webrtc/pc/session_description.h
Henrik Boström f8187e0a82 [Unified Plan] Support multiple BUNDLE groups.
In this CL, JsepTransportController and MediaSessionDescriptionFactory
are updated not to assume that there only exists at most a single BUNDLE
group but a list of N groups. This makes it possible to create multiple
BUNDLE groups by having multiple "a=group:BUNDLE" lines in the SDP.

This makes it possible to have some m= sections in one group and some
other m= sections in another group. For example, you could group all
audio m= sections in one group and all video m= sections in another
group. This enables "send all audio tracks on one transport and all
video tracks on another transport" in Unified Plan. This is something
that was possible in Plan B because all ssrcs in the same m= section
were implicitly bundled together forming a group of audio m= section and
video m= section (even without use of the BUNDLE tag).

PeerConnection will never create multiple BUNDLE groups by default, but
upon setting SDP with multiple BUNDLE groups the PeerConnection will
accept them if configured to accept BUNDLE. This makes it possible to
accept an SFU's BUNDLE offer without having to SDP munge the answer.

C++ unit tests are added. This fix has also been verified manually on:
https://jsfiddle.net/henbos/to89L6ce/43/

Without fix: 0+2 get bundled, 1+3 don't get bundled.
With fix: 0+2 get bundled in first group, 1+3 get bundled in second
group.

Bug: webrtc:10208
Change-Id: Iaf451fa5459c484730c8018274166ef154b19af8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214487
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33838}
2021-04-27 05:53:37 +00:00

627 lines
22 KiB
C++

/*
* 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.
*/
#ifndef PC_SESSION_DESCRIPTION_H_
#define PC_SESSION_DESCRIPTION_H_
#include <stddef.h>
#include <stdint.h>
#include <algorithm>
#include <iosfwd>
#include <memory>
#include <string>
#include <utility>
#include <vector>
#include "absl/memory/memory.h"
#include "api/crypto_params.h"
#include "api/media_types.h"
#include "api/rtp_parameters.h"
#include "api/rtp_transceiver_direction.h"
#include "api/rtp_transceiver_interface.h"
#include "media/base/codec.h"
#include "media/base/media_channel.h"
#include "media/base/media_constants.h"
#include "media/base/rid_description.h"
#include "media/base/stream_params.h"
#include "p2p/base/transport_description.h"
#include "p2p/base/transport_info.h"
#include "pc/media_protocol_names.h"
#include "pc/simulcast_description.h"
#include "rtc_base/checks.h"
#include "rtc_base/socket_address.h"
#include "rtc_base/system/rtc_export.h"
namespace cricket {
typedef std::vector<AudioCodec> AudioCodecs;
typedef std::vector<VideoCodec> VideoCodecs;
typedef std::vector<CryptoParams> CryptoParamsVec;
typedef std::vector<webrtc::RtpExtension> RtpHeaderExtensions;
// RTC4585 RTP/AVPF
extern const char kMediaProtocolAvpf[];
// RFC5124 RTP/SAVPF
extern const char kMediaProtocolSavpf[];
extern const char kMediaProtocolDtlsSavpf[];
// Options to control how session descriptions are generated.
const int kAutoBandwidth = -1;
class AudioContentDescription;
class VideoContentDescription;
class SctpDataContentDescription;
class UnsupportedContentDescription;
// Describes a session description media section. There are subclasses for each
// media type (audio, video, data) that will have additional information.
class MediaContentDescription {
public:
MediaContentDescription() = default;
virtual ~MediaContentDescription() = default;
virtual MediaType type() const = 0;
// Try to cast this media description to an AudioContentDescription. Returns
// nullptr if the cast fails.
virtual AudioContentDescription* as_audio() { return nullptr; }
virtual const AudioContentDescription* as_audio() const { return nullptr; }
// Try to cast this media description to a VideoContentDescription. Returns
// nullptr if the cast fails.
virtual VideoContentDescription* as_video() { return nullptr; }
virtual const VideoContentDescription* as_video() const { return nullptr; }
virtual SctpDataContentDescription* as_sctp() { return nullptr; }
virtual const SctpDataContentDescription* as_sctp() const { return nullptr; }
virtual UnsupportedContentDescription* as_unsupported() { return nullptr; }
virtual const UnsupportedContentDescription* as_unsupported() const {
return nullptr;
}
virtual bool has_codecs() const = 0;
// Copy operator that returns an unique_ptr.
