/* * 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_ENGINE_VIE_SYNC_MODULE_H_ #define WEBRTC_VIDEO_ENGINE_VIE_SYNC_MODULE_H_ #include "webrtc/modules/interface/module.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" #include "webrtc/system_wrappers/interface/tick_util.h" #include "webrtc/video_engine/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: ViESyncModule(VideoCodingModule* vcm, ViEChannel* vie_channel); ~ViESyncModule(); int ConfigureSync(int voe_channel_id, VoEVideoSync* voe_sync_interface, RtpRtcp* video_rtcp_module); int VoiceChannel(); // Set target delay for buffering mode (0 = real-time mode). int SetTargetBufferingDelay(int target_delay_ms); // Implements Module. virtual int32_t TimeUntilNextProcess(); virtual int32_t Process(); private: scoped_ptr data_cs_; VideoCodingModule* vcm_; ViEChannel* vie_channel_; RtpRtcp* video_rtp_rtcp_; int voe_channel_id_; VoEVideoSync* voe_sync_interface_; TickTime last_sync_time_; scoped_ptr sync_; StreamSynchronization::Measurements audio_measurement_; StreamSynchronization::Measurements video_measurement_; }; } // namespace webrtc #endif // WEBRTC_VIDEO_ENGINE_VIE_SYNC_MODULE_H_