From 36d22d82aa202bb199967e9512281e9a53db42c9 Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Sun, 7 Apr 2024 21:33:14 +0200 Subject: Adding upstream version 115.7.0esr. Signed-off-by: Daniel Baumann --- third_party/libwebrtc/pc/media_session.h | 423 +++++++++++++++++++++++++++++++ 1 file changed, 423 insertions(+) create mode 100644 third_party/libwebrtc/pc/media_session.h (limited to 'third_party/libwebrtc/pc/media_session.h') diff --git a/third_party/libwebrtc/pc/media_session.h b/third_party/libwebrtc/pc/media_session.h new file mode 100644 index 0000000000..6548f9c436 --- /dev/null +++ b/third_party/libwebrtc/pc/media_session.h @@ -0,0 +1,423 @@ +/* + * 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_MEDIA_SESSION_H_ +#define PC_MEDIA_SESSION_H_ + +#include +#include +#include +#include + +#include "api/crypto/crypto_options.h" +#include "api/field_trials_view.h" +#include "api/media_types.h" +#include "api/rtp_parameters.h" +#include "api/rtp_transceiver_direction.h" +#include "media/base/media_constants.h" +#include "media/base/rid_description.h" +#include "media/base/stream_params.h" +#include "p2p/base/ice_credentials_iterator.h" +#include "p2p/base/transport_description.h" +#include "p2p/base/transport_description_factory.h" +#include "p2p/base/transport_info.h" +#include "pc/jsep_transport.h" +#include "pc/media_protocol_names.h" +#include "pc/session_description.h" +#include "pc/simulcast_description.h" +#include "rtc_base/memory/always_valid_pointer.h" +#include "rtc_base/unique_id_generator.h" + +namespace webrtc { + +// Forward declaration due to circular dependecy. +class ConnectionContext; + +} // namespace webrtc + +namespace cricket { + +class MediaEngineInterface; + +// Default RTCP CNAME for unit tests. +const char kDefaultRtcpCname[] = "DefaultRtcpCname"; + +// Options for an RtpSender contained with an media description/"m=" section. +// Note: Spec-compliant Simulcast and legacy simulcast are mutually exclusive. +struct SenderOptions { + std::string track_id; + std::vector stream_ids; + // Use RIDs and Simulcast Layers to indicate spec-compliant Simulcast. + std::vector rids; + SimulcastLayerList simulcast_layers; + // Use `num_sim_layers` to indicate legacy simulcast. + int num_sim_layers; +}; + +// Options for an individual media description/"m=" section. +struct MediaDescriptionOptions { + MediaDescriptionOptions(MediaType type, + const std::string& mid, + webrtc::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& stream_ids); + void AddVideoSender(const std::string& track_id, + const std::vector& stream_ids, + const std::vector& rids, + const SimulcastLayerList& simulcast_layers, + int num_sim_layers); + + MediaType type; + std::string mid; + webrtc::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 sender_options; + std::vector codec_preferences; + std::vector header_extensions; + + private: + // Doesn't DCHECK on `type`. + void AddSenderInternal(const std::string& track_id, + const std::vector& stream_ids, + const std::vector& rids, + const SimulcastLayerList& simulcast_layers, + 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; + + bool vad_enabled = true; // When disabled, removes all CN codecs from SDP. + bool rtcp_mux_enabled = true; + bool bundle_enabled = false; + bool offer_extmap_allow_mixed = false; + bool raw_packetization_for_video = false; + std::string rtcp_cname = kDefaultRtcpCname; + webrtc::CryptoOptions crypto_options; + // List of media description options in the same order that the media + // descriptions will be generated. + std::vector media_description_options; + std::vector pooled_ice_credentials; + + // Use the draft-ietf-mmusic-sctp-sdp-03 obsolete syntax for SCTP + // datachannels. + // Default is true for backwards compatibility with clients that use + // this internal interface. + bool use_obsolete_sctp_sdp = 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: + // Simple constructor that does not set any configuration for the factory. + // When using this constructor, the methods below can be used to set the + // configuration. + // The TransportDescriptionFactory and the UniqueRandomIdGenerator are not + // owned by MediaSessionDescriptionFactory, so they must be kept alive by the + // user of this class. + MediaSessionDescriptionFactory(const TransportDescriptionFactory* factory, + rtc::UniqueRandomIdGenerator* ssrc_generator); + // This helper automatically sets up the factory to get its configuration + // from the specified MediaEngine + MediaSessionDescriptionFactory(cricket::MediaEngineInterface* media_engine, + bool rtx_enabled, + rtc::UniqueRandomIdGenerator* ssrc_generator, + 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); + const VideoCodecs& video_sendrecv_codecs() const; + const VideoCodecs& video_send_codecs() const; + const VideoCodecs& video_recv_codecs() const; + void set_video_codecs(const VideoCodecs& send_codecs, + const VideoCodecs& recv_codecs); + RtpHeaderExtensions filtered_rtp_header_extensions( + RtpHeaderExtensions extensions) const; + 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; + } + + void set_is_unified_plan(bool is_unified_plan) { + is_unified_plan_ = is_unified_plan; + } + + std::unique_ptr CreateOffer( + const MediaSessionOptions& options, + const SessionDescription* current_description) const; + std::unique_ptr CreateAnswer( + const SessionDescription* offer, + const MediaSessionOptions& options, + const SessionDescription* current_description) const; + + private: + struct AudioVideoRtpHeaderExtensions { + RtpHeaderExtensions audio; + RtpHeaderExtensions video; + }; + + const AudioCodecs& GetAudioCodecsForOffer( + const webrtc::RtpTransceiverDirection& direction) const; + const