| /* |
| * 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. |
| */ |
| |
| // Types and classes used in media session descriptions. |
| |
| #ifndef PC_MEDIASESSION_H_ |
| #define PC_MEDIASESSION_H_ |
| |
| #include <algorithm> |
| #include <map> |
| #include <string> |
| #include <vector> |
| |
| #include "api/mediatypes.h" |
| #include "media/base/codec.h" |
| #include "media/base/cryptoparams.h" |
| #include "media/base/mediachannel.h" |
| #include "media/base/mediaconstants.h" |
| #include "media/base/mediaengine.h" // For DataChannelType |
| #include "media/base/streamparams.h" |
| #include "p2p/base/sessiondescription.h" |
| #include "p2p/base/jseptransport.h" |
| #include "p2p/base/transportdescriptionfactory.h" |
| |
| namespace cricket { |
| |
| class ChannelManager; |
| typedef std::vector<AudioCodec> AudioCodecs; |
| typedef std::vector<VideoCodec> VideoCodecs; |
| typedef std::vector<DataCodec> DataCodecs; |
| typedef std::vector<CryptoParams> CryptoParamsVec; |
| typedef std::vector<webrtc::RtpExtension> RtpHeaderExtensions; |
| |
| enum MediaContentDirection { |
| MD_INACTIVE, |
| MD_SENDONLY, |
| MD_RECVONLY, |
| MD_SENDRECV |
| }; |
| |
| std::string MediaContentDirectionToString(MediaContentDirection direction); |
| |
| enum CryptoType { |
| CT_NONE, |
| CT_SDES, |
| CT_DTLS |
| }; |
| |
| // RTC4585 RTP/AVPF |
| extern const char kMediaProtocolAvpf[]; |
| // RFC5124 RTP/SAVPF |
| extern const char kMediaProtocolSavpf[]; |
| |
| extern const char kMediaProtocolDtlsSavpf[]; |
| |
| extern const char kMediaProtocolRtpPrefix[]; |
| |
| extern const char kMediaProtocolSctp[]; |
| extern const char kMediaProtocolDtlsSctp[]; |
| extern const char kMediaProtocolUdpDtlsSctp[]; |
| extern const char kMediaProtocolTcpDtlsSctp[]; |
| |
| // Options to control how session descriptions are generated. |
| const int kAutoBandwidth = -1; |
| const int kBufferedModeDisabled = 0; |
| |
| // Default RTCP CNAME for unit tests. |
| const char kDefaultRtcpCname[] = "DefaultRtcpCname"; |
| |
| struct RtpTransceiverDirection { |
| bool send; |
| bool recv; |
| |
| RtpTransceiverDirection(bool send, bool recv) : send(send), recv(recv) {} |
| |
| bool operator==(const RtpTransceiverDirection& o) const { |
| return send == o.send && recv == o.recv; |
| } |
| |
| bool operator!=(const RtpTransceiverDirection& o) const { |
| return !(*this == o); |
| } |
| |
| static RtpTransceiverDirection FromMediaContentDirection( |
| MediaContentDirection md); |
| |
| MediaContentDirection ToMediaContentDirection() const; |
| |
| RtpTransceiverDirection Reversed() const { |
| return RtpTransceiverDirection(recv, send); |
| } |
| }; |
| |
| RtpTransceiverDirection |
| NegotiateRtpTransceiverDirection(RtpTransceiverDirection offer, |
| RtpTransceiverDirection wants); |
| |
| // Options for an RtpSender contained with an media description/"m=" section. |
| struct SenderOptions { |
| std::string track_id; |
| // TODO(steveanton): As part of work towards Unified Plan, this has been |
| // changed to be a vector. But for now this can only have exactly one. |
| std::vector<std::string> stream_ids; |
| int num_sim_layers; |
| }; |
| |
| // Options for an individual media description/"m=" section. |
| struct MediaDescriptionOptions { |
| MediaDescriptionOptions(MediaType type, |
| const std::string& mid, |
| RtpTransceiverDirection direction, |
| bool stopped) |
| : type(type), mid(mid), direction(direction), stopped(stopped) {} |
| |
| // TODO(deadbeef): When we don't support Plan B, there will only be one |
| // sender per media description and this can be simplified. |
| void AddAudioSender(const std::string& track_id, |
| const std::vector<std::string>& stream_ids); |
