| /* |
| * 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 <iosfwd> |
| #include <memory> |
| #include <string> |
| #include <utility> |
| #include <vector> |
| |
| #include "absl/memory/memory.h" |
| #include "api/crypto_params.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 { |
| |
| typedef std::vector<AudioCodec> AudioCodecs; |
| typedef std::vector<VideoCodec> VideoCodecs; |
| typedef std::vector<CryptoParams> CryptoParamsVec; |
| typedef std::vector<webrtc::RtpExtension> RtpHeaderExtensions; |
| |
| // RTC4585 RTP/AVPF |
| extern const char kMediaProtocolAvpf[]; |
| // RFC5124 RTP/SAVPF |
| extern const char kMediaProtocolSavpf[]; |
| |
| extern const char kMediaProtocolDtlsSavpf[]; |
| |
| // 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; |
| } |
| |
| virtual bool has_codecs() const = 0; |
| |
| // 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. |
| virtual std::string protocol() const { return protocol_; } |
| virtual void set_protocol(const std::string& protocol) { |
| protocol_ = protocol; |
| } |
| |
| virtual webrtc::RtpTransceiverDirection direction() const { |
| return direction_; |
| } |
| virtual void set_direction(webrtc::RtpTransceiverDirection direction) { |
| direction_ = direction; |
| } |
| |
| virtual bool rtcp_mux() const { return rtcp_mux_; } |
| virtual void set_rtcp_mux(bool mux) { rtcp_mux_ = mux; } |
| |
| virtual bool rtcp_reduced_size() const { return rtcp_reduced_size_; } |
| virtual 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. |
| virtual bool remote_estimate() const { return remote_estimate_; } |
| virtual void set_remote_estimate(bool remote_estimate) { |
| remote_estimate_ = remote_estimate; |
| } |
| |
| virtual int bandwidth() const { return bandwidth_; } |
| virtual void set_bandwidth(int bandwidth) { bandwidth_ = bandwidth; } |
| virtual std::string bandwidth_type() const { return bandwidth_type_; } |
| virtual void set_bandwidth_type(std::string bandwidth_type) { |
| bandwidth_type_ = bandwidth_type; |
| } |
| |
| virtual const std::vector<CryptoParams>& cryptos() const { return cryptos_; } |
| virtual void AddCrypto(const CryptoParams& params) { |
| cryptos_.push_back(params); |
| } |
| virtual void set_cryptos(const std::vector<CryptoParams>& cryptos) { |
| cryptos_ = cryptos; |
| } |
| |
| // 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. |
| virtual const RtpHeaderExtensions& rtp_header_extensions() const { |
| return rtp_header_extensions_; |
| } |
| virtual void set_rtp_header_extensions( |
| const RtpHeaderExtensions& extensions) { |
| rtp_header_extensions_ = extensions; |
| rtp_header_extensions_set_ = true; |
| } |
| virtual void AddRtpHeaderExtension(const webrtc::RtpExtension& ext) { |
| rtp_header_extensions_.push_back(ext); |
| rtp_header_extensions_set_ = true; |
| } |
| virtual 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). |
| virtual bool rtp_header_extensions_set() const { |
| return rtp_header_extensions_set_; |
| } |
| virtual const StreamParamsVec& streams() const { return send_streams_; } |
| // TODO(pthatcher): Remove this by giving mediamessage.cc access |
| // to MediaContentDescription |
| virtual StreamParamsVec& mutable_streams() { return send_streams_; } |
| virtual 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); |
| } |
| |
| // Sets the CNAME of all StreamParams if it have not been set. |
| virtual void SetCnameIfEmpty(const std::string& cname) { |
| for (cricket::StreamParamsVec::iterator it = send_streams_.begin(); |
| it != send_streams_.end(); ++it) { |
| if (it->cname.empty()) |
| it->cname = cname; |
| } |
| } |
| virtual uint32_t first_ssrc() const { |
| if (send_streams_.empty()) { |
| return 0; |
| } |
| return send_streams_[0].first_ssrc(); |
| } |
| virtual bool has_ssrcs() const { |
| if (send_streams_.empty()) { |
| return false; |
| } |
| return send_streams_[0].has_ssrcs(); |
| } |
| |
| virtual void set_conference_mode(bool enable) { conference_mode_ = enable; } |
| virtual bool conference_mode() const { return conference_mode_; } |
| |
| // https://tools.ietf.org/html/rfc4566#section-5.7 |
| // 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. |
| virtual void set_connection_address(const rtc::SocketAddress& address) { |
| connection_address_ = address; |
| } |
| virtual 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 }; |
| virtual 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; |
| } |
| virtual ExtmapAllowMixed extmap_allow_mixed_enum() const { |
| return extmap_allow_mixed_enum_; |
| } |
| virtual bool extmap_allow_mixed() const { |
