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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_enabled = false; | |
| 66 bool srtp_required = true; | |
| 67 voice_channel_ = channel_manager_.CreateVoiceChannel( | |
| 68 &fake_media_controller_, &fake_transport_controller_, cricket::CN_AUDIO, | |
| 69 nullptr, rtcp_enabled, srtp_required, cricket::AudioOptions()); | |
| 70 video_channel_ = channel_manager_.CreateVideoChannel( | |
| 71 &fake_media_controller_, &fake_transport_controller_, cricket::CN_VIDEO, | |
| 72 nullptr, rtcp_enabled, srtp_required, cricket::VideoOptions()); | |
| 73 voice_media_channel_ = media_engine_->GetVoiceChannel(0); | |
| 74 video_media_channel_ = media_engine_->GetVideoChannel(0); | |
| 75 RTC_CHECK(voice_channel_); | |
| 76 RTC_CHECK(video_channel_); | |
| 77 RTC_CHECK(voice_media_channel_); | |
| 78 RTC_CHECK(video_media_channel_); | |
| 79 | |
| 80 // Create streams for predefined SSRCs. Streams need to exist in order | |
| 81 // for the senders and receievers to apply parameters to them. | |
| 82 // Normally these would be created by SetLocalDescription and | |
| 83 // SetRemoteDescription. | |
| 84 voice_media_channel_->AddSendStream( | |
| 85 cricket::StreamParams::CreateLegacy(kAudioSsrc)); | |
| 86 voice_media_channel_->AddRecvStream( | |
| 87 cricket::StreamParams::CreateLegacy(kAudioSsrc)); | |
| 88 voice_media_channel_->AddSendStream( | |
| 89 cricket::StreamParams::CreateLegacy(kAudioSsrc2)); | |
| 90 voice_media_channel_->AddRecvStream( | |
| 91 cricket::StreamParams::CreateLegacy(kAudioSsrc2)); | |
| 92 video_media_channel_->AddSendStream( | |
| 93 cricket::StreamParams::CreateLegacy(kVideoSsrc)); | |
| 94 video_media_channel_->AddRecvStream( | |
| 95 cricket::StreamParams::CreateLegacy(kVideoSsrc)); | |
| 96 video_media_channel_->AddSendStream( | |
| 97 cricket::StreamParams::CreateLegacy(kVideoSsrc2)); | |
| 98 video_media_channel_->AddRecvStream( | |
| 99 cricket::StreamParams::CreateLegacy(kVideoSsrc2)); | |
| 100 } | |
| 101 | |
| 102 void TearDown() override { channel_manager_.Terminate(); } | |
| 103 | |
| 104 void AddVideoTrack() { | |
| 105 rtc::scoped_refptr<VideoTrackSourceInterface> source( | |
| 106 FakeVideoTrackSource::Create()); | |
| 107 video_track_ = VideoTrack::Create(kVideoTrackId, source); | |
| 108 EXPECT_TRUE(stream_->AddTrack(video_track_)); | |
| 109 } | |
| 110 | |
| 111 void CreateAudioRtpSender() { CreateAudioRtpSender(nullptr); } | |
| 112 | |
| 113 void CreateAudioRtpSender(rtc::scoped_refptr<LocalAudioSource> source) { | |
| 114 audio_track_ = AudioTrack::Create(kAudioTrackId, source); | |
| 115 EXPECT_TRUE(stream_->AddTrack(audio_track_)); | |
| 116 audio_rtp_sender_ = | |
| 117 new AudioRtpSender(stream_->GetAudioTracks()[0], stream_->label(), | |
| 118 voice_channel_, nullptr); | |
| 119 audio_rtp_sender_->SetSsrc(kAudioSsrc); | |
| 120 VerifyVoiceChannelInput(); | |
| 121 } | |
| 122 | |
| 123 void CreateVideoRtpSender() { | |
| 124 AddVideoTrack(); | |
