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Comparing version 0.6.1 to 0.7.0

177

CHANGELOG.md

@@ -0,1 +1,8 @@

# 0.7.0
- Updated to WebRTC M96.
- Added `target` property to event objects
- Support for multiple sinks per audio source
- Fixed RTCRtpSender::GetParameters
# 0.6.1

@@ -24,3 +31,3 @@

The previous changelog entries correspond to releases of the [`wrtc`](https://github.com/node-webrtc/node-webrtc)
The previous changelog entries correspond to releases of the [`wrtc`](https://github.com/node-webrtc/node-webrtc)
package from which `@astronautlabs/webrtc` was forked.

@@ -80,4 +87,4 @@

```js
const stream = new MediaStream({ id: 'foo' });
stream.id === 'foo'; // true
const stream = new MediaStream({ id: "foo" });
stream.id === "foo"; // true
```

@@ -238,6 +245,6 @@

* Pass audio samples to RTCAudioSource via the `onData` method. Then use the
- Pass audio samples to RTCAudioSource via the `onData` method. Then use the
RTCAudioSource's `createTrack` method to create a local audio
MediaStreamTrack.
* Construct an RTCAudioSink from a local or remote audio MediaStreamTrack. The
- Construct an RTCAudioSink from a local or remote audio MediaStreamTrack. The
RTCAudioSink will emit a "data" event every time audio samples are received.

@@ -250,3 +257,3 @@ When you're finished, stop the RTCAudioSink by calling `stop`.

```js
const { RTCAudioSource, RTCAudioSink } = require('wrtc').nonstandard;
const { RTCAudioSource, RTCAudioSink } = require("wrtc").nonstandard;

@@ -258,7 +265,7 @@ const source = new RTCAudioSource();

const sampleRate = 8000;
const samples = new Int16Array(sampleRate / 100); // 10 ms of 16-bit mono audio
const samples = new Int16Array(sampleRate / 100); // 10 ms of 16-bit mono audio
const data = {
samples,
sampleRate
sampleRate,
};

@@ -271,3 +278,3 @@

sink.ondata = data => {
sink.ondata = (data) => {
// Do something with the received audio samples.

@@ -301,7 +308,7 @@ };

* Calling `createTrack` will return a local audio MediaStreamTrack whose source
- Calling `createTrack` will return a local audio MediaStreamTrack whose source
is the RTCAudioSource.
* Calling `onData` with RTCAudioData pushes a new audio samples to every
- Calling `onData` with RTCAudioData pushes a new audio samples to every
non-stopped local audio MediaStreamTrack created with `createTrack`.
* RTCAudioData should represent 10 ms worth of 16-bit audio samples.
- RTCAudioData should represent 10 ms worth of 16-bit audio samples.

@@ -319,8 +326,8 @@ #### RTCAudioSink

* RTCAudioSink's constructor accepts a local or remote audio MediaStreamTrack.
* As long as neither the RTCAudioSink nor the RTCAudioSink's MediaStreamTrack
- RTCAudioSink's constructor accepts a local or remote audio MediaStreamTrack.
- As long as neither the RTCAudioSink nor the RTCAudioSink's MediaStreamTrack
are stopped, the RTCAudioSink will raise a "data" event any time RTCAudioData
is received.
* The "data" event has all the properties of RTCAudioData.
* RTCAudioSink must be stopped by calling `stop`.
- The "data" event has all the properties of RTCAudioData.
- RTCAudioSink must be stopped by calling `stop`.

@@ -346,6 +353,6 @@ ### RTCVideoFrame `rotation`

* Pass [I420](https://wiki.videolan.org/YUV/#I420) frames to RTCVideoSource via
- Pass [I420](https://wiki.videolan.org/YUV/#I420) frames to RTCVideoSource via
the `onFrame` method. Then use RTCVideoSource's `createTrack` method to create
a local video MediaStreamTrack.
* Construct an RTCVideoSink from a local or remote video MediaStreamTrack. The
- Construct an RTCVideoSink from a local or remote video MediaStreamTrack. The
RTCVideoSink will emit a "frame" event every time an I420 frame is received.

@@ -358,3 +365,3 @@ When you're finished, stop the RTCVideoSink by calling `stop`.

