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This is intended to ensure compatibility between Plan B and Unified Plan endpoints for the single audio - single video case. If Unified Plan is the offerer, it will add a=msid and a=ssrc MSID entries to its offer. If Unified Plan is the answerer, it will use whatever MSID signaling mechanism was used in the offer (either a=msid or a=ssrc). Bug: webrtc:7600 Change-Id: I6192dec19123fbb56f5d04540d2175c7fb30b9b6 Reviewed-on: https://webrtc-review.googlesource.com/44162 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21859}
476 lines
17 KiB
C++
476 lines
17 KiB
C++
/*
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* Copyright 2004 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 PC_SESSIONDESCRIPTION_H_
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#define PC_SESSIONDESCRIPTION_H_
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#include <string>
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#include <vector>
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#include "api/cryptoparams.h"
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#include "api/rtpparameters.h"
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#include "api/rtptransceiverinterface.h"
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#include "media/base/codec.h"
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#include "media/base/mediachannel.h"
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#include "media/base/streamparams.h"
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#include "p2p/base/transportinfo.h"
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#include "rtc_base/constructormagic.h"
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namespace cricket {
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typedef std::vector<AudioCodec> AudioCodecs;
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typedef std::vector<VideoCodec> VideoCodecs;
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typedef std::vector<DataCodec> DataCodecs;
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typedef std::vector<CryptoParams> CryptoParamsVec;
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typedef std::vector<webrtc::RtpExtension> RtpHeaderExtensions;
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// RTC4585 RTP/AVPF
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extern const char kMediaProtocolAvpf[];
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// RFC5124 RTP/SAVPF
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extern const char kMediaProtocolSavpf[];
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extern const char kMediaProtocolDtlsSavpf[];
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extern const char kMediaProtocolRtpPrefix[];
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extern const char kMediaProtocolSctp[];
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extern const char kMediaProtocolDtlsSctp[];
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extern const char kMediaProtocolUdpDtlsSctp[];
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extern const char kMediaProtocolTcpDtlsSctp[];
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// Options to control how session descriptions are generated.
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const int kAutoBandwidth = -1;
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class AudioContentDescription;
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class VideoContentDescription;
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class DataContentDescription;
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// Describes a session description media section. There are subclasses for each
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// media type (audio, video, data) that will have additional information.
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class MediaContentDescription {
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public:
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MediaContentDescription() = default;
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virtual ~MediaContentDescription() = default;
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virtual MediaType type() const = 0;
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// Try to cast this media description to an AudioContentDescription. Returns
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// nullptr if the cast fails.
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virtual AudioContentDescription* as_audio() { return nullptr; }
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virtual const AudioContentDescription* as_audio() const { return nullptr; }
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// Try to cast this media description to a VideoContentDescription. Returns
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// nullptr if the cast fails.
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virtual VideoContentDescription* as_video() { return nullptr; }
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virtual const VideoContentDescription* as_video() const { return nullptr; }
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// Try to cast this media description to a DataContentDescription. Returns
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// nullptr if the cast fails.
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virtual DataContentDescription* as_data() { return nullptr; }
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virtual const DataContentDescription* as_data() const { return nullptr; }
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virtual bool has_codecs() const = 0;
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virtual MediaContentDescription* Copy() const = 0;
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// |protocol| is the expected media transport protocol, such as RTP/AVPF,
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// RTP/SAVPF or SCTP/DTLS.
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std::string protocol() const { return protocol_; }
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void set_protocol(const std::string& protocol) { protocol_ = protocol; }
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webrtc::RtpTransceiverDirection direction() const { return direction_; }
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void set_direction(webrtc::RtpTransceiverDirection direction) {
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direction_ = direction;
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}
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bool rtcp_mux() const { return rtcp_mux_; }
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void set_rtcp_mux(bool mux) { rtcp_mux_ = mux; }
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bool rtcp_reduced_size() const { return rtcp_reduced_size_; }
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void set_rtcp_reduced_size(bool reduced_size) {
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rtcp_reduced_size_ = reduced_size;
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}
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int bandwidth() const { return bandwidth_; }
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void set_bandwidth(int bandwidth) { bandwidth_ = bandwidth; }
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const std::vector<CryptoParams>& cryptos() const { return cryptos_; }
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void AddCrypto(const CryptoParams& params) { cryptos_.push_back(params); }
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void set_cryptos(const std::vector<CryptoParams>& cryptos) {
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cryptos_ = cryptos;
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}
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const RtpHeaderExtensions& rtp_header_extensions() const {
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return rtp_header_extensions_;
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}
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void set_rtp_header_extensions(const RtpHeaderExtensions& extensions) {
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rtp_header_extensions_ = extensions;
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rtp_header_extensions_set_ = true;
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}
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void AddRtpHeaderExtension(const webrtc::RtpExtension& ext) {
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rtp_header_extensions_.push_back(ext);
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rtp_header_extensions_set_ = true;
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}
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void AddRtpHeaderExtension(const cricket::RtpHeaderExtension& ext) {
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webrtc::RtpExtension webrtc_extension;
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webrtc_extension.uri = ext.uri;
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webrtc_extension.id = ext.id;
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rtp_header_extensions_.push_back(webrtc_extension);
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rtp_header_extensions_set_ = true;
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}
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void ClearRtpHeaderExtensions() {
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rtp_header_extensions_.clear();
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rtp_header_extensions_set_ = true;
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}
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// We can't always tell if an empty list of header extensions is
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// because the other side doesn't support them, or just isn't hooked up to
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// signal them. For now we assume an empty list means no signaling, but
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// provide the ClearRtpHeaderExtensions method to allow "no support" to be
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// clearly indicated (i.e. when derived from other information).
