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/*
* Copyright (c) 2020 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_synchronizer2.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 internal {
namespace {
// Time interval for logging stats.
constexpr int64_t kStatsLogIntervalMs = 10000;
constexpr TimeDelta kSyncInterval = TimeDelta::Millis(1000);
bool UpdateMeasurements(StreamSynchronization::Measurements* stream,
const Syncable::Info& info) {
stream->latest_timestamp = info.latest_received_capture_timestamp;
stream->latest_receive_time_ms = info.latest_receive_time_ms;
return stream->rtp_to_ntp.UpdateMeasurements(
NtpTime(info.capture_time_ntp_secs, info.capture_time_ntp_frac),
info.capture_time_source_clock) !=
RtpToNtpEstimator::kInvalidMeasurement;
}
} // namespace
RtpStreamsSynchronizer::RtpStreamsSynchronizer(TaskQueueBase* main_queue,
Syncable* syncable_video)
: task_queue_(main_queue),
syncable_video_(syncable_video),
last_stats_log_ms_(rtc::TimeMillis()) {
RTC_DCHECK(syncable_video);
}
RtpStreamsSynchronizer::~RtpStreamsSynchronizer() {
RTC_DCHECK_RUN_ON(&main_checker_);
repeating_task_.Stop();
}
void RtpStreamsSynchronizer::ConfigureSync(Syncable* syncable_audio) {
RTC_DCHECK_RUN_ON(&main_checker_);
// Prevent expensive no-ops.
if (syncable_audio == syncable_audio_)
return;
syncable_audio_ = syncable_audio;
sync_.reset(nullptr);
if (!syncable_audio_) {
repeating_task_.Stop();
return;
}
sync_.reset(
new StreamSynchronization(syncable_video_->id(), syncable_audio_->id()));
if (repeating_task_.Running())
return;
repeating_task_ =
RepeatingTaskHandle::DelayedStart(task_queue_, kSyncInterval, [this]() {
UpdateDelay();
return kSyncInterval;
});
}
void RtpStreamsSynchronizer::UpdateDelay() {
RTC_DCHECK_RUN_ON(&main_checker_);
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;
}
int64_t last_audio_receive_time_ms =
audio_measurement_.latest_receive_time_ms;
absl::optional<Syncable::Info> audio_info = syncable_audio_->GetInfo();
if (!audio_info || !UpdateMeasurements(&audio_measurement_, *audio_info)) {
return;
}
if (last_audio_receive_time_ms == audio_measurement_.latest_receive_time_ms) {
// No new audio packet has been received since last update.
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 << "} ";
}
if (!syncable_audio_->SetMinimumPlayoutDelay(target_audio_delay_ms)) {
sync_->ReduceAudioDelay();
}
if (!syncable_video_->SetMinimumPlayoutDelay(target_video_delay_ms)) {
sync_->ReduceVideoDelay();
}
}
// 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 {
RTC_DCHECK_RUN_ON(&main_checker_);
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;
}
NtpTime latest_audio_ntp =
audio_measurement_.rtp_to_ntp.Estimate(audio_rtp_timestamp);
if (!latest_audio_ntp.Valid()) {
return false;
}
int64_t latest_audio_ntp_ms = latest_audio_ntp.ToMs();
syncable_audio_->SetEstimatedPlayoutNtpTimestampMs(latest_audio_ntp_ms,
time_ms);
NtpTime latest_video_ntp =
video_measurement_.rtp_to_ntp.Estimate(rtp_timestamp);
if (!latest_video_ntp.Valid()) {
return false;
}
int64_t latest_video_ntp_ms = latest_video_ntp.ToMs();
// Current audio ntp.
int64_t now_ms = rtc::TimeMillis();
latest_audio_ntp_ms += (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_ms -= time_to_render_ms;
*video_playout_ntp_ms = latest_video_ntp_ms;
*stream_offset_ms = latest_audio_ntp_ms - latest_video_ntp_ms;
*estimated_freq_khz = video_measurement_.rtp_to_ntp.EstimatedFrequencyKhz();
return true;
}
} // namespace internal
} // namespace webrtc
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