# Enums String enums of the specifications. Members compare equal to their spec strings and print as them: `RTCSignalingState.have_local_offer == 'have-local-offer'`. Hyphens in the values become underscores in the member names. Parameters that take an enum take its string value too, as typed by the `...Value` literal next to it. ### *class* webrtc.enums.RTCPeerConnectionState The overall state of a connection, combined from its ICE and DTLS transports. See [MDN: RTCPeerConnection/connectionState](https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/connectionState). #### new *= 'new'* No transport has started connecting yet #### connecting *= 'connecting'* A transport is connecting, and none has failed #### connected *= 'connected'* Every transport in use is connected #### disconnected *= 'disconnected'* A transport lost connectivity, and may recover #### failed *= 'failed'* A transport failed and needs an ICE restart #### closed *= 'closed'* The connection was closed ### *class* webrtc.enums.RTCSignalingState Where a connection is in the offer/answer exchange. See [MDN: RTCPeerConnection/signalingState](https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/signalingState). #### stable *= 'stable'* No offer/answer exchange is in progress #### have_local_offer *= 'have-local-offer'* A local offer is set, waiting for an answer #### have_local_pranswer *= 'have-local-pranswer'* A remote offer and a local provisional answer are set #### have_remote_offer *= 'have-remote-offer'* A remote offer is set, waiting for a local answer #### have_remote_pranswer *= 'have-remote-pranswer'* A local offer and a remote provisional answer are set #### closed *= 'closed'* The connection was closed ### *class* webrtc.enums.RTCIceConnectionState The state of the ICE transports of a connection, taken together. See [MDN: RTCPeerConnection/iceConnectionState](https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/iceConnectionState). #### new *= 'new'* No transport is checking candidates yet #### checking *= 'checking'* A transport is checking candidate pairs, and none is connected #### connected *= 'connected'* Every transport has a usable pair #### completed *= 'completed'* Every transport finished checking and has a usable pair #### failed *= 'failed'* A transport found no usable pair #### disconnected *= 'disconnected'* A transport lost connectivity, and may recover #### closed *= 'closed'* The connection was closed ### *class* webrtc.enums.RTCIceGatheringState The candidate gathering state of the ICE transports of a connection, taken together. See [MDN: RTCPeerConnection/iceGatheringState](https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/iceGatheringState). #### new *= 'new'* No transport has started gathering #### gathering *= 'gathering'* A transport is gathering candidates #### complete *= 'complete'* Every transport finished gathering ### *class* webrtc.enums.RTCSdpType The type of a session description. See [MDN: RTCSessionDescription/type](https://developer.mozilla.org/en-US/docs/Web/API/RTCSessionDescription/type). #### offer *= 'offer'* An offer, which starts an exchange #### pranswer *= 'pranswer'* A provisional answer, which can still be replaced #### answer *= 'answer'* The final answer, which ends an exchange #### rollback *= 'rollback'* Discards a pending offer and returns to the last stable state ### webrtc.enums.RTCSdpTypeValue The string values of [`RTCSdpType`](#webrtc.enums.RTCSdpType), which parameters accept in place of its members alias of [`Literal`](https://docs.python.org/3/library/typing.html#typing.Literal)[‘offer’, ‘pranswer’, ‘answer’, ‘rollback’] ### *class* webrtc.enums.MediaStreamTrackState Whether a track still delivers media. See [MDN: MediaStreamTrack/readyState](https://developer.mozilla.org/en-US/docs/Web/API/MediaStreamTrack/readyState). #### live *= 'live'* The track delivers media, or will when it’s unmuted #### ended *= 'ended'* The track was stopped or its source ended, and it won’t deliver media again ### *class* webrtc.enums.MediaStreamSourceState The state of the native source of a track. This library adds it. The specifications have no such enum. #### initializing *= 'initializing'* The source is starting #### live *= 'live'* The source produces media #### ended *= 'ended'* The source stopped permanently #### muted *= 'muted'* The source produces no media for now ### *class* webrtc.enums.RTCRtpTransceiverDirection Whether a transceiver sends media, receives