mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 13:50:40 +01:00
Replace use of RecursiveCriticalSection in VirtualSocketServer
Also change listen_queue_ member to use std::unique_ptr to manage ownership. Bug: webrtc:11567 Change-Id: I85171c9cd0253fdbcbce38b1cfebb1adb5bddd9b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223063 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34353}
This commit is contained in:
parent
fe6580fb87
commit
c413c5581b
2 changed files with 25 additions and 24 deletions
|
@ -19,7 +19,6 @@
|
|||
|
||||
#include "absl/algorithm/container.h"
|
||||
#include "rtc_base/checks.h"
|
||||
#include "rtc_base/deprecated/recursive_critical_section.h"
|
||||
#include "rtc_base/fake_clock.h"
|
||||
#include "rtc_base/logging.h"
|
||||
#include "rtc_base/physical_socket_server.h"
|
||||
|
@ -164,7 +163,7 @@ int VirtualSocket::Close() {
|
|||
}
|
||||
|
||||
if (SOCK_STREAM == type_) {
|
||||
CritScope cs(&crit_);
|
||||
webrtc::MutexLock lock(&mutex_);
|
||||
|
||||
// Cancel pending sockets
|
||||
if (listen_queue_) {
|
||||
|
@ -175,7 +174,6 @@ int VirtualSocket::Close() {
|
|||
server_->Disconnect(addr);
|
||||
listen_queue_->pop_front();
|
||||
}
|
||||
delete listen_queue_;
|
||||
listen_queue_ = nullptr;
|
||||
}
|
||||
// Disconnect stream sockets
|
||||
|
@ -234,7 +232,7 @@ int VirtualSocket::RecvFrom(void* pv,
|
|||
*timestamp = -1;
|
||||
}
|
||||
|
||||
CritScope cs(&crit_);
|
||||
webrtc::MutexLock lock(&mutex_);
|
||||
// If we don't have a packet, then either error or wait for one to arrive.
|
||||
if (recv_buffer_.empty()) {
|
||||
if (async_) {
|
||||
|
@ -277,7 +275,7 @@ int VirtualSocket::RecvFrom(void* pv,
|
|||
}
|
||||
|
||||
int VirtualSocket::Listen(int backlog) {
|
||||
CritScope cs(&crit_);
|
||||
webrtc::MutexLock lock(&mutex_);
|
||||
RTC_DCHECK(SOCK_STREAM == type_);
|
||||
RTC_DCHECK(CS_CLOSED == state_);
|
||||
if (local_addr_.IsNil()) {
|
||||
|
@ -285,13 +283,13 @@ int VirtualSocket::Listen(int backlog) {
|
|||
return -1;
|
||||
}
|
||||
RTC_DCHECK(nullptr == listen_queue_);
|
||||
listen_queue_ = new ListenQueue;
|
||||
listen_queue_ = std::make_unique<ListenQueue>();
|
||||
state_ = CS_CONNECTING;
|
||||
return 0;
|
||||
}
|
||||
|
||||
VirtualSocket* VirtualSocket::Accept(SocketAddress* paddr) {
|
||||
CritScope cs(&crit_);
|
||||
webrtc::MutexLock lock(&mutex_);
|
||||
if (nullptr == listen_queue_) {
|
||||
error_ = EINVAL;
|
||||
return nullptr;
|
||||
|
@ -310,7 +308,7 @@ VirtualSocket* VirtualSocket::Accept(SocketAddress* paddr) {
|
|||
delete socket;
|
||||
continue;
|
||||
}
|
||||
socket->CompleteConnect(remote_addr, false);
|
||||
socket->CompleteConnect(remote_addr);
|
||||
if (paddr) {
|
||||
*paddr = remote_addr;
|
||||
}
|
||||
|
@ -349,9 +347,10 @@ int VirtualSocket::SetOption(Option opt, int value) {
|
|||
void VirtualSocket::OnMessage(Message* pmsg) {
|
||||
bool signal_read_event = false;
|
||||
bool signal_close_event = false;
|
||||
bool signal_connect_event = false;
|
||||
int error_to_signal = 0;
|
||||
{
|
||||
CritScope cs(&crit_);
|
||||
webrtc::MutexLock lock(&mutex_);
|
||||
if (pmsg->message_id == MSG_ID_PACKET) {
|
||||
RTC_DCHECK(nullptr != pmsg->pdata);
|
||||
Packet* packet = static_cast<Packet*>(pmsg->pdata);
|
||||
|
@ -365,7 +364,8 @@ void VirtualSocket::OnMessage(Message* pmsg) {
|
|||
listen_queue_->push_back(data->addr);
|
||||
signal_read_event = async_;
|
||||
} else if ((SOCK_STREAM == type_) && (CS_CONNECTING == state_)) {
|
||||
CompleteConnect(data->addr, true);
|
||||
CompleteConnect(data->addr);
|
||||
signal_connect_event = async_;
|
||||
} else {
|
||||
RTC_LOG(LS_VERBOSE)
|
||||
<< "Socket at " << local_addr_.ToString() << " is not listening";
|
||||
|
@ -386,14 +386,17 @@ void VirtualSocket::OnMessage(Message* pmsg) {
|
|||
RTC_NOTREACHED();
|
||||
}
|
||||
}
|
||||
// Signal events without holding `crit_`, to avoid lock order inversion with
|
||||
// sigslot locks.
