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| 1 /* | |
| 2 * Copyright 2012 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 */ | |
| 10 | |
| 11 #include <memory> | |
| 12 #include <string> | |
| 13 #include <utility> | |
| 14 | |
| 15 #include "webrtc/api/audiotrack.h" | |
| 16 #include "webrtc/api/fakemediacontroller.h" | |
| 17 #include "webrtc/api/localaudiosource.h" | |
| 18 #include "webrtc/api/mediastream.h" | |
| 19 #include "webrtc/api/remoteaudiosource.h" | |
| 20 #include "webrtc/api/rtpreceiver.h" | |
| 21 #include "webrtc/api/rtpsender.h" | |
| 22 #include "webrtc/api/streamcollection.h" | |
| 23 #include "webrtc/api/test/fakevideotracksource.h" | |
| 24 #include "webrtc/api/videotrack.h" | |
| 25 #include "webrtc/api/videotracksource.h" | |
| 26 #include "webrtc/base/gunit.h" | |
| 27 #include "webrtc/logging/rtc_event_log/rtc_event_log.h" | |
| 28 #include "webrtc/media/base/fakemediaengine.h" | |
| 29 #include "webrtc/media/base/mediachannel.h" | |
| 30 #include "webrtc/media/engine/fakewebrtccall.h" | |
| 31 #include "webrtc/p2p/base/faketransportcontroller.h" | |
| 32 #include "webrtc/pc/channelmanager.h" | |
| 33 #include "webrtc/test/gmock.h" | |
| 34 #include "webrtc/test/gtest.h" | |
| 35 | |
| 36 using ::testing::_; | |
| 37 using ::testing::Exactly; | |
| 38 using ::testing::InvokeWithoutArgs; | |
| 39 using ::testing::Return; | |
| 40 | |
| 41 static const char kStreamLabel1[] = "local_stream_1"; | |
| 42 static const char kVideoTrackId[] = "video_1"; | |
| 43 static const char kAudioTrackId[] = "audio_1"; | |
| 44 static const uint32_t kVideoSsrc = 98; | |
| 45 static const uint32_t kVideoSsrc2 = 100; | |
| 46 static const uint32_t kAudioSsrc = 99; | |
| 47 static const uint32_t kAudioSsrc2 = 101; | |
| 48 | |
| 49 namespace webrtc { | |
| 50 | |
| 51 class RtpSenderReceiverTest : public testing::Test { | |
| 52 public: | |
| 53 RtpSenderReceiverTest() | |
| 54 : // Create fake media engine/etc. so we can create channels to use to | |
| 55 // test RtpSenders/RtpReceivers. | |
| 56 media_engine_(new cricket::FakeMediaEngine()), | |
| 57 channel_manager_(media_engine_, | |
| 58 rtc::Thread::Current(), | |
| 59 rtc::Thread::Current()), | |
| 60 fake_call_(Call::Config(&event_log_)), | |
| 61 fake_media_controller_(&channel_manager_, &fake_call_), | |
| 62 stream_(MediaStream::Create(kStreamLabel1)) { | |
| 63 // Create channels to be used by the RtpSenders and RtpReceivers. | |
| 64 channel_manager_.Init(); | |
| 65 bool rtcp_mux_required = true; | |
| 66 bool srtp_required = true; | |
| 67 cricket::DtlsTransportInternal* rtp_transport = | |
| 68 fake_transport_controller_.CreateDtlsTransport( | |
| 69 cricket::CN_AUDIO, cricket::ICE_CANDIDATE_COMPONENT_RTP); | |
| 70 voice_channel_ = channel_manager_.CreateVoiceChannel( | |
| 71 &fake_media_controller_, rtp_transport, nullptr, rtc::Thread::Current(), | |
| 72 cricket::CN_AUDIO, nullptr, rtcp_mux_required, srtp_required, | |
| 73 cricket::AudioOptions()); | |
| 74 video_channel_ = channel_manager_.CreateVideoChannel( | |
| 75 &fake_media_controller_, rtp_transport, nullptr, rtc::Thread::Current(), | |
| 76 cricket::CN_VIDEO, nullptr, rtcp_mux_required, srtp_required, | |
| 77 cricket::VideoOptions()); | |
| 78 voice_media_channel_ = media_engine_->GetVoiceChannel(0); | |
| 79 video_media_channel_ = media_engine_->GetVideoChannel(0); | |
| 80 RTC_CHECK(voice_channel_); | |
| 81 RTC_CHECK(video_channel_); | |
| 82 RTC_CHECK(voice_media_channel_); | |
| 83 RTC_CHECK(video_media_channel_); | |
| 84 | |
| 85 // Create streams for predefined SSRCs. Streams need to exist in order | |
| 86 // for the senders and receievers to apply parameters to them. | |
| 87 // Normally these would be created by SetLocalDescription and | |
| 88 // SetRemoteDescription. | |
| 89 voice_media_channel_->AddSendStream( | |
| 90 cricket::StreamParams::CreateLegacy(kAudioSsrc)); | |
| 91 voice_media_channel_->AddRecvStream( | |
