| Index: webrtc/p2p/base/turnport.cc
|
| diff --git a/webrtc/p2p/base/turnport.cc b/webrtc/p2p/base/turnport.cc
|
| index f97642e6bd81e103e44682b6fd5221aac97cc99e..77c58b9c6af1c4e72ef55a403228c9158b775281 100644
|
| --- a/webrtc/p2p/base/turnport.cc
|
| +++ b/webrtc/p2p/base/turnport.cc
|
| @@ -16,6 +16,7 @@
|
| #include "webrtc/p2p/base/stun.h"
|
| #include "webrtc/base/asyncpacketsocket.h"
|
| #include "webrtc/base/byteorder.h"
|
| +#include "webrtc/base/checks.h"
|
| #include "webrtc/base/common.h"
|
| #include "webrtc/base/logging.h"
|
| #include "webrtc/base/nethelpers.h"
|
| @@ -317,14 +318,14 @@ void TurnPort::PrepareAddress() {
|
| }
|
|
|
| bool TurnPort::CreateTurnClientSocket() {
|
| - ASSERT(!socket_ || SharedSocket());
|
| + RTC_DCHECK(!socket_ || SharedSocket());
|
|
|
| if (server_address_.proto == PROTO_UDP && !SharedSocket()) {
|
| socket_ = socket_factory()->CreateUdpSocket(
|
| rtc::SocketAddress(ip(), 0), min_port(), max_port());
|
| } else if (server_address_.proto == PROTO_TCP ||
|
| server_address_.proto == PROTO_TLS) {
|
| - ASSERT(!SharedSocket());
|
| + RTC_DCHECK(!SharedSocket());
|
| int opts = rtc::PacketSocketFactory::OPT_STUN;
|
|
|
| // Apply server address TLS and insecure bits to options.
|
| @@ -375,7 +376,7 @@ bool TurnPort::CreateTurnClientSocket() {
|
| }
|
|
|
| void TurnPort::OnSocketConnect(rtc::AsyncPacketSocket* socket) {
|
| - ASSERT(server_address_.proto == PROTO_TCP);
|
| + RTC_DCHECK(server_address_.proto == PROTO_TCP);
|
| // Do not use this port if the socket bound to a different address than
|
| // the one we asked for. This is seen in Chrome, where TCP sockets cannot be
|
| // given a binding address, and the platform is expected to pick the
|
| @@ -420,7 +421,7 @@ void TurnPort::OnSocketConnect(rtc::AsyncPacketSocket* socket) {
|
|
|
| void TurnPort::OnSocketClose(rtc::AsyncPacketSocket* socket, int error) {
|
| LOG_J(LS_WARNING, this) << "Connection with server failed, error=" << error;
|
| - ASSERT(socket == socket_);
|
| + RTC_DCHECK(socket == socket_);
|
| Close();
|
| }
|
|
|
| @@ -680,7 +681,7 @@ void TurnPort::ResolveTurnAddress(const rtc::SocketAddress& address) {
|
| }
|
|
|
| void TurnPort::OnResolveResult(rtc::AsyncResolverInterface* resolver) {
|
| - ASSERT(resolver == resolver_);
|
| + RTC_DCHECK(resolver == resolver_);
|
| // If DNS resolve is failed when trying to connect to the server using TCP,
|
| // one of the reason could be due to DNS queries blocked by firewall.
|
| // In such cases we will try to connect to the server with hostname, assuming
|
| @@ -713,7 +714,7 @@ void TurnPort::OnResolveResult(rtc::AsyncResolverInterface* resolver) {
|
|
|
| void TurnPort::OnSendStunPacket(const void* data, size_t size,
|
| StunRequest* request) {
|
| - ASSERT(connected());
|
| + RTC_DCHECK(connected());
|
| rtc::PacketOptions options(DefaultDscpValue());
|
| if (Send(data, size, options) < 0) {
|
| LOG_J(LS_ERROR, this) << "Failed to send TURN message, err="
|
| @@ -804,8 +805,8 @@ void TurnPort::OnMessage(rtc::Message* message) {
|
| // Since it's TCP, we have to delete the connected socket and reconnect
|
| // with the alternate server. PrepareAddress will send stun binding once
|
| // the new socket is connected.
