7623ce4aeb
Reason for revert: Bot breakage caused by TickTime::UseFakeClock has been removed. Original issue's description: > Revert of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:20001 of https://codereview.webrtc.org/1506773002/ ) > > Reason for revert: > Breaks Dr Memory Light https://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Light/builds/4643 and all the Android Tests bots. > > Original issue's description: > > Merge webrtc/video_engine/ into webrtc/video/ > > > > BUG=webrtc:1695 > > R=mflodman@webrtc.org > > > > Committed: https://crrev.com/03ef053202bc5d5ab43460eebf5403232f157646 > > Cr-Commit-Position: refs/heads/master@{#10926} > > TBR=mflodman@webrtc.org,pbos@webrtc.org > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:1695 > > Committed: https://crrev.com/8237abf563bf4782ee104408b53cc8e55ce44518 > Cr-Commit-Position: refs/heads/master@{#10937} BUG=webrtc:1695 TBR=mflodman@webrtc.org,kjellander@webrtc.org NOPRESUBMIT=true Review URL: https://codereview.webrtc.org/1510183002 . Cr-Commit-Position: refs/heads/master@{#10948}
60 lines
1.9 KiB
C++
60 lines
1.9 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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struct ViESyncDelay;
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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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RtcpList 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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~StreamSynchronization();
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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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ViESyncDelay* 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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