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Side by Side Diff: talk/media/webrtc/webrtcvoiceengine.h

Issue 1364093002: Remove SetAudioDelayOffset() and friends. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@lj_remove_getcapabilities
Patch Set: rebase Created 5 years, 3 months 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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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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