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bool rtp_header_extensions_set() const { return rtp_header_extensions_set_; }
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const StreamParamsVec& streams() const { return streams_; }
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// TODO(pthatcher): Remove this by giving mediamessage.cc access
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// to MediaContentDescription
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StreamParamsVec& mutable_streams() { return streams_; }
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void AddStream(const StreamParams& stream) { streams_.push_back(stream); }
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// Legacy streams have an ssrc, but nothing else.
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void AddLegacyStream(uint32_t ssrc) {
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streams_.push_back(StreamParams::CreateLegacy(ssrc));
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}
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void AddLegacyStream(uint32_t ssrc, uint32_t fid_ssrc) {
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StreamParams sp = StreamParams::CreateLegacy(ssrc);
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sp.AddFidSsrc(ssrc, fid_ssrc);
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streams_.push_back(sp);
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}
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// Sets the CNAME of all StreamParams if it have not been set.
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void SetCnameIfEmpty(const std::string& cname) {
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for (cricket::StreamParamsVec::iterator it = streams_.begin();
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it != streams_.end(); ++it) {
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if (it->cname.empty())
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it->cname = cname;
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}
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}
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uint32_t first_ssrc() const {
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if (streams_.empty()) {
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return 0;
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}
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return streams_[0].first_ssrc();
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}
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bool has_ssrcs() const {
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if (streams_.empty()) {
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return false;
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}
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return streams_[0].has_ssrcs();
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}
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void set_conference_mode(bool enable) { conference_mode_ = enable; }
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bool conference_mode() const { return conference_mode_; }
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// https://tools.ietf.org/html/rfc4566#section-5.7
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// May be present at the media or session level of SDP. If present at both
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// levels, the media-level attribute overwrites the session-level one.
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void set_connection_address(const rtc::SocketAddress& address) {
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connection_address_ = address;
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}
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const rtc::SocketAddress& connection_address() const {
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return connection_address_;
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}
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protected:
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bool rtcp_mux_ = false;
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bool rtcp_reduced_size_ = false;
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int bandwidth_ = kAutoBandwidth;
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std::string protocol_;
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std::vector<CryptoParams> cryptos_;
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std::vector<webrtc::RtpExtension> rtp_header_extensions_;
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bool rtp_header_extensions_set_ = false;
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StreamParamsVec streams_;
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bool conference_mode_ = false;
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webrtc::RtpTransceiverDirection direction_ =
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webrtc::RtpTransceiverDirection::kSendRecv;
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rtc::SocketAddress connection_address_;
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};
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// TODO(bugs.webrtc.org/8620): Remove this alias once downstream projects have
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// updated.
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using ContentDescription = MediaContentDescription;
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template <class C>
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class MediaContentDescriptionImpl : public MediaContentDescription {
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public:
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typedef C CodecType;
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// Codecs should be in preference order (most preferred codec first).
