/*
 *  Copyright 2012 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.
 */

// This file contains the PeerConnection interface as defined in
// https://w3c.github.io/webrtc-pc/#peer-to-peer-connections
//
// The PeerConnectionFactory class provides factory methods to create
// PeerConnection, MediaStream and MediaStreamTrack objects.
//
// The following steps are needed to setup a typical call using WebRTC:
//
// 1. Create a PeerConnectionFactoryInterface. Check constructors for more
// information about input parameters.
//
// 2. Create a PeerConnection object. Provide a configuration struct which
// points to STUN and/or TURN servers used to generate ICE candidates, and
// provide an object that implements the PeerConnectionObserver interface,
// which is used to receive callbacks from the PeerConnection.
//
// 3. Create local MediaStreamTracks using the PeerConnectionFactory and add
// them to PeerConnection by calling AddTrack (or legacy method, AddStream).
//
// 4. Create an offer, call SetLocalDescription with it, serialize it, and send
// it to the remote peer
//
// 5. Once an ICE candidate has been gathered, the PeerConnection will call the
// observer function OnIceCandidate. The candidates must also be serialized and
// sent to the remote peer.
//
// 6. Once an answer is received from the remote peer, call
// SetRemoteDescription with the remote answer.
//
// 7. Once a remote candidate is received from the remote peer, provide it to
// the PeerConnection by calling AddIceCandidate.
//
// The receiver of a call (assuming the application is "call"-based) can decide
// to accept or reject the call; this decision will be taken by the application,
// not the PeerConnection.
//
// If the application decides to accept the call, it should:
//
// 1. Create PeerConnectionFactoryInterface if it doesn't exist.
//
// 2. Create a new PeerConnection.
//
// 3. Provide the remote offer to the new PeerConnection object by calling
// SetRemoteDescription.
//
// 4. Generate an answer to the remote offer by calling CreateAnswer and send it
// back to the remote peer.
//
// 5. Provide the local answer to the new PeerConnection by calling
// SetLocalDescription with the answer.
//
// 6. Provide the remote ICE candidates by calling AddIceCandidate.
//
// 7. Once a candidate has been gathered, the PeerConnection will call the
// observer function OnIceCandidate. Send these candidates to the remote peer.

#ifndef API_PEER_CONNECTION_INTERFACE_H_
#define API_PEER_CONNECTION_INTERFACE_H_

#include <stdio.h>

#include <memory>
#include <string>
#include <vector>

#include "api/async_resolver_factory.h"
#include "api/audio/audio_mixer.h"
#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/audio_codecs/audio_encoder_factory.h"
#include "api/audio_options.h"
#include "api/call/call_factory_interface.h"
#include "api/crypto/crypto_options.h"
#include "api/data_channel_interface.h"
#include "api/dtls_transport_interface.h"
#include "api/fec_controller.h"
#include "api/jsep.h"
#include "api/media_stream_interface.h"
#include "api/network_state_predictor.h"
#include "api/packet_socket_factory.h"
#include "api/rtc_error.h"
#include "api/rtc_event_log/rtc_event_log_factory_interface.h"
#include "api/rtc_event_log_output.h"
#include "api/rtp_receiver_interface.h"
#include "api/rtp_sender_interface.h"
#include "api/rtp_transceiver_interface.h"
#include "api/sctp_transport_interface.h"
#include "api/set_remote_description_observer_interface.h"
#include "api/stats/rtc_stats_collector_callback.h"
#include "api/stats_types.h"
#include "api/task_queue/task_queue_factory.h"
#include "api/transport/bitrate_settings.h"
#include "api/transport/media/media_transport_interface.h"
#include "api/transport/network_control.h"
#include "api/turn_customizer.h"
#include "media/base/media_config.h"
#include "media/base/media_engine.h"
// TODO(bugs.webrtc.org/7447): We plan to provide a way to let applications
// inject a PacketSocketFactory and/or NetworkManager, and not expose
// PortAllocator in the PeerConnection api.
#include "p2p/base/port_allocator.h"  // nogncheck
#include "rtc_base/network.h"
#include "rtc_base/rtc_certificate.h"
#include "rtc_base/rtc_certificate_generator.h"
#include "rtc_base/socket_address.h"
#include "rtc_base/ssl_certificate.h"
#include "rtc_base/ssl_stream_adapter.h"
#include "rtc_base/system/rtc_export.h"

namespace rtc {
class Thread;
}  // namespace rtc

namespace webrtc {

// MediaStream container interface.
class StreamCollectionInterface : public rtc::RefCountInterface {
 public:
  // TODO(ronghuawu): Update the function names to c++ style, e.g. find -> Find.
  virtual size_t count() = 0;
  virtual MediaStreamInterface* at(size_t index) = 0;
  virtual MediaStreamInterface* find(const std::string& label) = 0;
  virtual MediaStreamTrackInterface* FindAudioTrack(const std::string& id) = 0;
  virtual MediaStreamTrackInterface* FindVideoTrack(const std::string& id) = 0;

 protected:
  // Dtor protected as objects shouldn't be deleted via this interface.
  ~StreamCollectionInterface() override = default;
};

class StatsObserver : public rtc::RefCountInterface {
 public:
  virtual void OnComplete(const StatsReports& reports) = 0;

 protected:
  ~StatsObserver() override = default;
};

enum class SdpSemantics { kPlanB, kUnifiedPlan };

class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
 public:
  // See https://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate
  enum SignalingState {
    kStable,
    kHaveLocalOffer,
    kHaveLocalPrAnswer,
    kHaveRemoteOffer,
    kHaveRemotePrAnswer,
    kClosed,
  };

  // See https://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate
  enum IceGatheringState {
    kIceGatheringNew,
    kIceGatheringGathering,
    kIceGatheringComplete
  };

  // See https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate
  enum class PeerConnectionState {
    kNew,
    kConnecting,
    kConnected,
    kDisconnected,
    kFailed,
    kClosed,
  };

  // See https://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate
  enum IceConnectionState {
    kIceConnectionNew,
    kIceConnectionChecking,
    kIceConnectionConnected,
    kIceConnectionCompleted,
    kIceConnectionFailed,
    kIceConnectionDisconnected,
    kIceConnectionClosed,
    kIceConnectionMax,
  };

  // TLS certificate policy.
  enum TlsCertPolicy {
    // For TLS based protocols, ensure the connection is secure by not
    // circumventing certificate validation.
    kTlsCertPolicySecure,
    // For TLS based protocols, disregard security completely by skipping
    // certificate validation. This is insecure and should never be used unless
    // security is irrelevant in that particular context.
    kTlsCertPolicyInsecureNoCheck,
  };

  struct IceServer {
    IceServer();
    IceServer(const IceServer&);
    ~IceServer();

    // TODO(jbauch): Remove uri when all code using it has switched to urls.
    // List of URIs associated with this server. Valid formats are described
    // in RFC7064 and RFC7065, and more may be added in the future. The "host"
    // part of the URI may contain either an IP address or a hostname.
    std::string uri;
    std::vector<std::string> urls;
    std::string username;
    std::string password;
    TlsCertPolicy tls_cert_policy = kTlsCertPolicySecure;
    // If the URIs in |urls| only contain IP addresses, this field can be used
    // to indicate the hostname, which may be necessary for TLS (using the SNI
    // extension). If |urls| itself contains the hostname, this isn't
    // necessary.
    std::string hostname;
    // List of protocols to be used in the TLS ALPN extension.
    std::vector<std::string> tls_alpn_protocols;
    // List of elliptic curves to be used in the TLS elliptic curves extension.
    std::vector<std::string> tls_elliptic_curves;

    bool operator==(const IceServer& o) const {
      return uri == o.uri && urls == o.urls && username == o.username &&
             password == o.password && tls_cert_policy == o.tls_cert_policy &&
             hostname == o.hostname &&
             tls_alpn_protocols == o.tls_alpn_protocols &&
             tls_elliptic_curves == o.tls_elliptic_curves;
    }
    bool operator!=(const IceServer& o) const { return !(*this == o); }
  };
  typedef std::vector<IceServer> IceServers;

  enum IceTransportsType {
    // TODO(pthatcher): Rename these kTransporTypeXXX, but update
    // Chromium at the same time.
    kNone,
    kRelay,
    kNoHost,
    kAll
  };

