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

Issue 2934103002: Allow WebRtcMediaEngine to be created from any thread. (Closed)
Patch Set: Adding DCHECK for encoder_factory Created 3 years, 6 months ago
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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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