1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11 #include "video/rtp_streams_synchronizer.h"
12
13 #include "absl/types/optional.h"
14 #include "call/syncable.h"
15 #include "rtc_base/checks.h"
16 #include "rtc_base/logging.h"
17 #include "rtc_base/time_utils.h"
18 #include "rtc_base/trace_event.h"
19 #include "system_wrappers/include/rtp_to_ntp_estimator.h"
20
21 namespace webrtc {
22 namespace {
23 // Time interval for logging stats.
24 constexpr int64_t kStatsLogIntervalMs = 10000;
25
UpdateMeasurements(StreamSynchronization::Measurements * stream,const Syncable::Info & info)26 bool UpdateMeasurements(StreamSynchronization::Measurements* stream,
27 const Syncable::Info& info) {
28 RTC_DCHECK(stream);
29 stream->latest_timestamp = info.latest_received_capture_timestamp;
30 stream->latest_receive_time_ms = info.latest_receive_time_ms;
31 bool new_rtcp_sr = false;
32 if (!stream->rtp_to_ntp.UpdateMeasurements(
33 info.capture_time_ntp_secs, info.capture_time_ntp_frac,
34 info.capture_time_source_clock, &new_rtcp_sr)) {
35 return false;
36 }
37 return true;
38 }
39 } // namespace
40
RtpStreamsSynchronizer(Syncable * syncable_video)41 RtpStreamsSynchronizer::RtpStreamsSynchronizer(Syncable* syncable_video)
42 : syncable_video_(syncable_video),
43 syncable_audio_(nullptr),
44 sync_(),
45 last_sync_time_(rtc::TimeNanos()),
46 last_stats_log_ms_(rtc::TimeMillis()) {
47 RTC_DCHECK(syncable_video);
48 process_thread_checker_.Detach();
49 }
50
51 RtpStreamsSynchronizer::~RtpStreamsSynchronizer() = default;
52
ConfigureSync(Syncable * syncable_audio)53 void RtpStreamsSynchronizer::ConfigureSync(Syncable* syncable_audio) {
54 MutexLock lock(&mutex_);
55 if (syncable_audio == syncable_audio_) {
56 // This prevents expensive no-ops.
57 return;
58 }
59
60 syncable_audio_ = syncable_audio;
61 sync_.reset(nullptr);
62 if (syncable_audio_) {
63 sync_.reset(new StreamSynchronization(syncable_video_->id(),
64 syncable_audio_->id()));
65 }
66 }
67
TimeUntilNextProcess()68 int64_t RtpStreamsSynchronizer::TimeUntilNextProcess() {
69 RTC_DCHECK_RUN_ON(&process_thread_checker_);
70 const int64_t kSyncIntervalMs = 1000;
71 return kSyncIntervalMs -
72 (rtc::TimeNanos() - last_sync_time_) / rtc::kNumNanosecsPerMillisec;
73 }
74
Process()75 void RtpStreamsSynchronizer::Process() {
76 RTC_DCHECK_RUN_ON(&process_thread_checker_);
77 last_sync_time_ = rtc::TimeNanos();
78
79 MutexLock lock(&mutex_);
80 if (!syncable_audio_) {
81 return;
82 }
83 RTC_DCHECK(sync_.get());
84
85 bool log_stats = false;
86 const int64_t now_ms = rtc::TimeMillis();
87 if (now_ms - last_stats_log_ms_ > kStatsLogIntervalMs) {
88 last_stats_log_ms_ = now_ms;
89 log_stats = true;
90 }
91
92 int64_t last_audio_receive_time_ms =
93 audio_measurement_.latest_receive_time_ms;
94 absl::optional<Syncable::Info> audio_info = syncable_audio_->GetInfo();
95 if (!audio_info || !UpdateMeasurements(&audio_measurement_, *audio_info)) {
96 return;
97 }
98
99 if (last_audio_receive_time_ms == audio_measurement_.latest_receive_time_ms) {
100 // No new audio packet has been received since last update.
101 return;
102 }
103
104 int64_t last_video_receive_ms = video_measurement_.latest_receive_time_ms;
105 absl::optional<Syncable::Info> video_info = syncable_video_->GetInfo();
106 if (!video_info || !UpdateMeasurements(&video_measurement_, *video_info)) {
107 return;
108 }
109
110 if (last_video_receive_ms == video_measurement_.latest_receive_time_ms) {
111 // No new video packet has been received since last update.