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const std::vector<C>& codecs() const { return codecs_; }
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void set_codecs(const std::vector<C>& codecs) { codecs_ = codecs; }
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virtual bool has_codecs() const { return !codecs_.empty(); }
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bool HasCodec(int id) {
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bool found = false;
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for (typename std::vector<C>::iterator iter = codecs_.begin();
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iter != codecs_.end(); ++iter) {
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if (iter->id == id) {
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found = true;
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break;
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}
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}
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return found;
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}
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void AddCodec(const C& codec) { codecs_.push_back(codec); }
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void AddOrReplaceCodec(const C& codec) {
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for (typename std::vector<C>::iterator iter = codecs_.begin();
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iter != codecs_.end(); ++iter) {
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if (iter->id == codec.id) {
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*iter = codec;
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return;
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}
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}
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AddCodec(codec);
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}
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void AddCodecs(const std::vector<C>& codecs) {
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typename std::vector<C>::const_iterator codec;
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for (codec = codecs.begin(); codec != codecs.end(); ++codec) {
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AddCodec(*codec);
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}
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}
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private:
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std::vector<C> codecs_;
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};
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class AudioContentDescription : public MediaContentDescriptionImpl<AudioCodec> {
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public:
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AudioContentDescription() {}
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virtual AudioContentDescription* Copy() const {
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return new AudioContentDescription(*this);
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}
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virtual MediaType type() const { return MEDIA_TYPE_AUDIO; }
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virtual AudioContentDescription* as_audio() { return this; }
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virtual const AudioContentDescription* as_audio() const { return this; }
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};
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class VideoContentDescription : public MediaContentDescriptionImpl<VideoCodec> {
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public:
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virtual VideoContentDescription* Copy() const {
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return new VideoContentDescription(*this);
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}
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virtual MediaType type() const { return MEDIA_TYPE_VIDEO; }
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virtual VideoContentDescription* as_video() { return this; }
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virtual const VideoContentDescription* as_video() const { return this; }
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};
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class DataContentDescription : public MediaContentDescriptionImpl<DataCodec> {
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public:
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DataContentDescription() {}
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virtual DataContentDescription* Copy() const {
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return new DataContentDescription(*this);
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}
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virtual MediaType type() const { return MEDIA_TYPE_DATA; }
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virtual DataContentDescription* as_data() { return this; }
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virtual const DataContentDescription* as_data() const { return this; }
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bool use_sctpmap() const { return use_sctpmap_; }
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void set_use_sctpmap(bool enable) { use_sctpmap_ = enable; }
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private:
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bool use_sctpmap_ = true;
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};
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// Protocol used for encoding media. This is the "top level" protocol that may
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// be wrapped by zero or many transport protocols (UDP, ICE, etc.).
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enum class MediaProtocolType {
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kRtp, // Section will use the RTP protocol (e.g., for audio or video).
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// https://tools.ietf.org/html/rfc3550
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kSctp // Section will use the SCTP protocol (e.g., for a data channel).
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// https://tools.ietf.org/html/rfc4960
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};
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// TODO(bugs.webrtc.org/8620): Remove once downstream projects have updated.
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constexpr MediaProtocolType NS_JINGLE_RTP = MediaProtocolType::kRtp;
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constexpr MediaProtocolType NS_JINGLE_DRAFT_SCTP = MediaProtocolType::kSctp;
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// Represents a session description section. Most information about the section
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// is stored in the description, which is a subclass of MediaContentDescription.
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struct ContentInfo {
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friend class SessionDescription;
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explicit ContentInfo(MediaProtocolType type) : type(type) {}
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// Alias for |name|.
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std::string mid() const { return name; }
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void set_mid(const std::string& mid) { this->name = mid; }
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// Alias for |description|.
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MediaContentDescription* media_description() { return description; }
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const MediaContentDescription* media_description() const {
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return description;
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}
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void set_media_description(MediaContentDescription* desc) {
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description = desc;
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}
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// TODO(bugs.webrtc.org/8620): Rename this to mid.
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std::string name;
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MediaProtocolType type;
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bool rejected = false;
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bool bundle_only = false;
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// TODO(bugs.webrtc.org/8620): Switch to the getter and setter, and make this
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// private.
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MediaContentDescription* description = nullptr;
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};
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typedef std::vector<std::string> ContentNames;
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// This class provides a mechanism to aggregate different media contents into a
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// group. This group can also be shared with the peers in a pre-defined format.
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// GroupInfo should be populated only with the |content_name| of the
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// MediaDescription.
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class ContentGroup {
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public:
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explicit ContentGroup(const std::string& semantics);
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ContentGroup(const ContentGroup&);
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ContentGroup(ContentGroup&&);
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ContentGroup& operator=(const ContentGroup&);
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ContentGroup& operator=(ContentGroup&&);
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~ContentGroup();
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const std::string& semantics() const { return semantics_; }
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const ContentNames& content_names() const { return content_names_; }
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const std::string* FirstContentName() const;
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bool HasContentName(const std::string& content_name) const;
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void AddContentName(const std::string& content_name);
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bool RemoveContentName(const std::string& content_name);
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private:
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std::string semantics_;
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ContentNames content_names_;
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};
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typedef std::vector<ContentInfo> ContentInfos;
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typedef std::vector<ContentGroup> ContentGroups;
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const ContentInfo* FindContentInfoByName(const ContentInfos& contents,
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const std::string& name);
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const ContentInfo* FindContentInfoByType(const ContentInfos& contents,
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const std::string& type);
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// Determines how the MSID will be signaled in the SDP. These can be used as
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// flags to indicate both or none.
