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pbos@webrtc.org994d0b72014-06-27 08:47:521/*
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 */
Stefan Holmer9fea80f2016-01-07 16:43:1810#include "webrtc/base/checks.h"
11#include "webrtc/common.h"
12#include "webrtc/config.h"
pbos@webrtc.org994d0b72014-06-27 08:47:5213#include "webrtc/test/call_test.h"
pbos@webrtc.org994d0b72014-06-27 08:47:5214#include "webrtc/test/encoder_settings.h"
Stefan Holmer9fea80f2016-01-07 16:43:1815#include "webrtc/test/testsupport/fileutils.h"
16#include "webrtc/voice_engine/include/voe_base.h"
17#include "webrtc/voice_engine/include/voe_codec.h"
18#include "webrtc/voice_engine/include/voe_network.h"
pbos@webrtc.org994d0b72014-06-27 08:47:5219
20namespace webrtc {
21namespace test {
22
Guo-wei Shieh2c370782015-04-08 20:00:1023namespace {
24const int kVideoRotationRtpExtensionId = 4;
25}
26
pbos@webrtc.org994d0b72014-06-27 08:47:5227CallTest::CallTest()
pbos@webrtc.org2bb1bda2014-07-07 13:06:4828 : clock_(Clock::GetRealTimeClock()),
stefanff483612015-12-21 11:14:0029 video_send_config_(nullptr),
Stefan Holmer9fea80f2016-01-07 16:43:1830 video_send_stream_(nullptr),
31 audio_send_config_(nullptr),
32 audio_send_stream_(nullptr),
33 fake_encoder_(clock_),
Stefan Holmer04cb7632016-01-14 19:34:3034 num_video_streams_(1),
Stefan Holmer9fea80f2016-01-07 16:43:1835 num_audio_streams_(0),
36 fake_send_audio_device_(nullptr),
37 fake_recv_audio_device_(nullptr) {}
pbos@webrtc.org32452b22014-10-22 12:15:2438
pbos@webrtc.org994d0b72014-06-27 08:47:5239CallTest::~CallTest() {
40}
41
stefane74eef12016-01-08 14:47:1342void CallTest::RunBaseTest(BaseTest* test) {
Stefan Holmer9fea80f2016-01-07 16:43:1843 num_video_streams_ = test->GetNumVideoStreams();
44 num_audio_streams_ = test->GetNumAudioStreams();
45 RTC_DCHECK(num_video_streams_ > 0 || num_audio_streams_ > 0);
46 Call::Config send_config(test->GetSenderCallConfig());
47 if (num_audio_streams_ > 0) {
48 CreateVoiceEngines();
49 AudioState::Config audio_state_config;
50 audio_state_config.voice_engine = voe_send_.voice_engine;
51 send_config.audio_state = AudioState::Create(audio_state_config);
52 }
53 CreateSenderCall(send_config);
54 if (test->ShouldCreateReceivers()) {
55 Call::Config recv_config(test->GetReceiverCallConfig());
56 if (num_audio_streams_ > 0) {
57 AudioState::Config audio_state_config;
58 audio_state_config.voice_engine = voe_recv_.voice_engine;
59 recv_config.audio_state = AudioState::Create(audio_state_config);
60 }
61 CreateReceiverCall(recv_config);
62 }
Stefan Holmer04cb7632016-01-14 19:34:3063 test->OnCallsCreated(sender_call_.get(), receiver_call_.get());
stefane74eef12016-01-08 14:47:1364 send_transport_.reset(test->CreateSendTransport(sender_call_.get()));
65 receive_transport_.reset(test->CreateReceiveTransport());
pbos@webrtc.org994d0b72014-06-27 08:47:5266
67 if (test->ShouldCreateReceivers()) {
stefanf116bd02015-10-27 15:29:4268 send_transport_->SetReceiver(receiver_call_->Receiver());
69 receive_transport_->SetReceiver(sender_call_->Receiver());
pbos@webrtc.org994d0b72014-06-27 08:47:5270 } else {
71 // Sender-only call delivers to itself.