it, both or neither. See [MDN: RTCRtpTransceiver/direction](https://developer.mozilla.org/en-US/docs/Web/API/RTCRtpTransceiver/direction). #### sendrecv *= 'sendrecv'* Sends and receives #### sendonly *= 'sendonly'* Sends only #### recvonly *= 'recvonly'* Receives only #### inactive *= 'inactive'* Neither sends nor receives #### stopped *= 'stopped'* Stopped, which isn’t a direction that can be set ### webrtc.enums.RTCRtpTransceiverDirectionValue The string values of [`RTCRtpTransceiverDirection`](#webrtc.enums.RTCRtpTransceiverDirection), which parameters accept in place of its members alias of [`Literal`](https://docs.python.org/3/library/typing.html#typing.Literal)[‘sendrecv’, ‘sendonly’, ‘recvonly’, ‘inactive’, ‘stopped’] ### *class* webrtc.enums.MediaType The kind of a track or a transceiver. Tracks are audio or video. The data and unsupported kinds come from the media sections of the native WebRTC engine. See [MDN: MediaStreamTrack/kind](https://developer.mozilla.org/en-US/docs/Web/API/MediaStreamTrack/kind). #### audio *= 'audio'* Audio #### video *= 'video'* Video #### data *= 'data'* The media section of data channels #### unsupported *= 'unsupported'* A media section of a kind the native WebRTC engine doesn’t handle ### webrtc.enums.MediaTypeValue The string values of [`MediaType`](#webrtc.enums.MediaType), which parameters accept in place of its members alias of [`Literal`](https://docs.python.org/3/library/typing.html#typing.Literal)[‘audio’, ‘video’, ‘data’, ‘unsupported’] ### *class* webrtc.enums.MediaDeviceKind The kind of a media device. See [MDN: MediaDeviceInfo/kind](https://developer.mozilla.org/en-US/docs/Web/API/MediaDeviceInfo/kind). #### audioinput *= 'audioinput'* A microphone or another audio input #### audiooutput *= 'audiooutput'* Speakers, headphones or another audio output #### videoinput *= 'videoinput'* A camera or another video input ### *class* webrtc.enums.VideoFacingModeEnum The direction a camera faces, relative to the user. See [MDN: MediaTrackConstraints/facingMode](https://developer.mozilla.org/en-US/docs/Web/API/MediaTrackConstraints/facingMode). #### user *= 'user'* Towards the user, like a front camera #### environment *= 'environment'* Away from the user, like a back camera #### left *= 'left'* Towards the left of the user #### right *= 'right'* Towards the right of the user ### *class* webrtc.enums.VideoResizeModeEnum Whether the video of a source may be cropped and scaled to meet the constraints of a track. #### none *= 'none'* The video keeps the resolution and frame rate of the source #### crop_and_scale *= 'crop-and-scale'* The video may be cropped, scaled down and have frames dropped ### *class* webrtc.enums.EchoCancellationModeEnum Which audio played out is removed as echo from a microphone. See [MDN: MediaTrackConstraints/echoCancellation](https://developer.mozilla.org/en-US/docs/Web/API/MediaTrackConstraints/echoCancellation). #### all *= 'all'* Any audio the system plays #### remote_only *= 'remote-only'* Only the audio received from remote peers ### *class* webrtc.enums.RTCIceComponent Whether an ICE transport or candidate carries RTP or RTCP. See [MDN: RTCIceCandidate/component](https://developer.mozilla.org/en-US/docs/Web/API/RTCIceCandidate/component). #### rtp *= 'rtp'* RTP, or RTP and RTCP multiplexed together #### rtcp *= 'rtcp'* RTCP on its own transport ### *class* webrtc.enums.RTCIceRole The role of an ICE agent. The controlling agent picks the candidate pair. See [MDN: RTCIceTransport/role](https://developer.mozilla.org/en-US/docs/Web/API/RTCIceTransport/role). #### controlling *= 'controlling'* Nominates the candidate pair in use #### controlled *= 'controlled'* Accepts the pair the other agent nominates #### unknown *= 'unknown'* Not decided yet ### *class* webrtc.enums.RTCIceTransportState The state of an ICE transport. See [MDN: RTCIceTransport/state](https://developer.mozilla.org/en-US/docs/Web/API/RTCIceTransport/state). #### new *= 'new'* The transport is gathering candidates or waiting for remote ones, and hasn’t started checks #### checking *= 'checking'* Candidate pairs are being checked, and none works yet #### connected *= 'connected'* A usable pair was found, and checks may go on #### completed *= 'completed'* Checks finished, and a usable pair was found #### disconnected *= 'disconnected'* Connectivity was lost, and may recover #### failed *= 'failed'* Checks finished without a usable pair #### closed *= 'closed'* The transport