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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
stefane74eef12016-01-08 06:47:13 -080042void CallTest::RunBaseTest(BaseTest* test) {
Stefan Holmer9fea80f2016-01-07 17:43:18 +010043 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 }
stefane74eef12016-01-08 06:47:13 -080063 send_transport_.reset(test->CreateSendTransport(sender_call_.get()));
64 receive_transport_.reset(test->CreateReceiveTransport());
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000065 test->OnCallsCreated(sender_call_.get(), receiver_call_.get());
66
67 if (test->ShouldCreateReceivers()) {
stefanf116bd02015-10-27 08:29:42 -070068 send_transport_->SetReceiver(receiver_call_->Receiver());
69 receive_transport_->SetReceiver(sender_call_->Receiver());
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000070 } else {
71 // Sender-only call delivers to itself.
stefanf116bd02015-10-27 08:29:42 -070072 send_transport_->SetReceiver(sender_call_->Receiver());
73 receive_transport_->SetReceiver(nullptr);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000074 }
75
Stefan Holmer9fea80f2016-01-07 17:43:18 +010076 CreateSendConfig(num_video_streams_, num_audio_streams_,
77 send_transport_.get());
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000078 if (test->ShouldCreateReceivers()) {
stefanf116bd02015-10-27 08:29:42 -070079 CreateMatchingReceiveConfigs(receive_transport_.get());
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000080 }
Stefan Holmer9fea80f2016-01-07 17:43:18 +010081 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 08:47:52 +000099
100 CreateFrameGeneratorCapturer();
101 test->OnFrameGeneratorCapturerCreated(frame_generator_capturer_.get());
102
103 Start();
104 test->PerformTest();
stefanf116bd02015-10-27 08:29:42 -0700105 send_transport_->StopSending();
106 receive_transport_->StopSending();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000107 Stop();
108
109 DestroyStreams();
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100110 DestroyCalls();
111 if (num_audio_streams_ > 0)
112 DestroyVoiceEngines();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000113}
114
115void CallTest::Start() {
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100116 if (video_send_stream_)
117 video_send_stream_->Start();
118 for (VideoReceiveStream* video_recv_stream : video_receive_streams_)
119 video_recv_stream->Start();
120 if (audio_send_stream_) {
121 fake_send_audio_device_->Start();
122 audio_send_stream_->Start();
123 EXPECT_EQ(0, voe_send_.base->StartSend(voe_send_.channel_id));
124 }
125 for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_)
126 audio_recv_stream->Start();
127 if (!audio_receive_streams_.empty()) {
128 fake_recv_audio_device_->Start();
129 EXPECT_EQ(0, voe_recv_.base->StartPlayout(voe_recv_.channel_id));
130 EXPECT_EQ(0, voe_recv_.base->StartReceive(voe_recv_.channel_id));
131 }
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000132 if (frame_generator_capturer_.get() != NULL)
133 frame_generator_capturer_->Start();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000134}
135
136void CallTest::Stop() {
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000137 if (frame_generator_capturer_.get() != NULL)
138 frame_generator_capturer_->Stop();
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100139 if (!audio_receive_streams_.empty()) {
140 fake_recv_audio_device_->Stop();
141 EXPECT_EQ(0, voe_recv_.base->StopReceive(voe_recv_.channel_id));
142 EXPECT_EQ(0, voe_recv_.base->StopPlayout(voe_recv_.channel_id));
143 }
144 for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_)
145 audio_recv_stream->Stop();
146 if (audio_send_stream_) {
147 fake_send_audio_device_->Stop();