| 125 video_rtp_sender_ = new VideoRtpSender(stream_->GetVideoTracks()[0], | |
| 126 stream_->label(), video_channel_); | |
| 127 video_rtp_sender_->SetSsrc(kVideoSsrc); | |
| 128 VerifyVideoChannelInput(); | |
| 129 } | |
| 130 | |
| 131 void DestroyAudioRtpSender() { | |
| 132 audio_rtp_sender_ = nullptr; | |
| 133 VerifyVoiceChannelNoInput(); | |
| 134 } | |
| 135 | |
| 136 void DestroyVideoRtpSender() { | |
| 137 video_rtp_sender_ = nullptr; | |
| 138 VerifyVideoChannelNoInput(); | |
| 139 } | |
| 140 | |
| 141 void CreateAudioRtpReceiver() { | |
| 142 audio_track_ = AudioTrack::Create( | |
| 143 kAudioTrackId, RemoteAudioSource::Create(kAudioSsrc, NULL)); | |
| 144 EXPECT_TRUE(stream_->AddTrack(audio_track_)); | |
| 145 audio_rtp_receiver_ = new AudioRtpReceiver(stream_, kAudioTrackId, | |
| 146 kAudioSsrc, voice_channel_); | |
| 147 audio_track_ = audio_rtp_receiver_->audio_track(); | |
| 148 VerifyVoiceChannelOutput(); | |
| 149 } | |
| 150 | |
| 151 void CreateVideoRtpReceiver() { | |
| 152 video_rtp_receiver_ = | |
| 153 new VideoRtpReceiver(stream_, kVideoTrackId, rtc::Thread::Current(), | |
| 154 kVideoSsrc, video_channel_); | |
| 155 video_track_ = video_rtp_receiver_->video_track(); | |
| 156 VerifyVideoChannelOutput(); | |
| 157 } | |
| 158 | |
| 159 void DestroyAudioRtpReceiver() { | |
| 160 audio_rtp_receiver_ = nullptr; | |
| 161 VerifyVoiceChannelNoOutput(); | |
| 162 } | |
| 163 | |
| 164 void DestroyVideoRtpReceiver() { | |
| 165 video_rtp_receiver_ = nullptr; | |
| 166 VerifyVideoChannelNoOutput(); | |
| 167 } | |
| 168 | |
| 169 void VerifyVoiceChannelInput() { VerifyVoiceChannelInput(kAudioSsrc); } | |
| 170 | |
| 171 void VerifyVoiceChannelInput(uint32_t ssrc) { | |
| 172 // Verify that the media channel has an audio source, and the stream isn't | |
| 173 // muted. | |
| 174 EXPECT_TRUE(voice_media_channel_->HasSource(ssrc)); | |
| 175 EXPECT_FALSE(voice_media_channel_->IsStreamMuted(ssrc)); | |
| 176 } | |
| 177 | |
| 178 void VerifyVideoChannelInput() { VerifyVideoChannelInput(kVideoSsrc); } | |
| 179 | |
| 180 void VerifyVideoChannelInput(uint32_t ssrc) { | |
| 181 // Verify that the media channel has a video source, | |
| 182 EXPECT_TRUE(video_media_channel_->HasSource(ssrc)); | |
| 183 } | |
| 184 | |
| 185 void VerifyVoiceChannelNoInput() { VerifyVoiceChannelNoInput(kAudioSsrc); } | |
| 186 | |
| 187 void VerifyVoiceChannelNoInput(uint32_t ssrc) { | |
| 188 // Verify that the media channel's source is reset. | |
| 189 EXPECT_FALSE(voice_media_channel_->HasSource(ssrc)); | |
| 190 } | |
| 191 | |
| 192 void VerifyVideoChannelNoInput() { VerifyVideoChannelNoInput(kVideoSsrc); } | |
| 193 | |
| 194 void VerifyVideoChannelNoInput(uint32_t ssrc) { | |
| 195 // Verify that the media channel's source is reset. | |
| 196 EXPECT_FALSE(video_media_channel_->HasSource(ssrc)); | |
| 197 } | |
| 198 | |
| 199 void VerifyVoiceChannelOutput() { | |
| 200 // Verify that the volume is initialized to 1. | |
| 201 double volume; | |