| 92 cricket::StreamParams::CreateLegacy(kAudioSsrc)); | |
| 93 voice_media_channel_->AddSendStream( | |
| 94 cricket::StreamParams::CreateLegacy(kAudioSsrc2)); | |
| 95 voice_media_channel_->AddRecvStream( | |
| 96 cricket::StreamParams::CreateLegacy(kAudioSsrc2)); | |
| 97 video_media_channel_->AddSendStream( | |
| 98 cricket::StreamParams::CreateLegacy(kVideoSsrc)); | |
| 99 video_media_channel_->AddRecvStream( | |
| 100 cricket::StreamParams::CreateLegacy(kVideoSsrc)); | |
| 101 video_media_channel_->AddSendStream( | |
| 102 cricket::StreamParams::CreateLegacy(kVideoSsrc2)); | |
| 103 video_media_channel_->AddRecvStream( | |
| 104 cricket::StreamParams::CreateLegacy(kVideoSsrc2)); | |
| 105 } | |
| 106 | |
| 107 void TearDown() override { channel_manager_.Terminate(); } | |
| 108 | |
| 109 void AddVideoTrack() { AddVideoTrack(false); } | |
| 110 | |
| 111 void AddVideoTrack(bool is_screencast) { | |
| 112 rtc::scoped_refptr<VideoTrackSourceInterface> source( | |
| 113 FakeVideoTrackSource::Create(is_screencast)); | |
| 114 video_track_ = VideoTrack::Create(kVideoTrackId, source); | |
| 115 EXPECT_TRUE(stream_->AddTrack(video_track_)); | |
| 116 } | |
| 117 | |
| 118 void CreateAudioRtpSender() { CreateAudioRtpSender(nullptr); } | |
| 119 | |
| 120 void CreateAudioRtpSender(rtc::scoped_refptr<LocalAudioSource> source) { | |
| 121 audio_track_ = AudioTrack::Create(kAudioTrackId, source); | |
| 122 EXPECT_TRUE(stream_->AddTrack(audio_track_)); | |
| 123 audio_rtp_sender_ = | |
| 124 new AudioRtpSender(stream_->GetAudioTracks()[0], stream_->label(), | |
| 125 voice_channel_, nullptr); | |
| 126 audio_rtp_sender_->SetSsrc(kAudioSsrc); | |
| 127 VerifyVoiceChannelInput(); | |
| 128 } | |
| 129 | |
| 130 void CreateVideoRtpSender() { CreateVideoRtpSender(false); } | |
| 131 | |
| 132 void CreateVideoRtpSender(bool is_screencast) { | |
| 133 AddVideoTrack(is_screencast); | |
| 134 video_rtp_sender_ = new VideoRtpSender(stream_->GetVideoTracks()[0], | |
| 135 stream_->label(), video_channel_); | |
| 136 video_rtp_sender_->SetSsrc(kVideoSsrc); | |
| 137 VerifyVideoChannelInput(); | |
| 138 } | |
| 139 | |
| 140 void DestroyAudioRtpSender() { | |
| 141 audio_rtp_sender_ = nullptr; | |
| 142 VerifyVoiceChannelNoInput(); | |
| 143 } | |
| 144 | |
| 145 void DestroyVideoRtpSender() { | |
| 146 video_rtp_sender_ = nullptr; | |
| 147 VerifyVideoChannelNoInput(); | |
| 148 } | |
| 149 | |
| 150 void CreateAudioRtpReceiver() { | |
| 151 audio_track_ = AudioTrack::Create( | |
| 152 kAudioTrackId, RemoteAudioSource::Create(kAudioSsrc, NULL)); | |
| 153 EXPECT_TRUE(stream_->AddTrack(audio_track_)); | |
| 154 audio_rtp_receiver_ = new AudioRtpReceiver(stream_, kAudioTrackId, | |
| 155 kAudioSsrc, voice_channel_); | |
| 156 audio_track_ = audio_rtp_receiver_->audio_track(); | |
| 157 VerifyVoiceChannelOutput(); | |
| 158 } | |
| 159 | |
| 160 void CreateVideoRtpReceiver() { | |
| 161 video_rtp_receiver_ = | |
| 162 new VideoRtpReceiver(stream_, kVideoTrackId, rtc::Thread::Current(), | |
| 163 kVideoSsrc, video_channel_); | |
| 164 video_track_ = video_rtp_receiver_->video_track(); | |
| 165 VerifyVideoChannelOutput(); | |
| 166 } | |
| 167 | |
| 168 void DestroyAudioRtpReceiver() { | |
| 169 audio_rtp_receiver_ = nullptr; | |
| 170 VerifyVoiceChannelNoOutput(); | |
| 171 } | |
| 172 | |
| 173 void DestroyVideoRtpReceiver() { | |
| 174 video_rtp_receiver_ = nullptr; | |
| 175 VerifyVideoChannelNoOutput(); | |
| 176 } | |
| 177 | |
| 178 void VerifyVoiceChannelInput() { VerifyVoiceChannelInput(kAudioSsrc); } | |
| 179 | |
| 180 void VerifyVoiceChannelInput(uint32_t ssrc) { | |
| 181 // Verify that the media channel has an audio source, and the stream isn't | |
| 182 // muted. | |
| 183 EXPECT_TRUE(voice_media_channel_->HasSource(ssrc)); | |
| 184 EXPECT_FALSE(voice_media_channel_->IsStreamMuted(ssrc)); | |
| 185 } | |
| 186 | |
| 187 void VerifyVideoChannelInput() { VerifyVideoChannelInput(kVideoSsrc); } | |
