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/*
* 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/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.
absl::optional<int> maxRetransmitTime;
// The max number of retransmissions.
//
// Cannot be set along with |maxRetransmitTime|.
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;
};
// 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 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_