de9e5fffa2
- Add histogram: "WebRTC.Video.RtpToNtpFreqOffsetInKhz" The absolute value of the difference between the estimated frequency during RTP timestamp to NTP time conversion and the actual value (i.e. 90 kHz) is measured per received video frame. The max offset during 40 second intervals is stored. The average of these stored offsets per received video stream is recorded when a stream is removed. Updated rtp_to_ntp.cc: - Add validation for only inserting newer RTCP sender reports to the rtcp list. - Move calculation of frequency/offset (from RTP/NTP timestamp pairs) to UpdateRtcpList. Calculated when a new RTCP SR in inserted (and not in RtpToNtpMs per packet). BUG=webrtc:6579 Review-Url: https://codereview.webrtc.org/2385763002 Cr-Commit-Position: refs/heads/master@{#14891}
77 lines
2.5 KiB
C++
77 lines
2.5 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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// RtpStreamsSynchronizer is responsible for synchronization audio and video for
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// a given voice engine channel and video receive stream.
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#ifndef WEBRTC_VIDEO_RTP_STREAMS_SYNCHRONIZER_H_
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#define WEBRTC_VIDEO_RTP_STREAMS_SYNCHRONIZER_H_
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#include <memory>
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#include "webrtc/base/criticalsection.h"
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#include "webrtc/base/thread_checker.h"
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#include "webrtc/modules/include/module.h"
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#include "webrtc/video/rtp_stream_receiver.h"
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#include "webrtc/video/stream_synchronization.h"
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namespace webrtc {
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class Clock;
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class VideoFrame;
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class VoEVideoSync;
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namespace vcm {
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class VideoReceiver;
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} // namespace vcm
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class RtpStreamsSynchronizer : public Module {
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public:
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RtpStreamsSynchronizer(vcm::VideoReceiver* vcm,
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RtpStreamReceiver* rtp_stream_receiver);
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void ConfigureSync(int voe_channel_id,
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VoEVideoSync* voe_sync_interface);
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// Implements Module.
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int64_t TimeUntilNextProcess() override;
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void Process() override;
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// Gets the sync offset between the current played out audio frame and the
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// video |frame|. Returns true on success, false otherwise.
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// The estimated frequency is the frequency used in the RTP to NTP timestamp
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// conversion.
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bool GetStreamSyncOffsetInMs(const VideoFrame& frame,
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int64_t* stream_offset_ms,
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double* estimated_freq_khz) const;
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private:
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Clock* const clock_;
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vcm::VideoReceiver* const video_receiver_;
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RtpReceiver* const video_rtp_receiver_;
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RtpRtcp* const video_rtp_rtcp_;
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rtc::CriticalSection crit_;
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int voe_channel_id_ GUARDED_BY(crit_);
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VoEVideoSync* voe_sync_interface_ GUARDED_BY(crit_);
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RtpReceiver* audio_rtp_receiver_ GUARDED_BY(crit_);
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RtpRtcp* audio_rtp_rtcp_ GUARDED_BY(crit_);
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std::unique_ptr<StreamSynchronization> sync_ GUARDED_BY(crit_);
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StreamSynchronization::Measurements audio_measurement_ GUARDED_BY(crit_);
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StreamSynchronization::Measurements video_measurement_ GUARDED_BY(crit_);
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rtc::ThreadChecker process_thread_checker_;
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int64_t last_sync_time_ ACCESS_ON(&process_thread_checker_);
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};
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} // namespace webrtc
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#endif // WEBRTC_VIDEO_RTP_STREAMS_SYNCHRONIZER_H_
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