stefanf116bd02015-10-27 15:29:4272 send_transport_->SetReceiver(sender_call_->Receiver());
73 receive_transport_->SetReceiver(nullptr);
pbos@webrtc.org994d0b72014-06-27 08:47:5274 }
75
Stefan Holmer9fea80f2016-01-07 16:43:1876 CreateSendConfig(num_video_streams_, num_audio_streams_,
77 send_transport_.get());
pbos@webrtc.org994d0b72014-06-27 08:47:5278 if (test->ShouldCreateReceivers()) {
stefanf116bd02015-10-27 15:29:4279 CreateMatchingReceiveConfigs(receive_transport_.get());
pbos@webrtc.org994d0b72014-06-27 08:47:5280 }
Stefan Holmer9fea80f2016-01-07 16:43:1881 if (num_audio_streams_ > 0)
82 SetupVoiceEngineTransports(send_transport_.get(), receive_transport_.get());
83
84 if (num_video_streams_ > 0) {
85 test->ModifyVideoConfigs(&video_send_config_, &video_receive_configs_,
86 &video_encoder_config_);
87 }
88 if (num_audio_streams_ > 0)
89 test->ModifyAudioConfigs(&audio_send_config_, &audio_receive_configs_);
90
91 if (num_video_streams_ > 0) {
92 CreateVideoStreams();
93 test->OnVideoStreamsCreated(video_send_stream_, video_receive_streams_);
94 }
95 if (num_audio_streams_ > 0) {
96 CreateAudioStreams();
97 test->OnAudioStreamsCreated(audio_send_stream_, audio_receive_streams_);
98 }
pbos@webrtc.org994d0b72014-06-27 08:47:5299
Stefan Holmer04cb7632016-01-14 19:34:30100 if (num_video_streams_ > 0) {
101 CreateFrameGeneratorCapturer();
102 test->OnFrameGeneratorCapturerCreated(frame_generator_capturer_.get());
103 }
pbos@webrtc.org994d0b72014-06-27 08:47:52104
105 Start();
106 test->PerformTest();
stefanf116bd02015-10-27 15:29:42107 send_transport_->StopSending();
108 receive_transport_->StopSending();
pbos@webrtc.org994d0b72014-06-27 08:47:52109 Stop();
110
111 DestroyStreams();
Stefan Holmer9fea80f2016-01-07 16:43:18112 DestroyCalls();
113 if (num_audio_streams_ > 0)
114 DestroyVoiceEngines();
pbos@webrtc.org994d0b72014-06-27 08:47:52115}
116
117void CallTest::Start() {
Stefan Holmer9fea80f2016-01-07 16:43:18118 if (video_send_stream_)
119 video_send_stream_->Start();
120 for (VideoReceiveStream* video_recv_stream : video_receive_streams_)
121 video_recv_stream->Start();
122 if (audio_send_stream_) {
123 fake_send_audio_device_->Start();
124 audio_send_stream_->Start();
125 EXPECT_EQ(0, voe_send_.base->StartSend(voe_send_.channel_id));
126 }
127 for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_)
128 audio_recv_stream->Start();
129 if (!audio_receive_streams_.empty()) {
130 fake_recv_audio_device_->Start();
131 EXPECT_EQ(0, voe_recv_.base->StartPlayout(voe_recv_.channel_id));
132 EXPECT_EQ(0, voe_recv_.base->StartReceive(voe_recv_.channel_id));
133 }
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09134 if (frame_generator_capturer_.get() != NULL)
135 frame_generator_capturer_->Start();
pbos@webrtc.org994d0b72014-06-27 08:47:52136}
137
138void CallTest::Stop() {
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09139 if (frame_generator_capturer_.get() != NULL)
140 frame_generator_capturer_->Stop();
Stefan Holmer9fea80f2016-01-07 16:43:18141 if (!audio_receive_streams_.empty()) {
142 fake_recv_audio_device_->Stop();
143 EXPECT_EQ(0, voe_recv_.base->StopReceive(voe_recv_.channel_id));
144 EXPECT_EQ(0, voe_recv_.base->StopPlayout(voe_recv_.channel_id));
145 }
146 for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_)
147 audio_recv_stream->Stop();
148 if (audio_send_stream_) {
149 fake_send_audio_device_->Stop();
150 EXPECT_EQ(0, voe_send_.base->StopSend(voe_send_.channel_id));
151 audio_send_stream_->Stop();
152 }
153 for (VideoReceiveStream* video_recv_stream : video_receive_streams_)