  // https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1
  enum BundlePolicy {
    kBundlePolicyBalanced,
    kBundlePolicyMaxBundle,
    kBundlePolicyMaxCompat
  };

  // https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1
  enum RtcpMuxPolicy {
    kRtcpMuxPolicyNegotiate,
    kRtcpMuxPolicyRequire,
  };

  enum TcpCandidatePolicy {
    kTcpCandidatePolicyEnabled,
    kTcpCandidatePolicyDisabled
  };

  enum CandidateNetworkPolicy {
    kCandidateNetworkPolicyAll,
    kCandidateNetworkPolicyLowCost
  };

  enum ContinualGatheringPolicy { GATHER_ONCE, GATHER_CONTINUALLY };

  enum class RTCConfigurationType {
    // A configuration that is safer to use, despite not having the best
    // performance. Currently this is the default configuration.
    kSafe,
    // An aggressive configuration that has better performance, although it
    // may be riskier and may need extra support in the application.
    kAggressive
  };

  // TODO(hbos): Change into class with private data and public getters.
  // TODO(nisse): In particular, accessing fields directly from an
  // application is brittle, since the organization mirrors the
  // organization of the implementation, which isn't stable. So we
  // need getters and setters at least for fields which applications
  // are interested in.
  struct RTC_EXPORT RTCConfiguration {
    // This struct is subject to reorganization, both for naming
    // consistency, and to group settings to match where they are used
    // in the implementation. To do that, we need getter and setter
    // methods for all settings which are of interest to applications,
    // Chrome in particular.

    RTCConfiguration();
    RTCConfiguration(const RTCConfiguration&);
    explicit RTCConfiguration(RTCConfigurationType type);
    ~RTCConfiguration();

    bool operator==(const RTCConfiguration& o) const;
    bool operator!=(const RTCConfiguration& o) const;

    bool dscp() const { return media_config.enable_dscp; }
    void set_dscp(bool enable) { media_config.enable_dscp = enable; }

    bool cpu_adaptation() const {
      return media_config.video.enable_cpu_adaptation;
    }
    void set_cpu_adaptation(bool enable) {
      media_config.video.enable_cpu_adaptation = enable;
    }

    bool suspend_below_min_bitrate() const {
      return media_config.video.suspend_below_min_bitrate;
    }
    void set_suspend_below_min_bitrate(bool enable) {
      media_config.video.suspend_below_min_bitrate = enable;
    }

    bool prerenderer_smoothing() const {
      return media_config.video.enable_prerenderer_smoothing;
    }
    void set_prerenderer_smoothing(bool enable) {
      media_config.video.enable_prerenderer_smoothing = enable;
    }

    bool experiment_cpu_load_estimator() const {
      return media_config.video.experiment_cpu_load_estimator;
    }
    void set_experiment_cpu_load_estimator(bool enable) {
      media_config.video.experiment_cpu_load_estimator = enable;
    }

    int audio_rtcp_report_interval_ms() const {
      return media_config.audio.rtcp_report_interval_ms;
    }
    void set_audio_rtcp_report_interval_ms(int audio_rtcp_report_interval_ms) {
      media_config.audio.rtcp_report_interval_ms =
          audio_rtcp_report_interval_ms;
    }

    int video_rtcp_report_interval_ms() const {
      return media_config.video.rtcp_report_interval_ms;
    }
    void set_video_rtcp_report_interval_ms(int video_rtcp_report_interval_ms) {
      media_config.video.rtcp_report_interval_ms =
          video_rtcp_report_interval_ms;
    }

    static const int kUndefined = -1;
    // Default maximum number of packets in the audio jitter buffer.
    static const int kAudioJitterBufferMaxPackets = 200;
    // ICE connection receiving timeout for aggressive configuration.
    static const int kAggressiveIceConnectionReceivingTimeout = 1000;

    ////////////////////////////////////////////////////////////////////////
    // The below few fields mirror the standard RTCConfiguration dictionary:
    // https://w3c.github.io/webrtc-pc/#rtcconfiguration-dictionary
    ////////////////////////////////////////////////////////////////////////

    // TODO(pthatcher): Rename this ice_servers, but update Chromium
    // at the same time.
    IceServers servers;
    // TODO(pthatcher): Rename this ice_transport_type, but update
    // Chromium at the same time.
    IceTransportsType type = kAll;
    BundlePolicy bundle_policy = kBundlePolicyBalanced;
    RtcpMuxPolicy rtcp_mux_policy = kRtcpMuxPolicyRequire;
    std::vector<rtc::scoped_refptr<rtc::RTCCertificate>> certificates;
    int ice_candidate_pool_size = 0;

    //////////////////////////////////////////////////////////////////////////
    // The below fields correspond to constraints from the deprecated
    // constraints interface for constructing a PeerConnection.
    //
    // absl::optional fields can be "missing", in which case the implementation
    // default will be used.
    //////////////////////////////////////////////////////////////////////////

    // If set to true, don't gather IPv6 ICE candidates.
    // TODO(deadbeef): Remove this? IPv6 support has long stopped being
    // experimental
    bool disable_ipv6 = false;

    // If set to true, don't gather IPv6 ICE candidates on Wi-Fi.
    // Only intended to be used on specific devices. Certain phones disable IPv6
    // when the screen is turned off and it would be better to just disable the
    // IPv6 ICE candidates on Wi-Fi in those cases.
    bool disable_ipv6_on_wifi = false;

    // By default, the PeerConnection will use a limited number of IPv6 network
    // interfaces, in order to avoid too many ICE candidate pairs being created
    // and delaying ICE completion.
    //
    // Can be set to INT_MAX to effectively disable the limit.
    int max_ipv6_networks = cricket::kDefaultMaxIPv6Networks;

    // Exclude link-local network interfaces
    // from considertaion for gathering ICE candidates.
    bool disable_link_local_networks = false;

    // If set to true, use RTP data channels instead of SCTP.
    // TODO(deadbeef): Remove this. We no longer commit to supporting RTP data
    // channels, though some applications are still working on moving off of
    // them.
    bool enable_rtp_data_channel = false;

    // Minimum bitrate at which screencast video tracks will be encoded at.
    // This means adding padding bits up to this bitrate, which can help
    // when switching from a static scene to one with motion.
    absl::optional<int> screencast_min_bitrate;

    // Use new combined audio/video bandwidth estimation?
    absl::optional<bool> combined_audio_video_bwe;

    // TODO(bugs.webrtc.org/9891) - Move to crypto_options
    // Can be used to disable DTLS-SRTP. This should never be done, but can be
    // useful for testing purposes, for example in setting up a loopback call
    // with a single PeerConnection.
    absl::optional<bool> enable_dtls_srtp;

    /////////////////////////////////////////////////
    // The below fields are not part of the standard.
    /////////////////////////////////////////////////

    // Can be used to disable TCP candidate generation.
    TcpCandidatePolicy tcp_candidate_policy = kTcpCandidatePolicyEnabled;