| 202 EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume)); | |
| 203 EXPECT_EQ(1, volume); | |
| 204 } | |
| 205 | |
| 206 void VerifyVideoChannelOutput() { | |
| 207 // Verify that the media channel has a sink. | |
| 208 EXPECT_TRUE(video_media_channel_->HasSink(kVideoSsrc)); | |
| 209 } | |
| 210 | |
| 211 void VerifyVoiceChannelNoOutput() { | |
| 212 // Verify that the volume is reset to 0. | |
| 213 double volume; | |
| 214 EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume)); | |
| 215 EXPECT_EQ(0, volume); | |
| 216 } | |
| 217 | |
| 218 void VerifyVideoChannelNoOutput() { | |
| 219 // Verify that the media channel's sink is reset. | |
| 220 EXPECT_FALSE(video_media_channel_->HasSink(kVideoSsrc)); | |
| 221 } | |
| 222 | |
| 223 protected: | |
| 224 webrtc::RtcEventLogNullImpl event_log_; | |
| 225 cricket::FakeMediaEngine* media_engine_; | |
| 226 cricket::FakeTransportController fake_transport_controller_; | |
| 227 cricket::ChannelManager channel_manager_; | |
| 228 cricket::FakeCall fake_call_; | |
| 229 cricket::FakeMediaController fake_media_controller_; | |
| 230 cricket::VoiceChannel* voice_channel_; | |
| 231 cricket::VideoChannel* video_channel_; | |
| 232 cricket::FakeVoiceMediaChannel* voice_media_channel_; | |
| 233 cricket::FakeVideoMediaChannel* video_media_channel_; | |
| 234 rtc::scoped_refptr<AudioRtpSender> audio_rtp_sender_; | |
| 235 rtc::scoped_refptr<VideoRtpSender> video_rtp_sender_; | |
| 236 rtc::scoped_refptr<AudioRtpReceiver> audio_rtp_receiver_; | |
| 237 rtc::scoped_refptr<VideoRtpReceiver> video_rtp_receiver_; | |
| 238 rtc::scoped_refptr<MediaStreamInterface> stream_; | |
| 239 rtc::scoped_refptr<VideoTrackInterface> video_track_; | |
| 240 rtc::scoped_refptr<AudioTrackInterface> audio_track_; | |
| 241 }; | |
| 242 | |
| 243 // Test that |voice_channel_| is updated when an audio track is associated | |
| 244 // and disassociated with an AudioRtpSender. | |
| 245 TEST_F(RtpSenderReceiverTest, AddAndDestroyAudioRtpSender) { | |
| 246 CreateAudioRtpSender(); | |
| 247 DestroyAudioRtpSender(); | |
| 248 } | |
| 249 | |
| 250 // Test that |video_channel_| is updated when a video track is associated and | |
| 251 // disassociated with a VideoRtpSender. | |
| 252 TEST_F(RtpSenderReceiverTest, AddAndDestroyVideoRtpSender) { | |
| 253 CreateVideoRtpSender(); | |
| 254 DestroyVideoRtpSender(); | |
| 255 } | |
| 256 | |
| 257 // Test that |voice_channel_| is updated when a remote audio track is | |
| 258 // associated and disassociated with an AudioRtpReceiver. | |
| 259 TEST_F(RtpSenderReceiverTest, AddAndDestroyAudioRtpReceiver) { | |
| 260 CreateAudioRtpReceiver(); | |
| 261 DestroyAudioRtpReceiver(); | |
| 262 } | |
| 263 | |
| 264 // Test that |video_channel_| is updated when a remote video track is | |
| 265 // associated and disassociated with a VideoRtpReceiver. | |
| 266 TEST_F(RtpSenderReceiverTest, AddAndDestroyVideoRtpReceiver) { | |
| 267 CreateVideoRtpReceiver(); | |
| 268 DestroyVideoRtpReceiver(); | |
