| /* |
| * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| // ViESyncModule is responsible for synchronization audio and video for a given |
| // VoE and ViE channel couple. |
| |
| #ifndef WEBRTC_VIDEO_VIE_SYNC_MODULE_H_ |
| #define WEBRTC_VIDEO_VIE_SYNC_MODULE_H_ |
| |
| #include "webrtc/base/scoped_ptr.h" |
| #include "webrtc/modules/include/module.h" |
| #include "webrtc/system_wrappers/include/tick_util.h" |
| #include "webrtc/video/stream_synchronization.h" |
| #include "webrtc/voice_engine/include/voe_video_sync.h" |
| |
| namespace webrtc { |
| |
| class CriticalSectionWrapper; |
| class RtpRtcp; |
| class VideoCodingModule; |
| class ViEChannel; |
| class VoEVideoSync; |
| |
| class ViESyncModule : public Module { |
| public: |
| explicit ViESyncModule(VideoCodingModule* vcm); |
| ~ViESyncModule(); |
| |
| int ConfigureSync(int voe_channel_id, |
| VoEVideoSync* voe_sync_interface, |
| RtpRtcp* video_rtcp_module, |
| RtpReceiver* video_receiver); |
| |
| int VoiceChannel(); |
| |
| // Implements Module. |
| int64_t TimeUntilNextProcess() override; |
| int32_t Process() override; |
| |
| private: |
| rtc::scoped_ptr<CriticalSectionWrapper> data_cs_; |
| VideoCodingModule* const vcm_; |
| RtpReceiver* video_receiver_; |
| RtpRtcp* video_rtp_rtcp_; |
| int voe_channel_id_; |
| VoEVideoSync* voe_sync_interface_; |
| TickTime last_sync_time_; |
| rtc::scoped_ptr<StreamSynchronization> sync_; |
| StreamSynchronization::Measurements audio_measurement_; |
| StreamSynchronization::Measurements video_measurement_; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_VIDEO_VIE_SYNC_MODULE_H_ |