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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 Chapovalov83bbe912019-08-07 12:24:53 +020019#include "api/rtc_event_log/rtc_event_log.h"
Danil Chapovalov44db4362019-09-30 04:16:28 +020020#include "api/task_queue/task_queue_base.h"
Danil Chapovalova92e6242019-04-18 10:58:56 +020021#include "api/task_queue/task_queue_factory.h"
Danil Chapovalov99b71df2018-10-26 15:57:48 +020022#include "api/test/video/function_video_decoder_factory.h"
23#include "api/test/video/function_video_encoder_factory.h"
Erik Språng014dd3c2019-11-28 13:44:25 +010024#include "api/transport/field_trial_based_config.h"
Jiawei Ouc2ebe212018-11-08 10:02:56 -080025#include "api/video/video_bitrate_allocator_factory.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020026#include "call/call.h"
Artem Titov3faa8322018-03-07 14:44:00 +010027#include "modules/audio_device/include/test_audio_device.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020028#include "test/encoder_settings.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020029#include "test/fake_decoder.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020030#include "test/fake_videorenderer.h"
Ilya Nikolaevskiyb0588e62018-08-27 14:12:27 +020031#include "test/fake_vp8_encoder.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020032#include "test/frame_generator_capturer.h"
33#include "test/rtp_rtcp_observer.h"
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000034
35namespace webrtc {
36namespace test {
37
38class BaseTest;
39
40class CallTest : public ::testing::Test {
41 public:
42 CallTest();
Stefan Holmer9fea80f2016-01-07 17:43:18 +010043 virtual ~CallTest();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000044
Ilya Nikolaevskiy255d1cd2017-12-21 18:02:59 +010045 static constexpr size_t kNumSsrcs = 6;
46 static const int kNumSimulcastStreams = 3;
perkjfa10b552016-10-02 23:45:26 -070047 static const int kDefaultWidth = 320;
48 static const int kDefaultHeight = 180;
49 static const int kDefaultFramerate = 30;
Peter Boström5811a392015-12-10 13:02:50 +010050 static const int kDefaultTimeoutMs;
51 static const int kLongTimeoutMs;
Ilya Nikolaevskiy465a5d92018-03-16 11:12:06 +010052 enum classPayloadTypes : uint8_t {
53 kSendRtxPayloadType = 98,
54 kRtxRedPayloadType = 99,
55 kVideoSendPayloadType = 100,
56 kAudioSendPayloadType = 103,
57 kRedPayloadType = 118,
58 kUlpfecPayloadType = 119,
59 kFlexfecPayloadType = 120,
60 kPayloadTypeH264 = 122,
61 kPayloadTypeVP8 = 123,
62 kPayloadTypeVP9 = 124,
Rasmus Brandt5894b6a2019-06-13 16:28:14 +020063 kPayloadTypeGeneric = 125,
64 kFakeVideoSendPayloadType = 126,
Ilya Nikolaevskiy465a5d92018-03-16 11:12:06 +010065 };
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +000066 static const uint32_t kSendRtxSsrcs[kNumSsrcs];
Stefan Holmer9fea80f2016-01-07 17:43:18 +010067 static const uint32_t kVideoSendSsrcs[kNumSsrcs];
68 static const uint32_t kAudioSendSsrc;
brandtr841de6a2016-11-15 07:10:52 -080069 static const uint32_t kFlexfecSendSsrc;
Stefan Holmer9fea80f2016-01-07 17:43:18 +010070 static const uint32_t kReceiverLocalVideoSsrc;
71 static const uint32_t kReceiverLocalAudioSsrc;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000072 static const int kNackRtpHistoryMs;
minyue20c84cc2017-04-10 16:57:57 -070073 static const std::map<uint8_t, MediaType> payload_type_map_;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000074
75 protected:
Elad Alond8d32482019-02-18 23:45:57 +010076 void RegisterRtpExtension(const RtpExtension& extension);
77
Fredrik Solenberg8f5787a2018-01-11 13:52:30 +010078 // RunBaseTest overwrites the audio_state of the send and receive Call configs
79 // to simplify test code.
