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/*
* Copyright 2004 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
// Types and classes used in media session descriptions.
#ifndef PC_MEDIA_SESSION_H_
#define PC_MEDIA_SESSION_H_
#include <map>
#include <memory>
#include <string>
#include <vector>
#include "api/crypto/crypto_options.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<std::string> stream_ids;
// Use RIDs and Simulcast Layers to indicate spec-compliant Simulcast.
std::vector<RidDescription> 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<std::string>& stream_ids);
void AddVideoSender(const std::string& track_id,
const std::vector<std::string>& stream_ids,
const std::vector<RidDescription>& 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<SenderOptions> sender_options;
std::vector<webrtc::RtpCodecCapability> codec_preferences;
std::vector<webrtc::RtpHeaderExtensionCapability> header_extensions;
private:
// Doesn't DCHECK on `type`.
void AddSenderInternal(const std::string& track_id,
const std::vector<std::string>& stream_ids,
const std::vector<RidDescription>& 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<MediaDescriptionOptions> media_description_options;
std::vector<IceParameters> 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:
// This constructor automatically sets up the factory to get its configuration
// from the specified MediaEngine (when provided).
// The TransportDescriptionFactory and the UniqueRandomIdGenerator are not
// owned by MediaSessionDescriptionFactory, so they must be kept alive by the
// user of this class.
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;
}
webrtc::RTCErrorOr<std::unique_ptr<SessionDescription>> CreateOfferOrError(
const MediaSessionOptions& options,
const SessionDescription* current_description) const;
webrtc::RTCErrorOr<std::unique_ptr<SessionDescription>> CreateAnswerOrError(
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<const ContentInfo*>& current_active_contents,
AudioCodecs* audio_codecs,
VideoCodecs* video_codecs) const;
void GetCodecsForAnswer(
const std::vector<const ContentInfo*>& current_active_contents,
const SessionDescription& remote_offer,
AudioCodecs* audio_codecs,
VideoCodecs* video_codecs) const;
AudioVideoRtpHeaderExtensions GetOfferedRtpHeaderExtensionsWithIds(
const std::vector<const ContentInfo*>& current_active_contents,
bool extmap_allow_mixed,
const std::vector<MediaDescriptionOptions>& media_description_options)
const;
webrtc::RTCError AddTransportOffer(
const std::string& content_name,
const TransportOptions& transport_options,
const SessionDescription* current_desc,
SessionDescription* offer,
IceCredentialsIterator* ice_credentials) const;
std::unique_ptr<TransportDescription> 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;
webrtc::RTCError 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.
webrtc::RTCErrorOr<AudioCodecs> GetNegotiatedAudioCodecsForOffer(
const MediaDescriptionOptions& media_description_options,
const MediaSessionOptions& session_options,
const ContentInfo* current_content,
const AudioCodecs& audio_codecs) const;
webrtc::RTCError 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;
webrtc::RTCErrorOr<AudioCodecs> GetNegotiatedVideoCodecsForOffer(
const MediaDescriptionOptions& media_description_options,
const MediaSessionOptions& session_options,
const ContentInfo* current_content,
const VideoCodecs& video_codecs) const;
webrtc::RTCError 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;
webrtc::RTCError 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;
webrtc::RTCError AddUnsupportedContentForOffer(
const MediaDescriptionOptions& media_description_options,
const MediaSessionOptions& session_options,
const ContentInfo* current_content,
const SessionDescription* current_description,
SessionDescription* desc,
IceCredentialsIterator* ice_credentials) const;
webrtc::RTCErrorOr<AudioCodecs> GetNegotiatedAudioCodecsForAnswer(
const MediaDescriptionOptions& media_description_options,
const MediaSessionOptions& session_options,
const ContentInfo* current_content,
const AudioCodecs& audio_codecs,
const AudioCodecs& supported_audio_codecs) const;
webrtc::RTCError 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& rtp_header_extensions,
StreamParamsVec* current_streams,
SessionDescription* answer,
IceCredentialsIterator* ice_credentials) const;
webrtc::RTCErrorOr<AudioCodecs> GetNegotiatedVideoCodecsForAnswer(
const MediaDescriptionOptions& media_description_options,
const MediaSessionOptions& session_options,
const ContentInfo* current_content,
const VideoCodecs& video_codecs,
const VideoCodecs& supported_video_codecs) const;
webrtc::RTCError 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& rtp_header_extensions,
StreamParamsVec* current_streams,
SessionDescription* answer,
IceCredentialsIterator* ice_credentials) const;
webrtc::RTCError 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;
webrtc::RTCError 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<rtc::UniqueRandomIdGenerator> 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);
} // namespace cricket
#endif // PC_MEDIA_SESSION_H_