rhubarb-lip-sync/rhubarb/lib/webrtc-8d2248ff/webrtc/video/vie_sync_module.cc

195 lines
6.3 KiB
C++
Raw Normal View History

2016-06-21 20:13:05 +00:00
/*
* 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/vie_sync_module.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/timeutils.h"
#include "webrtc/base/trace_event.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/video_coding/video_coding_impl.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/video/stream_synchronization.h"
#include "webrtc/video_frame.h"
#include "webrtc/voice_engine/include/voe_video_sync.h"
namespace webrtc {
namespace {
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 (rtp_rtcp.RemoteNTP(&ntp_secs, &ntp_frac, nullptr, nullptr,
&rtp_timestamp) != 0) {
return -1;
}
bool new_rtcp_sr = false;
if (!UpdateRtcpList(
ntp_secs, ntp_frac, rtp_timestamp, &stream->rtcp, &new_rtcp_sr)) {
return -1;
}
return 0;
}
} // namespace
ViESyncModule::ViESyncModule(vcm::VideoReceiver* video_receiver)
: video_receiver_(video_receiver),
clock_(Clock::GetRealTimeClock()),
rtp_receiver_(nullptr),
video_rtp_rtcp_(nullptr),
voe_channel_id_(-1),
voe_sync_interface_(nullptr),
last_sync_time_(rtc::TimeNanos()),
sync_() {}
ViESyncModule::~ViESyncModule() {
}
void ViESyncModule::ConfigureSync(int voe_channel_id,
VoEVideoSync* voe_sync_interface,
RtpRtcp* video_rtcp_module,
RtpReceiver* rtp_receiver) {
if (voe_channel_id != -1)
RTC_DCHECK(voe_sync_interface);
rtc::CritScope lock(&data_cs_);
// Prevent expensive no-ops.
if (voe_channel_id_ == voe_channel_id &&
voe_sync_interface_ == voe_sync_interface &&
rtp_receiver_ == rtp_receiver && video_rtp_rtcp_ == video_rtcp_module) {
return;
}
voe_channel_id_ = voe_channel_id;
voe_sync_interface_ = voe_sync_interface;
rtp_receiver_ = rtp_receiver;
video_rtp_rtcp_ = video_rtcp_module;
sync_.reset(
new StreamSynchronization(video_rtp_rtcp_->SSRC(), voe_channel_id));
}
int64_t ViESyncModule::TimeUntilNextProcess() {
const int64_t kSyncIntervalMs = 1000;
return kSyncIntervalMs -
(rtc::TimeNanos() - last_sync_time_) / rtc::kNumNanosecsPerMillisec;
}
void ViESyncModule::Process() {
rtc::CritScope lock(&data_cs_);
last_sync_time_ = rtc::TimeNanos();
const int current_video_delay_ms = video_receiver_->Delay();
if (voe_channel_id_ == -1) {
return;
}
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;
}
const int current_audio_delay_ms = audio_jitter_buffer_delay_ms +
playout_buffer_delay_ms;
RtpRtcp* voice_rtp_rtcp = nullptr;
RtpReceiver* voice_receiver = nullptr;
if (voe_sync_interface_->GetRtpRtcp(voe_channel_id_, &voice_rtp_rtcp,
&voice_receiver) != 0) {
return;
}
assert(voice_rtp_rtcp);
assert(voice_receiver);
if (UpdateMeasurements(&video_measurement_, *video_rtp_rtcp_,
*rtp_receiver_) != 0) {
return;
}
if (UpdateMeasurements(&audio_measurement_, *voice_rtp_rtcp,
*voice_receiver) != 0) {
return;
}
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;
}
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;
}
if (voe_sync_interface_->SetMinimumPlayoutDelay(
voe_channel_id_, target_audio_delay_ms) == -1) {
LOG(LS_ERROR) << "Error setting voice delay.";
}
video_receiver_->SetMinimumPlayoutDelay(target_video_delay_ms);
}
bool ViESyncModule::GetStreamSyncOffsetInMs(const VideoFrame& frame,
int64_t* stream_offset_ms) const {
rtc::CritScope lock(&data_cs_);
if (voe_channel_id_ == -1)
return false;
uint32_t playout_timestamp = 0;
if (voe_sync_interface_->GetPlayoutTimestamp(voe_channel_id_,
playout_timestamp) != 0) {
return false;
}
int64_t latest_audio_ntp;
if (!RtpToNtpMs(playout_timestamp, audio_measurement_.rtcp,
&latest_audio_ntp)) {
return false;
}
int64_t latest_video_ntp;
if (!RtpToNtpMs(frame.timestamp(), video_measurement_.rtcp,
&latest_video_ntp)) {
return false;
}
int64_t time_to_render_ms =
frame.render_time_ms() - clock_->TimeInMilliseconds();
if (time_to_render_ms > 0)
latest_video_ntp += time_to_render_ms;
*stream_offset_ms = latest_audio_ntp - latest_video_ntp;
return true;
}
} // namespace webrtc