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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 */
Stefan Holmer9fea80f2016-01-07 17:43:18 +010010#include "webrtc/base/checks.h"
11#include "webrtc/common.h"
12#include "webrtc/config.h"
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000013#include "webrtc/test/call_test.h"
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000014#include "webrtc/test/encoder_settings.h"
Stefan Holmer9fea80f2016-01-07 17:43:18 +010015#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 08:47:52 +000019
20namespace webrtc {
21namespace test {
22
Guo-wei Shieh2c370782015-04-08 13:00:10 -070023namespace {
24const int kVideoRotationRtpExtensionId = 4;
25}
26
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000027CallTest::CallTest()
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +000028 : clock_(Clock::GetRealTimeClock()),
stefanff483612015-12-21 03:14:00 -080029 video_send_config_(nullptr),
Stefan Holmer9fea80f2016-01-07 17:43:18 +010030 video_send_stream_(nullptr),
31 audio_send_config_(nullptr),
32 audio_send_stream_(nullptr),
33 fake_encoder_(clock_),
34 num_video_streams_(0),
35 num_audio_streams_(0),
36 fake_send_audio_device_(nullptr),
37 fake_recv_audio_device_(nullptr) {}
pbos@webrtc.org32452b22014-10-22 12:15:24 +000038
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000039CallTest::~CallTest() {
40}
41
stefanf116bd02015-10-27 08:29:42 -070042void CallTest::RunBaseTest(BaseTest* test,
43 const FakeNetworkPipe::Config& config) {
Stefan Holmer9fea80f2016-01-07 17:43:18 +010044 num_video_streams_ = test->GetNumVideoStreams();
45 num_audio_streams_ = test->GetNumAudioStreams();
46 RTC_DCHECK(num_video_streams_ > 0 || num_audio_streams_ > 0);
47 Call::Config send_config(test->GetSenderCallConfig());
48 if (num_audio_streams_ > 0) {
49 CreateVoiceEngines();
50 AudioState::Config audio_state_config;
51 audio_state_config.voice_engine = voe_send_.voice_engine;
52 send_config.audio_state = AudioState::Create(audio_state_config);
53 }
54 CreateSenderCall(send_config);
55 if (test->ShouldCreateReceivers()) {
56 Call::Config recv_config(test->GetReceiverCallConfig());
57 if (num_audio_streams_ > 0) {
58 AudioState::Config audio_state_config;
59 audio_state_config.voice_engine = voe_recv_.voice_engine;
60 recv_config.audio_state = AudioState::Create(audio_state_config);
61 }
62 CreateReceiverCall(recv_config);
63 }
stefanf116bd02015-10-27 08:29:42 -070064 send_transport_.reset(new PacketTransport(
65 sender_call_.get(), test, test::PacketTransport::kSender, config));
66 receive_transport_.reset(new PacketTransport(
67 nullptr, test, test::PacketTransport::kReceiver, config));
68 test->OnTransportsCreated(send_transport_.get(), receive_transport_.get());
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000069 test->OnCallsCreated(sender_call_.get(), receiver_call_.get());
70
71 if (test->ShouldCreateReceivers()) {
stefanf116bd02015-10-27 08:29:42 -070072 send_transport_->SetReceiver(receiver_call_->Receiver());
73 receive_transport_->SetReceiver(sender_call_->Receiver());
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000074 } else {
75 // Sender-only call delivers to itself.
