| Index: webrtc/p2p/base/packettransportinternal.h
|
| diff --git a/webrtc/p2p/base/packettransportinterface.h b/webrtc/p2p/base/packettransportinternal.h
|
| similarity index 79%
|
| copy from webrtc/p2p/base/packettransportinterface.h
|
| copy to webrtc/p2p/base/packettransportinternal.h
|
| index 04130ef040b811cb726da637addad134ed587c5f..5789c62eeacaec610afc088475336e37c1c4efad 100644
|
| --- a/webrtc/p2p/base/packettransportinterface.h
|
| +++ b/webrtc/p2p/base/packettransportinternal.h
|
| @@ -8,8 +8,8 @@
|
| * be found in the AUTHORS file in the root of the source tree.
|
| */
|
|
|
| -#ifndef WEBRTC_P2P_BASE_PACKETTRANSPORTINTERFACE_H_
|
| -#define WEBRTC_P2P_BASE_PACKETTRANSPORTINTERFACE_H_
|
| +#ifndef WEBRTC_P2P_BASE_PACKETTRANSPORTINTERNAL_H_
|
| +#define WEBRTC_P2P_BASE_PACKETTRANSPORTINTERNAL_H_
|
|
|
| #include <string>
|
| #include <vector>
|
| @@ -28,9 +28,9 @@ struct PacketOptions;
|
| struct PacketTime;
|
| struct SentPacket;
|
|
|
| -class PacketTransportInterface : public sigslot::has_slots<> {
|
| +class PacketTransportInternal : public sigslot::has_slots<> {
|
| public:
|
| - virtual ~PacketTransportInterface() {}
|
| + virtual ~PacketTransportInternal() {}
|
|
|
| // Identify the object for logging and debug purpose.
|
| virtual std::string debug_name() const = 0;
|
| @@ -59,7 +59,7 @@ class PacketTransportInterface : public sigslot::has_slots<> {
|
| // supported by all transport types.
|
| virtual int SetOption(rtc::Socket::Option opt, int value) = 0;
|
|
|
| - // TODO(pthatcher): Once Chrome's MockPacketTransportInterface implements
|
| + // TODO(pthatcher): Once Chrome's MockPacketTransportInternal implements
|
| // this, remove the default implementation.
|
| virtual bool GetOption(rtc::Socket::Option opt, int* value) { return false; }
|
|
|
| @@ -67,20 +67,20 @@ class PacketTransportInterface : public sigslot::has_slots<> {
|
| virtual int GetError() = 0;
|
|
|
| // Emitted when the writable state, represented by |writable()|, changes.
|
| - sigslot::signal1<PacketTransportInterface*> SignalWritableState;
|
| + sigslot::signal1<PacketTransportInternal*> SignalWritableState;
|
|
|
| - // Emitted when the PacketTransportInterface is ready to send packets. "Ready
|
| + // Emitted when the PacketTransportInternal is ready to send packets. "Ready
|
| // to send" is more sensitive than the writable state; a transport may be
|
| // writable, but temporarily not able to send packets. For example, the
|
| // underlying transport's socket buffer may be full, as indicated by
|
| // SendPacket's return code and/or GetError.
|
| - sigslot::signal1<PacketTransportInterface*> SignalReadyToSend;
|
| + sigslot::signal1<PacketTransportInternal*> SignalReadyToSend;
|
|
|
| // Emitted when receiving state changes to true.
|
| - sigslot::signal1<PacketTransportInterface*> SignalReceivingState;
|
| + sigslot::signal1<PacketTransportInternal*> SignalReceivingState;
|
|
|
| // Signalled each time a packet is received on this channel.
|
| - sigslot::signal5<PacketTransportInterface*,
|
| + sigslot::signal5<PacketTransportInternal*,
|
| const char*,
|
| size_t,
|
| const rtc::PacketTime&,
|
| @@ -88,10 +88,10 @@ class PacketTransportInterface : public sigslot::has_slots<> {
|
| SignalReadPacket;
|
|
|
| // Signalled each time a packet is sent on this channel.
|
| - sigslot::signal2<PacketTransportInterface*, const rtc::SentPacket&>
|
| + sigslot::signal2<PacketTransportInternal*, const rtc::SentPacket&>
|
| SignalSentPacket;
|
| };
|
|
|
| } // namespace rtc
|
|
|
| -#endif // WEBRTC_P2P_BASE_PACKETTRANSPORTINTERFACE_H_
|
| +#endif // WEBRTC_P2P_BASE_PACKETTRANSPORTINTERNAL_H_
|
|
|