| 188 | |
| 189 void VerifyVideoChannelInput(uint32_t ssrc) { | |
| 190 // Verify that the media channel has a video source, | |
| 191 EXPECT_TRUE(video_media_channel_->HasSource(ssrc)); | |
| 192 } | |
| 193 | |
| 194 void VerifyVoiceChannelNoInput() { VerifyVoiceChannelNoInput(kAudioSsrc); } | |
| 195 | |
| 196 void VerifyVoiceChannelNoInput(uint32_t ssrc) { | |
| 197 // Verify that the media channel's source is reset. | |
| 198 EXPECT_FALSE(voice_media_channel_->HasSource(ssrc)); | |
| 199 } | |
| 200 | |
| 201 void VerifyVideoChannelNoInput() { VerifyVideoChannelNoInput(kVideoSsrc); } | |
| 202 | |
| 203 void VerifyVideoChannelNoInput(uint32_t ssrc) { | |
| 204 // Verify that the media channel's source is reset. | |
| 205 EXPECT_FALSE(video_media_channel_->HasSource(ssrc)); | |
| 206 } | |
| 207 | |
| 208 void VerifyVoiceChannelOutput() { | |
| 209 // Verify that the volume is initialized to 1. | |
| 210 double volume; | |
| 211 EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume)); | |
| 212 EXPECT_EQ(1, volume); | |
| 213 } | |
| 214 | |
| 215 void VerifyVideoChannelOutput() { | |
| 216 // Verify that the media channel has a sink. | |
| 217 EXPECT_TRUE(video_media_channel_->HasSink(kVideoSsrc)); | |
| 218 } | |
| 219 | |
| 220 void VerifyVoiceChannelNoOutput() { | |
| 221 // Verify that the volume is reset to 0. | |
| 222 double volume; | |
| 223 EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume)); | |
| 224 EXPECT_EQ(0, volume); | |
| 225 } | |
| 226 | |
| 227 void VerifyVideoChannelNoOutput() { | |
| 228 // Verify that the media channel's sink is reset. | |
| 229 EXPECT_FALSE(video_media_channel_->HasSink(kVideoSsrc)); | |
| 230 } | |
| 231 | |
| 232 protected: | |
| 233 webrtc::RtcEventLogNullImpl event_log_; | |
| 234 cricket::FakeMediaEngine* media_engine_; | |
| 235 cricket::FakeTransportController fake_transport_controller_; | |
| 236 cricket::ChannelManager channel_manager_; | |
| 237 cricket::FakeCall fake_call_; | |
| 238 cricket::FakeMediaController fake_media_controller_; | |
| 239 cricket::VoiceChannel* voice_channel_; | |
| 240 cricket::VideoChannel* video_channel_; | |
| 241 cricket::FakeVoiceMediaChannel* voice_media_channel_; | |
| 242 cricket::FakeVideoMediaChannel* video_media_channel_; | |
| 243 rtc::scoped_refptr<AudioRtpSender> audio_rtp_sender_; | |
| 244 rtc::scoped_refptr<VideoRtpSender> video_rtp_sender_; | |
| 245 rtc::scoped_refptr<AudioRtpReceiver> audio_rtp_receiver_; | |
| 246 rtc::scoped_refptr<VideoRtpReceiver> video_rtp_receiver_; | |
| 247 rtc::scoped_refptr<MediaStreamInterface> stream_; | |
| 248 rtc::scoped_refptr<VideoTrackInterface> video_track_; | |
| 249 rtc::scoped_refptr<AudioTrackInterface> audio_track_; | |
| 250 }; | |
| 251 | |
| 252 // Test that |voice_channel_| is updated when an audio track is associated | |
| 253 // and disassociated with an AudioRtpSender. | |
| 254 TEST_F(RtpSenderReceiverTest, AddAndDestroyAudioRtpSender) { | |
| 255 CreateAudioRtpSender(); | |
| 256 DestroyAudioRtpSender(); | |
| 257 } | |
| 258 | |
| 259 // Test that |video_channel_| is updated when a video track is associated and | |
| 260 // disassociated with a VideoRtpSender. | |
| 261 TEST_F(RtpSenderReceiverTest, AddAndDestroyVideoRtpSender) { | |
| 262 CreateVideoRtpSender(); | |
| 263 DestroyVideoRtpSender(); | |
| 264 } | |
| 265 | |
| 266 // Test that |voice_channel_| is updated when a remote audio track is | |
| 267 // associated and disassociated with an AudioRtpReceiver. | |
| 268 TEST_F(RtpSenderReceiverTest, AddAndDestroyAudioRtpReceiver) { | |
| 269 CreateAudioRtpReceiver(); | |
| 270 DestroyAudioRtpReceiver(); | |
| 271 } | |
| 272 | |
| 273 // Test that |video_channel_| is updated when a remote video track is | |
| 274 // associated and disassociated with a VideoRtpReceiver. | |
| 275 TEST_F(RtpSenderReceiverTest, AddAndDestroyVideoRtpReceiver) { | |
| 276 CreateVideoRtpReceiver(); | |
| 277 DestroyVideoRtpReceiver(); | |