    // Can be used to avoid gathering candidates for a "higher cost" network,
    // if a lower cost one exists. For example, if both Wi-Fi and cellular
    // interfaces are available, this could be used to avoid using the cellular
    // interface.
    CandidateNetworkPolicy candidate_network_policy =
        kCandidateNetworkPolicyAll;

    // The maximum number of packets that can be stored in the NetEq audio
    // jitter buffer. Can be reduced to lower tolerated audio latency.
    int audio_jitter_buffer_max_packets = kAudioJitterBufferMaxPackets;

    // Whether to use the NetEq "fast mode" which will accelerate audio quicker
    // if it falls behind.
    bool audio_jitter_buffer_fast_accelerate = false;

    // The minimum delay in milliseconds for the audio jitter buffer.
    int audio_jitter_buffer_min_delay_ms = 0;

    // Whether the audio jitter buffer adapts the delay to retransmitted
    // packets.
    bool audio_jitter_buffer_enable_rtx_handling = false;

    // Timeout in milliseconds before an ICE candidate pair is considered to be
    // "not receiving", after which a lower priority candidate pair may be
    // selected.
    int ice_connection_receiving_timeout = kUndefined;

    // Interval in milliseconds at which an ICE "backup" candidate pair will be
    // pinged. This is a candidate pair which is not actively in use, but may
    // be switched to if the active candidate pair becomes unusable.
    //
    // This is relevant mainly to Wi-Fi/cell handoff; the application may not
    // want this backup cellular candidate pair pinged frequently, since it
    // consumes data/battery.
    int ice_backup_candidate_pair_ping_interval = kUndefined;

    // Can be used to enable continual gathering, which means new candidates
    // will be gathered as network interfaces change. Note that if continual
    // gathering is used, the candidate removal API should also be used, to
    // avoid an ever-growing list of candidates.
    ContinualGatheringPolicy continual_gathering_policy = GATHER_ONCE;

    // If set to true, candidate pairs will be pinged in order of most likely
    // to work (which means using a TURN server, generally), rather than in
    // standard priority order.
    bool prioritize_most_likely_ice_candidate_pairs = false;

    // Implementation defined settings. A public member only for the benefit of
    // the implementation. Applications must not access it directly, and should
    // instead use provided accessor methods, e.g., set_cpu_adaptation.
    struct cricket::MediaConfig media_config;

    // If set to true, only one preferred TURN allocation will be used per
    // network interface. UDP is preferred over TCP and IPv6 over IPv4. This
    // can be used to cut down on the number of candidate pairings.
    bool prune_turn_ports = false;

    // If set to true, this means the ICE transport should presume TURN-to-TURN
    // candidate pairs will succeed, even before a binding response is received.
    // This can be used to optimize the initial connection time, since the DTLS
    // handshake can begin immediately.
    bool presume_writable_when_fully_relayed = false;

    // If true, "renomination" will be added to the ice options in the transport
    // description.
    // See: https://tools.ietf.org/html/draft-thatcher-ice-renomination-00
    bool enable_ice_renomination = false;

    // If true, the ICE role is re-determined when the PeerConnection sets a
    // local transport description that indicates an ICE restart.
    //
    // This is standard RFC5245 ICE behavior, but causes unnecessary role
    // thrashing, so an application may wish to avoid it. This role
    // re-determining was removed in ICEbis (ICE v2).
    bool redetermine_role_on_ice_restart = true;

    // This flag is only effective when |continual_gathering_policy| is
    // GATHER_CONTINUALLY.
    //
    // If true, after the ICE transport type is changed such that new types of
    // ICE candidates are allowed by the new transport type, e.g. from
    // IceTransportsType::kRelay to IceTransportsType::kAll, candidates that
    // have been gathered by the ICE transport but not matching the previous
    // transport type and as a result not observed by PeerConnectionObserver,
    // will be surfaced to the observer.
    bool surface_ice_candidates_on_ice_transport_type_changed = false;

    // The following fields define intervals in milliseconds at which ICE
    // connectivity checks are sent.
    //
    // We consider ICE is "strongly connected" for an agent when there is at
    // least one candidate pair that currently succeeds in connectivity check
    // from its direction i.e. sending a STUN ping and receives a STUN ping
    // response, AND all candidate pairs have sent a minimum number of pings for
    // connectivity (this number is implementation-specific). Otherwise, ICE is
    // considered in "weak connectivity".
    //
    // Note that the above notion of strong and weak connectivity is not defined
    // in RFC 5245, and they apply to our current ICE implementation only.
    //
    // 1) ice_check_interval_strong_connectivity defines the interval applied to
    // ALL candidate pairs when ICE is strongly connected, and it overrides the
    // default value of this interval in the ICE implementation;
    // 2) ice_check_interval_weak_connectivity defines the counterpart for ALL
    // pairs when ICE is weakly connected, and it overrides the default value of
    // this interval in the ICE implementation;
    // 3) ice_check_min_interval defines the minimal interval (equivalently the
    // maximum rate) that overrides the above two intervals when either of them
    // is less.
    absl::optional<int> ice_check_interval_strong_connectivity;
    absl::optional<int> ice_check_interval_weak_connectivity;
    absl::optional<int> ice_check_min_interval;

    // The min time period for which a candidate pair must wait for response to
    // connectivity checks before it becomes unwritable. This parameter
    // overrides the default value in the ICE implementation if set.
    absl::optional<int> ice_unwritable_timeout;

    // The min number of connectivity checks that a candidate pair must sent
    // without receiving response before it becomes unwritable. This parameter
    // overrides the default value in the ICE implementation if set.
    absl::optional<int> ice_unwritable_min_checks;

    // The min time period for which a candidate pair must wait for response to
    // connectivity checks it becomes inactive. This parameter overrides the
    // default value in the ICE implementation if set.
    absl::optional<int> ice_inactive_timeout;

    // The interval in milliseconds at which STUN candidates will resend STUN
    // binding requests to keep NAT bindings open.
    absl::optional<int> stun_candidate_keepalive_interval;

    // ICE Periodic Regathering
    // If set, WebRTC will periodically create and propose candidates without
    // starting a new ICE generation. The regathering happens continuously with
    // interval specified in milliseconds by the uniform distribution [a, b].
    absl::optional<rtc::IntervalRange> ice_regather_interval_range;

    // Optional TurnCustomizer.
    // With this class one can modify outgoing TURN messages.
    // The object passed in must remain valid until PeerConnection::Close() is
    // called.
    webrtc::TurnCustomizer* turn_customizer = nullptr;

    // Preferred network interface.
    // A candidate pair on a preferred network has a higher precedence in ICE
    // than one on an un-preferred network, regardless of priority or network
    // cost.
    absl::optional<rtc::AdapterType> network_preference;

    // Configure the SDP semantics used by this PeerConnection. Note that the
    // WebRTC 1.0 specification requires kUnifiedPlan semantics. The
    // RtpTransceiver API is only available with kUnifiedPlan semantics.
    //
    // kPlanB will cause PeerConnection to create offers and answers with at
    // most one audio and one video m= section with multiple RtpSenders and
    // RtpReceivers specified as multiple a=ssrc lines within the section. This
    // will also cause PeerConnection to ignore all but the first m= section of
    // the same media type.
    //
    // kUnifiedPlan will cause PeerConnection to create offers and answers with
    // multiple m= sections where each m= section maps to one RtpSender and one
    // RtpReceiver (an RtpTransceiver), either both audio or both video. This
    // will also cause PeerConnection to ignore all but the first a=ssrc lines
    // that form a Plan B stream.
    //
    // For users who wish to send multiple audio/video streams and need to stay
    // interoperable with legacy WebRTC implementations or use legacy APIs,
    // specify kPlanB.
    //
    // For all other users, specify kUnifiedPlan.
    SdpSemantics sdp_semantics = SdpSemantics::kPlanB;

