409 lines
17 KiB
C++
409 lines
17 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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// Types and classes used in media session descriptions.
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#ifndef PC_MEDIA_SESSION_H_
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#define PC_MEDIA_SESSION_H_
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#include <map>
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#include <memory>
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#include <string>
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#include <vector>
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#include "api/crypto/crypto_options.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_direction.h"
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#include "media/base/media_constants.h"
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#include "media/base/rid_description.h"
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#include "media/base/stream_params.h"
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#include "p2p/base/ice_credentials_iterator.h"
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#include "p2p/base/transport_description.h"
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#include "p2p/base/transport_description_factory.h"
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#include "p2p/base/transport_info.h"
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#include "pc/jsep_transport.h"
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#include "pc/media_protocol_names.h"
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#include "pc/session_description.h"
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#include "pc/simulcast_description.h"
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#include "rtc_base/unique_id_generator.h"
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namespace cricket {
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class ChannelManager;
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// Default RTCP CNAME for unit tests.
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const char kDefaultRtcpCname[] = "DefaultRtcpCname";
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// Options for an RtpSender contained with an media description/"m=" section.
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// Note: Spec-compliant Simulcast and legacy simulcast are mutually exclusive.
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struct SenderOptions {
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std::string track_id;
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std::vector<std::string> stream_ids;
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// Use RIDs and Simulcast Layers to indicate spec-compliant Simulcast.
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std::vector<RidDescription> rids;
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SimulcastLayerList simulcast_layers;
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// Use |num_sim_layers| to indicate legacy simulcast.
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int num_sim_layers;
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};
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// Options for an individual media description/"m=" section.
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struct MediaDescriptionOptions {
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MediaDescriptionOptions(MediaType type,
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const std::string& mid,
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webrtc::RtpTransceiverDirection direction,
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bool stopped)
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: type(type), mid(mid), direction(direction), stopped(stopped) {}
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// TODO(deadbeef): When we don't support Plan B, there will only be one
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// sender per media description and this can be simplified.
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void AddAudioSender(const std::string& track_id,
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const std::vector<std::string>& stream_ids);
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void AddVideoSender(const std::string& track_id,
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const std::vector<std::string>& stream_ids,
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const std::vector<RidDescription>& rids,
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const SimulcastLayerList& simulcast_layers,
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int num_sim_layers);
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MediaType type;
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std::string mid;
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webrtc::RtpTransceiverDirection direction;
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bool stopped;
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TransportOptions transport_options;
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// Note: There's no equivalent "RtpReceiverOptions" because only send
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// stream information goes in the local descriptions.
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std::vector<SenderOptions> sender_options;
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std::vector<webrtc::RtpCodecCapability> codec_preferences;
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std::vector<webrtc::RtpHeaderExtensionCapability> header_extensions;
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private:
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// Doesn't DCHECK on |type|.
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void AddSenderInternal(const std::string& track_id,
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const std::vector<std::string>& stream_ids,
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const std::vector<RidDescription>& rids,
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const SimulcastLayerList& simulcast_layers,
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int num_sim_layers);
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};
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// Provides a mechanism for describing how m= sections should be generated.
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// The m= section with index X will use media_description_options[X]. There
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// must be an option for each existing section if creating an answer, or a
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// subsequent offer.
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struct MediaSessionOptions {
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MediaSessionOptions() {}
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bool has_audio() const { return HasMediaDescription(MEDIA_TYPE_AUDIO); }
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bool has_video() const { return HasMediaDescription(MEDIA_TYPE_VIDEO); }
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bool has_data() const { return HasMediaDescription(MEDIA_TYPE_DATA); }
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bool HasMediaDescription(MediaType type) const;
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bool vad_enabled = true; // When disabled, removes all CN codecs from SDP.
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bool rtcp_mux_enabled = true;
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bool bundle_enabled = false;
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bool offer_extmap_allow_mixed = false;
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bool raw_packetization_for_video = false;
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std::string rtcp_cname = kDefaultRtcpCname;
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webrtc::CryptoOptions crypto_options;
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// List of media description options in the same order that the media
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// descriptions will be generated.
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std::vector<MediaDescriptionOptions> media_description_options;
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std::vector<IceParameters> pooled_ice_credentials;
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// Use the draft-ietf-mmusic-sctp-sdp-03 obsolete syntax for SCTP
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// datachannels.