| void AddVideoSender(const std::string& track_id, |
| const std::vector<std::string>& stream_ids, |
| int num_sim_layers); |
| |
| // Internally just uses sender_options. |
| void AddRtpDataChannel(const std::string& track_id, |
| const std::string& stream_id); |
| |
| MediaType type; |
| std::string mid; |
| RtpTransceiverDirection direction; |
| bool stopped; |
| TransportOptions transport_options; |
| // Note: There's no equivalent "RtpReceiverOptions" because only send |
| // stream information goes in the local descriptions. |
| std::vector<SenderOptions> sender_options; |
| |
| private: |
| // Doesn't DCHECK on |type|. |
| void AddSenderInternal(const std::string& track_id, |
| const std::vector<std::string>& stream_ids, |
| int num_sim_layers); |
| }; |
| |
| // Provides a mechanism for describing how m= sections should be generated. |
| // The m= section with index X will use media_description_options[X]. There |
| // must be an option for each existing section if creating an answer, or a |
| // subsequent offer. |
| struct MediaSessionOptions { |
| MediaSessionOptions() {} |
| |
| bool has_audio() const { return HasMediaDescription(MEDIA_TYPE_AUDIO); } |
| bool has_video() const { return HasMediaDescription(MEDIA_TYPE_VIDEO); } |
| bool has_data() const { return HasMediaDescription(MEDIA_TYPE_DATA); } |
| |
| bool HasMediaDescription(MediaType type) const; |
| |
| DataChannelType data_channel_type = DCT_NONE; |
| bool is_muc = false; |
| bool vad_enabled = true; // When disabled, removes all CN codecs from SDP. |
| bool rtcp_mux_enabled = true; |
| bool bundle_enabled = false; |
| std::string rtcp_cname = kDefaultRtcpCname; |
| rtc::CryptoOptions crypto_options; |
| // List of media description options in the same order that the media |
| // descriptions will be generated. |
| std::vector<MediaDescriptionOptions> media_description_options; |
| }; |
| |
| // "content" (as used in XEP-0166) descriptions for voice and video. |
| class MediaContentDescription : public ContentDescription { |
| public: |
| MediaContentDescription() {} |
| |
| virtual MediaType type() const = 0; |
| virtual bool has_codecs() const = 0; |
| |
| // |protocol| is the expected media transport protocol, such as RTP/AVPF, |
| // RTP/SAVPF or SCTP/DTLS. |
| std::string protocol() const { return protocol_; } |
| void set_protocol(const std::string& protocol) { protocol_ = protocol; } |
| |
| MediaContentDirection direction() const { return direction_; } |
| void set_direction(MediaContentDirection 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; |
| } |
| |
| int bandwidth() const { return bandwidth_; } |
| void set_bandwidth(int bandwidth) { bandwidth_ = bandwidth; } |
| |
| const std::vector<CryptoParams>& cryptos() const { return cryptos_; } |
| void AddCrypto(const CryptoParams& params) { |
| cryptos_.push_back(params); |
| } |
| void set_cryptos(const std::vector<CryptoParams>& cryptos) { |
| cryptos_ = cryptos; |
| } |
| |
| CryptoType crypto_required() const { return crypto_required_; } |
| void set_crypto_required(CryptoType type) { |
| crypto_required_ = type; |
| } |
| |
| 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 AddRtpHeaderExtension(const cricket::RtpHeaderExtension& ext) { |
| webrtc::RtpExtension webrtc_extension; |
| webrtc_extension.uri = ext.uri; |
| webrtc_extension.id = ext.id; |
| rtp_header_extensions_.push_back(webrtc_extension); |
| 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_; |
| } |
| // True iff the client supports multiple streams. |
| void set_multistream(bool multistream) { multistream_ = multistream; } |
| bool multistream() const { return multistream_; } |