was closed ### *class* webrtc.enums.RTCIceGathererState The candidate gathering state of an ICE transport. See [MDN: RTCIceTransport/gatheringState](https://developer.mozilla.org/en-US/docs/Web/API/RTCIceTransport/gatheringState). #### new *= 'new'* Gathering hasn’t started #### gathering *= 'gathering'* Candidates are being gathered #### complete *= 'complete'* Gathering finished ### *class* webrtc.enums.RTCDtlsTransportState The state of a DTLS transport. See [MDN: RTCDtlsTransport/state](https://developer.mozilla.org/en-US/docs/Web/API/RTCDtlsTransport/state). #### new *= 'new'* The handshake hasn’t started #### connecting *= 'connecting'* The handshake is in progress #### connected *= 'connected'* The handshake succeeded #### closed *= 'closed'* Either side closed the transport #### failed *= 'failed'* The handshake or the fingerprint check failed ### *class* webrtc.enums.RTCSctpTransportState The state of an SCTP transport. See [MDN: RTCSctpTransport/state](https://developer.mozilla.org/en-US/docs/Web/API/RTCSctpTransport/state). #### connecting *= 'connecting'* The association is being set up #### connected *= 'connected'* The association is up, and channels can open #### closed *= 'closed'* The association was closed ### *class* webrtc.enums.RTCDataChannelState The state of a data channel. See [MDN: RTCDataChannel/readyState](https://developer.mozilla.org/en-US/docs/Web/API/RTCDataChannel/readyState). #### connecting *= 'connecting'* The channel is being opened, and messages can’t be sent yet #### open *= 'open'* Messages can be sent and received #### closing *= 'closing'* The channel will close once the messages already queued are sent #### closed *= 'closed'* The channel was closed, or couldn’t be opened ### *class* webrtc.enums.RTCPriorityType The priority of a data channel or an encoding, relative to the others of the connection. Each level gets twice the bandwidth share of the one below it. See [MDN: RTCRtpSender/setParameters](https://developer.mozilla.org/en-US/docs/Web/API/RTCRtpSender/setParameters). #### very_low *= 'very-low'* The lowest priority, with half the share of `low` #### low *= 'low'* The default priority #### medium *= 'medium'* Twice the share of `low` #### high *= 'high'* Four times the share of `low` ### webrtc.enums.RTCPriorityTypeValue The string values of [`RTCPriorityType`](#webrtc.enums.RTCPriorityType), which parameters accept in place of its members alias of [`Literal`](https://docs.python.org/3/library/typing.html#typing.Literal)[‘very-low’, ‘low’, ‘medium’, ‘high’] ### *class* webrtc.enums.RTCDegradationPreference What a video sender gives up first when bandwidth or CPU runs short. See [MDN: RTCRtpSender/setParameters](https://developer.mozilla.org/en-US/docs/Web/API/RTCRtpSender/setParameters). #### maintain_framerate *= 'maintain-framerate'* Keeps the frame rate and lowers the resolution #### maintain_resolution *= 'maintain-resolution'* Keeps the resolution and lowers the frame rate #### balanced *= 'balanced'* Lowers both, as the native WebRTC engine sees fit #### maintain_framerate_and_resolution *= 'maintain-framerate-and-resolution'* Keeps both and lowers only the quality of the encoding ### webrtc.enums.RTCDegradationPreferenceValue The string values of [`RTCDegradationPreference`](#webrtc.enums.RTCDegradationPreference), which parameters accept in place of its members alias of [`Literal`](https://docs.python.org/3/library/typing.html#typing.Literal)[‘maintain-framerate’, ‘maintain-resolution’, ‘balanced’, ‘maintain-framerate-and-resolution’] ### *class* webrtc.enums.RTCIceTransportPolicy Which ICE candidates a connection may use. See [MDN: RTCPeerConnection/RTCPeerConnection](https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/RTCPeerConnection). #### all *= 'all'* Any candidate #### relay *= 'relay'* Only candidates relayed through a TURN server, which hides the host’s addresses ### webrtc.enums.RTCIceTransportPolicyValue The string values of [`RTCIceTransportPolicy`](#webrtc.enums.RTCIceTransportPolicy), which parameters accept in place of its members alias of [`Literal`](https://docs.python.org/3/library/typing.html#typing.Literal)[‘all’, ‘relay’] ### *class* webrtc.enums.RTCBundlePolicy How media sections share transports when the remote peer may not support bundling. See [MDN: RTCPeerConnection/RTCPeerConnection](https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/RTCPeerConnection). #### balanced *= 'balanced'* A transport per media kind