1 /*
2  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
3  *
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
6  *  tree. An additional intellectual property rights grant can be found
7  *  in the file PATENTS.  All contributing project authors may
8  *  be found in the AUTHORS file in the root of the source tree.
9  */
10 
11 // RtpStreamsSynchronizer is responsible for synchronizing audio and video for
12 // a given audio receive stream and video receive stream.
13 
14 #ifndef VIDEO_RTP_STREAMS_SYNCHRONIZER_H_
15 #define VIDEO_RTP_STREAMS_SYNCHRONIZER_H_
16 
17 #include <memory>
18 
19 #include "modules/include/module.h"
20 #include "rtc_base/synchronization/mutex.h"
21 #include "rtc_base/thread_checker.h"
22 #include "video/stream_synchronization.h"
23 
24 namespace webrtc {
25 
26 class Syncable;
27 
28 // DEPRECATED.
29 class RtpStreamsSynchronizer : public Module {
30  public:
31   explicit RtpStreamsSynchronizer(Syncable* syncable_video);
32   ~RtpStreamsSynchronizer() override;
33 
34   void ConfigureSync(Syncable* syncable_audio);
35 
36   // Implements Module.
37   int64_t TimeUntilNextProcess() override;
38   void Process() override;
39 
40   // Gets the estimated playout NTP timestamp for the video frame with
41   // |rtp_timestamp| and the sync offset between the current played out audio
42   // frame and the video frame. Returns true on success, false otherwise.
43   // The |estimated_freq_khz| is the frequency used in the RTP to NTP timestamp
44   // conversion.
45   bool GetStreamSyncOffsetInMs(uint32_t rtp_timestamp,
46                                int64_t render_time_ms,
47                                int64_t* video_playout_ntp_ms,
48                                int64_t* stream_offset_ms,
49                                double* estimated_freq_khz) const;
50 
51  private:
52   Syncable* syncable_video_;
53 
54   mutable Mutex mutex_;
55   Syncable* syncable_audio_ RTC_GUARDED_BY(mutex_);
56   std::unique_ptr<StreamSynchronization> sync_ RTC_GUARDED_BY(mutex_);
57   StreamSynchronization::Measurements audio_measurement_ RTC_GUARDED_BY(mutex_);
58   StreamSynchronization::Measurements video_measurement_ RTC_GUARDED_BY(mutex_);
59 
60   rtc::ThreadChecker process_thread_checker_;
61   int64_t last_sync_time_ RTC_GUARDED_BY(&process_thread_checker_);
62   int64_t last_stats_log_ms_ RTC_GUARDED_BY(&process_thread_checker_);
63 };
64 
65 }  // namespace webrtc
66 
67 #endif  // VIDEO_RTP_STREAMS_SYNCHRONIZER_H_
68