// Not a virtual function.
// If a type-specific variant of Clone() is desired, override it, or
// simply use std::make_unique<typename>(*this) instead of Clone().
std::unique_ptr<MediaContentDescription> Clone() const {
return absl::WrapUnique(CloneInternal());
}
// |protocol| is the expected media transport protocol, such as RTP/AVPF,
// RTP/SAVPF or SCTP/DTLS.
virtual std::string protocol() const { return protocol_; }
virtual void set_protocol(const std::string& protocol) {
protocol_ = protocol;
}
virtual webrtc::RtpTransceiverDirection direction() const {
return direction_;
}
virtual void set_direction(webrtc::RtpTransceiverDirection direction) {
direction_ = direction;
}
virtual bool rtcp_mux() const { return rtcp_mux_; }
virtual void set_rtcp_mux(bool mux) { rtcp_mux_ = mux; }
virtual bool rtcp_reduced_size() const { return rtcp_reduced_size_; }
virtual void set_rtcp_reduced_size(bool reduced_size) {
rtcp_reduced_size_ = reduced_size;
}
// Indicates support for the remote network estimate packet type. This
// functionality is experimental and subject to change without notice.
virtual bool remote_estimate() const { return remote_estimate_; }
virtual void set_remote_estimate(bool remote_estimate) {
remote_estimate_ = remote_estimate;
}
virtual int bandwidth() const { return bandwidth_; }
virtual void set_bandwidth(int bandwidth) { bandwidth_ = bandwidth; }
virtual std::string bandwidth_type() const { return bandwidth_type_; }
virtual void set_bandwidth_type(std::string bandwidth_type) {
bandwidth_type_ = bandwidth_type;
}
virtual const std::vector<CryptoParams>& cryptos() const { return cryptos_; }
virtual void AddCrypto(const CryptoParams& params) {
cryptos_.push_back(params);
}
virtual void set_cryptos(const std::vector<CryptoParams>& cryptos) {
cryptos_ = cryptos;
}
virtual const RtpHeaderExtensions& rtp_header_extensions() const {
return rtp_header_extensions_;
}
virtual void set_rtp_header_extensions(
const RtpHeaderExtensions& extensions) {
rtp_header_extensions_ = extensions;
rtp_header_extensions_set_ = true;
}
virtual void AddRtpHeaderExtension(const webrtc::RtpExtension& ext) {
rtp_header_extensions_.push_back(ext);
rtp_header_extensions_set_ = true;
}
virtual 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).
virtual bool rtp_header_extensions_set() const {
return rtp_header_extensions_set_;
}
virtual const StreamParamsVec& streams() const { return send_streams_; }
// TODO(pthatcher): Remove this by giving mediamessage.cc access
// to MediaContentDescription
virtual StreamParamsVec& mutable_streams() { return send_streams_; }
virtual void AddStream(const StreamParams& stream) {
send_streams_.push_back(stream);
}
// Legacy streams have an ssrc, but nothing else.
void AddLegacyStream(uint32_t ssrc) {
AddStream(StreamParams::CreateLegacy(ssrc));
}
void AddLegacyStream(uint32_t ssrc, uint32_t fid_ssrc) {
StreamParams sp = StreamParams::CreateLegacy(ssrc);
sp.AddFidSsrc(ssrc, fid_ssrc);
AddStream(sp);
}
// Sets the CNAME of all StreamParams if it have not been set.
virtual void SetCnameIfEmpty(const std::string& cname) {
for (cricket::StreamParamsVec::iterator it = send_streams_.begin();
it != send_streams_.end(); ++it) {
if (it->cname.empty())
it->cname = cname;
}
}
virtual uint32_t first_ssrc() const {
if (send_streams_.empty()) {
return 0;
}
return send_streams_[0].first_ssrc();
}
virtual bool has_ssrcs() const {
if (send_streams_.empty()) {
return false;
}
return send_streams_[0].has_ssrcs();
}
virtual void set_conference_mode(bool enable) { conference_mode_ = enable; }
virtual bool conference_mode() const { return conference_mode_; }
// 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.
virtual void set_connection_address(const rtc::SocketAddress& address) {
connection_address_ = address;
}
virtual const rtc::SocketAddress& connection_address() const {
return connection_address_;
}
// Determines if it's allowed to mix one- and two-byte rtp header extensions
// within the same rtp stream.
