| /* |
| * 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/fec_controller.h" |
| #include "api/jsep.h" |
| #include "api/media_stream_interface.h" |
| #include "api/media_transport_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/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/network_control.h" |
| #include "api/turn_customizer.h" |
| #include "media/base/media_config.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 "media/base/media_engine.h" // nogncheck |
| #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 SSLIdentity; |
| class Thread; |
| } // namespace rtc |
| |
| namespace webrtc { |
| class AudioDeviceModule; |
| class AudioMixer; |
| class AudioProcessing; |
| class DtlsTransportInterface; |
| class SctpTransportInterface; |
| class VideoDecoderFactory; |
| class VideoEncoderFactory; |
| |
| // 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; |
| |
| // 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; |
| |
| // |
| // 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); |
| |
| // Remove an RtpSender from this PeerConnection. |
| // Returns true on success. |
| // TODO(steveanton): Replace with signature that returns RTCError. |
| virtual bool RemoveTrack(RtpSenderInterface* sender); |
| |
| // 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. |
| // TODO(steveanton): Make these pure virtual once downstream projects have |
| // updated. |
| |
| // 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); |
| virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> |
| AddTransceiver(rtc::scoped_refptr<MediaStreamTrackInterface> track, |
| const RtpTransceiverInit& init); |
| |
| // 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); |
| virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> |
| AddTransceiver(cricket::MediaType media_type, const RtpTransceiverInit& init); |
| |
| // TODO(deadbeef): Make these pure virtual once all subclasses implement them. |
| |
| // 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); |
| |
| // 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; |
| |
| // 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; |
| |
| // 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; |
| |
| // 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) {} |
| // Spec-compliant getStats() performing the stats selection algorithm with the |
| // sender. https://w3c.github.io/webrtc-pc/#dom-rtcrtpsender-getstats |
| // TODO(hbos): Make abstract as soon as third party projects implement it. |
| virtual void GetStats( |
| rtc::scoped_refptr<RtpSenderInterface> selector, |
| rtc::scoped_refptr<RTCStatsCollectorCallback> callback) {} |
| // Spec-compliant getStats() performing the stats selection algorithm with the |
| // receiver. https://w3c.github.io/webrtc-pc/#dom-rtcrtpreceiver-getstats |
| // TODO(hbos): Make abstract as soon as third party projects implement it. |
| virtual void GetStats( |
| rtc::scoped_refptr<RtpReceiverInterface> selector, |
| rtc::scoped_refptr<RTCStatsCollectorCallback> callback) {} |
| // 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; |
| virtual const SessionDescriptionInterface* current_remote_description() const; |
| |
| // 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; |
| virtual const SessionDescriptionInterface* pending_remote_description() const; |
| |
| // 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() {} |
| |
| // 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) {} |
| // TODO(hbos): Make pure virtual when Chrome has updated its signature. |
| virtual void SetRemoteDescription( |
| std::unique_ptr<SessionDescriptionInterface> desc, |
| rtc::scoped_refptr<SetRemoteDescriptionObserverInterface> observer) {} |
| |
| // TODO(deadbeef): Make this pure virtual once all Chrome subclasses of |
| // PeerConnectionInterface implement it. |
| virtual PeerConnectionInterface::RTCConfiguration GetConfiguration(); |
| |
| // 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 <= 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. |
| // TODO(hta): Remove default implementation after updating Chrome. |
| virtual rtc::scoped_refptr<DtlsTransportInterface> LookupDtlsTransportByMid( |
| const std::string& mid); |
| |
| // Returns the SCTP transport, if any. |
| // TODO(hta): Remove default implementation after updating Chrome. |
| virtual rtc::scoped_refptr<SctpTransportInterface> GetSctpTransport() const; |
| |
| // 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(); |
| |
| // Returns an aggregated state of all ICE and DTLS transports. |
| virtual PeerConnectionState peer_connection_state(); |
| |
| 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); |
| virtual bool StartRtcEventLog(std::unique_ptr<RtcEventLogOutput> output); |
| |
| // Stops logging the RtcEventLog. |
| // TODO(ivoc): Make this pure virtual when Chrome is updat ed. |
| virtual void StopRtcEventLog() {} |
| |
| // 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_ |