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 "call/syncable.h"
14 #include "modules/video_coding/video_coding_impl.h"
15 #include "rtc_base/checks.h"
16 #include "rtc_base/logging.h"
17 #include "rtc_base/timeutils.h"
18 #include "rtc_base/trace_event.h"
19
20 namespace webrtc {
21 namespace {
UpdateMeasurements(StreamSynchronization::Measurements * stream,const Syncable::Info & info)22 bool UpdateMeasurements(StreamSynchronization::Measurements* stream,
23 const Syncable::Info& info) {
24 RTC_DCHECK(stream);
25 stream->latest_timestamp = info.latest_received_capture_timestamp;
26 stream->latest_receive_time_ms = info.latest_receive_time_ms;
27 bool new_rtcp_sr = false;
28 if (!stream->rtp_to_ntp.UpdateMeasurements(info.capture_time_ntp_secs,
29 info.capture_time_ntp_frac,
30 info.capture_time_source_clock,
31 &new_rtcp_sr)) {
32 return false;
33 }
34 return true;
35 }
36 } // namespace
37
RtpStreamsSynchronizer(Syncable * syncable_video)38 RtpStreamsSynchronizer::RtpStreamsSynchronizer(Syncable* syncable_video)
39 : syncable_video_(syncable_video),
40 syncable_audio_(nullptr),
41 sync_(),
42 last_sync_time_(rtc::TimeNanos()) {
43 RTC_DCHECK(syncable_video);
44 process_thread_checker_.DetachFromThread();
45 }
46
ConfigureSync(Syncable * syncable_audio)47 void RtpStreamsSynchronizer::ConfigureSync(Syncable* syncable_audio) {
48 rtc::CritScope lock(&crit_);
49 if (syncable_audio == syncable_audio_) {
50 // This prevents expensive no-ops.
51 return;
52 }
53
54 syncable_audio_ = syncable_audio;
55 sync_.reset(nullptr);
56 if (syncable_audio_) {
57 sync_.reset(new StreamSynchronization(syncable_video_->id(),
58 syncable_audio_->id()));
59 }
60 }
61
TimeUntilNextProcess()62 int64_t RtpStreamsSynchronizer::TimeUntilNextProcess() {
63 RTC_DCHECK_RUN_ON(&process_thread_checker_);
64 const int64_t kSyncIntervalMs = 1000;
65 return kSyncIntervalMs -
66 (rtc::TimeNanos() - last_sync_time_) / rtc::kNumNanosecsPerMillisec;
67 }
68
Process()69 void RtpStreamsSynchronizer::Process() {
70 RTC_DCHECK_RUN_ON(&process_thread_checker_);
71 last_sync_time_ = rtc::TimeNanos();
72
73 rtc::CritScope lock(&crit_);
74 if (!syncable_audio_) {
75 return;
76 }
77 RTC_DCHECK(sync_.get());
78
79 rtc::Optional<Syncable::Info> audio_info = syncable_audio_->GetInfo();
80 if (!audio_info || !UpdateMeasurements(&audio_measurement_, *audio_info)) {
81 return;
82 }
83
84 int64_t last_video_receive_ms = video_measurement_.latest_receive_time_ms;
85 rtc::Optional<Syncable::Info> video_info = syncable_video_->GetInfo();
86 if (!video_info || !UpdateMeasurements(&video_measurement_, *video_info)) {
87 return;
88 }
89
90 if (last_video_receive_ms == video_measurement_.latest_receive_time_ms) {
91 // No new video packet has been received since last update.
92 return;
93 }
94
95 int relative_delay_ms;
96 // Calculate how much later or earlier the audio stream is compared to video.
97 if (!sync_->ComputeRelativeDelay(audio_measurement_, video_measurement_,
98 &relative_delay_ms)) {
99 return;
100 }
101
102 TRACE_COUNTER1("webrtc", "SyncCurrentVideoDelay",
103 video_info->current_delay_ms);
104 TRACE_COUNTER1("webrtc", "SyncCurrentAudioDelay",
105 audio_info->current_delay_ms);
106 TRACE_COUNTER1("webrtc", "SyncRelativeDelay", relative_delay_ms);
107 int target_audio_delay_ms = 0;
108 int target_video_delay_ms = video_info->current_delay_ms;
109 // Calculate the necessary extra audio delay and desired total video
110 // delay to get the streams in sync.
111 if (!sync_->ComputeDelays(relative_delay_ms,
112 audio_info->current_delay_ms,
113 &target_audio_delay_ms,
114 &target_video_delay_ms)) {
115 return;
116 }
117
118 syncable_audio_->SetMinimumPlayoutDelay(target_audio_delay_ms);
119 syncable_video_->SetMinimumPlayoutDelay(target_video_delay_ms);
120 }
121
GetStreamSyncOffsetInMs(uint32_t timestamp,int64_t render_time_ms,int64_t * stream_offset_ms,double * estimated_freq_khz) const122 bool RtpStreamsSynchronizer::GetStreamSyncOffsetInMs(
123 uint32_t timestamp,
124 int64_t render_time_ms,
125 int64_t* stream_offset_ms,
126 double* estimated_freq_khz) const {
127 rtc::CritScope lock(&crit_);
128 if (!syncable_audio_) {
129 return false;
130 }
131
132 uint32_t playout_timestamp = syncable_audio_->GetPlayoutTimestamp();
133
134 int64_t latest_audio_ntp;
135 if (!audio_measurement_.rtp_to_ntp.Estimate(playout_timestamp,
136 &latest_audio_ntp)) {
137 return false;
138 }
139
140 int64_t latest_video_ntp;
141 if (!video_measurement_.rtp_to_ntp.Estimate(timestamp, &latest_video_ntp)) {
142 return false;
143 }
144
145 int64_t time_to_render_ms = render_time_ms - rtc::TimeMillis();
146 if (time_to_render_ms > 0)
147 latest_video_ntp += time_to_render_ms;
148
149 *stream_offset_ms = latest_audio_ntp - latest_video_ntp;
150 *estimated_freq_khz = video_measurement_.rtp_to_ntp.params()->frequency_khz;
151 return true;
152 }
153
154 } // namespace webrtc
155