enum ExtmapAllowMixed { kNo, kSession, kMedia };
virtual void set_extmap_allow_mixed_enum(
ExtmapAllowMixed new_extmap_allow_mixed) {
if (new_extmap_allow_mixed == kMedia &&
extmap_allow_mixed_enum_ == kSession) {
// Do not downgrade from session level to media level.
return;
}
extmap_allow_mixed_enum_ = new_extmap_allow_mixed;
}
virtual ExtmapAllowMixed extmap_allow_mixed_enum() const {
return extmap_allow_mixed_enum_;
}
virtual bool extmap_allow_mixed() const {
return extmap_allow_mixed_enum_ != kNo;
}
// Simulcast functionality.
virtual bool HasSimulcast() const { return !simulcast_.empty(); }
virtual SimulcastDescription& simulcast_description() { return simulcast_; }
virtual const SimulcastDescription& simulcast_description() const {
return simulcast_;
}
virtual void set_simulcast_description(
const SimulcastDescription& simulcast) {
simulcast_ = simulcast;
}
virtual const std::vector<RidDescription>& receive_rids() const {
return receive_rids_;
}
virtual void set_receive_rids(const std::vector<RidDescription>& rids) {
receive_rids_ = rids;
}
protected:
bool rtcp_mux_ = false;
bool rtcp_reduced_size_ = false;
bool remote_estimate_ = false;
int bandwidth_ = kAutoBandwidth;
std::string bandwidth_type_ = kApplicationSpecificBandwidth;
std::string protocol_;
std::vector<CryptoParams> cryptos_;
std::vector<webrtc::RtpExtension> rtp_header_extensions_;
bool rtp_header_extensions_set_ = false;
StreamParamsVec send_streams_;
bool conference_mode_ = false;
webrtc::RtpTransceiverDirection direction_ =
webrtc::RtpTransceiverDirection::kSendRecv;
rtc::SocketAddress connection_address_;
ExtmapAllowMixed extmap_allow_mixed_enum_ = kMedia;
SimulcastDescription simulcast_;
std::vector<RidDescription> receive_rids_;
private:
// Copy function that returns a raw pointer. Caller will assert ownership.
// Should only be called by the Clone() function. Must be implemented
// by each final subclass.
virtual MediaContentDescription* CloneInternal() const = 0;
};
template <class C>
class MediaContentDescriptionImpl : public MediaContentDescription {
public:
void set_protocol(const std::string& protocol) override {
RTC_DCHECK(IsRtpProtocol(protocol));
protocol_ = protocol;
}
typedef C CodecType;
// Codecs should be in preference order (most preferred codec first).
virtual const std::vector<C>& codecs() const { return codecs_; }
virtual void set_codecs(const std::vector<C>& codecs) { codecs_ = codecs; }
bool has_codecs() const override { return !codecs_.empty(); }
virtual 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;
}
virtual void AddCodec(const C& codec) { codecs_.push_back(codec); }
virtual 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);
}
virtual 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() {}
virtual MediaType type() const { return MEDIA_TYPE_AUDIO; }
virtual AudioContentDescription* as_audio() { return this; }
virtual const AudioContentDescription* as_audio() const { return this; }
private:
virtual AudioContentDescription* CloneInternal() const {
return new AudioContentDescription(*this);
}
};
class VideoContentDescription : public MediaContentDescriptionImpl<VideoCodec> {
public:
virtual MediaType type() const { return MEDIA_TYPE_VIDEO; }
virtual VideoContentDescription* as_video() { return this; }
virtual const VideoContentDescription* as_video() const { return this; }
private:
virtual VideoContentDescription* CloneInternal() const {
return new VideoContentDescription(*this);
}
};
class SctpDataContentDescription : public MediaContentDescription {
public:
SctpDataContentDescription() {}
SctpDataContentDescription(const SctpDataContentDescription& o)
: MediaContentDescription(o),
use_sctpmap_(o.use_sctpmap_),
port_(o.port_),
max_message_size_(o.max_message_size_) {}
MediaType type() const override { return MEDIA_TYPE_DATA; }
SctpDataContentDescription* as_sctp() override { return this; }
const SctpDataContentDescription* as_sctp() const override { return this; }
bool has_codecs() const override { return false; }
void set_protocol(const std::string& protocol) override {
RTC_DCHECK(IsSctpProtocol(protocol));
protocol_ = protocol;
}
bool use_sctpmap() const { return use_sctpmap_; }
void set_use_sctpmap(bool enable) { use_sctpmap_ = enable; }
int port() const { return port_; }
void set_port(int port) { port_ = port; }
int max_message_size() const { return max_message_size_; }
void set_max_message_size(int max_message_size) {
max_message_size_ = max_message_size;
}
private:
SctpDataContentDescription* CloneInternal() const override {
return new SctpDataContentDescription(*this);
}
bool use_sctpmap_ = true; // Note: "true" is no longer conformant.
