e590416722
This means all channels within the same group will share the same pacing queue and scheduler. It also means padding will be computed and sent by a single pacer. To accomplish this I also introduce a PacketRouter which finds the RTP module which owns the packet to be paced out. BUG=4323 R=mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/45549004 Cr-Commit-Position: refs/heads/master@{#8864}
62 lines
2.0 KiB
C++
62 lines
2.0 KiB
C++
/*
|
|
* Copyright (c) 2015 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/modules/pacing/include/packet_router.h"
|
|
|
|
#include "webrtc/base/checks.h"
|
|
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
|
|
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
|
|
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
|
|
|
|
namespace webrtc {
|
|
|
|
PacketRouter::PacketRouter()
|
|
: crit_(CriticalSectionWrapper::CreateCriticalSection()) {
|
|
}
|
|
|
|
PacketRouter::~PacketRouter() {
|
|
}
|
|
|
|
void PacketRouter::AddRtpModule(RtpRtcp* rtp_module) {
|
|
CriticalSectionScoped cs(crit_.get());
|
|
DCHECK(std::find(rtp_modules_.begin(), rtp_modules_.end(), rtp_module) ==
|
|
rtp_modules_.end());
|
|
rtp_modules_.push_back(rtp_module);
|
|
}
|
|
|
|
void PacketRouter::RemoveRtpModule(RtpRtcp* rtp_module) {
|
|
CriticalSectionScoped cs(crit_.get());
|
|
rtp_modules_.remove(rtp_module);
|
|
}
|
|
|
|
bool PacketRouter::TimeToSendPacket(uint32_t ssrc,
|
|
uint16_t sequence_number,
|
|
int64_t capture_timestamp,
|
|
bool retransmission) {
|
|
CriticalSectionScoped cs(crit_.get());
|
|
for (auto* rtp_module : rtp_modules_) {
|
|
if (rtp_module->SendingMedia() && ssrc == rtp_module->SSRC()) {
|
|
return rtp_module->TimeToSendPacket(ssrc, sequence_number,
|
|
capture_timestamp, retransmission);
|
|
}
|
|
}
|
|
return true;
|
|
}
|
|
|
|
size_t PacketRouter::TimeToSendPadding(size_t bytes) {
|
|
CriticalSectionScoped cs(crit_.get());
|
|
for (auto* rtp_module : rtp_modules_) {
|
|
if (rtp_module->SendingMedia())
|
|
return rtp_module->TimeToSendPadding(bytes);
|
|
}
|
|
return 0;
|
|
}
|
|
} // namespace webrtc
|