| /* |
| * 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 <stddef.h> |
| #include <stdint.h> |
| |
| #include <string> |
| |
| #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<int> 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<int> 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> 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<int> maxRetransmitsOpt() const; |
| virtual absl::optional<int> 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. |
| 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 16MB. If Send is called while this buffer is full, the |
| // data channel will be closed abruptly. |
| // |
| // So, it's important to use buffered_amount() and OnBufferedAmountChange to |
| // ensure the data channel is used efficiently but without filling this |
| // buffer. |
| virtual bool Send(const DataBuffer& buffer) = 0; |
| |
| protected: |
| ~DataChannelInterface() override = default; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // API_DATA_CHANNEL_INTERFACE_H_ |