stefanf116bd02015-10-27 08:29:42 -070076 send_transport_->SetReceiver(sender_call_->Receiver());
77 receive_transport_->SetReceiver(nullptr);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000078 }
79
Stefan Holmer9fea80f2016-01-07 17:43:18 +010080 CreateSendConfig(num_video_streams_, num_audio_streams_,
81 send_transport_.get());
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000082 if (test->ShouldCreateReceivers()) {
stefanf116bd02015-10-27 08:29:42 -070083 CreateMatchingReceiveConfigs(receive_transport_.get());
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000084 }
Stefan Holmer9fea80f2016-01-07 17:43:18 +010085 if (num_audio_streams_ > 0)
86 SetupVoiceEngineTransports(send_transport_.get(), receive_transport_.get());
87
88 if (num_video_streams_ > 0) {
89 test->ModifyVideoConfigs(&video_send_config_, &video_receive_configs_,
90 &video_encoder_config_);
91 }
92 if (num_audio_streams_ > 0)
93 test->ModifyAudioConfigs(&audio_send_config_, &audio_receive_configs_);
94
95 if (num_video_streams_ > 0) {
96 CreateVideoStreams();
97 test->OnVideoStreamsCreated(video_send_stream_, video_receive_streams_);
98 }
99 if (num_audio_streams_ > 0) {
100 CreateAudioStreams();
101 test->OnAudioStreamsCreated(audio_send_stream_, audio_receive_streams_);
102 }
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000103
104 CreateFrameGeneratorCapturer();
105 test->OnFrameGeneratorCapturerCreated(frame_generator_capturer_.get());
106
107 Start();
108 test->PerformTest();
stefanf116bd02015-10-27 08:29:42 -0700109 send_transport_->StopSending();
110 receive_transport_->StopSending();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000111 Stop();
112
113 DestroyStreams();
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100114 DestroyCalls();
115 if (num_audio_streams_ > 0)
116 DestroyVoiceEngines();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000117}
118
119void CallTest::Start() {
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100120 if (video_send_stream_)
121 video_send_stream_->Start();
122 for (VideoReceiveStream* video_recv_stream : video_receive_streams_)
123 video_recv_stream->Start();
124 if (audio_send_stream_) {
125 fake_send_audio_device_->Start();
126 audio_send_stream_->Start();
127 EXPECT_EQ(0, voe_send_.base->StartSend(voe_send_.channel_id));
128 }
129 for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_)
130 audio_recv_stream->Start();
131 if (!audio_receive_streams_.empty()) {
132 fake_recv_audio_device_->Start();
133 EXPECT_EQ(0, voe_recv_.base->StartPlayout(voe_recv_.channel_id));
134 EXPECT_EQ(0, voe_recv_.base->StartReceive(voe_recv_.channel_id));
135 }
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000136 if (frame_generator_capturer_.get() != NULL)
137 frame_generator_capturer_->Start();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000138}
139
140void CallTest::Stop() {
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000141 if (frame_generator_capturer_.get() != NULL)
142 frame_generator_capturer_->Stop();
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100143 if (!audio_receive_streams_.empty()) {
144 fake_recv_audio_device_->Stop();
145 EXPECT_EQ(0, voe_recv_.base->StopReceive(voe_recv_.channel_id));
146 EXPECT_EQ(0, voe_recv_.base->StopPlayout(voe_recv_.channel_id));
147 }
148 for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_)
149 audio_recv_stream->Stop();
150 if (audio_send_stream_) {
151 fake_send_audio_device_->Stop();
152 EXPECT_EQ(0, voe_send_.base->StopSend(voe_send_.channel_id));
153 audio_send_stream_->Stop();
154 }
155 for (VideoReceiveStream* video_recv_stream : video_receive_streams_)
156 video_recv_stream->Stop();
157 if (video_send_stream_)
158 video_send_stream_->Stop();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000159}
160
161void CallTest::CreateCalls(const Call::Config& sender_config,
162 const Call::Config& receiver_config) {
163 CreateSenderCall(sender_config);
164 CreateReceiverCall(receiver_config);
165}
166
167void CallTest::CreateSenderCall(const Call::Config& config) {
168 sender_call_.reset(Call::Create(config));
169}
170
171void CallTest::CreateReceiverCall(const Call::Config& config) {
172 receiver_call_.reset(Call::Create(config));
173}
174
Fredrik Solenberg4f4ec0a2015-10-22 10:49:27 +0200175void CallTest::DestroyCalls() {
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100176 sender_call_.reset();
177 receiver_call_.reset();
Fredrik Solenberg4f4ec0a2015-10-22 10:49:27 +0200178}
179
