WebRTC Encoded Transform

1. Introduction

The [WEBRTC-NV-USE-CASES] document describes the use-case of

• Untrusted JavaScript Cloud Conferencing

This specification provides access to encoded media, which is the output of the encoder part of a codec and the input to the decoder part of a codec which allows the user agent to apply encryption locally.

The interface is inspired by [WEBCODECS] to provide access to such functionality while retaining the setup flow of RTCPeerConnection

2. Specification

The Streams definition doesn’t use WebIDL much, but the WebRTC spec does. This specification shows the IDL extensions for WebRTC.

It uses an additional API on RTCRtpSender and RTCRtpReceiver to insert the processing into the pipeline.

typedef (SFrameTransform or RTCRtpScriptTransform) RTCRtpTransform;
// New methods for RTCRtpSender and RTCRtpReceiver
partial interface RTCRtpSender {
    attribute RTCRtpTransform? transform;
};
partial interface RTCRtpReceiver {
    attribute RTCRtpTransform? transform;
};

2.1. Extension operation

At the time when a codec is initialized as part of the encoder, and the corresponding flag is set in the RTCPeerConnection ’s RTCConfiguration argument, ensure that the codec is disabled and produces no output.

2.1.1. Stream creation

At construction of each RTCRtpSender or RTCRtpReceiver , run the following steps:

1. Initialize this . [[transform]] to null.

2. Initialize this . [[readable]] to a new ReadableStream .

3. Set up this . [[readable]] . this . [[readable]] is provided frames using the readEncodedData algorithm given this as parameter.

4. Initialize this . [[writable]] to a new WritableStream .

5. Set up this . [[writable]] with its writeAlgorithm set to writeEncodedData given this as parameter and its highWaterMark set to Infinity .

   highWaterMark is set to Infinity to explicitly disable backpressure.

6. Initialize this . [[pipeToController]] to null.

7. Initialize this . [[lastReceivedFrameCounter]] to 0 .

8. Initialize this . [[lastEnqueuedFrameCounter]] to 0 .

9. Queue a task to run the following steps:

   1. If this . [[pipeToController]] is not null, abort these steps.

   2. Set this . [[pipeToController]] to a new AbortController .

   3. Call pipeTo with this . [[readable]] , this . [[writable]] , preventClose equal to true, preventAbort equal to true, preventCancel equal to true and this . [[pipeToController]] ’s signal .

Streams backpressure can optimize throughput while limiting processing and memory consumption by pausing data production as early as possible in a data pipeline. This proves useful in contexts where reliability is essential and latency is less of a concern. On the other hand, WebRTC media pipelines favour low latency over reliability, for instance by allowing to drop frames at various places and by using recovery mechanisms. Buffering within a transform would add latency without allowing web applications to adapt much. The User Agent is responsible for doing these adaptations, especially since it controls both ends of the transform. For those reasons, streams backpressure is disabled in WebRTC encoded transforms.

2.1.2. Stream processing

The readEncodedData algorithm is given a rtcObject as parameter. It is defined by running the following steps:

1. Wait for a frame to be produced by rtcObject ’s encoder if it is a RTCRtpSender or rtcObject ’s packetizer if it is a RTCRtpReceiver .

2. Increment rtcObject . [[lastEnqueuedFrameCounter]] by 1 .

3. Let frame be the newly produced frame.

4. Set frame . [[owner]] to rtcObject .

5. Set this . [[writable]] to this . [[transform]] . [[writable]] .

6. Set frame . [[counter]] to rtcObject . [[lastEnqueuedFrameCounter]] .

7. If the frame has been produced by a RTCRtpReceiver :

   1. If the relevant RTP packet contains the RTP Header Extension for Absolute Capture Time , set frame . [[captureTime]] to the absolute capture timestamp field and set frame . [[senderCaptureTimeOffset]] to the capture clock offset field if it is present.

   2. Otherwise, if the relevant RTP packet does not contain the RTP Header Extension for Absolute Capture Time but a previous RTP packet did, set frame . [[captureTime]] to the result of calculating the absolute capture timestamp according to timestamp interpolation and set frame . [[senderCaptureTimeOffset]] to the most recent value that was present.

   3. Otherwise, set frame . [[captureTime]] to undefined and set frame . [[senderCaptureTimeOffset]] to undefined.

   4. If frame was produced by a SFrame depacketizer , set frame . [[useSFrame]] to true.

8. If the frame has been produced by a RTCRtpSender , set frame . [[captureTime]] to the capture timestamp using the methodology described in RTP Header Extension for Absolute Capture Time § absolute-capture-timestamp and set frame. [[senderCaptureTimeOffset]] to undefined.