|
||||
// Signal events without holding `mutex_`, to avoid recursive locking, as well
|
||||
// as issues with sigslot and lock order.
|
||||
if (signal_read_event) {
|
||||
SignalReadEvent(this);
|
||||
}
|
||||
if (signal_close_event) {
|
||||
SignalCloseEvent(this, error_to_signal);
|
||||
}
|
||||
if (signal_connect_event) {
|
||||
SignalConnectEvent(this);
|
||||
}
|
||||
}
|
||||
|
||||
int VirtualSocket::InitiateConnect(const SocketAddress& addr, bool use_delay) {
|
||||
|
@ -427,14 +430,11 @@ int VirtualSocket::InitiateConnect(const SocketAddress& addr, bool use_delay) {
|
|||
return 0;
|
||||
}
|
||||
|
||||
void VirtualSocket::CompleteConnect(const SocketAddress& addr, bool notify) {
|
||||
void VirtualSocket::CompleteConnect(const SocketAddress& addr) {
|
||||
RTC_DCHECK(CS_CONNECTING == state_);
|
||||
remote_addr_ = addr;
|
||||
state_ = CS_CONNECTED;
|
||||
server_->AddConnection(remote_addr_, local_addr_, this);
|
||||
if (async_ && notify) {
|
||||
SignalConnectEvent(this);
|
||||
}
|
||||
}
|
||||
|
||||
int VirtualSocket::SendUdp(const void* pv,
|
||||
|
@ -486,7 +486,7 @@ void VirtualSocket::OnSocketServerReadyToSend() {
|
|||
}
|
||||
|
||||
void VirtualSocket::SetToBlocked() {
|
||||
CritScope cs(&crit_);
|
||||
webrtc::MutexLock lock(&mutex_);
|
||||
ready_to_send_ = false;
|
||||
error_ = EWOULDBLOCK;
|
||||
}
|
||||
|
@ -536,7 +536,7 @@ int64_t VirtualSocket::UpdateOrderedDelivery(int64_t ts) {
|
|||
}
|
||||
|
||||
size_t VirtualSocket::PurgeNetworkPackets(int64_t cur_time) {
|
||||
CritScope cs(&crit_);
|
||||
webrtc::MutexLock lock(&mutex_);
|
||||
|
||||
while (!network_.empty() && (network_.front().done_time <= cur_time)) {
|
||||
RTC_DCHECK(network_size_ >= network_.front().size);
|
||||
|
|
|
@ -17,11 +17,11 @@
|
|||
|
||||
#include "rtc_base/checks.h"
|
||||
#include "rtc_base/constructor_magic.h"
|
||||
#include "rtc_base/deprecated/recursive_critical_section.h"
|
||||
#include "rtc_base/event.h"
|
||||
#include "rtc_base/fake_clock.h"
|
||||
#include "rtc_base/message_handler.h"
|
||||
#include "rtc_base/socket_server.h"
|
||||
#include "rtc_base/synchronization/mutex.h"
|
||||
|
||||
namespace rtc {
|
||||
|
||||
|
@ -394,7 +394,7 @@ class VirtualSocket : public AsyncSocket,
|
|||
typedef std::map<Option, int> OptionsMap;
|
||||
|
||||
int InitiateConnect(const SocketAddress& addr, bool use_delay);
|
||||
void CompleteConnect(const SocketAddress& addr, bool notify);
|
||||
void CompleteConnect(const SocketAddress& addr);
|
||||
int SendUdp(const void* pv, size_t cb, const SocketAddress& addr);
|
||||
int SendTcp(const void* pv, size_t cb);
|
||||
|
||||
|
@ -409,7 +409,8 @@ class VirtualSocket : public AsyncSocket,
|
|||
SocketAddress remote_addr_;
|
||||
|
||||
// Pending sockets which can be Accepted
|
||||
ListenQueue* listen_queue_ RTC_GUARDED_BY(crit_) RTC_PT_GUARDED_BY(crit_);
|
||||
std::unique_ptr<ListenQueue> listen_queue_ RTC_GUARDED_BY(mutex_)
|
||||
RTC_PT_GUARDED_BY(mutex_);
|
||||
|
||||
// Data which tcp has buffered for sending
|
||||
SendBuffer send_buffer_;
|
||||
|
@ -417,8 +418,8 @@ class VirtualSocket : public AsyncSocket,
|
|||
// Set back to true when the socket can send again.
|
||||
bool ready_to_send_ = true;
|
||||
|
||||
// Critical section to protect the recv_buffer and listen_queue_
|
||||
RecursiveCriticalSection crit_;
|
||||
// Mutex to protect the recv_buffer and listen_queue_
|
||||
webrtc::Mutex mutex_;
|
||||
|
||||
// Network model that enforces bandwidth and capacity constraints
|
||||
NetworkQueue network_;
|
||||
|
@ -428,7 +429,7 @@ class VirtualSocket : public AsyncSocket,
|
|||
int64_t last_delivery_time_ = 0;
|
||||
|
||||
// Data which has been received from the network
|
||||
RecvBuffer recv_buffer_ RTC_GUARDED_BY(crit_);
|
||||
RecvBuffer recv_buffer_ RTC_GUARDED_BY(mutex_);
|
||||
// The amount of data which is in flight or in recv_buffer_
|
||||
size_t recv_buffer_size_;
|
||||
|
||||
|
|
Loading…
Reference in a new issue