615 lines
22 KiB
C++
615 lines
22 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_SESSION_DESCRIPTION_H_
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#define PC_SESSION_DESCRIPTION_H_
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#include <stddef.h>
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#include <stdint.h>
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#include <iosfwd>
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#include <memory>
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#include <string>
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#include <utility>
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#include <vector>
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#include "absl/memory/memory.h"
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#include "api/crypto_params.h"
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#include "api/media_types.h"
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#include "api/rtp_parameters.h"
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#include "api/rtp_transceiver_interface.h"
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#include "media/base/media_channel.h"
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#include "media/base/stream_params.h"
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#include "p2p/base/transport_description.h"
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#include "p2p/base/transport_info.h"
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#include "pc/media_protocol_names.h"
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#include "pc/simulcast_description.h"
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#include "rtc_base/deprecation.h"
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#include "rtc_base/socket_address.h"
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#include "rtc_base/system/rtc_export.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<RtpDataCodec> RtpDataCodecs;
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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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// 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 RtpDataContentDescription;
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class SctpDataContentDescription;
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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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virtual RtpDataContentDescription* as_rtp_data() { return nullptr; }
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virtual const RtpDataContentDescription* as_rtp_data() const {
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return nullptr;
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}
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virtual SctpDataContentDescription* as_sctp() { return nullptr; }
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virtual const SctpDataContentDescription* as_sctp() const { return nullptr; }
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virtual bool has_codecs() const = 0;
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// Copy operator that returns an unique_ptr.
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// Not a virtual function.
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// If a type-specific variant of Clone() is desired, override it, or
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// simply use std::make_unique<typename>(*this) instead of Clone().
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std::unique_ptr<MediaContentDescription> Clone() const {
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return absl::WrapUnique(CloneInternal());
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}
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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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virtual std::string protocol() const { return protocol_; }
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virtual void set_protocol(const std::string& protocol) {
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protocol_ = protocol;
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}
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virtual webrtc::RtpTransceiverDirection direction() const {
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return direction_;
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}
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virtual void set_direction(webrtc::RtpTransceiverDirection direction) {
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direction_ = direction;
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}
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virtual bool rtcp_mux() const { return rtcp_mux_; }
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virtual void set_rtcp_mux(bool mux) { rtcp_mux_ = mux; }
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virtual bool rtcp_reduced_size() const { return rtcp_reduced_size_; }
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virtual 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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// Indicates support for the remote network estimate packet type. This
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// functionality is experimental and subject to change without notice.
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virtual bool remote_estimate() const { return remote_estimate_; }
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virtual void set_remote_estimate(bool remote_estimate) {
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remote_estimate_ = remote_estimate;
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}
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virtual int bandwidth() const { return bandwidth_; }
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virtual void set_bandwidth(int bandwidth) { bandwidth_ = bandwidth; }
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virtual const std::vector<CryptoParams>& cryptos() const { return cryptos_; }
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virtual void AddCrypto(const CryptoParams& params) {
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cryptos_.push_back(params);
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}
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virtual void set_cryptos(const std::vector<CryptoParams>& cryptos) {
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cryptos_ = cryptos;
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}
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virtual const RtpHeaderExtensions& rtp_header_extensions() const {
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return rtp_header_extensions_;
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}
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virtual void set_rtp_header_extensions(
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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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virtual 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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virtual 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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virtual bool rtp_header_extensions_set() const {
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return rtp_header_extensions_set_;
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}
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virtual const StreamParamsVec& streams() const { return send_streams_; }
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// TODO(pthatcher): Remove this by giving mediamessage.cc access
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// to MediaContentDescription
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virtual StreamParamsVec& mutable_streams() { return send_streams_; }
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virtual void AddStream(const StreamParams& stream) {
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send_streams_.push_back(stream);
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}
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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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AddStream(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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AddStream(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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virtual void SetCnameIfEmpty(const std::string& cname) {
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for (cricket::StreamParamsVec::iterator it = send_streams_.begin();
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it != send_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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virtual uint32_t first_ssrc() const {
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if (send_streams_.empty()) {
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return 0;
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}
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return send_streams_[0].first_ssrc();
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}
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virtual bool has_ssrcs() const {
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if (send_streams_.empty()) {
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return false;
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}
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return send_streams_[0].has_ssrcs();
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}
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virtual void set_conference_mode(bool enable) { conference_mode_ = enable; }
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virtual 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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virtual void set_connection_address(const rtc::SocketAddress& address) {
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connection_address_ = address;
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}
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virtual const rtc::SocketAddress& connection_address() const {
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return connection_address_;
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}
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// Determines if it's allowed to mix one- and two-byte rtp header extensions
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// within the same rtp stream.
