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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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237 void OnError(int error); | 237 void OnError(int error); |
238 | 238 |
239 int GetReceiveChannelId(uint32 ssrc) const; | 239 int GetReceiveChannelId(uint32 ssrc) const; |
240 int GetSendChannelId(uint32 ssrc) const; | 240 int GetSendChannelId(uint32 ssrc) const; |
241 | 241 |
242 private: | 242 private: |
243 bool SetSendCodecs(const std::vector<AudioCodec>& codecs); | 243 bool SetSendCodecs(const std::vector<AudioCodec>& codecs); |
244 bool SetSendRtpHeaderExtensions( | 244 bool SetSendRtpHeaderExtensions( |
245 const std::vector<RtpHeaderExtension>& extensions); | 245 const std::vector<RtpHeaderExtension>& extensions); |
246 bool SetOptions(const AudioOptions& options); | 246 bool SetOptions(const AudioOptions& options); |
| 247 bool SetRecvOptions(int channel_id, const AudioOptions& options); |
247 bool SetMaxSendBandwidth(int bps); | 248 bool SetMaxSendBandwidth(int bps); |
248 bool SetRecvCodecs(const std::vector<AudioCodec>& codecs); | 249 bool SetRecvCodecs(const std::vector<AudioCodec>& codecs); |
249 bool SetRecvRtpHeaderExtensions( | 250 bool SetRecvRtpHeaderExtensions( |
250 const std::vector<RtpHeaderExtension>& extensions); | 251 const std::vector<RtpHeaderExtension>& extensions); |
251 bool SetLocalRenderer(uint32 ssrc, AudioRenderer* renderer); | 252 bool SetLocalRenderer(uint32 ssrc, AudioRenderer* renderer); |
252 bool MuteStream(uint32 ssrc, bool mute); | 253 bool MuteStream(uint32 ssrc, bool mute); |
253 | 254 |
254 WebRtcVoiceEngine* engine() { return engine_; } | 255 WebRtcVoiceEngine* engine() { return engine_; } |
255 int GetLastEngineError() { return engine()->GetLastEngineError(); } | 256 int GetLastEngineError() { return engine()->GetLastEngineError(); } |
256 int GetOutputLevel(int channel); | 257 int GetOutputLevel(int channel); |
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342 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; | 343 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; |
343 | 344 |
344 // Do not lock this on the VoE media processor thread; potential for deadlock | 345 // Do not lock this on the VoE media processor thread; potential for deadlock |
345 // exists. | 346 // exists. |
346 mutable rtc::CriticalSection receive_channels_cs_; | 347 mutable rtc::CriticalSection receive_channels_cs_; |
347 }; | 348 }; |
348 | 349 |
349 } // namespace cricket | 350 } // namespace cricket |
350 | 351 |
351 #endif // TALK_MEDIA_WEBRTCVOICEENGINE_H_ | 352 #endif // TALK_MEDIA_WEBRTCVOICEENGINE_H_ |
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