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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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414 ADD_FAILURE() << "GetNetworkState called with unknown parameter."; | 414 ADD_FAILURE() << "GetNetworkState called with unknown parameter."; |
415 return webrtc::kNetworkDown; | 415 return webrtc::kNetworkDown; |
416 } | 416 } |
417 // Even though all the values for the enum class are listed above,the compiler | 417 // Even though all the values for the enum class are listed above,the compiler |
418 // will emit a warning as the method may be called with a value outside of the | 418 // will emit a warning as the method may be called with a value outside of the |
419 // valid enum range, unless this case is also handled. | 419 // valid enum range, unless this case is also handled. |
420 ADD_FAILURE() << "GetNetworkState called with unknown parameter."; | 420 ADD_FAILURE() << "GetNetworkState called with unknown parameter."; |
421 return webrtc::kNetworkDown; | 421 return webrtc::kNetworkDown; |
422 } | 422 } |
423 | 423 |
| 424 void FakeCall::SetVideoReceiveRtpHeaderExtensions( |
| 425 const std::vector<webrtc::RtpExtension>& extensions) {} |
| 426 |
424 webrtc::AudioSendStream* FakeCall::CreateAudioSendStream( | 427 webrtc::AudioSendStream* FakeCall::CreateAudioSendStream( |
425 const webrtc::AudioSendStream::Config& config) { | 428 const webrtc::AudioSendStream::Config& config) { |
426 FakeAudioSendStream* fake_stream = new FakeAudioSendStream(next_stream_id_++, | 429 FakeAudioSendStream* fake_stream = new FakeAudioSendStream(next_stream_id_++, |
427 config); | 430 config); |
428 audio_send_streams_.push_back(fake_stream); | 431 audio_send_streams_.push_back(fake_stream); |
429 ++num_created_send_streams_; | 432 ++num_created_send_streams_; |
430 return fake_stream; | 433 return fake_stream; |
431 } | 434 } |
432 | 435 |
433 void FakeCall::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) { | 436 void FakeCall::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) { |
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616 } | 619 } |
617 | 620 |
618 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) { | 621 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) { |
619 last_sent_packet_ = sent_packet; | 622 last_sent_packet_ = sent_packet; |
620 if (sent_packet.packet_id >= 0) { | 623 if (sent_packet.packet_id >= 0) { |
621 last_sent_nonnegative_packet_id_ = sent_packet.packet_id; | 624 last_sent_nonnegative_packet_id_ = sent_packet.packet_id; |
622 } | 625 } |
623 } | 626 } |
624 | 627 |
625 } // namespace cricket | 628 } // namespace cricket |
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