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Side by Side Diff: talk/media/base/fakemediaengine.h

Issue 1403363003: Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly i… (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: that's all folks! (incl rebase), or is it? Created 5 years, 1 month ago
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1 /* 1 /*
2 * libjingle 2 * libjingle
3 * Copyright 2004 Google Inc. 3 * Copyright 2004 Google Inc.
4 * 4 *
5 * Redistribution and use in source and binary forms, with or without 5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met: 6 * modification, are permitted provided that the following conditions are met:
7 * 7 *
8 * 1. Redistributions of source code must retain the above copyright notice, 8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer. 9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice, 10 * 2. Redistributions in binary form must reproduce the above copyright notice,
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726 class FakeVoiceEngine : public FakeBaseEngine { 726 class FakeVoiceEngine : public FakeBaseEngine {
727 public: 727 public:
728 FakeVoiceEngine() 728 FakeVoiceEngine()
729 : output_volume_(-1) { 729 : output_volume_(-1) {
730 // Add a fake audio codec. Note that the name must not be "" as there are 730 // Add a fake audio codec. Note that the name must not be "" as there are
731 // sanity checks against that. 731 // sanity checks against that.
732 codecs_.push_back(AudioCodec(101, "fake_audio_codec", 0, 0, 1, 0)); 732 codecs_.push_back(AudioCodec(101, "fake_audio_codec", 0, 0, 1, 0));
733 } 733 }
734 bool Init(rtc::Thread* worker_thread) { return true; } 734 bool Init(rtc::Thread* worker_thread) { return true; }
735 void Terminate() {} 735 void Terminate() {}
736 webrtc::VoiceEngine* GetVoE() { return nullptr; } 736 rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const {
737 return rtc::scoped_refptr<webrtc::AudioState>();
738 }
737 AudioOptions GetOptions() const { 739 AudioOptions GetOptions() const {
738 return options_; 740 return options_;
739 } 741 }
740 bool SetOptions(const AudioOptions& options) { 742 bool SetOptions(const AudioOptions& options) {
741 options_ = options; 743 options_ = options;
742 options_changed_ = true; 744 options_changed_ = true;
743 return true; 745 return true;
744 } 746 }
745 747
746 VoiceMediaChannel* CreateChannel(webrtc::Call* call, 748 VoiceMediaChannel* CreateChannel(webrtc::Call* call,
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985 987
986 private: 988 private:
987 std::vector<FakeDataMediaChannel*> channels_; 989 std::vector<FakeDataMediaChannel*> channels_;
988 std::vector<DataCodec> data_codecs_; 990 std::vector<DataCodec> data_codecs_;
989 DataChannelType last_channel_type_; 991 DataChannelType last_channel_type_;
990 }; 992 };
991 993
992 } // namespace cricket 994 } // namespace cricket
993 995
994 #endif // TALK_MEDIA_BASE_FAKEMEDIAENGINE_H_ 996 #endif // TALK_MEDIA_BASE_FAKEMEDIAENGINE_H_
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