148 EXPECT_EQ(0, voe_send_.base->StopSend(voe_send_.channel_id));
149 audio_send_stream_->Stop();
150 }
151 for (VideoReceiveStream* video_recv_stream : video_receive_streams_)
152 video_recv_stream->Stop();
153 if (video_send_stream_)
154 video_send_stream_->Stop();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000155}
156
157void CallTest::CreateCalls(const Call::Config& sender_config,
158 const Call::Config& receiver_config) {
159 CreateSenderCall(sender_config);
160 CreateReceiverCall(receiver_config);
161}
162
163void CallTest::CreateSenderCall(const Call::Config& config) {
164 sender_call_.reset(Call::Create(config));
165}
166
167void CallTest::CreateReceiverCall(const Call::Config& config) {
168 receiver_call_.reset(Call::Create(config));
169}
170
Fredrik Solenberg4f4ec0a2015-10-22 10:49:27 +0200171void CallTest::DestroyCalls() {
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100172 sender_call_.reset();
173 receiver_call_.reset();
Fredrik Solenberg4f4ec0a2015-10-22 10:49:27 +0200174}
175
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100176void CallTest::CreateSendConfig(size_t num_video_streams,
177 size_t num_audio_streams,
pbos2d566682015-09-28 09:59:31 -0700178 Transport* send_transport) {
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100179 RTC_DCHECK(num_video_streams <= kNumSsrcs);
180 RTC_DCHECK_LE(num_audio_streams, 1u);
181 RTC_DCHECK(num_audio_streams == 0 || voe_send_.channel_id >= 0);
stefanff483612015-12-21 03:14:00 -0800182 video_send_config_ = VideoSendStream::Config(send_transport);
183 video_send_config_.encoder_settings.encoder = &fake_encoder_;
184 video_send_config_.encoder_settings.payload_name = "FAKE";
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100185 video_send_config_.encoder_settings.payload_type = kFakeVideoSendPayloadType;
stefanff483612015-12-21 03:14:00 -0800186 video_send_config_.rtp.extensions.push_back(
Erik SprĂ¥ng95261872015-04-10 11:58:49 +0200187 RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100188 video_encoder_config_.streams = test::CreateVideoStreams(num_video_streams);
189 for (size_t i = 0; i < num_video_streams; ++i)
190 video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[i]);
stefanff483612015-12-21 03:14:00 -0800191 video_send_config_.rtp.extensions.push_back(
Guo-wei Shieh2c370782015-04-08 13:00:10 -0700192 RtpExtension(RtpExtension::kVideoRotation, kVideoRotationRtpExtensionId));
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100193
194 if (num_audio_streams > 0) {
195 audio_send_config_ = AudioSendStream::Config(send_transport);
196 audio_send_config_.voe_channel_id = voe_send_.channel_id;
197 audio_send_config_.rtp.ssrc = kAudioSendSsrc;
198 }
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000199}
200
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100201void CallTest::CreateMatchingReceiveConfigs(Transport* rtcp_send_transport) {
202 RTC_DCHECK(!video_send_config_.rtp.ssrcs.empty());
203 RTC_DCHECK(video_receive_configs_.empty());
204 RTC_DCHECK(allocated_decoders_.empty());
205 RTC_DCHECK(num_audio_streams_ == 0 || voe_send_.channel_id >= 0);
206 VideoReceiveStream::Config video_config(rtcp_send_transport);
207 video_config.rtp.remb = true;
208 video_config.rtp.local_ssrc = kReceiverLocalVideoSsrc;
stefanff483612015-12-21 03:14:00 -0800209 for (const RtpExtension& extension : video_send_config_.rtp.extensions)
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100210 video_config.rtp.extensions.push_back(extension);
stefanff483612015-12-21 03:14:00 -0800211 for (size_t i = 0; i < video_send_config_.rtp.ssrcs.size(); ++i) {
pbos@webrtc.org776e6f22014-10-29 15:28:39 +0000212 VideoReceiveStream::Decoder decoder =
