Index: talk/app/webrtc/datachannelinterface.h |
diff --git a/talk/app/webrtc/datachannelinterface.h b/talk/app/webrtc/datachannelinterface.h |
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-/* |
- * libjingle |
- * Copyright 2012 Google Inc. |
- * |
- * Redistribution and use in source and binary forms, with or without |
- * modification, are permitted provided that the following conditions are met: |
- * |
- * 1. Redistributions of source code must retain the above copyright notice, |
- * this list of conditions and the following disclaimer. |
- * 2. Redistributions in binary form must reproduce the above copyright notice, |
- * this list of conditions and the following disclaimer in the documentation |
- * and/or other materials provided with the distribution. |
- * 3. The name of the author may not be used to endorse or promote products |
- * derived from this software without specific prior written permission. |
- * |
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED |
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF |
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO |
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, |
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, |
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; |
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, |
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR |
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF |
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. |
- */ |
- |
-// This file contains interfaces for DataChannels |
-// http://dev.w3.org/2011/webrtc/editor/webrtc.html#rtcdatachannel |
- |
-#ifndef TALK_APP_WEBRTC_DATACHANNELINTERFACE_H_ |
-#define TALK_APP_WEBRTC_DATACHANNELINTERFACE_H_ |
- |
-#include <string> |
- |
-#include "webrtc/base/basictypes.h" |
-#include "webrtc/base/buffer.h" |
-#include "webrtc/base/checks.h" |
-#include "webrtc/base/refcount.h" |
- |
- |
-namespace webrtc { |
- |
-struct DataChannelInit { |
- DataChannelInit() |
- : reliable(false), |
- ordered(true), |
- maxRetransmitTime(-1), |
- maxRetransmits(-1), |
- negotiated(false), |
- id(-1) { |
- } |
- |
- bool reliable; // Deprecated. |
- bool ordered; // True if ordered delivery is required. |
- int maxRetransmitTime; // The max period of time in milliseconds in which |
- // retransmissions will be sent. After this time, no |
- // more retransmissions will be sent. -1 if unset. |
- int maxRetransmits; // The max number of retransmissions. -1 if unset. |
- std::string protocol; // This is set by the application and opaque to the |
- // WebRTC implementation. |
- bool negotiated; // True if the channel has been externally negotiated |
- // and we do not send an in-band signalling in the |
- // form of an "open" message. |
- int id; // The stream id, or SID, for SCTP data channels. -1 |
- // if unset. |
-}; |
- |
-struct DataBuffer { |
- DataBuffer(const rtc::Buffer& data, bool binary) |
- : data(data), |
- binary(binary) { |
- } |
- // For convenience for unit tests. |
- explicit DataBuffer(const std::string& text) |
- : data(text.data(), text.length()), |
- binary(false) { |
- } |
- size_t size() const { return data.size(); } |
- |
- rtc::Buffer data; |
- // Indicates if the received data contains UTF-8 or binary data. |
- // Note that the upper layers are left to verify the UTF-8 encoding. |
- // TODO(jiayl): prefer to use an enum instead of a bool. |
- bool binary; |
-}; |
- |
-class DataChannelObserver { |
- public: |
- // The data channel state have changed. |
- virtual void OnStateChange() = 0; |
- // A data buffer was successfully received. |
- virtual void OnMessage(const DataBuffer& buffer) = 0; |
- // The data channel's buffered_amount has changed. |
- virtual void OnBufferedAmountChange(uint64_t previous_amount){}; |
- |
- protected: |
- virtual ~DataChannelObserver() {} |
-}; |
- |
-class DataChannelInterface : public rtc::RefCountInterface { |
- public: |
- // Keep in sync with DataChannel.java:State and |
- // RTCDataChannel.h:RTCDataChannelState. |
- enum DataState { |
- kConnecting, |
- kOpen, // The DataChannel is ready to send data. |
- kClosing, |
- kClosed |
- }; |
- |
- static const char* DataStateString(DataState state) { |
- switch (state) { |
- case kConnecting: |
- return "connecting"; |
- case kOpen: |
- return "open"; |
- case kClosing: |
- return "closing"; |
- case kClosed: |
- return "closed"; |
- } |
- RTC_CHECK(false) << "Unknown DataChannel state: " << state; |
- return ""; |
- } |
- |
- virtual void RegisterObserver(DataChannelObserver* observer) = 0; |
- virtual void UnregisterObserver() = 0; |
- // The label attribute represents a label that can be used to distinguish this |
- // DataChannel object from other DataChannel objects. |
- virtual std::string label() const = 0; |
- virtual bool reliable() const = 0; |
- |
- // TODO(tommyw): Remove these dummy implementations when all classes have |
- // implemented these APIs. They should all just return the values the |
- // DataChannel was created with. |
- virtual bool ordered() const { return false; } |
- virtual uint16_t maxRetransmitTime() const { return 0; } |
- virtual uint16_t maxRetransmits() const { return 0; } |
- virtual std::string protocol() const { return std::string(); } |
- virtual bool negotiated() const { return false; } |
- |
- virtual int id() const = 0; |
- virtual DataState state() const = 0; |
- // The buffered_amount returns the number of bytes of application data |
- // (UTF-8 text and binary data) that have been queued using SendBuffer but |
- // have not yet been transmitted to the network. |
- virtual uint64_t buffered_amount() const = 0; |
- virtual void Close() = 0; |
- // Sends |data| to the remote peer. |
- virtual bool Send(const DataBuffer& buffer) = 0; |
- |
- protected: |
- virtual ~DataChannelInterface() {} |
-}; |
- |
-} // namespace webrtc |
- |
-#endif // TALK_APP_WEBRTC_DATACHANNELINTERFACE_H_ |