112 return;
113 }
114
115 int relative_delay_ms;
116 // Calculate how much later or earlier the audio stream is compared to video.
117 if (!sync_->ComputeRelativeDelay(audio_measurement_, video_measurement_,
118 &relative_delay_ms)) {
119 return;
120 }
121
122 if (log_stats) {
123 RTC_LOG(LS_INFO) << "Sync info stats: " << now_ms
124 << ", {ssrc: " << sync_->audio_stream_id() << ", "
125 << "cur_delay_ms: " << audio_info->current_delay_ms
126 << "} {ssrc: " << sync_->video_stream_id() << ", "
127 << "cur_delay_ms: " << video_info->current_delay_ms
128 << "} {relative_delay_ms: " << relative_delay_ms << "} ";
129 }
130
131 TRACE_COUNTER1("webrtc", "SyncCurrentVideoDelay",
132 video_info->current_delay_ms);
133 TRACE_COUNTER1("webrtc", "SyncCurrentAudioDelay",
134 audio_info->current_delay_ms);
135 TRACE_COUNTER1("webrtc", "SyncRelativeDelay", relative_delay_ms);
136
137 int target_audio_delay_ms = 0;
138 int target_video_delay_ms = video_info->current_delay_ms;
139 // Calculate the necessary extra audio delay and desired total video
140 // delay to get the streams in sync.
141 if (!sync_->ComputeDelays(relative_delay_ms, audio_info->current_delay_ms,
142 &target_audio_delay_ms, &target_video_delay_ms)) {
143 return;
144 }
145
146 if (log_stats) {
147 RTC_LOG(LS_INFO) << "Sync delay stats: " << now_ms
148 << ", {ssrc: " << sync_->audio_stream_id() << ", "
149 << "target_delay_ms: " << target_audio_delay_ms
150 << "} {ssrc: " << sync_->video_stream_id() << ", "
151 << "target_delay_ms: " << target_video_delay_ms << "} ";
152 }
153
154 syncable_audio_->SetMinimumPlayoutDelay(target_audio_delay_ms);
155 syncable_video_->SetMinimumPlayoutDelay(target_video_delay_ms);
156 }
157
158 // TODO(https://bugs.webrtc.org/7065): Move RtpToNtpEstimator out of
159 // RtpStreamsSynchronizer and into respective receive stream to always populate
160 // the estimated playout timestamp.
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) const161 bool RtpStreamsSynchronizer::GetStreamSyncOffsetInMs(
162 uint32_t rtp_timestamp,
163 int64_t render_time_ms,
164 int64_t* video_playout_ntp_ms,
165 int64_t* stream_offset_ms,
166 double* estimated_freq_khz) const {
167 MutexLock lock(&mutex_);
168 if (!syncable_audio_) {
169 return false;
170 }
171
172 uint32_t audio_rtp_timestamp;
173 int64_t time_ms;
174 if (!syncable_audio_->GetPlayoutRtpTimestamp(&audio_rtp_timestamp,
175 &time_ms)) {
176 return false;
177 }
178
179 int64_t latest_audio_ntp;
180 if (!audio_measurement_.rtp_to_ntp.Estimate(audio_rtp_timestamp,
181 &latest_audio_ntp)) {
182 return false;
183 }
184
185 syncable_audio_->SetEstimatedPlayoutNtpTimestampMs(latest_audio_ntp, time_ms);
186
187 int64_t latest_video_ntp;
188 if (!video_measurement_.rtp_to_ntp.Estimate(rtp_timestamp,
189 &latest_video_ntp)) {
190 return false;
191 }
192
193 // Current audio ntp.
194 int64_t now_ms = rtc::TimeMillis();
195 latest_audio_ntp += (now_ms - time_ms);
196
197 // Remove video playout delay.
198 int64_t time_to_render_ms = render_time_ms - now_ms;
199 if (time_to_render_ms > 0)
200 latest_video_ntp -= time_to_render_ms;
201
202 *video_playout_ntp_ms = latest_video_ntp;
203 *stream_offset_ms = latest_audio_ntp - latest_video_ntp;
204 *estimated_freq_khz = video_measurement_.rtp_to_ntp.params()->frequency_khz;
205 return true;
206 }
207
208 } // namespace webrtc
209