9. Enqueue frame in rtcObject . [[readable]] .

The writeEncodedData algorithm is given a rtcObject as parameter and a frame as input. It is defined by running the following steps:

1. If frame . [[owner]] is not equal to rtcObject , abort these steps and return a promise resolved with undefined. A processor cannot create frames, or move frames between streams.

2. If frame . [[counter]] is equal or smaller than rtcObject . [[lastReceivedFrameCounter]] , abort these steps and return a promise resolved with undefined. A processor cannot reorder frames, although it may delay them or drop them.

3. Set rtcObject . [[lastReceivedFrameCounter]] to frame [[counter]] .

4. Let data be frame . [[data]] .

5. Let serializedFrame be StructuredSerializeWithTransfer ( frame , « data »).

6. Let frameCopy be StructuredDeserializeWithTransfer ( serializedFrame , frame ’s relevant realm ).

7. If frame . [[useSFrame]] is true, set frameCopy . [[useSFrame]] to true.

8. Enqueue frameCopy for processing as if it came directly from the encoded data source, by running one of the following steps:

   • If rtcObject is a RTCRtpSender , enqueue frameCopy to rtcObject ’s packetizer, to be processed in parallel . If frameCopy . [[useSFrame]] is true, rtcObject ’s MUST use a SFrame packetizer or skip processing of frameCopy .

   • If rtcObject is a RTCRtpReceiver , enqueue frameCopy it to rtcObject ’s decoder, to be processed in parallel .

9. Return a promise resolved with undefined.

On sender side, as part of readEncodedData , frames produced by rtcObject ’s encoder MUST be enqueued in rtcObject . [[readable]] in the encoder’s output order. As writeEncodedData ensures that the transform cannot reorder frames, the encoder’s output order is also the order followed by packetizers to generate RTP packets and assign RTP packet sequence numbers. The packetizer may expect the transformed data to still conform to the original format, e.g. a series of NAL units separated by Annex B start codes.

On receiver side, as part of readEncodedData , frames produced by rtcObject ’s packetizer MUST be enqueued in rtcObject . [[readable]] in the same encoder’s output order. To ensure the order is respected, the depacketizer will typically use RTP packet sequence numbers to reorder RTP packets as needed before enqueuing frames in rtcObject . [[readable]] . As writeEncodedData ensures that the transform cannot reorder frames, this will be the order expected by rtcObject ’s decoder.

2.2. Extension attribute

A RTCRtpTransform has private slots:

• [[readable]] of type ReadableStream .

• [[writable]] of type WritableStream .

• [[owner]] of type RTCRtpSender or RTCRtpReceiver , initialized to null.

• [[useSFrame]] of type boolean. // FIXME: Decide whether augmenting this boolean with either cipher suite support or whether doing frame vs. packet based encryption.

Each RTCRtpTransform has an association steps set, which is empty by default.

The transform getter steps are:

1. Return this . [[transform]] .

The transform setter steps are:

1. Let transform be the argument to the setter.

2. Let checkedTransform set to transform if it is not null or to an identity transform stream otherwise.

3. If checkedTransform . [[owner]] is not null, throw a InvalidStateError and abort these steps.

4. Let reader be the result of getting a reader for checkedTransform . [[readable]] .

5. Let writer be the result of getting a writer for checkedTransform . [[writable]] .

6. Set checkedTransform . [[owner]] to this .

7. Initialize newPipeToController to a new AbortController .

8. If this . [[pipeToController]] is not null, run the following steps:

   1. Add the chain transform algorithm to this . [[pipeToController]] ’s signal .

   2. signal abort on this . [[pipeToController]] .

9. Else, run the chain transform algorithm steps.

10. Set If this is a RTCRtpSender , run the following substeps:

    1. Let useSFrame be true if this is configured to use a SFrame packetizer and false otherwise.

    2. If useSFrame is equal to checkedTransform . [[useSFrame]] , abort these substeps.

    3. Configure this ’s packetizer to use SFrame if checkedTransform . [[useSFrame]] is true and to not use SFrame if checkedTransform . [[useSFrame]] is false.

    4. Update the negotiation-needed flag for this ’s connection.

11. Otherwise, run the following steps:

    1. Let useSFrame be true if this is configured to use a SFrame depacketizer and false otherwise.

    2. If useSFrame is equal to checkedTransform . [[useSFrame]] , abort these substeps.

    3. Configure this ’s depacketizer to use SFrame if checkedTransform . [[useSFrame]] is true and to not use SFrame if checkedTransform . [[useSFrame]] is false.

    4. Update the negotiation-needed flag for this ’s connection.

12. Set this . [[pipeToController]] to newPipeToController .

13. Set this . [[transform]] to transform .

14. Run the steps in the set of association steps of transform with this .

The chain transform algorithm steps are defined as:

1. If newPipeToController ’s signal is aborted , abort these steps.

2. Release reader .

3. Release writer .

4. Assert that newPipeToController is the same object as rtcObject . [[pipeToController]] .

5. Call pipeTo with rtcObject . [[readable]] , checkedTransform . [[writable]] , preventClose equal to false, preventAbort equal to false, preventCancel equal to true and newPipeToController ’s signal .