stefane74eef12016-01-08 06:47:13 -080080 void RunBaseTest(BaseTest* test);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000081
Sebastian Jansson8e6602f2018-07-13 10:43:20 +020082 void CreateCalls();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000083 void CreateCalls(const Call::Config& sender_config,
84 const Call::Config& receiver_config);
Sebastian Jansson8e6602f2018-07-13 10:43:20 +020085 void CreateSenderCall();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000086 void CreateSenderCall(const Call::Config& config);
87 void CreateReceiverCall(const Call::Config& config);
Fredrik Solenberg4f4ec0a2015-10-22 10:49:27 +020088 void DestroyCalls();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000089
Ilya Nikolaevskiy255d1cd2017-12-21 18:02:59 +010090 void CreateVideoSendConfig(VideoSendStream::Config* video_config,
91 size_t num_video_streams,
92 size_t num_used_ssrcs,
93 Transport* send_transport);
94 void CreateAudioAndFecSendConfigs(size_t num_audio_streams,
95 size_t num_flexfec_streams,
96 Transport* send_transport);
Sebastian Jansson3bd2c792018-07-13 13:29:03 +020097 void SetAudioConfig(const AudioSendStream::Config& config);
98
99 void SetSendFecConfig(std::vector<uint32_t> video_send_ssrcs);
100 void SetSendUlpFecConfig(VideoSendStream::Config* send_config);
101 void SetReceiveUlpFecConfig(VideoReceiveStream::Config* receive_config);
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100102 void CreateSendConfig(size_t num_video_streams,
103 size_t num_audio_streams,
brandtr841de6a2016-11-15 07:10:52 -0800104 size_t num_flexfec_streams,
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100105 Transport* send_transport);
ilnika014cc52017-03-07 04:21:04 -0800106
Sebastian Jansson3bd2c792018-07-13 13:29:03 +0200107 void CreateMatchingVideoReceiveConfigs(
Ilya Nikolaevskiy255d1cd2017-12-21 18:02:59 +0100108 const VideoSendStream::Config& video_send_config,
109 Transport* rtcp_send_transport);
Sebastian Jansson3bd2c792018-07-13 13:29:03 +0200110 void CreateMatchingVideoReceiveConfigs(
111 const VideoSendStream::Config& video_send_config,
112 Transport* rtcp_send_transport,
113 bool send_side_bwe,
Niels Möllercbcbc222018-09-28 09:07:24 +0200114 VideoDecoderFactory* decoder_factory,
Sebastian Jansson3bd2c792018-07-13 13:29:03 +0200115 absl::optional<size_t> decode_sub_stream,
116 bool receiver_reference_time_report,
117 int rtp_history_ms);
118 void AddMatchingVideoReceiveConfigs(
119 std::vector<VideoReceiveStream::Config>* receive_configs,
120 const VideoSendStream::Config& video_send_config,
121 Transport* rtcp_send_transport,
122 bool send_side_bwe,
Niels Möllercbcbc222018-09-28 09:07:24 +0200123 VideoDecoderFactory* decoder_factory,
Sebastian Jansson3bd2c792018-07-13 13:29:03 +0200124 absl::optional<size_t> decode_sub_stream,
125 bool receiver_reference_time_report,
126 int rtp_history_ms);
127
Ilya Nikolaevskiy255d1cd2017-12-21 18:02:59 +0100128 void CreateMatchingAudioAndFecConfigs(Transport* rtcp_send_transport);
Sebastian Jansson3bd2c792018-07-13 13:29:03 +0200129 void CreateMatchingAudioConfigs(Transport* transport, std::string sync_group);
130 static AudioReceiveStream::Config CreateMatchingAudioConfig(
131 const AudioSendStream::Config& send_config,
132 rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
133 Transport* transport,
134 std::string sync_group);
135 void CreateMatchingFecConfig(
136 Transport* transport,
137 const VideoSendStream::Config& video_send_config);
pbos2d566682015-09-28 09:59:31 -0700138 void CreateMatchingReceiveConfigs(Transport* rtcp_send_transport);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000139
perkjfa10b552016-10-02 23:45:26 -0700140 void CreateFrameGeneratorCapturerWithDrift(Clock* drift_clock,
141 float speed,
142 int framerate,
143 int width,
144 int height);
