Index: webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc |
diff --git a/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc b/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc |
index 70f68298f5df2a664ee2b02afb7e0f5a492b8811..086eeab7b1bf5393a0f56e06310b92e827a01adb 100644 |
--- a/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc |
+++ b/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc |
@@ -37,9 +37,7 @@ namespace { |
namespace voetest { |
ConferenceTransport::ConferenceTransport() |
- : pq_crit_(webrtc::CriticalSectionWrapper::CreateCriticalSection()), |
- stream_crit_(webrtc::CriticalSectionWrapper::CreateCriticalSection()), |
- packet_event_(webrtc::EventWrapper::Create()), |
+ : packet_event_(webrtc::EventWrapper::Create()), |
thread_(Run, this, "ConferenceTransport"), |
rtt_ms_(0), |
stream_count_(0), |
@@ -120,7 +118,7 @@ bool ConferenceTransport::SendRtcp(const uint8_t* data, size_t len) { |
int ConferenceTransport::GetReceiverChannelForSsrc(unsigned int sender_ssrc) |
const { |
- webrtc::CriticalSectionScoped lock(stream_crit_.get()); |
+ rtc::CritScope lock(&stream_crit_); |
auto it = streams_.find(sender_ssrc); |
if (it != streams_.end()) { |
return it->second.second; |
@@ -132,7 +130,7 @@ void ConferenceTransport::StorePacket(Packet::Type type, |
const void* data, |
size_t len) { |
{ |
- webrtc::CriticalSectionScoped lock(pq_crit_.get()); |
+ rtc::CritScope lock(&pq_crit_); |
packet_queue_.push_back(Packet(type, data, len, rtc::Time())); |
} |
packet_event_->Set(); |
@@ -198,7 +196,7 @@ bool ConferenceTransport::DispatchPackets() { |
while (true) { |
Packet packet; |
{ |
- webrtc::CriticalSectionScoped lock(pq_crit_.get()); |
+ rtc::CritScope lock(&pq_crit_); |
if (packet_queue_.empty()) |
break; |
packet = packet_queue_.front(); |
@@ -245,14 +243,14 @@ unsigned int ConferenceTransport::AddStream(std::string file_name, |
EXPECT_EQ(0, local_rtp_rtcp_->SetLocalSSRC(new_receiver, kLocalSsrc)); |
{ |
- webrtc::CriticalSectionScoped lock(stream_crit_.get()); |
+ rtc::CritScope lock(&stream_crit_); |
streams_[remote_ssrc] = std::make_pair(new_sender, new_receiver); |
} |
return remote_ssrc; // remote ssrc used as stream id. |
} |
bool ConferenceTransport::RemoveStream(unsigned int id) { |
- webrtc::CriticalSectionScoped lock(stream_crit_.get()); |
+ rtc::CritScope lock(&stream_crit_); |
auto it = streams_.find(id); |
if (it == streams_.end()) { |
return false; |