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// Default is true for backwards compatibility with clients that use
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// this internal interface.
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bool use_obsolete_sctp_sdp = true;
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};
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// Creates media session descriptions according to the supplied codecs and
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// other fields, as well as the supplied per-call options.
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// When creating answers, performs the appropriate negotiation
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// of the various fields to determine the proper result.
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class MediaSessionDescriptionFactory {
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public:
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// Simple constructor that does not set any configuration for the factory.
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// When using this constructor, the methods below can be used to set the
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// configuration.
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// The TransportDescriptionFactory and the UniqueRandomIdGenerator are not
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// owned by MediaSessionDescriptionFactory, so they must be kept alive by the
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// user of this class.
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MediaSessionDescriptionFactory(const TransportDescriptionFactory* factory,
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rtc::UniqueRandomIdGenerator* ssrc_generator);
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// This helper automatically sets up the factory to get its configuration
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// from the specified ChannelManager.
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MediaSessionDescriptionFactory(ChannelManager* cmanager,
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const TransportDescriptionFactory* factory,
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rtc::UniqueRandomIdGenerator* ssrc_generator);
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const AudioCodecs& audio_sendrecv_codecs() const;
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const AudioCodecs& audio_send_codecs() const;
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const AudioCodecs& audio_recv_codecs() const;
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void set_audio_codecs(const AudioCodecs& send_codecs,
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const AudioCodecs& recv_codecs);
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const VideoCodecs& video_sendrecv_codecs() const;
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const VideoCodecs& video_send_codecs() const;
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const VideoCodecs& video_recv_codecs() const;
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void set_video_codecs(const VideoCodecs& send_codecs,
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const VideoCodecs& recv_codecs);
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RtpHeaderExtensions filtered_rtp_header_extensions(
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RtpHeaderExtensions extensions) const;
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SecurePolicy secure() const { return secure_; }
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void set_secure(SecurePolicy s) { secure_ = s; }
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void set_enable_encrypted_rtp_header_extensions(bool enable) {
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enable_encrypted_rtp_header_extensions_ = enable;
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}
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void set_is_unified_plan(bool is_unified_plan) {
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is_unified_plan_ = is_unified_plan;
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}
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std::unique_ptr<SessionDescription> CreateOffer(
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const MediaSessionOptions& options,
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const SessionDescription* current_description) const;
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std::unique_ptr<SessionDescription> CreateAnswer(
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const SessionDescription* offer,
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const MediaSessionOptions& options,
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const SessionDescription* current_description) const;
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private:
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struct AudioVideoRtpHeaderExtensions {
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RtpHeaderExtensions audio;
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RtpHeaderExtensions video;
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};
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const AudioCodecs& GetAudioCodecsForOffer(
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const webrtc::RtpTransceiverDirection& direction) const;
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const AudioCodecs& GetAudioCodecsForAnswer(
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const webrtc::RtpTransceiverDirection& offer,
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const webrtc::RtpTransceiverDirection& answer) const;
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const VideoCodecs& GetVideoCodecsForOffer(
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const webrtc::RtpTransceiverDirection& direction) const;
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const VideoCodecs& GetVideoCodecsForAnswer(
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const webrtc::RtpTransceiverDirection& offer,
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const webrtc::RtpTransceiverDirection& answer) const;
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void GetCodecsForOffer(
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const std::vector<const ContentInfo*>& current_active_contents,
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AudioCodecs* audio_codecs,
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VideoCodecs* video_codecs) const;
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void GetCodecsForAnswer(
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const std::vector<const ContentInfo*>& current_active_contents,
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const SessionDescription& remote_offer,
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AudioCodecs* audio_codecs,
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VideoCodecs* video_codecs) const;
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AudioVideoRtpHeaderExtensions GetOfferedRtpHeaderExtensionsWithIds(
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const std::vector<const ContentInfo*>& current_active_contents,
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bool extmap_allow_mixed,
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const std::vector<MediaDescriptionOptions>& media_description_options)
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const;
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bool AddTransportOffer(const std::string& content_name,
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const TransportOptions& transport_options,
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const SessionDescription* current_desc,
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SessionDescription* offer,
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IceCredentialsIterator* ice_credentials) const;
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std::unique_ptr<TransportDescription> CreateTransportAnswer(
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const std::string& content_name,
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const SessionDescription* offer_desc,
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const TransportOptions& transport_options,
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const SessionDescription* current_desc,
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bool require_transport_attributes,
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IceCredentialsIterator* ice_credentials) const;
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bool AddTransportAnswer(const std::string& content_name,
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const TransportDescription& transport_desc,
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SessionDescription* answer_desc) const;
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// Helpers for adding media contents to the SessionDescription. Returns true
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// it succeeds or the media content is not needed, or false if there is any
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// error.