```js
const { RTCVideoSource, RTCVideoSink } = require('wrtc').nonstandard;
const { RTCVideoSource, RTCVideoSink } = require("wrtc").nonstandard;

@@ -413,7 +420,7 @@ const source = new RTCVideoSource();

* Calling `createTrack` will return a local video MediaStreamTrack whose source
- Calling `createTrack` will return a local video MediaStreamTrack whose source
is the RTCVideoSource.
* Calling `onFrame` with an RTCVideoFrame pushes a new video frame to every
- Calling `onFrame` with an RTCVideoFrame pushes a new video frame to every
non-stopped local video MediaStreamTrack created with `createTrack`.
* An RTCVideoFrame represents an I420 frame.
- An RTCVideoFrame represents an I420 frame.

@@ -431,8 +438,8 @@ #### RTCVideoSink

* RTCVideoSink's constructor accepts a local or remote video MediaStreamTrack.
* As long as neither the RTCVideoSink nor the RTCVideoSink's MediaStreamTrack
- RTCVideoSink's constructor accepts a local or remote video MediaStreamTrack.
- As long as neither the RTCVideoSink nor the RTCVideoSink's MediaStreamTrack
are stopped, the RTCVideoSink will raise a "frame" event any time an
RTCVideoFrame is received.
* The "frame" event has a property, `frame`, of type RTCVideoFrame.
* RTCVideoSink must be stopped by calling `stop`.
- The "frame" event has a property, `frame`, of type RTCVideoFrame.
- RTCVideoSink must be stopped by calling `stop`.

@@ -447,3 +454,3 @@ #### `i420ToRgba` and `rgbaToI420`

```js
const { i420ToRgba, rgbaToI420 } = require('wrtc').nonstandard;
const { i420ToRgba, rgbaToI420 } = require("wrtc").nonstandard;

@@ -526,9 +533,9 @@ const width = 640;

```js
const { getUserMedia } = require('wrtc');
const { getUserMedia } = require("wrtc");
getUserMedia({
audio: true,
video: true
}).then(stream => {
stream.getTracks().forEach(track => stop());
video: true,
}).then((stream) => {
stream.getTracks().forEach((track) => stop());
});

@@ -550,12 +557,10 @@ ```

// Legacy API
pc.getStats(
response => { /* ... */ },
console.error
);
pc.getStats((response) => {
/* ... */
}, console.error);
// Standards-compliant API
pc.getStats().then(
report => { /* ... */ },
console.error
);
pc.getStats().then((report) => {
/* ... */
}, console.error);
```

@@ -567,4 +572,4 @@

* A non-standard RTCConfiguration option, `sdpSemantics`, and
* An environment variable, `SDP_SEMANTICS`.
- A non-standard RTCConfiguration option, `sdpSemantics`, and
- An environment variable, `SDP_SEMANTICS`.

@@ -576,6 +581,6 @@ Construct an RTCPeerConnection with `sdpSemantics` set to "unified-plan" or

```js
const { RTCPeerConnection } = require('wrtc');
const { RTCPeerConnection } = require("wrtc");
const pc = new RTCPeerConnection({
sdpSemantics: 'unified-plan' // default is "plan-b"
sdpSemantics: "unified-plan", // default is "plan-b"
});

