Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 400 insertions, 0 deletions

diff --git a/third_party/libwebrtc/pc/media_session.h b/third_party/libwebrtc/pc/media_session.h
new file mode 100644
index 0000000000..3100fb6fdb
--- /dev/null
+++ b/third_party/libwebrtc/pc/media_session.h
@@ -0,0 +1,400 @@
+/*
+ *  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/field_trials_view.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::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::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::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::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_