6. Call pipeTo with checkedTransform . [[readable]] , rtcObject . [[writable]] , preventClose equal to true, preventAbort equal to true, preventCancel equal to false and newPipeToController ’s signal .

This algorithm is defined so that transforms can be updated dynamically. There is no guarantee on which frame will happen the switch from the previous transform to the new transform.

If a web application sets the transform synchronously at creation of the RTCRtpSender (for instance when calling addTrack), the transform will receive the first frame generated by the RTCRtpSender ’s encoder. Similarly, if a web application sets the transform synchronously at creation of the RTCRtpReceiver (for instance when calling addTrack, or at track event handler), the transform will receive the first full frame generated by the RTCRtpReceiver ’s packetizer.

3. SFrameTransform

The API presented in this section allows applications to process SFrame data as defined in [SFrame] .

enum SFrameTransformRole {
    "encrypt",
    "decrypt"
};
dictionary SFrameTransformOptions {
    SFrameTransformRole role = "encrypt";
};
typedef [EnforceRange] unsigned long long SmallCryptoKeyID;
typedef (SmallCryptoKeyID or bigint) CryptoKeyID;
[Exposed=(Window,DedicatedWorker)]
interface SFrameTransform : EventTarget {
    constructor(optional SFrameTransformOptions options = {});
    Promise<undefined> setEncryptionKey(CryptoKey key, optional CryptoKeyID keyID);
    attribute EventHandler onerror;
};
SFrameTransform includes GenericTransformStream;
enum SFrameTransformErrorEventType {
    "authentication",
    "keyID",
    

    "syntax",
    "packetization"
};
[Exposed=(Window,DedicatedWorker)]
interface SFrameTransformErrorEvent : Event {
    constructor(DOMString type, SFrameTransformErrorEventInit eventInitDict);
    readonly attribute SFrameTransformErrorEventType errorType;
    readonly attribute CryptoKeyID? keyID;
    readonly attribute any frame;
};
dictionary SFrameTransformErrorEventInit : EventInit {
    required SFrameTransformErrorEventType errorType;
    required any frame;
    CryptoKeyID? keyID;
};

The new SFrameTransform( options ) constructor steps are:

1. Let transformAlgorithm be an algorithm which takes a frame as input and runs the SFrame transform algorithm with this and frame .

2. Set this . [[transform]] to a new TransformStream .

3. Set up this . [[transform]] with transformAlgorithm set to transformAlgorithm .

4. Let options be the method’s first argument.

5. Set this . [[role]] to options [" role "].

6. Set this . [[readable]] to this . [[transform]] . [[readable]] .

7. Set this . [[writable]] to this . [[transform]] . [[writable]] .

8. Set this . [[useSFrame]] to true.

3.1. Algorithm

The SFrame transform algorithm, given sframe as a SFrameTransform object and frame , runs these steps:

1. Let role be sframe . [[role]] .

2. If frame sframe . [[owner]] is a RTCRtpSender , set role to 'encrypt'.

3. If frame sframe . [[owner]] is a RTCRtpReceiver , set role to 'decrypt'.

4. If sframe . [[owner]] is a RTCRtpReceiver and frame . [[useSFrame]] is not true, queue a task to run the following steps:

   1. fire an event named error at sframe , using the SFrameTransformErrorEvent interface with its errorType attribute set to packetization and its frame attribute set to frame .

   2. Abort these steps.

5. Let data be undefined.

6. If frame is a BufferSource , set data to frame .

7. If frame is a RTCEncodedAudioFrame , set data to frame . data

8. If frame is a RTCEncodedVideoFrame , set data to frame . data

9. If data is undefined, abort these steps.

10. Let buffer be the result of running the SFrame algorithm with data and role as parameters. This algorithm is defined by the SFrame specification and returns an ArrayBuffer .

11. If the SFrame algorithm exits abruptly with an error, queue a task to run the following sub steps:

    1. If the processing fails on decryption side due to data not following the SFrame format, fire an event named error at sframe , using the SFrameTransformErrorEvent interface with its errorType attribute set to syntax and its frame attribute set to frame .

    2. If the processing fails on decryption side due to the key identifier parsed in data being unknown, fire an event named error at sframe , using the SFrameTransformErrorEvent interface with its errorType attribute set to keyID , its frame attribute set to frame and its keyID attribute set to the keyID value parsed in the SFrame header.

    3. If the processing fails on decryption side due to validation of the authentication tag, fire an event named error at sframe , using the SFrameTransformErrorEvent interface with its errorType attribute set to authentication and its frame attribute set to frame .

