blob: e98ab0c2b02539963758ce0259c412e5657f385f [file] [log] [blame]
/*
* 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/packet_router.h"
#include "webrtc/base/atomicops.h"
#include "webrtc/base/checks.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
namespace webrtc {
namespace {
void AddModule(RtpRtcp* rtp_module, std::list<RtpRtcp*>* rtp_modules) {
RTC_DCHECK(std::find(rtp_modules->begin(), rtp_modules->end(), rtp_module) ==
rtp_modules->end());
rtp_modules->push_back(rtp_module);
}
void RemoveModule(RtpRtcp* rtp_module, std::list<RtpRtcp*>* rtp_modules) {
RTC_DCHECK(std::find(rtp_modules->begin(), rtp_modules->end(), rtp_module) !=
rtp_modules->end());
rtp_modules->remove(rtp_module);
}
bool SendFeedback(rtcp::TransportFeedback* packet,
std::list<RtpRtcp*>* rtp_modules) {
for (auto* rtp_module : *rtp_modules) {
packet->WithPacketSenderSsrc(rtp_module->SSRC());
if (rtp_module->SendFeedbackPacket(*packet))
return true;
}
return false;
}
}
PacketRouter::PacketRouter() : transport_seq_(0) {
pacer_thread_checker_.DetachFromThread();
}
PacketRouter::~PacketRouter() {
RTC_DCHECK(send_rtp_modules_.empty());
RTC_DCHECK(recv_rtp_modules_.empty());
}
void PacketRouter::AddRtpModule(RtpRtcp* rtp_module, bool sender) {
rtc::CritScope cs(&modules_crit_);
if (sender) {
AddModule(rtp_module, &send_rtp_modules_);
} else {
AddModule(rtp_module, &recv_rtp_modules_);
}
}
void PacketRouter::RemoveRtpModule(RtpRtcp* rtp_module, bool sender) {
rtc::CritScope cs(&modules_crit_);
if (sender) {
RemoveModule(rtp_module, &send_rtp_modules_);
} else {
RemoveModule(rtp_module, &recv_rtp_modules_);
}
}
bool PacketRouter::TimeToSendPacket(uint32_t ssrc,
uint16_t sequence_number,
int64_t capture_timestamp,
bool retransmission) {
RTC_DCHECK(pacer_thread_checker_.CalledOnValidThread());
rtc::CritScope cs(&modules_crit_);
for (auto* rtp_module : send_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_to_send) {
RTC_DCHECK(pacer_thread_checker_.CalledOnValidThread());
size_t total_bytes_sent = 0;
rtc::CritScope cs(&modules_crit_);
for (RtpRtcp* module : send_rtp_modules_) {
if (module->SendingMedia()) {
size_t bytes_sent =
module->TimeToSendPadding(bytes_to_send - total_bytes_sent);
total_bytes_sent += bytes_sent;
if (total_bytes_sent >= bytes_to_send)
break;
}
}
return total_bytes_sent;
}
void PacketRouter::SetTransportWideSequenceNumber(uint16_t sequence_number) {
rtc::AtomicOps::ReleaseStore(&transport_seq_, sequence_number);
}
uint16_t PacketRouter::AllocateSequenceNumber() {
int prev_seq = rtc::AtomicOps::AcquireLoad(&transport_seq_);
int desired_prev_seq;
int new_seq;
do {
desired_prev_seq = prev_seq;
new_seq = (desired_prev_seq + 1) & 0xFFFF;
// Note: CompareAndSwap returns the actual value of transport_seq at the
// time the CAS operation was executed. Thus, if prev_seq is returned, the
// operation was successful - otherwise we need to retry. Saving the
// return value saves us a load on retry.
prev_seq = rtc::AtomicOps::CompareAndSwap(&transport_seq_, desired_prev_seq,
new_seq);
} while (prev_seq != desired_prev_seq);
return new_seq;
}
bool PacketRouter::SendFeedback(rtcp::TransportFeedback* packet) {
RTC_DCHECK(pacer_thread_checker_.CalledOnValidThread());
rtc::CritScope cs(&modules_crit_);
if (::webrtc::SendFeedback(packet, &recv_rtp_modules_))
return true;
if (::webrtc::SendFeedback(packet, &send_rtp_modules_))
return true;
return false;
}
} // namespace webrtc