/* * 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.
*/
// 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_;
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