4cd2790f17
This CL removes (almost) the last RTP references in VideoReceiveStream. There are still references to RTPFragmentationHeader and SSRCs, which will be dealt with later. There are also new GUARDED_BY and thred checker added to the synchronization class. When there are othre transports than RTP, there will instead be an interface + inheritance for RtpStreamReceiver and RtpStreamSynchronizattion in VideoReceiveStream. This work will be done when we actually know how we want to make thee transport interface. BUG=webrtc:5838 Review-Url: https://codereview.webrtc.org/2216533002 Cr-Commit-Position: refs/heads/master@{#13655}
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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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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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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