From 6bf0a5cb5034a7e684dcc3500e841785237ce2dd Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Sun, 7 Apr 2024 19:32:43 +0200 Subject: Adding upstream version 1:115.7.0. Signed-off-by: Daniel Baumann --- third_party/libwebrtc/api/data_channel_interface.h | 202 +++++++++++++++++++++ 1 file changed, 202 insertions(+) create mode 100644 third_party/libwebrtc/api/data_channel_interface.h (limited to 'third_party/libwebrtc/api/data_channel_interface.h') diff --git a/third_party/libwebrtc/api/data_channel_interface.h b/third_party/libwebrtc/api/data_channel_interface.h new file mode 100644 index 0000000000..4f74918ff9 --- /dev/null +++ b/third_party/libwebrtc/api/data_channel_interface.h @@ -0,0 +1,202 @@ +/* + * 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_ -- cgit v1.2.3