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1 /* | 1 /* |
2 * Copyright (c) 2004 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2004 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
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215 void OnReadyToSend(bool ready) override; | 215 void OnReadyToSend(bool ready) override; |
216 void OnTransportOverheadChanged(int transport_overhead_per_packet) override; | 216 void OnTransportOverheadChanged(int transport_overhead_per_packet) override; |
217 bool GetStats(VoiceMediaInfo* info) override; | 217 bool GetStats(VoiceMediaInfo* info) override; |
218 | 218 |
219 // SSRC=0 will set the audio sink on the latest unsignaled stream, future or | 219 // SSRC=0 will set the audio sink on the latest unsignaled stream, future or |
220 // current. Only one stream at a time will use the sink. | 220 // current. Only one stream at a time will use the sink. |
221 void SetRawAudioSink( | 221 void SetRawAudioSink( |
222 uint32_t ssrc, | 222 uint32_t ssrc, |
223 std::unique_ptr<webrtc::AudioSinkInterface> sink) override; | 223 std::unique_ptr<webrtc::AudioSinkInterface> sink) override; |
224 | 224 |
225 std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const; | 225 std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const override; |
226 | 226 |
227 // implements Transport interface | 227 // implements Transport interface |
228 bool SendRtp(const uint8_t* data, | 228 bool SendRtp(const uint8_t* data, |
229 size_t len, | 229 size_t len, |
230 const webrtc::PacketOptions& options) override { | 230 const webrtc::PacketOptions& options) override { |
231 rtc::CopyOnWriteBuffer packet(data, len, kMaxRtpPacketLen); | 231 rtc::CopyOnWriteBuffer packet(data, len, kMaxRtpPacketLen); |
232 rtc::PacketOptions rtc_options; | 232 rtc::PacketOptions rtc_options; |
233 rtc_options.packet_id = options.packet_id; | 233 rtc_options.packet_id = options.packet_id; |
234 return VoiceMediaChannel::SendPacket(&packet, rtc_options); | 234 return VoiceMediaChannel::SendPacket(&packet, rtc_options); |
235 } | 235 } |
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303 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; | 303 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; |
304 | 304 |
305 rtc::Optional<webrtc::AudioSendStream::Config::SendCodecSpec> | 305 rtc::Optional<webrtc::AudioSendStream::Config::SendCodecSpec> |
306 send_codec_spec_; | 306 send_codec_spec_; |
307 | 307 |
308 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); | 308 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); |
309 }; | 309 }; |
310 } // namespace cricket | 310 } // namespace cricket |
311 | 311 |
312 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ | 312 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ |
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