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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: Connect to SignalSentPacket when enabling bundle. 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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220 const rtc::PacketTime& packet_time) override; 220 const rtc::PacketTime& packet_time) override;
221 void OnReadyToSend(bool ready) override {} 221 void OnReadyToSend(bool ready) override {}
222 bool GetStats(VoiceMediaInfo* info) override; 222 bool GetStats(VoiceMediaInfo* info) override;
223 223
224 // implements Transport interface 224 // implements Transport interface
225 bool SendRtp(const uint8_t* data, 225 bool SendRtp(const uint8_t* data,
226 size_t len, 226 size_t len,
227 const webrtc::PacketOptions& options) override { 227 const webrtc::PacketOptions& options) override {
228 rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len, 228 rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len,
229 kMaxRtpPacketLen); 229 kMaxRtpPacketLen);
230 return VoiceMediaChannel::SendPacket(&packet); 230 rtc::PacketOptions rtc_options;
231 rtc_options.packet_id = options.packet_id;
232 return VoiceMediaChannel::SendPacket(&packet, rtc_options);
231 } 233 }
232 234
233 bool SendRtcp(const uint8_t* data, size_t len) override { 235 bool SendRtcp(const uint8_t* data, size_t len) override {
234 rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len, 236 rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len,
235 kMaxRtpPacketLen); 237 kMaxRtpPacketLen);
236 return VoiceMediaChannel::SendRtcp(&packet); 238 return VoiceMediaChannel::SendRtcp(&packet, rtc::PacketOptions());
237 } 239 }
238 240
239 void OnError(int error); 241 void OnError(int error);
240 242
241 int GetReceiveChannelId(uint32_t ssrc) const; 243 int GetReceiveChannelId(uint32_t ssrc) const;
242 int GetSendChannelId(uint32_t ssrc) const; 244 int GetSendChannelId(uint32_t ssrc) const;
243 245
244 private: 246 private:
245 bool SetSendCodecs(const std::vector<AudioCodec>& codecs); 247 bool SetSendCodecs(const std::vector<AudioCodec>& codecs);
246 bool SetSendRtpHeaderExtensions( 248 bool SetSendRtpHeaderExtensions(
(...skipping 97 matching lines...) Expand 10 before | Expand all | Expand 10 after
344 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; 346 std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
345 347
346 // Do not lock this on the VoE media processor thread; potential for deadlock 348 // Do not lock this on the VoE media processor thread; potential for deadlock
347 // exists. 349 // exists.
348 mutable rtc::CriticalSection receive_channels_cs_; 350 mutable rtc::CriticalSection receive_channels_cs_;
349 }; 351 };
350 352
351 } // namespace cricket 353 } // namespace cricket
352 354
353 #endif // TALK_MEDIA_WEBRTCVOICEENGINE_H_ 355 #endif // TALK_MEDIA_WEBRTCVOICEENGINE_H_
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