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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/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
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