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Diffstat (limited to 'third_party/libwebrtc/video/rtp_streams_synchronizer2.h')
-rw-r--r-- | third_party/libwebrtc/video/rtp_streams_synchronizer2.h | 75 |
1 files changed, 75 insertions, 0 deletions
diff --git a/third_party/libwebrtc/video/rtp_streams_synchronizer2.h b/third_party/libwebrtc/video/rtp_streams_synchronizer2.h new file mode 100644 index 0000000000..7042b1bd9a --- /dev/null +++ b/third_party/libwebrtc/video/rtp_streams_synchronizer2.h @@ -0,0 +1,75 @@ +/* + * Copyright (c) 2020 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. + */ + +#ifndef VIDEO_RTP_STREAMS_SYNCHRONIZER2_H_ +#define VIDEO_RTP_STREAMS_SYNCHRONIZER2_H_ + +#include <memory> + +#include "api/sequence_checker.h" +#include "api/task_queue/task_queue_base.h" +#include "rtc_base/system/no_unique_address.h" +#include "rtc_base/task_utils/repeating_task.h" +#include "video/stream_synchronization.h" + +namespace webrtc { + +class Syncable; + +namespace internal { + +// RtpStreamsSynchronizer is responsible for synchronizing audio and video for +// a given audio receive stream and video receive stream. +class RtpStreamsSynchronizer { + public: + RtpStreamsSynchronizer(TaskQueueBase* main_queue, Syncable* syncable_video); + ~RtpStreamsSynchronizer(); + + void ConfigureSync(Syncable* syncable_audio); + + // 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: + void UpdateDelay(); + + TaskQueueBase* const task_queue_; + + // Used to check if we're running on the main thread/task queue. + // The reason we currently don't use RTC_DCHECK_RUN_ON(task_queue_) is because + // we might be running on an rtc::Thread implementation of TaskQueue, which + // does not consistently set itself as the active TaskQueue. + // Instead, we rely on a SequenceChecker for now. + RTC_NO_UNIQUE_ADDRESS SequenceChecker main_checker_; + + Syncable* const syncable_video_; + + Syncable* syncable_audio_ RTC_GUARDED_BY(main_checker_) = nullptr; + std::unique_ptr<StreamSynchronization> sync_ RTC_GUARDED_BY(main_checker_); + StreamSynchronization::Measurements audio_measurement_ + RTC_GUARDED_BY(main_checker_); + StreamSynchronization::Measurements video_measurement_ + RTC_GUARDED_BY(main_checker_); + RepeatingTaskHandle repeating_task_ RTC_GUARDED_BY(main_checker_); + int64_t last_stats_log_ms_ RTC_GUARDED_BY(&main_checker_); +}; + +} // namespace internal +} // namespace webrtc + +#endif // VIDEO_RTP_STREAMS_SYNCHRONIZER2_H_ |