| const StreamParamsVec& streams() const { |
| return streams_; |
| } |
| // TODO(pthatcher): Remove this by giving mediamessage.cc access |
| // to MediaContentDescription |
| StreamParamsVec& mutable_streams() { |
| return streams_; |
| } |
| void AddStream(const StreamParams& stream) { |
| streams_.push_back(stream); |
| } |
| // Legacy streams have an ssrc, but nothing else. |
| void AddLegacyStream(uint32_t ssrc) { |
| streams_.push_back(StreamParams::CreateLegacy(ssrc)); |
| } |
| void AddLegacyStream(uint32_t ssrc, uint32_t fid_ssrc) { |
| StreamParams sp = StreamParams::CreateLegacy(ssrc); |
| sp.AddFidSsrc(ssrc, fid_ssrc); |
| streams_.push_back(sp); |
| } |
| // Sets the CNAME of all StreamParams if it have not been set. |
| void SetCnameIfEmpty(const std::string& cname) { |
| for (cricket::StreamParamsVec::iterator it = streams_.begin(); |
| it != streams_.end(); ++it) { |
| if (it->cname.empty()) |
| it->cname = cname; |
| } |
| } |
| uint32_t first_ssrc() const { |
| if (streams_.empty()) { |
| return 0; |
| } |
| return streams_[0].first_ssrc(); |
| } |
| bool has_ssrcs() const { |
| if (streams_.empty()) { |
| return false; |
| } |
| return streams_[0].has_ssrcs(); |
| } |
| |
| void set_conference_mode(bool enable) { conference_mode_ = enable; } |
| bool conference_mode() const { return conference_mode_; } |
| |
| void set_partial(bool partial) { partial_ = partial; } |
| bool partial() const { return partial_; } |
| |
| void set_buffered_mode_latency(int latency) { |
| buffered_mode_latency_ = latency; |
| } |
| int buffered_mode_latency() const { return buffered_mode_latency_; } |
| |
| // 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. |
| void set_connection_address(const rtc::SocketAddress& address) { |
| connection_address_ = address; |
| } |
| const rtc::SocketAddress& connection_address() const { |
| return connection_address_; |
| } |
| |
| protected: |
| bool rtcp_mux_ = false; |
| bool rtcp_reduced_size_ = false; |
| int bandwidth_ = kAutoBandwidth; |
| std::string protocol_; |
| std::vector<CryptoParams> cryptos_; |
| CryptoType crypto_required_ = CT_NONE; |
| std::vector<webrtc::RtpExtension> rtp_header_extensions_; |
| bool rtp_header_extensions_set_ = false; |
| bool multistream_ = false; |
| StreamParamsVec streams_; |
| bool conference_mode_ = false; |
| bool partial_ = false; |
| int buffered_mode_latency_ = kBufferedModeDisabled; |
| MediaContentDirection direction_ = MD_SENDRECV; |
| rtc::SocketAddress connection_address_; |
| }; |
| |
| template <class C> |
| class MediaContentDescriptionImpl : public MediaContentDescription { |
| public: |
| typedef C CodecType; |
| |
| // Codecs should be in preference order (most preferred codec first). |
| const std::vector<C>& codecs() const { return codecs_; } |
| void set_codecs(const std::vector<C>& codecs) { codecs_ = codecs; } |
| virtual bool has_codecs() const { return !codecs_.empty(); } |
| 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; |
| } |
| void AddCodec(const C& codec) { |
| codecs_.push_back(codec); |
| } |
| 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); |
| } |
| 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() : |
| agc_minus_10db_(false) {} |
| |
| virtual ContentDescription* Copy() const { |
| return new AudioContentDescription(*this); |
| } |
| virtual MediaType type() const { return MEDIA_TYPE_AUDIO; } |
| |
| const std::string &lang() const { return lang_; } |
| void set_lang(const std::string &lang) { lang_ = lang; } |
| |
| bool agc_minus_10db() const { return agc_minus_10db_; } |
| void set_agc_minus_10db(bool enable) { |
| agc_minus_10db_ = enable; |
| } |
| |
| private: |
| bool agc_minus_10db_; |