stefanff483612015-12-21 03:14:00 -0800213 test::CreateMatchingDecoder(video_send_config_.encoder_settings);
pbos@webrtc.org776e6f22014-10-29 15:28:39 +0000214 allocated_decoders_.push_back(decoder.decoder);
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100215 video_config.decoders.clear();
216 video_config.decoders.push_back(decoder);
217 video_config.rtp.remote_ssrc = video_send_config_.rtp.ssrcs[i];
218 video_receive_configs_.push_back(video_config);
219 }
220
221 RTC_DCHECK(num_audio_streams_ <= 1);
222 if (num_audio_streams_ == 1) {
223 AudioReceiveStream::Config audio_config;
224 audio_config.rtp.local_ssrc = kReceiverLocalAudioSsrc;
225 audio_config.rtcp_send_transport = rtcp_send_transport;
226 audio_config.voe_channel_id = voe_recv_.channel_id;
227 audio_config.rtp.remote_ssrc = audio_send_config_.rtp.ssrc;
228 audio_receive_configs_.push_back(audio_config);
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000229 }
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000230}
231
232void CallTest::CreateFrameGeneratorCapturer() {
stefanff483612015-12-21 03:14:00 -0800233 VideoStream stream = video_encoder_config_.streams.back();
234 frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create(
235 video_send_stream_->Input(), stream.width, stream.height,
236 stream.max_framerate, clock_));
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000237}
pbosf1828e82015-07-28 08:20:59 -0700238
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100239void CallTest::CreateFakeAudioDevices() {
240 fake_send_audio_device_.reset(new FakeAudioDevice(
241 clock_, test::ResourcePath("voice_engine/audio_long16", "pcm")));
242 fake_recv_audio_device_.reset(new FakeAudioDevice(
243 clock_, test::ResourcePath("voice_engine/audio_long16", "pcm")));
244}
245
246void CallTest::CreateVideoStreams() {
247 RTC_DCHECK(video_send_stream_ == nullptr);
248 RTC_DCHECK(video_receive_streams_.empty());
249 RTC_DCHECK(audio_send_stream_ == nullptr);
250 RTC_DCHECK(audio_receive_streams_.empty());
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000251
stefanff483612015-12-21 03:14:00 -0800252 video_send_stream_ = sender_call_->CreateVideoSendStream(
253 video_send_config_, video_encoder_config_);
stefanff483612015-12-21 03:14:00 -0800254 for (size_t i = 0; i < video_receive_configs_.size(); ++i) {
255 video_receive_streams_.push_back(
256 receiver_call_->CreateVideoReceiveStream(video_receive_configs_[i]));
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000257 }
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000258}
259
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100260void CallTest::CreateAudioStreams() {
261 audio_send_stream_ = sender_call_->CreateAudioSendStream(audio_send_config_);
262 for (size_t i = 0; i < audio_receive_configs_.size(); ++i) {
263 audio_receive_streams_.push_back(
264 receiver_call_->CreateAudioReceiveStream(audio_receive_configs_[i]));
265 }
266 CodecInst isac = {kAudioSendPayloadType, "ISAC", 16000, 480, 1, 32000};
267 EXPECT_EQ(0, voe_send_.codec->SetSendCodec(voe_send_.channel_id, isac));
268}
269
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000270void CallTest::DestroyStreams() {
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100271 if (video_send_stream_)
stefanff483612015-12-21 03:14:00 -0800272 sender_call_->DestroyVideoSendStream(video_send_stream_);
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100273 video_send_stream_ = nullptr;
274 for (VideoReceiveStream* video_recv_stream : video_receive_streams_)
275 receiver_call_->DestroyVideoReceiveStream(video_recv_stream);
276
277 if (audio_send_stream_)
278 sender_call_->DestroyAudioSendStream(audio_send_stream_);
279 audio_send_stream_ = nullptr;
