| 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 9f5546eecd514ad4b8f468d0789b3e4b3244624b..10bf411a3d90d0b0d8b5e9414bde1e1818cf6282 100644
|
| --- a/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h
|
| +++ b/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h
|
| @@ -19,6 +19,7 @@
|
| #include "webrtc/base/basictypes.h"
|
| #include "webrtc/base/scoped_ptr.h"
|
| #include "webrtc/common_types.h"
|
| +#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
|
| #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
|
| #include "webrtc/system_wrappers/interface/event_wrapper.h"
|
| #include "webrtc/system_wrappers/interface/thread_wrapper.h"
|
| @@ -27,7 +28,7 @@
|
| #include "webrtc/voice_engine/include/voe_file.h"
|
| #include "webrtc/voice_engine/include/voe_network.h"
|
| #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;
|
|
|
| @@ -57,9 +58,13 @@ class ConferenceTransport: public webrtc::Transport {
|
| /* AddStream()
|
| * Adds a stream in the conference.
|
| *
|
| + * Input:
|
| + * file_name : name of the file to be added as microphone input.
|
| + * format : format of the input file.
|
| + *
|
| * Returns stream id.
|
| */
|
| - unsigned int AddStream();
|
| + unsigned int AddStream(std::string file_name, webrtc::FileFormats format);
|
|
|
| /* RemoveStream()
|
| * Removes a stream with specified ID from the conference.
|
| @@ -123,7 +128,7 @@ class ConferenceTransport: public webrtc::Transport {
|
| int GetReceiverChannelForSsrc(unsigned int sender_ssrc) const;
|
| void StorePacket(Packet::Type type, int channel, const void* data,
|
| size_t len);
|
| - void SendPacket(const Packet& packet) const;
|
| + void SendPacket(const Packet& packet);
|
| bool DispatchPackets();
|
|
|
| const rtc::scoped_ptr<webrtc::CriticalSectionWrapper> pq_crit_;
|
| @@ -152,6 +157,10 @@ class ConferenceTransport: public webrtc::Transport {
|
| webrtc::VoERTP_RTCP* remote_rtp_rtcp_;
|
| webrtc::VoENetwork* remote_network_;
|
| webrtc::VoEFile* remote_file_;
|
| +
|
| + LoudestFilter loudest_filter_;
|
| +
|
| + const rtc::scoped_ptr<webrtc::RtpHeaderParser> rtp_header_parser_;
|
| };
|
| } // namespace voetest
|
|
|
|
|