| |
| private: |
| std::string lang_; |
| }; |
| |
| class VideoContentDescription : public MediaContentDescriptionImpl<VideoCodec> { |
| public: |
| virtual ContentDescription* Copy() const { |
| return new VideoContentDescription(*this); |
| } |
| virtual MediaType type() const { return MEDIA_TYPE_VIDEO; } |
| }; |
| |
| class DataContentDescription : public MediaContentDescriptionImpl<DataCodec> { |
| public: |
| DataContentDescription() {} |
| |
| virtual ContentDescription* Copy() const { |
| return new DataContentDescription(*this); |
| } |
| virtual MediaType type() const { return MEDIA_TYPE_DATA; } |
| |
| bool use_sctpmap() const { return use_sctpmap_; } |
| void set_use_sctpmap(bool enable) { use_sctpmap_ = enable; } |
| |
| private: |
| bool use_sctpmap_ = true; |
| }; |
| |
| // Creates media session descriptions according to the supplied codecs and |
| // other fields, as well as the supplied per-call options. |
| // When creating answers, performs the appropriate negotiation |
| // of the various fields to determine the proper result. |
| class MediaSessionDescriptionFactory { |
| public: |
| // Default ctor; use methods below to set configuration. |
| // The TransportDescriptionFactory is not owned by MediaSessionDescFactory, |
| // so it must be kept alive by the user of this class. |
| explicit MediaSessionDescriptionFactory( |
| const TransportDescriptionFactory* factory); |
| // This helper automatically sets up the factory to get its configuration |
| // from the specified ChannelManager. |
| MediaSessionDescriptionFactory(ChannelManager* cmanager, |
| const TransportDescriptionFactory* factory); |
| |
| const AudioCodecs& audio_sendrecv_codecs() const; |
| const AudioCodecs& audio_send_codecs() const; |
| const AudioCodecs& audio_recv_codecs() const; |
| void set_audio_codecs(const AudioCodecs& send_codecs, |
| const AudioCodecs& recv_codecs); |
| void set_audio_rtp_header_extensions(const RtpHeaderExtensions& extensions) { |
| audio_rtp_extensions_ = extensions; |
| } |
| const RtpHeaderExtensions& audio_rtp_header_extensions() const { |
| return audio_rtp_extensions_; |
| } |
| const VideoCodecs& video_codecs() const { return video_codecs_; } |
| void set_video_codecs(const VideoCodecs& codecs) { video_codecs_ = codecs; } |
| void set_video_rtp_header_extensions(const RtpHeaderExtensions& extensions) { |
| video_rtp_extensions_ = extensions; |
| } |
| const RtpHeaderExtensions& video_rtp_header_extensions() const { |
| return video_rtp_extensions_; |
| } |
| const DataCodecs& data_codecs() const { return data_codecs_; } |
| void set_data_codecs(const DataCodecs& codecs) { data_codecs_ = codecs; } |
| SecurePolicy secure() const { return secure_; } |
| void set_secure(SecurePolicy s) { secure_ = s; } |
| |
| void set_enable_encrypted_rtp_header_extensions(bool enable) { |
| enable_encrypted_rtp_header_extensions_ = enable; |
| } |
| |
| SessionDescription* CreateOffer( |
| const MediaSessionOptions& options, |
| const SessionDescription* current_description) const; |
| SessionDescription* CreateAnswer( |
| const SessionDescription* offer, |
| const MediaSessionOptions& options, |
| const SessionDescription* current_description) const; |
| |
| private: |
| const AudioCodecs& GetAudioCodecsForOffer( |
| const RtpTransceiverDirection& direction) const; |
| const AudioCodecs& GetAudioCodecsForAnswer( |
| const RtpTransceiverDirection& offer, |
| const RtpTransceiverDirection& answer) const; |
| void GetCodecsForOffer(const SessionDescription* current_description, |
| AudioCodecs* audio_codecs, |
| VideoCodecs* video_codecs, |
| DataCodecs* data_codecs) const; |
| void GetCodecsForAnswer(const SessionDescription* current_description, |