| 269 } | |
| 270 | |
| 271 // Test that the AudioRtpSender applies options from the local audio source. | |
| 272 TEST_F(RtpSenderReceiverTest, LocalAudioSourceOptionsApplied) { | |
| 273 cricket::AudioOptions options; | |
| 274 options.echo_cancellation = rtc::Optional<bool>(true); | |
| 275 auto source = LocalAudioSource::Create( | |
| 276 PeerConnectionFactoryInterface::Options(), &options); | |
| 277 CreateAudioRtpSender(source.get()); | |
| 278 | |
| 279 EXPECT_EQ(rtc::Optional<bool>(true), | |
| 280 voice_media_channel_->options().echo_cancellation); | |
| 281 | |
| 282 DestroyAudioRtpSender(); | |
| 283 } | |
| 284 | |
| 285 // Test that the stream is muted when the track is disabled, and unmuted when | |
| 286 // the track is enabled. | |
| 287 TEST_F(RtpSenderReceiverTest, LocalAudioTrackDisable) { | |
| 288 CreateAudioRtpSender(); | |
| 289 | |
| 290 audio_track_->set_enabled(false); | |
| 291 EXPECT_TRUE(voice_media_channel_->IsStreamMuted(kAudioSsrc)); | |
| 292 | |
| 293 audio_track_->set_enabled(true); | |
| 294 EXPECT_FALSE(voice_media_channel_->IsStreamMuted(kAudioSsrc)); | |
| 295 | |
| 296 DestroyAudioRtpSender(); | |
| 297 } | |
| 298 | |
| 299 // Test that the volume is set to 0 when the track is disabled, and back to | |
| 300 // 1 when the track is enabled. | |
| 301 TEST_F(RtpSenderReceiverTest, RemoteAudioTrackDisable) { | |
| 302 CreateAudioRtpReceiver(); | |
| 303 | |
| 304 double volume; | |
| 305 EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume)); | |
| 306 EXPECT_EQ(1, volume); | |
| 307 | |
| 308 audio_track_->set_enabled(false); | |
| 309 EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume)); | |
| 310 EXPECT_EQ(0, volume); | |
| 311 | |
| 312 audio_track_->set_enabled(true); | |
| 313 EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume)); | |
| 314 EXPECT_EQ(1, volume); | |
| 315 | |
| 316 DestroyAudioRtpReceiver(); | |
| 317 } | |
| 318 | |
| 319 // Currently no action is taken when a remote video track is disabled or | |
| 320 // enabled, so there's nothing to test here, other than what is normally | |
| 321 // verified in DestroyVideoRtpSender. | |
| 322 TEST_F(RtpSenderReceiverTest, LocalVideoTrackDisable) { | |
| 323 CreateVideoRtpSender(); | |
| 324 | |
| 325 video_track_->set_enabled(false); | |
| 326 video_track_->set_enabled(true); | |
| 327 | |
| 328 DestroyVideoRtpSender(); | |
| 329 } | |
| 330 | |
| 331 // Test that the state of the video track created by the VideoRtpReceiver is | |
| 332 // updated when the receiver is destroyed. | |
| 333 TEST_F(RtpSenderReceiverTest, RemoteVideoTrackState) { | |
| 334 CreateVideoRtpReceiver(); | |
| 335 | |
| 336 EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, video_track_->state()); | |
| 337 EXPECT_EQ(webrtc::MediaSourceInterface::kLive, | |
| 338 video_track_->GetSource()->state()); | |
| 339 | |
| 340 DestroyVideoRtpReceiver(); | |
| 341 | |
| 342 EXPECT_EQ(webrtc::MediaStreamTrackInterface::kEnded, video_track_->state()); | |