154 video_recv_stream->Stop();
155 if (video_send_stream_)
156 video_send_stream_->Stop();
pbos@webrtc.org994d0b72014-06-27 08:47:52157}
158
159void CallTest::CreateCalls(const Call::Config& sender_config,
160 const Call::Config& receiver_config) {
161 CreateSenderCall(sender_config);
162 CreateReceiverCall(receiver_config);
163}
164
165void CallTest::CreateSenderCall(const Call::Config& config) {
166 sender_call_.reset(Call::Create(config));
167}
168
169void CallTest::CreateReceiverCall(const Call::Config& config) {
170 receiver_call_.reset(Call::Create(config));
171}
172
Fredrik Solenberg4f4ec0a2015-10-22 08:49:27173void CallTest::DestroyCalls() {
Stefan Holmer9fea80f2016-01-07 16:43:18174 sender_call_.reset();
175 receiver_call_.reset();
Fredrik Solenberg4f4ec0a2015-10-22 08:49:27176}
177
Stefan Holmer9fea80f2016-01-07 16:43:18178void CallTest::CreateSendConfig(size_t num_video_streams,
179 size_t num_audio_streams,
pbos2d566682015-09-28 16:59:31180 Transport* send_transport) {
Stefan Holmer9fea80f2016-01-07 16:43:18181 RTC_DCHECK(num_video_streams <= kNumSsrcs);
182 RTC_DCHECK_LE(num_audio_streams, 1u);
183 RTC_DCHECK(num_audio_streams == 0 || voe_send_.channel_id >= 0);
Stefan Holmer04cb7632016-01-14 19:34:30184 if (num_video_streams > 0) {
185 video_send_config_ = VideoSendStream::Config(send_transport);
186 video_send_config_.encoder_settings.encoder = &fake_encoder_;
187 video_send_config_.encoder_settings.payload_name = "FAKE";
188 video_send_config_.encoder_settings.payload_type =
189 kFakeVideoSendPayloadType;
190 video_send_config_.rtp.extensions.push_back(
191 RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));
192 video_encoder_config_.streams = test::CreateVideoStreams(num_video_streams);
193 for (size_t i = 0; i < num_video_streams; ++i)
194 video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[i]);
195 video_send_config_.rtp.extensions.push_back(RtpExtension(
196 RtpExtension::kVideoRotation, kVideoRotationRtpExtensionId));
197 }
Stefan Holmer9fea80f2016-01-07 16:43:18198
199 if (num_audio_streams > 0) {
200 audio_send_config_ = AudioSendStream::Config(send_transport);
201 audio_send_config_.voe_channel_id = voe_send_.channel_id;
202 audio_send_config_.rtp.ssrc = kAudioSendSsrc;
203 }
pbos@webrtc.org994d0b72014-06-27 08:47:52204}
205
Stefan Holmer9fea80f2016-01-07 16:43:18206void CallTest::CreateMatchingReceiveConfigs(Transport* rtcp_send_transport) {
Stefan Holmer9fea80f2016-01-07 16:43:18207 RTC_DCHECK(video_receive_configs_.empty());
208 RTC_DCHECK(allocated_decoders_.empty());
Stefan Holmer04cb7632016-01-14 19:34:30209 if (num_video_streams_ > 0) {
210 RTC_DCHECK(!video_send_config_.rtp.ssrcs.empty());
211 VideoReceiveStream::Config video_config(rtcp_send_transport);
212 video_config.rtp.remb = true;
213 video_config.rtp.local_ssrc = kReceiverLocalVideoSsrc;
214 for (const RtpExtension& extension : video_send_config_.rtp.extensions)
215 video_config.rtp.extensions.push_back(extension);
216 for (size_t i = 0; i < video_send_config_.rtp.ssrcs.size(); ++i) {
217 VideoReceiveStream::Decoder decoder =
218 test::CreateMatchingDecoder(video_send_config_.encoder_settings);
219 allocated_decoders_.push_back(decoder.decoder);
220 video_config.decoders.clear();
221 video_config.decoders.push_back(decoder);
222 video_config.rtp.remote_ssrc = video_send_config_.rtp.ssrcs[i];
223 video_receive_configs_.push_back(video_config);
224 }
Stefan Holmer9fea80f2016-01-07 16:43:18225 }
226
227 RTC_DCHECK(num_audio_streams_ <= 1);
228 if (num_audio_streams_ == 1) {
Stefan Holmer04cb7632016-01-14 19:34:30229 RTC_DCHECK(voe_send_.channel_id >= 0);
