Index: webrtc/p2p/base/packettransportinterface.h |
diff --git a/webrtc/p2p/base/packettransportinterface.h b/webrtc/p2p/base/packettransportinterface.h |
index a18081cf72b896a5a1160e08f32e838b9b0b641d..c796c6c2fa11194a1391e7b0b48ba7b277dafdea 100644 |
--- a/webrtc/p2p/base/packettransportinterface.h |
+++ b/webrtc/p2p/base/packettransportinterface.h |
@@ -11,12 +11,78 @@ |
#ifndef WEBRTC_P2P_BASE_PACKETTRANSPORTINTERFACE_H_ |
#define WEBRTC_P2P_BASE_PACKETTRANSPORTINTERFACE_H_ |
+#include <string> |
+#include <vector> |
+ |
+#include "webrtc/base/sigslot.h" |
+#include "webrtc/base/socket.h" |
+ |
namespace cricket { |
class TransportChannel; |
} |
namespace rtc { |
-typedef cricket::TransportChannel PacketTransportInterface; |
-} |
+struct PacketOptions; |
+struct PacketTime; |
+struct SentPacket; |
+ |
+class PacketTransportInterface : public sigslot::has_slots<> { |
+ public: |
+ virtual ~PacketTransportInterface() {} |
+ |
+ // Identify the object for logging and debug purpose. |
+ virtual const std::string debug_name() const = 0; |
+ |
+ // The transport has been established. |
+ virtual bool writable() const = 0; |
+ |
+ // Attempts to send the given packet. |
+ // The return value is < 0 on failure. The return value in failure case is not |
+ // descriptive. Depending on failure cause and implementation details |
+ // GetError() returns an descriptive errno.h error value. |
+ // This mimics posix socket send() or sendto() behavior. |
+ // TODO(johan): Reliable, meaningful, consistent error codes for all |
+ // implementations would be nice. |
+ // TODO(johan): Remove the default argument once channel code is updated. |
+ virtual int SendPacket(const char* data, |
+ size_t len, |
+ const rtc::PacketOptions& options, |
+ int flags = 0) = 0; |
+ |
+ // Sets a socket option. Note that not all options are |
+ // supported by all transport types. |
+ virtual int SetOption(rtc::Socket::Option opt, int value) = 0; |
+ |
+ // TODO(pthatcher): Once Chrome's MockPacketTransportInterface implements |
+ // this, remove the default implementation. |
+ virtual bool GetOption(rtc::Socket::Option opt, int* value) { return false; } |
+ |
+ // Returns the most recent error that occurred on this channel. |
+ virtual int GetError() = 0; |
+ |
+ // Emitted when the writable state, represented by |writable()|, changes. |
+ sigslot::signal1<PacketTransportInterface*> SignalWritableState; |
+ |
+ // Emitted when the PacketTransportInterface 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; |
+ |
+ // Signalled each time a packet is received on this channel. |
+ sigslot::signal5<PacketTransportInterface*, |
+ const char*, |
+ size_t, |
+ const rtc::PacketTime&, |
+ int> |
+ SignalReadPacket; |
+ |
+ // Signalled each time a packet is sent on this channel. |
+ sigslot::signal2<PacketTransportInterface*, const rtc::SentPacket&> |
+ SignalSentPacket; |
+}; |
+ |
+} // namespace rtc |
#endif // WEBRTC_P2P_BASE_PACKETTRANSPORTINTERFACE_H_ |