    // TODO(bugs.webrtc.org/9891) - Move to crypto_options or remove.
    // Actively reset the SRTP parameters whenever the DTLS transports
    // underneath are reset for every offer/answer negotiation.
    // This is only intended to be a workaround for crbug.com/835958
    // WARNING: This would cause RTP/RTCP packets decryption failure if not used
    // correctly. This flag will be deprecated soon. Do not rely on it.
    bool active_reset_srtp_params = false;

    // If MediaTransportFactory is provided in PeerConnectionFactory, this flag
    // informs PeerConnection that it should use the MediaTransportInterface for
    // media (audio/video). It's invalid to set it to |true| if the
    // MediaTransportFactory wasn't provided.
    bool use_media_transport = false;

    // If MediaTransportFactory is provided in PeerConnectionFactory, this flag
    // informs PeerConnection that it should use the MediaTransportInterface for
    // data channels.  It's invalid to set it to |true| if the
    // MediaTransportFactory wasn't provided.  Data channels over media
    // transport are not compatible with RTP or SCTP data channels.  Setting
    // both |use_media_transport_for_data_channels| and
    // |enable_rtp_data_channel| is invalid.
    bool use_media_transport_for_data_channels = false;

    // If MediaTransportFactory is provided in PeerConnectionFactory, this flag
    // informs PeerConnection that it should use the DatagramTransportInterface
    // for packets instead DTLS. It's invalid to set it to |true| if the
    // MediaTransportFactory wasn't provided.
    absl::optional<bool> use_datagram_transport;

    // If MediaTransportFactory is provided in PeerConnectionFactory, this flag
    // informs PeerConnection that it should use the DatagramTransport's
    // implementation of DataChannelTransportInterface for data channels instead
    // of SCTP-DTLS.
    absl::optional<bool> use_datagram_transport_for_data_channels;

    // If true, this PeerConnection will only use datagram transport for data
    // channels when receiving an incoming offer that includes datagram
    // transport parameters.  It will not request use of a datagram transport
    // when it creates the initial, outgoing offer.
    // This setting only applies when |use_datagram_transport_for_data_channels|
    // is true.
    absl::optional<bool> use_datagram_transport_for_data_channels_receive_only;

    // Defines advanced optional cryptographic settings related to SRTP and
    // frame encryption for native WebRTC. Setting this will overwrite any
    // settings set in PeerConnectionFactory (which is deprecated).
    absl::optional<CryptoOptions> crypto_options;

    // Configure if we should include the SDP attribute extmap-allow-mixed in
    // our offer. Although we currently do support this, it's not included in
    // our offer by default due to a previous bug that caused the SDP parser to
    // abort parsing if this attribute was present. This is fixed in Chrome 71.
    // TODO(webrtc:9985): Change default to true once sufficient time has
    // passed.
    bool offer_extmap_allow_mixed = false;

    // TURN logging identifier.
    // This identifier is added to a TURN allocation
    // and it intended to be used to be able to match client side
    // logs with TURN server logs. It will not be added if it's an empty string.
    std::string turn_logging_id;

    // Added to be able to control rollout of this feature.
    bool enable_implicit_rollback = false;

    //
    // Don't forget to update operator== if adding something.
    //
  };

  // See: https://www.w3.org/TR/webrtc/#idl-def-rtcofferansweroptions
  struct RTCOfferAnswerOptions {
    static const int kUndefined = -1;
    static const int kMaxOfferToReceiveMedia = 1;

    // The default value for constraint offerToReceiveX:true.
    static const int kOfferToReceiveMediaTrue = 1;

    // These options are left as backwards compatibility for clients who need
    // "Plan B" semantics. Clients who have switched to "Unified Plan" semantics
    // should use the RtpTransceiver API (AddTransceiver) instead.
    //
    // offer_to_receive_X set to 1 will cause a media description to be
    // generated in the offer, even if no tracks of that type have been added.
    // Values greater than 1 are treated the same.
    //
    // If set to 0, the generated directional attribute will not include the
    // "recv" direction (meaning it will be "sendonly" or "inactive".
    int offer_to_receive_video = kUndefined;
    int offer_to_receive_audio = kUndefined;

    bool voice_activity_detection = true;
    bool ice_restart = false;

    // If true, will offer to BUNDLE audio/video/data together. Not to be
    // confused with RTCP mux (multiplexing RTP and RTCP together).
    bool use_rtp_mux = true;

    // If true, "a=packetization:<payload_type> raw" attribute will be offered
    // in the SDP for all video payload and accepted in the answer if offered.
    bool raw_packetization_for_video = false;

    // This will apply to all video tracks with a Plan B SDP offer/answer.
    int num_simulcast_layers = 1;

    // If true: Use SDP format from draft-ietf-mmusic-scdp-sdp-03
    // If false: Use SDP format from draft-ietf-mmusic-sdp-sdp-26 or later
    bool use_obsolete_sctp_sdp = false;

    RTCOfferAnswerOptions() = default;

    RTCOfferAnswerOptions(int offer_to_receive_video,
                          int offer_to_receive_audio,
                          bool voice_activity_detection,
                          bool ice_restart,
                          bool use_rtp_mux)
        : offer_to_receive_video(offer_to_receive_video),
          offer_to_receive_audio(offer_to_receive_audio),
          voice_activity_detection(voice_activity_detection),
          ice_restart(ice_restart),
          use_rtp_mux(use_rtp_mux) {}
  };

  // Used by GetStats to decide which stats to include in the stats reports.
  // |kStatsOutputLevelStandard| includes the standard stats for Javascript API;
  // |kStatsOutputLevelDebug| includes both the standard stats and additional
  // stats for debugging purposes.
  enum StatsOutputLevel {
    kStatsOutputLevelStandard,
    kStatsOutputLevelDebug,
  };

  // Accessor methods to active local streams.
  // This method is not supported with kUnifiedPlan semantics. Please use
  // GetSenders() instead.
  virtual rtc::scoped_refptr<StreamCollectionInterface> local_streams() = 0;

  // Accessor methods to remote streams.
  // This method is not supported with kUnifiedPlan semantics. Please use
  // GetReceivers() instead.
  virtual rtc::scoped_refptr<StreamCollectionInterface> remote_streams() = 0;

  // Add a new MediaStream to be sent on this PeerConnection.
  // Note that a SessionDescription negotiation is needed before the
  // remote peer can receive the stream.
  //
  // This has been removed from the standard in favor of a track-based API. So,
  // this is equivalent to simply calling AddTrack for each track within the
  // stream, with the one difference that if "stream->AddTrack(...)" is called
  // later, the PeerConnection will automatically pick up the new track. Though
  // this functionality will be deprecated in the future.
  //
  // This method is not supported with kUnifiedPlan semantics. Please use
  // AddTrack instead.
  virtual bool AddStream(MediaStreamInterface* stream) = 0;

  // Remove a MediaStream from this PeerConnection.
  // Note that a SessionDescription negotiation is needed before the
  // remote peer is notified.
  //
  // This method is not supported with kUnifiedPlan semantics. Please use
  // RemoveTrack instead.
  virtual void RemoveStream(MediaStreamInterface* stream) = 0;

