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| 1 /* | 1 /* |
| 2 * Copyright (c) 2004 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2004 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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| 658 // sanity checks against that. | 658 // sanity checks against that. |
| 659 codecs_.push_back(AudioCodec(101, "fake_audio_codec", 0, 0, 1, 0)); | 659 codecs_.push_back(AudioCodec(101, "fake_audio_codec", 0, 0, 1, 0)); |
| 660 } | 660 } |
| 661 bool Init(rtc::Thread* worker_thread) { return true; } | 661 bool Init(rtc::Thread* worker_thread) { return true; } |
| 662 void Terminate() {} | 662 void Terminate() {} |
| 663 rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const { | 663 rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const { |
| 664 return rtc::scoped_refptr<webrtc::AudioState>(); | 664 return rtc::scoped_refptr<webrtc::AudioState>(); |
| 665 } | 665 } |
| 666 | 666 |
| 667 VoiceMediaChannel* CreateChannel(webrtc::Call* call, | 667 VoiceMediaChannel* CreateChannel(webrtc::Call* call, |
| 668 const MediaConfig& config, |
| 668 const AudioOptions& options) { | 669 const AudioOptions& options) { |
| 669 if (fail_create_channel_) { | 670 if (fail_create_channel_) { |
| 670 return nullptr; | 671 return nullptr; |
| 671 } | 672 } |
| 672 | 673 |
| 673 FakeVoiceMediaChannel* ch = new FakeVoiceMediaChannel(this, options); | 674 FakeVoiceMediaChannel* ch = new FakeVoiceMediaChannel(this, options); |
| 674 channels_.push_back(ch); | 675 channels_.push_back(ch); |
| 675 return ch; | 676 return ch; |
| 676 } | 677 } |
| 677 FakeVoiceMediaChannel* GetChannel(size_t index) { | 678 FakeVoiceMediaChannel* GetChannel(size_t index) { |
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| 721 codecs_.push_back(VideoCodec(0, "fake_video_codec", 0, 0, 0, 0)); | 722 codecs_.push_back(VideoCodec(0, "fake_video_codec", 0, 0, 0, 0)); |
| 722 } | 723 } |
| 723 void Init() {} | 724 void Init() {} |
| 724 bool SetOptions(const VideoOptions& options) { | 725 bool SetOptions(const VideoOptions& options) { |
| 725 options_ = options; | 726 options_ = options; |
| 726 options_changed_ = true; | 727 options_changed_ = true; |
| 727 return true; | 728 return true; |
| 728 } | 729 } |
| 729 | 730 |
| 730 VideoMediaChannel* CreateChannel(webrtc::Call* call, | 731 VideoMediaChannel* CreateChannel(webrtc::Call* call, |
| 732 const MediaConfig& config, |
| 731 const VideoOptions& options) { | 733 const VideoOptions& options) { |
| 732 if (fail_create_channel_) { | 734 if (fail_create_channel_) { |
| 733 return NULL; | 735 return NULL; |
| 734 } | 736 } |
| 735 | 737 |
| 736 FakeVideoMediaChannel* ch = new FakeVideoMediaChannel(this, options); | 738 FakeVideoMediaChannel* ch = new FakeVideoMediaChannel(this, options); |
| 737 channels_.push_back(ch); | 739 channels_.push_back(ch); |
| 738 return ch; | 740 return ch; |
| 739 } | 741 } |
| 740 FakeVideoMediaChannel* GetChannel(size_t index) { | 742 FakeVideoMediaChannel* GetChannel(size_t index) { |
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| 866 | 868 |
| 867 private: | 869 private: |
| 868 std::vector<FakeDataMediaChannel*> channels_; | 870 std::vector<FakeDataMediaChannel*> channels_; |
| 869 std::vector<DataCodec> data_codecs_; | 871 std::vector<DataCodec> data_codecs_; |
| 870 DataChannelType last_channel_type_; | 872 DataChannelType last_channel_type_; |
| 871 }; | 873 }; |
| 872 | 874 |
| 873 } // namespace cricket | 875 } // namespace cricket |
| 874 | 876 |
| 875 #endif // WEBRTC_MEDIA_BASE_FAKEMEDIAENGINE_H_ | 877 #endif // WEBRTC_MEDIA_BASE_FAKEMEDIAENGINE_H_ |
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