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Add support for transport wide sequence numbers

Browse Source 
Also refactor packet router to use a map rather than iterate over all
rtp modules for each packet sent.

BUG=webrtc:4311

Review URL: https://codereview.webrtc.org/1247293002

Cr-Commit-Position: refs/heads/master@{#9670}
This commit is contained in:
sprang
2015-08-03 04:38:41 -07:00
committed by Commit bot
parent d67a219bec
commit 867fb5224e
39 changed files with 818 additions and 334 deletions
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57
webrtc/modules/pacing/packet_router.cc
View File

@ -10,37 +10,39 @@
#include "webrtc/modules/pacing/include/packet_router.h"
#include "webrtc/base/atomicops.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() : transport_seq_(0) {
}
PacketRouter::~PacketRouter() {
DCHECK(rtp_modules_.empty());
}
void PacketRouter::AddRtpModule(RtpRtcp* rtp_module) {
CriticalSectionScoped cs(crit_.get());
rtc::CritScope cs(&modules_lock_);
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);
rtc::CritScope cs(&modules_lock_);
auto it = std::find(rtp_modules_.begin(), rtp_modules_.end(), rtp_module);
DCHECK(it != rtp_modules_.end());
rtp_modules_.erase(it);
}
bool PacketRouter::TimeToSendPacket(uint32_t ssrc,
uint16_t sequence_number,
int64_t capture_timestamp,
bool retransmission) {
CriticalSectionScoped cs(crit_.get());
rtc::CritScope cs(&modules_lock_);
for (auto* rtp_module : rtp_modules_) {
if (rtp_module->SendingMedia() && ssrc == rtp_module->SSRC()) {
return rtp_module->TimeToSendPacket(ssrc, sequence_number,
@ -50,12 +52,41 @@ bool PacketRouter::TimeToSendPacket(uint32_t ssrc,
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);
size_t PacketRouter::TimeToSendPadding(size_t bytes_to_send) {
size_t total_bytes_sent = 0;
rtc::CritScope cs(&modules_lock_);
for (RtpRtcp* module : 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 0;
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;
}
} // namespace webrtc