until the remote peer accepts bundling #### max_compat *= 'max-compat'* A transport per media section until the remote peer accepts bundling #### max_bundle *= 'max-bundle'* One transport for every media section, which needs a remote peer that supports bundling ### webrtc.enums.RTCBundlePolicyValue The string values of [`RTCBundlePolicy`](#webrtc.enums.RTCBundlePolicy), which parameters accept in place of its members alias of [`Literal`](https://docs.python.org/3/library/typing.html#typing.Literal)[‘balanced’, ‘max-compat’, ‘max-bundle’] ### *class* webrtc.enums.RTCRtcpMuxPolicy Whether RTCP shares the transport of RTP. See [MDN: RTCPeerConnection/RTCPeerConnection](https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/RTCPeerConnection). #### negotiate *= 'negotiate'* RTCP may use its own transport; removed from the specification #### require *= 'require'* RTCP shares the transport of RTP, and a remote description without RTCP multiplexing fails ### webrtc.enums.RTCRtcpMuxPolicyValue The string values of [`RTCRtcpMuxPolicy`](#webrtc.enums.RTCRtcpMuxPolicy), which parameters accept in place of its members alias of [`Literal`](https://docs.python.org/3/library/typing.html#typing.Literal)[‘negotiate’, ‘require’] ### *class* webrtc.enums.RTCRtpHeaderEncryptionPolicy Whether RTP header extensions are encrypted with cryptex (RFC 9335). #### disable *= 'disable'* Never encrypts them #### negotiate *= 'negotiate'* Encrypts them when the remote peer supports it #### require *= 'require'* Always encrypts them, so a remote description without cryptex fails ### webrtc.enums.RTCRtpHeaderEncryptionPolicyValue The string values of [`RTCRtpHeaderEncryptionPolicy`](#webrtc.enums.RTCRtpHeaderEncryptionPolicy), which parameters accept in place of its members alias of [`Literal`](https://docs.python.org/3/library/typing.html#typing.Literal)[‘disable’, ‘negotiate’, ‘require’] ### *class* webrtc.enums.RTCIceCandidateType Where the address of an ICE candidate comes from. See [MDN: RTCIceCandidate/type](https://developer.mozilla.org/en-US/docs/Web/API/RTCIceCandidate/type). #### host *= 'host'* An address of a local network interface #### srflx *= 'srflx'* The public address a STUN server saw (server reflexive) #### prflx *= 'prflx'* An address learned from the checks of the remote peer (peer reflexive) #### relay *= 'relay'* An address allocated on a TURN server ### *class* webrtc.enums.RTCIceProtocol The transport protocol of an ICE candidate. See [MDN: RTCIceCandidate/protocol](https://developer.mozilla.org/en-US/docs/Web/API/RTCIceCandidate/protocol). #### udp *= 'udp'* UDP #### tcp *= 'tcp'* TCP ### *class* webrtc.enums.RTCIceTcpCandidateType How a TCP ICE candidate opens its connections. See [MDN: RTCIceCandidate/tcpType](https://developer.mozilla.org/en-US/docs/Web/API/RTCIceCandidate/tcpType). #### active *= 'active'* Opens connections but doesn’t accept them #### passive *= 'passive'* Accepts connections but doesn’t open them #### so *= 'so'* Both sides try to open the connection at once (simultaneous open) ### *class* webrtc.enums.RTCIceServerTransportProtocol The protocol between the client and the TURN server of a relay candidate. See [MDN: RTCIceCandidateStats/relayProtocol](https://developer.mozilla.org/en-US/docs/Web/API/RTCIceCandidateStats/relayProtocol). #### udp *= 'udp'* UDP #### tcp *= 'tcp'* TCP #### tls *= 'tls'* TLS over TCP ### webrtc.enums.RTCIceServerTransportProtocolValue The string values of [`RTCIceServerTransportProtocol`](#webrtc.enums.RTCIceServerTransportProtocol), which parameters accept in place of its members alias of [`Literal`](https://docs.python.org/3/library/typing.html#typing.Literal)[‘udp’, ‘tcp’, ‘tls’] ### *class* webrtc.enums.RTCStatsType The type of a stats object, which tells which stats class it is. See [MDN: RTCStatsReport](https://developer.mozilla.org/en-US/docs/Web/API/RTCStatsReport). #### codec *= 'codec'* [`webrtc.RTCCodecStats`](models/rtc_stats.md#webrtc.models.rtc_stats.RTCCodecStats) #### inbound_rtp *= 'inbound-rtp'* [`webrtc.RTCInboundRtpStreamStats`](models/rtc_stats.md#webrtc.models.rtc_stats.RTCInboundRtpStreamStats) #### outbound_rtp *= 'outbound-rtp'* [`webrtc.RTCOutboundRtpStreamStats`](models/rtc_stats.md#webrtc.models.rtc_stats.RTCOutboundRtpStreamStats) #### remote_inbound_rtp *= 'remote-inbound-rtp'* [`webrtc.RTCRemoteInboundRtpStreamStats`](models/rtc_stats.md#webrtc.models.rtc_stats.RTCRemoteInboundRtpStreamStats) #### remote_outbound_rtp *= 