bef7b058f5
Setting a minimum delay can fail in some cases. It is important that the AV sync code is aware of failures and can act accordingly to recover and prevent sync delays that keep increasing indefinitely. Bug: webrtc:11805 Change-Id: I0deed951dc6c6d0905536a949af875e0a6d9f7fa Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183360 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32062}
72 lines
2.3 KiB
C++
72 lines
2.3 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 <stdint.h>
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#include "system_wrappers/include/rtp_to_ntp_estimator.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(uint32_t video_stream_id, uint32_t 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* total_audio_delay_target_ms,
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int* total_video_delay_target_ms);
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// On success |relative_delay_ms| contains the number of milliseconds later
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// video is rendered relative audio. If audio is played back later than video
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// |relative_delay_ms| 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. Audio and video will be delayed by at least
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// |target_delay_ms|.
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void SetTargetBufferingDelay(int target_delay_ms);
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// Lowers the audio delay by 10%. Can be used to recover from errors.
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void ReduceAudioDelay();
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// Lowers the video delay by 10%. Can be used to recover from errors.
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void ReduceVideoDelay();
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uint32_t audio_stream_id() const { return audio_stream_id_; }
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uint32_t video_stream_id() const { return video_stream_id_; }
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private:
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struct SynchronizationDelays {
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int extra_ms = 0;
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int last_ms = 0;
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};
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const uint32_t video_stream_id_;
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const uint32_t audio_stream_id_;
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SynchronizationDelays audio_delay_;
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SynchronizationDelays video_delay_;
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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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