280 for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_)
281 receiver_call_->DestroyAudioReceiveStream(audio_recv_stream);
stefanff483612015-12-21 03:14:00 -0800282 video_receive_streams_.clear();
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100283
pbos@webrtc.org776e6f22014-10-29 15:28:39 +0000284 allocated_decoders_.clear();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000285}
286
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100287void CallTest::CreateVoiceEngines() {
288 CreateFakeAudioDevices();
289 voe_send_.voice_engine = VoiceEngine::Create();
290 voe_send_.base = VoEBase::GetInterface(voe_send_.voice_engine);
291 voe_send_.network = VoENetwork::GetInterface(voe_send_.voice_engine);
292 voe_send_.codec = VoECodec::GetInterface(voe_send_.voice_engine);
293 EXPECT_EQ(0, voe_send_.base->Init(fake_send_audio_device_.get(), nullptr));
294 Config voe_config;
295 voe_config.Set<VoicePacing>(new VoicePacing(true));
296 voe_send_.channel_id = voe_send_.base->CreateChannel(voe_config);
297 EXPECT_GE(voe_send_.channel_id, 0);
298
299 voe_recv_.voice_engine = VoiceEngine::Create();
300 voe_recv_.base = VoEBase::GetInterface(voe_recv_.voice_engine);
301 voe_recv_.network = VoENetwork::GetInterface(voe_recv_.voice_engine);
302 voe_recv_.codec = VoECodec::GetInterface(voe_recv_.voice_engine);
303 EXPECT_EQ(0, voe_recv_.base->Init(fake_recv_audio_device_.get(), nullptr));
304 voe_recv_.channel_id = voe_recv_.base->CreateChannel();
305 EXPECT_GE(voe_recv_.channel_id, 0);
306}
307
308void CallTest::SetupVoiceEngineTransports(PacketTransport* send_transport,
309 PacketTransport* recv_transport) {
310 voe_send_.transport_adapter.reset(
311 new internal::TransportAdapter(send_transport));
312 voe_send_.transport_adapter->Enable();
313 EXPECT_EQ(0, voe_send_.network->RegisterExternalTransport(
314 voe_send_.channel_id, *voe_send_.transport_adapter.get()));
315
316 voe_recv_.transport_adapter.reset(
317 new internal::TransportAdapter(recv_transport));
318 voe_recv_.transport_adapter->Enable();
319 EXPECT_EQ(0, voe_recv_.network->RegisterExternalTransport(
320 voe_recv_.channel_id, *voe_recv_.transport_adapter.get()));
321}
322
323void CallTest::DestroyVoiceEngines() {
324 voe_recv_.base->DeleteChannel(voe_recv_.channel_id);
325 voe_recv_.channel_id = -1;
326 voe_recv_.base->Release();
327 voe_recv_.base = nullptr;
328 voe_recv_.network->Release();
329 voe_recv_.network = nullptr;
330 voe_recv_.codec->Release();
331 voe_recv_.codec = nullptr;
332
333 voe_send_.base->DeleteChannel(voe_send_.channel_id);
334 voe_send_.channel_id = -1;
335 voe_send_.base->Release();
336 voe_send_.base = nullptr;
337 voe_send_.network->Release();
338 voe_send_.network = nullptr;
339 voe_send_.codec->Release();
340 voe_send_.codec = nullptr;
341
342 VoiceEngine::Delete(voe_send_.voice_engine);
343 voe_send_.voice_engine = nullptr;
344 VoiceEngine::Delete(voe_recv_.voice_engine);
345 voe_recv_.voice_engine = nullptr;
346}
347
Peter Boström5811a392015-12-10 13:02:50 +0100348const int CallTest::kDefaultTimeoutMs = 30 * 1000;
349const int CallTest::kLongTimeoutMs = 120 * 1000;
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100350const uint8_t CallTest::kVideoSendPayloadType = 100;
351const uint8_t CallTest::kFakeVideoSendPayloadType = 125;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000352const uint8_t CallTest::kSendRtxPayloadType = 98;
stefan@webrtc.org01581da2014-09-04 06:48:14 +0000353const uint8_t CallTest::kRedPayloadType = 118;
Shao Changbine62202f2015-04-21 20:24:50 +0800354const uint8_t CallTest::kRtxRedPayloadType = 99;
stefan@webrtc.org01581da2014-09-04 06:48:14 +0000355const uint8_t CallTest::kUlpfecPayloadType = 119;
