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/*
* Copyright 2004 The WebRTC Project Authors. All rights reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef PC_SESSION_DESCRIPTION_H_
#define PC_SESSION_DESCRIPTION_H_
#include <stddef.h>
#include <stdint.h>
#include <algorithm>
#include <memory>
#include <string>
#include <type_traits>
#include <utility>
#include <vector>
#include "absl/memory/memory.h"
#include "absl/strings/string_view.h"
#include "api/media_types.h"
#include "api/rtp_parameters.h"
#include "api/rtp_transceiver_direction.h"
#include "api/rtp_transceiver_interface.h"
#include "media/base/codec.h"
#include "media/base/media_channel.h"
#include "media/base/media_constants.h"
#include "media/base/rid_description.h"
#include "media/base/stream_params.h"
#include "p2p/base/transport_description.h"
#include "p2p/base/transport_info.h"
#include "pc/media_protocol_names.h"
#include "pc/simulcast_description.h"
#include "rtc_base/checks.h"
#include "rtc_base/socket_address.h"
#include "rtc_base/system/rtc_export.h"
namespace cricket {
using RtpHeaderExtensions = std::vector<webrtc::RtpExtension>;
// Options to control how session descriptions are generated.
const int kAutoBandwidth = -1;
class AudioContentDescription;
class VideoContentDescription;
class SctpDataContentDescription;
class UnsupportedContentDescription;
// Describes a session description media section. There are subclasses for each
// media type (audio, video, data) that will have additional information.
class MediaContentDescription {
public:
MediaContentDescription() = default;
virtual ~MediaContentDescription() = default;
virtual MediaType type() const = 0;
// Try to cast this media description to an AudioContentDescription. Returns
// nullptr if the cast fails.
virtual AudioContentDescription* as_audio() { return nullptr; }
virtual const AudioContentDescription* as_audio() const { return nullptr; }
// Try to cast this media description to a VideoContentDescription. Returns
// nullptr if the cast fails.
virtual VideoContentDescription* as_video() { return nullptr; }
virtual const VideoContentDescription* as_video() const { return nullptr; }
virtual SctpDataContentDescription* as_sctp() { return nullptr; }
virtual const SctpDataContentDescription* as_sctp() const { return nullptr; }
virtual UnsupportedContentDescription* as_unsupported() { return nullptr; }
virtual const UnsupportedContentDescription* as_unsupported() const {
return nullptr;
}
// Copy operator that returns an unique_ptr.
// Not a virtual function.
// If a type-specific variant of Clone() is desired, override it, or
// simply use std::make_unique<typename>(*this) instead of Clone().
std::unique_ptr<MediaContentDescription> Clone() const {
return absl::WrapUnique(CloneInternal());
}
// `protocol` is the expected media transport protocol, such as RTP/AVPF,
// RTP/SAVPF or SCTP/DTLS.
std::string protocol() const { return protocol_; }
virtual void set_protocol(absl::string_view protocol) {
protocol_ = std::string(protocol);
}
webrtc::RtpTransceiverDirection direction() const { return direction_; }
void set_direction(webrtc::RtpTransceiverDirection direction) {
direction_ = direction;
}
bool rtcp_mux() const { return rtcp_mux_; }
void set_rtcp_mux(bool mux) { rtcp_mux_ = mux; }
bool rtcp_reduced_size() const { return rtcp_reduced_size_; }
void set_rtcp_reduced_size(bool reduced_size) {
rtcp_reduced_size_ = reduced_size;
}
// Indicates support for the remote network estimate packet type. This
// functionality is experimental and subject to change without notice.
bool remote_estimate() const { return remote_estimate_; }
void set_remote_estimate(bool remote_estimate) {
remote_estimate_ = remote_estimate;
}
int bandwidth() const { return bandwidth_; }
void set_bandwidth(int bandwidth) { bandwidth_ = bandwidth; }
std::string bandwidth_type() const { return bandwidth_type_; }
void set_bandwidth_type(std::string bandwidth_type) {
bandwidth_type_ = bandwidth_type;
}
// List of RTP header extensions. URIs are **NOT** guaranteed to be unique
// as they can appear twice when both encrypted and non-encrypted extensions
// are present.
