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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 "video_engine/vie_remb.h"
#include <algorithm>
#include <cassert>
#include "modules/rtp_rtcp/interface/rtp_rtcp.h"
#include "modules/utility/interface/process_thread.h"
#include "system_wrappers/interface/critical_section_wrapper.h"
#include "system_wrappers/interface/tick_util.h"
#include "system_wrappers/interface/trace.h"
namespace webrtc {
const int kRembTimeOutThresholdMs = 2000;
const int kRembSendIntervallMs = 1000;
const unsigned int kRembMinimumBitrateKbps = 50;
// % threshold for if we should send a new REMB asap.
const unsigned int kSendThresholdPercent = 97;
VieRemb::VieRemb()
: list_crit_(CriticalSectionWrapper::CreateCriticalSection()),
last_remb_time_(TickTime::MillisecondTimestamp()),
last_send_bitrate_(0),
bitrate_(0) {}
VieRemb::~VieRemb() {}
void VieRemb::AddReceiveChannel(RtpRtcp* rtp_rtcp) {
assert(rtp_rtcp);
WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, -1,
"VieRemb::AddReceiveChannel(%p)", rtp_rtcp);
CriticalSectionScoped cs(list_crit_.get());
if (std::find(receive_modules_.begin(), receive_modules_.end(), rtp_rtcp) !=
receive_modules_.end())
return;
WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1, "AddRembChannel");
// 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);
WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, -1,
"VieRemb::RemoveReceiveChannel(%p)", 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);
WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, -1,
"VieRemb::AddRembSender(%p)", 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);
WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, -1,
"VieRemb::RemoveRembSender(%p)", 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(std::vector<unsigned int>* ssrcs,
unsigned int bitrate) {
WEBRTC_TRACE(kTraceStream, kTraceVideo, -1,
"VieRemb::UpdateBitrateEstimate(bitrate: %u)", bitrate);
assert(ssrcs);
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_ = TickTime::MillisecondTimestamp() - kRembSendIntervallMs;
}
}
bitrate_ = bitrate;
// TODO(holmer): Remove |ssrcs_| from this class as the REMB is produced
// immediately upon a call to this function.
ssrcs_.resize(ssrcs->size());
std::copy(ssrcs->begin(), ssrcs->end(), ssrcs_.begin());
// Calculate total receive bitrate estimate.
int64_t now = TickTime::MillisecondTimestamp();
if (now - last_remb_time_ < kRembSendIntervallMs) {
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_;
// Never send a REMB lower than last_send_bitrate_.
if (last_send_bitrate_ < kRembMinimumBitrateKbps) {
last_send_bitrate_ = kRembMinimumBitrateKbps;
}
// Copy SSRCs to avoid race conditions.
int ssrcs_length = ssrcs_.size();
unsigned int* ssrcs_copy = new unsigned int[ssrcs_length];
for (int i = 0; i < ssrcs_length; ++i) {
ssrcs_copy[i] = ssrcs_[i];
}
list_crit_->Leave();
if (sender) {
// TODO(holmer): Change RTP module API to take a vector pointer.
sender->SetREMBData(bitrate_, ssrcs_length, ssrcs_copy);
}
delete [] ssrcs_copy;
}
} // namespace webrtc
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