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10542f21c8
Mechanically generated by running this command: tools_webrtc/do-renames.sh update all-renames.txt && git cl format Then manually updating: tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc Bug: webrtc:10159 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833 Reviewed-on: https://webrtc-review.googlesource.com/c/115653 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26226}
94 lines
3.5 KiB
C++
94 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 API_ORTC_RTP_TRANSPORT_INTERFACE_H_
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#define API_ORTC_RTP_TRANSPORT_INTERFACE_H_
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#include <string>
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#include "absl/types/optional.h"
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#include "api/ortc/packet_transport_interface.h"
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#include "api/rtc_error.h"
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#include "api/rtp_headers.h"
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#include "api/rtp_parameters.h"
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namespace webrtc {
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class RtpTransportAdapter;
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struct RtpTransportParameters final {
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RtcpParameters rtcp;
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// Enabled periodic sending of keep-alive packets, that help prevent timeouts
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// on the network level, such as NAT bindings. See RFC6263 section 4.6.
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RtpKeepAliveConfig keepalive;
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bool operator==(const RtpTransportParameters& o) const {
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return rtcp == o.rtcp && keepalive == o.keepalive;
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}
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bool operator!=(const RtpTransportParameters& o) const {
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return !(*this == o);
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}
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};
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// Base class for different types of RTP transports that can be created by an
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// OrtcFactory. Used by RtpSenders/RtpReceivers.
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//
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// This is not present in the standard ORTC API, but exists here for a few
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// reasons. Firstly, it allows different types of RTP transports to be used:
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// DTLS-SRTP (which is required for the web), but also SDES-SRTP and
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// unencrypted RTP. It also simplifies the handling of RTCP muxing, and
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// provides a better API point for it.
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//
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// Note that Edge's implementation of ORTC provides a similar API point, called
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// RTCSrtpSdesTransport:
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// https://msdn.microsoft.com/en-us/library/mt502527(v=vs.85).aspx
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class RtpTransportInterface {
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public:
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virtual ~RtpTransportInterface() {}
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// Returns packet transport that's used to send RTP packets.
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virtual PacketTransportInterface* GetRtpPacketTransport() const = 0;
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// Returns separate packet transport that's used to send RTCP packets. If
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// RTCP multiplexing is being used, returns null.
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virtual PacketTransportInterface* GetRtcpPacketTransport() const = 0;
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// Set/get RTP/RTCP transport params. Can be used to enable RTCP muxing or
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// reduced-size RTCP if initially not enabled.
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//
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// Changing |mux| from "true" to "false" is not allowed, and changing the
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// CNAME is currently unsupported.
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// RTP keep-alive settings need to be set before before an RtpSender has
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// started sending, altering the payload type or timeout interval after this
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// point is not supported. The parameters must also match across all RTP
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// transports for a given RTP transport controller.
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virtual RTCError SetParameters(const RtpTransportParameters& parameters) = 0;
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// Returns last set or constructed-with parameters. If |cname| was empty in
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// construction, the generated CNAME will be present in the returned
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// parameters (see above).
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virtual RtpTransportParameters GetParameters() const = 0;
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protected:
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// Only for internal use. Returns a pointer to an internal interface, for use
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// by the implementation.
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virtual RtpTransportAdapter* GetInternal() = 0;
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// Classes that can use this internal interface.
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friend class OrtcFactory;
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friend class OrtcRtpSenderAdapter;
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friend class OrtcRtpReceiverAdapter;
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friend class RtpTransportControllerAdapter;
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friend class RtpTransportAdapter;
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};
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} // namespace webrtc
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#endif // API_ORTC_RTP_TRANSPORT_INTERFACE_H_
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