// Defaults should be constants imported from SCTP. Quick hack.
int port_ = 5000;
// draft-ietf-mmusic-sdp-sctp-23: Max message size default is 64K
int max_message_size_ = 64 * 1024;
};
class UnsupportedContentDescription : public MediaContentDescription {
public:
explicit UnsupportedContentDescription(const std::string& media_type)
: media_type_(media_type) {}
MediaType type() const override { return MEDIA_TYPE_UNSUPPORTED; }
UnsupportedContentDescription* as_unsupported() override { return this; }
const UnsupportedContentDescription* as_unsupported() const override {
return this;
}
bool has_codecs() const override { return false; }
const std::string& media_type() const { return media_type_; }
private:
UnsupportedContentDescription* CloneInternal() const override {
return new UnsupportedContentDescription(*this);
}
std::string media_type_;
};
// Protocol used for encoding media. This is the "top level" protocol that may
// be wrapped by zero or many transport protocols (UDP, ICE, etc.).
enum class MediaProtocolType {
kRtp, // Section will use the RTP protocol (e.g., for audio or video).
// https://tools.ietf.org/html/rfc3550
kSctp, // Section will use the SCTP protocol (e.g., for a data channel).
// https://tools.ietf.org/html/rfc4960
kOther // Section will use another top protocol which is not
// explicitly supported.
};
// Represents a session description section. Most information about the section
// is stored in the description, which is a subclass of MediaContentDescription.
// Owns the description.
class RTC_EXPORT ContentInfo {
public:
explicit ContentInfo(MediaProtocolType type) : type(type) {}
~ContentInfo();
// Copy
ContentInfo(const ContentInfo& o);
ContentInfo& operator=(const ContentInfo& o);
ContentInfo(ContentInfo&& o) = default;
ContentInfo& operator=(ContentInfo&& o) = default;
// Alias for |name|.
std::string mid() const { return name; }
void set_mid(const std::string& mid) { this->name = mid; }
// Alias for |description|.
MediaContentDescription* media_description();
const MediaContentDescription* media_description() const;
void set_media_description(std::unique_ptr<MediaContentDescription> desc) {
description_ = std::move(desc);
}
// TODO(bugs.webrtc.org/8620): Rename this to mid.
std::string name;
MediaProtocolType type;
bool rejected = false;
bool bundle_only = false;
private:
friend class SessionDescription;
std::unique_ptr<MediaContentDescription> description_;
};
typedef std::vector<std::string> ContentNames;
// This class provides a mechanism to aggregate different media contents into a
// group. This group can also be shared with the peers in a pre-defined format.
// GroupInfo should be populated only with the |content_name| of the
// MediaDescription.
class ContentGroup {
public:
explicit ContentGroup(const std::string& semantics);
ContentGroup(const ContentGroup&);
ContentGroup(ContentGroup&&);
ContentGroup& operator=(const ContentGroup&);
ContentGroup& operator=(ContentGroup&&);
~ContentGroup();
const std::string& semantics() const { return semantics_; }
const ContentNames& content_names() const { return content_names_; }
const std::string* FirstContentName() const;
bool HasContentName(const std::string& content_name) const;
void AddContentName(const std::string& content_name);
bool RemoveContentName(const std::string& content_name);
private:
std::string semantics_;
ContentNames content_names_;
};
typedef std::vector<ContentInfo> ContentInfos;
typedef std::vector<ContentGroup> ContentGroups;
const ContentInfo* FindContentInfoByName(const ContentInfos& contents,
const std::string& name);
const ContentInfo* FindContentInfoByType(const ContentInfos& contents,
const std::string& type);
// Determines how the MSID will be signaled in the SDP. These can be used as
// flags to indicate both or none.
enum MsidSignaling {
// Signal MSID with one a=msid line in the media section.
kMsidSignalingMediaSection = 0x1,
// Signal MSID with a=ssrc: msid lines in the media section.
