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 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 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 67 ViESyncModule::~ViESyncModule() { 68 } 69 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 92 int ViESyncModule::VoiceChannel() { 93 return voe_channel_id_; 94 } 95 96 int32_t ViESyncModule::TimeUntilNextProcess() { 97 return static_cast<int32_t>(kSyncInterval - 98 (TickTime::Now() - last_sync_time_).Milliseconds()); 99 } 100 101 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 } 170 171 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
