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 |