:27 +020075 void DestroyCalls();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000076
Ilya Nikolaevskiy255d1cd2017-12-21 18:02:59 +010077 void CreateVideoSendConfig(VideoSendStream::Config* video_config,
78 size_t num_video_streams,
79 size_t num_used_ssrcs,
80 Transport* send_transport);
81 void CreateAudioAndFecSendConfigs(size_t num_audio_streams,
82 size_t num_flexfec_streams,
83 Transport* send_transport);
Stefan Holmer9fea80f2016-01-07 17:43:18 +010084 void CreateSendConfig(size_t num_video_streams,
85 size_t num_audio_streams,
brandtr841de6a2016-11-15 07:10:52 -080086 size_t num_flexfec_streams,
Stefan Holmer9fea80f2016-01-07 17:43:18 +010087 Transport* send_transport);
ilnika014cc52017-03-07 04:21:04 -080088
Ilya Nikolaevskiy255d1cd2017-12-21 18:02:59 +010089 std::vector<VideoReceiveStream::Config> CreateMatchingVideoReceiveConfigs(
90 const VideoSendStream::Config& video_send_config,
91 Transport* rtcp_send_transport);
92 void CreateMatchingAudioAndFecConfigs(Transport* rtcp_send_transport);
pbos2d566682015-09-28 09:59:31 -070093 void CreateMatchingReceiveConfigs(Transport* rtcp_send_transport);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000094
perkjfa10b552016-10-02 23:45:26 -070095 void CreateFrameGeneratorCapturerWithDrift(Clock* drift_clock,
96 float speed,
97 int framerate,
98 int width,
99 int height);
100 void CreateFrameGeneratorCapturer(int framerate, int width, int height);
oprypin92220ff2017-03-23 03:40:03 -0700101 void CreateFakeAudioDevices(
Artem Titov3faa8322018-03-07 14:44:00 +0100102 std::unique_ptr<TestAudioDeviceModule::Capturer> capturer,
103 std::unique_ptr<TestAudioDeviceModule::Renderer> renderer);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000104
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100105 void CreateVideoStreams();
106 void CreateAudioStreams();
brandtr841de6a2016-11-15 07:10:52 -0800107 void CreateFlexfecStreams();
eladalonc0d481a2017-08-02 07:39:07 -0700108
109 void AssociateFlexfecStreamsWithVideoStreams();
110 void DissociateFlexfecStreamsFromVideoStreams();
111
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000112 void Start();
113 void Stop();
114 void DestroyStreams();
Perba7dc722016-04-19 15:01:23 +0200115 void SetFakeVideoCaptureRotation(VideoRotation rotation);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000116
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +0000117 Clock* const clock_;
118
philipel4fb651d2017-04-10 03:54:05 -0700119 std::unique_ptr<webrtc::RtcEventLog> event_log_;
kwibergbfefb032016-05-01 14:53:46 -0700120 std::unique_ptr<Call> sender_call_;
sprangdb2a9fc2017-08-09 06:42:32 -0700121 RtpTransportControllerSend* sender_call_transport_controller_;
kwibergbfefb032016-05-01 14:53:46 -0700122 std::unique_ptr<PacketTransport> send_transport_;
stefanff483612015-12-21 03:14:00 -0800123 VideoSendStream::Config video_send_config_;
124 VideoEncoderConfig video_encoder_config_;
125 VideoSendStream* video_send_stream_;
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100126 AudioSendStream::Config audio_send_config_;
127 AudioSendStream* audio_send_stream_;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000128
kwibergbfefb032016-05-01 14:53:46 -0700129 std::unique_ptr<Call> receiver_call_;
130 std::unique_ptr<PacketTransport> receive_transport_;
stefanff483612015-12-21 03:14:00 -0800131 std::vector<VideoReceiveStream::Config> video_receive_configs_;
132 std::vector<VideoReceiveStream*> video_receive_streams_;
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100133 std::vector<AudioReceiveStream::Config> audio_receive_configs_;
134 std::vector<AudioReceiveStream*> audio_receive_streams_;
brandtr841de6a2016-11-15 07:10:52 -0800135 std::vector<FlexfecReceiveStream::Config> flexfec_receive_configs_;
136 std::vector<FlexfecReceiveStream*> flexfec_receive_streams_;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000137
kwibergbfefb032016-05-01 14:53:46 -0700138 std::unique_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000139 test::FakeEncoder fake_encoder_;
kwiberg4a206a92016-03-31 10:24:26 -0700140 std::vector<std::unique_ptr<VideoDecoder>> allocated_decoders_;
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100141 size_t num_video_streams_;
142 size_t num_audio_streams_;
brandtr841de6a2016-11-15 07:10:52 -0800143 size_t num_flexfec_streams_;
Rasmus Brandt31027342017-09-29 13:48:12 +0000144 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
145 rtc::scoped_refptr<AudioEncoderFactory> encoder_factory_;
sakal55d932b2016-09-30 06:19:08 -0700146 test::FakeVideoRenderer fake_renderer_;
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100147
eladalon413ee9a2017-08-22 04:02:52 -0700148 SingleThreadedTaskQueueForTesting task_queue_;
149
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100150 private:
peaha9cc40b2017-06-29 08:32:09 -0700151 rtc::scoped_refptr<AudioProcessing> apm_send_;