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100356const uint8_t CallTest::kAudioSendPayloadType = 103;
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +0000357const uint32_t CallTest::kSendRtxSsrcs[kNumSsrcs] = {0xBADCAFD, 0xBADCAFE,
358 0xBADCAFF};
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100359const uint32_t CallTest::kVideoSendSsrcs[kNumSsrcs] = {0xC0FFED, 0xC0FFEE,
360 0xC0FFEF};
361const uint32_t CallTest::kAudioSendSsrc = 0xDEADBEEF;
362const uint32_t CallTest::kReceiverLocalVideoSsrc = 0x123456;
363const uint32_t CallTest::kReceiverLocalAudioSsrc = 0x1234567;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000364const int CallTest::kNackRtpHistoryMs = 1000;
365
366BaseTest::BaseTest(unsigned int timeout_ms) : RtpRtcpObserver(timeout_ms) {
367}
368
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000369BaseTest::~BaseTest() {
370}
371
372Call::Config BaseTest::GetSenderCallConfig() {
solenberg4fbae2b2015-08-28 04:07:10 -0700373 return Call::Config();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000374}
375
376Call::Config BaseTest::GetReceiverCallConfig() {
solenberg4fbae2b2015-08-28 04:07:10 -0700377 return Call::Config();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000378}
379
380void BaseTest::OnCallsCreated(Call* sender_call, Call* receiver_call) {
381}
382
stefane74eef12016-01-08 06:47:13 -0800383test::PacketTransport* BaseTest::CreateSendTransport(Call* sender_call) {
384 return new PacketTransport(sender_call, this, test::PacketTransport::kSender,
385 FakeNetworkPipe::Config());
386}
387
388test::PacketTransport* BaseTest::CreateReceiveTransport() {
389 return new PacketTransport(nullptr, this, test::PacketTransport::kReceiver,
390 FakeNetworkPipe::Config());
391}
stefanf116bd02015-10-27 08:29:42 -0700392
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100393size_t BaseTest::GetNumVideoStreams() const {
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000394 return 1;
395}
396
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100397size_t BaseTest::GetNumAudioStreams() const {
398 return 0;
399}
400
stefanff483612015-12-21 03:14:00 -0800401void BaseTest::ModifyVideoConfigs(
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000402 VideoSendStream::Config* send_config,
403 std::vector<VideoReceiveStream::Config>* receive_configs,
stefanff483612015-12-21 03:14:00 -0800404 VideoEncoderConfig* encoder_config) {}
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000405
stefanff483612015-12-21 03:14:00 -0800406void BaseTest::OnVideoStreamsCreated(
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000407 VideoSendStream* send_stream,
stefanff483612015-12-21 03:14:00 -0800408 const std::vector<VideoReceiveStream*>& receive_streams) {}
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000409
Stefan Holmer9fea80f2016-01-07 17:43:18 +0100410void BaseTest::ModifyAudioConfigs(
411 AudioSendStream::Config* send_config,
412 std::vector<AudioReceiveStream::Config>* receive_configs) {}
413
414void BaseTest::OnAudioStreamsCreated(
415 AudioSendStream* send_stream,
416 const std::vector<AudioReceiveStream*>& receive_streams) {}
417
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000418void BaseTest::OnFrameGeneratorCapturerCreated(
419 FrameGeneratorCapturer* frame_generator_capturer) {
420}
421
422SendTest::SendTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
423}
424
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000425bool SendTest::ShouldCreateReceivers() const {
426 return false;
427}
428
429EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
430}
431
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000432bool EndToEndTest::ShouldCreateReceivers() const {
433 return true;
434}
435
436} // namespace test
437} // namespace webrtc