    4. Abort these steps.

12. If frame is a BufferSource , set frame to buffer .

13. If frame is a RTCEncodedAudioFrame , set frame . data to buffer .

14. If frame is a RTCEncodedVideoFrame , set frame . data to buffer .

15. Set frame . [[useSFrame]] to true.

16. Enqueue frame in sframe . [[transform]] .

3.2. Methods

1. Let promise be a new promise .

2. If keyID is a bigint which cannot be represented as a integer between 0 and 2 ⁶⁴ -1 inclusive, reject promise with a RangeError exception.

3. Otherwise, in parallel , run the following steps:

   1. Set key with its optional keyID as key material to use for the SFrame transform algorithm, as defined by the SFrame specification .

   2. If setting the key material fails, reject promise with an InvalidModificationError exception and abort these steps.

   3. Resolve promise with undefined.

4. Return promise .

3.3. SFrame packetization integration

A SFrame packetizer is responsible to generate SFrame packets from media content. In the context of this specification, the SFrame packetizer is not responsible for doing the actual encryption. Instead, the transform is responsible for doing so. The SFrame packetizer is responsible for splitting SFrame frames as needed so that they fit in RTP packets.

Similarly, a SFrame depacketizer is responsible to assemble RTP packets into a complete SFrame frame. It is not responsible for doing the actual decryption, the transform is responsible for doing so.

WebRTC encoded transform model is a per frame processing. SFrame can either be applied on each frame or on subframes. WebRTC encoded transform model is naturally aligned with aplying SFrame on a frame as a whole. To preserve WebRTC encoded transform model when applying SFrame on subframes, the following conceptual steps can be done:

1. On sending side, the SFrameTransform may first split the media frame in subframes like would do a regular media packetizer, and apply the SFrame encryption on each subframe. It then concatenates the encrypted subframes as a unique encrypted frame. The transform provides the encrypted frame and information of each subframe so that the SFrame packetizer generates individual packets for each subframe.

2. On receiving side, the SFrame depacketizer assembles all individual subframe RTP packets as a unique encrypted frame. It is responsible to give the necessary subframe information to the transform so that the transform can apply the SFrame decryption on each individual subframe contained in the unique encrypted frame and concatenate each decrypted subframe as a unique decrypted media frame. If decryption of a single subframe fails, the whole encrypted frame is discarded.

4. RTCRtpScriptTransform

In this section, the capture system refers to the system where media is sourced from and the sender system refers to the system that is sending RTP and RTCP packets to the receiver system where RTCEncodedFrameMetadata data is populated.

4.1. RTCEncodedFrameMetadata dictionary

dictionary RTCEncodedFrameMetadata {
    unsigned long synchronizationSource;
    octet payloadType;
    sequence<unsigned long> contributingSources;
    unsigned long rtpTimestamp;
    DOMHighResTimeStamp receiveTime;
    DOMHighResTimeStamp captureTime;
    DOMHighResTimeStamp senderCaptureTimeOffset;
    DOMString mimeType;
};

4.1.1. Members

synchronizationSource , of type unsigned long unsigned long

The synchronization source (ssrc) identifier is an unsigned integer value per [RFC3550] used to identify the stream of RTP packets that the encoded frame object is describing.

payloadType , of type octet octet

The payload type is an unsigned integer value in the range from 0 to 127 per [RFC3550] that is used to describe the format of the RTP payload.

contributingSources , of type sequence<unsigned long> sequence<unsigned long>

The list of contribution sources (csrc list) as defined in [RFC3550] .

rtpTimestamp , of type unsigned long unsigned long

The RTP timestamp identifier is an unsigned integer value per [RFC3550] that reflects the sampling instant of the first octet in the RTP data packet.

receiveTime , of type DOMHighResTimeStamp DOMHighResTimeStamp

For frames coming from an RTCRtpReceiver, represents the timestamp of the last received packet used to produce this media frame. This timestamp is relative to Performance . timeOrigin .

captureTime , of type DOMHighResTimeStamp DOMHighResTimeStamp

The capture time of this frame in the capture system’s clock. On populating this member, the user agent MUST return the value of the frame’s [[captureTime]] slot, shifted to be relative to Performance . timeOrigin .

senderCaptureTimeOffset , of type DOMHighResTimeStamp DOMHighResTimeStamp

The senderCaptureTimeOffset is the sender system’s estimate of the offset between its own NTP clock and the capture system’s NTP clock, for the same frame that the captureTime was originated from. On populating this member, the user agent MUST return the value of the frame’s [[senderCaptureTimeOffset]] slot.

mimeType , of type DOMString DOMString

The codec MIME media type/subtype defined in the IANA media types registry [IANA-MEDIA-TYPES] , e.g. audio/opus or video/VP8.

