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| 1 /* | 1 /* |
| 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. |
| 3 * | 3 * |
| 4 * Use of this source code is governed by a BSD-style license | 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 | 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 | 6 * tree. An additional intellectual property rights grant can be found |
| 7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
| 8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
| 9 */ | 9 */ |
| 10 | 10 |
| (...skipping 278 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... | |
| 289 void FakeVideoReceiveStream::SetStats( | 289 void FakeVideoReceiveStream::SetStats( |
| 290 const webrtc::VideoReceiveStream::Stats& stats) { | 290 const webrtc::VideoReceiveStream::Stats& stats) { |
| 291 stats_ = stats; | 291 stats_ = stats; |
| 292 } | 292 } |
| 293 | 293 |
| 294 void FakeVideoReceiveStream::EnableEncodedFrameRecording(rtc::PlatformFile file, | 294 void FakeVideoReceiveStream::EnableEncodedFrameRecording(rtc::PlatformFile file, |
| 295 size_t byte_limit) { | 295 size_t byte_limit) { |
| 296 rtc::ClosePlatformFile(file); | 296 rtc::ClosePlatformFile(file); |
| 297 } | 297 } |
| 298 | 298 |
| 299 FakeFlexfecReceiveStream::FakeFlexfecReceiveStream( | |
| 300 const webrtc::FlexfecReceiveStream::Config& config) | |
| 301 : config_(config), receiving_(false) {} | |
| 302 | |
| 303 const webrtc::FlexfecReceiveStream::Config& | |
| 304 FakeFlexfecReceiveStream::GetConfig() { | |
| 305 return config_; | |
| 306 } | |
| 307 | |
| 308 void FakeFlexfecReceiveStream::Start() { | |
| 309 receiving_ = true; | |
| 310 } | |
| 311 | |
| 312 void FakeFlexfecReceiveStream::Stop() { | |
| 313 receiving_ = false; | |
| 314 } | |
| 315 | |
| 316 // TODO(brandtr): Implement when the stats have been designed. | |
| 317 webrtc::FlexfecReceiveStream::Stats FakeFlexfecReceiveStream::GetStats() const { | |
| 318 return webrtc::FlexfecReceiveStream::Stats(); | |
| 319 } | |
| 320 | |
| 299 FakeCall::FakeCall(const webrtc::Call::Config& config) | 321 FakeCall::FakeCall(const webrtc::Call::Config& config) |
| 300 : config_(config), | 322 : config_(config), |
| 301 audio_network_state_(webrtc::kNetworkUp), | 323 audio_network_state_(webrtc::kNetworkUp), |
| 302 video_network_state_(webrtc::kNetworkUp), | 324 video_network_state_(webrtc::kNetworkUp), |
| 303 num_created_send_streams_(0), | 325 num_created_send_streams_(0), |
| 304 num_created_receive_streams_(0) {} | 326 num_created_receive_streams_(0) {} |
| 305 | 327 |
| 306 FakeCall::~FakeCall() { | 328 FakeCall::~FakeCall() { |
| 307 EXPECT_EQ(0u, video_send_streams_.size()); | 329 EXPECT_EQ(0u, video_send_streams_.size()); |
| 308 EXPECT_EQ(0u, audio_send_streams_.size()); | 330 EXPECT_EQ(0u, audio_send_streams_.size()); |
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| 341 | 363 |
| 342 const FakeAudioReceiveStream* FakeCall::GetAudioReceiveStream(uint32_t ssrc) { | 364 const FakeAudioReceiveStream* FakeCall::GetAudioReceiveStream(uint32_t ssrc) { |
| 343 for (const auto* p : GetAudioReceiveStreams()) { | 365 for (const auto* p : GetAudioReceiveStreams()) { |
| 344 if (p->GetConfig().rtp.remote_ssrc == ssrc) { | 366 if (p->GetConfig().rtp.remote_ssrc == ssrc) { |
| 345 return p; | 367 return p; |
| 346 } | 368 } |
| 347 } | 369 } |
| 348 return nullptr; | 370 return nullptr; |
| 349 } | 371 } |
| 350 | 372 |
| 373 const std::vector<FakeFlexfecReceiveStream*>& | |
| 374 FakeCall::GetFlexfecReceiveStreams() { | |
| 375 return flexfec_receive_streams_; | |
| 376 } | |
| 377 | |
| 351 webrtc::NetworkState FakeCall::GetNetworkState(webrtc::MediaType media) const { | 378 webrtc::NetworkState FakeCall::GetNetworkState(webrtc::MediaType media) const { |
| 352 switch (media) { | 379 switch (media) { |
| 353 case webrtc::MediaType::AUDIO: | 380 case webrtc::MediaType::AUDIO: |