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100180void CallTest::CreateSendConfig(size_t num_video_streams,
181 size_t num_audio_streams,
pbos2d566682015-09-28 09:59:31 -0700182 Transport* send_transport) {
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100183 RTC_DCHECK(num_video_streams <= kNumSsrcs);
184 RTC_DCHECK_LE(num_audio_streams, 1u);
185 RTC_DCHECK(num_audio_streams == 0 || voe_send_.channel_id >= 0);
stefanff483612015-12-21 03:14:00 -0800186 video_send_config_ = VideoSendStream::Config(send_transport);
187 video_send_config_.encoder_settings.encoder = &fake_encoder_;
188 video_send_config_.encoder_settings.payload_name = "FAKE";
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100189 video_send_config_.encoder_settings.payload_type = kFakeVideoSendPayloadType;
stefanff483612015-12-21 03:14:00 -0800190 video_send_config_.rtp.extensions.push_back(
Erik SprĂ¥ng95261872015-04-10 11:58:49 +0200191 RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100192 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]);
stefanff483612015-12-21 03:14:00 -0800195 video_send_config_.rtp.extensions.push_back(
Guo-wei Shieh2c370782015-04-08 13:00:10 -0700196 RtpExtension(RtpExtension::kVideoRotation, kVideoRotationRtpExtensionId));
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100197
198 if (num_audio_streams > 0) {
199 audio_send_config_ = AudioSendStream::Config(send_transport);
200 audio_send_config_.voe_channel_id = voe_send_.channel_id;
201 audio_send_config_.rtp.ssrc = kAudioSendSsrc;
202 }
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000203}
204
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100205void CallTest::CreateMatchingReceiveConfigs(Transport* rtcp_send_transport) {
206 RTC_DCHECK(!video_send_config_.rtp.ssrcs.empty());
207 RTC_DCHECK(video_receive_configs_.empty());
208 RTC_DCHECK(allocated_decoders_.empty());
209 RTC_DCHECK(num_audio_streams_ == 0 || voe_send_.channel_id >= 0);
210 VideoReceiveStream::Config video_config(rtcp_send_transport);
211 video_config.rtp.remb = true;
212 video_config.rtp.local_ssrc = kReceiverLocalVideoSsrc;
stefanff483612015-12-21 03:14:00 -0800213 for (const RtpExtension& extension : video_send_config_.rtp.extensions)
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100214 video_config.rtp.extensions.push_back(extension);
stefanff483612015-12-21 03:14:00 -0800215 for (size_t i = 0; i < video_send_config_.rtp.ssrcs.size(); ++i) {
pbos@webrtc.org776e6f22014-10-29 15:28:39 +0000216 VideoReceiveStream::Decoder decoder =
stefanff483612015-12-21 03:14:00 -0800217 test::CreateMatchingDecoder(video_send_config_.encoder_settings);
pbos@webrtc.org776e6f22014-10-29 15:28:39 +0000218 allocated_decoders_.push_back(decoder.decoder);
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100219 video_config.decoders.clear();
220 video_config.decoders.push_back(decoder);
221 video_config.rtp.remote_ssrc = video_send_config_.rtp.ssrcs[i];
222 video_receive_configs_.push_back(video_config);
223 }
224
225 RTC_DCHECK(num_audio_streams_ <= 1);
226 if (num_audio_streams_ == 1) {
227 AudioReceiveStream::Config audio_config;
228 audio_config.rtp.local_ssrc = kReceiverLocalAudioSsrc;
229 audio_config.rtcp_send_transport = rtcp_send_transport;
230 audio_config.voe_channel_id = voe_recv_.channel_id;
231 audio_config.rtp.remote_ssrc = audio_send_config_.rtp.ssrc;
232 audio_receive_configs_.push_back(audio_config);
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000233 }
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000234}
235
236void CallTest::CreateFrameGeneratorCapturer() {
stefanff483612015-12-21 03:14:00 -0800237 VideoStream stream = video_encoder_config_.streams.back();
238 frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create(
239 video_send_stream_->Input(), stream.width, stream.height,
240 stream.max_framerate, clock_));
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000241}
pbosf1828e82015-07-28 08:20:59 -0700242
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100243void CallTest::CreateFakeAudioDevices() {
244 fake_send_audio_device_.reset(new FakeAudioDevice(
245 clock_, test::ResourcePath("voice_engine/audio_long16", "pcm")));
246 fake_recv_audio_device_.reset(new FakeAudioDevice(
247 clock_, test::ResourcePath("voice_engine/audio_long16", "pcm")));
248}
249
250void CallTest::CreateVideoStreams() {
251 RTC_DCHECK(video_send_stream_ == nullptr);
252 RTC_DCHECK(video_receive_streams_.empty());