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bool AddAudioContentForOffer(
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const MediaDescriptionOptions& media_description_options,
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const MediaSessionOptions& session_options,
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const ContentInfo* current_content,
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const SessionDescription* current_description,
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const RtpHeaderExtensions& audio_rtp_extensions,
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const AudioCodecs& audio_codecs,
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StreamParamsVec* current_streams,
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SessionDescription* desc,
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IceCredentialsIterator* ice_credentials) const;
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bool AddVideoContentForOffer(
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const MediaDescriptionOptions& media_description_options,
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const MediaSessionOptions& session_options,
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const ContentInfo* current_content,
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const SessionDescription* current_description,
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const RtpHeaderExtensions& video_rtp_extensions,
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const VideoCodecs& video_codecs,
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StreamParamsVec* current_streams,
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SessionDescription* desc,
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IceCredentialsIterator* ice_credentials) const;
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bool AddDataContentForOffer(
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const MediaDescriptionOptions& media_description_options,
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const MediaSessionOptions& session_options,
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const ContentInfo* current_content,
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const SessionDescription* current_description,
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StreamParamsVec* current_streams,
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SessionDescription* desc,
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IceCredentialsIterator* ice_credentials) const;
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bool AddUnsupportedContentForOffer(
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const MediaDescriptionOptions& media_description_options,
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const MediaSessionOptions& session_options,
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const ContentInfo* current_content,
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const SessionDescription* current_description,
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SessionDescription* desc,
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IceCredentialsIterator* ice_credentials) const;
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bool AddAudioContentForAnswer(
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const MediaDescriptionOptions& media_description_options,
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const MediaSessionOptions& session_options,
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const ContentInfo* offer_content,
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const SessionDescription* offer_description,
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const ContentInfo* current_content,
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const SessionDescription* current_description,
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const TransportInfo* bundle_transport,
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const AudioCodecs& audio_codecs,
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const RtpHeaderExtensions& default_audio_rtp_header_extensions,
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StreamParamsVec* current_streams,
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SessionDescription* answer,
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IceCredentialsIterator* ice_credentials) const;
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bool AddVideoContentForAnswer(
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const MediaDescriptionOptions& media_description_options,
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const MediaSessionOptions& session_options,
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const ContentInfo* offer_content,
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const SessionDescription* offer_description,
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const ContentInfo* current_content,
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const SessionDescription* current_description,
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const TransportInfo* bundle_transport,
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const VideoCodecs& video_codecs,
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const RtpHeaderExtensions& default_video_rtp_header_extensions,
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StreamParamsVec* current_streams,
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SessionDescription* answer,
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IceCredentialsIterator* ice_credentials) const;
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bool AddDataContentForAnswer(
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const MediaDescriptionOptions& media_description_options,
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const MediaSessionOptions& session_options,
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const ContentInfo* offer_content,
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const SessionDescription* offer_description,
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const ContentInfo* current_content,
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const SessionDescription* current_description,
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const TransportInfo* bundle_transport,
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StreamParamsVec* current_streams,
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SessionDescription* answer,
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IceCredentialsIterator* ice_credentials) const;
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bool AddUnsupportedContentForAnswer(
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const MediaDescriptionOptions& media_description_options,
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const MediaSessionOptions& session_options,
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const ContentInfo* offer_content,
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const SessionDescription* offer_description,
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const ContentInfo* current_content,
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const SessionDescription* current_description,
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const TransportInfo* bundle_transport,
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SessionDescription* answer,
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IceCredentialsIterator* ice_credentials) const;
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void ComputeAudioCodecsIntersectionAndUnion();
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void ComputeVideoCodecsIntersectionAndUnion();
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bool is_unified_plan_ = false;
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AudioCodecs audio_send_codecs_;
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AudioCodecs audio_recv_codecs_;
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// Intersection of send and recv.
