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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 |
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421 ADD_FAILURE() << "GetNetworkState called with unknown parameter."; | 421 ADD_FAILURE() << "GetNetworkState called with unknown parameter."; |
422 return webrtc::kNetworkDown; | 422 return webrtc::kNetworkDown; |
423 } | 423 } |
424 // Even though all the values for the enum class are listed above,the compiler | 424 // Even though all the values for the enum class are listed above,the compiler |
425 // will emit a warning as the method may be called with a value outside of the | 425 // will emit a warning as the method may be called with a value outside of the |
426 // valid enum range, unless this case is also handled. | 426 // valid enum range, unless this case is also handled. |
427 ADD_FAILURE() << "GetNetworkState called with unknown parameter."; | 427 ADD_FAILURE() << "GetNetworkState called with unknown parameter."; |
428 return webrtc::kNetworkDown; | 428 return webrtc::kNetworkDown; |
429 } | 429 } |
430 | 430 |
| 431 void FakeCall::SetVideoReceiveRtpHeaderExtensions( |
| 432 const std::vector<webrtc::RtpExtension>& extensions) {} |
| 433 |
431 webrtc::AudioSendStream* FakeCall::CreateAudioSendStream( | 434 webrtc::AudioSendStream* FakeCall::CreateAudioSendStream( |
432 const webrtc::AudioSendStream::Config& config) { | 435 const webrtc::AudioSendStream::Config& config) { |
433 FakeAudioSendStream* fake_stream = new FakeAudioSendStream(next_stream_id_++, | 436 FakeAudioSendStream* fake_stream = new FakeAudioSendStream(next_stream_id_++, |
434 config); | 437 config); |
435 audio_send_streams_.push_back(fake_stream); | 438 audio_send_streams_.push_back(fake_stream); |
436 ++num_created_send_streams_; | 439 ++num_created_send_streams_; |
437 return fake_stream; | 440 return fake_stream; |
438 } | 441 } |
439 | 442 |
440 void FakeCall::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) { | 443 void FakeCall::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) { |
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628 } | 631 } |
629 | 632 |
630 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) { | 633 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) { |
631 last_sent_packet_ = sent_packet; | 634 last_sent_packet_ = sent_packet; |
632 if (sent_packet.packet_id >= 0) { | 635 if (sent_packet.packet_id >= 0) { |
633 last_sent_nonnegative_packet_id_ = sent_packet.packet_id; | 636 last_sent_nonnegative_packet_id_ = sent_packet.packet_id; |
634 } | 637 } |
635 } | 638 } |
636 | 639 |
637 } // namespace cricket | 640 } // namespace cricket |
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