92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
64 lines
2.0 KiB
C++
64 lines
2.0 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 VIDEO_STREAM_SYNCHRONIZATION_H_
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#define VIDEO_STREAM_SYNCHRONIZATION_H_
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#include <list>
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#include "system_wrappers/include/rtp_to_ntp_estimator.h"
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#include "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() : latest_receive_time_ms(0), latest_timestamp(0) {}
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RtpToNtpEstimator rtp_to_ntp;
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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(int video_stream_id, int audio_stream_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 int video_stream_id_;
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const int audio_stream_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 // VIDEO_STREAM_SYNCHRONIZATION_H_
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