4.2. RTCEncodedVideoFrameType dictionary

// New enum for video frame types. Will eventually re-use the equivalent defined
// by WebCodecs.
enum RTCEncodedVideoFrameType {
    "empty",
    "key",
    "delta",
};

4.3. RTCEncodedVideoFrameMetadata dictionary

dictionary RTCEncodedVideoFrameMetadata : RTCEncodedFrameMetadata {
    unsigned long long frameId;
    sequence<unsigned long long> dependencies;
    unsigned short width;
    unsigned short height;
    unsigned long spatialIndex;
    unsigned long temporalIndex;
    long long timestamp;    // microseconds
};

4.3.1. Members

frameId , of type unsigned long long unsigned long long

An identifier for the encoded frame, monotonically increasing in decode order. Its lower 16 bits match the frame_number of the AV1 Dependency Descriptor Header Extension defined in Appendix A of [AV1-RTP-SPEC] , if present. Only present for received frames if the Dependency Descriptor Header Extension is present.

dependencies , of type sequence<unsigned long long> sequence<unsigned long long>

List of frameIds of frames this frame references. Only present for received frames if the AV1 Dependency Descriptor Header Extension defined in Appendix A of [AV1-RTP-SPEC] is present.

timestamp , of type long long long long

The media presentation timestamp (PTS) in microseconds of raw frame, matching the timestamp for raw frames which correspond to this frame.

4.4. RTCEncodedVideoFrame interface

dictionary RTCEncodedVideoFrameOptions {
    RTCEncodedVideoFrameMetadata metadata;
};
// New interfaces to define encoded video and audio frames. Will eventually
// re-use or extend the equivalent defined in WebCodecs.
[Exposed=(Window,DedicatedWorker), Serializable]
interface RTCEncodedVideoFrame {
    constructor(RTCEncodedVideoFrame originalFrame, optional RTCEncodedVideoFrameOptions options = {});
    readonly attribute RTCEncodedVideoFrameType type;
    attribute ArrayBuffer data;
    RTCEncodedVideoFrameMetadata getMetadata();
};

4.4.1. Constructor

constructor()

Creates a new RTCEncodedVideoFrame from the given originalFrame and options . [metadata] . The newly created frame is completely independent of originalFrame , with its [[data]] being a deep copy of originalFrame . [[data]] . The new frame’s [[metadata]] is a deep copy of originalFrame . [[metadata]] , with fields replaced with deep copies of the fields present in options . [metadata] .

When called, run the following steps:

1. Set this. [[type]] to originalFrame . [[type]] .

2. Let this. [[data]] be the result of [CloneArrayBuffer] ( originalFrame . [[data]] , 0, originalFrame . [[data]] . [[ArrayBufferByteLength]] ).

3. Let [[metadata]] represent the metadata associated with this newly constructed frame.

   1. For each { [[key]] , [[value]] } pair of originalFrame . [[getMetadata()]] , set [[metadata]] . [[key]] to a deep copy of [[value]] .

   2. For each { [[key]] , [[value]] } pair of options . [metadata] , set [[metadata]] . [[key]] to a deep copy of [[value]] .

4.4.2. Members

type , of type RTCEncodedVideoFrameType , readonly RTCEncodedVideoFrameType

The type attribute allows the application to determine when a key frame is being sent or received.

data , of type ArrayBuffer ArrayBuffer

The encoded frame data. The format of the data depends on the video codec that is used to encode/decode the frame which can be determined by looking at the mimeType . For SVC , each spatial layer is transformed separately.

Since packetizers may drop certain elements, e.g. AV1 temporal delimiter OBUs, the input to a receive-side transform may be different from the output of a send-side transform.

The following table gives a number of examples:

mimeType	Data format
video/VP8	The data starts with the "uncompressed data chunk" defined in section 9.1 of [RFC6386] and is followed by the rest of the frame data. The VP8 payload descriptor is not accessible.
video/VP9	The data is a frame as described in Section 6 of [VP9] . The VP9 payload descriptor is not accessible.
video/H264	The data is a series of NAL units in Annex B format, as defined in [ITU-T-REC-H.264] Annex B.
video/AV1	The data is a series of OBUs compliant to the low-overhead bitstream format as described in Section 5 of [AV1] . The AV1 aggregation header is not accessible.

4.4.3. Methods

getMetadata()

Returns the metadata associated with the frame.

