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/*
* 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) {
if (!receiver.Timestamp(&stream->latest_timestamp))
return -1;
if (!receiver.LastReceivedTimeMs(&stream->latest_receive_time_ms))
return -1;
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
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