@@ -620,16 +625,16 @@ ```

```js
const assert = require('assert');
const { MediaStream, RTCPeerConnection, RTCRtpTransceiver } = require('wrtc');
const assert = require("assert");
const { MediaStream, RTCPeerConnection, RTCRtpTransceiver } = require("wrtc");
const pc = new RTCPeerConnection({
sdpSemantics: 'unified-plan'
sdpSemantics: "unified-plan",
});
const t1 = pc.addTransceiver('audio', {
direction: 'recvonly'
const t1 = pc.addTransceiver("audio", {
direction: "recvonly",
});
const t2 = pc.addTransceiver(t1.receiver.track, {
direction: 'sendonly',
streams: [new MediaStream()]
direction: "sendonly",
streams: [new MediaStream()],
});

@@ -727,3 +732,3 @@ ```

```js
const { MediaStream } = require('wrtc');
const { MediaStream } = require("wrtc");

@@ -878,7 +883,7 @@ const stream1 = new MediaStream();

* `bundlePolicy`
* `iceCandidatePoolSize`
* `iceServers` (no support for OAuth yet)
* `iceTransportPolicy`
* `rtcpMuxPolicy`
- `bundlePolicy`
- `iceCandidatePoolSize`
- `iceServers` (no support for OAuth yet)
- `iceTransportPolicy`
- `rtcpMuxPolicy`

