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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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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 } |
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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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