/* * Copyright 2012 The WebRTC project authors. All Rights Reserved. * * Use of this source code is governed by a BSD-style license * that can be found in the LICENSE file in the root of the source * tree. An additional intellectual property rights grant can be found * in the file PATENTS. All contributing project authors may * be found in the AUTHORS file in the root of the source tree. */ // This file contains interfaces for DataChannels // http://dev.w3.org/2011/webrtc/editor/webrtc.html#rtcdatachannel #ifndef API_DATA_CHANNEL_INTERFACE_H_ #define API_DATA_CHANNEL_INTERFACE_H_ #include #include #include #include "absl/types/optional.h" #include "api/priority.h" #include "api/rtc_error.h" #include "rtc_base/checks.h" #include "rtc_base/copy_on_write_buffer.h" #include "rtc_base/ref_count.h" #include "rtc_base/system/rtc_export.h" namespace webrtc { // C++ version of: https://www.w3.org/TR/webrtc/#idl-def-rtcdatachannelinit // TODO(deadbeef): Use absl::optional for the "-1 if unset" things. struct DataChannelInit { // Deprecated. Reliability is assumed, and channel will be unreliable if // maxRetransmitTime or MaxRetransmits is set. bool reliable = false; // True if ordered delivery is required. bool ordered = true; // The max period of time in milliseconds in which retransmissions will be // sent. After this time, no more retransmissions will be sent. // // Cannot be set along with `maxRetransmits`. // This is called `maxPacketLifeTime` in the WebRTC JS API. // Negative values are ignored, and positive values are clamped to [0-65535] absl::optional maxRetransmitTime; // The max number of retransmissions. // // Cannot be set along with `maxRetransmitTime`. // Negative values are ignored, and positive values are clamped to [0-65535] absl::optional maxRetransmits; // This is set by the application and opaque to the WebRTC implementation. std::string protocol; // True if the channel has been externally negotiated and we do not send an // in-band signalling in the form of an "open" message. If this is true, `id` // below must be set; otherwise it should be unset and will be negotiated // in-band. bool negotiated = false; // The stream id, or SID, for SCTP data channels. -1 if unset (see above). int id = -1; // https://w3c.github.io/webrtc-priority/#new-rtcdatachannelinit-member absl::optional priority; }; // At the JavaScript level, data can be passed in as a string or a blob, so // this structure's `binary` flag tells whether the data should be interpreted // as binary or text. struct DataBuffer { DataBuffer(const rtc::CopyOnWriteBuffer& 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::CopyOnWriteBuffer 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; }; // Used to implement RTCDataChannel events. // // The code responding to these callbacks should unwind the stack before // using any other webrtc APIs; re-entrancy is not supported. 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 sent_data_size) {} protected: virtual ~DataChannelObserver() = default; }; class RTC_EXPORT DataChannelInterface : public rtc::RefCountInterface { public: // C++ version of: https://www.w3.org/TR/webrtc/#idl-def-rtcdatachannelstate // Unlikely to change, but 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 ""; } // Used to receive events from the data channel. Only one observer can be // registered at a time. UnregisterObserver should be called before the // observer object is destroyed. 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; // The accessors below simply return the properties from the DataChannelInit // the data channel was constructed with. virtual bool reliable() const = 0; // TODO(deadbeef): 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; // TODO(hta): Deprecate and remove the following two functions. virtual uint16_t maxRetransmitTime() const; virtual uint16_t maxRetransmits() const; virtual absl::optional maxRetransmitsOpt() const; virtual absl::optional maxPacketLifeTime() const; virtual std::string protocol() const; virtual bool negotiated() const; // Returns the ID from the DataChannelInit, if it was negotiated out-of-band. // If negotiated in-band, this ID will be populated once the DTLS role is // determined, and until then this will return -1. virtual int id() const = 0; virtual Priority priority() const { return Priority::kLow; } virtual DataState state() const = 0; // When state is kClosed, and the DataChannel was not closed using // the closing procedure, returns the error information about the closing. // The default implementation returns "no error". virtual RTCError error() const { return RTCError(); } virtual uint32_t messages_sent() const = 0; virtual uint64_t bytes_sent() const = 0; virtual uint32_t messages_received() const = 0; virtual uint64_t bytes_received() const = 0; // Returns the number of bytes of application data (UTF-8 text and binary // data) that have been queued using Send but have not yet been processed at // the SCTP level. See comment above Send below. // Values are less or equal to MaxSendQueueSize(). virtual uint64_t buffered_amount() const = 0; // Begins the graceful data channel closing procedure. See: // https://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-13#section-6.7 virtual void Close() = 0; // Sends `data` to the remote peer. If the data can't be sent at the SCTP // level (due to congestion control), it's buffered at the data channel level, // up to a maximum of MaxSendQueueSize(). // Returns false if the data channel is not in open state or if the send // buffer is full. // TODO(webrtc:13289): Return an RTCError with information about the failure. virtual bool Send(const DataBuffer& buffer) = 0; // Amount of bytes that can be queued for sending on the data channel. // Those are bytes that have not yet been processed at the SCTP level. static uint64_t MaxSendQueueSize(); protected: ~DataChannelInterface() override = default; }; } // namespace webrtc #endif // API_DATA_CHANNEL_INTERFACE_H_