| 354 return audio_network_state_; | 381 return audio_network_state_; |
| 355 case webrtc::MediaType::VIDEO: | 382 case webrtc::MediaType::VIDEO: |
| 356 return video_network_state_; | 383 return video_network_state_; |
| 357 case webrtc::MediaType::DATA: | 384 case webrtc::MediaType::DATA: |
| 358 case webrtc::MediaType::ANY: | 385 case webrtc::MediaType::ANY: |
| 359 ADD_FAILURE() << "GetNetworkState called with unknown parameter."; | 386 ADD_FAILURE() << "GetNetworkState called with unknown parameter."; |
| 360 return webrtc::kNetworkDown; | 387 return webrtc::kNetworkDown; |
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| 443 static_cast<FakeVideoReceiveStream*>(receive_stream)); | 470 static_cast<FakeVideoReceiveStream*>(receive_stream)); |
| 444 if (it == video_receive_streams_.end()) { | 471 if (it == video_receive_streams_.end()) { |
| 445 ADD_FAILURE() << "DestroyVideoReceiveStream called with unknown parameter."; | 472 ADD_FAILURE() << "DestroyVideoReceiveStream called with unknown parameter."; |
| 446 } else { | 473 } else { |
| 447 delete *it; | 474 delete *it; |
| 448 video_receive_streams_.erase(it); | 475 video_receive_streams_.erase(it); |
| 449 } | 476 } |
| 450 } | 477 } |
| 451 | 478 |
| 452 webrtc::FlexfecReceiveStream* FakeCall::CreateFlexfecReceiveStream( | 479 webrtc::FlexfecReceiveStream* FakeCall::CreateFlexfecReceiveStream( |
| 453 webrtc::FlexfecReceiveStream::Config config) { | 480 webrtc::FlexfecReceiveStream::Config config) { |
|
magjed_webrtc
2016/11/18 14:28:00
I would like this to be a const-ref as well (and/o
brandtr
2016/11/21 08:52:03
Can't do it here alone, because FakeCall is a webr
| |
| 454 // TODO(brandtr): Implement when adding integration with WebRtcVideoEngine2. | 481 flexfec_receive_streams_.push_back( |
| 455 return nullptr; | 482 new FakeFlexfecReceiveStream(std::move(config))); |
| 483 ++num_created_receive_streams_; | |
| 484 return flexfec_receive_streams_.back(); | |
| 456 } | 485 } |
| 457 | 486 |
| 458 void FakeCall::DestroyFlexfecReceiveStream( | 487 void FakeCall::DestroyFlexfecReceiveStream( |
| 459 webrtc::FlexfecReceiveStream* receive_stream) { | 488 webrtc::FlexfecReceiveStream* receive_stream) { |
| 460 // TODO(brandtr): Implement when adding integration with WebRtcVideoEngine2. | 489 auto it = std::find(flexfec_receive_streams_.begin(), |
| 490 flexfec_receive_streams_.end(), | |
| 491 static_cast<FakeFlexfecReceiveStream*>(receive_stream)); | |
| 492 if (it == flexfec_receive_streams_.end()) { | |
| 493 ADD_FAILURE() | |
| 494 << "DestroyFlexfecReceiveStream called with unknown parameter."; | |
| 495 } else { | |
| 496 delete *it; | |
| 497 flexfec_receive_streams_.erase(it); | |
| 498 } | |
| 461 } | 499 } |
| 462 | 500 |
| 463 webrtc::PacketReceiver* FakeCall::Receiver() { | 501 webrtc::PacketReceiver* FakeCall::Receiver() { |
| 464 return this; | 502 return this; |
| 465 } | 503 } |
| 466 | 504 |
| 467 FakeCall::DeliveryStatus FakeCall::DeliverPacket( | 505 FakeCall::DeliveryStatus FakeCall::DeliverPacket( |
| 468 webrtc::MediaType media_type, | 506 webrtc::MediaType media_type, |
| 469 const uint8_t* packet, | 507 const uint8_t* packet, |
| 470 size_t length, | 508 size_t length, |
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| 547 } | 585 } |
| 548 | 586 |
| 549 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) { | 587 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) { |
| 550 last_sent_packet_ = sent_packet; | 588 last_sent_packet_ = sent_packet; |
| 551 if (sent_packet.packet_id >= 0) { | 589 if (sent_packet.packet_id >= 0) { |
| 552 last_sent_nonnegative_packet_id_ = sent_packet.packet_id; | 590 last_sent_nonnegative_packet_id_ = sent_packet.packet_id; |
| 553 } | 591 } |
| 554 } | 592 } |
| 555 | 593 |
| 556 } // namespace cricket | 594 } // namespace cricket |
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