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enum ExtmapAllowMixed { kNo, kSession, kMedia };
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virtual void set_extmap_allow_mixed_enum(
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ExtmapAllowMixed new_extmap_allow_mixed) {
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if (new_extmap_allow_mixed == kMedia &&
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extmap_allow_mixed_enum_ == kSession) {
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// Do not downgrade from session level to media level.
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return;
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}
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extmap_allow_mixed_enum_ = new_extmap_allow_mixed;
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}
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virtual ExtmapAllowMixed extmap_allow_mixed_enum() const {
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return extmap_allow_mixed_enum_;
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}
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virtual bool extmap_allow_mixed() const {
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return extmap_allow_mixed_enum_ != kNo;
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}
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// Simulcast functionality.
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virtual bool HasSimulcast() const { return !simulcast_.empty(); }
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virtual SimulcastDescription& simulcast_description() { return simulcast_; }
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virtual const SimulcastDescription& simulcast_description() const {
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return simulcast_;
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}
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virtual void set_simulcast_description(
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const SimulcastDescription& simulcast) {
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simulcast_ = simulcast;
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}
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virtual const std::vector<RidDescription>& receive_rids() const {
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return receive_rids_;
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}
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virtual void set_receive_rids(const std::vector<RidDescription>& rids) {
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receive_rids_ = rids;
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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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bool remote_estimate_ = 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 send_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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// Mixed one- and two-byte header not included in offer on media level or
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// session level, but we will respond that we support it. The plan is to add
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// it to our offer on session level. See todo in SessionDescription.
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ExtmapAllowMixed extmap_allow_mixed_enum_ = kNo;
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SimulcastDescription simulcast_;
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std::vector<RidDescription> receive_rids_;
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private:
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// Copy function that returns a raw pointer. Caller will assert ownership.
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// Should only be called by the Clone() function. Must be implemented
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// by each final subclass.
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virtual MediaContentDescription* CloneInternal() const = 0;
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};
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template <class C>
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class MediaContentDescriptionImpl : public MediaContentDescription {
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public:
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void set_protocol(const std::string& protocol) override {
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RTC_DCHECK(IsRtpProtocol(protocol));
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protocol_ = protocol;
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}
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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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virtual const std::vector<C>& codecs() const { return codecs_; }
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virtual void set_codecs(const std::vector<C>& codecs) { codecs_ = codecs; }
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bool has_codecs() const override { return !codecs_.empty(); }
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virtual 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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virtual void AddCodec(const C& codec) { codecs_.push_back(codec); }
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virtual 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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virtual 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 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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private:
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virtual AudioContentDescription* CloneInternal() const {
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return new AudioContentDescription(*this);
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}
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};
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class VideoContentDescription : public MediaContentDescriptionImpl<VideoCodec> {
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public:
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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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private:
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virtual VideoContentDescription* CloneInternal() const {
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return new VideoContentDescription(*this);
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}
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};
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class RtpDataContentDescription
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: public MediaContentDescriptionImpl<RtpDataCodec> {
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public:
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RtpDataContentDescription() {}
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MediaType type() const override { return MEDIA_TYPE_DATA; }