|
| - ASSERT(server_address().proto == PROTO_TCP);
|
| - ASSERT(!SharedSocket());
|
| + RTC_DCHECK(server_address().proto == PROTO_TCP);
|
| + RTC_DCHECK(!SharedSocket());
|
| delete socket_;
|
| socket_ = NULL;
|
| PrepareAddress();
|
| @@ -1021,7 +1022,7 @@ void TurnPort::CreateOrRefreshEntry(const rtc::SocketAddress& addr) {
|
| }
|
|
|
| void TurnPort::DestroyEntry(TurnEntry* entry) {
|
| - ASSERT(entry != NULL);
|
| + RTC_DCHECK(entry != NULL);
|
| entry->SignalDestroyed(entry);
|
| entries_.remove(entry);
|
| delete entry;
|
| @@ -1042,12 +1043,12 @@ void TurnPort::HandleConnectionDestroyed(Connection* conn) {
|
| // already destroyed.
|
| const rtc::SocketAddress& remote_address = conn->remote_candidate().address();
|
| TurnEntry* entry = FindEntry(remote_address);
|
| - ASSERT(entry != NULL);
|
| + RTC_DCHECK(entry != NULL);
|
| ScheduleEntryDestruction(entry);
|
| }
|
|
|
| void TurnPort::ScheduleEntryDestruction(TurnEntry* entry) {
|
| - ASSERT(entry->destruction_timestamp() == 0);
|
| + RTC_DCHECK(entry->destruction_timestamp() == 0);
|
| int64_t timestamp = rtc::TimeMillis();
|
| entry->set_destruction_timestamp(timestamp);
|
| invoker_.AsyncInvokeDelayed<void>(
|
| @@ -1057,7 +1058,7 @@ void TurnPort::ScheduleEntryDestruction(TurnEntry* entry) {
|
| }
|
|
|
| void TurnPort::CancelEntryDestruction(TurnEntry* entry) {
|
| - ASSERT(entry->destruction_timestamp() != 0);
|
| + RTC_DCHECK(entry->destruction_timestamp() != 0);
|
| entry->set_destruction_timestamp(0);
|
| }
|
|
|
| @@ -1368,7 +1369,7 @@ void TurnCreatePermissionRequest::OnTimeout() {
|
| }
|
|
|
| void TurnCreatePermissionRequest::OnEntryDestroyed(TurnEntry* entry) {
|
| - ASSERT(entry_ == entry);
|
| + RTC_DCHECK(entry_ == entry);
|
| entry_ = NULL;
|
| }
|
|
|
| @@ -1438,7 +1439,7 @@ void TurnChannelBindRequest::OnTimeout() {
|
| }
|
|
|
| void TurnChannelBindRequest::OnEntryDestroyed(TurnEntry* entry) {
|
| - ASSERT(entry_ == entry);
|
| + RTC_DCHECK(entry_ == entry);
|
| entry_ = NULL;
|
| }
|
|
|
| @@ -1533,7 +1534,7 @@ void TurnEntry::OnCreatePermissionTimeout() {
|
| void TurnEntry::OnChannelBindSuccess() {
|
| LOG_J(LS_INFO, port_) << "Channel bind for " << ext_addr_.ToSensitiveString()
|
| << " succeeded";
|
| - ASSERT(state_ == STATE_BINDING || state_ == STATE_BOUND);
|
| + RTC_DCHECK(state_ == STATE_BINDING || state_ == STATE_BOUND);
|
| state_ = STATE_BOUND;
|
| }
|
|
|
|
|