145 void CreateFrameGeneratorCapturer(int framerate, int width, int height);
oprypin92220ff2017-03-23 03:40:03 -0700146 void CreateFakeAudioDevices(
Artem Titov3faa8322018-03-07 14:44:00 +0100147 std::unique_ptr<TestAudioDeviceModule::Capturer> capturer,
148 std::unique_ptr<TestAudioDeviceModule::Renderer> renderer);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000149
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100150 void CreateVideoStreams();
Sebastian Janssonf33905d2018-07-13 09:49:00 +0200151 void CreateVideoSendStreams();
152 void CreateVideoSendStream(const VideoEncoderConfig& encoder_config);
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100153 void CreateAudioStreams();
brandtr841de6a2016-11-15 07:10:52 -0800154 void CreateFlexfecStreams();
eladalonc0d481a2017-08-02 07:39:07 -0700155
Sebastian Janssonf33905d2018-07-13 09:49:00 +0200156 void ConnectVideoSourcesToStreams();
157
eladalonc0d481a2017-08-02 07:39:07 -0700158 void AssociateFlexfecStreamsWithVideoStreams();
159 void DissociateFlexfecStreamsFromVideoStreams();
160
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000161 void Start();
Sebastian Janssonf33905d2018-07-13 09:49:00 +0200162 void StartVideoStreams();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000163 void Stop();
Sebastian Jansson3bd2c792018-07-13 13:29:03 +0200164 void StopVideoStreams();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000165 void DestroyStreams();
Sebastian Janssonf33905d2018-07-13 09:49:00 +0200166 void DestroyVideoSendStreams();
Perba7dc722016-04-19 15:01:23 +0200167 void SetFakeVideoCaptureRotation(VideoRotation rotation);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000168
Sebastian Janssonf33905d2018-07-13 09:49:00 +0200169 void SetVideoDegradation(DegradationPreference preference);
170
171 VideoSendStream::Config* GetVideoSendConfig();
172 void SetVideoSendConfig(const VideoSendStream::Config& config);
173 VideoEncoderConfig* GetVideoEncoderConfig();
174 void SetVideoEncoderConfig(const VideoEncoderConfig& config);
175 VideoSendStream* GetVideoSendStream();
Sebastian Jansson3bd2c792018-07-13 13:29:03 +0200176 FlexfecReceiveStream::Config* GetFlexFecConfig();
Danil Chapovalov1b668902019-11-13 11:19:53 +0100177 TaskQueueBase* task_queue() { return task_queue_.get(); }
Sebastian Janssonf33905d2018-07-13 09:49:00 +0200178
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +0000179 Clock* const clock_;
Erik Språng014dd3c2019-11-28 13:44:25 +0100180 const FieldTrialBasedConfig field_trials_;
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +0000181
Danil Chapovalova92e6242019-04-18 10:58:56 +0200182 std::unique_ptr<TaskQueueFactory> task_queue_factory_;
Sebastian Jansson8e6602f2018-07-13 10:43:20 +0200183 std::unique_ptr<webrtc::RtcEventLog> send_event_log_;
184 std::unique_ptr<webrtc::RtcEventLog> recv_event_log_;
kwibergbfefb032016-05-01 14:53:46 -0700185 std::unique_ptr<Call> sender_call_;
186 std::unique_ptr<PacketTransport> send_transport_;
Sebastian Jansson3bd2c792018-07-13 13:29:03 +0200187 std::vector<VideoSendStream::Config> video_send_configs_;
188 std::vector<VideoEncoderConfig> video_encoder_configs_;
189 std::vector<VideoSendStream*> video_send_streams_;
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100190 AudioSendStream::Config audio_send_config_;
191 AudioSendStream* audio_send_stream_;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000192
kwibergbfefb032016-05-01 14:53:46 -0700193 std::unique_ptr<Call> receiver_call_;
194 std::unique_ptr<PacketTransport> receive_transport_;
stefanff483612015-12-21 03:14:00 -0800195 std::vector<VideoReceiveStream::Config> video_receive_configs_;
196 std::vector<VideoReceiveStream*> video_receive_streams_;
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100197 std::vector<AudioReceiveStream::Config> audio_receive_configs_;