AudioCodecs& GetAudioCodecsForAnswer( + const webrtc::RtpTransceiverDirection& offer, + const webrtc::RtpTransceiverDirection& answer) const; + const VideoCodecs& GetVideoCodecsForOffer( + const webrtc::RtpTransceiverDirection& direction) const; + const VideoCodecs& GetVideoCodecsForAnswer( + const webrtc::RtpTransceiverDirection& offer, + const webrtc::RtpTransceiverDirection& answer) const; + void GetCodecsForOffer( + const std::vector& current_active_contents, + AudioCodecs* audio_codecs, + VideoCodecs* video_codecs) const; + void GetCodecsForAnswer( + const std::vector& current_active_contents, + const SessionDescription& remote_offer, + AudioCodecs* audio_codecs, + VideoCodecs* video_codecs) const; + AudioVideoRtpHeaderExtensions GetOfferedRtpHeaderExtensionsWithIds( + const std::vector& current_active_contents, + bool extmap_allow_mixed, + const std::vector& media_description_options) + const; + bool AddTransportOffer(const std::string& content_name, + const TransportOptions& transport_options, + const SessionDescription* current_desc, + SessionDescription* offer, + IceCredentialsIterator* ice_credentials) const; + + std::unique_ptr CreateTransportAnswer( + const std::string& content_name, + const SessionDescription* offer_desc, + const TransportOptions& transport_options, + const SessionDescription* current_desc, + bool require_transport_attributes, + IceCredentialsIterator* ice_credentials) 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, + IceCredentialsIterator* ice_credentials) 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, + IceCredentialsIterator* ice_credentials) const; + + bool AddDataContentForOffer( + const MediaDescriptionOptions& media_description_options, + const MediaSessionOptions& session_options, + const ContentInfo* current_content, + const SessionDescription* current_description, + StreamParamsVec* current_streams, + SessionDescription* desc, + IceCredentialsIterator* ice_credentials) const; + + bool AddUnsupportedContentForOffer( + const MediaDescriptionOptions& media_description_options, + const MediaSessionOptions& session_options, + const ContentInfo* current_content, + const SessionDescription* current_description, + SessionDescription* desc, + IceCredentialsIterator* ice_credentials) 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, + const RtpHeaderExtensions& default_audio_rtp_header_extensions, + StreamParamsVec* current_streams, + SessionDescription* answer, + IceCredentialsIterator* ice_credentials) 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, + const RtpHeaderExtensions& default_video_rtp_header_extensions, + StreamParamsVec* current_streams, + SessionDescription* answer, + IceCredentialsIterator* ice_credentials) 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, + StreamParamsVec* current_streams, + SessionDescription* answer, + IceCredentialsIterator* ice_credentials) const; + + bool AddUnsupportedContentForAnswer( + 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, + SessionDescription* answer, + IceCredentialsIterator* ice_credentials) const; + + void ComputeAudioCodecsIntersectionAndUnion(); + + void ComputeVideoCodecsIntersectionAndUnion(); + + rtc::UniqueRandomIdGenerator* ssrc_generator() const { + return ssrc_generator_.get(); + } + + bool is_unified_plan_ = false; + 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_; + VideoCodecs video_send_codecs_; + VideoCodecs video_recv_codecs_; + // Intersection of send and recv. + VideoCodecs video_sendrecv_codecs_; + // Union of send and recv. + VideoCodecs all_video_codecs_; + // This object may or may not be owned by this class. + webrtc::AlwaysValidPointer const + ssrc_generator_; + bool enable_encrypted_rtp_header_extensions_ = false; + // TODO(zhihuang): Rename secure_ to sdec_policy_; rename the related getter + // and setter. + SecurePolicy secure_ = SEC_DISABLED; + 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); +bool IsUnsupportedContent(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* GetFirstMediaContent(const SessionDescription* sdesc, + MediaType media_type); +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 SctpDataContentDescription* GetFirstSctpDataContentDescription( + 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* GetFirstMediaContent(SessionDescription* sdesc, + MediaType media_type); +ContentInfo* GetFirstAudioContent(SessionDescription* sdesc); +ContentInfo* GetFirstVideoContent(SessionDescription* sdesc); +ContentInfo* GetFirstDataContent(SessionDescription* sdesc); +AudioContentDescription* GetFirstAudioContentDescription( + SessionDescription* sdesc); +VideoContentDescription* GetFirstVideoContentDescription( + SessionDescription* sdesc); +SctpDataContentDescription* GetFirstSctpDataContentDescription( + SessionDescription* sdesc); + +// Helper functions to return crypto suites used for SDES. +void GetSupportedAudioSdesCryptoSuites( + const webrtc::CryptoOptions& crypto_options, + std::vector* crypto_suites); +void GetSupportedVideoSdesCryptoSuites( + const webrtc::CryptoOptions& crypto_options, + std::vector* crypto_suites); +void GetSupportedDataSdesCryptoSuites( + const webrtc::CryptoOptions& crypto_options, + std::vector* crypto_suites); +void GetSupportedAudioSdesCryptoSuiteNames( + const webrtc::CryptoOptions& crypto_options, + std::vector* crypto_suite_names); +void GetSupportedVideoSdesCryptoSuiteNames( + const webrtc::CryptoOptions& crypto_options, + std::vector* crypto_suite_names); +void GetSupportedDataSdesCryptoSuiteNames( + const webrtc::CryptoOptions& crypto_options, + std::vector* crypto_suite_names); + +} // namespace cricket + +#endif // PC_MEDIA_SESSION_H_ -- cgit v1.2.3