253 RTC_DCHECK(audio_send_stream_ == nullptr);
254 RTC_DCHECK(audio_receive_streams_.empty());
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000255
stefanff483612015-12-21 03:14:00 -0800256 video_send_stream_ = sender_call_->CreateVideoSendStream(
257 video_send_config_, video_encoder_config_);
stefanff483612015-12-21 03:14:00 -0800258 for (size_t i = 0; i < video_receive_configs_.size(); ++i) {
259 video_receive_streams_.push_back(
260 receiver_call_->CreateVideoReceiveStream(video_receive_configs_[i]));
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000261 }
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000262}
263
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100264void CallTest::CreateAudioStreams() {
265 audio_send_stream_ = sender_call_->CreateAudioSendStream(audio_send_config_);
266 for (size_t i = 0; i < audio_receive_configs_.size(); ++i) {
267 audio_receive_streams_.push_back(
268 receiver_call_->CreateAudioReceiveStream(audio_receive_configs_[i]));
269 }
270 CodecInst isac = {kAudioSendPayloadType, "ISAC", 16000, 480, 1, 32000};
271 EXPECT_EQ(0, voe_send_.codec->SetSendCodec(voe_send_.channel_id, isac));
272}
273
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000274void CallTest::DestroyStreams() {
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100275 if (video_send_stream_)
stefanff483612015-12-21 03:14:00 -0800276 sender_call_->DestroyVideoSendStream(video_send_stream_);
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100277 video_send_stream_ = nullptr;
278 for (VideoReceiveStream* video_recv_stream : video_receive_streams_)
279 receiver_call_->DestroyVideoReceiveStream(video_recv_stream);
280
281 if (audio_send_stream_)
282 sender_call_->DestroyAudioSendStream(audio_send_stream_);
283 audio_send_stream_ = nullptr;
284 for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_)
285 receiver_call_->DestroyAudioReceiveStream(audio_recv_stream);
stefanff483612015-12-21 03:14:00 -0800286 video_receive_streams_.clear();
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100287
pbos@webrtc.org776e6f22014-10-29 15:28:39 +0000288 allocated_decoders_.clear();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000289}
290
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100291void CallTest::CreateVoiceEngines() {
292 CreateFakeAudioDevices();
293 voe_send_.voice_engine = VoiceEngine::Create();
294 voe_send_.base = VoEBase::GetInterface(voe_send_.voice_engine);
295 voe_send_.network = VoENetwork::GetInterface(voe_send_.voice_engine);
296 voe_send_.codec = VoECodec::GetInterface(voe_send_.voice_engine);
297 EXPECT_EQ(0, voe_send_.base->Init(fake_send_audio_device_.get(), nullptr));
298 Config voe_config;
299 voe_config.Set<VoicePacing>(new VoicePacing(true));
300 voe_send_.channel_id = voe_send_.base->CreateChannel(voe_config);
301 EXPECT_GE(voe_send_.channel_id, 0);
302
303 voe_recv_.voice_engine = VoiceEngine::Create();
304 voe_recv_.base = VoEBase::GetInterface(voe_recv_.voice_engine);
305 voe_recv_.network = VoENetwork::GetInterface(voe_recv_.voice_engine);
306 voe_recv_.codec = VoECodec::GetInterface(voe_recv_.voice_engine);
307 EXPECT_EQ(0, voe_recv_.base->Init(fake_recv_audio_device_.get(), nullptr));
308 voe_recv_.channel_id = voe_recv_.base->CreateChannel();
309 EXPECT_GE(voe_recv_.channel_id, 0);
310}
311
312void CallTest::SetupVoiceEngineTransports(PacketTransport* send_transport,
313 PacketTransport* recv_transport) {
314 voe_send_.transport_adapter.reset(
315 new internal::TransportAdapter(send_transport));
316 voe_send_.transport_adapter->Enable();
317 EXPECT_EQ(0, voe_send_.network->RegisterExternalTransport(
318 voe_send_.channel_id, *voe_send_.transport_adapter.get()));
319
320 voe_recv_.transport_adapter.reset(
321 new internal::TransportAdapter(recv_transport));
322 voe_recv_.transport_adapter->Enable();
323 EXPECT_EQ(0, voe_recv_.network->RegisterExternalTransport(
324 voe_recv_.channel_id, *voe_recv_.transport_adapter.get()));
325}
326
327void CallTest::DestroyVoiceEngines() {
328 voe_recv_.base->DeleteChannel(voe_recv_.channel_id);
329 voe_recv_.channel_id = -1;
330 voe_recv_.base->Release();
331 voe_recv_.base = nullptr;
332 voe_recv_.network->Release();
333 voe_recv_.network = nullptr;
334 voe_recv_.codec->Release();
335 voe_recv_.codec = nullptr;
336
337 voe_send_.base->DeleteChannel(voe_send_.channel_id);
338 voe_send_.channel_id = -1;
339 voe_send_.base->Release();
340 voe_send_.base = nullptr;
341 voe_send_.network->Release();
342 voe_send_.network = nullptr;
