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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/vie_remb.h"
#include <assert.h>
#include <algorithm>
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/utility/include/process_thread.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/tick_util.h"
#include "webrtc/system_wrappers/include/trace.h"
namespace webrtc {
const int kRembSendIntervalMs = 200;
// % threshold for if we should send a new REMB asap.
const unsigned int kSendThresholdPercent = 97;
VieRemb::VieRemb(Clock* clock)
: clock_(clock),
list_crit_(CriticalSectionWrapper::CreateCriticalSection()),
last_remb_time_(clock_->TimeInMilliseconds()),
last_send_bitrate_(0),
bitrate_(0) {}
VieRemb::~VieRemb() {}
void VieRemb::AddReceiveChannel(RtpRtcp* rtp_rtcp) {
assert(rtp_rtcp);
CriticalSectionScoped cs(list_crit_.get());
if (std::find(receive_modules_.begin(), receive_modules_.end(), rtp_rtcp) !=
receive_modules_.end())
return;
// The module probably doesn't have a remote SSRC yet, so don't add it to the
// map.
receive_modules_.push_back(rtp_rtcp);
}
void VieRemb::RemoveReceiveChannel(RtpRtcp* rtp_rtcp) {
assert(rtp_rtcp);
CriticalSectionScoped cs(list_crit_.get());
for (RtpModules::iterator it = receive_modules_.begin();
it != receive_modules_.end(); ++it) {
if ((*it) == rtp_rtcp) {
receive_modules_.erase(it);
break;
}
}
}
void VieRemb::AddRembSender(RtpRtcp* rtp_rtcp) {
assert(rtp_rtcp);
CriticalSectionScoped cs(list_crit_.get());
// Verify this module hasn't been added earlier.
if (std::find(rtcp_sender_.begin(), rtcp_sender_.end(), rtp_rtcp) !=
rtcp_sender_.end())
return;
rtcp_sender_.push_back(rtp_rtcp);
}
void VieRemb::RemoveRembSender(RtpRtcp* rtp_rtcp) {
assert(rtp_rtcp);
CriticalSectionScoped cs(list_crit_.get());
for (RtpModules::iterator it = rtcp_sender_.begin();
it != rtcp_sender_.end(); ++it) {
if ((*it) == rtp_rtcp) {
rtcp_sender_.erase(it);
return;
}
}
}
bool VieRemb::InUse() const {
CriticalSectionScoped cs(list_crit_.get());
if (receive_modules_.empty() && rtcp_sender_.empty())
return false;
else
return true;
}
void VieRemb::OnReceiveBitrateChanged(const std::vector<unsigned int>& ssrcs,
unsigned int bitrate) {
list_crit_->Enter();
// If we already have an estimate, check if the new total estimate is below
// kSendThresholdPercent of the previous estimate.
if (last_send_bitrate_ > 0) {
unsigned int new_remb_bitrate = last_send_bitrate_ - bitrate_ + bitrate;
if (new_remb_bitrate < kSendThresholdPercent * last_send_bitrate_ / 100) {
// The new bitrate estimate is less than kSendThresholdPercent % of the
// last report. Send a REMB asap.
last_remb_time_ = clock_->TimeInMilliseconds() - kRembSendIntervalMs;
}
}
bitrate_ = bitrate;
// Calculate total receive bitrate estimate.
int64_t now = clock_->TimeInMilliseconds();
if (now - last_remb_time_ < kRembSendIntervalMs) {
list_crit_->Leave();
return;
}
last_remb_time_ = now;
if (ssrcs.empty() || receive_modules_.empty()) {
list_crit_->Leave();
return;
}
// Send a REMB packet.
RtpRtcp* sender = NULL;
if (!rtcp_sender_.empty()) {
sender = rtcp_sender_.front();
} else {
sender = receive_modules_.front();
}
last_send_bitrate_ = bitrate_;
list_crit_->Leave();
if (sender) {
sender->SetREMBData(bitrate_, ssrcs);
}
}
} // namespace webrtc
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