| 343 EXPECT_EQ(webrtc::MediaSourceInterface::kEnded, | |
| 344 video_track_->GetSource()->state()); | |
| 345 } | |
| 346 | |
| 347 // Currently no action is taken when a remote video track is disabled or | |
| 348 // enabled, so there's nothing to test here, other than what is normally | |
| 349 // verified in DestroyVideoRtpReceiver. | |
| 350 TEST_F(RtpSenderReceiverTest, RemoteVideoTrackDisable) { | |
| 351 CreateVideoRtpReceiver(); | |
| 352 | |
| 353 video_track_->set_enabled(false); | |
| 354 video_track_->set_enabled(true); | |
| 355 | |
| 356 DestroyVideoRtpReceiver(); | |
| 357 } | |
| 358 | |
| 359 // Test that the AudioRtpReceiver applies volume changes from the track source | |
| 360 // to the media channel. | |
| 361 TEST_F(RtpSenderReceiverTest, RemoteAudioTrackSetVolume) { | |
| 362 CreateAudioRtpReceiver(); | |
| 363 | |
| 364 double volume; | |
| 365 audio_track_->GetSource()->SetVolume(0.5); | |
| 366 EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume)); | |
| 367 EXPECT_EQ(0.5, volume); | |
| 368 | |
| 369 // Disable the audio track, this should prevent setting the volume. | |
| 370 audio_track_->set_enabled(false); | |
| 371 audio_track_->GetSource()->SetVolume(0.8); | |
| 372 EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume)); | |
| 373 EXPECT_EQ(0, volume); | |
| 374 | |
| 375 // When the track is enabled, the previously set volume should take effect. | |
| 376 audio_track_->set_enabled(true); | |
| 377 EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume)); | |
| 378 EXPECT_EQ(0.8, volume); | |
| 379 | |
| 380 // Try changing volume one more time. | |
| 381 audio_track_->GetSource()->SetVolume(0.9); | |
| 382 EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume)); | |
| 383 EXPECT_EQ(0.9, volume); | |
| 384 | |
| 385 DestroyAudioRtpReceiver(); | |
| 386 } | |
| 387 | |
| 388 // Test that the media channel isn't enabled for sending if the audio sender | |
| 389 // doesn't have both a track and SSRC. | |
| 390 TEST_F(RtpSenderReceiverTest, AudioSenderWithoutTrackAndSsrc) { | |
| 391 audio_rtp_sender_ = new AudioRtpSender(voice_channel_, nullptr); | |
| 392 rtc::scoped_refptr<AudioTrackInterface> track = | |
| 393 AudioTrack::Create(kAudioTrackId, nullptr); | |
| 394 | |
| 395 // Track but no SSRC. | |
| 396 EXPECT_TRUE(audio_rtp_sender_->SetTrack(track)); | |
| 397 VerifyVoiceChannelNoInput(); | |
| 398 | |
| 399 // SSRC but no track. | |
| 400 EXPECT_TRUE(audio_rtp_sender_->SetTrack(nullptr)); | |
| 401 audio_rtp_sender_->SetSsrc(kAudioSsrc); | |
| 402 VerifyVoiceChannelNoInput(); | |
| 403 } | |
| 404 | |
| 405 // Test that the media channel isn't enabled for sending if the video sender | |
| 406 // doesn't have both a track and SSRC. | |
| 407 TEST_F(RtpSenderReceiverTest, VideoSenderWithoutTrackAndSsrc) { | |
| 408 video_rtp_sender_ = new VideoRtpSender(video_channel_); | |
| 409 | |
| 410 // Track but no SSRC. | |
| 411 EXPECT_TRUE(video_rtp_sender_->SetTrack(video_track_)); | |
| 412 VerifyVideoChannelNoInput(); | |
