| Index: webrtc/video_engine/stream_synchronization.h
|
| diff --git a/webrtc/video_engine/stream_synchronization.h b/webrtc/video_engine/stream_synchronization.h
|
| deleted file mode 100644
|
| index 1209062f9b24c0744059a52d4eb67e74ff335580..0000000000000000000000000000000000000000
|
| --- a/webrtc/video_engine/stream_synchronization.h
|
| +++ /dev/null
|
| @@ -1,59 +0,0 @@
|
| -/*
|
| - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
|
| - *
|
| - * Use of this source code is governed by a BSD-style license
|
| - * that can be found in the LICENSE file in the root of the source
|
| - * tree. An additional intellectual property rights grant can be found
|
| - * in the file PATENTS. All contributing project authors may
|
| - * be found in the AUTHORS file in the root of the source tree.
|
| - */
|
| -
|
| -#ifndef WEBRTC_VIDEO_ENGINE_STREAM_SYNCHRONIZATION_H_
|
| -#define WEBRTC_VIDEO_ENGINE_STREAM_SYNCHRONIZATION_H_
|
| -
|
| -#include <list>
|
| -
|
| -#include "webrtc/system_wrappers/include/rtp_to_ntp.h"
|
| -#include "webrtc/typedefs.h"
|
| -
|
| -namespace webrtc {
|
| -
|
| -struct ViESyncDelay;
|
| -
|
| -class StreamSynchronization {
|
| - public:
|
| - struct Measurements {
|
| - Measurements() : rtcp(), latest_receive_time_ms(0), latest_timestamp(0) {}
|
| - RtcpList rtcp;
|
| - int64_t latest_receive_time_ms;
|
| - uint32_t latest_timestamp;
|
| - };
|
| -
|
| - StreamSynchronization(uint32_t video_primary_ssrc, int audio_channel_id);
|
| - ~StreamSynchronization();
|
| -
|
| - bool ComputeDelays(int relative_delay_ms,
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| - int current_audio_delay_ms,
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| - int* extra_audio_delay_ms,
|
| - int* total_video_delay_target_ms);
|
| -
|
| - // On success |relative_delay| contains the number of milliseconds later video
|
| - // is rendered relative audio. If audio is played back later than video a
|
| - // |relative_delay| will be negative.
|
| - static bool ComputeRelativeDelay(const Measurements& audio_measurement,
|
| - const Measurements& video_measurement,
|
| - int* relative_delay_ms);
|
| - // Set target buffering delay - All audio and video will be delayed by at
|
| - // least target_delay_ms.
|
| - void SetTargetBufferingDelay(int target_delay_ms);
|
| -
|
| - private:
|
| - ViESyncDelay* channel_delay_;
|
| - const uint32_t video_primary_ssrc_;
|
| - const int audio_channel_id_;
|
| - int base_target_delay_ms_;
|
| - int avg_diff_ms_;
|
| -};
|
| -} // namespace webrtc
|
| -
|
| -#endif // WEBRTC_VIDEO_ENGINE_STREAM_SYNCHRONIZATION_H_
|
|
|