diff options
Diffstat (limited to 'webrtc/video_engine/vie_sync_module.cc')
| -rw-r--r-- | webrtc/video_engine/vie_sync_module.cc | 188 |
1 files changed, 0 insertions, 188 deletions
diff --git a/webrtc/video_engine/vie_sync_module.cc b/webrtc/video_engine/vie_sync_module.cc deleted file mode 100644 index 1c5d877cd2..0000000000 --- a/webrtc/video_engine/vie_sync_module.cc +++ /dev/null @@ -1,188 +0,0 @@ -/* - * 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/base/logging.h" -#include "webrtc/base/trace_event.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/include/critical_section_wrapper.h" -#include "webrtc/video_engine/stream_synchronization.h" -#include "webrtc/voice_engine/include/voe_video_sync.h" - -namespace webrtc { - -int UpdateMeasurements(StreamSynchronization::Measurements* stream, - const RtpRtcp& rtp_rtcp, const RtpReceiver& receiver) { - if (!receiver.Timestamp(&stream->latest_timestamp)) - return -1; - if (!receiver.LastReceivedTimeMs(&stream->latest_receive_time_ms)) - return -1; - - uint32_t ntp_secs = 0; - uint32_t ntp_frac = 0; - uint32_t rtp_timestamp = 0; - if (0 != rtp_rtcp.RemoteNTP(&ntp_secs, - &ntp_frac, - NULL, - NULL, - &rtp_timestamp)) { - return -1; - } - - bool new_rtcp_sr = false; - if (!UpdateRtcpList( - ntp_secs, ntp_frac, rtp_timestamp, &stream->rtcp, &new_rtcp_sr)) { - return -1; - } - - return 0; -} - -ViESyncModule::ViESyncModule(VideoCodingModule* vcm) - : data_cs_(CriticalSectionWrapper::CreateCriticalSection()), - vcm_(vcm), - 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()); - // Prevent expensive no-ops. - if (voe_channel_id_ == voe_channel_id && - voe_sync_interface_ == voe_sync_interface && - video_receiver_ == video_receiver && - video_rtp_rtcp_ == video_rtcp_module) { - return 0; - } - 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(video_rtp_rtcp_->SSRC(), voe_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_; -} - -int64_t ViESyncModule::TimeUntilNextProcess() { - const int64_t kSyncIntervalMs = 1000; - return kSyncIntervalMs - (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(); - - 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) { - 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; - } - - if (voe_sync_interface_->SetMinimumPlayoutDelay( - voe_channel_id_, target_audio_delay_ms) == -1) { - LOG(LS_ERROR) << "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_) { - LOG(LS_ERROR) << "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 |