4.4.4. Serialization

RTCEncodedVideoFrame objects are serializable objects . Their serialization steps , given value , serialized , and forStorage , are:

1. If forStorage is true, then throw a DataCloneError .

2. Set serialized . [[type]] to the value of value . type .

3. Set serialized . [[metadata]] to an internal representation of value ’s metadata.

4. Set serialized . [[data]] to the sub-serialization of value . [[data]] .

Their deserialization steps , given serialized , value and realm , are:

1. Set value . type to serialized . [[type]] .

2. Set value ’s metadata to the platform object representation of serialized . [[metadata]] .

3. Set value . [[data]] to the sub-deserialization of serialized . [[data]] .

The internal form of a serialized RTCEncodedVideoFrame is not observable; it is defined chiefly so that it can be used with frame cloning in the writeEncodedData algorithm and in the structuredClone() operation. An implementation is therefore free to choose whatever method works best.

4.5. RTCEncodedAudioFrameMetadata dictionary

dictionary RTCEncodedAudioFrameMetadata : RTCEncodedFrameMetadata {
    short sequenceNumber;
    double audioLevel;
};

4.5.1. Members

sequenceNumber , of type short short

The RTP sequence number as defined in [RFC3550] . Only exists for incoming audio frames.

Comparing two sequence numbers requires serial number arithmetic described in [RFC1982] .

audioLevel , of type double double

The audio level of this frame. The value is between 0..1 (linear), where 1.0 represents 0 dBov, 0 represents silence, and 0.5 represents approximately 6 dBSPL change in the sound pressure level from 0 dBov.

4.6. RTCEncodedAudioFrame interface

dictionary RTCEncodedAudioFrameOptions {
    RTCEncodedAudioFrameMetadata metadata;
};
[Exposed=(Window,DedicatedWorker), Serializable]
interface RTCEncodedAudioFrame {
    constructor(RTCEncodedAudioFrame originalFrame, optional RTCEncodedAudioFrameOptions options = {});
    attribute ArrayBuffer data;
    RTCEncodedAudioFrameMetadata getMetadata();
};

4.6.1. Constructor

constructor()

Creates a new RTCEncodedAudioFrame from the given originalFrame and options . [metadata] . The newly created frame is completely independent of originalFrame , with its [[data]] being a deep copy of originalFrame . [[data]] . The new frame’s [[metadata]] is a deep copy of originalFrame . [[metadata]] , with fields replaced with deep copies of the fields present in options . [metadata] .

When called, run the following steps:

1. Let this. [[data]] be the result of [CloneArrayBuffer] ( originalFrame . [[data]] , 0, originalFrame . [[data]] . [[ArrayBufferByteLength]] ).

2. Let [[metadata]] represent the metadata associated with this newly constructed frame.

   1. For each { [[key]] , [[value]] } pair of originalFrame . [[getMetadata()]] , set [[metadata]] . [[key]] to a deep copy of [[value]] .

   2. For each { [[key]] , [[value]] } pair of options . [metadata] , set [[metadata]] . [[key]] to a deep copy of [[value]] .

4.6.2. Members

data , of type ArrayBuffer ArrayBuffer

The encoded frame data. The format of the data depends on the audio codec that is used to encode/decode the frame which can be determined by looking at the mimeType . The following table gives a number of examples:

mimeType	Data format
audio/opus	The data is Opus packets, as described in section 3 of [RFC6716] .
audio/PCMU	The data is a sequence of bytes of arbitrary length, where each byte is a u-law encoded PCM sample as defined by Table 2a and 2b in [ITU-G.711] .
audio/PCMA	The data is a sequence of bytes of arbitrary length, where each byte is an A-law encoded PCM sample as defined by Tables 1a and 1b in [ITU-G.711] .
audio/G722	The data is G.722 audio as described in [ITU-G.722] .
audio/RED	The data is Redundant Audio Data as described in section 3 of [RFC2198] .
audio/CN	The data is Comfort Noise as described in section 3 of [RFC3389] .

4.6.3. Methods

getMetadata()

Returns the metadata associated with the frame.

4.6.4. Serialization

RTCEncodedAudioFrame objects are serializable objects . Their serialization steps , given value , serialized , and forStorage , are:

1. If forStorage is true, then throw a DataCloneError .

2. Set serialized . [[metadata]] to an internal representation of value ’s metadata.

3. Set serialized . [[data]] to the sub-serialization of value . [[data]] .

Their deserialization steps , given serialized , value and realm , are:

1. Set value ’s metadata to the platform object representation of serialized . [[metadata]]

2. Set value . [[data]] to the sub-deserialization of serialized . [[data]] .

4.7. Interfaces

[Exposed=DedicatedWorker]
interface RTCTransformEvent : Event {
    readonly attribute RTCRtpScriptTransformer transformer;
};
partial interface DedicatedWorkerGlobalScope {
    attribute EventHandler onrtctransform;
};
[Exposed=DedicatedWorker]
interface RTCRtpScriptTransformer : EventTarget {
    // Attributes and methods related to the transformer source
    readonly attribute ReadableStream readable;
    Promise<unsigned long long> generateKeyFrame(optional DOMString rid);
    Promise<undefined> sendKeyFrameRequest();
    // Attributes and methods related to the transformer sink
    readonly attribute WritableStream writable;
    attribute EventHandler onkeyframerequest;
    // Attributes for configuring the Javascript code
    readonly attribute any options;
};
enum RTCRtpScriptTransformType {
    "sframe"
};

