RTCRtpSendParameters
dictionary's
encodings
property is an
RTCRtpEncodingParameters
object providing configuration settings for the encoder being used for the
RTCRtpSender
's
track
.
sendParameters.encodings = encodingParameterList; encodingParameterList = sendParameters.encodings;
An array of objects conforming to the
RTCRtpEncodingParameters
dictionary, each of which contains properties which provide settings and parameters that describe and configure the codec used for a single destination. Each object's properties are:
active
true
, the described encoding is currently actively being used. That is, for RTP senders, the encoding is currently being used to send data, while for receivers, the encoding is being used to decode received data. The default value is
true
.
codecPayloadType
RTCRtpSender
,
codecPayloadType
is a single 8-bit byte (or octet) specifying the codec to use for sending the stream; the value matches one from the owning
RTCRtpParameters
对象的
codecs
parameter. This value can only be set when creating the transceiver; after that, this value is read only.
dtx
RTCRtpSender
whose
kind
is
audio
, this property indicates whether or not to use discontinuous transmission (a feature by which a phone is turned off or the microphone muted automatically in the absence of voice activity). The value is taken from the enumerated string type
RTCDtxStatus
.
maxBitrate
maxFramerate
or transport or physical network limitations.
maxFramerate
A double-precision floating-point value specifying the maximum number of frames per second to allow for this encoding.
ptime
An unsigned long integer value indicating the preferred duration of a media packet in milliseconds. This is typically only relevant for audio encodings. The user agent will try to match this as well as it can, but there is no guarantee.
rid
DOMString
which, if set, specifies an
RTP stream ID
(
RID
) to be sent using the RID header extension. This parameter cannot be modified using
setParameters()
. Its value can only be set when the transceiver is first created.
scaleResolutionDownBy
kind
is
视频
, this is a double-precision floating-point value specifying a factor by which to scale down the video during encoding. The default value, 1.0, means that the sent video's size will be the same as the original. A value of 2.0 scales the video frames down by a factor of 2 in each dimension, resulting in a video 1/4 the size of the original. The value must not be less than 1.0 (you can't use this to scale the video up).
In a connection in which there's only one remote peer, the
encodings
array will have just one object in it, describing the encoding to use when transmitting to that peer. For each peer you add the
RTCRtpSender
to, another entry is added to
encodings
to describe its configuration.
| 规范 | 状态 | 注释 |
|---|---|---|
|
WebRTC 1.0: Real-time Communication Between Browsers
The definition of 'RTCRtpSendParameters.encodings' in that specification. |
候选推荐 | 初始定义。 |
| 桌面 | 移动 | |||||||||||
|---|---|---|---|---|---|---|---|---|---|---|---|---|
encodings
|
Chrome 69 | Edge ≤79 |
Firefox
No
|
IE No | Opera ? | Safari ? | WebView Android 69 | Chrome Android 69 |
Firefox Android
No
|
Opera Android ? | Safari iOS ? | Samsung Internet Android 10.0 |
完整支持
不支持
兼容性未知
见实现注意事项。
RTCRtpSendParameters
MediaDevices.getUserMedia()
Navigator.mediaDevices
RTCCertificate
RTCDTMFSender
RTCDTMFToneChangeEvent
RTCDataChannel
RTCDataChannelEvent
RTCDtlsTransport
RTCErrorEvent
RTCIceCandidate
RTCIceTransport
RTCPeerConnection
RTCPeerConnectionIceErrorEvent
RTCPeerConnectionIceEvent
RTCRtpReceiver
RTCRtpSender
RTCRtpTransceiver
RTCSctpTransport
RTCSessionDescription
RTCStatsReport
RTCTrackEvent