| Index: webrtc/p2p/base/packettransportinternal.h
|
| diff --git a/webrtc/p2p/base/packettransportinterface.h b/webrtc/p2p/base/packettransportinternal.h
|
| similarity index 78%
|
| rename from webrtc/p2p/base/packettransportinterface.h
|
| rename to webrtc/p2p/base/packettransportinternal.h
|
| index 04130ef040b811cb726da637addad134ed587c5f..77c409c019801a257948566812fc0d84cd5d5150 100644
|
| --- a/webrtc/p2p/base/packettransportinterface.h
|
| +++ b/webrtc/p2p/base/packettransportinternal.h
|
| @@ -8,12 +8,13 @@
|
| * 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>
|
|
|
| +#include "webrtc/api/ortc/packettransportinterface.h"
|
| // This is included for PacketOptions.
|
| #include "webrtc/base/asyncpacketsocket.h"
|
| #include "webrtc/base/sigslot.h"
|
| @@ -28,10 +29,9 @@ struct PacketOptions;
|
| struct PacketTime;
|
| struct SentPacket;
|
|
|
| -class PacketTransportInterface : public sigslot::has_slots<> {
|
| +class PacketTransportInternal : public virtual webrtc::PacketTransportInterface,
|
| + public sigslot::has_slots<> {
|
| public:
|
| - virtual ~PacketTransportInterface() {}
|
| -
|
| // Identify the object for logging and debug purpose.
|
| virtual std::string debug_name() const = 0;
|
|
|
| @@ -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 +88,13 @@ 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;
|
| +
|
| + protected:
|
| + PacketTransportInternal* GetInternal() { return this; }
|
| };
|
|
|
| } // namespace rtc
|
|
|
| -#endif // WEBRTC_P2P_BASE_PACKETTRANSPORTINTERFACE_H_
|
| +#endif // WEBRTC_P2P_BASE_PACKETTRANSPORTINTERNAL_H_
|
|
|