137 lines
3.6 KiB
C++
137 lines
3.6 KiB
C++
/*
|
|
* 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/trace.h"
|
|
|
|
namespace webrtc {
|
|
|
|
const int kRembSendIntervalMs = 200;
|
|
|
|
// % threshold for if we should send a new REMB asap.
|
|
const uint32_t kSendThresholdPercent = 97;
|
|
|
|
VieRemb::VieRemb(Clock* clock)
|
|
: clock_(clock),
|
|
last_remb_time_(clock_->TimeInMilliseconds()),
|
|
last_send_bitrate_(0),
|
|
bitrate_(0) {}
|
|
|
|
VieRemb::~VieRemb() {}
|
|
|
|
void VieRemb::AddReceiveChannel(RtpRtcp* rtp_rtcp) {
|
|
assert(rtp_rtcp);
|
|
|
|
rtc::CritScope lock(&list_crit_);
|
|
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);
|
|
|
|
rtc::CritScope lock(&list_crit_);
|
|
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);
|
|
|
|
rtc::CritScope lock(&list_crit_);
|
|
|
|
// 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);
|
|
|
|
rtc::CritScope lock(&list_crit_);
|
|
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 {
|
|
rtc::CritScope lock(&list_crit_);
|
|
return !receive_modules_.empty() || !rtcp_sender_.empty();
|
|
}
|
|
|
|
void VieRemb::OnReceiveBitrateChanged(const std::vector<uint32_t>& ssrcs,
|
|
uint32_t bitrate) {
|
|
RtpRtcp* sender = nullptr;
|
|
{
|
|
rtc::CritScope lock(&list_crit_);
|
|
// If we already have an estimate, check if the new total estimate is below
|
|
// kSendThresholdPercent of the previous estimate.
|
|
if (last_send_bitrate_ > 0) {
|
|
uint32_t 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) {
|
|
return;
|
|
}
|
|
last_remb_time_ = now;
|
|
|
|
if (ssrcs.empty() || receive_modules_.empty()) {
|
|
return;
|
|
}
|
|
|
|
// Send a REMB packet.
|
|
if (!rtcp_sender_.empty()) {
|
|
sender = rtcp_sender_.front();
|
|
} else {
|
|
sender = receive_modules_.front();
|
|
}
|
|
last_send_bitrate_ = bitrate_;
|
|
}
|
|
|
|
if (sender) {
|
|
sender->SetREMBData(bitrate_, ssrcs);
|
|
}
|
|
}
|
|
|
|
} // namespace webrtc
|