blob: 28e9a0ba9d02b5f4af6d576876bc293908246b23 [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.
*/
#include "video/rtp_streams_synchronizer.h"
#include "absl/types/optional.h"
#include "call/syncable.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/time_utils.h"
#include "rtc_base/trace_event.h"
#include "system_wrappers/include/rtp_to_ntp_estimator.h"
namespace webrtc {
namespace {
// Time interval for logging stats.
constexpr int64_t kStatsLogIntervalMs = 10000;
bool UpdateMeasurements(StreamSynchronization::Measurements* stream,
const Syncable::Info& info) {
RTC_DCHECK(stream);
stream->latest_timestamp = info.latest_received_capture_timestamp;
stream->latest_receive_time_ms = info.latest_receive_time_ms;
bool new_rtcp_sr = false;
if (!stream->rtp_to_ntp.UpdateMeasurements(
info.capture_time_ntp_secs, info.capture_time_ntp_frac,
info.capture_time_source_clock, &new_rtcp_sr)) {
return false;
}
return true;
}
} // namespace
RtpStreamsSynchronizer::RtpStreamsSynchronizer(Syncable* syncable_video)
: syncable_video_(syncable_video),
syncable_audio_(nullptr),
sync_(),
last_sync_time_(rtc::TimeNanos()),
last_stats_log_ms_(rtc::TimeMillis()) {
RTC_DCHECK(syncable_video);
process_thread_checker_.Detach();
}
RtpStreamsSynchronizer::~RtpStreamsSynchronizer() = default;
void RtpStreamsSynchronizer::ConfigureSync(Syncable* syncable_audio) {
MutexLock lock(&mutex_);
if (syncable_audio == syncable_audio_) {
// This prevents expensive no-ops.
return;
}
syncable_audio_ = syncable_audio;
sync_.reset(nullptr);
if (syncable_audio_) {
sync_.reset(new StreamSynchronization(syncable_video_->id(),
syncable_audio_->id()));
}
}
int64_t RtpStreamsSynchronizer::TimeUntilNextProcess() {
RTC_DCHECK_RUN_ON(&process_thread_checker_);
const int64_t kSyncIntervalMs = 1000;
return kSyncIntervalMs -
(rtc::TimeNanos() - last_sync_time_) / rtc::kNumNanosecsPerMillisec;
}
void RtpStreamsSynchronizer::Process() {
RTC_DCHECK_RUN_ON(&process_thread_checker_);
last_sync_time_ = rtc::TimeNanos();
MutexLock lock(&mutex_);
if (!syncable_audio_) {
return;
}
RTC_DCHECK(sync_.get());
bool log_stats = false;
const int64_t now_ms = rtc::TimeMillis();
if (now_ms - last_stats_log_ms_ > kStatsLogIntervalMs) {
last_stats_log_ms_ = now_ms;
log_stats = true;
}
absl::optional<Syncable::Info> audio_info = syncable_audio_->GetInfo();
if (!audio_info || !UpdateMeasurements(&audio_measurement_, *audio_info)) {
return;
}
int64_t last_video_receive_ms = video_measurement_.latest_receive_time_ms;
absl::optional<Syncable::Info> video_info = syncable_video_->GetInfo();
if (!video_info || !UpdateMeasurements(&video_measurement_, *video_info)) {
return;
}
if (last_video_receive_ms == video_measurement_.latest_receive_time_ms) {
// No new video packet has been received since last update.
return;
}
int relative_delay_ms;
// Calculate how much later or earlier the audio stream is compared to video.
if (!sync_->ComputeRelativeDelay(audio_measurement_, video_measurement_,
&relative_delay_ms)) {
return;
}
if (log_stats) {
RTC_LOG(LS_INFO) << "Sync info stats: " << now_ms
<< ", {ssrc: " << sync_->audio_stream_id() << ", "
<< "cur_delay_ms: " << audio_info->current_delay_ms
<< "} {ssrc: " << sync_->video_stream_id() << ", "
<< "cur_delay_ms: " << video_info->current_delay_ms
<< "} {relative_delay_ms: " << relative_delay_ms << "} ";
}
TRACE_COUNTER1("webrtc", "SyncCurrentVideoDelay",
video_info->current_delay_ms);
TRACE_COUNTER1("webrtc", "SyncCurrentAudioDelay",
audio_info->current_delay_ms);
TRACE_COUNTER1("webrtc", "SyncRelativeDelay", relative_delay_ms);
int target_audio_delay_ms = 0;
int target_video_delay_ms = video_info->current_delay_ms;
// Calculate the necessary extra audio delay and desired total video
// delay to get the streams in sync.
if (!sync_->ComputeDelays(relative_delay_ms, audio_info->current_delay_ms,
&target_audio_delay_ms, &target_video_delay_ms)) {
return;
}
if (log_stats) {
RTC_LOG(LS_INFO) << "Sync delay stats: " << now_ms
<< ", {ssrc: " << sync_->audio_stream_id() << ", "
<< "target_delay_ms: " << target_audio_delay_ms
<< "} {ssrc: " << sync_->video_stream_id() << ", "
<< "target_delay_ms: " << target_video_delay_ms << "} ";
}
syncable_audio_->SetMinimumPlayoutDelay(target_audio_delay_ms);
syncable_video_->SetMinimumPlayoutDelay(target_video_delay_ms);
}
// TODO(https://bugs.webrtc.org/7065): Move RtpToNtpEstimator out of
// RtpStreamsSynchronizer and into respective receive stream to always populate
// the estimated playout timestamp.
bool RtpStreamsSynchronizer::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 {
MutexLock lock(&mutex_);
if (!syncable_audio_) {
return false;
}
uint32_t audio_rtp_timestamp;
int64_t time_ms;
if (!syncable_audio_->GetPlayoutRtpTimestamp(&audio_rtp_timestamp,
&time_ms)) {
return false;
}
int64_t latest_audio_ntp;
if (!audio_measurement_.rtp_to_ntp.Estimate(audio_rtp_timestamp,
&latest_audio_ntp)) {
return false;
}
syncable_audio_->SetEstimatedPlayoutNtpTimestampMs(latest_audio_ntp, time_ms);
int64_t latest_video_ntp;
if (!video_measurement_.rtp_to_ntp.Estimate(rtp_timestamp,
&latest_video_ntp)) {
return false;
}
// Current audio ntp.
int64_t now_ms = rtc::TimeMillis();
latest_audio_ntp += (now_ms - time_ms);
// Remove video playout delay.
int64_t time_to_render_ms = render_time_ms - now_ms;
if (time_to_render_ms > 0)
latest_video_ntp -= time_to_render_ms;
*video_playout_ntp_ms = latest_video_ntp;
*stream_offset_ms = latest_audio_ntp - latest_video_ntp;
*estimated_freq_khz = video_measurement_.rtp_to_ntp.params()->frequency_khz;
return true;
}
} // namespace webrtc