| const SessionDescription* remote_offer, |
| AudioCodecs* audio_codecs, |
| VideoCodecs* video_codecs, |
| DataCodecs* data_codecs) const; |
| void GetRtpHdrExtsToOffer(const SessionDescription* current_description, |
| RtpHeaderExtensions* audio_extensions, |
| RtpHeaderExtensions* video_extensions) const; |
| bool AddTransportOffer( |
| const std::string& content_name, |
| const TransportOptions& transport_options, |
| const SessionDescription* current_desc, |
| SessionDescription* offer) const; |
| |
| TransportDescription* CreateTransportAnswer( |
| const std::string& content_name, |
| const SessionDescription* offer_desc, |
| const TransportOptions& transport_options, |
| const SessionDescription* current_desc, |
| bool require_transport_attributes) const; |
| |
| bool AddTransportAnswer( |
| const std::string& content_name, |
| const TransportDescription& transport_desc, |
| SessionDescription* answer_desc) const; |
| |
| // Helpers for adding media contents to the SessionDescription. Returns true |
| // it succeeds or the media content is not needed, or false if there is any |
| // error. |
| |
| bool AddAudioContentForOffer( |
| const MediaDescriptionOptions& media_description_options, |
| const MediaSessionOptions& session_options, |
| const ContentInfo* current_content, |
| const SessionDescription* current_description, |
| const RtpHeaderExtensions& audio_rtp_extensions, |
| const AudioCodecs& audio_codecs, |
| StreamParamsVec* current_streams, |
| SessionDescription* desc) const; |
| |
| bool AddVideoContentForOffer( |
| const MediaDescriptionOptions& media_description_options, |
| const MediaSessionOptions& session_options, |
| const ContentInfo* current_content, |
| const SessionDescription* current_description, |
| const RtpHeaderExtensions& video_rtp_extensions, |
| const VideoCodecs& video_codecs, |
| StreamParamsVec* current_streams, |
| SessionDescription* desc) const; |
| |
| bool AddDataContentForOffer( |
| const MediaDescriptionOptions& media_description_options, |
| const MediaSessionOptions& session_options, |
| const ContentInfo* current_content, |
| const SessionDescription* current_description, |
| const DataCodecs& data_codecs, |
| StreamParamsVec* current_streams, |
| SessionDescription* desc) const; |
| |
| bool AddAudioContentForAnswer( |
| const MediaDescriptionOptions& media_description_options, |
| const MediaSessionOptions& session_options, |
| const ContentInfo* offer_content, |
| const SessionDescription* offer_description, |
| const ContentInfo* current_content, |
| const SessionDescription* current_description, |
| const TransportInfo* bundle_transport, |
| const AudioCodecs& audio_codecs, |
| StreamParamsVec* current_streams, |
| SessionDescription* answer) const; |
| |
| bool AddVideoContentForAnswer( |
| const MediaDescriptionOptions& media_description_options, |
| const MediaSessionOptions& session_options, |
| const ContentInfo* offer_content, |
| const SessionDescription* offer_description, |
| const ContentInfo* current_content, |
| const SessionDescription* current_description, |
| const TransportInfo* bundle_transport, |
| const VideoCodecs& video_codecs, |
| StreamParamsVec* current_streams, |
| SessionDescription* answer) const; |
| |
| bool AddDataContentForAnswer( |
| const MediaDescriptionOptions& media_description_options, |
| const MediaSessionOptions& session_options, |
| const ContentInfo* offer_content, |
| const SessionDescription* offer_description, |
| const ContentInfo* current_content, |
| const SessionDescription* current_description, |
| const TransportInfo* bundle_transport, |
| const DataCodecs& data_codecs, |