  // Add a new MediaStreamTrack to be sent on this PeerConnection, and return
  // the newly created RtpSender. The RtpSender will be associated with the
  // streams specified in the |stream_ids| list.
  //
  // Errors:
  // - INVALID_PARAMETER: |track| is null, has a kind other than audio or video,
  //       or a sender already exists for the track.
  // - INVALID_STATE: The PeerConnection is closed.
  virtual RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> AddTrack(
      rtc::scoped_refptr<MediaStreamTrackInterface> track,
      const std::vector<std::string>& stream_ids) = 0;

  // Remove an RtpSender from this PeerConnection.
  // Returns true on success.
  // TODO(steveanton): Replace with signature that returns RTCError.
  virtual bool RemoveTrack(RtpSenderInterface* sender) = 0;

  // Plan B semantics: Removes the RtpSender from this PeerConnection.
  // Unified Plan semantics: Stop sending on the RtpSender and mark the
  // corresponding RtpTransceiver direction as no longer sending.
  //
  // Errors:
  // - INVALID_PARAMETER: |sender| is null or (Plan B only) the sender is not
  //       associated with this PeerConnection.
  // - INVALID_STATE: PeerConnection is closed.
  // TODO(bugs.webrtc.org/9534): Rename to RemoveTrack once the other signature
  // is removed.
  virtual RTCError RemoveTrackNew(
      rtc::scoped_refptr<RtpSenderInterface> sender);

  // AddTransceiver creates a new RtpTransceiver and adds it to the set of
  // transceivers. Adding a transceiver will cause future calls to CreateOffer
  // to add a media description for the corresponding transceiver.
  //
  // The initial value of |mid| in the returned transceiver is null. Setting a
  // new session description may change it to a non-null value.
  //
  // https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-addtransceiver
  //
  // Optionally, an RtpTransceiverInit structure can be specified to configure
  // the transceiver from construction. If not specified, the transceiver will
  // default to having a direction of kSendRecv and not be part of any streams.
  //
  // These methods are only available when Unified Plan is enabled (see
  // RTCConfiguration).
  //
  // Common errors:
  // - INTERNAL_ERROR: The configuration does not have Unified Plan enabled.

  // Adds a transceiver with a sender set to transmit the given track. The kind
  // of the transceiver (and sender/receiver) will be derived from the kind of
  // the track.
  // Errors:
  // - INVALID_PARAMETER: |track| is null.
  virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
  AddTransceiver(rtc::scoped_refptr<MediaStreamTrackInterface> track) = 0;
  virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
  AddTransceiver(rtc::scoped_refptr<MediaStreamTrackInterface> track,
                 const RtpTransceiverInit& init) = 0;

  // Adds a transceiver with the given kind. Can either be MEDIA_TYPE_AUDIO or
  // MEDIA_TYPE_VIDEO.
  // Errors:
  // - INVALID_PARAMETER: |media_type| is not MEDIA_TYPE_AUDIO or
  //                      MEDIA_TYPE_VIDEO.
  virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
  AddTransceiver(cricket::MediaType media_type) = 0;
  virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
  AddTransceiver(cricket::MediaType media_type,
                 const RtpTransceiverInit& init) = 0;

  // Creates a sender without a track. Can be used for "early media"/"warmup"
  // use cases, where the application may want to negotiate video attributes
  // before a track is available to send.
  //
  // The standard way to do this would be through "addTransceiver", but we
  // don't support that API yet.
  //
  // |kind| must be "audio" or "video".
  //
  // |stream_id| is used to populate the msid attribute; if empty, one will
  // be generated automatically.
  //
  // This method is not supported with kUnifiedPlan semantics. Please use
  // AddTransceiver instead.
  virtual rtc::scoped_refptr<RtpSenderInterface> CreateSender(
      const std::string& kind,
      const std::string& stream_id) = 0;

  // If Plan B semantics are specified, gets all RtpSenders, created either
  // through AddStream, AddTrack, or CreateSender. All senders of a specific
  // media type share the same media description.
  //
  // If Unified Plan semantics are specified, gets the RtpSender for each
  // RtpTransceiver.
  virtual std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders()
      const = 0;

  // If Plan B semantics are specified, gets all RtpReceivers created when a
  // remote description is applied. All receivers of a specific media type share
  // the same media description. It is also possible to have a media description
  // with no associated RtpReceivers, if the directional attribute does not
  // indicate that the remote peer is sending any media.
  //
  // If Unified Plan semantics are specified, gets the RtpReceiver for each
  // RtpTransceiver.
  virtual std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceivers()
      const = 0;

  // Get all RtpTransceivers, created either through AddTransceiver, AddTrack or
  // by a remote description applied with SetRemoteDescription.
  //
  // Note: This method is only available when Unified Plan is enabled (see
  // RTCConfiguration).
  virtual std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>
  GetTransceivers() const = 0;

  // The legacy non-compliant GetStats() API. This correspond to the
  // callback-based version of getStats() in JavaScript. The returned metrics
  // are UNDOCUMENTED and many of them rely on implementation-specific details.
  // The goal is to DELETE THIS VERSION but we can't today because it is heavily
  // relied upon by third parties. See https://crbug.com/822696.
  //
  // This version is wired up into Chrome. Any stats implemented are
  // automatically exposed to the Web Platform. This has BYPASSED the Chrome
  // release processes for years and lead to cross-browser incompatibility
  // issues and web application reliance on Chrome-only behavior.
  //
  // This API is in "maintenance mode", serious regressions should be fixed but
  // adding new stats is highly discouraged.
  //
  // TODO(hbos): Deprecate and remove this when third parties have migrated to
  // the spec-compliant GetStats() API. https://crbug.com/822696
  virtual bool GetStats(StatsObserver* observer,
                        MediaStreamTrackInterface* track,  // Optional
                        StatsOutputLevel level) = 0;
  // The spec-compliant GetStats() API. This correspond to the promise-based
  // version of getStats() in JavaScript. Implementation status is described in
  // api/stats/rtcstats_objects.h. For more details on stats, see spec:
  // https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-getstats
  // TODO(hbos): Takes shared ownership, use rtc::scoped_refptr<> instead. This
  // requires stop overriding the current version in third party or making third
  // party calls explicit to avoid ambiguity during switch. Make the future
  // version abstract as soon as third party projects implement it.
  virtual void GetStats(RTCStatsCollectorCallback* callback) = 0;
  // Spec-compliant getStats() performing the stats selection algorithm with the
  // sender. https://w3c.github.io/webrtc-pc/#dom-rtcrtpsender-getstats
  virtual void GetStats(
      rtc::scoped_refptr<RtpSenderInterface> selector,
      rtc::scoped_refptr<RTCStatsCollectorCallback> callback) = 0;
  // Spec-compliant getStats() performing the stats selection algorithm with the
  // receiver. https://w3c.github.io/webrtc-pc/#dom-rtcrtpreceiver-getstats
  virtual void GetStats(
      rtc::scoped_refptr<RtpReceiverInterface> selector,
      rtc::scoped_refptr<RTCStatsCollectorCallback> callback) = 0;
  // Clear cached stats in the RTCStatsCollector.
  // Exposed for testing while waiting for automatic cache clear to work.
  // https://bugs.webrtc.org/8693
  virtual void ClearStatsCache() {}

  // Create a data channel with the provided config, or default config if none
  // is provided. Note that an offer/answer negotiation is still necessary
  // before the data channel can be used.
  //
  // Also, calling CreateDataChannel is the only way to get a data "m=" section
  // in SDP, so it should be done before CreateOffer is called, if the
  // application plans to use data channels.
  virtual rtc::scoped_refptr<DataChannelInterface> CreateDataChannel(
      const std::string& label,
      const DataChannelInit* config) = 0;

  // Returns the more recently applied description; "pending" if it exists, and
  // otherwise "current". See below.
  virtual const SessionDescriptionInterface* local_description() const = 0;
  virtual const SessionDescriptionInterface* remote_description() const = 0;