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RtpDataContentDescription* as_rtp_data() override { return this; }
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const RtpDataContentDescription* as_rtp_data() const override { return this; }
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private:
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RtpDataContentDescription* CloneInternal() const override {
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return new RtpDataContentDescription(*this);
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}
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};
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class SctpDataContentDescription : public MediaContentDescription {
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public:
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SctpDataContentDescription() {}
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SctpDataContentDescription(const SctpDataContentDescription& o)
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: MediaContentDescription(o),
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use_sctpmap_(o.use_sctpmap_),
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port_(o.port_),
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max_message_size_(o.max_message_size_) {}
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MediaType type() const override { return MEDIA_TYPE_DATA; }
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SctpDataContentDescription* as_sctp() override { return this; }
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const SctpDataContentDescription* as_sctp() const override { return this; }
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bool has_codecs() const override { return false; }
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void set_protocol(const std::string& protocol) override {
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RTC_DCHECK(IsSctpProtocol(protocol));
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protocol_ = protocol;
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}
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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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int port() const { return port_; }
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void set_port(int port) { port_ = port; }
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int max_message_size() const { return max_message_size_; }
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void set_max_message_size(int max_message_size) {
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max_message_size_ = max_message_size;
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}
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private:
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SctpDataContentDescription* CloneInternal() const override {
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return new SctpDataContentDescription(*this);
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}
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bool use_sctpmap_ = true; // Note: "true" is no longer conformant.
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// Defaults should be constants imported from SCTP. Quick hack.
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int port_ = 5000;
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// draft-ietf-mmusic-sdp-sctp-23: Max message size default is 64K
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int max_message_size_ = 64 * 1024;
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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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// 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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// Owns the description.
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class RTC_EXPORT ContentInfo {
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public:
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explicit ContentInfo(MediaProtocolType type) : type(type) {}
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~ContentInfo();
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// Copy
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ContentInfo(const ContentInfo& o);
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ContentInfo& operator=(const ContentInfo& o);
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ContentInfo(ContentInfo&& o) = default;
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ContentInfo& operator=(ContentInfo&& o) = default;
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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();
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const MediaContentDescription* media_description() const;
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void set_media_description(std::unique_ptr<MediaContentDescription> desc) {
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description_ = std::move(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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private:
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friend class SessionDescription;
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std::unique_ptr<MediaContentDescription> description_;
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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
|
|
};
|
|
|
|
// Describes a collection of contents, each with its own name and
|
|
// type. Analogous to a <jingle> or <session> stanza. Assumes that
|
|
// contents are unique be name, but doesn't enforce that.
|
|
class SessionDescription {
|
|
public:
|
|
SessionDescription();
|
|
~SessionDescription();
|
|
|
|
std::unique_ptr<SessionDescription> Clone() const;
|
|
|
|
// Content accessors.
|
|
const ContentInfos& contents() const { return contents_; }
|
|
ContentInfos& contents() { return contents_; }
|
|
const ContentInfo* GetContentByName(const std::string& name) const;
|
|
ContentInfo* GetContentByName(const std::string& name);
|
|
const MediaContentDescription* GetContentDescriptionByName(
|
|
const std::string& name) const;
|
|
MediaContentDescription* GetContentDescriptionByName(const std::string& name);
|
|
const ContentInfo* FirstContentByType(MediaProtocolType type) const;
|
|
const ContentInfo* FirstContent() const;
|
|
|
|
// Content mutators.
|
|
// Adds a content to this description. Takes ownership of ContentDescription*.
|
|
void AddContent(const std::string& name,
|
|
MediaProtocolType type,
|
|
std::unique_ptr<MediaContentDescription> description);
|
|
void AddContent(const std::string& name,
|
|
MediaProtocolType type,
|
|
bool rejected,
|
|
std::unique_ptr<MediaContentDescription> description);
|
|
void AddContent(const std::string& name,
|
|
MediaProtocolType type,
|
|
bool rejected,
|
|
bool bundle_only,
|
|
std::unique_ptr<MediaContentDescription> description);
|
|
void AddContent(ContentInfo&& content);
|
|
|
|
bool RemoveContentByName(const std::string& name);
|
|
|
|
// Transport accessors.
