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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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768 FakeVoiceEngine(webrtc::AudioDeviceModule* adm, | 768 FakeVoiceEngine(webrtc::AudioDeviceModule* adm, |
769 const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& | 769 const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& |
770 audio_encoder_factory, | 770 audio_encoder_factory, |
771 const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& | 771 const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& |
772 audio_decoder_factory, | 772 audio_decoder_factory, |
773 rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer) { | 773 rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer) { |
774 // Add a fake audio codec. Note that the name must not be "" as there are | 774 // Add a fake audio codec. Note that the name must not be "" as there are |
775 // sanity checks against that. | 775 // sanity checks against that. |
776 codecs_.push_back(AudioCodec(101, "fake_audio_codec", 0, 0, 1)); | 776 codecs_.push_back(AudioCodec(101, "fake_audio_codec", 0, 0, 1)); |
777 } | 777 } |
| 778 void Init() {} |
778 rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const { | 779 rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const { |
779 return rtc::scoped_refptr<webrtc::AudioState>(); | 780 return rtc::scoped_refptr<webrtc::AudioState>(); |
780 } | 781 } |
781 | 782 |
782 VoiceMediaChannel* CreateChannel(webrtc::Call* call, | 783 VoiceMediaChannel* CreateChannel(webrtc::Call* call, |
783 const MediaConfig& config, | 784 const MediaConfig& config, |
784 const AudioOptions& options) { | 785 const AudioOptions& options) { |
785 if (fail_create_channel_) { | 786 if (fail_create_channel_) { |
786 return nullptr; | 787 return nullptr; |
787 } | 788 } |
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966 virtual const std::vector<DataCodec>& data_codecs() { return data_codecs_; } | 967 virtual const std::vector<DataCodec>& data_codecs() { return data_codecs_; } |
967 | 968 |
968 private: | 969 private: |
969 std::vector<FakeDataMediaChannel*> channels_; | 970 std::vector<FakeDataMediaChannel*> channels_; |
970 std::vector<DataCodec> data_codecs_; | 971 std::vector<DataCodec> data_codecs_; |
971 }; | 972 }; |
972 | 973 |
973 } // namespace cricket | 974 } // namespace cricket |
974 | 975 |
975 #endif // WEBRTC_MEDIA_BASE_FAKEMEDIAENGINE_H_ | 976 #endif // WEBRTC_MEDIA_BASE_FAKEMEDIAENGINE_H_ |
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