152 rtc::scoped_refptr<AudioProcessing> apm_recv_;
Artem Titov3faa8322018-03-07 14:44:00 +0100153 rtc::scoped_refptr<TestAudioDeviceModule> fake_send_audio_device_;
154 rtc::scoped_refptr<TestAudioDeviceModule> fake_recv_audio_device_;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000155};
156
157class BaseTest : public RtpRtcpObserver {
158 public:
philipele828c962017-03-21 03:24:27 -0700159 BaseTest();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000160 explicit BaseTest(unsigned int timeout_ms);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000161 virtual ~BaseTest();
162
163 virtual void PerformTest() = 0;
164 virtual bool ShouldCreateReceivers() const = 0;
165
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100166 virtual size_t GetNumVideoStreams() const;
167 virtual size_t GetNumAudioStreams() const;
brandtr841de6a2016-11-15 07:10:52 -0800168 virtual size_t GetNumFlexfecStreams() const;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000169
Artem Titov3faa8322018-03-07 14:44:00 +0100170 virtual std::unique_ptr<TestAudioDeviceModule::Capturer> CreateCapturer();
171 virtual std::unique_ptr<TestAudioDeviceModule::Renderer> CreateRenderer();
172 virtual void OnFakeAudioDevicesCreated(
173 TestAudioDeviceModule* send_audio_device,
174 TestAudioDeviceModule* recv_audio_device);
oprypin92220ff2017-03-23 03:40:03 -0700175
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000176 virtual Call::Config GetSenderCallConfig();
177 virtual Call::Config GetReceiverCallConfig();
sprangdb2a9fc2017-08-09 06:42:32 -0700178 virtual void OnRtpTransportControllerSendCreated(
179 RtpTransportControllerSend* controller);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000180 virtual void OnCallsCreated(Call* sender_call, Call* receiver_call);
stefane74eef12016-01-08 06:47:13 -0800181
eladalon413ee9a2017-08-22 04:02:52 -0700182 virtual test::PacketTransport* CreateSendTransport(
183 SingleThreadedTaskQueueForTesting* task_queue,
184 Call* sender_call);
185 virtual test::PacketTransport* CreateReceiveTransport(
186 SingleThreadedTaskQueueForTesting* task_queue);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000187
stefanff483612015-12-21 03:14:00 -0800188 virtual void ModifyVideoConfigs(
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000189 VideoSendStream::Config* send_config,
190 std::vector<VideoReceiveStream::Config>* receive_configs,
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +0000191 VideoEncoderConfig* encoder_config);
perkjfa10b552016-10-02 23:45:26 -0700192 virtual void ModifyVideoCaptureStartResolution(int* width,
193 int* heigt,
194 int* frame_rate);
stefanff483612015-12-21 03:14:00 -0800195 virtual void OnVideoStreamsCreated(
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000196 VideoSendStream* send_stream,
197 const std::vector<VideoReceiveStream*>& receive_streams);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000198
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100199 virtual void ModifyAudioConfigs(
200 AudioSendStream::Config* send_config,
201 std::vector<AudioReceiveStream::Config>* receive_configs);
202 virtual void OnAudioStreamsCreated(
203 AudioSendStream* send_stream,
204 const std::vector<AudioReceiveStream*>& receive_streams);
205
brandtr841de6a2016-11-15 07:10:52 -0800206 virtual void ModifyFlexfecConfigs(
207 std::vector<FlexfecReceiveStream::Config>* receive_configs);
208 virtual void OnFlexfecStreamsCreated(
209 const std::vector<FlexfecReceiveStream*>& receive_streams);
210
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000211 virtual void OnFrameGeneratorCapturerCreated(
212 FrameGeneratorCapturer* frame_generator_capturer);
skvlad11a9cbf2016-10-07 11:53:05 -0700213
Fredrik Solenberg73276ad2017-09-14 14:46:47 +0200214 virtual void OnStreamsStopped();
oprypin92220ff2017-03-23 03:40:03 -0700215
philipel4fb651d2017-04-10 03:54:05 -0700216 std::unique_ptr<webrtc::RtcEventLog> event_log_;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000217};
218
219class SendTest : public BaseTest {
220 public:
221 explicit SendTest(unsigned int timeout_ms);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000222
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000223 bool ShouldCreateReceivers() const override;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000224};
225
226class EndToEndTest : public BaseTest {
227 public:
philipele828c962017-03-21 03:24:27 -0700228 EndToEndTest();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000229 explicit EndToEndTest(unsigned int timeout_ms);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000230
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000231 bool ShouldCreateReceivers() const override;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000232};
233
234} // namespace test
235} // namespace webrtc
236
Mirko Bonadei92ea95e2017-09-15 06:47:31 +0200237#endif // TEST_CALL_TEST_H_