/* * 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 #include "webrtc/base/criticalsection.h" #include "webrtc/modules/include/module.h" #include "webrtc/video/stream_synchronization.h" #include "webrtc/voice_engine/include/voe_video_sync.h" namespace webrtc { class Clock; class RtpRtcp; class VideoFrame; class ViEChannel; class VoEVideoSync; namespace vcm { class VideoReceiver; } // namespace vcm class ViESyncModule : public Module { public: explicit ViESyncModule(vcm::VideoReceiver* vcm); ~ViESyncModule(); void ConfigureSync(int voe_channel_id, VoEVideoSync* voe_sync_interface, RtpRtcp* video_rtcp_module, RtpReceiver* rtp_receiver); // Implements Module. int64_t TimeUntilNextProcess() override; void Process() override; // Gets the sync offset between the current played out audio frame and the // video |frame|. Returns true on success, false otherwise. bool GetStreamSyncOffsetInMs(const VideoFrame& frame, int64_t* stream_offset_ms) const; private: rtc::CriticalSection data_cs_; vcm::VideoReceiver* const video_receiver_; Clock* const clock_; RtpReceiver* rtp_receiver_; RtpRtcp* video_rtp_rtcp_; int voe_channel_id_; VoEVideoSync* voe_sync_interface_; int64_t last_sync_time_; std::unique_ptr sync_; StreamSynchronization::Measurements audio_measurement_; StreamSynchronization::Measurements video_measurement_; }; } // namespace webrtc #endif // WEBRTC_VIDEO_VIE_SYNC_MODULE_H_