blob: c8f2d2ac1d10e2e7dd77f2496a9c9f0800294201 [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 synchronization audio and video for
// a given voice engine channel and video receive stream.
#include <memory>
#include "webrtc/modules/include/module.h"
#include "webrtc/rtc_base/criticalsection.h"
#include "webrtc/rtc_base/thread_checker.h"
#include "webrtc/video/stream_synchronization.h"
namespace webrtc {
class Syncable;
namespace vcm {
class VideoReceiver;
} // namespace vcm
class RtpStreamsSynchronizer : public Module {
explicit RtpStreamsSynchronizer(Syncable* syncable_video);
void ConfigureSync(Syncable* syncable_audio);
// Implements Module.
int64_t TimeUntilNextProcess() override;
void Process() override;
// Gets the sync offset between the current played out audio frame and the
// video |frame|. Returns true on success, false otherwise.
// The estimated frequency is the frequency used in the RTP to NTP timestamp
// conversion.
bool GetStreamSyncOffsetInMs(uint32_t timestamp,
int64_t render_time_ms,
int64_t* stream_offset_ms,
double* estimated_freq_khz) const;
Syncable* syncable_video_;
rtc::CriticalSection crit_;
Syncable* syncable_audio_ RTC_GUARDED_BY(crit_);
std::unique_ptr<StreamSynchronization> sync_ RTC_GUARDED_BY(crit_);
StreamSynchronization::Measurements audio_measurement_ RTC_GUARDED_BY(crit_);
StreamSynchronization::Measurements video_measurement_ RTC_GUARDED_BY(crit_);
rtc::ThreadChecker process_thread_checker_;
int64_t last_sync_time_ RTC_ACCESS_ON(&process_thread_checker_);
} // namespace webrtc