| /* |
| * 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. |
| */ |
| |
| // RtpStreamsSynchronizer is responsible for synchronizing audio and video for |
| // a given audio receive stream and video receive stream. |
| |
| #ifndef VIDEO_RTP_STREAMS_SYNCHRONIZER_H_ |
| #define VIDEO_RTP_STREAMS_SYNCHRONIZER_H_ |
| |
| #include <memory> |
| |
| #include "modules/include/module.h" |
| #include "rtc_base/synchronization/mutex.h" |
| #include "rtc_base/thread_checker.h" |
| #include "video/stream_synchronization.h" |
| |
| namespace webrtc { |
| |
| class Syncable; |
| |
| // DEPRECATED. |
| class RtpStreamsSynchronizer : public Module { |
| public: |
| explicit RtpStreamsSynchronizer(Syncable* syncable_video); |
| ~RtpStreamsSynchronizer() override; |
| |
| void ConfigureSync(Syncable* syncable_audio); |
| |
| // Implements Module. |
| int64_t TimeUntilNextProcess() override; |
| void Process() override; |
| |
| // Gets the estimated playout NTP timestamp for the video frame with |
| // |rtp_timestamp| and the sync offset between the current played out audio |
| // frame and the video frame. Returns true on success, false otherwise. |
| // The |estimated_freq_khz| is the frequency used in the RTP to NTP timestamp |
| // conversion. |
| bool GetStreamSyncOffsetInMs(uint32_t rtp_timestamp, |
| int64_t render_time_ms, |
| int64_t* video_playout_ntp_ms, |
| int64_t* stream_offset_ms, |
| double* estimated_freq_khz) const; |
| |
| private: |
| Syncable* syncable_video_; |
| |
| mutable Mutex mutex_; |
| Syncable* syncable_audio_ RTC_GUARDED_BY(mutex_); |
| std::unique_ptr<StreamSynchronization> sync_ RTC_GUARDED_BY(mutex_); |
| StreamSynchronization::Measurements audio_measurement_ RTC_GUARDED_BY(mutex_); |
| StreamSynchronization::Measurements video_measurement_ RTC_GUARDED_BY(mutex_); |
| |
| rtc::ThreadChecker process_thread_checker_; |
| int64_t last_sync_time_ RTC_GUARDED_BY(&process_thread_checker_); |
| int64_t last_stats_log_ms_ RTC_GUARDED_BY(&process_thread_checker_); |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // VIDEO_RTP_STREAMS_SYNCHRONIZER_H_ |