| 413 | |
| 414 // SSRC but no track. | |
| 415 EXPECT_TRUE(video_rtp_sender_->SetTrack(nullptr)); | |
| 416 video_rtp_sender_->SetSsrc(kVideoSsrc); | |
| 417 VerifyVideoChannelNoInput(); | |
| 418 } | |
| 419 | |
| 420 // Test that the media channel is enabled for sending when the audio sender | |
| 421 // has a track and SSRC, when the SSRC is set first. | |
| 422 TEST_F(RtpSenderReceiverTest, AudioSenderEarlyWarmupSsrcThenTrack) { | |
| 423 audio_rtp_sender_ = new AudioRtpSender(voice_channel_, nullptr); | |
| 424 rtc::scoped_refptr<AudioTrackInterface> track = | |
| 425 AudioTrack::Create(kAudioTrackId, nullptr); | |
| 426 audio_rtp_sender_->SetSsrc(kAudioSsrc); | |
| 427 audio_rtp_sender_->SetTrack(track); | |
| 428 VerifyVoiceChannelInput(); | |
| 429 | |
| 430 DestroyAudioRtpSender(); | |
| 431 } | |
| 432 | |
| 433 // Test that the media channel is enabled for sending when the audio sender | |
| 434 // has a track and SSRC, when the SSRC is set last. | |
| 435 TEST_F(RtpSenderReceiverTest, AudioSenderEarlyWarmupTrackThenSsrc) { | |
| 436 audio_rtp_sender_ = new AudioRtpSender(voice_channel_, nullptr); | |
| 437 rtc::scoped_refptr<AudioTrackInterface> track = | |
| 438 AudioTrack::Create(kAudioTrackId, nullptr); | |
| 439 audio_rtp_sender_->SetTrack(track); | |
| 440 audio_rtp_sender_->SetSsrc(kAudioSsrc); | |
| 441 VerifyVoiceChannelInput(); | |
| 442 | |
| 443 DestroyAudioRtpSender(); | |
| 444 } | |
| 445 | |
| 446 // Test that the media channel is enabled for sending when the video sender | |
| 447 // has a track and SSRC, when the SSRC is set first. | |
| 448 TEST_F(RtpSenderReceiverTest, VideoSenderEarlyWarmupSsrcThenTrack) { | |
| 449 AddVideoTrack(); | |
| 450 video_rtp_sender_ = new VideoRtpSender(video_channel_); | |
| 451 video_rtp_sender_->SetSsrc(kVideoSsrc); | |
| 452 video_rtp_sender_->SetTrack(video_track_); | |
| 453 VerifyVideoChannelInput(); | |
| 454 | |
| 455 DestroyVideoRtpSender(); | |
| 456 } | |
| 457 | |
| 458 // Test that the media channel is enabled for sending when the video sender | |
| 459 // has a track and SSRC, when the SSRC is set last. | |
| 460 TEST_F(RtpSenderReceiverTest, VideoSenderEarlyWarmupTrackThenSsrc) { | |
| 461 AddVideoTrack(); | |
| 462 video_rtp_sender_ = new VideoRtpSender(video_channel_); | |
| 463 video_rtp_sender_->SetTrack(video_track_); | |
| 464 video_rtp_sender_->SetSsrc(kVideoSsrc); | |
| 465 VerifyVideoChannelInput(); | |
| 466 | |
| 467 DestroyVideoRtpSender(); | |
| 468 } | |
| 469 | |
| 470 // Test that the media channel stops sending when the audio sender's SSRC is set | |
| 471 // to 0. | |
| 472 TEST_F(RtpSenderReceiverTest, AudioSenderSsrcSetToZero) { | |
| 473 CreateAudioRtpSender(); | |
| 474 | |
| 475 audio_rtp_sender_->SetSsrc(0); | |
| 476 VerifyVoiceChannelNoInput(); | |
| 477 } | |
| 478 | |
| 479 // Test that the media channel stops sending when the video sender's SSRC is set | |
| 480 // to 0. | |
| 481 TEST_F(RtpSenderReceiverTest, VideoSenderSsrcSetToZero) { | |
