ef9f76a59d
At the same time, renaming the send side API. Review URL: https://webrtc-codereview.appspot.com/1104004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3525 4adac7df-926f-26a2-2b94-8c16560cd09d
59 lines
1.9 KiB
C++
59 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_ENGINE_STREAM_SYNCHRONIZATION_H_
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#define WEBRTC_VIDEO_ENGINE_STREAM_SYNCHRONIZATION_H_
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#include <list>
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#include "modules/remote_bitrate_estimator/include/rtp_to_ntp.h"
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#include "typedefs.h" // NOLINT
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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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synchronization::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(int audio_channel_id, int video_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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int audio_channel_id_;
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int video_channel_id_;
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int base_target_delay_ms_;
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};
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} // namespace webrtc
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#endif // WEBRTC_VIDEO_ENGINE_STREAM_SYNCHRONIZATION_H_
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