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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 <memory>
#include <string>
#include <vector>
#include "api/media_types.h"
#include "api/rtc_error.h"
#include "media/base/media_engine.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/codec_vendor.h"
#include "pc/media_options.h"
#include "pc/session_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 webrtc {
// 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, the UniqueRandomIdGenerator, and the
// PayloadTypeSuggester are not owned by MediaSessionDescriptionFactory, so
// they must be kept alive by the user of this class.
MediaSessionDescriptionFactory(MediaEngineInterface* media_engine,
bool rtx_enabled,
UniqueRandomIdGenerator* ssrc_generator,
const TransportDescriptionFactory* factory,
CodecLookupHelper* codec_lookup_helper);
RtpHeaderExtensions filtered_rtp_header_extensions(
RtpHeaderExtensions extensions) const;
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;
}
RTCErrorOr<std::unique_ptr<SessionDescription>> CreateOfferOrError(
const MediaSessionOptions& options,
const SessionDescription* current_description) const;
RTCErrorOr<std::unique_ptr<SessionDescription>> CreateAnswerOrError(
const SessionDescription* offer,
const MediaSessionOptions& options,
const SessionDescription* current_description) const;
private:
struct AudioVideoRtpHeaderExtensions {
RtpHeaderExtensions audio;
RtpHeaderExtensions video;
};
AudioVideoRtpHeaderExtensions GetOfferedRtpHeaderExtensionsWithIds(
const std::vector<const ContentInfo*>& current_active_contents,
bool extmap_allow_mixed,
const std::vector<MediaDescriptionOptions>& media_description_options)
const;
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;
RTCError AddTransportAnswer(const std::string& content_name,
const TransportDescription& transport_desc,
SessionDescription* answer_desc) const;
// Helpers for adding media contents to the SessionDescription.
RTCError AddRtpContentForOffer(
const MediaDescriptionOptions& media_description_options,
const MediaSessionOptions& session_options,
const ContentInfo* current_content,
const SessionDescription* current_description,
const RtpHeaderExtensions& header_extensions,
StreamParamsVec* current_streams,
SessionDescription* desc,
IceCredentialsIterator* ice_credentials) const;
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;
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;
RTCError AddRtpContentForAnswer(
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 RtpHeaderExtensions& header_extensions,
StreamParamsVec* current_streams,
SessionDescription* answer,
IceCredentialsIterator* ice_credentials) const;
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;
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;
UniqueRandomIdGenerator* ssrc_generator() const {
return ssrc_generator_.get();
}
bool is_unified_plan_ = false;
// This object may or may not be owned by this class.
AlwaysValidPointer<UniqueRandomIdGenerator> const ssrc_generator_;
bool enable_encrypted_rtp_header_extensions_ = true;
const TransportDescriptionFactory* transport_desc_factory_;
CodecLookupHelper* codec_lookup_helper_;
bool payload_types_in_transport_trial_enabled_;
};
// 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,
webrtc::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,
webrtc::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,
webrtc::MediaType media_type);
ContentInfo* GetFirstAudioContent(ContentInfos* contents);
ContentInfo* GetFirstVideoContent(ContentInfos* contents);
ContentInfo* GetFirstDataContent(ContentInfos* contents);
ContentInfo* GetFirstMediaContent(SessionDescription* sdesc,
webrtc::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 webrtc
// Re-export symbols from the webrtc namespace for backwards compatibility.
// TODO(bugs.webrtc.org/4222596): Remove once all references are updated.
#ifdef WEBRTC_ALLOW_DEPRECATED_NAMESPACES
namespace cricket {
using ::webrtc::GetFirstAudioContent;
using ::webrtc::GetFirstAudioContentDescription;
using ::webrtc::GetFirstDataContent;
using ::webrtc::GetFirstMediaContent;
using ::webrtc::GetFirstSctpDataContentDescription;
using ::webrtc::GetFirstVideoContent;
using ::webrtc::GetFirstVideoContentDescription;
using ::webrtc::IsAudioContent;
using ::webrtc::IsDataContent;
using ::webrtc::IsMediaContent;
using ::webrtc::IsUnsupportedContent;
using ::webrtc::IsVideoContent;
using ::webrtc::MediaSessionDescriptionFactory;
} // namespace cricket
#endif // WEBRTC_ALLOW_DEPRECATED_NAMESPACES
#endif // PC_MEDIA_SESSION_H_
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