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

Issue 1505253004: Support for remote audio into tracks (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Address comments Created 5 years 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,
(...skipping 178 matching lines...) Expand 10 before | Expand all | Expand 10 after
189 bool CanInsertDtmf() override; 189 bool CanInsertDtmf() override;
190 bool InsertDtmf(uint32_t ssrc, int event, int duration) override; 190 bool InsertDtmf(uint32_t ssrc, int event, int duration) override;
191 191
192 void OnPacketReceived(rtc::Buffer* packet, 192 void OnPacketReceived(rtc::Buffer* packet,
193 const rtc::PacketTime& packet_time) override; 193 const rtc::PacketTime& packet_time) override;
194 void OnRtcpReceived(rtc::Buffer* packet, 194 void OnRtcpReceived(rtc::Buffer* packet,
195 const rtc::PacketTime& packet_time) override; 195 const rtc::PacketTime& packet_time) override;
196 void OnReadyToSend(bool ready) override {} 196 void OnReadyToSend(bool ready) override {}
197 bool GetStats(VoiceMediaInfo* info) override; 197 bool GetStats(VoiceMediaInfo* info) override;
198 198
199 void SetRawAudioSink(
200 uint32_t ssrc,
201 rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) override;
202
199 // implements Transport interface 203 // implements Transport interface
200 bool SendRtp(const uint8_t* data, 204 bool SendRtp(const uint8_t* data,
201 size_t len, 205 size_t len,
202 const webrtc::PacketOptions& options) override { 206 const webrtc::PacketOptions& options) override {
203 rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len, 207 rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len,
204 kMaxRtpPacketLen); 208 kMaxRtpPacketLen);
205 rtc::PacketOptions rtc_options; 209 rtc::PacketOptions rtc_options;
206 rtc_options.packet_id = options.packet_id; 210 rtc_options.packet_id = options.packet_id;
207 return VoiceMediaChannel::SendPacket(&packet, rtc_options); 211 return VoiceMediaChannel::SendPacket(&packet, rtc_options);
208 } 212 }
(...skipping 67 matching lines...) Expand 10 before | Expand all | Expand 10 after
276 280
277 class WebRtcAudioReceiveStream; 281 class WebRtcAudioReceiveStream;
278 std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_; 282 std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_;
279 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; 283 std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
280 284
281 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); 285 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
282 }; 286 };
283 } // namespace cricket 287 } // namespace cricket
284 288
285 #endif // TALK_MEDIA_WEBRTCVOICEENGINE_H_ 289 #endif // TALK_MEDIA_WEBRTCVOICEENGINE_H_
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