@@ -890,3 +895,3 @@ RTCConfiguration also accepts a non-standard property, `portRange`. This

```js
const { RTCPeerConnection } = require('wrtc');
const { RTCPeerConnection } = require("wrtc");

@@ -896,4 +901,4 @@ const pc = new RTCPeerConnection({

min: 10000, // defaults to 0
max: 20000 // defaults to 65535
}
max: 20000, // defaults to 65535
},
});

@@ -906,18 +911,18 @@ ```

* `getConfiguration`
* `setConfiguration`
- `getConfiguration`
- `setConfiguration`
RTCPeerConnection now supports the following properties:
* `canTrickleIceCandidates` (always returns `null` for now)
* `connectionState` (derived from `iceConnectionState`)
* `currentLocalDescription`
* `currentRemoteDescription`
* `pendingLocalDescription`
* `pendingRemoteDescription`
- `canTrickleIceCandidates` (always returns `null` for now)
- `connectionState` (derived from `iceConnectionState`)
- `currentLocalDescription`
- `currentRemoteDescription`
- `pendingLocalDescription`
- `pendingRemoteDescription`
RTCPeerConnection now supports the following events:
* "connectionstatechange"
* "negotiationneeded"
- "connectionstatechange"
- "negotiationneeded"

@@ -930,5 +935,5 @@ ### RTCOfferOptions and RTCAnswerOptions

* `iceRestart`
* `offerToReceiveAudio`
* `offerToReceiveVideo`
- `iceRestart`
- `offerToReceiveAudio`
- `offerToReceiveVideo`

@@ -941,8 +946,8 @@ Both RTCOfferOptions and RTCAnswerOptions support `voiceActivityDetection`.

* `id`
* `maxPacketLifeTime`
* `maxRetransmits`
* `negotiated`
* `ordered`
* `protocol`
- `id`
- `maxPacketLifeTime`
- `maxRetransmits`
- `negotiated`
- `ordered`
- `protocol`

@@ -953,7 +958,7 @@ ### RTCDataChannel

* `id`
* `maxRetransmits`
* `ordered`
* `priority` (always returns "high")
* `protocol`
- `id`
- `maxRetransmits`
- `ordered`
- `priority` (always returns "high")
- `protocol`

@@ -968,5 +973,5 @@ RTCDataChannel's `send` method now supports sending Blobs provided by

* RTCDataChannel
* RTCDataChannelEvent
* RTCPeerConnectionIceEvent
- RTCDataChannel
- RTCDataChannelEvent
- RTCPeerConnectionIceEvent

@@ -973,0 +978,0 @@ ## Bug Fixes

2

CMakeLists.txt

@@ -133,3 +133,3 @@ cmake_minimum_required(VERSION 3.12)

set(WEBRTC_REVISION branch-heads/4638) # M95 https://groups.google.com/g/discuss-webrtc/c/SfzpFc-dH-E/m/JHlMpLO1AAAJ
set(WEBRTC_REVISION branch-heads/4664) # M96 https://groups.google.com/g/discuss-webrtc/c/Bp8OzBzipSc/m/0AC4OGhdAgAJ

@@ -136,0 +136,0 @@ list(APPEND GN_GEN_ARGS

32

dist/browser/index.d.ts

@@ -7,3 +7,3 @@ export declare const MediaStream: {

};
export declare type MediaStream = typeof window.MediaStream;
export type MediaStream = typeof window.MediaStream;
export declare const MediaStreamTrack: {

@@ -13,3 +13,3 @@ new (): globalThis.MediaStreamTrack;

};
export declare type MediaStreamTrack = typeof window.MediaStreamTrack;
export type MediaStreamTrack = typeof window.MediaStreamTrack;
export declare const RTCDataChannel: {

@@ -19,3 +19,3 @@ new (): globalThis.RTCDataChannel;

};
export declare type RTCDataChannel = typeof window.RTCDataChannel;
export type RTCDataChannel = typeof window.RTCDataChannel;
export declare const RTCDataChannelEvent: {

@@ -25,3 +25,3 @@ new (type: string, eventInitDict: RTCDataChannelEventInit): globalThis.RTCDataChannelEvent;

};
export declare type RTCDataChannelEvent = typeof window.RTCDataChannelEvent;
export type RTCDataChannelEvent = typeof window.RTCDataChannelEvent;
export declare const RTCDtlsTransport: {

@@ -31,3 +31,3 @@ new (): globalThis.RTCDtlsTransport;

};
export declare type RTCDtlsTransport = typeof window.RTCDtlsTransport;
export type RTCDtlsTransport = typeof window.RTCDtlsTransport;
export declare const RTCIceCandidate: {

@@ -37,3 +37,3 @@ new (candidateInitDict?: RTCIceCandidateInit): globalThis.RTCIceCandidate;

};
export declare type RTCIceCandidate = typeof window.RTCIceCandidate;
export type RTCIceCandidate = typeof window.RTCIceCandidate;
export declare const RTCIceTransport: {