| return extmap_allow_mixed_enum_ != kNo; |
| } |
| |
| // Simulcast functionality. |
| virtual bool HasSimulcast() const { return !simulcast_.empty(); } |
| virtual SimulcastDescription& simulcast_description() { return simulcast_; } |
| virtual const SimulcastDescription& simulcast_description() const { |
| return simulcast_; |
| } |
| virtual void set_simulcast_description( |
| const SimulcastDescription& simulcast) { |
| simulcast_ = simulcast; |
| } |
| virtual const std::vector<RidDescription>& receive_rids() const { |
| return receive_rids_; |
| } |
| virtual void set_receive_rids(const std::vector<RidDescription>& rids) { |
| receive_rids_ = rids; |
| } |
| |
| protected: |
| bool rtcp_mux_ = false; |
| bool rtcp_reduced_size_ = false; |
| bool remote_estimate_ = false; |
| int bandwidth_ = kAutoBandwidth; |
| std::string bandwidth_type_ = kApplicationSpecificBandwidth; |
| std::string protocol_; |
| std::vector<CryptoParams> cryptos_; |
| 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_; |
| |
| private: |
| // 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; |
| }; |
| |
| template <class C> |
| class MediaContentDescriptionImpl : public MediaContentDescription { |
| public: |
| void set_protocol(const std::string& protocol) override { |
| RTC_DCHECK(IsRtpProtocol(protocol)); |
| protocol_ = protocol; |
| } |
| |
| typedef C CodecType; |
| |
| // Codecs should be in preference order (most preferred codec first). |
| virtual const std::vector<C>& codecs() const { return codecs_; } |
| virtual void set_codecs(const std::vector<C>& codecs) { codecs_ = codecs; } |
| bool has_codecs() const override { return !codecs_.empty(); } |
| virtual bool HasCodec(int id) { |
| bool found = false; |
| for (typename std::vector<C>::iterator iter = codecs_.begin(); |
| iter != codecs_.end(); ++iter) { |
| if (iter->id == id) { |
| found = true; |
| break; |
| } |
| } |
| return found; |
| } |
| virtual void AddCodec(const C& codec) { codecs_.push_back(codec); } |
| virtual void AddOrReplaceCodec(const C& codec) { |
| for (typename std::vector<C>::iterator iter = codecs_.begin(); |
| iter != codecs_.end(); ++iter) { |
| if (iter->id == codec.id) { |
| *iter = codec; |
| return; |
| } |
| } |
| AddCodec(codec); |
| } |
| virtual void AddCodecs(const std::vector<C>& codecs) { |
| typename std::vector<C>::const_iterator codec; |
| for (codec = codecs.begin(); codec != codecs.end(); ++codec) { |
| AddCodec(*codec); |
| } |
| } |
| |
| private: |
| std::vector<C> codecs_; |
| }; |
| |
| class AudioContentDescription : public MediaContentDescriptionImpl<AudioCodec> { |
| public: |
| AudioContentDescription() {} |
| |
| virtual MediaType type() const { return MEDIA_TYPE_AUDIO; } |
| virtual AudioContentDescription* as_audio() { return this; } |
| virtual const AudioContentDescription* as_audio() const { return this; } |
| |
| private: |
| virtual AudioContentDescription* CloneInternal() const { |
| return new AudioContentDescription(*this); |
| } |
| }; |
| |
| class VideoContentDescription : public MediaContentDescriptionImpl<VideoCodec> { |
| public: |
| virtual MediaType type() const { return MEDIA_TYPE_VIDEO; } |
| virtual VideoContentDescription* as_video() { return this; } |
| virtual const VideoContentDescription* as_video() const { return this; } |
| |
| private: |
| virtual VideoContentDescription* CloneInternal() const { |
| 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(const std::string& protocol) override { |
| RTC_DCHECK(IsSctpProtocol(protocol)); |
| protocol_ = 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(const std::string& 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). |
| // https://tools.ietf.org/html/rfc3550 |
| kSctp, // Section will use the SCTP protocol (e.g., for a data channel). |
| // https://tools.ietf.org/html/rfc4960 |
| 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(const std::string& content_name) const; |
| void AddContentName(const std::string& content_name); |
| bool RemoveContentName(const std::string& content_name); |
| |
| 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 |
| // flags to indicate both or none. |
| enum MsidSignaling { |
| // Signal MSID with one a=msid line in the media section. |
| kMsidSignalingMediaSection = 0x1, |
| // Signal MSID with a=ssrc: msid lines in the media section. |
| 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; |
| 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. |
| 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; |
| 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_ |