| /* |
| * 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 "webrtc/video_engine/vie_sync_module.h" |
| |
| #include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h" |
| #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" |
| #include "webrtc/modules/video_coding/main/interface/video_coding.h" |
| #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" |
| #include "webrtc/system_wrappers/interface/trace.h" |
| #include "webrtc/system_wrappers/interface/trace_event.h" |
| #include "webrtc/video_engine/stream_synchronization.h" |
| #include "webrtc/video_engine/vie_channel.h" |
| #include "webrtc/voice_engine/include/voe_video_sync.h" |
| |
| namespace webrtc { |
| |
| enum { kSyncInterval = 1000}; |
| |
| int UpdateMeasurements(StreamSynchronization::Measurements* stream, |
| const RtpRtcp& rtp_rtcp, const RtpReceiver& receiver) { |
| stream->latest_timestamp = receiver.Timestamp(); |
| stream->latest_receive_time_ms = receiver.LastReceivedTimeMs(); |
| synchronization::RtcpMeasurement measurement; |
| if (0 != rtp_rtcp.RemoteNTP(&measurement.ntp_secs, |
| &measurement.ntp_frac, |
| NULL, |
| NULL, |
| &measurement.rtp_timestamp)) { |
| return -1; |
| } |
| if (measurement.ntp_secs == 0 && measurement.ntp_frac == 0) { |
| return -1; |
| } |
| for (synchronization::RtcpList::iterator it = stream->rtcp.begin(); |
| it != stream->rtcp.end(); ++it) { |
| if (measurement.ntp_secs == (*it).ntp_secs && |
| measurement.ntp_frac == (*it).ntp_frac) { |
| // This RTCP has already been added to the list. |
| return 0; |
| } |
| } |
| // We need two RTCP SR reports to map between RTP and NTP. More than two will |
| // not improve the mapping. |
| if (stream->rtcp.size() == 2) { |
| stream->rtcp.pop_back(); |
| } |
| stream->rtcp.push_front(measurement); |
| return 0; |
| } |
| |
| ViESyncModule::ViESyncModule(VideoCodingModule* vcm, |
| ViEChannel* vie_channel) |
| : data_cs_(CriticalSectionWrapper::CreateCriticalSection()), |
| vcm_(vcm), |
| vie_channel_(vie_channel), |
| video_receiver_(NULL), |
| video_rtp_rtcp_(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, |
| RtpReceiver* video_receiver) { |
| CriticalSectionScoped cs(data_cs_.get()); |
| voe_channel_id_ = voe_channel_id; |
| voe_sync_interface_ = voe_sync_interface; |
| video_receiver_ = video_receiver; |
| video_rtp_rtcp_ = video_rtcp_module; |
| sync_.reset(new StreamSynchronization(voe_channel_id, vie_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_; |
| } |
| |
| int32_t ViESyncModule::TimeUntilNextProcess() { |
| return static_cast<int32_t>(kSyncInterval - |
| (TickTime::Now() - last_sync_time_).Milliseconds()); |
| } |
| |
| int32_t ViESyncModule::Process() { |
| CriticalSectionScoped cs(data_cs_.get()); |
| last_sync_time_ = TickTime::Now(); |
| |
| const int current_video_delay_ms = vcm_->Delay(); |
| WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, vie_channel_->Id(), |
| "Video delay (JB + decoder) is %d ms", current_video_delay_ms); |
| |
| if (voe_channel_id_ == -1) { |
| return 0; |
| } |
| assert(video_rtp_rtcp_ && voe_sync_interface_); |
| assert(sync_.get()); |
| |
| int audio_jitter_buffer_delay_ms = 0; |
| int playout_buffer_delay_ms = 0; |
| if (voe_sync_interface_->GetDelayEstimate(voe_channel_id_, |
| &audio_jitter_buffer_delay_ms, |
| &playout_buffer_delay_ms) != 0) { |
| // Could not get VoE delay value, probably not a valid channel Id or |
| // the channel have not received enough packets. |
| WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceVideo, vie_channel_->Id(), |
| "%s: VE_GetDelayEstimate error for voice_channel %d", |
| __FUNCTION__, voe_channel_id_); |
| return 0; |
| } |
| const int current_audio_delay_ms = audio_jitter_buffer_delay_ms + |
| playout_buffer_delay_ms; |
| |
| RtpRtcp* voice_rtp_rtcp = NULL; |
| RtpReceiver* voice_receiver = NULL; |
| if (0 != voe_sync_interface_->GetRtpRtcp(voe_channel_id_, &voice_rtp_rtcp, |
| &voice_receiver)) { |
| return 0; |
| } |
| assert(voice_rtp_rtcp); |
| assert(voice_receiver); |
| |
| if (UpdateMeasurements(&video_measurement_, *video_rtp_rtcp_, |
| *video_receiver_) != 0) { |
| return 0; |
| } |
| |
| if (UpdateMeasurements(&audio_measurement_, *voice_rtp_rtcp, |
| *voice_receiver) != 0) { |
| return 0; |
| } |
| |
| int relative_delay_ms; |
| // Calculate how much later or earlier the audio stream is compared to video. |
| if (!sync_->ComputeRelativeDelay(audio_measurement_, video_measurement_, |
| &relative_delay_ms)) { |
| return 0; |
| } |
| |
| TRACE_COUNTER1("webrtc", "SyncCurrentVideoDelay", current_video_delay_ms); |
| TRACE_COUNTER1("webrtc", "SyncCurrentAudioDelay", current_audio_delay_ms); |
| TRACE_COUNTER1("webrtc", "SyncRelativeDelay", relative_delay_ms); |
| int target_audio_delay_ms = 0; |
| int target_video_delay_ms = current_video_delay_ms; |
| // Calculate the necessary extra audio delay and desired total video |
| // delay to get the streams in sync. |
| if (!sync_->ComputeDelays(relative_delay_ms, |
| current_audio_delay_ms, |
| &target_audio_delay_ms, |
| &target_video_delay_ms)) { |
| return 0; |
| } |
| |
| WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, vie_channel_->Id(), |
| "Set delay current(a=%d v=%d rel=%d) target(a=%d v=%d)", |
| current_audio_delay_ms, current_video_delay_ms, |
| relative_delay_ms, |
| target_audio_delay_ms, target_video_delay_ms); |
| if (voe_sync_interface_->SetMinimumPlayoutDelay( |
| voe_channel_id_, target_audio_delay_ms) == -1) { |
| WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideo, vie_channel_->Id(), |
| "Error setting voice delay"); |
| } |
| vcm_->SetMinimumPlayoutDelay(target_video_delay_ms); |
| return 0; |
| } |
| |
| int ViESyncModule::SetTargetBufferingDelay(int target_delay_ms) { |
| CriticalSectionScoped cs(data_cs_.get()); |
| if (!voe_sync_interface_) { |
| WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, vie_channel_->Id(), |
| "voe_sync_interface_ NULL, can't set playout delay."); |
| return -1; |
| } |
| sync_->SetTargetBufferingDelay(target_delay_ms); |
| // Setting initial playout delay to voice engine (video engine is updated via |
| // the VCM interface). |
| voe_sync_interface_->SetInitialPlayoutDelay(voe_channel_id_, |
| target_delay_ms); |
| return 0; |
| } |
| |
| } // namespace webrtc |
