822fbd8b68
Together with Stefan's http://review.webrtc.org/1960004/. R=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2048004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4556 4adac7df-926f-26a2-2b94-8c16560cd09d
205 lines
7.3 KiB
C++
205 lines
7.3 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/video_engine/vie_sync_module.h"
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#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
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#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
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#include "webrtc/modules/video_coding/main/interface/video_coding.h"
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#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
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#include "webrtc/system_wrappers/interface/trace.h"
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#include "webrtc/system_wrappers/interface/trace_event.h"
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#include "webrtc/video_engine/stream_synchronization.h"
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#include "webrtc/video_engine/vie_channel.h"
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#include "webrtc/voice_engine/include/voe_video_sync.h"
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namespace webrtc {
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enum { kSyncInterval = 1000};
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int UpdateMeasurements(StreamSynchronization::Measurements* stream,
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const RtpRtcp& rtp_rtcp, const RtpReceiver& receiver) {
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stream->latest_timestamp = receiver.Timestamp();
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stream->latest_receive_time_ms = receiver.LastReceivedTimeMs();
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synchronization::RtcpMeasurement measurement;
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if (0 != rtp_rtcp.RemoteNTP(&measurement.ntp_secs,
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&measurement.ntp_frac,
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NULL,
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NULL,
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&measurement.rtp_timestamp)) {
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return -1;
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}
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if (measurement.ntp_secs == 0 && measurement.ntp_frac == 0) {
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return -1;
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}
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for (synchronization::RtcpList::iterator it = stream->rtcp.begin();
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it != stream->rtcp.end(); ++it) {
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if (measurement.ntp_secs == (*it).ntp_secs &&
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measurement.ntp_frac == (*it).ntp_frac) {
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// This RTCP has already been added to the list.
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return 0;
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}
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}
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// We need two RTCP SR reports to map between RTP and NTP. More than two will
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// not improve the mapping.
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if (stream->rtcp.size() == 2) {
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stream->rtcp.pop_back();
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}
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stream->rtcp.push_front(measurement);
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return 0;
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}
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ViESyncModule::ViESyncModule(VideoCodingModule* vcm,
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ViEChannel* vie_channel)
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: data_cs_(CriticalSectionWrapper::CreateCriticalSection()),
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vcm_(vcm),
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vie_channel_(vie_channel),
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video_receiver_(NULL),
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video_rtp_rtcp_(NULL),
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voe_channel_id_(-1),
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voe_sync_interface_(NULL),
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last_sync_time_(TickTime::Now()),
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sync_() {
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}
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ViESyncModule::~ViESyncModule() {
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}
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int ViESyncModule::ConfigureSync(int voe_channel_id,
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VoEVideoSync* voe_sync_interface,
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RtpRtcp* video_rtcp_module,
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RtpReceiver* video_receiver) {
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CriticalSectionScoped cs(data_cs_.get());
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voe_channel_id_ = voe_channel_id;
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voe_sync_interface_ = voe_sync_interface;
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video_receiver_ = video_receiver;
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video_rtp_rtcp_ = video_rtcp_module;
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sync_.reset(new StreamSynchronization(voe_channel_id, vie_channel_->Id()));
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if (!voe_sync_interface) {
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voe_channel_id_ = -1;
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if (voe_channel_id >= 0) {
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// Trying to set a voice channel but no interface exist.
