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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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