  // A "current" description the one currently negotiated from a complete
  // offer/answer exchange.
  virtual const SessionDescriptionInterface* current_local_description()
      const = 0;
  virtual const SessionDescriptionInterface* current_remote_description()
      const = 0;

  // A "pending" description is one that's part of an incomplete offer/answer
  // exchange (thus, either an offer or a pranswer). Once the offer/answer
  // exchange is finished, the "pending" description will become "current".
  virtual const SessionDescriptionInterface* pending_local_description()
      const = 0;
  virtual const SessionDescriptionInterface* pending_remote_description()
      const = 0;

  // Tells the PeerConnection that ICE should be restarted. This triggers a need
  // for negotiation and subsequent CreateOffer() calls will act as if
  // RTCOfferAnswerOptions::ice_restart is true.
  // https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-restartice
  // TODO(hbos): Remove default implementation when downstream projects
  // implement this.
  virtual void RestartIce() = 0;

  // Create a new offer.
  // The CreateSessionDescriptionObserver callback will be called when done.
  virtual void CreateOffer(CreateSessionDescriptionObserver* observer,
                           const RTCOfferAnswerOptions& options) = 0;

  // Create an answer to an offer.
  // The CreateSessionDescriptionObserver callback will be called when done.
  virtual void CreateAnswer(CreateSessionDescriptionObserver* observer,
                            const RTCOfferAnswerOptions& options) = 0;

  // Sets the local session description.
  // The PeerConnection takes the ownership of |desc| even if it fails.
  // The |observer| callback will be called when done.
  // TODO(deadbeef): Change |desc| to be a unique_ptr, to make it clear
  // that this method always takes ownership of it.
  virtual void SetLocalDescription(SetSessionDescriptionObserver* observer,
                                   SessionDescriptionInterface* desc) = 0;
  // Sets the remote session description.
  // The PeerConnection takes the ownership of |desc| even if it fails.
  // The |observer| callback will be called when done.
  // TODO(hbos): Remove when Chrome implements the new signature.
  virtual void SetRemoteDescription(SetSessionDescriptionObserver* observer,
                                    SessionDescriptionInterface* desc) {}
  virtual void SetRemoteDescription(
      std::unique_ptr<SessionDescriptionInterface> desc,
      rtc::scoped_refptr<SetRemoteDescriptionObserverInterface> observer) = 0;

  virtual PeerConnectionInterface::RTCConfiguration GetConfiguration() = 0;

  // Sets the PeerConnection's global configuration to |config|.
  //
  // The members of |config| that may be changed are |type|, |servers|,
  // |ice_candidate_pool_size| and |prune_turn_ports| (though the candidate
  // pool size can't be changed after the first call to SetLocalDescription).
  // Note that this means the BUNDLE and RTCP-multiplexing policies cannot be
  // changed with this method.
  //
  // Any changes to STUN/TURN servers or ICE candidate policy will affect the
  // next gathering phase, and cause the next call to createOffer to generate
  // new ICE credentials, as described in JSEP. This also occurs when
  // |prune_turn_ports| changes, for the same reasoning.
  //
  // If an error occurs, returns false and populates |error| if non-null:
  // - INVALID_MODIFICATION if |config| contains a modified parameter other
  //   than one of the parameters listed above.
  // - INVALID_RANGE if |ice_candidate_pool_size| is out of range.
  // - SYNTAX_ERROR if parsing an ICE server URL failed.
  // - INVALID_PARAMETER if a TURN server is missing |username| or |password|.
  // - INTERNAL_ERROR if an unexpected error occurred.
  //
  // TODO(nisse): Make this pure virtual once all Chrome subclasses of
  // PeerConnectionInterface implement it.
  virtual RTCError SetConfiguration(
      const PeerConnectionInterface::RTCConfiguration& config);

  // Provides a remote candidate to the ICE Agent.
  // A copy of the |candidate| will be created and added to the remote
  // description. So the caller of this method still has the ownership of the
  // |candidate|.
  virtual bool AddIceCandidate(const IceCandidateInterface* candidate) = 0;

  // Removes a group of remote candidates from the ICE agent. Needed mainly for
  // continual gathering, to avoid an ever-growing list of candidates as
  // networks come and go.
  virtual bool RemoveIceCandidates(
      const std::vector<cricket::Candidate>& candidates) = 0;

  // 0 <= min <= current <= max should hold for set parameters.
  struct BitrateParameters {
    BitrateParameters();
    ~BitrateParameters();

    absl::optional<int> min_bitrate_bps;
    absl::optional<int> current_bitrate_bps;
    absl::optional<int> max_bitrate_bps;
  };

  // SetBitrate limits the bandwidth allocated for all RTP streams sent by
  // this PeerConnection. Other limitations might affect these limits and
  // are respected (for example "b=AS" in SDP).
  //
  // Setting |current_bitrate_bps| will reset the current bitrate estimate
  // to the provided value.
  virtual RTCError SetBitrate(const BitrateSettings& bitrate);

  // TODO(nisse): Deprecated - use version above. These two default
  // implementations require subclasses to implement one or the other
  // of the methods.
  virtual RTCError SetBitrate(const BitrateParameters& bitrate_parameters);

  // Enable/disable playout of received audio streams. Enabled by default. Note
  // that even if playout is enabled, streams will only be played out if the
  // appropriate SDP is also applied. Setting |playout| to false will stop
  // playout of the underlying audio device but starts a task which will poll
  // for audio data every 10ms to ensure that audio processing happens and the
  // audio statistics are updated.
  // TODO(henrika): deprecate and remove this.
  virtual void SetAudioPlayout(bool playout) {}

  // Enable/disable recording of transmitted audio streams. Enabled by default.
  // Note that even if recording is enabled, streams will only be recorded if
  // the appropriate SDP is also applied.
  // TODO(henrika): deprecate and remove this.
  virtual void SetAudioRecording(bool recording) {}

  // Looks up the DtlsTransport associated with a MID value.
  // In the Javascript API, DtlsTransport is a property of a sender, but
  // because the PeerConnection owns the DtlsTransport in this implementation,
  // it is better to look them up on the PeerConnection.
  virtual rtc::scoped_refptr<DtlsTransportInterface> LookupDtlsTransportByMid(
      const std::string& mid) = 0;

  // Returns the SCTP transport, if any.
  virtual rtc::scoped_refptr<SctpTransportInterface> GetSctpTransport()
      const = 0;

  // Returns the current SignalingState.
  virtual SignalingState signaling_state() = 0;

  // Returns an aggregate state of all ICE *and* DTLS transports.
  // This is left in place to avoid breaking native clients who expect our old,
  // nonstandard behavior.
  // TODO(jonasolsson): deprecate and remove this.
  virtual IceConnectionState ice_connection_state() = 0;

  // Returns an aggregated state of all ICE transports.
  virtual IceConnectionState standardized_ice_connection_state() = 0;

  // Returns an aggregated state of all ICE and DTLS transports.
  virtual PeerConnectionState peer_connection_state() = 0;

  virtual IceGatheringState ice_gathering_state() = 0;

  // Start RtcEventLog using an existing output-sink. Takes ownership of
  // |output| and passes it on to Call, which will take the ownership. If the
  // operation fails the output will be closed and deallocated. The event log
  // will send serialized events to the output object every |output_period_ms|.
  // Applications using the event log should generally make their own trade-off
  // regarding the output period. A long period is generally more efficient,
  // with potential drawbacks being more bursty thread usage, and more events
  // lost in case the application crashes. If the |output_period_ms| argument is
  // omitted, webrtc selects a default deemed to be workable in most cases.
  virtual bool StartRtcEventLog(std::unique_ptr<RtcEventLogOutput> output,
                                int64_t output_period_ms) = 0;
  virtual bool StartRtcEventLog(std::unique_ptr<RtcEventLogOutput> output) = 0;

