Add datagram_transport and congestion_control interface

This change introduces experimental datagram_transport interface and congestion_control interfaces. The goal is to integrate support for datagram transport in DTLS transport and set it up in a similar way we currently setup media_transport. Datagram transport will be injected in peer connection factory the same way media_transport is injected (we might even keep using the same factory which creates both media and datagram transports for now until we decided what to do next).

Bug: webrtc:9719
Change-Id: I80e70ce8d3827664ac5f5f7e55b706fe2dd2fbef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136782
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27943}
This commit is contained in:
Anton Sukhanov
2019-05-14 14:53:42 -07:00
committed by Commit Bot
parent 39068db06d
commit c136b06326
3 changed files with 169 additions and 0 deletions

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@ -86,6 +86,7 @@ rtc_static_library("libjingle_peerconnection_api") {
"bitrate_constraints.h",
"candidate.cc",
"candidate.h",
"congestion_control_interface.h",
"crypto/crypto_options.cc",
"crypto/crypto_options.h",
"crypto/frame_decryptor_interface.h",
@ -93,6 +94,7 @@ rtc_static_library("libjingle_peerconnection_api") {
"crypto_params.h",
"data_channel_interface.cc",
"data_channel_interface.h",
"datagram_transport_interface.h",
"dtls_transport_interface.cc",
"dtls_transport_interface.h",
"dtmf_sender_interface.h",

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@ -0,0 +1,67 @@
/* Copyright 2018 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 is EXPERIMENTAL interface for media and datagram transports.
#ifndef API_CONGESTION_CONTROL_INTERFACE_H_
#define API_CONGESTION_CONTROL_INTERFACE_H_
#include <memory>
#include <string>
#include <utility>
#include "api/media_transport_interface.h"
#include "api/units/data_rate.h"
namespace webrtc {
// Defines congestion control feedback interface for media and datagram
// transports.
class CongestionControlInterface {
public:
virtual ~CongestionControlInterface() = default;
// Updates allocation limits.
virtual void SetAllocatedBitrateLimits(
const MediaTransportAllocatedBitrateLimits& limits) = 0;
// Sets starting rate.
virtual void SetTargetBitrateLimits(
const MediaTransportTargetRateConstraints& target_rate_constraints) = 0;
// Intended for receive side. AddRttObserver registers an observer to be
// called for each RTT measurement, typically once per ACK. Before media
// transport is destructed the observer must be unregistered.
//
// TODO(sukhanov): Looks like AddRttObserver and RemoveRttObserver were
// never implemented for media transport, so keeping noop implementation.
virtual void AddRttObserver(MediaTransportRttObserver* observer) {}
virtual void RemoveRttObserver(MediaTransportRttObserver* observer) {}
// Adds a target bitrate observer. Before media transport is destructed
// the observer must be unregistered (by calling
// RemoveTargetTransferRateObserver).
// A newly registered observer will be called back with the latest recorded
// target rate, if available.
virtual void AddTargetTransferRateObserver(
TargetTransferRateObserver* observer) = 0;
// Removes an existing |observer| from observers. If observer was never
// registered, an error is logged and method does nothing.
virtual void RemoveTargetTransferRateObserver(
TargetTransferRateObserver* observer) = 0;
// Returns the last known target transfer rate as reported to the above
// observers.
virtual absl::optional<TargetTransferRate> GetLatestTargetTransferRate() = 0;
};
} // namespace webrtc
#endif // API_CONGESTION_CONTROL_INTERFACE_H_

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@ -0,0 +1,100 @@
/* Copyright 2018 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 is EXPERIMENTAL interface for media and datagram transports.
#ifndef API_DATAGRAM_TRANSPORT_INTERFACE_H_
#define API_DATAGRAM_TRANSPORT_INTERFACE_H_
#include <memory>
#include <string>
#include <utility>
#include "api/array_view.h"
#include "api/congestion_control_interface.h"
#include "api/media_transport_interface.h"
#include "api/rtc_error.h"
#include "api/units/data_rate.h"
namespace rtc {
class PacketTransportInternal;
} // namespace rtc
namespace webrtc {
typedef int64_t DatagramId;
// All sink methods are called on network thread.
class DatagramSinkInterface {
public:
virtual ~DatagramSinkInterface() {}
// Called when new packet is received.
virtual void OnDatagramReceived(rtc::ArrayView<const uint8_t> data) = 0;
// Called when datagram is actually sent (datragram can be delayed due
// to congestion control or fusing). |datagram_id| is same as passed in
// QuicTransportInterface::SendDatagram.
virtual void OnDatagramSent(DatagramId datagram_id) = 0;
};
// Datagram transport allows to send and receive unreliable packets (datagrams)
// and receive feedback from congestion control (via
// CongestionControlInterface). The idea is to send RTP packets as datagrams and
// have underlying implementation of datagram transport to use QUIC datagram
// protocol.
class DatagramTransportInterface {
public:
virtual ~DatagramTransportInterface() = default;
// Connect the datagram transport to the ICE transport.
// The implementation must be able to ignore incoming packets that don't
// belong to it.
virtual void Connect(rtc::PacketTransportInternal* packet_transport) = 0;
// Returns congestion control feedback interface or nullptr if datagram
// transport does not implement congestion control.
//
// Note that right now datagram transport is used without congestion control,
// but we plan to use it in the future.
virtual CongestionControlInterface* congestion_control() = 0;
// Sets a state observer callback. Before datagram transport is destroyed, the
// callback must be unregistered by setting it to nullptr.
// A newly registered callback will be called with the current state.
// Datagram transport does not invoke this callback concurrently.
virtual void SetTransportStateCallback(
MediaTransportStateCallback* callback) = 0;
// Start asynchronous send of datagram. The status returned by this method
// only pertains to the synchronous operations (e.g. serialization /
// packetization), not to the asynchronous operation.
//
// Datagrams larger than GetLargestDatagramSize() will fail and return error.
//
// Datagrams are sent in FIFO order.
virtual RTCError SendDatagram(rtc::ArrayView<const uint8_t> data,
DatagramId datagram_id) = 0;
// Returns maximum size of datagram message, does not change.
// TODO(sukhanov): Because value may be undefined before connection setup
// is complete, consider returning error when called before connection is
// established. Currently returns hardcoded const, because integration
// prototype may call before connection is established.
virtual size_t GetLargestDatagramSize() const = 0;
// Sets packet sink. Sink must be unset by calling
// SetDataTransportSink(nullptr) before the data transport is destroyed or
// before new sink is set.
virtual void SetDatagramSink(DatagramSinkInterface* sink) = 0;
};
} // namespace webrtc
#endif // API_DATAGRAM_TRANSPORT_INTERFACE_H_