// Use RtpExtension::FindHeaderExtensionByUri for finding and
// RtpExtension::DeduplicateHeaderExtensions for filtering.
const RtpHeaderExtensions& rtp_header_extensions() const {
return rtp_header_extensions_;
}
void set_rtp_header_extensions(const RtpHeaderExtensions& extensions) {
rtp_header_extensions_ = extensions;
rtp_header_extensions_set_ = true;
}
void AddRtpHeaderExtension(const webrtc::RtpExtension& ext) {
rtp_header_extensions_.push_back(ext);
rtp_header_extensions_set_ = true;
}
void ClearRtpHeaderExtensions() {
rtp_header_extensions_.clear();
rtp_header_extensions_set_ = true;
}
// We can't always tell if an empty list of header extensions is
// because the other side doesn't support them, or just isn't hooked up to
// signal them. For now we assume an empty list means no signaling, but
// provide the ClearRtpHeaderExtensions method to allow "no support" to be
// clearly indicated (i.e. when derived from other information).
bool rtp_header_extensions_set() const { return rtp_header_extensions_set_; }
const StreamParamsVec& streams() const { return send_streams_; }
// TODO(pthatcher): Remove this by giving mediamessage.cc access
// to MediaContentDescription
StreamParamsVec& mutable_streams() { return send_streams_; }
void AddStream(const StreamParams& stream) {
send_streams_.push_back(stream);
}
// Legacy streams have an ssrc, but nothing else.
void AddLegacyStream(uint32_t ssrc) {
AddStream(StreamParams::CreateLegacy(ssrc));
}
void AddLegacyStream(uint32_t ssrc, uint32_t fid_ssrc) {
StreamParams sp = StreamParams::CreateLegacy(ssrc);
sp.AddFidSsrc(ssrc, fid_ssrc);
AddStream(sp);
}
uint32_t first_ssrc() const {
if (send_streams_.empty()) {
return 0;
}
return send_streams_[0].first_ssrc();
}
bool has_ssrcs() const {
if (send_streams_.empty()) {
return false;
}
return send_streams_[0].has_ssrcs();
}
void set_conference_mode(bool enable) { conference_mode_ = enable; }
bool conference_mode() const { return conference_mode_; }
// May be present at the media or session level of SDP. If present at both
// levels, the media-level attribute overwrites the session-level one.
void set_connection_address(const rtc::SocketAddress& address) {
connection_address_ = address;
}
const rtc::SocketAddress& connection_address() const {
return connection_address_;
}
// Determines if it's allowed to mix one- and two-byte rtp header extensions
// within the same rtp stream.
enum ExtmapAllowMixed { kNo, kSession, kMedia };
void set_extmap_allow_mixed_enum(ExtmapAllowMixed new_extmap_allow_mixed) {
if (new_extmap_allow_mixed == kMedia &&
extmap_allow_mixed_enum_ == kSession) {
// Do not downgrade from session level to media level.
return;
}
extmap_allow_mixed_enum_ = new_extmap_allow_mixed;
}
ExtmapAllowMixed extmap_allow_mixed_enum() const {
return extmap_allow_mixed_enum_;
}
bool extmap_allow_mixed() const { return extmap_allow_mixed_enum_ != kNo; }
// Simulcast functionality.
bool HasSimulcast() const { return !simulcast_.empty(); }
SimulcastDescription& simulcast_description() { return simulcast_; }
const SimulcastDescription& simulcast_description() const {
return simulcast_;
}
void set_simulcast_description(const SimulcastDescription& simulcast) {
simulcast_ = simulcast;
}
const std::vector<RidDescription>& receive_rids() const {
return receive_rids_;
}
void set_receive_rids(const std::vector<RidDescription>& rids) {
receive_rids_ = rids;
}
// Codecs should be in preference order (most preferred codec first).
