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* 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.
#include <stdint.h>
#include "system_wrappers/include/rtp_to_ntp_estimator.h"
namespace webrtc {
class StreamSynchronization {
struct Measurements {
Measurements() : latest_receive_time_ms(0), latest_timestamp(0) {}
RtpToNtpEstimator rtp_to_ntp;
int64_t latest_receive_time_ms;
uint32_t latest_timestamp;
StreamSynchronization(int video_stream_id, int audio_stream_id);
bool ComputeDelays(int relative_delay_ms,
int current_audio_delay_ms,
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);
struct SynchronizationDelays {
int extra_video_delay_ms = 0;
int last_video_delay_ms = 0;
int extra_audio_delay_ms = 0;
int last_audio_delay_ms = 0;
SynchronizationDelays channel_delay_;
const int video_stream_id_;
const int audio_stream_id_;
int base_target_delay_ms_;
int avg_diff_ms_;
} // namespace webrtc