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}
97 lines
3.9 KiB
C++
97 lines
3.9 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_ORTC_RTPPARAMETERSCONVERSION_H_
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#define WEBRTC_ORTC_RTPPARAMETERSCONVERSION_H_
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#include <memory>
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#include <vector>
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#include "webrtc/api/rtcerror.h"
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#include "webrtc/api/rtpparameters.h"
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#include "webrtc/base/optional.h"
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#include "webrtc/pc/mediasession.h"
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#include "webrtc/media/base/codec.h"
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namespace webrtc {
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// NOTE: Some functions are templated for convenience, such that template-based
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// code dealing with AudioContentDescription and VideoContentDescription can
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// use this easily. Such methods are usable with cricket::AudioCodec and
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// cricket::VideoCodec.
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//***************************************************************************
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// Functions for converting from new webrtc:: structures to old cricket::
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// structures.
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//
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// As the return values imply, all of these functions do validation of the
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// parameters and return an error if they're invalid. It's expected that any
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// default values (such as video clock rate of 90000) have been filled by the
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// time the webrtc:: structure is being converted to the cricket:: one.
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//
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// These are expected to be used when parameters are passed into an RtpSender
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// or RtpReceiver, and need to be validated and converted so they can be
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// applied to the media engine level.
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//***************************************************************************
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// Returns error on invalid input. Certain message types are only valid for
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// certain feedback types.
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RTCErrorOr<cricket::FeedbackParam> ToCricketFeedbackParam(
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const RtcpFeedback& feedback);
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// Verifies that the codec kind is correct, and it has mandatory parameters
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// filled, with values in valid ranges.
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template <typename C>
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RTCErrorOr<C> ToCricketCodec(const RtpCodecParameters& codec);
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// Verifies that payload types aren't duplicated, in addition to normal
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// validation.
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template <typename C>
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RTCErrorOr<std::vector<C>> ToCricketCodecs(
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const std::vector<RtpCodecParameters>& codecs);
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// Validates that header extension IDs aren't duplicated.
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RTCErrorOr<cricket::RtpHeaderExtensions> ToCricketRtpHeaderExtensions(
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const std::vector<RtpHeaderExtensionParameters>& extensions);
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// SSRCs are allowed to be ommitted. This may be used for receive parameters
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// where SSRCs are unsignaled.
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RTCErrorOr<cricket::StreamParamsVec> ToCricketStreamParamsVec(
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const std::vector<RtpEncodingParameters>& encodings);
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//*****************************************************************************
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// Functions for converting from old cricket:: structures to new webrtc::
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// structures. Unlike the above functions, these are permissive with regards to
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// input validation; it's assumed that any necessary validation already
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// occurred.
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//
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// These are expected to be used either to convert from audio/video engine
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// capabilities to RtpCapabilities, or to convert from already-parsed SDP
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// (in the form of cricket:: structures) to webrtc:: structures. The latter
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// functionality is not yet implemented.
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//*****************************************************************************
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// Returns empty value if |cricket_feedback| is a feedback type not
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// supported/recognized.
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rtc::Optional<RtcpFeedback> ToRtcpFeedback(
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const cricket::FeedbackParam& cricket_feedback);
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template <typename C>
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RtpCodecCapability ToRtpCodecCapability(const C& cricket_codec);
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template <class C>
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RtpCapabilities ToRtpCapabilities(
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const std::vector<C>& cricket_codecs,
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const cricket::RtpHeaderExtensions& cricket_extensions);
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} // namespace webrtc
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#endif // WEBRTC_ORTC_RTPPARAMETERSCONVERSION_H_
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