'remote-outbound-rtp'* [`webrtc.RTCRemoteOutboundRtpStreamStats`](models/rtc_stats.md#webrtc.models.rtc_stats.RTCRemoteOutboundRtpStreamStats) #### media_source *= 'media-source'* [`webrtc.RTCAudioSourceStats`](models/rtc_stats.md#webrtc.models.rtc_stats.RTCAudioSourceStats) or [`webrtc.RTCVideoSourceStats`](models/rtc_stats.md#webrtc.models.rtc_stats.RTCVideoSourceStats) #### media_playout *= 'media-playout'* [`webrtc.RTCAudioPlayoutStats`](models/rtc_stats.md#webrtc.models.rtc_stats.RTCAudioPlayoutStats) #### peer_connection *= 'peer-connection'* [`webrtc.RTCPeerConnectionStats`](models/rtc_stats.md#webrtc.models.rtc_stats.RTCPeerConnectionStats) #### data_channel *= 'data-channel'* [`webrtc.RTCDataChannelStats`](models/rtc_stats.md#webrtc.models.rtc_stats.RTCDataChannelStats) #### transport *= 'transport'* [`webrtc.RTCTransportStats`](models/rtc_stats.md#webrtc.models.rtc_stats.RTCTransportStats) #### candidate_pair *= 'candidate-pair'* [`webrtc.RTCIceCandidatePairStats`](models/rtc_stats.md#webrtc.models.rtc_stats.RTCIceCandidatePairStats) #### local_candidate *= 'local-candidate'* [`webrtc.RTCIceCandidateStats`](models/rtc_stats.md#webrtc.models.rtc_stats.RTCIceCandidateStats) of a local candidate #### remote_candidate *= 'remote-candidate'* [`webrtc.RTCIceCandidateStats`](models/rtc_stats.md#webrtc.models.rtc_stats.RTCIceCandidateStats) of a remote candidate #### certificate *= 'certificate'* [`webrtc.RTCCertificateStats`](models/rtc_stats.md#webrtc.models.rtc_stats.RTCCertificateStats) ### *class* webrtc.enums.RTCQualityLimitationReason What limits the resolution or frame rate of a video sender the most. See [MDN: RTCOutboundRtpStreamStats/qualityLimitationReason](https://developer.mozilla.org/en-US/docs/Web/API/RTCOutboundRtpStreamStats/qualityLimitationReason). #### none *= 'none'* Nothing #### cpu *= 'cpu'* The CPU, which can’t encode fast enough #### bandwidth *= 'bandwidth'* The estimated bandwidth #### other *= 'other'* Something else ### *class* webrtc.enums.RTCDtlsRole The role of a DTLS transport in the handshake. See [MDN: RTCTransportStats/dtlsRole](https://developer.mozilla.org/en-US/docs/Web/API/RTCTransportStats/dtlsRole). #### client *= 'client'* Starts the handshake #### server *= 'server'* Answers the handshake #### unknown *= 'unknown'* Not decided yet ### *class* webrtc.enums.RTCStatsIceCandidatePairState The state of an ICE candidate pair in the checklist. See [MDN: RTCIceCandidatePairStats/state](https://developer.mozilla.org/en-US/docs/Web/API/RTCIceCandidatePairStats/state). #### frozen *= 'frozen'* Waits for a check of another pair of the same foundation #### waiting *= 'waiting'* Waits to be checked #### in_progress *= 'in-progress'* Being checked #### failed *= 'failed'* The check failed #### succeeded *= 'succeeded'* The check succeeded ### *class* webrtc.enums.RTCErrorDetailType The WebRTC-specific cause of an [`webrtc.RTCError`](exceptions.md#webrtc.exceptions.RTCError). See [MDN: RTCError/errorDetail](https://developer.mozilla.org/en-US/docs/Web/API/RTCError/errorDetail). #### data_channel_failure *= 'data-channel-failure'* A data channel failed (see [`webrtc.RTCError.sctp_cause_code`](exceptions.md#webrtc.exceptions.RTCError.sctp_cause_code)) #### dtls_failure *= 'dtls-failure'* The DTLS handshake failed or a DTLS alert was received #### fingerprint_failure *= 'fingerprint-failure'* The certificate of the remote peer doesn’t match the fingerprint of its description #### sctp_failure *= 'sctp-failure'* The SCTP association failed (see [`webrtc.RTCError.sctp_cause_code`](exceptions.md#webrtc.exceptions.RTCError.sctp_cause_code)) #### sdp_syntax_error *= 'sdp-syntax-error'* A description doesn’t parse (see [`webrtc.RTCError.sdp_line_number`](exceptions.md#webrtc.exceptions.RTCError.sdp_line_number)) #### hardware_encoder_not_available *= 'hardware-encoder-not-available'* A hardware encoder was required and none is available #### hardware_encoder_error *= 'hardware-encoder-error'* The hardware encoder failed ### webrtc.enums.RTCErrorDetailTypeValue The string values of [`webrtc.RTCErrorDetailType`](#webrtc.enums.RTCErrorDetailType), which parameters accept in place of its members alias of [`Literal`](https://docs.python.org/3/library/typing.html#typing.Literal)[‘data-channel-failure’, ‘dtls-failure’, ‘fingerprint-failure’, ‘sctp-failure’, ‘sdp-syntax-error’, ‘hardware-encoder-not-available’, ‘hardware-encoder-error’] ### *class* webrtc.enums.BinaryType What the binary messages of