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enum MsidSignaling {
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// Signal MSID with one a=msid line in the media section.
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kMsidSignalingMediaSection = 0x1,
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// Signal MSID with a=ssrc: msid lines in the media section.
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kMsidSignalingSsrcAttribute = 0x2
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};
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// Describes a collection of contents, each with its own name and
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// type. Analogous to a <jingle> or <session> stanza. Assumes that
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// contents are unique be name, but doesn't enforce that.
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class SessionDescription {
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public:
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SessionDescription();
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explicit SessionDescription(const ContentInfos& contents);
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SessionDescription(const ContentInfos& contents, const ContentGroups& groups);
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SessionDescription(const ContentInfos& contents,
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const TransportInfos& transports,
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const ContentGroups& groups);
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~SessionDescription();
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SessionDescription* Copy() const;
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// Content accessors.
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const ContentInfos& contents() const { return contents_; }
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ContentInfos& contents() { return contents_; }
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const ContentInfo* GetContentByName(const std::string& name) const;
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ContentInfo* GetContentByName(const std::string& name);
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const MediaContentDescription* GetContentDescriptionByName(
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const std::string& name) const;
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MediaContentDescription* GetContentDescriptionByName(const std::string& name);
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const ContentInfo* FirstContentByType(MediaProtocolType type) const;
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const ContentInfo* FirstContent() const;
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// Content mutators.
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// Adds a content to this description. Takes ownership of ContentDescription*.
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void AddContent(const std::string& name,
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MediaProtocolType type,
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MediaContentDescription* description);
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void AddContent(const std::string& name,
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MediaProtocolType type,
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bool rejected,
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MediaContentDescription* description);
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void AddContent(const std::string& name,
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MediaProtocolType type,
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bool rejected,
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bool bundle_only,
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MediaContentDescription* description);
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bool RemoveContentByName(const std::string& name);
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// Transport accessors.
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const TransportInfos& transport_infos() const { return transport_infos_; }
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TransportInfos& transport_infos() { return transport_infos_; }
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const TransportInfo* GetTransportInfoByName(const std::string& name) const;
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TransportInfo* GetTransportInfoByName(const std::string& name);
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const TransportDescription* GetTransportDescriptionByName(
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const std::string& name) const {
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const TransportInfo* tinfo = GetTransportInfoByName(name);
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return tinfo ? &tinfo->description : NULL;
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}
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// Transport mutators.
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void set_transport_infos(const TransportInfos& transport_infos) {
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transport_infos_ = transport_infos;
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}
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// Adds a TransportInfo to this description.
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// Returns false if a TransportInfo with the same name already exists.
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bool AddTransportInfo(const TransportInfo& transport_info);
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bool RemoveTransportInfoByName(const std::string& name);
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// Group accessors.
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const ContentGroups& groups() const { return content_groups_; }
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const ContentGroup* GetGroupByName(const std::string& name) const;
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bool HasGroup(const std::string& name) const;
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// Group mutators.
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void AddGroup(const ContentGroup& group) { content_groups_.push_back(group); }
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// Remove the first group with the same semantics specified by |name|.
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void RemoveGroupByName(const std::string& name);
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// Global attributes.
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void set_msid_supported(bool supported) { msid_supported_ = supported; }
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bool msid_supported() const { return msid_supported_; }
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// Determines how the MSIDs were/will be signaled. Flag value composed of
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// MsidSignaling bits (see enum above).
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void set_msid_signaling(int msid_signaling) {
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msid_signaling_ = msid_signaling;
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}
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int msid_signaling() const { return msid_signaling_; }
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private:
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SessionDescription(const SessionDescription&);
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ContentInfos contents_;
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TransportInfos transport_infos_;
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ContentGroups content_groups_;
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bool msid_supported_ = true;
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// Default to what Plan B would do.
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// TODO(bugs.webrtc.org/8530): Change default to kMsidSignalingMediaSection.
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int msid_signaling_ = kMsidSignalingSsrcAttribute;
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};
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// Indicates whether a session description was sent by the local client or
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// received from the remote client.
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enum ContentSource { CS_LOCAL, CS_REMOTE };
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} // namespace cricket
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#endif // PC_SESSIONDESCRIPTION_H_
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