Repo created

TMessagesProj/jni/voip/webrtc/pc/session_description.h

@@ -0,0 +1,593 @@
+/*
+ *  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 <memory>
+#include <string>
+#include <type_traits>
+#include <utility>
+#include <vector>
+
+#include "absl/memory/memory.h"
+#include "absl/strings/string_view.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 {
+
+using RtpHeaderExtensions = std::vector<webrtc::RtpExtension>;
+
+// 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;
+  }
+
+  // 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.
+  std::string protocol() const { return protocol_; }
+  virtual void set_protocol(absl::string_view protocol) {
+    protocol_ = std::string(protocol);
+  }
+
+  webrtc::RtpTransceiverDirection direction() const { return direction_; }
+  void set_direction(webrtc::RtpTransceiverDirection direction) {
+    direction_ = direction;
+  }
+
+  bool rtcp_mux() const { return rtcp_mux_; }
+  void set_rtcp_mux(bool mux) { rtcp_mux_ = mux; }
+
+  bool rtcp_reduced_size() const { return rtcp_reduced_size_; }
+  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.
+  bool remote_estimate() const { return remote_estimate_; }
+  void set_remote_estimate(bool remote_estimate) {
+    remote_estimate_ = remote_estimate;
+  }
+
+  int bandwidth() const { return bandwidth_; }
+  void set_bandwidth(int bandwidth) { bandwidth_ = bandwidth; }
+  std::string bandwidth_type() const { return bandwidth_type_; }
+  void set_bandwidth_type(std::string bandwidth_type) {
+    bandwidth_type_ = bandwidth_type;
+  }
+
+  // List of RTP header extensions. URIs are **NOT** guaranteed to be unique
+  // as they can appear twice when both encrypted and non-encrypted extensions
+  // are present.
+  // Use RtpExtension::FindHeaderExtensionByUri for finding and
+  // RtpExtension::DeduplicateHeaderExtensions for filtering.
+  const RtpHeaderExtensions& rtp_header_extensions() const {
+    return rtp_header_extensions_;
+  }
+  void set_rtp_header_extensions(const RtpHeaderExtensions& extensions) {
+    rtp_header_extensions_ = extensions;
+    rtp_header_extensions_set_ = true;
+  }
+  void AddRtpHeaderExtension(const webrtc::RtpExtension& ext) {
+    rtp_header_extensions_.push_back(ext);
+    rtp_header_extensions_set_ = true;
+  }
+  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).
+  bool rtp_header_extensions_set() const { return rtp_header_extensions_set_; }
+  const StreamParamsVec& streams() const { return send_streams_; }
+  // TODO(pthatcher): Remove this by giving mediamessage.cc access
+  // to MediaContentDescription
+  StreamParamsVec& mutable_streams() { return send_streams_; }
+  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);
+  }
+
+  uint32_t first_ssrc() const {
+    if (send_streams_.empty()) {
+      return 0;
+    }
+    return send_streams_[0].first_ssrc();
+  }
+  bool has_ssrcs() const {
+    if (send_streams_.empty()) {
+      return false;
+    }
+    return send_streams_[0].has_ssrcs();
+  }
+
+  void set_conference_mode(bool enable) { conference_mode_ = enable; }
+  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.
+  void set_connection_address(const rtc::SocketAddress& address) {
+    connection_address_ = address;
+  }
+  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 };
+  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;
+  }
+  ExtmapAllowMixed extmap_allow_mixed_enum() const {
+    return extmap_allow_mixed_enum_;
+  }
+  bool extmap_allow_mixed() const { return extmap_allow_mixed_enum_ != kNo; }
+
+  // Simulcast functionality.
+  bool HasSimulcast() const { return !simulcast_.empty(); }
+  SimulcastDescription& simulcast_description() { return simulcast_; }
+  const SimulcastDescription& simulcast_description() const {
+    return simulcast_;
+  }
+  void set_simulcast_description(const SimulcastDescription& simulcast) {
+    simulcast_ = simulcast;
+  }
+  const std::vector<RidDescription>& receive_rids() const {
+    return receive_rids_;
+  }
+  void set_receive_rids(const std::vector<RidDescription>& rids) {
+    receive_rids_ = rids;
+  }
+
+  // Codecs should be in preference order (most preferred codec first).