| 278 } | |
| 279 | |
| 280 // Test that the AudioRtpSender applies options from the local audio source. | |
| 281 TEST_F(RtpSenderReceiverTest, LocalAudioSourceOptionsApplied) { | |
| 282 cricket::AudioOptions options; | |
| 283 options.echo_cancellation = rtc::Optional<bool>(true); | |
| 284 auto source = LocalAudioSource::Create( | |
| 285 PeerConnectionFactoryInterface::Options(), &options); | |
| 286 CreateAudioRtpSender(source.get()); | |
| 287 | |
| 288 EXPECT_EQ(rtc::Optional<bool>(true), | |
| 289 voice_media_channel_->options().echo_cancellation); | |
| 290 | |
| 291 DestroyAudioRtpSender(); | |
| 292 } | |
| 293 | |
| 294 // Test that the stream is muted when the track is disabled, and unmuted when | |
| 295 // the track is enabled. | |
| 296 TEST_F(RtpSenderReceiverTest, LocalAudioTrackDisable) { | |
| 297 CreateAudioRtpSender(); | |
| 298 | |
| 299 audio_track_->set_enabled(false); | |
| 300 EXPECT_TRUE(voice_media_channel_->IsStreamMuted(kAudioSsrc)); | |
| 301 | |
| 302 audio_track_->set_enabled(true); | |
| 303 EXPECT_FALSE(voice_media_channel_->IsStreamMuted(kAudioSsrc)); | |
| 304 | |
| 305 DestroyAudioRtpSender(); | |
| 306 } | |
| 307 | |
| 308 // Test that the volume is set to 0 when the track is disabled, and back to | |
| 309 // 1 when the track is enabled. | |
| 310 TEST_F(RtpSenderReceiverTest, RemoteAudioTrackDisable) { | |
| 311 CreateAudioRtpReceiver(); | |
| 312 | |
| 313 double volume; | |
| 314 EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume)); | |
| 315 EXPECT_EQ(1, volume); | |
| 316 | |
| 317 audio_track_->set_enabled(false); | |
| 318 EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume)); | |
| 319 EXPECT_EQ(0, volume); | |
| 320 | |
| 321 audio_track_->set_enabled(true); | |
| 322 EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume)); | |
| 323 EXPECT_EQ(1, volume); | |
| 324 | |
| 325 DestroyAudioRtpReceiver(); | |
| 326 } | |
| 327 | |
| 328 // Currently no action is taken when a remote video track is disabled or | |
| 329 // enabled, so there's nothing to test here, other than what is normally | |
| 330 // verified in DestroyVideoRtpSender. | |
| 331 TEST_F(RtpSenderReceiverTest, LocalVideoTrackDisable) { | |
| 332 CreateVideoRtpSender(); | |
| 333 | |
| 334 video_track_->set_enabled(false); | |
| 335 video_track_->set_enabled(true); | |
| 336 | |
| 337 DestroyVideoRtpSender(); | |
| 338 } | |
| 339 | |
| 340 // Test that the state of the video track created by the VideoRtpReceiver is | |
| 341 // updated when the receiver is destroyed. | |
| 342 TEST_F(RtpSenderReceiverTest, RemoteVideoTrackState) { | |
| 343 CreateVideoRtpReceiver(); | |
| 344 | |
| 345 EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, video_track_->state()); | |
| 346 EXPECT_EQ(webrtc::MediaSourceInterface::kLive, | |
| 347 video_track_->GetSource()->state()); | |
| 348 | |
| 349 DestroyVideoRtpReceiver(); | |
| 350 | |
| 351 EXPECT_EQ(webrtc::MediaStreamTrackInterface::kEnded, video_track_->state()); | |
| 352 EXPECT_EQ(webrtc::MediaSourceInterface::kEnded, | |
| 353 video_track_->GetSource()->state()); | |
| 354 } | |
| 355 | |
| 356 // Currently no action is taken when a remote video track is disabled or | |
| 357 // enabled, so there's nothing to test here, other than what is normally | |
| 358 // verified in DestroyVideoRtpReceiver. | |
| 359 TEST_F(RtpSenderReceiverTest, RemoteVideoTrackDisable) { | |
| 360 CreateVideoRtpReceiver(); | |
| 361 | |
| 362 video_track_->set_enabled(false); | |
| 363 video_track_->set_enabled(true); | |
| 364 | |
| 365 DestroyVideoRtpReceiver(); | |
| 366 } | |
| 367 | |
| 368 // Test that the AudioRtpReceiver applies volume changes from the track source | |
| 369 // to the media channel. | |
| 370 TEST_F(RtpSenderReceiverTest, RemoteAudioTrackSetVolume) { | |
| 371 CreateAudioRtpReceiver(); | |
| 372 | |
| 373 double volume; | |
| 374 audio_track_->GetSource()->SetVolume(0.5); | |