  // Stops logging the RtcEventLog.
  virtual void StopRtcEventLog() = 0;

  // Terminates all media, closes the transports, and in general releases any
  // resources used by the PeerConnection. This is an irreversible operation.
  //
  // Note that after this method completes, the PeerConnection will no longer
  // use the PeerConnectionObserver interface passed in on construction, and
  // thus the observer object can be safely destroyed.
  virtual void Close() = 0;

 protected:
  // Dtor protected as objects shouldn't be deleted via this interface.
  ~PeerConnectionInterface() override = default;
};

// PeerConnection callback interface, used for RTCPeerConnection events.
// Application should implement these methods.
class PeerConnectionObserver {
 public:
  virtual ~PeerConnectionObserver() = default;

  // Triggered when the SignalingState changed.
  virtual void OnSignalingChange(
      PeerConnectionInterface::SignalingState new_state) = 0;

  // Triggered when media is received on a new stream from remote peer.
  virtual void OnAddStream(rtc::scoped_refptr<MediaStreamInterface> stream) {}

  // Triggered when a remote peer closes a stream.
  virtual void OnRemoveStream(rtc::scoped_refptr<MediaStreamInterface> stream) {
  }

  // Triggered when a remote peer opens a data channel.
  virtual void OnDataChannel(
      rtc::scoped_refptr<DataChannelInterface> data_channel) = 0;

  // Triggered when renegotiation is needed. For example, an ICE restart
  // has begun.
  virtual void OnRenegotiationNeeded() = 0;

  // Called any time the legacy IceConnectionState changes.
  //
  // Note that our ICE states lag behind the standard slightly. The most
  // notable differences include the fact that "failed" occurs after 15
  // seconds, not 30, and this actually represents a combination ICE + DTLS
  // state, so it may be "failed" if DTLS fails while ICE succeeds.
  //
  // TODO(jonasolsson): deprecate and remove this.
  virtual void OnIceConnectionChange(
      PeerConnectionInterface::IceConnectionState new_state) {}

  // Called any time the standards-compliant IceConnectionState changes.
  virtual void OnStandardizedIceConnectionChange(
      PeerConnectionInterface::IceConnectionState new_state) {}

  // Called any time the PeerConnectionState changes.
  virtual void OnConnectionChange(
      PeerConnectionInterface::PeerConnectionState new_state) {}

  // Called any time the IceGatheringState changes.
  virtual void OnIceGatheringChange(
      PeerConnectionInterface::IceGatheringState new_state) = 0;

  // A new ICE candidate has been gathered.
  virtual void OnIceCandidate(const IceCandidateInterface* candidate) = 0;

  // Gathering of an ICE candidate failed.
  // See https://w3c.github.io/webrtc-pc/#event-icecandidateerror
  // |host_candidate| is a stringified socket address.
  virtual void OnIceCandidateError(const std::string& host_candidate,
                                   const std::string& url,
                                   int error_code,
                                   const std::string& error_text) {}

  // Ice candidates have been removed.
  // TODO(honghaiz): Make this a pure virtual method when all its subclasses
  // implement it.
  virtual void OnIceCandidatesRemoved(
      const std::vector<cricket::Candidate>& candidates) {}

  // Called when the ICE connection receiving status changes.
  virtual void OnIceConnectionReceivingChange(bool receiving) {}

  // Called when the selected candidate pair for the ICE connection changes.
  virtual void OnIceSelectedCandidatePairChanged(
      const cricket::CandidatePairChangeEvent& event) {}

  // This is called when a receiver and its track are created.
  // TODO(zhihuang): Make this pure virtual when all subclasses implement it.
  // Note: This is called with both Plan B and Unified Plan semantics. Unified
  // Plan users should prefer OnTrack, OnAddTrack is only called as backwards
  // compatibility (and is called in the exact same situations as OnTrack).
  virtual void OnAddTrack(
      rtc::scoped_refptr<RtpReceiverInterface> receiver,
      const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams) {}

  // This is called when signaling indicates a transceiver will be receiving
  // media from the remote endpoint. This is fired during a call to
  // SetRemoteDescription. The receiving track can be accessed by:
  // |transceiver->receiver()->track()| and its associated streams by
  // |transceiver->receiver()->streams()|.
  // Note: This will only be called if Unified Plan semantics are specified.
  // This behavior is specified in section 2.2.8.2.5 of the "Set the
  // RTCSessionDescription" algorithm:
  // https://w3c.github.io/webrtc-pc/#set-description
  virtual void OnTrack(
      rtc::scoped_refptr<RtpTransceiverInterface> transceiver) {}

  // Called when signaling indicates that media will no longer be received on a
  // track.
  // With Plan B semantics, the given receiver will have been removed from the
  // PeerConnection and the track muted.
  // With Unified Plan semantics, the receiver will remain but the transceiver
  // will have changed direction to either sendonly or inactive.
  // https://w3c.github.io/webrtc-pc/#process-remote-track-removal
  // TODO(hbos,deadbeef): Make pure virtual when all subclasses implement it.
  virtual void OnRemoveTrack(
      rtc::scoped_refptr<RtpReceiverInterface> receiver) {}

  // Called when an interesting usage is detected by WebRTC.
  // An appropriate action is to add information about the context of the
  // PeerConnection and write the event to some kind of "interesting events"
  // log function.
  // The heuristics for defining what constitutes "interesting" are
  // implementation-defined.
  virtual void OnInterestingUsage(int usage_pattern) {}
};

// PeerConnectionDependencies holds all of PeerConnections dependencies.
// A dependency is distinct from a configuration as it defines significant
// executable code that can be provided by a user of the API.
//
// All new dependencies should be added as a unique_ptr to allow the
// PeerConnection object to be the definitive owner of the dependencies
// lifetime making injection safer.
struct PeerConnectionDependencies final {
  explicit PeerConnectionDependencies(PeerConnectionObserver* observer_in);
  // This object is not copyable or assignable.
  PeerConnectionDependencies(const PeerConnectionDependencies&) = delete;
  PeerConnectionDependencies& operator=(const PeerConnectionDependencies&) =
      delete;
  // This object is only moveable.
  PeerConnectionDependencies(PeerConnectionDependencies&&);
  PeerConnectionDependencies& operator=(PeerConnectionDependencies&&) = default;
  ~PeerConnectionDependencies();
  // Mandatory dependencies
  PeerConnectionObserver* observer = nullptr;
  // Optional dependencies
  // TODO(bugs.webrtc.org/7447): remove port allocator once downstream is
  // updated. For now, you can only set one of allocator and
  // packet_socket_factory, not both.
  std::unique_ptr<cricket::PortAllocator> allocator;
  std::unique_ptr<rtc::PacketSocketFactory> packet_socket_factory;
  std::unique_ptr<webrtc::AsyncResolverFactory> async_resolver_factory;
  std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator;
  std::unique_ptr<rtc::SSLCertificateVerifier> tls_cert_verifier;
  std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>
      video_bitrate_allocator_factory;
};

