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| 1 /* | 1 /* |
| 2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2014 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 |
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| 485 return Call::Config(&event_log_); | 485 return Call::Config(&event_log_); |
| 486 } | 486 } |
| 487 | 487 |
| 488 Call::Config BaseTest::GetReceiverCallConfig() { | 488 Call::Config BaseTest::GetReceiverCallConfig() { |
| 489 return Call::Config(&event_log_); | 489 return Call::Config(&event_log_); |
| 490 } | 490 } |
| 491 | 491 |
| 492 void BaseTest::OnCallsCreated(Call* sender_call, Call* receiver_call) { | 492 void BaseTest::OnCallsCreated(Call* sender_call, Call* receiver_call) { |
| 493 } | 493 } |
| 494 | 494 |
| 495 MediaType BaseTest::SelectMediaType() { |
| 496 if (GetNumVideoStreams() > 0) { |
| 497 if (GetNumAudioStreams() > 0) { |
| 498 // Relies on PayloadDemuxer to set media type from payload type. |
| 499 return MediaType::ANY; |
| 500 } else { |
| 501 return MediaType::VIDEO; |
| 502 } |
| 503 } else { |
| 504 return MediaType::AUDIO; |
| 505 } |
| 506 } |
| 507 |
| 495 test::PacketTransport* BaseTest::CreateSendTransport(Call* sender_call) { | 508 test::PacketTransport* BaseTest::CreateSendTransport(Call* sender_call) { |
| 496 return new PacketTransport(sender_call, this, test::PacketTransport::kSender, | 509 return new PacketTransport(sender_call, this, test::PacketTransport::kSender, |
| 497 MediaType::VIDEO, | 510 SelectMediaType(), |
| 498 FakeNetworkPipe::Config()); | 511 FakeNetworkPipe::Config()); |
| 499 } | 512 } |
| 500 | 513 |
| 501 test::PacketTransport* BaseTest::CreateReceiveTransport() { | 514 test::PacketTransport* BaseTest::CreateReceiveTransport() { |
| 502 return new PacketTransport(nullptr, this, test::PacketTransport::kReceiver, | 515 return new PacketTransport(nullptr, this, test::PacketTransport::kReceiver, |
| 503 MediaType::VIDEO, | 516 SelectMediaType(), |
| 504 FakeNetworkPipe::Config()); | 517 FakeNetworkPipe::Config()); |
| 505 } | 518 } |
| 506 | 519 |
| 507 size_t BaseTest::GetNumVideoStreams() const { | 520 size_t BaseTest::GetNumVideoStreams() const { |
| 508 return 1; | 521 return 1; |
| 509 } | 522 } |
| 510 | 523 |
| 511 size_t BaseTest::GetNumAudioStreams() const { | 524 size_t BaseTest::GetNumAudioStreams() const { |
| 512 return 0; | 525 return 0; |
| 513 } | 526 } |
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| 561 | 574 |
| 562 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) { | 575 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) { |
| 563 } | 576 } |
| 564 | 577 |
| 565 bool EndToEndTest::ShouldCreateReceivers() const { | 578 bool EndToEndTest::ShouldCreateReceivers() const { |
| 566 return true; | 579 return true; |
| 567 } | 580 } |
| 568 | 581 |
| 569 } // namespace test | 582 } // namespace test |
| 570 } // namespace webrtc | 583 } // namespace webrtc |
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