const std::vector<Codec>& codecs() const { return codecs_; }
void set_codecs(const std::vector<Codec>& codecs) { codecs_ = codecs; }
virtual bool has_codecs() const { return !codecs_.empty(); }
bool HasCodec(int id) {
return absl::c_find_if(codecs_, [id](const cricket::Codec codec) {
return codec.id == id;
}) != codecs_.end();
}
void AddCodec(const Codec& codec) { codecs_.push_back(codec); }
void AddOrReplaceCodec(const Codec& codec) {
for (auto it = codecs_.begin(); it != codecs_.end(); ++it) {
if (it->id == codec.id) {
*it = codec;
return;
}
}
AddCodec(codec);
}
void AddCodecs(const std::vector<Codec>& codecs) {
for (const auto& codec : codecs) {
AddCodec(codec);
}
}
protected:
// TODO(bugs.webrtc.org/15214): move all RTP related things to
// RtpMediaDescription that the SCTP content description does
// not inherit from.
std::string protocol_;
private:
bool rtcp_mux_ = false;
bool rtcp_reduced_size_ = false;
bool remote_estimate_ = false;
int bandwidth_ = kAutoBandwidth;
std::string bandwidth_type_ = kApplicationSpecificBandwidth;
std::vector<webrtc::RtpExtension> rtp_header_extensions_;
bool rtp_header_extensions_set_ = false;
StreamParamsVec send_streams_;
bool conference_mode_ = false;
webrtc::RtpTransceiverDirection direction_ =
webrtc::RtpTransceiverDirection::kSendRecv;
rtc::SocketAddress connection_address_;
ExtmapAllowMixed extmap_allow_mixed_enum_ = kMedia;
SimulcastDescription simulcast_;
std::vector<RidDescription> receive_rids_;
// Copy function that returns a raw pointer. Caller will assert ownership.
// Should only be called by the Clone() function. Must be implemented
// by each final subclass.
virtual MediaContentDescription* CloneInternal() const = 0;
std::vector<Codec> codecs_;
};
class RtpMediaContentDescription : public MediaContentDescription {};
class AudioContentDescription : public RtpMediaContentDescription {
public:
void set_protocol(absl::string_view protocol) override {
RTC_DCHECK(IsRtpProtocol(protocol));
protocol_ = std::string(protocol);
}
MediaType type() const override { return MEDIA_TYPE_AUDIO; }
AudioContentDescription* as_audio() override { return this; }
const AudioContentDescription* as_audio() const override { return this; }
private:
AudioContentDescription* CloneInternal() const override {
return new AudioContentDescription(*this);
}
};
class VideoContentDescription : public RtpMediaContentDescription {
public:
void set_protocol(absl::string_view protocol) override {
RTC_DCHECK(IsRtpProtocol(protocol));
protocol_ = std::string(protocol);
}
MediaType type() const override { return MEDIA_TYPE_VIDEO; }
VideoContentDescription* as_video() override { return this; }
const VideoContentDescription* as_video() const override { return this; }
private:
VideoContentDescription* CloneInternal() const override {
return new VideoContentDescription(*this);
}
};
class SctpDataContentDescription : public MediaContentDescription {
public:
SctpDataContentDescription() {}
SctpDataContentDescription(const SctpDataContentDescription& o)
: MediaContentDescription(o),
use_sctpmap_(o.use_sctpmap_),
port_(o.port_),
max_message_size_(o.max_message_size_) {}
MediaType type() const override { return MEDIA_TYPE_DATA; }
SctpDataContentDescription* as_sctp() override { return this; }
const SctpDataContentDescription* as_sctp() const override { return this; }
bool has_codecs() const override { return false; }
void set_protocol(absl::string_view protocol) override {
RTC_DCHECK(IsSctpProtocol(protocol));
protocol_ = std::string(protocol);
}
bool use_sctpmap() const { return use_sctpmap_; }
void set_use_sctpmap(bool enable) { use_sctpmap_ = enable; }
int port() const { return port_; }
void set_port(int port) { port_ = port; }
int max_message_size() const { return max_message_size_; }
void set_max_message_size(int max_message_size) {
max_message_size_ = max_message_size;
}
private:
SctpDataContentDescription* CloneInternal() const override {
return new SctpDataContentDescription(*this);
}
bool use_sctpmap_ = true; // Note: "true" is no longer conformant.