| StreamParamsVec* current_streams, |
| SessionDescription* answer) const; |
| |
| void ComputeAudioCodecsIntersectionAndUnion(); |
| |
| AudioCodecs audio_send_codecs_; |
| AudioCodecs audio_recv_codecs_; |
| // Intersection of send and recv. |
| AudioCodecs audio_sendrecv_codecs_; |
| // Union of send and recv. |
| AudioCodecs all_audio_codecs_; |
| RtpHeaderExtensions audio_rtp_extensions_; |
| VideoCodecs video_codecs_; |
| RtpHeaderExtensions video_rtp_extensions_; |
| DataCodecs data_codecs_; |
| bool enable_encrypted_rtp_header_extensions_ = false; |
| // TODO(zhihuang): Rename secure_ to sdec_policy_; rename the related getter |
| // and setter. |
| SecurePolicy secure_ = SEC_DISABLED; |
| std::string lang_; |
| const TransportDescriptionFactory* transport_desc_factory_; |
| }; |
| |
| // Convenience functions. |
| bool IsMediaContent(const ContentInfo* content); |
| bool IsAudioContent(const ContentInfo* content); |
| bool IsVideoContent(const ContentInfo* content); |
| bool IsDataContent(const ContentInfo* content); |
| const ContentInfo* GetFirstMediaContent(const ContentInfos& contents, |
| MediaType media_type); |
| const ContentInfo* GetFirstAudioContent(const ContentInfos& contents); |
| const ContentInfo* GetFirstVideoContent(const ContentInfos& contents); |
| const ContentInfo* GetFirstDataContent(const ContentInfos& contents); |
| const ContentInfo* GetFirstAudioContent(const SessionDescription* sdesc); |
| const ContentInfo* GetFirstVideoContent(const SessionDescription* sdesc); |
| const ContentInfo* GetFirstDataContent(const SessionDescription* sdesc); |
| const AudioContentDescription* GetFirstAudioContentDescription( |
| const SessionDescription* sdesc); |
| const VideoContentDescription* GetFirstVideoContentDescription( |
| const SessionDescription* sdesc); |
| const DataContentDescription* GetFirstDataContentDescription( |
| const SessionDescription* sdesc); |
| // Non-const versions of the above functions. |
| // Useful when modifying an existing description. |
| ContentInfo* GetFirstMediaContent(ContentInfos& contents, MediaType media_type); |
| ContentInfo* GetFirstAudioContent(ContentInfos& contents); |
| ContentInfo* GetFirstVideoContent(ContentInfos& contents); |
| ContentInfo* GetFirstDataContent(ContentInfos& contents); |
| ContentInfo* GetFirstAudioContent(SessionDescription* sdesc); |
| ContentInfo* GetFirstVideoContent(SessionDescription* sdesc); |
| ContentInfo* GetFirstDataContent(SessionDescription* sdesc); |
| AudioContentDescription* GetFirstAudioContentDescription( |
| SessionDescription* sdesc); |
| VideoContentDescription* GetFirstVideoContentDescription( |
| SessionDescription* sdesc); |
| DataContentDescription* GetFirstDataContentDescription( |
| SessionDescription* sdesc); |
| |
| // Helper functions to return crypto suites used for SDES. |
| void GetSupportedAudioSdesCryptoSuites(const rtc::CryptoOptions& crypto_options, |
| std::vector<int>* crypto_suites); |
| void GetSupportedVideoSdesCryptoSuites(const rtc::CryptoOptions& crypto_options, |
| std::vector<int>* crypto_suites); |
| void GetSupportedDataSdesCryptoSuites(const rtc::CryptoOptions& crypto_options, |
| std::vector<int>* crypto_suites); |
| void GetSupportedAudioSdesCryptoSuiteNames( |
| const rtc::CryptoOptions& crypto_options, |
| std::vector<std::string>* crypto_suite_names); |
| void GetSupportedVideoSdesCryptoSuiteNames( |
| const rtc::CryptoOptions& crypto_options, |
| std::vector<std::string>* crypto_suite_names); |
| void GetSupportedDataSdesCryptoSuiteNames( |
| const rtc::CryptoOptions& crypto_options, |
| std::vector<std::string>* crypto_suite_names); |
| |
| } // namespace cricket |
| |
| #endif // PC_MEDIASESSION_H_ |