| 375 EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume)); | |
| 376 EXPECT_EQ(0.5, volume); | |
| 377 | |
| 378 // Disable the audio track, this should prevent setting the volume. | |
| 379 audio_track_->set_enabled(false); | |
| 380 audio_track_->GetSource()->SetVolume(0.8); | |
| 381 EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume)); | |
| 382 EXPECT_EQ(0, volume); | |
| 383 | |
| 384 // When the track is enabled, the previously set volume should take effect. | |
| 385 audio_track_->set_enabled(true); | |
| 386 EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume)); | |
| 387 EXPECT_EQ(0.8, volume); | |
| 388 | |
| 389 // Try changing volume one more time. | |
| 390 audio_track_->GetSource()->SetVolume(0.9); | |
| 391 EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume)); | |
| 392 EXPECT_EQ(0.9, volume); | |
| 393 | |
| 394 DestroyAudioRtpReceiver(); | |
| 395 } | |
| 396 | |
| 397 // Test that the media channel isn't enabled for sending if the audio sender | |
| 398 // doesn't have both a track and SSRC. | |
| 399 TEST_F(RtpSenderReceiverTest, AudioSenderWithoutTrackAndSsrc) { | |
| 400 audio_rtp_sender_ = new AudioRtpSender(voice_channel_, nullptr); | |
| 401 rtc::scoped_refptr<AudioTrackInterface> track = | |
| 402 AudioTrack::Create(kAudioTrackId, nullptr); | |
| 403 | |
| 404 // Track but no SSRC. | |
| 405 EXPECT_TRUE(audio_rtp_sender_->SetTrack(track)); | |
| 406 VerifyVoiceChannelNoInput(); | |
| 407 | |
| 408 // SSRC but no track. | |
| 409 EXPECT_TRUE(audio_rtp_sender_->SetTrack(nullptr)); | |
| 410 audio_rtp_sender_->SetSsrc(kAudioSsrc); | |
| 411 VerifyVoiceChannelNoInput(); | |
| 412 } | |
| 413 | |
| 414 // Test that the media channel isn't enabled for sending if the video sender | |
| 415 // doesn't have both a track and SSRC. | |
| 416 TEST_F(RtpSenderReceiverTest, VideoSenderWithoutTrackAndSsrc) { | |
| 417 video_rtp_sender_ = new VideoRtpSender(video_channel_); | |
| 418 | |
| 419 // Track but no SSRC. | |
| 420 EXPECT_TRUE(video_rtp_sender_->SetTrack(video_track_)); | |
| 421 VerifyVideoChannelNoInput(); | |
| 422 | |
| 423 // SSRC but no track. | |
| 424 EXPECT_TRUE(video_rtp_sender_->SetTrack(nullptr)); | |
| 425 video_rtp_sender_->SetSsrc(kVideoSsrc); | |
| 426 VerifyVideoChannelNoInput(); | |
| 427 } | |
| 428 | |
| 429 // Test that the media channel is enabled for sending when the audio sender | |
| 430 // has a track and SSRC, when the SSRC is set first. | |
| 431 TEST_F(RtpSenderReceiverTest, AudioSenderEarlyWarmupSsrcThenTrack) { | |
| 432 audio_rtp_sender_ = new AudioRtpSender(voice_channel_, nullptr); | |
| 433 rtc::scoped_refptr<AudioTrackInterface> track = | |
| 434 AudioTrack::Create(kAudioTrackId, nullptr); | |
| 435 audio_rtp_sender_->SetSsrc(kAudioSsrc); | |
| 436 audio_rtp_sender_->SetTrack(track); | |
| 437 VerifyVoiceChannelInput(); | |
| 438 | |
| 439 DestroyAudioRtpSender(); | |
| 440 } | |
| 441 | |
| 442 // Test that the media channel is enabled for sending when the audio sender | |
| 443 // has a track and SSRC, when the SSRC is set last. | |
| 444 TEST_F(RtpSenderReceiverTest, AudioSenderEarlyWarmupTrackThenSsrc) { | |
| 445 audio_rtp_sender_ = new AudioRtpSender(voice_channel_, nullptr); | |
| 446 rtc::scoped_refptr<AudioTrackInterface> track = | |
| 447 AudioTrack::Create(kAudioTrackId, nullptr); | |
| 448 audio_rtp_sender_->SetTrack(track); | |
| 449 audio_rtp_sender_->SetSsrc(kAudioSsrc); | |
| 450 VerifyVoiceChannelInput(); | |
| 451 | |
| 452 DestroyAudioRtpSender(); | |
| 453 } | |
| 454 | |
| 455 // Test that the media channel is enabled for sending when the video sender | |
| 456 // has a track and SSRC, when the SSRC is set first. | |
| 457 TEST_F(RtpSenderReceiverTest, VideoSenderEarlyWarmupSsrcThenTrack) { | |
| 458 AddVideoTrack(); | |
| 459 video_rtp_sender_ = new VideoRtpSender(video_channel_); | |
| 460 video_rtp_sender_->SetSsrc(kVideoSsrc); | |
| 461 video_rtp_sender_->SetTrack(video_track_); | |
