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_engine/vie_sync_module.h"
12
13 #include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
14 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
15 #include "webrtc/modules/video_coding/main/interface/video_coding.h"
16 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
17 #include "webrtc/system_wrappers/interface/logging.h"
18 #include "webrtc/system_wrappers/interface/trace_event.h"
19 #include "webrtc/video_engine/stream_synchronization.h"
20 #include "webrtc/video_engine/vie_channel.h"
21 #include "webrtc/voice_engine/include/voe_video_sync.h"
22
23 namespace webrtc {
24
25 enum { kSyncInterval = 1000};
26
UpdateMeasurements(StreamSynchronization::Measurements * stream,const RtpRtcp & rtp_rtcp,const RtpReceiver & receiver)27 int UpdateMeasurements(StreamSynchronization::Measurements* stream,
28 const RtpRtcp& rtp_rtcp, const RtpReceiver& receiver) {
29 if (!receiver.Timestamp(&stream->latest_timestamp))
30 return -1;
31 if (!receiver.LastReceivedTimeMs(&stream->latest_receive_time_ms))
32 return -1;
33
34 uint32_t ntp_secs = 0;
35 uint32_t ntp_frac = 0;
36 uint32_t rtp_timestamp = 0;
37 if (0 != rtp_rtcp.RemoteNTP(&ntp_secs,
38 &ntp_frac,
39 NULL,
40 NULL,
41 &rtp_timestamp)) {
42 return -1;
43 }
44
45 bool new_rtcp_sr = false;
46 if (!UpdateRtcpList(
47 ntp_secs, ntp_frac, rtp_timestamp, &stream->rtcp, &new_rtcp_sr)) {
48 return -1;
49 }
50
51 return 0;
52 }
53
ViESyncModule(VideoCodingModule * vcm,ViEChannel * vie_channel)54 ViESyncModule::ViESyncModule(VideoCodingModule* vcm,
55 ViEChannel* vie_channel)
56 : data_cs_(CriticalSectionWrapper::CreateCriticalSection()),
57 vcm_(vcm),
58 vie_channel_(vie_channel),
59 video_receiver_(NULL),
60 video_rtp_rtcp_(NULL),
61 voe_channel_id_(-1),
62 voe_sync_interface_(NULL),
63 last_sync_time_(TickTime::Now()),
64 sync_() {
65 }
66
~ViESyncModule()67 ViESyncModule::~ViESyncModule() {
68 }
69
ConfigureSync(int voe_channel_id,VoEVideoSync * voe_sync_interface,RtpRtcp * video_rtcp_module,RtpReceiver * video_receiver)70 int ViESyncModule::ConfigureSync(int voe_channel_id,
71 VoEVideoSync* voe_sync_interface,
72 RtpRtcp* video_rtcp_module,
73 RtpReceiver* video_receiver) {
74 CriticalSectionScoped cs(data_cs_.get());
75 voe_channel_id_ = voe_channel_id;
76 voe_sync_interface_ = voe_sync_interface;
77 video_receiver_ = video_receiver;
78 video_rtp_rtcp_ = video_rtcp_module;
79 sync_.reset(new StreamSynchronization(voe_channel_id, vie_channel_->Id()));
80
81 if (!voe_sync_interface) {
82 voe_channel_id_ = -1;
83 if (voe_channel_id >= 0) {
84 // Trying to set a voice channel but no interface exist.
85 return -1;
86 }
87 return 0;
88 }
89 return 0;
90 }
91
VoiceChannel()92 int ViESyncModule::VoiceChannel() {
93 return voe_channel_id_;
94 }
95
TimeUntilNextProcess()96 int32_t ViESyncModule::TimeUntilNextProcess() {
97 return static_cast<int32_t>(kSyncInterval -
98 (TickTime::Now() - last_sync_time_).Milliseconds());
99 }
100
Process()101 int32_t ViESyncModule::Process() {
102 CriticalSectionScoped cs(data_cs_.get());
103 last_sync_time_ = TickTime::Now();
104
105 const int current_video_delay_ms = vcm_->Delay();
106
107 if (voe_channel_id_ == -1) {
108 return 0;
109 }
110 assert(video_rtp_rtcp_ && voe_sync_interface_);
111 assert(sync_.get());
112
113 int audio_jitter_buffer_delay_ms = 0;
114 int playout_buffer_delay_ms = 0;
115 if (voe_sync_interface_->GetDelayEstimate(voe_channel_id_,
116 &audio_jitter_buffer_delay_ms,
117 &playout_buffer_delay_ms) != 0) {
118 return 0;
119 }
120 const int current_audio_delay_ms = audio_jitter_buffer_delay_ms +
121 playout_buffer_delay_ms;
122
123 RtpRtcp* voice_rtp_rtcp = NULL;
124 RtpReceiver* voice_receiver = NULL;
125 if (0 != voe_sync_interface_->GetRtpRtcp(voe_channel_id_, &voice_rtp_rtcp,
126 &voice_receiver)) {
127 return 0;
128 }
129 assert(voice_rtp_rtcp);
130 assert(voice_receiver);
131
132 if (UpdateMeasurements(&video_measurement_, *video_rtp_rtcp_,
133 *video_receiver_) != 0) {
134 return 0;
135 }
136
137 if (UpdateMeasurements(&audio_measurement_, *voice_rtp_rtcp,
138 *voice_receiver) != 0) {
139 return 0;
140 }
141
142 int relative_delay_ms;
143 // Calculate how much later or earlier the audio stream is compared to video.
144 if (!sync_->ComputeRelativeDelay(audio_measurement_, video_measurement_,
145 &relative_delay_ms)) {
146 return 0;
147 }
148
149 TRACE_COUNTER1("webrtc", "SyncCurrentVideoDelay", current_video_delay_ms);
150 TRACE_COUNTER1("webrtc", "SyncCurrentAudioDelay", current_audio_delay_ms);
151 TRACE_COUNTER1("webrtc", "SyncRelativeDelay", relative_delay_ms);
152 int target_audio_delay_ms = 0;
153 int target_video_delay_ms = current_video_delay_ms;
154 // Calculate the necessary extra audio delay and desired total video
155 // delay to get the streams in sync.
156 if (!sync_->ComputeDelays(relative_delay_ms,
157 current_audio_delay_ms,
158 &target_audio_delay_ms,
159 &target_video_delay_ms)) {
160 return 0;
161 }
162
163 if (voe_sync_interface_->SetMinimumPlayoutDelay(
164 voe_channel_id_, target_audio_delay_ms) == -1) {
165 LOG(LS_ERROR) << "Error setting voice delay.";
166 }
167 vcm_->SetMinimumPlayoutDelay(target_video_delay_ms);
168 return 0;
169 }
170
SetTargetBufferingDelay(int target_delay_ms)171 int ViESyncModule::SetTargetBufferingDelay(int target_delay_ms) {
172 CriticalSectionScoped cs(data_cs_.get());
173 if (!voe_sync_interface_) {
174 LOG(LS_ERROR) << "voe_sync_interface_ NULL, can't set playout delay.";
175 return -1;
176 }
177 sync_->SetTargetBufferingDelay(target_delay_ms);
178 // Setting initial playout delay to voice engine (video engine is updated via
179 // the VCM interface).
180 voe_sync_interface_->SetInitialPlayoutDelay(voe_channel_id_,
181 target_delay_ms);
182 return 0;
183 }
184
185 } // namespace webrtc
186