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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: more 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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237 237
238 bool CanInsertDtmf() override; 238 bool CanInsertDtmf() override;
239 bool InsertDtmf(uint32 ssrc, int event, int duration, int flags) override; 239 bool InsertDtmf(uint32 ssrc, int event, int duration, int flags) override;
240 240
241 void OnPacketReceived(rtc::Buffer* packet, 241 void OnPacketReceived(rtc::Buffer* packet,
242 const rtc::PacketTime& packet_time) override; 242 const rtc::PacketTime& packet_time) override;
243 void OnRtcpReceived(rtc::Buffer* packet, 243 void OnRtcpReceived(rtc::Buffer* packet,
244 const rtc::PacketTime& packet_time) override; 244 const rtc::PacketTime& packet_time) override;
245 void OnReadyToSend(bool ready) override {} 245 void OnReadyToSend(bool ready) override {}
246 bool GetStats(VoiceMediaInfo* info) override; 246 bool GetStats(VoiceMediaInfo* info) override;
247 // Gets last reported error from WebRtc voice engine. This should be only
248 // called in response a failure.
249 void GetLastMediaError(uint32* ssrc,
250 VoiceMediaChannel::Error* error) override;
251 247
252 // implements Transport interface 248 // implements Transport interface
253 int SendPacket(const void* data, size_t len) override { 249 int SendPacket(const void* data, size_t len) override {
254 rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len, 250 rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len,
255 kMaxRtpPacketLen); 251 kMaxRtpPacketLen);
256 return VoiceMediaChannel::SendPacket(&packet) ? static_cast<int>(len) : -1; 252 return VoiceMediaChannel::SendPacket(&packet) ? static_cast<int>(len) : -1;
257 } 253 }
258 254
259 int SendRTCPPacket(const void* data, size_t len) override { 255 int SendRTCPPacket(const void* data, size_t len) override {
260 rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len, 256 rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len,
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372 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; 368 std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
373 369
374 // Do not lock this on the VoE media processor thread; potential for deadlock 370 // Do not lock this on the VoE media processor thread; potential for deadlock
375 // exists. 371 // exists.
376 mutable rtc::CriticalSection receive_channels_cs_; 372 mutable rtc::CriticalSection receive_channels_cs_;
377 }; 373 };
378 374
379 } // namespace cricket 375 } // namespace cricket
380 376
381 #endif // TALK_MEDIA_WEBRTCVOICEENGINE_H_ 377 #endif // TALK_MEDIA_WEBRTCVOICEENGINE_H_
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