| 462 VerifyVideoChannelInput(); | |
| 463 | |
| 464 DestroyVideoRtpSender(); | |
| 465 } | |
| 466 | |
| 467 // Test that the media channel is enabled for sending when the video sender | |
| 468 // has a track and SSRC, when the SSRC is set last. | |
| 469 TEST_F(RtpSenderReceiverTest, VideoSenderEarlyWarmupTrackThenSsrc) { | |
| 470 AddVideoTrack(); | |
| 471 video_rtp_sender_ = new VideoRtpSender(video_channel_); | |
| 472 video_rtp_sender_->SetTrack(video_track_); | |
| 473 video_rtp_sender_->SetSsrc(kVideoSsrc); | |
| 474 VerifyVideoChannelInput(); | |
| 475 | |
| 476 DestroyVideoRtpSender(); | |
| 477 } | |
| 478 | |
| 479 // Test that the media channel stops sending when the audio sender's SSRC is set | |
| 480 // to 0. | |
| 481 TEST_F(RtpSenderReceiverTest, AudioSenderSsrcSetToZero) { | |
| 482 CreateAudioRtpSender(); | |
| 483 | |
| 484 audio_rtp_sender_->SetSsrc(0); | |
| 485 VerifyVoiceChannelNoInput(); | |
| 486 } | |
| 487 | |
| 488 // Test that the media channel stops sending when the video sender's SSRC is set | |
| 489 // to 0. | |
| 490 TEST_F(RtpSenderReceiverTest, VideoSenderSsrcSetToZero) { | |
| 491 CreateAudioRtpSender(); | |
| 492 | |
| 493 audio_rtp_sender_->SetSsrc(0); | |
| 494 VerifyVideoChannelNoInput(); | |
| 495 } | |
| 496 | |
| 497 // Test that the media channel stops sending when the audio sender's track is | |
| 498 // set to null. | |
| 499 TEST_F(RtpSenderReceiverTest, AudioSenderTrackSetToNull) { | |
| 500 CreateAudioRtpSender(); | |
| 501 | |
| 502 EXPECT_TRUE(audio_rtp_sender_->SetTrack(nullptr)); | |
| 503 VerifyVoiceChannelNoInput(); | |
| 504 } | |
| 505 | |
| 506 // Test that the media channel stops sending when the video sender's track is | |
| 507 // set to null. | |
| 508 TEST_F(RtpSenderReceiverTest, VideoSenderTrackSetToNull) { | |
| 509 CreateVideoRtpSender(); | |
| 510 | |
| 511 video_rtp_sender_->SetSsrc(0); | |
| 512 VerifyVideoChannelNoInput(); | |
| 513 } | |
| 514 | |
| 515 // Test that when the audio sender's SSRC is changed, the media channel stops | |
| 516 // sending with the old SSRC and starts sending with the new one. | |
| 517 TEST_F(RtpSenderReceiverTest, AudioSenderSsrcChanged) { | |
| 518 CreateAudioRtpSender(); | |
| 519 | |
| 520 audio_rtp_sender_->SetSsrc(kAudioSsrc2); | |
| 521 VerifyVoiceChannelNoInput(kAudioSsrc); | |
| 522 VerifyVoiceChannelInput(kAudioSsrc2); | |
| 523 | |
| 524 audio_rtp_sender_ = nullptr; | |
| 525 VerifyVoiceChannelNoInput(kAudioSsrc2); | |
| 526 } | |
| 527 | |
| 528 // Test that when the audio sender's SSRC is changed, the media channel stops | |
| 529 // sending with the old SSRC and starts sending with the new one. | |
| 530 TEST_F(RtpSenderReceiverTest, VideoSenderSsrcChanged) { | |
| 531 CreateVideoRtpSender(); | |
| 532 | |
| 533 video_rtp_sender_->SetSsrc(kVideoSsrc2); | |
| 534 VerifyVideoChannelNoInput(kVideoSsrc); | |
| 535 VerifyVideoChannelInput(kVideoSsrc2); | |
| 536 | |
| 537 video_rtp_sender_ = nullptr; | |
| 538 VerifyVideoChannelNoInput(kVideoSsrc2); | |
| 539 } | |
| 540 | |
| 541 TEST_F(RtpSenderReceiverTest, AudioSenderCanSetParameters) { | |
| 542 CreateAudioRtpSender(); | |
| 543 | |
| 544 RtpParameters params = audio_rtp_sender_->GetParameters(); | |
| 545 EXPECT_EQ(1u, params.encodings.size()); | |
| 546 EXPECT_TRUE(audio_rtp_sender_->SetParameters(params)); | |
| 547 | |
| 548 DestroyAudioRtpSender(); | |
| 549 } | |
| 550 | |
| 551 TEST_F(RtpSenderReceiverTest, SetAudioMaxSendBitrate) { | |
| 552 CreateAudioRtpSender(); | |
| 553 | |
| 554 EXPECT_EQ(-1, voice_media_channel_->max_bps()); | |
| 555 webrtc::RtpParameters params = audio_rtp_sender_->GetParameters(); | |
| 556 EXPECT_EQ(1, params.encodings.size()); | |
| 557 EXPECT_EQ(-1, params.encodings[0].max_bitrate_bps); | |
| 558 params.encodings[0].max_bitrate_bps = 1000; | |
| 559 EXPECT_TRUE(audio_rtp_sender_->SetParameters(params)); | |
| 560 | |
| 561 // Read back the parameters and verify they have been changed. | |