// PeerConnectionFactoryDependencies holds all of the PeerConnectionFactory
// dependencies. All new dependencies should be added here instead of
// overloading the function. This simplifies dependency injection and makes it
// clear which are mandatory and optional. If possible please allow the peer
// connection factory to take ownership of the dependency by adding a unique_ptr
// to this structure.
struct PeerConnectionFactoryDependencies final {
  PeerConnectionFactoryDependencies();
  // This object is not copyable or assignable.
  PeerConnectionFactoryDependencies(const PeerConnectionFactoryDependencies&) =
      delete;
  PeerConnectionFactoryDependencies& operator=(
      const PeerConnectionFactoryDependencies&) = delete;
  // This object is only moveable.
  PeerConnectionFactoryDependencies(PeerConnectionFactoryDependencies&&);
  PeerConnectionFactoryDependencies& operator=(
      PeerConnectionFactoryDependencies&&) = default;
  ~PeerConnectionFactoryDependencies();

  // Optional dependencies
  rtc::Thread* network_thread = nullptr;
  rtc::Thread* worker_thread = nullptr;
  rtc::Thread* signaling_thread = nullptr;
  std::unique_ptr<TaskQueueFactory> task_queue_factory;
  std::unique_ptr<cricket::MediaEngineInterface> media_engine;
  std::unique_ptr<CallFactoryInterface> call_factory;
  std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory;
  std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory;
  std::unique_ptr<NetworkStatePredictorFactoryInterface>
      network_state_predictor_factory;
  std::unique_ptr<NetworkControllerFactoryInterface> network_controller_factory;
  std::unique_ptr<MediaTransportFactory> media_transport_factory;
};

// PeerConnectionFactoryInterface is the factory interface used for creating
// PeerConnection, MediaStream and MediaStreamTrack objects.
//
// The simplest method for obtaiing one, CreatePeerConnectionFactory will
// create the required libjingle threads, socket and network manager factory
// classes for networking if none are provided, though it requires that the
// application runs a message loop on the thread that called the method (see
// explanation below)
//
// If an application decides to provide its own threads and/or implementation
// of networking classes, it should use the alternate
// CreatePeerConnectionFactory method which accepts threads as input, and use
// the CreatePeerConnection version that takes a PortAllocator as an argument.
class PeerConnectionFactoryInterface : public rtc::RefCountInterface {
 public:
  class Options {
   public:
    Options() {}

    // If set to true, created PeerConnections won't enforce any SRTP
    // requirement, allowing unsecured media. Should only be used for
    // testing/debugging.
    bool disable_encryption = false;

    // Deprecated. The only effect of setting this to true is that
    // CreateDataChannel will fail, which is not that useful.
    bool disable_sctp_data_channels = false;

    // If set to true, any platform-supported network monitoring capability
    // won't be used, and instead networks will only be updated via polling.
    //
    // This only has an effect if a PeerConnection is created with the default
    // PortAllocator implementation.
    bool disable_network_monitor = false;

    // Sets the network types to ignore. For instance, calling this with
    // ADAPTER_TYPE_ETHERNET | ADAPTER_TYPE_LOOPBACK will ignore Ethernet and
    // loopback interfaces.
    int network_ignore_mask = rtc::kDefaultNetworkIgnoreMask;

    // Sets the maximum supported protocol version. The highest version
    // supported by both ends will be used for the connection, i.e. if one
    // party supports DTLS 1.0 and the other DTLS 1.2, DTLS 1.0 will be used.
    rtc::SSLProtocolVersion ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;

    // Sets crypto related options, e.g. enabled cipher suites.
    CryptoOptions crypto_options = CryptoOptions::NoGcm();
  };

  // Set the options to be used for subsequently created PeerConnections.
  virtual void SetOptions(const Options& options) = 0;

  // The preferred way to create a new peer connection. Simply provide the
  // configuration and a PeerConnectionDependencies structure.
  // TODO(benwright): Make pure virtual once downstream mock PC factory classes
  // are updated.
  virtual rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
      const PeerConnectionInterface::RTCConfiguration& configuration,
      PeerConnectionDependencies dependencies);

  // Deprecated; |allocator| and |cert_generator| may be null, in which case
  // default implementations will be used.
  //
  // |observer| must not be null.
  //
  // Note that this method does not take ownership of |observer|; it's the
  // responsibility of the caller to delete it. It can be safely deleted after
  // Close has been called on the returned PeerConnection, which ensures no
  // more observer callbacks will be invoked.
  virtual rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
      const PeerConnectionInterface::RTCConfiguration& configuration,
      std::unique_ptr<cricket::PortAllocator> allocator,
      std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
      PeerConnectionObserver* observer);

  // Returns the capabilities of an RTP sender of type |kind|.
  // If for some reason you pass in MEDIA_TYPE_DATA, returns an empty structure.
  // TODO(orphis): Make pure virtual when all subclasses implement it.
  virtual RtpCapabilities GetRtpSenderCapabilities(
      cricket::MediaType kind) const;

  // Returns the capabilities of an RTP receiver of type |kind|.
  // If for some reason you pass in MEDIA_TYPE_DATA, returns an empty structure.
  // TODO(orphis): Make pure virtual when all subclasses implement it.
  virtual RtpCapabilities GetRtpReceiverCapabilities(
      cricket::MediaType kind) const;

  virtual rtc::scoped_refptr<MediaStreamInterface> CreateLocalMediaStream(
      const std::string& stream_id) = 0;

  // Creates an AudioSourceInterface.
  // |options| decides audio processing settings.
  virtual rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
      const cricket::AudioOptions& options) = 0;

  // Creates a new local VideoTrack. The same |source| can be used in several
  // tracks.
  virtual rtc::scoped_refptr<VideoTrackInterface> CreateVideoTrack(
      const std::string& label,
      VideoTrackSourceInterface* source) = 0;

  // Creates an new AudioTrack. At the moment |source| can be null.
  virtual rtc::scoped_refptr<AudioTrackInterface> CreateAudioTrack(
      const std::string& label,
      AudioSourceInterface* source) = 0;

  // Starts AEC dump using existing file. Takes ownership of |file| and passes
  // it on to VoiceEngine (via other objects) immediately, which will take
  // the ownerhip. If the operation fails, the file will be closed.
  // A maximum file size in bytes can be specified. When the file size limit is
  // reached, logging is stopped automatically. If max_size_bytes is set to a
  // value <= 0, no limit will be used, and logging will continue until the
  // StopAecDump function is called.
  // TODO(webrtc:6463): Delete default implementation when downstream mocks
  // classes are updated.
  virtual bool StartAecDump(FILE* file, int64_t max_size_bytes) {
    return false;
  }

  // Stops logging the AEC dump.
  virtual void StopAecDump() = 0;

 protected:
  // Dtor and ctor protected as objects shouldn't be created or deleted via
  // this interface.
  PeerConnectionFactoryInterface() {}
  ~PeerConnectionFactoryInterface() override = default;
};

// CreateModularPeerConnectionFactory is implemented in the "peerconnection"
// build target, which doesn't pull in the implementations of every module
// webrtc may use.
//
// If an application knows it will only require certain modules, it can reduce
// webrtc's impact on its binary size by depending only on the "peerconnection"
// target and the modules the application requires, using
// CreateModularPeerConnectionFactory. For example, if an application
// only uses WebRTC for audio, it can pass in null pointers for the
// video-specific interfaces, and omit the corresponding modules from its
// build.
//
// If |network_thread| or |worker_thread| are null, the PeerConnectionFactory
// will create the necessary thread internally. If |signaling_thread| is null,
// the PeerConnectionFactory will use the thread on which this method is called
// as the signaling thread, wrapping it in an rtc::Thread object if needed.
rtc::scoped_refptr<PeerConnectionFactoryInterface>
CreateModularPeerConnectionFactory(
    PeerConnectionFactoryDependencies dependencies);

}  // namespace webrtc

#endif  // API_PEER_CONNECTION_INTERFACE_H_