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

Issue 1398823003: Remove MediaChannel::SetRemoteRenderer(). (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 5 years, 2 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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193 bool PauseSend(); 193 bool PauseSend();
194 bool ResumeSend(); 194 bool ResumeSend();
195 bool SetAudioSend(uint32_t ssrc, 195 bool SetAudioSend(uint32_t ssrc,
196 bool enable, 196 bool enable,
197 const AudioOptions* options, 197 const AudioOptions* options,
198 AudioRenderer* renderer) override; 198 AudioRenderer* renderer) override;
199 bool AddSendStream(const StreamParams& sp) override; 199 bool AddSendStream(const StreamParams& sp) override;
200 bool RemoveSendStream(uint32_t ssrc) override; 200 bool RemoveSendStream(uint32_t ssrc) override;
201 bool AddRecvStream(const StreamParams& sp) override; 201 bool AddRecvStream(const StreamParams& sp) override;
202 bool RemoveRecvStream(uint32_t ssrc) override; 202 bool RemoveRecvStream(uint32_t ssrc) override;
203 bool SetRemoteRenderer(uint32_t ssrc, AudioRenderer* renderer) override;
204 bool GetActiveStreams(AudioInfo::StreamList* actives) override; 203 bool GetActiveStreams(AudioInfo::StreamList* actives) override;
205 int GetOutputLevel() override; 204 int GetOutputLevel() override;
206 int GetTimeSinceLastTyping() override; 205 int GetTimeSinceLastTyping() override;
207 void SetTypingDetectionParameters(int time_window, 206 void SetTypingDetectionParameters(int time_window,
208 int cost_per_typing, 207 int cost_per_typing,
209 int reporting_threshold, 208 int reporting_threshold,
210 int penalty_decay, 209 int penalty_decay,
211 int type_event_delay) override; 210 int type_event_delay) override;
212 bool SetOutputVolume(uint32_t ssrc, double volume) override; 211 bool SetOutputVolume(uint32_t ssrc, double volume) override;
213 212
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344 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; 343 std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
345 344
346 // 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
347 // exists. 346 // exists.
348 mutable rtc::CriticalSection receive_channels_cs_; 347 mutable rtc::CriticalSection receive_channels_cs_;
349 }; 348 };
350 349
351 } // namespace cricket 350 } // namespace cricket
352 351
353 #endif // TALK_MEDIA_WEBRTCVOICEENGINE_H_ 352 #endif // TALK_MEDIA_WEBRTCVOICEENGINE_H_
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