kMsidSignalingSsrcAttribute = 0x2
};
// Describes a collection of contents, each with its own name and
// type. Analogous to a <jingle> or <session> stanza. Assumes that
// contents are unique be name, but doesn't enforce that.
class SessionDescription {
public:
SessionDescription();
~SessionDescription();
std::unique_ptr<SessionDescription> Clone() const;
// Content accessors.
const ContentInfos& contents() const { return contents_; }
ContentInfos& contents() { return contents_; }
const ContentInfo* GetContentByName(const std::string& name) const;
ContentInfo* GetContentByName(const std::string& name);
const MediaContentDescription* GetContentDescriptionByName(
const std::string& name) const;
MediaContentDescription* GetContentDescriptionByName(const std::string& name);
const ContentInfo* FirstContentByType(MediaProtocolType type) const;
const ContentInfo* FirstContent() const;
// Content mutators.
// Adds a content to this description. Takes ownership of ContentDescription*.
void AddContent(const std::string& name,
MediaProtocolType type,
std::unique_ptr<MediaContentDescription> description);
void AddContent(const std::string& name,
MediaProtocolType type,
bool rejected,
std::unique_ptr<MediaContentDescription> description);
void AddContent(const std::string& name,
MediaProtocolType type,
bool rejected,
bool bundle_only,
std::unique_ptr<MediaContentDescription> description);
void AddContent(ContentInfo&& content);
bool RemoveContentByName(const std::string& name);
// Transport accessors.
const TransportInfos& transport_infos() const { return transport_infos_; }
TransportInfos& transport_infos() { return transport_infos_; }
const TransportInfo* GetTransportInfoByName(const std::string& name) const;
TransportInfo* GetTransportInfoByName(const std::string& name);
const TransportDescription* GetTransportDescriptionByName(
const std::string& name) const {
const TransportInfo* tinfo = GetTransportInfoByName(name);
return tinfo ? &tinfo->description : NULL;
}
// Transport mutators.
void set_transport_infos(const TransportInfos& transport_infos) {
transport_infos_ = transport_infos;
}
// Adds a TransportInfo to this description.
void AddTransportInfo(const TransportInfo& transport_info);
bool RemoveTransportInfoByName(const std::string& name);
// Group accessors.
const ContentGroups& groups() const { return content_groups_; }
const ContentGroup* GetGroupByName(const std::string& name) const;
std::vector<const ContentGroup*> GetGroupsByName(
const std::string& name) const;
bool HasGroup(const std::string& name) const;
// Group mutators.
void AddGroup(const ContentGroup& group) { content_groups_.push_back(group); }
// Remove the first group with the same semantics specified by |name|.
void RemoveGroupByName(const std::string& name);
// Global attributes.
void set_msid_supported(bool supported) { msid_supported_ = supported; }
bool msid_supported() const { return msid_supported_; }
// Determines how the MSIDs were/will be signaled. Flag value composed of
// MsidSignaling bits (see enum above).
void set_msid_signaling(int msid_signaling) {
msid_signaling_ = msid_signaling;
}
int msid_signaling() const { return msid_signaling_; }
// Determines if it's allowed to mix one- and two-byte rtp header extensions
// within the same rtp stream.
void set_extmap_allow_mixed(bool supported) {
extmap_allow_mixed_ = supported;
MediaContentDescription::ExtmapAllowMixed media_level_setting =
supported ? MediaContentDescription::kSession
: MediaContentDescription::kNo;
for (auto& content : contents_) {
// Do not set to kNo if the current setting is kMedia.
if (supported || content.media_description()->extmap_allow_mixed_enum() !=
MediaContentDescription::kMedia) {
content.media_description()->set_extmap_allow_mixed_enum(
media_level_setting);
}
}
}
bool extmap_allow_mixed() const { return extmap_allow_mixed_; }
private:
SessionDescription(const SessionDescription&);
ContentInfos contents_;
TransportInfos transport_infos_;
ContentGroups content_groups_;
bool msid_supported_ = true;
// Default to what Plan B would do.
// TODO(bugs.webrtc.org/8530): Change default to kMsidSignalingMediaSection.
int msid_signaling_ = kMsidSignalingSsrcAttribute;
bool extmap_allow_mixed_ = true;
};
// Indicates whether a session description was sent by the local client or
// received from the remote client.
enum ContentSource { CS_LOCAL, CS_REMOTE };
} // namespace cricket
#endif // PC_SESSION_DESCRIPTION_H_