[Exposed=Window]
interface RTCRtpScriptTransform {
    );

    constructor(Worker worker, optional any options, optional sequence<object> transfer, optional RTCRtpScriptTransformType type);
};
[Exposed=DedicatedWorker]
interface KeyFrameRequestEvent : Event {
  constructor(DOMString type, optional DOMString rid);
  readonly attribute DOMString? rid;
};

4.8. Operations

The new RTCRtpScriptTransform( worker , options , transfer , type ) constructor steps are:

1. Set t1 to an identity transform stream .

2. Set t2 to an identity transform stream .

3. Set this . [[writable]] to t1 . [[writable]] .

4. Set this . [[readable]] to t2 . [[readable]] .

5. If type is equal to "sframe" , set this . [[useSFrame]] to true.

6. Let serializedOptions be the result of StructuredSerializeWithTransfer ( options , transfer ).

7. Let serializedReadable be the result of StructuredSerializeWithTransfer ( t1 . [[readable]] , « t1 . [[readable]] »).

8. Let serializedWritable be the result of StructuredSerializeWithTransfer ( t2 . [[writable]] , « t2 . [[writable]] »).

9. Queue a task on the DOM manipulation task source worker ’s global scope to run the following steps:

   1. Let transformerOptions be the result of StructuredDeserializeWithTransfer ( serializedOptions , the current Realm).

   2. Let readable be the result of StructuredDeserializeWithTransfer ( serializedReadable , the current Realm).

   3. Let writable be the result of StructuredDeserializeWithTransfer ( serializedWritable , the current Realm).

   4. Let transformer be a new RTCRtpScriptTransformer .

   5. Set transformer . [[options]] to transformerOptions .

   6. Set transformer . [[readable]] to readable .

   7. Set transformer . [[writable]] to writable .

   8. Fire an event named rtctransform using RTCTransformEvent with transformer set to transformer on worker ’s global scope.

// FIXME: Describe error handling (worker closing flag true at RTCRtpScriptTransform creation time. And worker being terminated while transform is processing data).

Each RTCRtpScriptTransform has the following set of association steps , given rtcObject :

1. Let transform be the RTCRtpScriptTransform object that owns the association steps .

2. Let encoder be rtcObject ’s encoder if rtcObject is a RTCRtpSender or undefined otherwise.

3. Let depacketizer be rtcObject ’s depacketizer if rtcObject is a RTCRtpReceiver or undefined otherwise.

4. Queue a task on the DOM manipulation task source worker ’s global scope to run the following steps:

   1. Let transformer be the RTCRtpScriptTransformer object associated to transform .

   2. Set transformer . [[encoder]] to encoder .

   3. Set transformer . [[depacketizer]] to depacketizer .

The generateKeyFrame( rid ) method steps are:

1. Let promise be a new promise.

2. Run the generate key frame algorithm with promise , this . [[encoder]] and rid .

3. Return promise .

The sendKeyFrameRequest() method steps are:

1. Let promise be a new promise.

2. Run the send request key frame algorithm with promise and this . [[depacketizer]] .

3. Return promise .

4.9. Attributes

A RTCRtpScriptTransformer has the following private slots called [[depacketizer]] , [[encoder]] , [[options]] , [[readable]] and [[writable]] . In addition, a RTCRtpScriptTransformer is always associated with its parent RTCRtpScriptTransform transform. This allows algorithms to go from an RTCRtpScriptTransformer object to its RTCRtpScriptTransform parent and vice versa.

The options getter steps are:

1. Return this . [[options]] .

The readable getter steps are:

1. Return this . [[readable]] .

The writable getter steps are:

1. Return this . [[writable]] .

The onbandwidthestimate EventHandler has type bandwidthestimate.

The onkeyframerequest EventHandler has type keyframerequest.

4.10. Events

The following event fires on an RTCRtpScriptTransformer :

• keyframerequest of type KeyFrameRequestEvent - fired when the sink determines that a key frame has been requested.

The steps that generate an event of type KeyFrameRequestEvent are as follows:

Given a RTCRtpScriptTransformer transform :

When transform ’s [[encoder]] receives a keyframe request, for instance from an incoming RTCP Picture Loss Indication (PLI) or Full Intra Refresh (FIR), queue a task to perform the following steps:

1. Set rid to the RID of the appropriate layer, or undefined if the request is not for a specific layer.

2. Fire an event named keyframerequest at transform using KeyFrameRequestEvent with its cancelable attribute initialized to "true", and with rid set to rid .