|
|
const TransportInfos& transport_infos() const { return transport_infos_; }
|
|
TransportInfos& transport_infos() { return transport_infos_; }
|
|
const TransportInfo* GetTransportInfoByName(const std::string& name) const;
|
|
TransportInfo* GetTransportInfoByName(const std::string& name);
|
|
const TransportDescription* GetTransportDescriptionByName(
|
|
const std::string& name) const {
|
|
const TransportInfo* tinfo = GetTransportInfoByName(name);
|
|
return tinfo ? &tinfo->description : NULL;
|
|
}
|
|
|
|
// Transport mutators.
|
|
void set_transport_infos(const TransportInfos& transport_infos) {
|
|
transport_infos_ = transport_infos;
|
|
}
|
|
// Adds a TransportInfo to this description.
|
|
void AddTransportInfo(const TransportInfo& transport_info);
|
|
bool RemoveTransportInfoByName(const std::string& name);
|
|
|
|
// Group accessors.
|
|
const ContentGroups& groups() const { return content_groups_; }
|
|
const ContentGroup* GetGroupByName(const std::string& name) const;
|
|
bool HasGroup(const std::string& name) const;
|
|
|
|
// Group mutators.
|
|
void AddGroup(const ContentGroup& group) { content_groups_.push_back(group); }
|
|
// Remove the first group with the same semantics specified by |name|.
|
|
void RemoveGroupByName(const std::string& name);
|
|
|
|
// Global attributes.
|
|
void set_msid_supported(bool supported) { msid_supported_ = supported; }
|
|
bool msid_supported() const { return msid_supported_; }
|
|
|
|
// Determines how the MSIDs were/will be signaled. Flag value composed of
|
|
// MsidSignaling bits (see enum above).
|
|
void set_msid_signaling(int msid_signaling) {
|
|
msid_signaling_ = msid_signaling;
|
|
}
|
|
int msid_signaling() const { return msid_signaling_; }
|
|
|
|
// Determines if it's allowed to mix one- and two-byte rtp header extensions
|
|
// within the same rtp stream.
|
|
void set_extmap_allow_mixed(bool supported) {
|
|
extmap_allow_mixed_ = supported;
|
|
MediaContentDescription::ExtmapAllowMixed media_level_setting =
|
|
supported ? MediaContentDescription::kSession
|
|
: MediaContentDescription::kNo;
|
|
for (auto& content : contents_) {
|
|
// Do not set to kNo if the current setting is kMedia.
|
|
if (supported || content.media_description()->extmap_allow_mixed_enum() !=
|
|
MediaContentDescription::kMedia) {
|
|
content.media_description()->set_extmap_allow_mixed_enum(
|
|
media_level_setting);
|
|
}
|
|
}
|
|
}
|
|
bool extmap_allow_mixed() const { return extmap_allow_mixed_; }
|
|
|
|
private:
|
|
SessionDescription(const SessionDescription&);
|
|
|
|
ContentInfos contents_;
|
|
TransportInfos transport_infos_;
|
|
ContentGroups content_groups_;
|
|
bool msid_supported_ = true;
|
|
// Default to what Plan B would do.
|
|
// TODO(bugs.webrtc.org/8530): Change default to kMsidSignalingMediaSection.
|
|
int msid_signaling_ = kMsidSignalingSsrcAttribute;
|
|
// TODO(webrtc:9985): Activate mixed one- and two-byte header extension in
|
|
// offer at session level. It's currently not included in offer by default
|
|
// because clients prior to https://bugs.webrtc.org/9712 cannot parse this
|
|
// correctly. If it's included in offer to us we will respond that we support
|
|
// it.
|
|
bool extmap_allow_mixed_ = false;
|
|
};
|
|
|
|
// Indicates whether a session description was sent by the local client or
|
|
// received from the remote client.
|
|
enum ContentSource { CS_LOCAL, CS_REMOTE };
|
|
|
|
} // namespace cricket
|
|
|
|
#endif // PC_SESSION_DESCRIPTION_H_
|