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return -1;
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}
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return 0;
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}
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return 0;
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}
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int ViESyncModule::VoiceChannel() {
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return voe_channel_id_;
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}
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int32_t ViESyncModule::TimeUntilNextProcess() {
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return static_cast<int32_t>(kSyncInterval -
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(TickTime::Now() - last_sync_time_).Milliseconds());
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}
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int32_t ViESyncModule::Process() {
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CriticalSectionScoped cs(data_cs_.get());
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last_sync_time_ = TickTime::Now();
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const int current_video_delay_ms = vcm_->Delay();
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WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, vie_channel_->Id(),
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"Video delay (JB + decoder) is %d ms", current_video_delay_ms);
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if (voe_channel_id_ == -1) {
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return 0;
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}
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assert(video_rtp_rtcp_ && voe_sync_interface_);
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assert(sync_.get());
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int audio_jitter_buffer_delay_ms = 0;
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int playout_buffer_delay_ms = 0;
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if (voe_sync_interface_->GetDelayEstimate(voe_channel_id_,
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&audio_jitter_buffer_delay_ms,
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&playout_buffer_delay_ms) != 0) {
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// Could not get VoE delay value, probably not a valid channel Id or
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// the channel have not received enough packets.
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WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceVideo, vie_channel_->Id(),
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"%s: VE_GetDelayEstimate error for voice_channel %d",
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__FUNCTION__, voe_channel_id_);
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return 0;
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}
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const int current_audio_delay_ms = audio_jitter_buffer_delay_ms +
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playout_buffer_delay_ms;
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RtpRtcp* voice_rtp_rtcp = NULL;
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RtpReceiver* voice_receiver = NULL;
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if (0 != voe_sync_interface_->GetRtpRtcp(voe_channel_id_, &voice_rtp_rtcp,
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&voice_receiver)) {
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return 0;
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}
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assert(voice_rtp_rtcp);
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assert(voice_receiver);
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if (UpdateMeasurements(&video_measurement_, *video_rtp_rtcp_,
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*video_receiver_) != 0) {
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return 0;
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}
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if (UpdateMeasurements(&audio_measurement_, *voice_rtp_rtcp,
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*voice_receiver) != 0) {
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return 0;
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}
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int relative_delay_ms;
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// Calculate how much later or earlier the audio stream is compared to video.
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if (!sync_->ComputeRelativeDelay(audio_measurement_, video_measurement_,
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&relative_delay_ms)) {
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return 0;
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}
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TRACE_COUNTER1("webrtc", "SyncCurrentVideoDelay", current_video_delay_ms);
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TRACE_COUNTER1("webrtc", "SyncCurrentAudioDelay", current_audio_delay_ms);
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TRACE_COUNTER1("webrtc", "SyncRelativeDelay", relative_delay_ms);
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int target_audio_delay_ms = 0;
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int target_video_delay_ms = current_video_delay_ms;
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// Calculate the necessary extra audio delay and desired total video
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// delay to get the streams in sync.
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if (!sync_->ComputeDelays(relative_delay_ms,
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current_audio_delay_ms,
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&target_audio_delay_ms,
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&target_video_delay_ms)) {
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return 0;
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}
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WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, vie_channel_->Id(),
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"Set delay current(a=%d v=%d rel=%d) target(a=%d v=%d)",
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current_audio_delay_ms, current_video_delay_ms,
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relative_delay_ms,
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target_audio_delay_ms, target_video_delay_ms);
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if (voe_sync_interface_->SetMinimumPlayoutDelay(
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voe_channel_id_, target_audio_delay_ms) == -1) {
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WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideo, vie_channel_->Id(),
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"Error setting voice delay");
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}
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vcm_->SetMinimumPlayoutDelay(target_video_delay_ms);
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return 0;
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}
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int ViESyncModule::SetTargetBufferingDelay(int target_delay_ms) {
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CriticalSectionScoped cs(data_cs_.get());
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if (!voe_sync_interface_) {
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WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, vie_channel_->Id(),
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"voe_sync_interface_ NULL, can't set playout delay.");
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return -1;
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}
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sync_->SetTargetBufferingDelay(target_delay_ms);
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// Setting initial playout delay to voice engine (video engine is updated via
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// the VCM interface).
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voe_sync_interface_->SetInitialPlayoutDelay(voe_channel_id_,
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target_delay_ms);
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return 0;
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}
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} // namespace webrtc
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