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}
64 lines
2.1 KiB
C++
64 lines
2.1 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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#ifndef WEBRTC_VIDEO_STREAM_SYNCHRONIZATION_H_
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#define WEBRTC_VIDEO_STREAM_SYNCHRONIZATION_H_
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#include <list>
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#include "webrtc/system_wrappers/include/rtp_to_ntp.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class StreamSynchronization {
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public:
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struct Measurements {
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Measurements() : rtcp(), latest_receive_time_ms(0), latest_timestamp(0) {}
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RtcpMeasurements rtcp;
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int64_t latest_receive_time_ms;
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uint32_t latest_timestamp;
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};
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StreamSynchronization(uint32_t video_primary_ssrc, int audio_channel_id);
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bool ComputeDelays(int relative_delay_ms,
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int current_audio_delay_ms,
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int* extra_audio_delay_ms,
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int* total_video_delay_target_ms);
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// On success |relative_delay| contains the number of milliseconds later video
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// is rendered relative audio. If audio is played back later than video a
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// |relative_delay| will be negative.
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static bool ComputeRelativeDelay(const Measurements& audio_measurement,
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const Measurements& video_measurement,
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int* relative_delay_ms);
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// Set target buffering delay - All audio and video will be delayed by at
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// least target_delay_ms.
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void SetTargetBufferingDelay(int target_delay_ms);
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private:
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struct SynchronizationDelays {
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int extra_video_delay_ms = 0;
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int last_video_delay_ms = 0;
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int extra_audio_delay_ms = 0;
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int last_audio_delay_ms = 0;
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};
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SynchronizationDelays channel_delay_;
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const uint32_t video_primary_ssrc_;
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const int audio_channel_id_;
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int base_target_delay_ms_;
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int avg_diff_ms_;
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};
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} // namespace webrtc
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#endif // WEBRTC_VIDEO_STREAM_SYNCHRONIZATION_H_
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