| /* |
| * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include "video_engine/vie_sync_module.h" |
| |
| #include "modules/rtp_rtcp/interface/rtp_rtcp.h" |
| #include "modules/video_coding/main/interface/video_coding.h" |
| #include "system_wrappers/interface/critical_section_wrapper.h" |
| #include "system_wrappers/interface/trace.h" |
| #include "video_engine/stream_synchronization.h" |
| #include "voice_engine/include/voe_video_sync.h" |
| |
| namespace webrtc { |
| |
| enum { kSyncInterval = 1000}; |
| |
| ViESyncModule::ViESyncModule(const int32_t channel_id, VideoCodingModule* vcm) |
| : data_cs_(CriticalSectionWrapper::CreateCriticalSection()), |
| channel_id_(channel_id), |
| vcm_(vcm), |
| video_rtcp_module_(NULL), |
| voe_channel_id_(-1), |
| voe_sync_interface_(NULL), |
| last_sync_time_(TickTime::Now()), |
| sync_() { |
| } |
| |
| ViESyncModule::~ViESyncModule() { |
| } |
| |
| int ViESyncModule::ConfigureSync(int voe_channel_id, |
| VoEVideoSync* voe_sync_interface, |
| RtpRtcp* video_rtcp_module) { |
| CriticalSectionScoped cs(data_cs_.get()); |
| voe_channel_id_ = voe_channel_id; |
| voe_sync_interface_ = voe_sync_interface; |
| video_rtcp_module_ = video_rtcp_module; |
| sync_.reset(new StreamSynchronization(voe_channel_id, channel_id_)); |
| |
| if (!voe_sync_interface) { |
| voe_channel_id_ = -1; |
| if (voe_channel_id >= 0) { |
| // Trying to set a voice channel but no interface exist. |
| return -1; |
| } |
| return 0; |
| } |
| return 0; |
| } |
| |
| int ViESyncModule::VoiceChannel() { |
| return voe_channel_id_; |
| } |
| |
| WebRtc_Word32 ViESyncModule::TimeUntilNextProcess() { |
| return static_cast<WebRtc_Word32>(kSyncInterval - |
| (TickTime::Now() - last_sync_time_).Milliseconds()); |
| } |
| |
| WebRtc_Word32 ViESyncModule::Process() { |
| CriticalSectionScoped cs(data_cs_.get()); |
| last_sync_time_ = TickTime::Now(); |
| |
| int total_video_delay_target_ms = vcm_->Delay(); |
| WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, channel_id_, |
| "Video delay (JB + decoder) is %d ms", |
| total_video_delay_target_ms); |
| |
| if (voe_channel_id_ == -1) { |
| return 0; |
| } |
| assert(video_rtcp_module_ && voe_sync_interface_); |
| assert(sync_.get()); |
| |
| int current_audio_delay_ms = 0; |
| if (voe_sync_interface_->GetDelayEstimate(voe_channel_id_, |
| current_audio_delay_ms) != 0) { |
| // Could not get VoE delay value, probably not a valid channel Id. |
| WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceVideo, channel_id_, |
| "%s: VE_GetDelayEstimate error for voice_channel %d", |
| __FUNCTION__, total_video_delay_target_ms, voe_channel_id_); |
| return 0; |
| } |
| |
| // VoiceEngine report delay estimates even when not started, ignore if the |
| // reported value is lower than 40 ms. |
| if (current_audio_delay_ms < 40) { |
| WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, channel_id_, |
| "A/V Sync: Audio delay < 40, skipping."); |
| return 0; |
| } |
| |
| RtpRtcp* voice_rtcp_module = NULL; |
| if (0 != voe_sync_interface_->GetRtpRtcp(voe_channel_id_, |
| voice_rtcp_module)) { |
| return 0; |
| } |
| assert(voice_rtcp_module); |
| |
| StreamSynchronization::Measurements video; |
| if (0 != video_rtcp_module_->RemoteNTP(&video.received_ntp_secs, |
| &video.received_ntp_frac, |
| &video.rtcp_arrivaltime_secs, |
| &video.rtcp_arrivaltime_frac)) { |
| // Failed to get video NTP. |
| return 0; |
| } |
| |
| StreamSynchronization::Measurements audio; |
| if (0 != voice_rtcp_module->RemoteNTP(&audio.received_ntp_secs, |
| &audio.received_ntp_frac, |
| &audio.rtcp_arrivaltime_secs, |
| &audio.rtcp_arrivaltime_frac)) { |
| // Failed to get audio NTP. |
| return 0; |
| } |
| int extra_audio_delay_ms = 0; |
| if (sync_->ComputeDelays(audio, current_audio_delay_ms, &extra_audio_delay_ms, |
| video, &total_video_delay_target_ms) != 0) { |
| return 0; |
| } |
| // Set the extra audio delay.synchronization |
| if (voe_sync_interface_->SetMinimumPlayoutDelay( |
| voe_channel_id_, extra_audio_delay_ms) == -1) { |
| WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideo, channel_id_, |
| "Error setting voice delay"); |
| } |
| vcm_->SetMinimumPlayoutDelay(total_video_delay_target_ms); |
| WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, channel_id_, |
| "New Video delay target is: %d", total_video_delay_target_ms); |
| return 0; |
| } |
| |
| } // namespace webrtc |