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

Issue 1362913004: Remove unused SignalMediaError and infrastructure. (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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214 214
215 bool CanInsertDtmf() override; 215 bool CanInsertDtmf() override;
216 bool InsertDtmf(uint32 ssrc, int event, int duration, int flags) override; 216 bool InsertDtmf(uint32 ssrc, int event, int duration, int flags) override;
217 217
218 void OnPacketReceived(rtc::Buffer* packet, 218 void OnPacketReceived(rtc::Buffer* packet,
219 const rtc::PacketTime& packet_time) override; 219 const rtc::PacketTime& packet_time) override;
220 void OnRtcpReceived(rtc::Buffer* packet, 220 void OnRtcpReceived(rtc::Buffer* packet,
221 const rtc::PacketTime& packet_time) override; 221 const rtc::PacketTime& packet_time) override;
222 void OnReadyToSend(bool ready) override {} 222 void OnReadyToSend(bool ready) override {}
223 bool GetStats(VoiceMediaInfo* info) override; 223 bool GetStats(VoiceMediaInfo* info) override;
224 // Gets last reported error from WebRtc voice engine. This should be only
225 // called in response a failure.
226 void GetLastMediaError(uint32* ssrc,
227 VoiceMediaChannel::Error* error) override;
228 224
229 // implements Transport interface 225 // implements Transport interface
230 bool SendRtp(const uint8_t* data, size_t len) override { 226 bool SendRtp(const uint8_t* data, size_t len) override {
231 rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len, 227 rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len,
232 kMaxRtpPacketLen); 228 kMaxRtpPacketLen);
233 return VoiceMediaChannel::SendPacket(&packet); 229 return VoiceMediaChannel::SendPacket(&packet);
234 } 230 }
235 231
236 bool SendRtcp(const uint8_t* data, size_t len) override { 232 bool SendRtcp(const uint8_t* data, size_t len) override {
237 rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len, 233 rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len,
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348 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; 344 std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
349 345
350 // Do not lock this on the VoE media processor thread; potential for deadlock 346 // Do not lock this on the VoE media processor thread; potential for deadlock
351 // exists. 347 // exists.
352 mutable rtc::CriticalSection receive_channels_cs_; 348 mutable rtc::CriticalSection receive_channels_cs_;
353 }; 349 };
354 350
355 } // namespace cricket 351 } // namespace cricket
356 352
357 #endif // TALK_MEDIA_WEBRTCVOICEENGINE_H_ 353 #endif // TALK_MEDIA_WEBRTCVOICEENGINE_H_
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