blob: 574ccba70b8884103e06e1052ff339c32ae82262 [file] [log] [blame]
* 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.
// RtpStreamsSynchronizer is responsible for synchronizing audio and video for
// a given audio receive stream and video receive stream.
#include <memory>
#include "api/sequence_checker.h"
#include "modules/include/module.h"
#include "rtc_base/synchronization/mutex.h"
#include "video/stream_synchronization.h"
namespace webrtc {
class Syncable;
class RtpStreamsSynchronizer : public Module {
explicit RtpStreamsSynchronizer(Syncable* syncable_video);
~RtpStreamsSynchronizer() override;
void ConfigureSync(Syncable* syncable_audio);
// Implements Module.
int64_t TimeUntilNextProcess() override;
void Process() override;
// Gets the estimated playout NTP timestamp for the video frame with
// |rtp_timestamp| and the sync offset between the current played out audio
// frame and the video frame. Returns true on success, false otherwise.
// The |estimated_freq_khz| is the frequency used in the RTP to NTP timestamp
// conversion.
bool GetStreamSyncOffsetInMs(uint32_t rtp_timestamp,
int64_t render_time_ms,
int64_t* video_playout_ntp_ms,
int64_t* stream_offset_ms,
double* estimated_freq_khz) const;
Syncable* syncable_video_;
mutable Mutex mutex_;
Syncable* syncable_audio_ RTC_GUARDED_BY(mutex_);
std::unique_ptr<StreamSynchronization> sync_ RTC_GUARDED_BY(mutex_);
StreamSynchronization::Measurements audio_measurement_ RTC_GUARDED_BY(mutex_);
StreamSynchronization::Measurements video_measurement_ RTC_GUARDED_BY(mutex_);
SequenceChecker process_thread_checker_;
int64_t last_sync_time_ RTC_GUARDED_BY(&process_thread_checker_);
int64_t last_stats_log_ms_ RTC_GUARDED_BY(&process_thread_checker_);
} // namespace webrtc