Stefan Holmer9fea80f2016-01-07 16:43:18230 AudioReceiveStream::Config audio_config;
231 audio_config.rtp.local_ssrc = kReceiverLocalAudioSsrc;
232 audio_config.rtcp_send_transport = rtcp_send_transport;
233 audio_config.voe_channel_id = voe_recv_.channel_id;
234 audio_config.rtp.remote_ssrc = audio_send_config_.rtp.ssrc;
235 audio_receive_configs_.push_back(audio_config);
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09236 }
pbos@webrtc.org994d0b72014-06-27 08:47:52237}
238
danilchap9c6a0c72016-02-10 18:54:47239void CallTest::CreateFrameGeneratorCapturerWithDrift(Clock* clock,
240 float speed) {
241 VideoStream stream = video_encoder_config_.streams.back();
242 frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create(
243 video_send_stream_->Input(), stream.width, stream.height,
244 stream.max_framerate * speed, clock));
245}
246
pbos@webrtc.org994d0b72014-06-27 08:47:52247void CallTest::CreateFrameGeneratorCapturer() {
stefanff483612015-12-21 11:14:00248 VideoStream stream = video_encoder_config_.streams.back();
249 frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create(
250 video_send_stream_->Input(), stream.width, stream.height,
251 stream.max_framerate, clock_));
pbos@webrtc.org994d0b72014-06-27 08:47:52252}
pbosf1828e82015-07-28 15:20:59253
Stefan Holmer9fea80f2016-01-07 16:43:18254void CallTest::CreateFakeAudioDevices() {
255 fake_send_audio_device_.reset(new FakeAudioDevice(
danilchap9c6a0c72016-02-10 18:54:47256 clock_, test::ResourcePath("voice_engine/audio_long16", "pcm"),
257 DriftingClock::kNoDrift));
Stefan Holmer9fea80f2016-01-07 16:43:18258 fake_recv_audio_device_.reset(new FakeAudioDevice(
danilchap9c6a0c72016-02-10 18:54:47259 clock_, test::ResourcePath("voice_engine/audio_long16", "pcm"),
260 DriftingClock::kNoDrift));
Stefan Holmer9fea80f2016-01-07 16:43:18261}
262
263void CallTest::CreateVideoStreams() {
264 RTC_DCHECK(video_send_stream_ == nullptr);
265 RTC_DCHECK(video_receive_streams_.empty());
266 RTC_DCHECK(audio_send_stream_ == nullptr);
267 RTC_DCHECK(audio_receive_streams_.empty());
pbos@webrtc.org994d0b72014-06-27 08:47:52268
stefanff483612015-12-21 11:14:00269 video_send_stream_ = sender_call_->CreateVideoSendStream(
270 video_send_config_, video_encoder_config_);
stefanff483612015-12-21 11:14:00271 for (size_t i = 0; i < video_receive_configs_.size(); ++i) {
272 video_receive_streams_.push_back(
273 receiver_call_->CreateVideoReceiveStream(video_receive_configs_[i]));
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09274 }
pbos@webrtc.org994d0b72014-06-27 08:47:52275}
276
Stefan Holmer9fea80f2016-01-07 16:43:18277void CallTest::CreateAudioStreams() {
278 audio_send_stream_ = sender_call_->CreateAudioSendStream(audio_send_config_);
279 for (size_t i = 0; i < audio_receive_configs_.size(); ++i) {
280 audio_receive_streams_.push_back(
281 receiver_call_->CreateAudioReceiveStream(audio_receive_configs_[i]));
282 }
283 CodecInst isac = {kAudioSendPayloadType, "ISAC", 16000, 480, 1, 32000};
284 EXPECT_EQ(0, voe_send_.codec->SetSendCodec(voe_send_.channel_id, isac));
285}
286
pbos@webrtc.org994d0b72014-06-27 08:47:52287void CallTest::DestroyStreams() {
Stefan Holmer9fea80f2016-01-07 16:43:18288 if (video_send_stream_)
stefanff483612015-12-21 11:14:00289 sender_call_->DestroyVideoSendStream(video_send_stream_);
Stefan Holmer9fea80f2016-01-07 16:43:18290 video_send_stream_ = nullptr;
291 for (VideoReceiveStream* video_recv_stream : video_receive_streams_)
292 receiver_call_->DestroyVideoReceiveStream(video_recv_stream);
293
294 if (audio_send_stream_)