3. If the event’s canceled flag is true, abort these steps.

4. Run the generate key frame algorithm with a new promise, transform . [[encoder]] and rid .

4.11. KeyFrame Algorithms

The generate key frame algorithm , given promise , encoder and rid , is defined by running these steps:

1. If encoder is undefined, reject promise with InvalidStateError , abort these steps.

2. If encoder is not processing video frames, reject promise with InvalidStateError , abort these steps.

3. If rid is defined, but does not conform to the grammar requirements specified in Section 10 of [RFC8851] , then reject promise with TypeError and abort these steps.

4. In parallel , run the following steps:

   1. Gather a list of video encoders, named videoEncoders from encoder , ordered according negotiated RIDs if any.

   2. If rid is defined, remove from videoEncoders any video encoder that does not match rid .

   3. If rid is undefined, remove from videoEncoders all video encoders except the first one.

   4. If videoEncoders is empty, queue a task to reject promise with NotFoundError and abort these steps. videoEncoders is expected to be empty if the corresponding RTCRtpSender is not active, or the corresponding RTCRtpSender track is ended.

   5. Let videoEncoder be the first encoder in videoEncoders .

   6. If rid is undefined, set rid to the RID value corresponding to videoEncoder .

   7. Create a pending key frame task called task with task . [[rid]] set to rid and task . [[promise]] | set to promise .

   8. If encoder . [[pendingKeyFrameTasks]] is undefined, initialize encoder . [[pendingKeyFrameTasks]] to an empty set.

   9. Let shouldTriggerKeyFrame be false if encoder . [[pendingKeyFrameTasks]] contains a task whose [[rid]] value is equal to rid , and true otherwise.

   10. Add task to encoder . [[pendingKeyFrameTasks]] .

   11. If shouldTriggerKeyFrame is true , instruct videoEncoder to generate a key frame for the next provided video frame.

For any RTCRtpScriptTransformer named transformer , the following steps are run just before any frame is enqueued in transformer . [[readable]] :

1. Let encoder be transformer . [[encoder]] .

2. If encoder or encoder . [[pendingKeyFrameTasks]] is undefined, abort these steps.

3. If frame is not a video "key" frame, abort these steps.

4. For each task in encoder . [[pendingKeyFrameTasks]] , run the following steps:

   1. If frame was generated by a video encoder identified by task . [[rid]] , run the following steps:

      1. Remove task from encoder . [[pendingKeyFrameTasks]] .

      2. Resolve task . [[promise]] with frame ’s timestamp.

By resolving the promises just before enqueuing the corresponding key frame in a RTCRtpScriptTransformer ’s readable, the resolution callbacks of the promises are always executed just before the corresponding key frame is exposed. If the promise is associated to several rid values, it will be resolved when the first key frame corresponding to one the rid value is enqueued.

The send request key frame algorithm , given promise and depacketizer , is defined by running these steps:

1. If depacketizer is undefined, reject promise with InvalidStateError , abort these steps.

2. If depacketizer does not belong to a video RTCRtpReceiver , reject promise with InvalidStateError , abort these steps.

3. In parallel , run the following steps:

   1. If sending a Full Intra Request (FIR) by depacketizer ’s receiver is not deemed appropriate, resolve promise with undefined and abort these steps. Section 4.3.1 of [RFC5104] provides guidelines of how and when it is appropriate to sending a Full Intra Request.

   2. Generate a Full Intra Request (FIR) packet as defined in section 4.3.1 of [RFC5104] and send it through depacketizer ’s receiver.

   3. Queue a task to resolve promise with undefined.

5. RTCRtpSender extension

An additional API on RTCRtpSender is added to complement the generation of key frame added to RTCRtpScriptTransformer .

partial interface RTCRtpSender {
    Promise<undefined> generateKeyFrame(optional sequence <DOMString> rids);
};

5.1. Extension operation

The generateKeyFrame( rids ) method steps are:

1. Let promise be a new promise.

2. In parallel , run the generate key frame algorithm with promise , this ’s encoder and rids .

3. Return promise .

6. Privacy and security considerations

This API gives Javascript access to the content of media streams. This is also available from other sources, such as Canvas and WebAudio.

However, streams that are isolated (as specified in [WEBRTC-IDENTITY] ) or tainted with another origin, cannot be accessed using this API, since that would break the isolation rule.

The API will allow access to some aspects of timing information that are otherwise unavailable, which allows some fingerprinting surface.

The API will give access to encoded media, which means that the JS application will have full control over what’s delivered to internal components like the packetizer or the decoder. This may require additional care with auditing how data is handled inside these components.

For instance, packetizers may expect to see data only from trusted encoders, and may not be audited for reception of data from untrusted sources.

7. Examples

See the explainer document .