@@ -43,3 +43,3 @@ new (): globalThis.RTCIceTransport;

};
export declare type RTCIceTransport = typeof window.RTCIceTransport;
export type RTCIceTransport = typeof window.RTCIceTransport;
export declare const RTCPeerConnection: {

@@ -50,3 +50,3 @@ new (configuration?: RTCConfiguration): globalThis.RTCPeerConnection;

};
export declare type RTCPeerConnection = typeof window.RTCPeerConnection;
export type RTCPeerConnection = typeof window.RTCPeerConnection;
export declare const RTCPeerConnectionIceEvent: {

@@ -56,3 +56,3 @@ new (type: string, eventInitDict?: RTCPeerConnectionIceEventInit): globalThis.RTCPeerConnectionIceEvent;

};
export declare type RTCPeerConnectionIceEvent = typeof window.RTCPeerConnectionIceEvent;
export type RTCPeerConnectionIceEvent = typeof window.RTCPeerConnectionIceEvent;
export declare const RTCRtpReceiver: {

@@ -63,3 +63,3 @@ new (): globalThis.RTCRtpReceiver;

};
export declare type RTCRtpReceiver = typeof window.RTCRtpReceiver;
export type RTCRtpReceiver = typeof window.RTCRtpReceiver;
export declare const RTCRtpSender: {

@@ -70,3 +70,3 @@ new (): globalThis.RTCRtpSender;

};
export declare type RTCRtpSender = typeof window.RTCRtpSender;
export type RTCRtpSender = typeof window.RTCRtpSender;
export declare const RTCRtpTransceiver: {

@@ -76,3 +76,3 @@ new (): globalThis.RTCRtpTransceiver;

};
export declare type RTCRtpTransceiver = typeof window.RTCRtpTransceiver;
export type RTCRtpTransceiver = typeof window.RTCRtpTransceiver;
export declare const RTCSctpTransport: {

@@ -82,3 +82,3 @@ new (): globalThis.RTCSctpTransport;

};
export declare type RTCSctpTransport = typeof window.RTCSctpTransport;
export type RTCSctpTransport = typeof window.RTCSctpTransport;
export declare const RTCSessionDescription: {

@@ -88,6 +88,6 @@ new (descriptionInitDict: RTCSessionDescriptionInit): globalThis.RTCSessionDescription;

};
export declare type RTCSessionDescription = typeof window.RTCSessionDescription;
export type RTCSessionDescription = typeof window.RTCSessionDescription;
export declare const getUserMedia: any;
export declare type getUserMedia = typeof globalThis.getUserMedia;
export type getUserMedia = typeof globalThis.getUserMedia;
export declare const mediaDevices: MediaDevices;
export declare type mediaDevices = typeof globalThis.mediaDevices;
export type mediaDevices = typeof globalThis.mediaDevices;

2

dist/nodejs/audiosink.d.ts
export declare const RTCAudioSink: any;
export declare type RTCAudioSink = typeof RTCAudioSinkT;
export type RTCAudioSink = typeof RTCAudioSinkT;
declare class RTCAudioSinkT {
}
export {};

2

dist/nodejs/audiosource.d.ts
export declare const RTCAudioSource: typeof RTCAudioSourceT;
export declare type RTCAudioSource = RTCAudioSourceT;
export type RTCAudioSource = RTCAudioSourceT;
export interface RTCAudioData {

@@ -4,0 +4,0 @@ samples: Int16Array;

2

dist/nodejs/datachannel.d.ts
export declare const RTCDataChannel: typeof globalThis.RTCDataChannel;
export declare type RTCDataChannel = globalThis.RTCDataChannel;
export type RTCDataChannel = globalThis.RTCDataChannel;

2

dist/nodejs/dtlstransport.d.ts
import type { RTCIceTransport } from './icetransport';
export declare const RTCDtlsTransport: (typeof globalThis.RTCDtlsTransport);
export declare type RTCDtlsTransport = RTCDtlsTransportExtensions;
export type RTCDtlsTransport = RTCDtlsTransportExtensions;
declare class RTCDtlsTransportExtensions extends globalThis.RTCDtlsTransport {

@@ -5,0 +5,0 @@ iceTransport: RTCIceTransport;

1

dist/nodejs/eventemitter.js

@@ -25,2 +25,3 @@ "use strict";

let listeners = this._listeners = this._listeners || {};
event.target = this;
process.nextTick(() => {

@@ -27,0 +28,0 @@ listeners = new Set(listeners[event.type] || []);

1

dist/nodejs/eventtarget.js

@@ -22,2 +22,3 @@ "use strict";

let listeners = this._listeners = this._listeners || {};
event.target = this;
process.nextTick(() => {

@@ -24,0 +25,0 @@ listeners = new Set(listeners[event.type] || []);