a [`webrtc.RTCDataChannel`](interfaces/rtc_data_channel.md#webrtc.interfaces.rtc_data_channel.RTCDataChannel) are delivered as. See [MDN: RTCDataChannel/binaryType](https://developer.mozilla.org/en-US/docs/Web/API/RTCDataChannel/binaryType). #### arraybuffer *= 'arraybuffer'* [`bytes`](https://docs.python.org/3/builtins/stdtypes.html#bytes) #### blob *= 'blob'* [`webrtc.Blob`](models/blob.md#webrtc.models.blob.Blob) ### webrtc.enums.BinaryTypeValue The string values of [`BinaryType`](#webrtc.enums.BinaryType), which parameters accept in place of its members alias of [`Literal`](https://docs.python.org/3/library/typing.html#typing.Literal)[‘arraybuffer’, ‘blob’] ### *class* webrtc.enums.EndingType How a [`webrtc.Blob`](models/blob.md#webrtc.models.blob.Blob) writes the line endings of its string parts. See [MDN: Blob/Blob](https://developer.mozilla.org/en-US/docs/Web/API/Blob/Blob). #### transparent *= 'transparent'* As they are #### native *= 'native'* As the platform’s, which are `\r\n` on Windows and `\n` elsewhere ### webrtc.enums.EndingTypeValue The string values of [`EndingType`](#webrtc.enums.EndingType), which parameters accept in place of its members alias of [`Literal`](https://docs.python.org/3/library/typing.html#typing.Literal)[‘transparent’, ‘native’] ### *class* webrtc.enums.VideoPixelFormat The layout of the pixels of a [`webrtc.VideoFrame`](models/video_frame.md#webrtc.models.video_frame.VideoFrame). `I4xx` formats are planar Y, U and V, plus alpha with `A`. Chroma is subsampled `4:2:0` in `I420` and `4:2:2` in `I422`, and has full resolution in `I444`. Samples are 8-bit, or 10- and 12-bit stored in 16 bits with `P10` and `P12`. See [MDN: VideoFrame/format](https://developer.mozilla.org/en-US/docs/Web/API/VideoFrame/format). #### I420 *= 'I420'* 8-bit Y, U, V, `4:2:0` #### I420P10 *= 'I420P10'* 10-bit Y, U, V, `4:2:0` #### I420P12 *= 'I420P12'* 12-bit Y, U, V, `4:2:0` #### I420A *= 'I420A'* 8-bit Y, U, V, alpha, `4:2:0` #### I420AP10 *= 'I420AP10'* 10-bit Y, U, V, alpha, `4:2:0` #### I420AP12 *= 'I420AP12'* 12-bit Y, U, V, alpha, `4:2:0` #### I422 *= 'I422'* 8-bit Y, U, V, `4:2:2` #### I422P10 *= 'I422P10'* 10-bit Y, U, V, `4:2:2` #### I422P12 *= 'I422P12'* 12-bit Y, U, V, `4:2:2` #### I422A *= 'I422A'* 8-bit Y, U, V, alpha, `4:2:2` #### I422AP10 *= 'I422AP10'* 10-bit Y, U, V, alpha, `4:2:2` #### I422AP12 *= 'I422AP12'* 12-bit Y, U, V, alpha, `4:2:2` #### I444 *= 'I444'* 8-bit Y, U, V, `4:4:4` #### I444P10 *= 'I444P10'* 10-bit Y, U, V, `4:4:4` #### I444P12 *= 'I444P12'* 12-bit Y, U, V, `4:4:4` #### I444A *= 'I444A'* 8-bit Y, U, V, alpha, `4:4:4` #### I444AP10 *= 'I444AP10'* 10-bit Y, U, V, alpha, `4:4:4` #### I444AP12 *= 'I444AP12'* 12-bit Y, U, V, alpha, `4:4:4` #### NV12 *= 'NV12'* 8-bit Y plane and a plane of interleaved U and V, `4:2:0` #### RGBA *= 'RGBA'* 8-bit red, green, blue and alpha per pixel #### RGBX *= 'RGBX'* 8-bit red, green, blue and an unused byte per pixel #### BGRA *= 'BGRA'* 8-bit blue, green, red and alpha per pixel #### BGRX *= 'BGRX'* 8-bit blue, green, red and an unused byte per pixel ### webrtc.enums.VideoPixelFormatValue The string values of [`VideoPixelFormat`](#webrtc.enums.VideoPixelFormat), which parameters accept in place of its members alias of [`Literal`](https://docs.python.org/3/library/typing.html#typing.Literal)[‘I420’, ‘I420P10’, ‘I420P12’, ‘I420A’, ‘I420AP10’, ‘I420AP12’, ‘I422’, ‘I422P10’, ‘I422P12’, ‘I422A’, ‘I422AP10’, ‘I422AP12’, ‘I444’, ‘I444P10’, ‘I444P12’, ‘I444A’, ‘I444AP10’, ‘I444AP12’, ‘NV12’, ‘RGBA’, ‘RGBX’, ‘BGRA’, ‘BGRX’] ### *class* webrtc.enums.VideoColorPrimaries The color primaries of a [`webrtc.VideoColorSpace`](models/video_frame.md#webrtc.models.video_frame.VideoColorSpace). See [MDN: VideoColorSpace/primaries](https://developer.mozilla.org/en-US/docs/Web/API/VideoColorSpace/primaries). #### bt709 *= 'bt709'* BT.709, used by HD video and sRGB #### bt470bg *= 'bt470bg'* BT.470 B/G, used by PAL video #### smpte170m *= 'smpte170m'* SMPTE 170M, used by NTSC video #### bt2020 *= 'bt2020'* BT.2020, used by UHD video #### smpte432 *= 'smpte432'* SMPTE EG 432-1, used by Display P3 ### webrtc.enums.VideoColorPrimariesValue The string values of [`VideoColorPrimaries`](#webrtc.enums.VideoColorPrimaries), which parameters accept in place of its members alias of [`Literal`](https://docs.python.org/3/library/typing.html#typing.Literal)[‘bt709’, ‘bt470bg’, ‘smpte170m’, ‘bt2020’, ‘smpte432’] ### *class* webrtc.enums.VideoTransferCharacteristics