343 voe_send_.codec->Release();
344 voe_send_.codec = nullptr;
345
346 VoiceEngine::Delete(voe_send_.voice_engine);
347 voe_send_.voice_engine = nullptr;
348 VoiceEngine::Delete(voe_recv_.voice_engine);
349 voe_recv_.voice_engine = nullptr;
350}
351
Peter Boström5811a392015-12-10 13:02:50 +0100352const int CallTest::kDefaultTimeoutMs = 30 * 1000;
353const int CallTest::kLongTimeoutMs = 120 * 1000;
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100354const uint8_t CallTest::kVideoSendPayloadType = 100;
355const uint8_t CallTest::kFakeVideoSendPayloadType = 125;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000356const uint8_t CallTest::kSendRtxPayloadType = 98;
stefan@webrtc.org01581da2014-09-04 06:48:14 +0000357const uint8_t CallTest::kRedPayloadType = 118;
Shao Changbine62202f2015-04-21 20:24:50 +0800358const uint8_t CallTest::kRtxRedPayloadType = 99;
stefan@webrtc.org01581da2014-09-04 06:48:14 +0000359const uint8_t CallTest::kUlpfecPayloadType = 119;
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100360const uint8_t CallTest::kAudioSendPayloadType = 103;
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +0000361const uint32_t CallTest::kSendRtxSsrcs[kNumSsrcs] = {0xBADCAFD, 0xBADCAFE,
362 0xBADCAFF};
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100363const uint32_t CallTest::kVideoSendSsrcs[kNumSsrcs] = {0xC0FFED, 0xC0FFEE,
364 0xC0FFEF};
365const uint32_t CallTest::kAudioSendSsrc = 0xDEADBEEF;
366const uint32_t CallTest::kReceiverLocalVideoSsrc = 0x123456;
367const uint32_t CallTest::kReceiverLocalAudioSsrc = 0x1234567;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000368const int CallTest::kNackRtpHistoryMs = 1000;
369
370BaseTest::BaseTest(unsigned int timeout_ms) : RtpRtcpObserver(timeout_ms) {
371}
372
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000373BaseTest::~BaseTest() {
374}
375
376Call::Config BaseTest::GetSenderCallConfig() {
solenberg4fbae2b2015-08-28 04:07:10 -0700377 return Call::Config();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000378}
379
380Call::Config BaseTest::GetReceiverCallConfig() {
solenberg4fbae2b2015-08-28 04:07:10 -0700381 return Call::Config();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000382}
383
384void BaseTest::OnCallsCreated(Call* sender_call, Call* receiver_call) {
385}
386
stefanf116bd02015-10-27 08:29:42 -0700387void BaseTest::OnTransportsCreated(PacketTransport* send_transport,
388 PacketTransport* receive_transport) {}
389
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100390size_t BaseTest::GetNumVideoStreams() const {
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000391 return 1;
392}
393
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100394size_t BaseTest::GetNumAudioStreams() const {
395 return 0;
396}
397
stefanff483612015-12-21 03:14:00 -0800398void BaseTest::ModifyVideoConfigs(
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000399 VideoSendStream::Config* send_config,
400 std::vector<VideoReceiveStream::Config>* receive_configs,
stefanff483612015-12-21 03:14:00 -0800401 VideoEncoderConfig* encoder_config) {}
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000402
stefanff483612015-12-21 03:14:00 -0800403void BaseTest::OnVideoStreamsCreated(
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000404 VideoSendStream* send_stream,
stefanff483612015-12-21 03:14:00 -0800405 const std::vector<VideoReceiveStream*>& receive_streams) {}
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000406
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100407void BaseTest::ModifyAudioConfigs(
408 AudioSendStream::Config* send_config,
409 std::vector<AudioReceiveStream::Config>* receive_configs) {}
410
411void BaseTest::OnAudioStreamsCreated(
412 AudioSendStream* send_stream,
413 const std::vector<AudioReceiveStream*>& receive_streams) {}
414
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000415void BaseTest::OnFrameGeneratorCapturerCreated(
416 FrameGeneratorCapturer* frame_generator_capturer) {
417}
418
419SendTest::SendTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
420}
421
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000422bool SendTest::ShouldCreateReceivers() const {
423 return false;
424}
425
426EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
427}
428
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000429bool EndToEndTest::ShouldCreateReceivers() const {
430 return true;
431}
432
433} // namespace test
434} // namespace webrtc