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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 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 | 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 | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
11 // RtpStreamsSynchronizer is responsible for synchronization audio and video for | 11 // RtpStreamsSynchronizer is responsible for synchronization audio and video for |
12 // a given voice engine channel and video receive stream. | 12 // a given voice engine channel and video receive stream. |
13 | 13 |
14 #ifndef WEBRTC_VIDEO_RTP_STREAMS_SYNCHRONIZER_H_ | 14 #ifndef WEBRTC_VIDEO_RTP_STREAMS_SYNCHRONIZER_H_ |
15 #define WEBRTC_VIDEO_RTP_STREAMS_SYNCHRONIZER_H_ | 15 #define WEBRTC_VIDEO_RTP_STREAMS_SYNCHRONIZER_H_ |
16 | 16 |
17 #include <memory> | 17 #include <memory> |
18 | 18 |
19 #include "webrtc/base/criticalsection.h" | 19 #include "webrtc/base/criticalsection.h" |
20 #include "webrtc/base/thread_checker.h" | 20 #include "webrtc/base/thread_checker.h" |
21 #include "webrtc/modules/include/module.h" | 21 #include "webrtc/modules/include/module.h" |
22 #include "webrtc/video/rtp_stream_receiver.h" | 22 #include "webrtc/video/rtp_stream_receiver.h" |
stefan-webrtc
2017/01/26 08:40:50
Remove this.
| |
23 #include "webrtc/video/stream_synchronization.h" | 23 #include "webrtc/video/stream_synchronization.h" |
24 | 24 |
25 namespace webrtc { | 25 namespace webrtc { |
26 | 26 |
27 class Clock; | 27 class Syncable; |
28 class VideoFrame; | 28 class VideoFrame; |
29 class VoEVideoSync; | |
30 | 29 |
31 namespace vcm { | 30 namespace vcm { |
32 class VideoReceiver; | 31 class VideoReceiver; |
33 } // namespace vcm | 32 } // namespace vcm |
34 | 33 |
35 class RtpStreamsSynchronizer : public Module { | 34 class RtpStreamsSynchronizer : public Module { |
36 public: | 35 public: |
37 RtpStreamsSynchronizer(vcm::VideoReceiver* vcm, | 36 explicit RtpStreamsSynchronizer(Syncable* syncable_video); |
38 RtpStreamReceiver* rtp_stream_receiver); | |
39 | 37 |
40 void ConfigureSync(int voe_channel_id, | 38 void ConfigureSync(Syncable* syncable_audio); |
41 VoEVideoSync* voe_sync_interface); | |
42 | 39 |
43 // Implements Module. | 40 // Implements Module. |
44 int64_t TimeUntilNextProcess() override; | 41 int64_t TimeUntilNextProcess() override; |
45 void Process() override; | 42 void Process() override; |
46 | 43 |
47 // Gets the sync offset between the current played out audio frame and the | 44 // Gets the sync offset between the current played out audio frame and the |
48 // video |frame|. Returns true on success, false otherwise. | 45 // video |frame|. Returns true on success, false otherwise. |
49 // The estimated frequency is the frequency used in the RTP to NTP timestamp | 46 // The estimated frequency is the frequency used in the RTP to NTP timestamp |
50 // conversion. | 47 // conversion. |
51 bool GetStreamSyncOffsetInMs(const VideoFrame& frame, | 48 bool GetStreamSyncOffsetInMs(const VideoFrame& frame, |
stefan-webrtc
2017/01/26 08:40:50
If you feel like doing some more cleanup here, you
the sun
2017/01/30 15:43:12
Done.
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52 int64_t* stream_offset_ms, | 49 int64_t* stream_offset_ms, |
53 double* estimated_freq_khz) const; | 50 double* estimated_freq_khz) const; |
54 | 51 |
55 private: | 52 private: |
56 Clock* const clock_; | 53 Syncable* syncable_video_; |
57 vcm::VideoReceiver* const video_receiver_; | |
58 RtpReceiver* const video_rtp_receiver_; | |
59 RtpRtcp* const video_rtp_rtcp_; | |
60 | 54 |
61 rtc::CriticalSection crit_; | 55 rtc::CriticalSection crit_; |
62 int voe_channel_id_ GUARDED_BY(crit_); | 56 Syncable* syncable_audio_ GUARDED_BY(crit_); |
63 VoEVideoSync* voe_sync_interface_ GUARDED_BY(crit_); | |
64 RtpReceiver* audio_rtp_receiver_ GUARDED_BY(crit_); | |
65 RtpRtcp* audio_rtp_rtcp_ GUARDED_BY(crit_); | |
66 std::unique_ptr<StreamSynchronization> sync_ GUARDED_BY(crit_); | 57 std::unique_ptr<StreamSynchronization> sync_ GUARDED_BY(crit_); |
67 StreamSynchronization::Measurements audio_measurement_ GUARDED_BY(crit_); | 58 StreamSynchronization::Measurements audio_measurement_ GUARDED_BY(crit_); |
68 StreamSynchronization::Measurements video_measurement_ GUARDED_BY(crit_); | 59 StreamSynchronization::Measurements video_measurement_ GUARDED_BY(crit_); |
69 | 60 |
70 rtc::ThreadChecker process_thread_checker_; | 61 rtc::ThreadChecker process_thread_checker_; |
71 int64_t last_sync_time_ ACCESS_ON(&process_thread_checker_); | 62 int64_t last_sync_time_ ACCESS_ON(&process_thread_checker_); |
72 }; | 63 }; |
73 | 64 |
74 } // namespace webrtc | 65 } // namespace webrtc |
75 | 66 |
76 #endif // WEBRTC_VIDEO_RTP_STREAMS_SYNCHRONIZER_H_ | 67 #endif // WEBRTC_VIDEO_RTP_STREAMS_SYNCHRONIZER_H_ |
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