Index: webrtc/voice_engine/test/auto_test/fakes/conference_transport.h |
diff --git a/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h b/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h |
index d8a5a49bb7fd1820630f08263afb0253e92bf0bd..37919e1e99d1403dca95e71eefc2da03458adcfe 100644 |
--- a/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h |
+++ b/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h |
@@ -30,10 +30,11 @@ |
#include "webrtc/voice_engine/include/voe_rtp_rtcp.h" |
#include "webrtc/voice_engine/test/auto_test/fakes/loudest_filter.h" |
-static const size_t kMaxPacketSizeByte = 1500; |
- |
+namespace webrtc { |
namespace voetest { |
+static const size_t kMaxPacketSizeByte = 1500; |
+ |
// This class is to simulate a conference call. There are two Voice Engines, one |
// for local channels and the other for remote channels. There is a simulated |
// reflector, which exchanges RTCP with local channels. For simplicity, it |
@@ -158,6 +159,8 @@ class ConferenceTransport: public webrtc::Transport { |
const std::unique_ptr<webrtc::RtpHeaderParser> rtp_header_parser_; |
}; |
+ |
} // namespace voetest |
+} // namespace webrtc |
#endif // WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_ |