198 std::vector<AudioReceiveStream*> audio_receive_streams_;
brandtr841de6a2016-11-15 07:10:52 -0800199 std::vector<FlexfecReceiveStream::Config> flexfec_receive_configs_;
200 std::vector<FlexfecReceiveStream*> flexfec_receive_streams_;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000201
Sebastian Jansson3bd2c792018-07-13 13:29:03 +0200202 test::FrameGeneratorCapturer* frame_generator_capturer_;
Niels Möller1c931c42018-12-18 16:08:11 +0100203 std::vector<std::unique_ptr<rtc::VideoSourceInterface<VideoFrame>>>
204 video_sources_;
Sebastian Jansson3bd2c792018-07-13 13:29:03 +0200205 DegradationPreference degradation_preference_ =
206 DegradationPreference::MAINTAIN_FRAMERATE;
207
208 std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory_;
Ying Wangcab77fd2019-04-16 11:12:49 +0200209 std::unique_ptr<NetworkStatePredictorFactoryInterface>
210 network_state_predictor_factory_;
Sebastian Jansson1391ed22019-04-30 14:23:51 +0200211 std::unique_ptr<NetworkControllerFactoryInterface>
212 network_controller_factory_;
Sebastian Jansson3bd2c792018-07-13 13:29:03 +0200213
Niels Möller4db138e2018-04-19 09:04:13 +0200214 test::FunctionVideoEncoderFactory fake_encoder_factory_;
215 int fake_encoder_max_bitrate_ = -1;
Niels Möllercbcbc222018-09-28 09:07:24 +0200216 test::FunctionVideoDecoderFactory fake_decoder_factory_;
Jiawei Ouc2ebe212018-11-08 10:02:56 -0800217 std::unique_ptr<VideoBitrateAllocatorFactory> bitrate_allocator_factory_;
Sebastian Jansson3bd2c792018-07-13 13:29:03 +0200218 // Number of simulcast substreams.
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100219 size_t num_video_streams_;
220 size_t num_audio_streams_;
brandtr841de6a2016-11-15 07:10:52 -0800221 size_t num_flexfec_streams_;
Niels Möller2784a032018-03-28 14:16:04 +0200222 rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory_;
223 rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory_;
sakal55d932b2016-09-30 06:19:08 -0700224 test::FakeVideoRenderer fake_renderer_;
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100225
eladalon413ee9a2017-08-22 04:02:52 -0700226
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100227 private:
Elad Alond8d32482019-02-18 23:45:57 +0100228 absl::optional<RtpExtension> GetRtpExtensionByUri(
229 const std::string& uri) const;
230
231 void AddRtpExtensionByUri(const std::string& uri,
232 std::vector<RtpExtension>* extensions) const;
233
Danil Chapovalov1b668902019-11-13 11:19:53 +0100234 std::unique_ptr<TaskQueueBase, TaskQueueDeleter> task_queue_;
Elad Alond8d32482019-02-18 23:45:57 +0100235 std::vector<RtpExtension> rtp_extensions_;
peaha9cc40b2017-06-29 08:32:09 -0700236 rtc::scoped_refptr<AudioProcessing> apm_send_;
237 rtc::scoped_refptr<AudioProcessing> apm_recv_;
Artem Titov3faa8322018-03-07 14:44:00 +0100238 rtc::scoped_refptr<TestAudioDeviceModule> fake_send_audio_device_;
239 rtc::scoped_refptr<TestAudioDeviceModule> fake_recv_audio_device_;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000240};
241
242class BaseTest : public RtpRtcpObserver {
243 public:
philipele828c962017-03-21 03:24:27 -0700244 BaseTest();
Sebastian Jansson72582242018-07-13 13:19:42 +0200245 explicit BaseTest(int timeout_ms);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000246 virtual ~BaseTest();
247
248 virtual void PerformTest() = 0;
249 virtual bool ShouldCreateReceivers() const = 0;
250
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100251 virtual size_t GetNumVideoStreams() const;
252 virtual size_t GetNumAudioStreams() const;
brandtr841de6a2016-11-15 07:10:52 -0800253 virtual size_t GetNumFlexfecStreams() const;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000254
Artem Titov3faa8322018-03-07 14:44:00 +0100255 virtual std::unique_ptr<TestAudioDeviceModule::Capturer> CreateCapturer();
256 virtual std::unique_ptr<TestAudioDeviceModule::Renderer> CreateRenderer();