| 482 CreateAudioRtpSender(); | |
| 483 | |
| 484 audio_rtp_sender_->SetSsrc(0); | |
| 485 VerifyVideoChannelNoInput(); | |
| 486 } | |
| 487 | |
| 488 // Test that the media channel stops sending when the audio sender's track is | |
| 489 // set to null. | |
| 490 TEST_F(RtpSenderReceiverTest, AudioSenderTrackSetToNull) { | |
| 491 CreateAudioRtpSender(); | |
| 492 | |
| 493 EXPECT_TRUE(audio_rtp_sender_->SetTrack(nullptr)); | |
| 494 VerifyVoiceChannelNoInput(); | |
| 495 } | |
| 496 | |
| 497 // Test that the media channel stops sending when the video sender's track is | |
| 498 // set to null. | |
| 499 TEST_F(RtpSenderReceiverTest, VideoSenderTrackSetToNull) { | |
| 500 CreateVideoRtpSender(); | |
| 501 | |
| 502 video_rtp_sender_->SetSsrc(0); | |
| 503 VerifyVideoChannelNoInput(); | |
| 504 } | |
| 505 | |
| 506 // Test that when the audio sender's SSRC is changed, the media channel stops | |
| 507 // sending with the old SSRC and starts sending with the new one. | |
| 508 TEST_F(RtpSenderReceiverTest, AudioSenderSsrcChanged) { | |
| 509 CreateAudioRtpSender(); | |
| 510 | |
| 511 audio_rtp_sender_->SetSsrc(kAudioSsrc2); | |
| 512 VerifyVoiceChannelNoInput(kAudioSsrc); | |
| 513 VerifyVoiceChannelInput(kAudioSsrc2); | |
| 514 | |
| 515 audio_rtp_sender_ = nullptr; | |
| 516 VerifyVoiceChannelNoInput(kAudioSsrc2); | |
| 517 } | |
| 518 | |
| 519 // Test that when the audio sender's SSRC is changed, the media channel stops | |
| 520 // sending with the old SSRC and starts sending with the new one. | |
| 521 TEST_F(RtpSenderReceiverTest, VideoSenderSsrcChanged) { | |
| 522 CreateVideoRtpSender(); | |
| 523 | |
| 524 video_rtp_sender_->SetSsrc(kVideoSsrc2); | |
| 525 VerifyVideoChannelNoInput(kVideoSsrc); | |
| 526 VerifyVideoChannelInput(kVideoSsrc2); | |
| 527 | |
| 528 video_rtp_sender_ = nullptr; | |
| 529 VerifyVideoChannelNoInput(kVideoSsrc2); | |
| 530 } | |
| 531 | |
| 532 TEST_F(RtpSenderReceiverTest, AudioSenderCanSetParameters) { | |
| 533 CreateAudioRtpSender(); | |
| 534 | |
| 535 RtpParameters params = audio_rtp_sender_->GetParameters(); | |
| 536 EXPECT_EQ(1u, params.encodings.size()); | |
| 537 EXPECT_TRUE(audio_rtp_sender_->SetParameters(params)); | |
| 538 | |
| 539 DestroyAudioRtpSender(); | |
| 540 } | |
| 541 | |
| 542 TEST_F(RtpSenderReceiverTest, SetAudioMaxSendBitrate) { | |
| 543 CreateAudioRtpSender(); | |
| 544 | |
| 545 EXPECT_EQ(-1, voice_media_channel_->max_bps()); | |
| 546 webrtc::RtpParameters params = audio_rtp_sender_->GetParameters(); | |
| 547 EXPECT_EQ(1, params.encodings.size()); | |
| 548 EXPECT_EQ(-1, params.encodings[0].max_bitrate_bps); | |
| 549 params.encodings[0].max_bitrate_bps = 1000; | |
| 550 EXPECT_TRUE(audio_rtp_sender_->SetParameters(params)); | |
| 551 | |
| 552 // Read back the parameters and verify they have been changed. | |
| 553 params = audio_rtp_sender_->GetParameters(); | |
| 554 EXPECT_EQ(1, params.encodings.size()); | |
| 555 EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps); | |
