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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 | |
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 | |
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 | |
62 ViESyncModule::~ViESyncModule() { | |
63 } | |
64 | |
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 | |
95 int ViESyncModule::VoiceChannel() { | |
96 return voe_channel_id_; | |
97 } | |
98 | |
99 int64_t ViESyncModule::TimeUntilNextProcess() { | |
100 const int64_t kSyncIntervalMs = 1000; | |
101 return kSyncIntervalMs - (TickTime::Now() - last_sync_time_).Milliseconds(); | |
102 } | |
103 | |
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 | |
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