// Defaults should be constants imported from SCTP. Quick hack.
int port_ = 5000;
// draft-ietf-mmusic-sdp-sctp-23: Max message size default is 64K
int max_message_size_ = 64 * 1024;
};
class UnsupportedContentDescription : public MediaContentDescription {
public:
explicit UnsupportedContentDescription(absl::string_view media_type)
: media_type_(media_type) {}
MediaType type() const override { return MEDIA_TYPE_UNSUPPORTED; }
UnsupportedContentDescription* as_unsupported() override { return this; }
const UnsupportedContentDescription* as_unsupported() const override {
return this;
}
bool has_codecs() const override { return false; }
const std::string& media_type() const { return media_type_; }
private:
UnsupportedContentDescription* CloneInternal() const override {
return new UnsupportedContentDescription(*this);
}
std::string media_type_;
};
// Protocol used for encoding media. This is the "top level" protocol that may
// be wrapped by zero or many transport protocols (UDP, ICE, etc.).
enum class MediaProtocolType {
kRtp, // Section will use the RTP protocol (e.g., for audio or video).
kSctp, // Section will use the SCTP protocol (e.g., for a data channel).
kOther // Section will use another top protocol which is not
// explicitly supported.
};
// Represents a session description section. Most information about the section
// is stored in the description, which is a subclass of MediaContentDescription.
// Owns the description.
class RTC_EXPORT ContentInfo {
public:
explicit ContentInfo(MediaProtocolType type) : type(type) {}
~ContentInfo();
// Copy
ContentInfo(const ContentInfo& o);
ContentInfo& operator=(const ContentInfo& o);
ContentInfo(ContentInfo&& o) = default;
ContentInfo& operator=(ContentInfo&& o) = default;
// Alias for `name`.
std::string mid() const { return name; }
void set_mid(const std::string& mid) { this->name = mid; }
// Alias for `description`.
MediaContentDescription* media_description();
const MediaContentDescription* media_description() const;
void set_media_description(std::unique_ptr<MediaContentDescription> desc) {
description_ = std::move(desc);
}
// TODO(bugs.webrtc.org/8620): Rename this to mid.
std::string name;
MediaProtocolType type;
bool rejected = false;
bool bundle_only = false;
private:
friend class SessionDescription;
std::unique_ptr<MediaContentDescription> description_;
};
typedef std::vector<std::string> ContentNames;
// This class provides a mechanism to aggregate different media contents into a
// group. This group can also be shared with the peers in a pre-defined format.
// GroupInfo should be populated only with the `content_name` of the
// MediaDescription.
class ContentGroup {
public:
explicit ContentGroup(const std::string& semantics);
ContentGroup(const ContentGroup&);
ContentGroup(ContentGroup&&);
ContentGroup& operator=(const ContentGroup&);
ContentGroup& operator=(ContentGroup&&);
~ContentGroup();
const std::string& semantics() const { return semantics_; }
const ContentNames& content_names() const { return content_names_; }
const std::string* FirstContentName() const;
bool HasContentName(absl::string_view content_name) const;
void AddContentName(absl::string_view content_name);
bool RemoveContentName(absl::string_view content_name);
// for debugging
std::string ToString() const;
private:
std::string semantics_;
ContentNames content_names_;
};
typedef std::vector<ContentInfo> ContentInfos;
typedef std::vector<ContentGroup> ContentGroups;
const ContentInfo* FindContentInfoByName(const ContentInfos& contents,
const std::string& name);
const ContentInfo* FindContentInfoByType(const ContentInfos& contents,
const std::string& type);
// Determines how the MSID will be signaled in the SDP.
// These can be used as bit flags to indicate both or the special value none.
enum MsidSignaling {
// MSID is not signaled. This is not a bit flag and must be compared for
// equality.
kMsidSignalingNotUsed = 0x0,
// Signal MSID with at least one a=msid line in the media section.
// This requires unified plan.
kMsidSignalingMediaSection = 0x1,
// Signal MSID with a=ssrc: msid lines in the media section.
// This should only be used with plan-b but is signalled in
// offers for backward compability reasons.
kMsidSignalingSsrcAttribute = 0x2,
// Signal MSID with a=msid-semantic: WMS in the session section.
// This is deprecated but signalled for backward compability reasons.
// It is typically combined with 0x1 or 0x2.
kMsidSignalingSemantic = 0x4
};
// 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;
std::vector<const ContentGroup*> GetGroupsByName(
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.
// 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_;
int msid_signaling_ = kMsidSignalingMediaSection | kMsidSignalingSemantic;
bool extmap_allow_mixed_ = true;
};
// 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_