257 virtual void OnFakeAudioDevicesCreated(
258 TestAudioDeviceModule* send_audio_device,
259 TestAudioDeviceModule* recv_audio_device);
oprypin92220ff2017-03-23 03:40:03 -0700260
Niels Möllerde8e6e62018-11-13 15:10:33 +0100261 virtual void ModifySenderBitrateConfig(BitrateConstraints* bitrate_config);
262 virtual void ModifyReceiverBitrateConfig(BitrateConstraints* bitrate_config);
Sebastian Jansson72582242018-07-13 13:19:42 +0200263
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000264 virtual void OnCallsCreated(Call* sender_call, Call* receiver_call);
stefane74eef12016-01-08 06:47:13 -0800265
Danil Chapovalov44db4362019-09-30 04:16:28 +0200266 virtual std::unique_ptr<test::PacketTransport> CreateSendTransport(
267 TaskQueueBase* task_queue,
eladalon413ee9a2017-08-22 04:02:52 -0700268 Call* sender_call);
Danil Chapovalov44db4362019-09-30 04:16:28 +0200269 virtual std::unique_ptr<test::PacketTransport> CreateReceiveTransport(
270 TaskQueueBase* task_queue);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000271
stefanff483612015-12-21 03:14:00 -0800272 virtual void ModifyVideoConfigs(
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000273 VideoSendStream::Config* send_config,
274 std::vector<VideoReceiveStream::Config>* receive_configs,
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +0000275 VideoEncoderConfig* encoder_config);
perkjfa10b552016-10-02 23:45:26 -0700276 virtual void ModifyVideoCaptureStartResolution(int* width,
277 int* heigt,
278 int* frame_rate);
Åsa Perssoncb7eddb2018-11-05 14:11:44 +0100279 virtual void ModifyVideoDegradationPreference(
280 DegradationPreference* degradation_preference);
281
stefanff483612015-12-21 03:14:00 -0800282 virtual void OnVideoStreamsCreated(
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000283 VideoSendStream* send_stream,
284 const std::vector<VideoReceiveStream*>& receive_streams);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000285
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100286 virtual void ModifyAudioConfigs(
287 AudioSendStream::Config* send_config,
288 std::vector<AudioReceiveStream::Config>* receive_configs);
289 virtual void OnAudioStreamsCreated(
290 AudioSendStream* send_stream,
291 const std::vector<AudioReceiveStream*>& receive_streams);
292
brandtr841de6a2016-11-15 07:10:52 -0800293 virtual void ModifyFlexfecConfigs(
294 std::vector<FlexfecReceiveStream::Config>* receive_configs);
295 virtual void OnFlexfecStreamsCreated(
296 const std::vector<FlexfecReceiveStream*>& receive_streams);
297
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000298 virtual void OnFrameGeneratorCapturerCreated(
299 FrameGeneratorCapturer* frame_generator_capturer);
skvlad11a9cbf2016-10-07 11:53:05 -0700300
Fredrik Solenberg73276ad2017-09-14 14:46:47 +0200301 virtual void OnStreamsStopped();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000302};
303
304class SendTest : public BaseTest {
305 public:
Sebastian Jansson72582242018-07-13 13:19:42 +0200306 explicit SendTest(int timeout_ms);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000307
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000308 bool ShouldCreateReceivers() const override;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000309};
310
311class EndToEndTest : public BaseTest {
312 public:
philipele828c962017-03-21 03:24:27 -0700313 EndToEndTest();
Sebastian Jansson72582242018-07-13 13:19:42 +0200314 explicit EndToEndTest(int timeout_ms);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000315
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000316 bool ShouldCreateReceivers() const override;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000317};
318
319} // namespace test
320} // namespace webrtc
321
Mirko Bonadei92ea95e2017-09-15 06:47:31 +0200322#endif // TEST_CALL_TEST_H_