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| 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 synchronization audio and video for |
| 12 // a given voice engine channel and video receive stream. |
| 13 |
| 14 #ifndef WEBRTC_VIDEO_RTP_STREAMS_SYNCHRONIZER_H_ |
| 15 #define WEBRTC_VIDEO_RTP_STREAMS_SYNCHRONIZER_H_ |
| 16 |
| 17 #include <memory> |
| 18 |
| 19 #include "webrtc/base/criticalsection.h" |
| 20 #include "webrtc/base/thread_checker.h" |
| 21 #include "webrtc/modules/include/module.h" |
| 22 #include "webrtc/video/rtp_stream_receiver.h" |
| 23 #include "webrtc/video/stream_synchronization.h" |
| 24 |
| 25 namespace webrtc { |
| 26 |
| 27 class Clock; |
| 28 class VideoFrame; |
| 29 class VoEVideoSync; |
| 30 |
| 31 namespace vcm { |
| 32 class VideoReceiver; |
| 33 } // namespace vcm |
| 34 |
| 35 class RtpStreamsSynchronizer : public Module { |
| 36 public: |
| 37 RtpStreamsSynchronizer(vcm::VideoReceiver* vcm, |
| 38 RtpStreamReceiver* rtp_stream_receiver); |
| 39 |
| 40 void ConfigureSync(int voe_channel_id, |
| 41 VoEVideoSync* voe_sync_interface); |
| 42 |
| 43 // Implements Module. |
| 44 int64_t TimeUntilNextProcess() override; |
| 45 void Process() override; |
| 46 |
| 47 // Gets the sync offset between the current played out audio frame and the |
| 48 // video |frame|. Returns true on success, false otherwise. |
| 49 bool GetStreamSyncOffsetInMs(const VideoFrame& frame, |
| 50 int64_t* stream_offset_ms) const; |
| 51 |
| 52 private: |
| 53 Clock* const clock_; |
| 54 vcm::VideoReceiver* const video_receiver_; |
| 55 RtpReceiver* const video_rtp_receiver_; |
| 56 RtpRtcp* const video_rtp_rtcp_; |
| 57 |
| 58 rtc::CriticalSection crit_; |
| 59 int voe_channel_id_ GUARDED_BY(crit_); |
| 60 VoEVideoSync* voe_sync_interface_ GUARDED_BY(crit_); |
| 61 RtpReceiver* audio_rtp_receiver_ GUARDED_BY(crit_); |
| 62 RtpRtcp* audio_rtp_rtcp_ GUARDED_BY(crit_); |
| 63 std::unique_ptr<StreamSynchronization> sync_ GUARDED_BY(crit_); |
| 64 StreamSynchronization::Measurements audio_measurement_ GUARDED_BY(crit_); |
| 65 StreamSynchronization::Measurements video_measurement_ GUARDED_BY(crit_); |
| 66 |
| 67 rtc::ThreadChecker process_thread_checker_; |
| 68 int64_t last_sync_time_ ACCESS_ON(&process_thread_checker_); |
| 69 }; |
| 70 |
| 71 } // namespace webrtc |
| 72 |
| 73 #endif // WEBRTC_VIDEO_RTP_STREAMS_SYNCHRONIZER_H_ |
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