| 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;
|
|
|