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 "webrtc/video/vie_sync_module.h"
12
13 #include "webrtc/base/logging.h"
14 #include "webrtc/base/trace_event.h"
15 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
16 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
17 #include "webrtc/modules/video_coding/include/video_coding.h"
18 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
19 #include "webrtc/video/stream_synchronization.h"
20 #include "webrtc/voice_engine/include/voe_video_sync.h"
21
22 namespace webrtc {
23
UpdateMeasurements(StreamSynchronization::Measurements * stream,const RtpRtcp & rtp_rtcp,const RtpReceiver & receiver)24 int UpdateMeasurements(StreamSynchronization::Measurements* stream,
25 const RtpRtcp& rtp_rtcp, const RtpReceiver& receiver) {
26 if (!receiver.Timestamp(&stream->latest_timestamp))
27 return -1;
28 if (!receiver.LastReceivedTimeMs(&stream->latest_receive_time_ms))
29 return -1;
30
31 uint32_t ntp_secs = 0;
32 uint32_t ntp_frac = 0;
33 uint32_t rtp_timestamp = 0;
34 if (0 != rtp_rtcp.RemoteNTP(&ntp_secs,
35 &ntp_frac,
36 NULL,
37 NULL,
38 &rtp_timestamp)) {
39 return -1;
40 }
41
42 bool new_rtcp_sr = false;
43 if (!UpdateRtcpList(
44 ntp_secs, ntp_frac, rtp_timestamp, &stream->rtcp, &new_rtcp_sr)) {
45 return -1;
46 }
47
48 return 0;
49 }
50
ViESyncModule(VideoCodingModule * vcm)51 ViESyncModule::ViESyncModule(VideoCodingModule* vcm)
52 : data_cs_(CriticalSectionWrapper::CreateCriticalSection()),
53 vcm_(vcm),
54 video_receiver_(NULL),
55 video_rtp_rtcp_(NULL),
56 voe_channel_id_(-1),
57 voe_sync_interface_(NULL),
58 last_sync_time_(TickTime::Now()),
59 sync_() {
60 }
61
~ViESyncModule()62 ViESyncModule::~ViESyncModule() {
63 }
64
ConfigureSync(int voe_channel_id,VoEVideoSync * voe_sync_interface,RtpRtcp * video_rtcp_module,RtpReceiver * video_receiver)65 int ViESyncModule::ConfigureSync(int voe_channel_id,
66 VoEVideoSync* voe_sync_interface,
67 RtpRtcp* video_rtcp_module,
68 RtpReceiver* video_receiver) {
69 CriticalSectionScoped cs(data_cs_.get());
70 // Prevent expensive no-ops.
71 if (voe_channel_id_ == voe_channel_id &&
72 voe_sync_interface_ == voe_sync_interface &&
73 video_receiver_ == video_receiver &&
74 video_rtp_rtcp_ == video_rtcp_module) {
75 return 0;
76 }
77 voe_channel_id_ = voe_channel_id;
78 voe_sync_interface_ = voe_sync_interface;
79 video_receiver_ = video_receiver;
80 video_rtp_rtcp_ = video_rtcp_module;
81 sync_.reset(
82 new StreamSynchronization(video_rtp_rtcp_->SSRC(), voe_channel_id));
83
84 if (!voe_sync_interface) {
85 voe_channel_id_ = -1;
86 if (voe_channel_id >= 0) {
87 // Trying to set a voice channel but no interface exist.
88 return -1;
89 }
90 return 0;
91 }
92 return 0;
93 }
94
VoiceChannel()95 int ViESyncModule::VoiceChannel() {
96 return voe_channel_id_;
97 }
98
TimeUntilNextProcess()99 int64_t ViESyncModule::TimeUntilNextProcess() {
100 const int64_t kSyncIntervalMs = 1000;
101 return kSyncIntervalMs - (TickTime::Now() - last_sync_time_).Milliseconds();
102 }
103
Process()104 int32_t ViESyncModule::Process() {
105 CriticalSectionScoped cs(data_cs_.get());
106 last_sync_time_ = TickTime::Now();
107
108 const int current_video_delay_ms = vcm_->Delay();
109
110 if (voe_channel_id_ == -1) {
111 return 0;
112 }
113 assert(video_rtp_rtcp_ && voe_sync_interface_);
114 assert(sync_.get());
115
116 int audio_jitter_buffer_delay_ms = 0;
117 int playout_buffer_delay_ms = 0;
118 if (voe_sync_interface_->GetDelayEstimate(voe_channel_id_,
119 &audio_jitter_buffer_delay_ms,
120 &playout_buffer_delay_ms) != 0) {
121 return 0;
122 }
123 const int current_audio_delay_ms = audio_jitter_buffer_delay_ms +
124 playout_buffer_delay_ms;
125
126 RtpRtcp* voice_rtp_rtcp = NULL;
127 RtpReceiver* voice_receiver = NULL;
128 if (0 != voe_sync_interface_->GetRtpRtcp(voe_channel_id_, &voice_rtp_rtcp,
129 &voice_receiver)) {
130 return 0;
131 }
132 assert(voice_rtp_rtcp);
133 assert(voice_receiver);
134
135 if (UpdateMeasurements(&video_measurement_, *video_rtp_rtcp_,
136 *video_receiver_) != 0) {
137 return 0;
138 }
139
140 if (UpdateMeasurements(&audio_measurement_, *voice_rtp_rtcp,
141 *voice_receiver) != 0) {
142 return 0;
143 }
144
145 int relative_delay_ms;
146 // Calculate how much later or earlier the audio stream is compared to video.
147 if (!sync_->ComputeRelativeDelay(audio_measurement_, video_measurement_,
148 &relative_delay_ms)) {
149 return 0;
150 }
151
152 TRACE_COUNTER1("webrtc", "SyncCurrentVideoDelay", current_video_delay_ms);
153 TRACE_COUNTER1("webrtc", "SyncCurrentAudioDelay", current_audio_delay_ms);
154 TRACE_COUNTER1("webrtc", "SyncRelativeDelay", relative_delay_ms);
155 int target_audio_delay_ms = 0;
156 int target_video_delay_ms = current_video_delay_ms;
157 // Calculate the necessary extra audio delay and desired total video
158 // delay to get the streams in sync.
159 if (!sync_->ComputeDelays(relative_delay_ms,
160 current_audio_delay_ms,
161 &target_audio_delay_ms,
162 &target_video_delay_ms)) {
163 return 0;
164 }
165
166 if (voe_sync_interface_->SetMinimumPlayoutDelay(
167 voe_channel_id_, target_audio_delay_ms) == -1) {
168 LOG(LS_ERROR) << "Error setting voice delay.";
169 }
170 vcm_->SetMinimumPlayoutDelay(target_video_delay_ms);
171 return 0;
172 }
173
174 } // namespace webrtc
175