webrtc/api/data_channel_interface.h
Marina Ciocea e448a3fb54 Update DataChannel bufferedamount implementation.
Call DataChannelObserver::OnBufferedAmountChange on each successful send.
Previously, the observer would get notified of buffered amount changes only when
queued send data is consumed. Data gets queued only if it cannot be sent right
away. According to the WebRTC standard[1], bufferedamount should be increased
before each sent and decreased after each successful sent. Update implementation
to be standard compliant.

Design doc: http://doc/1lorHBn-GMn5U0T0RQANxrsW0pXhw8XGZM-xZyVUOW90

[1] https://w3c.github.io/webrtc-pc/#dom-datachannel-bufferedamount

Bug: chromium:878682
Change-Id: Ife009d30c4a18dced9a54cf600a445bb1f02561d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123237
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27057}
2019-03-11 13:21:36 +00:00

180 lines
6.5 KiB
C++

/*
* 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 "rtc_base/checks.h"
#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/ref_count.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. -1 if unset.
//
// Cannot be set along with |maxRetransmits|.
int maxRetransmitTime = -1;
// The max number of retransmissions. -1 if unset.
//
// Cannot be set along with |maxRetransmitTime|.
int maxRetransmits = -1;
// 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 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;
virtual uint16_t maxRetransmitTime() const;
virtual uint16_t maxRetransmits() 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;
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_