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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: Add missing updated_options 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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219 const rtc::PacketTime& packet_time) override; 219 const rtc::PacketTime& packet_time) override;
220 void OnReadyToSend(bool ready) override {} 220 void OnReadyToSend(bool ready) override {}
221 bool GetStats(VoiceMediaInfo* info) override; 221 bool GetStats(VoiceMediaInfo* info) override;
222 222
223 // implements Transport interface 223 // implements Transport interface
224 bool SendRtp(const uint8_t* data, 224 bool SendRtp(const uint8_t* data,
225 size_t len, 225 size_t len,
226 const webrtc::PacketOptions& options) override { 226 const webrtc::PacketOptions& options) override {
227 rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len, 227 rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len,
228 kMaxRtpPacketLen); 228 kMaxRtpPacketLen);
229 return VoiceMediaChannel::SendPacket(&packet); 229 rtc::PacketOptions rtc_options;
230 rtc_options.packet_id = options.packet_id;
231 return VoiceMediaChannel::SendPacket(&packet, rtc_options);
230 } 232 }
231 233
232 bool SendRtcp(const uint8_t* data, size_t len) override { 234 bool SendRtcp(const uint8_t* data, size_t len) override {
233 rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len, 235 rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len,
234 kMaxRtpPacketLen); 236 kMaxRtpPacketLen);
235 return VoiceMediaChannel::SendRtcp(&packet); 237 return VoiceMediaChannel::SendRtcp(&packet, rtc::PacketOptions());
236 } 238 }
237 239
238 void OnError(int error); 240 void OnError(int error);
239 241
240 int GetReceiveChannelId(uint32_t ssrc) const; 242 int GetReceiveChannelId(uint32_t ssrc) const;
241 int GetSendChannelId(uint32_t ssrc) const; 243 int GetSendChannelId(uint32_t ssrc) const;
242 244
243 private: 245 private:
244 bool SetSendCodecs(const std::vector<AudioCodec>& codecs); 246 bool SetSendCodecs(const std::vector<AudioCodec>& codecs);
245 bool SetSendRtpHeaderExtensions( 247 bool SetSendRtpHeaderExtensions(
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338 // receive_channels_ can be read from WebRtc callback thread. Access from 340 // receive_channels_ can be read from WebRtc callback thread. Access from
339 // the WebRtc thread must be synchronized with edits on the worker thread. 341 // the WebRtc thread must be synchronized with edits on the worker thread.
340 // Reads on the worker thread are ok. 342 // Reads on the worker thread are ok.
341 std::vector<RtpHeaderExtension> receive_extensions_; 343 std::vector<RtpHeaderExtension> receive_extensions_;
342 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; 344 std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
343 }; 345 };
344 346
345 } // namespace cricket 347 } // namespace cricket
346 348
347 #endif // TALK_MEDIA_WEBRTCVOICEENGINE_H_ 349 #endif // TALK_MEDIA_WEBRTCVOICEENGINE_H_
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