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

Issue 1363573002: Wire up transport sequence number / send time callbacks to webrtc via libjingle. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: No changes to const parameters. 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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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;
224 224
225 // implements Transport interface 225 // implements Transport interface
226 bool SendRtp(const uint8_t* data, 226 bool SendRtp(const uint8_t* data,
227 size_t len, 227 size_t len,
228 const webrtc::PacketOptions& options) override { 228 const webrtc::PacketOptions& options) override {
229 rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len, 229 rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len,
230 kMaxRtpPacketLen); 230 kMaxRtpPacketLen);
231 return VoiceMediaChannel::SendPacket(&packet); 231 rtc::PacketOptions rtc_options;
232 rtc_options.packet_id = options.packet_id;
233 return VoiceMediaChannel::SendPacket(&packet, rtc_options);
232 } 234 }
233 235
234 bool SendRtcp(const uint8_t* data, size_t len) override { 236 bool SendRtcp(const uint8_t* data, size_t len) override {
235 rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len, 237 rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len,
236 kMaxRtpPacketLen); 238 kMaxRtpPacketLen);
237 return VoiceMediaChannel::SendRtcp(&packet); 239 return VoiceMediaChannel::SendRtcp(&packet, rtc::PacketOptions());
238 } 240 }
239 241
240 bool FindSsrc(int channel_num, uint32* ssrc); 242 bool FindSsrc(int channel_num, uint32* ssrc);
241 void OnError(uint32 ssrc, int error); 243 void OnError(uint32 ssrc, int error);
242 244
243 int GetReceiveChannelNum(uint32 ssrc) const; 245 int GetReceiveChannelNum(uint32 ssrc) const;
244 int GetSendChannelNum(uint32 ssrc) const; 246 int GetSendChannelNum(uint32 ssrc) const;
245 247
246 private: 248 private:
247 bool SetSendCodecs(const std::vector<AudioCodec>& codecs); 249 bool SetSendCodecs(const std::vector<AudioCodec>& codecs);
(...skipping 98 matching lines...) Expand 10 before | Expand all | Expand 10 after
346 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; 348 std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
347 349
348 // Do not lock this on the VoE media processor thread; potential for deadlock 350 // Do not lock this on the VoE media processor thread; potential for deadlock
349 // exists. 351 // exists.
350 mutable rtc::CriticalSection receive_channels_cs_; 352 mutable rtc::CriticalSection receive_channels_cs_;
351 }; 353 };
352 354
353 } // namespace cricket 355 } // namespace cricket
354 356
355 #endif // TALK_MEDIA_WEBRTCVOICEENGINE_H_ 357 #endif // TALK_MEDIA_WEBRTCVOICEENGINE_H_
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