295 sender_call_->DestroyAudioSendStream(audio_send_stream_);
296 audio_send_stream_ = nullptr;
297 for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_)
298 receiver_call_->DestroyAudioReceiveStream(audio_recv_stream);
stefanff483612015-12-21 11:14:00299 video_receive_streams_.clear();
Stefan Holmer9fea80f2016-01-07 16:43:18300
pbos@webrtc.org776e6f22014-10-29 15:28:39301 allocated_decoders_.clear();
pbos@webrtc.org994d0b72014-06-27 08:47:52302}
303
Stefan Holmer9fea80f2016-01-07 16:43:18304void CallTest::CreateVoiceEngines() {
305 CreateFakeAudioDevices();
306 voe_send_.voice_engine = VoiceEngine::Create();
307 voe_send_.base = VoEBase::GetInterface(voe_send_.voice_engine);
308 voe_send_.network = VoENetwork::GetInterface(voe_send_.voice_engine);
309 voe_send_.codec = VoECodec::GetInterface(voe_send_.voice_engine);
310 EXPECT_EQ(0, voe_send_.base->Init(fake_send_audio_device_.get(), nullptr));
311 Config voe_config;
312 voe_config.Set<VoicePacing>(new VoicePacing(true));
313 voe_send_.channel_id = voe_send_.base->CreateChannel(voe_config);
314 EXPECT_GE(voe_send_.channel_id, 0);
315
316 voe_recv_.voice_engine = VoiceEngine::Create();
317 voe_recv_.base = VoEBase::GetInterface(voe_recv_.voice_engine);
318 voe_recv_.network = VoENetwork::GetInterface(voe_recv_.voice_engine);
319 voe_recv_.codec = VoECodec::GetInterface(voe_recv_.voice_engine);
320 EXPECT_EQ(0, voe_recv_.base->Init(fake_recv_audio_device_.get(), nullptr));
321 voe_recv_.channel_id = voe_recv_.base->CreateChannel();
322 EXPECT_GE(voe_recv_.channel_id, 0);
323}
324
325void CallTest::SetupVoiceEngineTransports(PacketTransport* send_transport,
326 PacketTransport* recv_transport) {
327 voe_send_.transport_adapter.reset(
328 new internal::TransportAdapter(send_transport));
329 voe_send_.transport_adapter->Enable();
330 EXPECT_EQ(0, voe_send_.network->RegisterExternalTransport(
331 voe_send_.channel_id, *voe_send_.transport_adapter.get()));
332
333 voe_recv_.transport_adapter.reset(
334 new internal::TransportAdapter(recv_transport));
335 voe_recv_.transport_adapter->Enable();
336 EXPECT_EQ(0, voe_recv_.network->RegisterExternalTransport(
337 voe_recv_.channel_id, *voe_recv_.transport_adapter.get()));
338}
339
340void CallTest::DestroyVoiceEngines() {
341 voe_recv_.base->DeleteChannel(voe_recv_.channel_id);
342 voe_recv_.channel_id = -1;
343 voe_recv_.base->Release();
344 voe_recv_.base = nullptr;
345 voe_recv_.network->Release();
346 voe_recv_.network = nullptr;
347 voe_recv_.codec->Release();
348 voe_recv_.codec = nullptr;
349
350 voe_send_.base->DeleteChannel(voe_send_.channel_id);
351 voe_send_.channel_id = -1;
352 voe_send_.base->Release();
353 voe_send_.base = nullptr;
354 voe_send_.network->Release();
355 voe_send_.network = nullptr;
356 voe_send_.codec->Release();
357 voe_send_.codec = nullptr;
358
359 VoiceEngine::Delete(voe_send_.voice_engine);
360 voe_send_.voice_engine = nullptr;
361 VoiceEngine::Delete(voe_recv_.voice_engine);
362 voe_recv_.voice_engine = nullptr;
363}
364
Peter Boström5811a392015-12-10 12:02:50365const int CallTest::kDefaultTimeoutMs = 30 * 1000;
366const int CallTest::kLongTimeoutMs = 120 * 1000;
Stefan Holmer9fea80f2016-01-07 16:43:18367const uint8_t CallTest::kVideoSendPayloadType = 100;
368const uint8_t CallTest::kFakeVideoSendPayloadType = 125;
pbos@webrtc.org994d0b72014-06-27 08:47:52369const uint8_t CallTest::kSendRtxPayloadType = 98;
stefan@webrtc.org01581da2014-09-04 06:48:14370const uint8_t CallTest::kRedPayloadType = 118;
Shao Changbine62202f2015-04-21 12:24:50371const uint8_t CallTest::kRtxRedPayloadType = 99;
