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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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| 69 ~WebRtcVoiceEngine(); | 69 ~WebRtcVoiceEngine(); |
| 70 bool Init(rtc::Thread* worker_thread); | 70 bool Init(rtc::Thread* worker_thread); |
| 71 void Terminate(); | 71 void Terminate(); |
| 72 | 72 |
| 73 webrtc::VoiceEngine* GetVoE() { return voe()->engine(); } | 73 webrtc::VoiceEngine* GetVoE() { return voe()->engine(); } |
| 74 VoiceMediaChannel* CreateChannel(webrtc::Call* call, | 74 VoiceMediaChannel* CreateChannel(webrtc::Call* call, |
| 75 const AudioOptions& options); | 75 const AudioOptions& options); |
| 76 | 76 |
| 77 AudioOptions GetOptions() const { return options_; } | 77 AudioOptions GetOptions() const { return options_; } |
| 78 bool SetOptions(const AudioOptions& options); | 78 bool SetOptions(const AudioOptions& options); |
| 79 bool SetDelayOffset(int offset); | |
| 80 bool SetDevices(const Device* in_device, const Device* out_device); | 79 bool SetDevices(const Device* in_device, const Device* out_device); |
| 81 bool GetOutputVolume(int* level); | 80 bool GetOutputVolume(int* level); |
| 82 bool SetOutputVolume(int level); | 81 bool SetOutputVolume(int level); |
| 83 int GetInputLevel(); | 82 int GetInputLevel(); |
| 84 | 83 |
| 85 const std::vector<AudioCodec>& codecs(); | 84 const std::vector<AudioCodec>& codecs(); |
| 86 bool FindCodec(const AudioCodec& codec); | 85 bool FindCodec(const AudioCodec& codec); |
| 87 bool FindWebRtcCodec(const AudioCodec& codec, webrtc::CodecInst* gcodec); | 86 bool FindWebRtcCodec(const AudioCodec& codec, webrtc::CodecInst* gcodec); |
| 88 | 87 |
| 89 const std::vector<RtpHeaderExtension>& rtp_header_extensions() const; | 88 const std::vector<RtpHeaderExtension>& rtp_header_extensions() const; |
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| 373 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; | 372 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; |
| 374 | 373 |
| 375 // Do not lock this on the VoE media processor thread; potential for deadlock | 374 // Do not lock this on the VoE media processor thread; potential for deadlock |
| 376 // exists. | 375 // exists. |
| 377 mutable rtc::CriticalSection receive_channels_cs_; | 376 mutable rtc::CriticalSection receive_channels_cs_; |
| 378 }; | 377 }; |
| 379 | 378 |
| 380 } // namespace cricket | 379 } // namespace cricket |
| 381 | 380 |
| 382 #endif // TALK_MEDIA_WEBRTCVOICEENGINE_H_ | 381 #endif // TALK_MEDIA_WEBRTCVOICEENGINE_H_ |
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