| 556 | |
| 557 // Verify that the audio channel received the new parameters. | |
| 558 params = voice_media_channel_->GetRtpSendParameters(kAudioSsrc); | |
| 559 EXPECT_EQ(1, params.encodings.size()); | |
| 560 EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps); | |
| 561 | |
| 562 // Verify that the global bitrate limit has not been changed. | |
| 563 EXPECT_EQ(-1, voice_media_channel_->max_bps()); | |
| 564 | |
| 565 DestroyAudioRtpSender(); | |
| 566 } | |
| 567 | |
| 568 TEST_F(RtpSenderReceiverTest, VideoSenderCanSetParameters) { | |
| 569 CreateVideoRtpSender(); | |
| 570 | |
| 571 RtpParameters params = video_rtp_sender_->GetParameters(); | |
| 572 EXPECT_EQ(1u, params.encodings.size()); | |
| 573 EXPECT_TRUE(video_rtp_sender_->SetParameters(params)); | |
| 574 | |
| 575 DestroyVideoRtpSender(); | |
| 576 } | |
| 577 | |
| 578 TEST_F(RtpSenderReceiverTest, SetVideoMaxSendBitrate) { | |
| 579 CreateVideoRtpSender(); | |
| 580 | |
| 581 EXPECT_EQ(-1, video_media_channel_->max_bps()); | |
| 582 webrtc::RtpParameters params = video_rtp_sender_->GetParameters(); | |
| 583 EXPECT_EQ(1, params.encodings.size()); | |
| 584 EXPECT_EQ(-1, params.encodings[0].max_bitrate_bps); | |
| 585 params.encodings[0].max_bitrate_bps = 1000; | |
| 586 EXPECT_TRUE(video_rtp_sender_->SetParameters(params)); | |
| 587 | |
| 588 // Read back the parameters and verify they have been changed. | |
| 589 params = video_rtp_sender_->GetParameters(); | |
| 590 EXPECT_EQ(1, params.encodings.size()); | |
| 591 EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps); | |
| 592 | |
| 593 // Verify that the video channel received the new parameters. | |
| 594 params = video_media_channel_->GetRtpSendParameters(kVideoSsrc); | |
| 595 EXPECT_EQ(1, params.encodings.size()); | |
| 596 EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps); | |
| 597 | |
| 598 // Verify that the global bitrate limit has not been changed. | |
| 599 EXPECT_EQ(-1, video_media_channel_->max_bps()); | |
| 600 | |
| 601 DestroyVideoRtpSender(); | |
| 602 } | |
| 603 | |
| 604 TEST_F(RtpSenderReceiverTest, AudioReceiverCanSetParameters) { | |
| 605 CreateAudioRtpReceiver(); | |
| 606 | |
| 607 RtpParameters params = audio_rtp_receiver_->GetParameters(); | |
| 608 EXPECT_EQ(1u, params.encodings.size()); | |
| 609 EXPECT_TRUE(audio_rtp_receiver_->SetParameters(params)); | |
| 610 | |
| 611 DestroyAudioRtpReceiver(); | |
| 612 } | |
| 613 | |
| 614 TEST_F(RtpSenderReceiverTest, VideoReceiverCanSetParameters) { | |
| 615 CreateVideoRtpReceiver(); | |
| 616 | |
| 617 RtpParameters params = video_rtp_receiver_->GetParameters(); | |
| 618 EXPECT_EQ(1u, params.encodings.size()); | |
| 619 EXPECT_TRUE(video_rtp_receiver_->SetParameters(params)); | |
| 620 | |
| 621 DestroyVideoRtpReceiver(); | |
| 622 } | |
| 623 | |
| 624 } // namespace webrtc | |
| OLD | NEW |
|---|
Chromium Code Reviews
Issue 2514883002:
Create //webrtc/api:libjingle_peerconnection_api + refactorings. (Closed)
