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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 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 |
11 #ifndef WEBRTC_VOICE_ENGINE_CHANNEL_H_ | 11 #ifndef WEBRTC_VOICE_ENGINE_CHANNEL_H_ |
12 #define WEBRTC_VOICE_ENGINE_CHANNEL_H_ | 12 #define WEBRTC_VOICE_ENGINE_CHANNEL_H_ |
13 | 13 |
| 14 #include "webrtc/audio/audio_sink.h" |
14 #include "webrtc/base/criticalsection.h" | 15 #include "webrtc/base/criticalsection.h" |
15 #include "webrtc/base/scoped_ptr.h" | 16 #include "webrtc/base/scoped_ptr.h" |
16 #include "webrtc/common_audio/resampler/include/push_resampler.h" | 17 #include "webrtc/common_audio/resampler/include/push_resampler.h" |
17 #include "webrtc/common_types.h" | 18 #include "webrtc/common_types.h" |
18 #include "webrtc/modules/audio_coding/include/audio_coding_module.h" | 19 #include "webrtc/modules/audio_coding/include/audio_coding_module.h" |
19 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_d
efines.h" | 20 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_d
efines.h" |
20 #include "webrtc/modules/audio_processing/rms_level.h" | 21 #include "webrtc/modules/audio_processing/rms_level.h" |
21 #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h" | 22 #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h" |
22 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" | 23 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" |
23 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" | 24 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" |
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185 int32_t SetEngineInformation( | 186 int32_t SetEngineInformation( |
186 Statistics& engineStatistics, | 187 Statistics& engineStatistics, |
187 OutputMixer& outputMixer, | 188 OutputMixer& outputMixer, |
188 TransmitMixer& transmitMixer, | 189 TransmitMixer& transmitMixer, |
189 ProcessThread& moduleProcessThread, | 190 ProcessThread& moduleProcessThread, |
190 AudioDeviceModule& audioDeviceModule, | 191 AudioDeviceModule& audioDeviceModule, |
191 VoiceEngineObserver* voiceEngineObserver, | 192 VoiceEngineObserver* voiceEngineObserver, |
192 CriticalSectionWrapper* callbackCritSect); | 193 CriticalSectionWrapper* callbackCritSect); |
193 int32_t UpdateLocalTimeStamp(); | 194 int32_t UpdateLocalTimeStamp(); |
194 | 195 |
| 196 void SetSink(rtc::scoped_ptr<AudioSinkInterface> sink); |
| 197 |
195 // API methods | 198 // API methods |
196 | 199 |
197 // VoEBase | 200 // VoEBase |
198 int32_t StartPlayout(); | 201 int32_t StartPlayout(); |
199 int32_t StopPlayout(); | 202 int32_t StopPlayout(); |
200 int32_t StartSend(); | 203 int32_t StartSend(); |
201 int32_t StopSend(); | 204 int32_t StopSend(); |
202 int32_t StartReceiving(); | 205 int32_t StartReceiving(); |
203 int32_t StopReceiving(); | 206 int32_t StopReceiving(); |
204 | 207 |
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501 RtcEventLog* const event_log_; | 504 RtcEventLog* const event_log_; |
502 | 505 |
503 rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_; | 506 rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_; |
504 rtc::scoped_ptr<RTPPayloadRegistry> rtp_payload_registry_; | 507 rtc::scoped_ptr<RTPPayloadRegistry> rtp_payload_registry_; |
505 rtc::scoped_ptr<ReceiveStatistics> rtp_receive_statistics_; | 508 rtc::scoped_ptr<ReceiveStatistics> rtp_receive_statistics_; |
506 rtc::scoped_ptr<StatisticsProxy> statistics_proxy_; | 509 rtc::scoped_ptr<StatisticsProxy> statistics_proxy_; |
507 rtc::scoped_ptr<RtpReceiver> rtp_receiver_; | 510 rtc::scoped_ptr<RtpReceiver> rtp_receiver_; |
508 TelephoneEventHandler* telephone_event_handler_; | 511 TelephoneEventHandler* telephone_event_handler_; |
509 rtc::scoped_ptr<RtpRtcp> _rtpRtcpModule; | 512 rtc::scoped_ptr<RtpRtcp> _rtpRtcpModule; |
510 rtc::scoped_ptr<AudioCodingModule> audio_coding_; | 513 rtc::scoped_ptr<AudioCodingModule> audio_coding_; |
| 514 rtc::scoped_ptr<AudioSinkInterface> audio_sink_; |
511 AudioLevel _outputAudioLevel; | 515 AudioLevel _outputAudioLevel; |
512 bool _externalTransport; | 516 bool _externalTransport; |
513 AudioFrame _audioFrame; | 517 AudioFrame _audioFrame; |
514 // Downsamples to the codec rate if necessary. | 518 // Downsamples to the codec rate if necessary. |
515 PushResampler<int16_t> input_resampler_; | 519 PushResampler<int16_t> input_resampler_; |
516 FilePlayer* _inputFilePlayerPtr; | 520 FilePlayer* _inputFilePlayerPtr; |
517 FilePlayer* _outputFilePlayerPtr; | 521 FilePlayer* _outputFilePlayerPtr; |
518 FileRecorder* _outputFileRecorderPtr; | 522 FileRecorder* _outputFileRecorderPtr; |
519 int _inputFilePlayerId; | 523 int _inputFilePlayerId; |
520 int _outputFilePlayerId; | 524 int _outputFilePlayerId; |
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600 PacketRouter* packet_router_ = nullptr; | 604 PacketRouter* packet_router_ = nullptr; |
601 rtc::scoped_ptr<TransportFeedbackProxy> feedback_observer_proxy_; | 605 rtc::scoped_ptr<TransportFeedbackProxy> feedback_observer_proxy_; |
602 rtc::scoped_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_; | 606 rtc::scoped_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_; |
603 rtc::scoped_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_; | 607 rtc::scoped_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_; |
604 }; | 608 }; |
605 | 609 |
606 } // namespace voe | 610 } // namespace voe |
607 } // namespace webrtc | 611 } // namespace webrtc |
608 | 612 |
609 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ | 613 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ |
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