e814a0dee0
This CL adds the following interfaces: * RtpTransportController * RtpTransport * RtpSender * RtpReceiver They're implemented on top of the "BaseChannel" object, which is normally used in a PeerConnection, and roughly corresponds to an SDP "m=" section. As a result of this, there are several limitations: * You can only have one of each type of sender and receiver (audio/video) on top of the same transport controller. * The sender/receiver with the same media type must use the same RTP transport. * You can't change the transport after creating the sender or receiver. * Some of the parameters aren't supported. Later, these "adapter" objects will be gradually replaced by real objects that don't have these limitations, as "BaseChannel", "MediaChannel" and related code is restructured. In this CL, we essentially have: ORTC adapter objects -> BaseChannel -> Media engine PeerConnection -> BaseChannel -> Media engine And later we hope to have simply: PeerConnection -> "Real" ORTC objects -> Media engine See the linked bug for more context. BUG=webrtc:7013 TBR=stefan@webrtc.org Review-Url: https://codereview.webrtc.org/2675173003 Cr-Commit-Position: refs/heads/master@{#16842}
96 lines
3.5 KiB
C++
96 lines
3.5 KiB
C++
/*
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* Copyright 2017 The WebRTC Project Authors. All rights reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_P2P_BASE_PACKETTRANSPORTINTERNAL_H_
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#define WEBRTC_P2P_BASE_PACKETTRANSPORTINTERNAL_H_
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#include <string>
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#include <vector>
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// This is included for PacketOptions.
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#include "webrtc/base/asyncpacketsocket.h"
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#include "webrtc/base/sigslot.h"
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#include "webrtc/base/socket.h"
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namespace cricket {
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class TransportChannel;
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}
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namespace rtc {
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struct PacketOptions;
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struct PacketTime;
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struct SentPacket;
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class PacketTransportInternal : public sigslot::has_slots<> {
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public:
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// Identify the object for logging and debug purpose.
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virtual std::string debug_name() const = 0;
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// The transport has been established.
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virtual bool writable() const = 0;
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// The transport has received a packet in the last X milliseconds, here X is
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// configured by each implementation.
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virtual bool receiving() const = 0;
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// Attempts to send the given packet.
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// The return value is < 0 on failure. The return value in failure case is not
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// descriptive. Depending on failure cause and implementation details
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// GetError() returns an descriptive errno.h error value.
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// This mimics posix socket send() or sendto() behavior.
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// TODO(johan): Reliable, meaningful, consistent error codes for all
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// implementations would be nice.
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// TODO(johan): Remove the default argument once channel code is updated.
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virtual int SendPacket(const char* data,
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size_t len,
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const rtc::PacketOptions& options,
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int flags = 0) = 0;
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// Sets a socket option. Note that not all options are
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// supported by all transport types.
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virtual int SetOption(rtc::Socket::Option opt, int value) = 0;
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// TODO(pthatcher): Once Chrome's MockPacketTransportInterface implements
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// this, remove the default implementation.
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virtual bool GetOption(rtc::Socket::Option opt, int* value) { return false; }
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// Returns the most recent error that occurred on this channel.
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virtual int GetError() = 0;
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// Emitted when the writable state, represented by |writable()|, changes.
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sigslot::signal1<PacketTransportInternal*> SignalWritableState;
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// Emitted when the PacketTransportInternal is ready to send packets. "Ready
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// to send" is more sensitive than the writable state; a transport may be
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// writable, but temporarily not able to send packets. For example, the
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// underlying transport's socket buffer may be full, as indicated by
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// SendPacket's return code and/or GetError.
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sigslot::signal1<PacketTransportInternal*> SignalReadyToSend;
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// Emitted when receiving state changes to true.
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sigslot::signal1<PacketTransportInternal*> SignalReceivingState;
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// Signalled each time a packet is received on this channel.
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sigslot::signal5<PacketTransportInternal*,
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const char*,
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size_t,
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const rtc::PacketTime&,
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int>
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SignalReadPacket;
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// Signalled each time a packet is sent on this channel.
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sigslot::signal2<PacketTransportInternal*, const rtc::SentPacket&>
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SignalSentPacket;
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};
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} // namespace rtc
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#endif // WEBRTC_P2P_BASE_PACKETTRANSPORTINTERNAL_H_
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