| 562 params = audio_rtp_sender_->GetParameters(); | |
| 563 EXPECT_EQ(1, params.encodings.size()); | |
| 564 EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps); | |
| 565 | |
| 566 // Verify that the audio channel received the new parameters. | |
| 567 params = voice_media_channel_->GetRtpSendParameters(kAudioSsrc); | |
| 568 EXPECT_EQ(1, params.encodings.size()); | |
| 569 EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps); | |
| 570 | |
| 571 // Verify that the global bitrate limit has not been changed. | |
| 572 EXPECT_EQ(-1, voice_media_channel_->max_bps()); | |
| 573 | |
| 574 DestroyAudioRtpSender(); | |
| 575 } | |
| 576 | |
| 577 TEST_F(RtpSenderReceiverTest, VideoSenderCanSetParameters) { | |
| 578 CreateVideoRtpSender(); | |
| 579 | |
| 580 RtpParameters params = video_rtp_sender_->GetParameters(); | |
| 581 EXPECT_EQ(1u, params.encodings.size()); | |
| 582 EXPECT_TRUE(video_rtp_sender_->SetParameters(params)); | |
| 583 | |
| 584 DestroyVideoRtpSender(); | |
| 585 } | |
| 586 | |
| 587 TEST_F(RtpSenderReceiverTest, SetVideoMaxSendBitrate) { | |
| 588 CreateVideoRtpSender(); | |
| 589 | |
| 590 EXPECT_EQ(-1, video_media_channel_->max_bps()); | |
| 591 webrtc::RtpParameters params = video_rtp_sender_->GetParameters(); | |
| 592 EXPECT_EQ(1, params.encodings.size()); | |
| 593 EXPECT_EQ(-1, params.encodings[0].max_bitrate_bps); | |
| 594 params.encodings[0].max_bitrate_bps = 1000; | |
| 595 EXPECT_TRUE(video_rtp_sender_->SetParameters(params)); | |
| 596 | |
| 597 // Read back the parameters and verify they have been changed. | |
| 598 params = video_rtp_sender_->GetParameters(); | |
| 599 EXPECT_EQ(1, params.encodings.size()); | |
| 600 EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps); | |
| 601 | |
| 602 // Verify that the video channel received the new parameters. | |
| 603 params = video_media_channel_->GetRtpSendParameters(kVideoSsrc); | |
| 604 EXPECT_EQ(1, params.encodings.size()); | |
| 605 EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps); | |
| 606 | |
| 607 // Verify that the global bitrate limit has not been changed. | |
| 608 EXPECT_EQ(-1, video_media_channel_->max_bps()); | |
| 609 | |
| 610 DestroyVideoRtpSender(); | |
| 611 } | |
| 612 | |
| 613 TEST_F(RtpSenderReceiverTest, AudioReceiverCanSetParameters) { | |
| 614 CreateAudioRtpReceiver(); | |
| 615 | |
| 616 RtpParameters params = audio_rtp_receiver_->GetParameters(); | |
| 617 EXPECT_EQ(1u, params.encodings.size()); | |
| 618 EXPECT_TRUE(audio_rtp_receiver_->SetParameters(params)); | |
| 619 | |
| 620 DestroyAudioRtpReceiver(); | |
| 621 } | |
| 622 | |
| 623 TEST_F(RtpSenderReceiverTest, VideoReceiverCanSetParameters) { | |
| 624 CreateVideoRtpReceiver(); | |
| 625 | |
| 626 RtpParameters params = video_rtp_receiver_->GetParameters(); | |
| 627 EXPECT_EQ(1u, params.encodings.size()); | |
| 628 EXPECT_TRUE(video_rtp_receiver_->SetParameters(params)); | |
| 629 | |
| 630 DestroyVideoRtpReceiver(); | |
| 631 } | |
| 632 | |
| 633 // Test that makes sure that a video track content hint translates to the proper | |
| 634 // value for sources that are not screencast. | |
| 635 TEST_F(RtpSenderReceiverTest, PropagatesVideoTrackContentHint) { | |
| 636 CreateVideoRtpSender(); | |
| 637 | |
| 638 video_track_->set_enabled(true); | |
| 639 | |
| 640 // |video_track_| is not screencast by default. | |
| 641 EXPECT_EQ(rtc::Optional<bool>(false), | |
| 642 video_media_channel_->options().is_screencast); | |
| 643 // No content hint should be set by default. | |
| 644 EXPECT_EQ(VideoTrackInterface::ContentHint::kNone, | |
| 645 video_track_->content_hint()); | |
| 646 // Setting detailed should turn a non-screencast source into screencast mode. | |
| 647 video_track_->set_content_hint(VideoTrackInterface::ContentHint::kDetailed); | |
| 648 EXPECT_EQ(rtc::Optional<bool>(true), | |
| 649 video_media_channel_->options().is_screencast); | |
