| 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_
|
|
|