+  const std::vector<Codec>& codecs() const { return codecs_; }
+  void set_codecs(const std::vector<Codec>& codecs) { codecs_ = codecs; }
+  virtual bool has_codecs() const { return !codecs_.empty(); }
+  bool HasCodec(int id) {
+    return absl::c_find_if(codecs_, [id](const cricket::Codec codec) {
+             return codec.id == id;
+           }) != codecs_.end();
+  }
+  void AddCodec(const Codec& codec) { codecs_.push_back(codec); }
+  void AddOrReplaceCodec(const Codec& codec) {
+    for (auto it = codecs_.begin(); it != codecs_.end(); ++it) {
+      if (it->id == codec.id) {
+        *it = codec;
+        return;
+      }
+    }
+    AddCodec(codec);
+  }
+  void AddCodecs(const std::vector<Codec>& codecs) {
+    for (const auto& codec : codecs) {
+      AddCodec(codec);
+    }
+  }
+
+ protected:
+  // TODO(bugs.webrtc.org/15214): move all RTP related things to
+  // RtpMediaDescription that the SCTP content description does
+  // not inherit from.
+  std::string protocol_;
+
+ private:
+  bool rtcp_mux_ = false;
+  bool rtcp_reduced_size_ = false;
+  bool remote_estimate_ = false;
+  int bandwidth_ = kAutoBandwidth;
+  std::string bandwidth_type_ = kApplicationSpecificBandwidth;
+
+  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_;
+
+  // 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;
+
+  std::vector<Codec> codecs_;
+};
+
+class RtpMediaContentDescription : public MediaContentDescription {};
+
+class AudioContentDescription : public RtpMediaContentDescription {
+ public:
+  void set_protocol(absl::string_view protocol) override {
+    RTC_DCHECK(IsRtpProtocol(protocol));
+    protocol_ = std::string(protocol);
+  }
+  MediaType type() const override { return MEDIA_TYPE_AUDIO; }
+  AudioContentDescription* as_audio() override { return this; }
+  const AudioContentDescription* as_audio() const override { return this; }
+
+ private:
+  AudioContentDescription* CloneInternal() const override {
+    return new AudioContentDescription(*this);
+  }
+};
+
+class VideoContentDescription : public RtpMediaContentDescription {
+ public:
+  void set_protocol(absl::string_view protocol) override {
+    RTC_DCHECK(IsRtpProtocol(protocol));
+    protocol_ = std::string(protocol);
+  }
+  MediaType type() const override { return MEDIA_TYPE_VIDEO; }
+  VideoContentDescription* as_video() override { return this; }
+  const VideoContentDescription* as_video() const override { return this; }
+
+ private:
+  VideoContentDescription* CloneInternal() const override {
+    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(absl::string_view protocol) override {
+    RTC_DCHECK(IsSctpProtocol(protocol));
+    protocol_ = std::string(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(absl::string_view 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(absl::string_view content_name) const;
+  void AddContentName(absl::string_view content_name);
+  bool RemoveContentName(absl::string_view content_name);
+  // for debugging
+  std::string ToString() const;
+
+ 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 bit flags to indicate both or the special value none.
+enum MsidSignaling {
+  // MSID is not signaled. This is not a bit flag and must be compared for
+  // equality.
+  kMsidSignalingNotUsed = 0x0,
+  // Signal MSID with at least one a=msid line in the media section.
+  // This requires unified plan.
+  kMsidSignalingMediaSection = 0x1,
+  // Signal MSID with a=ssrc: msid lines in the media section.
+  // This should only be used with plan-b but is signalled in
+  // offers for backward compability reasons.
+  kMsidSignalingSsrcAttribute = 0x2,
+  // Signal MSID with a=msid-semantic: WMS in the session section.
+  // This is deprecated but signalled for backward compability reasons.
+  // It is typically combined with 0x1 or 0x2.
+  kMsidSignalingSemantic = 0x4
+};
+
+// 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.
+  // 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_;
+  int msid_signaling_ = kMsidSignalingMediaSection | kMsidSignalingSemantic;
+  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_