The transfer characteristics of a [`webrtc.VideoColorSpace`](models/video_frame.md#webrtc.models.video_frame.VideoColorSpace). See [MDN: VideoColorSpace/transfer](https://developer.mozilla.org/en-US/docs/Web/API/VideoColorSpace/transfer). #### bt709 *= 'bt709'* BT.709 #### smpte170m *= 'smpte170m'* SMPTE 170M #### iec61966_2_1 *= 'iec61966-2-1'* IEC 61966-2-1, the sRGB curve #### linear *= 'linear'* Linear light #### pq *= 'pq'* Perceptual quantizer (SMPTE ST 2084), used by HDR video #### hlg *= 'hlg'* Hybrid log-gamma (ARIB STD-B67), used by HDR video ### webrtc.enums.VideoTransferCharacteristicsValue The string values of [`VideoTransferCharacteristics`](#webrtc.enums.VideoTransferCharacteristics), which parameters accept in place of its members alias of [`Literal`](https://docs.python.org/3/library/typing.html#typing.Literal)[‘bt709’, ‘smpte170m’, ‘iec61966-2-1’, ‘linear’, ‘pq’, ‘hlg’] ### *class* webrtc.enums.VideoMatrixCoefficients The matrix coefficients of a [`webrtc.VideoColorSpace`](models/video_frame.md#webrtc.models.video_frame.VideoColorSpace), which convert between RGB and YUV. See [MDN: VideoColorSpace/matrix](https://developer.mozilla.org/en-US/docs/Web/API/VideoColorSpace/matrix). #### rgb *= 'rgb'* No conversion, because the samples are already RGB #### bt709 *= 'bt709'* BT.709 #### bt470bg *= 'bt470bg'* BT.470 B/G #### smpte170m *= 'smpte170m'* SMPTE 170M #### bt2020_ncl *= 'bt2020-ncl'* BT.2020, non-constant luminance ### webrtc.enums.VideoMatrixCoefficientsValue The string values of [`VideoMatrixCoefficients`](#webrtc.enums.VideoMatrixCoefficients), which parameters accept in place of its members alias of [`Literal`](https://docs.python.org/3/library/typing.html#typing.Literal)[‘rgb’, ‘bt709’, ‘bt470bg’, ‘smpte170m’, ‘bt2020-ncl’] ### *class* webrtc.enums.AlphaOption Whether a [`webrtc.VideoFrame`](models/video_frame.md#webrtc.models.video_frame.VideoFrame) created from another one keeps its alpha channel. See [MDN: VideoFrame/VideoFrame](https://developer.mozilla.org/en-US/docs/Web/API/VideoFrame/VideoFrame). #### keep *= 'keep'* Keeps the alpha channel #### discard *= 'discard'* Drops the alpha channel and uses the matching format without alpha ### webrtc.enums.AlphaOptionValue The string values of [`AlphaOption`](#webrtc.enums.AlphaOption), which parameters accept in place of its members alias of [`Literal`](https://docs.python.org/3/library/typing.html#typing.Literal)[‘keep’, ‘discard’] ### *class* webrtc.enums.PredefinedColorSpace The color space [`webrtc.VideoFrame.copy_to()`](models/video_frame.md#webrtc.models.video_frame.VideoFrame.copy_to) writes RGB pixels in. Only `srgb` is supported. See [MDN: VideoFrame/copyTo](https://developer.mozilla.org/en-US/docs/Web/API/VideoFrame/copyTo). #### srgb *= 'srgb'* sRGB #### srgb_linear *= 'srgb-linear'* sRGB primaries with linear light #### display_p3 *= 'display-p3'* Display P3 #### display_p3_linear *= 'display-p3-linear'* Display P3 primaries with linear light ### webrtc.enums.PredefinedColorSpaceValue The string values of [`PredefinedColorSpace`](#webrtc.enums.PredefinedColorSpace), which parameters accept in place of its members alias of [`Literal`](https://docs.python.org/3/library/typing.html#typing.Literal)[‘srgb’, ‘srgb-linear’, ‘display-p3’, ‘display-p3-linear’] ### *class* webrtc.enums.AudioSampleFormat The type of the samples of an [`webrtc.AudioData`](models/audio_data.md#webrtc.models.audio_data.AudioData), and whether channels are interleaved or planar. See [MDN: AudioData/format](https://developer.mozilla.org/en-US/docs/Web/API/AudioData/format). #### u8 *= 'u8'* Unsigned 8-bit, interleaved #### s16 *= 's16'* Signed 16-bit, interleaved #### s32 *= 's32'* Signed 32-bit, interleaved #### f32 *= 'f32'* 32-bit float, interleaved #### u8_planar *= 'u8-planar'* Unsigned 8-bit, a plane per channel #### s16_planar *= 's16-planar'* Signed 16-bit, a plane per channel #### s32_planar *= 's32-planar'* Signed 32-bit, a plane per channel #### f32_planar *= 'f32-planar'* 32-bit float, a plane per channel ### webrtc.enums.AudioSampleFormatValue The string values of [`AudioSampleFormat`](#webrtc.enums.AudioSampleFormat), which parameters accept in place of its members alias of [`Literal`](https://docs.python.org/3/library/typing.html#typing.Literal)[‘u8’, ‘s16’, ‘s32’, ‘f32’, ‘u8-planar’, ‘s16-planar’, ‘s32-planar’, ‘f32-planar’] ### *class* webrtc.enums.ReadableStreamReaderMode The type of reader [`webrtc.ReadableStream.get_reader()`](streams.md#webrtc.streams.ReadableStream.get_reader) returns. Its only member is rejected. See [MDN: ReadableStream/getReader](https://developer.mozilla.org/en-US/docs/Web/API/ReadableStream/getReader). #### byob *= 'byob'* A reader that fills a buffer of the caller. It needs a byte stream, and this library has none ### webrtc.enums.ReadableStreamReaderModeValue The string values of [`ReadableStreamReaderMode`](#webrtc.enums.ReadableStreamReaderMode), which parameters accept in place of its members alias of [`Literal`](https://docs.python.org/3/library/typing.html#typing.Literal)[‘byob’] ### *class* webrtc.enums.EncodedVideoChunkType Whether an encoded video frame is a key frame. See [MDN: RTCEncodedVideoFrame/type](https://developer.mozilla.org/en-US/docs/Web/API/RTCEncodedVideoFrame/type). #### key *= 'key'* A key frame, which decodes on its own #### delta *= 'delta'* A delta frame, which needs the frames before it ### *class* webrtc.enums.RTCRtpScriptTransformType What the worker of an [`webrtc.RTCRtpScriptTransform`](interfaces/rtc_rtp_script_transform.md#webrtc.interfaces.rtc_rtp_script_transform.RTCRtpScriptTransform) outputs. The value is accepted but has no effect. Frames are packetized as usual, because the native WebRTC engine has no SFrame packetization. #### sframe *= 'sframe'* The worker outputs SFrame-encrypted frames ### webrtc.enums.RTCRtpScriptTransformTypeValue The string values of [`RTCRtpScriptTransformType`](#webrtc.enums.RTCRtpScriptTransformType), which parameters accept in place of its members alias of [`Literal`](https://docs.python.org/3/library/typing.html#typing.Literal)[‘sframe’] ### *class* webrtc.enums.SFrameCipherSuite An SFrame cipher suite of RFC 9605 or draft-barnes-sframe-iana-256. The CTR suites authenticate with a truncated HMAC tag, whose bit length ends the name. The GCM suites use a 128-bit tag. #### AES_128_CTR_HMAC_SHA256_80 *= 'AES_128_CTR_HMAC_SHA256_80'* AES-128-CTR, HMAC-SHA-256, 80-bit tag #### AES_128_CTR_HMAC_SHA256_64 *= 'AES_128_CTR_HMAC_SHA256_64'* AES-128-CTR, HMAC-SHA-256, 64-bit tag #### AES_128_CTR_HMAC_SHA256_32 *= 'AES_128_CTR_HMAC_SHA256_32'* AES-128-CTR, HMAC-SHA-256, 32-bit tag #### AES_128_GCM_SHA256_128 *= 'AES_128_GCM_SHA256_128'* AES-128-GCM, SHA-256 key derivation, 128-bit tag #### AES_256_GCM_SHA512_128 *= 'AES_256_GCM_SHA512_128'* AES-256-GCM, SHA-512 key derivation, 128-bit tag #### AES_256_CTR_HMAC_SHA512_80 *= 'AES_256_CTR_HMAC_SHA512_80'* AES-256-CTR, HMAC-SHA-512, 80-bit tag #### AES_256_CTR_HMAC_SHA512_64 *= 'AES_256_CTR_HMAC_SHA512_64'* AES-256-CTR, HMAC-SHA-512, 64-bit tag #### AES_256_CTR_HMAC_SHA512_32 *= 'AES_256_CTR_HMAC_SHA512_32'* AES-256-CTR, HMAC-SHA-512, 32-bit tag ### webrtc.enums.SFrameCipherSuiteValue The string values of [`SFrameCipherSuite`](#webrtc.enums.SFrameCipherSuite), which parameters accept in place of its members alias of [`Literal`](https://docs.python.org/3/library/typing.html#typing.Literal)[‘AES_128_CTR_HMAC_SHA256_80’, ‘AES_128_CTR_HMAC_SHA256_64’, ‘AES_128_CTR_HMAC_SHA256_32’, ‘AES_128_GCM_SHA256_128’, ‘AES_256_GCM_SHA512_128’, ‘AES_256_CTR_HMAC_SHA512_80’, ‘AES_256_CTR_HMAC_SHA512_64’, ‘AES_256_CTR_HMAC_SHA512_32’] ### *class* webrtc.enums.SFrameType Whether an [`webrtc.RTCRtpSFrameEncryptor`](interfaces/sframe_transform.md#webrtc.interfaces.sframe_transform.RTCRtpSFrameEncryptor) encrypts whole frames or each RTP packet. #### per_frame *= 'per-frame'* Encrypts each frame before packetization #### per_packet *= 'per-packet'* Would encrypt each RTP packet, but isn’t supported because the native WebRTC engine has no SFrame packetization ### webrtc.enums.SFrameTypeValue The string values of [`SFrameType`](#webrtc.enums.SFrameType), which parameters accept in place of its members alias of [`Literal`](https://docs.python.org/3/library/typing.html#typing.Literal)[‘per-frame’, ‘per-packet’] ### *class* webrtc.enums.SFrameTransformErrorEventType Why an SFrame decryptor dropped a frame. #### authentication *= 'authentication'* The frame didn’t authenticate with the key of its key id #### key_id *= 'keyID'* No key is set for the key id of the frame #### syntax *= 'syntax'* The frame isn’t valid SFrame ### webrtc.enums.SFrameTransformErrorEventTypeValue The string values of [`SFrameTransformErrorEventType`](#webrtc.enums.SFrameTransformErrorEventType), which parameters accept in place of its members alias of [`Literal`](https://docs.python.org/3/library/typing.html#typing.Literal)[‘authentication’, ‘keyID’, ‘syntax’]