stefan@webrtc.org01581da2014-09-04 06:48:14372const uint8_t CallTest::kUlpfecPayloadType = 119;
Stefan Holmer9fea80f2016-01-07 16:43:18373const uint8_t CallTest::kAudioSendPayloadType = 103;
pbos@webrtc.org2bb1bda2014-07-07 13:06:48374const uint32_t CallTest::kSendRtxSsrcs[kNumSsrcs] = {0xBADCAFD, 0xBADCAFE,
375 0xBADCAFF};
Stefan Holmer9fea80f2016-01-07 16:43:18376const uint32_t CallTest::kVideoSendSsrcs[kNumSsrcs] = {0xC0FFED, 0xC0FFEE,
377 0xC0FFEF};
378const uint32_t CallTest::kAudioSendSsrc = 0xDEADBEEF;
379const uint32_t CallTest::kReceiverLocalVideoSsrc = 0x123456;
380const uint32_t CallTest::kReceiverLocalAudioSsrc = 0x1234567;
pbos@webrtc.org994d0b72014-06-27 08:47:52381const int CallTest::kNackRtpHistoryMs = 1000;
382
383BaseTest::BaseTest(unsigned int timeout_ms) : RtpRtcpObserver(timeout_ms) {
384}
385
pbos@webrtc.org994d0b72014-06-27 08:47:52386BaseTest::~BaseTest() {
387}
388
389Call::Config BaseTest::GetSenderCallConfig() {
solenberg4fbae2b2015-08-28 11:07:10390 return Call::Config();
pbos@webrtc.org994d0b72014-06-27 08:47:52391}
392
393Call::Config BaseTest::GetReceiverCallConfig() {
solenberg4fbae2b2015-08-28 11:07:10394 return Call::Config();
pbos@webrtc.org994d0b72014-06-27 08:47:52395}
396
397void BaseTest::OnCallsCreated(Call* sender_call, Call* receiver_call) {
398}
399
stefane74eef12016-01-08 14:47:13400test::PacketTransport* BaseTest::CreateSendTransport(Call* sender_call) {
401 return new PacketTransport(sender_call, this, test::PacketTransport::kSender,
402 FakeNetworkPipe::Config());
403}
404
405test::PacketTransport* BaseTest::CreateReceiveTransport() {
406 return new PacketTransport(nullptr, this, test::PacketTransport::kReceiver,
407 FakeNetworkPipe::Config());
408}
stefanf116bd02015-10-27 15:29:42409
Stefan Holmer9fea80f2016-01-07 16:43:18410size_t BaseTest::GetNumVideoStreams() const {
pbos@webrtc.org994d0b72014-06-27 08:47:52411 return 1;
412}
413
Stefan Holmer9fea80f2016-01-07 16:43:18414size_t BaseTest::GetNumAudioStreams() const {
415 return 0;
416}
417
stefanff483612015-12-21 11:14:00418void BaseTest::ModifyVideoConfigs(
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09419 VideoSendStream::Config* send_config,
420 std::vector<VideoReceiveStream::Config>* receive_configs,
stefanff483612015-12-21 11:14:00421 VideoEncoderConfig* encoder_config) {}
pbos@webrtc.org994d0b72014-06-27 08:47:52422
stefanff483612015-12-21 11:14:00423void BaseTest::OnVideoStreamsCreated(
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09424 VideoSendStream* send_stream,
stefanff483612015-12-21 11:14:00425 const std::vector<VideoReceiveStream*>& receive_streams) {}
pbos@webrtc.org994d0b72014-06-27 08:47:52426
Stefan Holmer9fea80f2016-01-07 16:43:18427void BaseTest::ModifyAudioConfigs(
428 AudioSendStream::Config* send_config,
429 std::vector<AudioReceiveStream::Config>* receive_configs) {}
430
431void BaseTest::OnAudioStreamsCreated(
432 AudioSendStream* send_stream,
433 const std::vector<AudioReceiveStream*>& receive_streams) {}
434
pbos@webrtc.org994d0b72014-06-27 08:47:52435void BaseTest::OnFrameGeneratorCapturerCreated(
436 FrameGeneratorCapturer* frame_generator_capturer) {
437}
438
439SendTest::SendTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
440}
441
pbos@webrtc.org994d0b72014-06-27 08:47:52442bool SendTest::ShouldCreateReceivers() const {
443 return false;
444}
445
446EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
447}
448
pbos@webrtc.org994d0b72014-06-27 08:47:52449bool EndToEndTest::ShouldCreateReceivers() const {
450 return true;
451}
452
453} // namespace test
454} // namespace webrtc