| 650 // Removing the content hint should turn the track back into non-screencast | |
| 651 // mode. | |
| 652 video_track_->set_content_hint(VideoTrackInterface::ContentHint::kNone); | |
| 653 EXPECT_EQ(rtc::Optional<bool>(false), | |
| 654 video_media_channel_->options().is_screencast); | |
| 655 // Setting fluid should remain in non-screencast mode (its default). | |
| 656 video_track_->set_content_hint(VideoTrackInterface::ContentHint::kFluid); | |
| 657 EXPECT_EQ(rtc::Optional<bool>(false), | |
| 658 video_media_channel_->options().is_screencast); | |
| 659 | |
| 660 DestroyVideoRtpSender(); | |
| 661 } | |
| 662 | |
| 663 // Test that makes sure that a video track content hint translates to the proper | |
| 664 // value for screencast sources. | |
| 665 TEST_F(RtpSenderReceiverTest, | |
| 666 PropagatesVideoTrackContentHintForScreencastSource) { | |
| 667 CreateVideoRtpSender(true); | |
| 668 | |
| 669 video_track_->set_enabled(true); | |
| 670 | |
| 671 // |video_track_| with a screencast source should be screencast by default. | |
| 672 EXPECT_EQ(rtc::Optional<bool>(true), | |
| 673 video_media_channel_->options().is_screencast); | |
| 674 // No content hint should be set by default. | |
| 675 EXPECT_EQ(VideoTrackInterface::ContentHint::kNone, | |
| 676 video_track_->content_hint()); | |
| 677 // Setting fluid should turn a screencast source into non-screencast mode. | |
| 678 video_track_->set_content_hint(VideoTrackInterface::ContentHint::kFluid); | |
| 679 EXPECT_EQ(rtc::Optional<bool>(false), | |
| 680 video_media_channel_->options().is_screencast); | |
| 681 // Removing the content hint should turn the track back into screencast mode. | |
| 682 video_track_->set_content_hint(VideoTrackInterface::ContentHint::kNone); | |
| 683 EXPECT_EQ(rtc::Optional<bool>(true), | |
| 684 video_media_channel_->options().is_screencast); | |
| 685 // Setting detailed should still remain in screencast mode (its default). | |
| 686 video_track_->set_content_hint(VideoTrackInterface::ContentHint::kDetailed); | |
| 687 EXPECT_EQ(rtc::Optional<bool>(true), | |
| 688 video_media_channel_->options().is_screencast); | |
| 689 | |
| 690 DestroyVideoRtpSender(); | |
| 691 } | |
| 692 | |
| 693 // Test that makes sure any content hints that are set on a track before | |
| 694 // VideoRtpSender is ready to send are still applied when it gets ready to send. | |
| 695 TEST_F(RtpSenderReceiverTest, | |
| 696 PropagatesVideoTrackContentHintSetBeforeEnabling) { | |
| 697 AddVideoTrack(); | |
| 698 // Setting detailed overrides the default non-screencast mode. This should be | |
| 699 // applied even if the track is set on construction. | |
| 700 video_track_->set_content_hint(VideoTrackInterface::ContentHint::kDetailed); | |
| 701 video_rtp_sender_ = new VideoRtpSender(stream_->GetVideoTracks()[0], | |
| 702 stream_->label(), video_channel_); | |
| 703 video_track_->set_enabled(true); | |
| 704 | |
| 705 // Sender is not ready to send (no SSRC) so no option should have been set. | |
| 706 EXPECT_EQ(rtc::Optional<bool>(), | |
| 707 video_media_channel_->options().is_screencast); | |
| 708 | |
| 709 // Verify that the content hint is accounted for when video_rtp_sender_ does | |
| 710 // get enabled. | |
| 711 video_rtp_sender_->SetSsrc(kVideoSsrc); | |
| 712 EXPECT_EQ(rtc::Optional<bool>(true), | |
| 713 video_media_channel_->options().is_screencast); | |
| 714 | |
| 715 // And removing the hint should go back to false (to verify that false was | |
| 716 // default correctly). | |
| 717 video_track_->set_content_hint(VideoTrackInterface::ContentHint::kNone); | |
| 718 EXPECT_EQ(rtc::Optional<bool>(false), | |
| 719 video_media_channel_->options().is_screencast); | |
| 720 | |
| 721 DestroyVideoRtpSender(); | |
| 722 } | |
| 723 | |
| 724 } // namespace webrtc | |
| OLD | NEW |
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Chromium Code Reviews
Issue 2514883002:
Create //webrtc/api:libjingle_peerconnection_api + refactorings. (Closed)