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AudioCodecs audio_sendrecv_codecs_;
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// Union of send and recv.
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AudioCodecs all_audio_codecs_;
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VideoCodecs video_send_codecs_;
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VideoCodecs video_recv_codecs_;
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// Intersection of send and recv.
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VideoCodecs video_sendrecv_codecs_;
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// Union of send and recv.
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VideoCodecs all_video_codecs_;
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// This object is not owned by the channel so it must outlive it.
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rtc::UniqueRandomIdGenerator* const ssrc_generator_;
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bool enable_encrypted_rtp_header_extensions_ = false;
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// TODO(zhihuang): Rename secure_ to sdec_policy_; rename the related getter
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// and setter.
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SecurePolicy secure_ = SEC_DISABLED;
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const TransportDescriptionFactory* transport_desc_factory_;
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};
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// Convenience functions.
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bool IsMediaContent(const ContentInfo* content);
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bool IsAudioContent(const ContentInfo* content);
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bool IsVideoContent(const ContentInfo* content);
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bool IsDataContent(const ContentInfo* content);
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bool IsUnsupportedContent(const ContentInfo* content);
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const ContentInfo* GetFirstMediaContent(const ContentInfos& contents,
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MediaType media_type);
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const ContentInfo* GetFirstAudioContent(const ContentInfos& contents);
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const ContentInfo* GetFirstVideoContent(const ContentInfos& contents);
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const ContentInfo* GetFirstDataContent(const ContentInfos& contents);
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const ContentInfo* GetFirstMediaContent(const SessionDescription* sdesc,
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MediaType media_type);
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const ContentInfo* GetFirstAudioContent(const SessionDescription* sdesc);
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const ContentInfo* GetFirstVideoContent(const SessionDescription* sdesc);
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const ContentInfo* GetFirstDataContent(const SessionDescription* sdesc);
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const AudioContentDescription* GetFirstAudioContentDescription(
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const SessionDescription* sdesc);
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const VideoContentDescription* GetFirstVideoContentDescription(
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const SessionDescription* sdesc);
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const SctpDataContentDescription* GetFirstSctpDataContentDescription(
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const SessionDescription* sdesc);
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// Non-const versions of the above functions.
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// Useful when modifying an existing description.
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ContentInfo* GetFirstMediaContent(ContentInfos* contents, MediaType media_type);
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ContentInfo* GetFirstAudioContent(ContentInfos* contents);
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ContentInfo* GetFirstVideoContent(ContentInfos* contents);
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ContentInfo* GetFirstDataContent(ContentInfos* contents);
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ContentInfo* GetFirstMediaContent(SessionDescription* sdesc,
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MediaType media_type);
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ContentInfo* GetFirstAudioContent(SessionDescription* sdesc);
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ContentInfo* GetFirstVideoContent(SessionDescription* sdesc);
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ContentInfo* GetFirstDataContent(SessionDescription* sdesc);
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AudioContentDescription* GetFirstAudioContentDescription(
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SessionDescription* sdesc);
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VideoContentDescription* GetFirstVideoContentDescription(
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SessionDescription* sdesc);
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SctpDataContentDescription* GetFirstSctpDataContentDescription(
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SessionDescription* sdesc);
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// Helper functions to return crypto suites used for SDES.
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void GetSupportedAudioSdesCryptoSuites(
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const webrtc::CryptoOptions& crypto_options,
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std::vector<int>* crypto_suites);
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void GetSupportedVideoSdesCryptoSuites(
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const webrtc::CryptoOptions& crypto_options,
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std::vector<int>* crypto_suites);
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void GetSupportedDataSdesCryptoSuites(
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const webrtc::CryptoOptions& crypto_options,
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std::vector<int>* crypto_suites);
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void GetSupportedAudioSdesCryptoSuiteNames(
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const webrtc::CryptoOptions& crypto_options,
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std::vector<std::string>* crypto_suite_names);
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void GetSupportedVideoSdesCryptoSuiteNames(
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const webrtc::CryptoOptions& crypto_options,
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std::vector<std::string>* crypto_suite_names);
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void GetSupportedDataSdesCryptoSuiteNames(
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const webrtc::CryptoOptions& crypto_options,
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std::vector<std::string>* crypto_suite_names);
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} // namespace cricket
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#endif // PC_MEDIA_SESSION_H_
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