2

dist/nodejs/getusermedia.d.ts
export declare const getUserMedia: typeof globalThis.getUserMedia;
export declare type getUserMedia = typeof globalThis.getUserMedia;
export type getUserMedia = typeof globalThis.getUserMedia;

2

dist/nodejs/icecandidate.d.ts

@@ -1,2 +0,2 @@

declare type NRTCIceCandidate = globalThis.RTCIceCandidate;
type NRTCIceCandidate = globalThis.RTCIceCandidate;
export declare class RTCIceCandidate implements NRTCIceCandidate {

@@ -3,0 +3,0 @@ /**

2

dist/nodejs/icetransport.d.ts
export declare const RTCIceTransport: (typeof globalThis.RTCIceTransport & typeof RTCIceTransportExtensions);
export declare type RTCIceTransport = RTCIceTransportExtensions;
export type RTCIceTransport = RTCIceTransportExtensions;
declare class RTCIceTransportExtensions extends globalThis.RTCIceTransport {

@@ -4,0 +4,0 @@ component: string;

14

dist/nodejs/index.d.ts

@@ -1,5 +0,5 @@

export * from './audiosink';
export * from './audiosource';
export * from './videosink';
export * from './videosource';
export * from "./audiosink";
export * from "./audiosource";
export * from "./videosink";
export * from "./videosource";
export * from "./datachannel";

@@ -9,4 +9,4 @@ export * from "./datachannelevent";

export * from "./peerconnection";
export * from './rtcpeerconnectioniceevent';
export * from './sessiondescription';
export * from "./rtcpeerconnectioniceevent";
export * from "./sessiondescription";
export * from "./mediastream";

@@ -21,3 +21,3 @@ export * from "./mediastreamtrack";

export * from "./getusermedia";
import { MediaDevices } from './mediadevices';
import { MediaDevices } from "./mediadevices";
export declare const mediaDevices: MediaDevices;

4

dist/nodejs/index.js

@@ -32,3 +32,3 @@ "use strict";

try {
native.setDOMException(require('domexception'));
native.setDOMException(require("domexception"));
}

@@ -62,3 +62,3 @@ catch (error) {

datachannel_1.RTCDataChannel.prototype.send = function send(data) {
const implSymbol = Object.getOwnPropertySymbols(data).find(symbol => symbol.toString() === 'Symbol(impl)');
const implSymbol = Object.getOwnPropertySymbols(data).find((symbol) => symbol.toString() === "Symbol(impl)");
if (data[implSymbol] && data[implSymbol]._buffer) {

@@ -65,0 +65,0 @@ data = data[implSymbol]._buffer;

2

dist/nodejs/mediastream.d.ts
export declare const MediaStream: (typeof globalThis.MediaStream & typeof MediaStreamExtensions);
export declare type MediaStream = globalThis.MediaStream;
export type MediaStream = globalThis.MediaStream;
declare class MediaStreamExtensions extends globalThis.MediaStream {

@@ -4,0 +4,0 @@ constructor(options: {

2

dist/nodejs/mediastreamtrack.d.ts
export declare const MediaStreamTrack: typeof globalThis.MediaStreamTrack;
export declare type MediaStreamTrack = globalThis.MediaStreamTrack;
export type MediaStreamTrack = globalThis.MediaStreamTrack;

2

dist/nodejs/rtpreceiver.d.ts
export declare const RTCRtpReceiver: typeof globalThis.RTCRtpReceiver;
export declare type RTCRtpReceiver = globalThis.RTCRtpReceiver;
export type RTCRtpReceiver = globalThis.RTCRtpReceiver;

2

dist/nodejs/rtpsender.d.ts
export declare const RTCRtpSender: typeof globalThis.RTCRtpSender;
export declare type RTCRtpSender = globalThis.RTCRtpSender;
export type RTCRtpSender = globalThis.RTCRtpSender;

2

dist/nodejs/rtptransceiver.d.ts
export declare const RTCRtpTransceiver: typeof globalThis.RTCRtpTransceiver;
export declare type RTCRtpTransceiver = globalThis.RTCRtpTransceiver;
export type RTCRtpTransceiver = globalThis.RTCRtpTransceiver;

2

dist/nodejs/sctptransport.d.ts
export declare const RTCSctpTransport: typeof globalThis.RTCSctpTransport;
export declare type RTCSctpTransport = globalThis.RTCSctpTransport;
export type RTCSctpTransport = globalThis.RTCSctpTransport;

2

dist/nodejs/videosink.d.ts
import { EventTarget } from './eventtarget';
export declare const RTCVideoSink: typeof RTCVideoSinkT;
export declare type RTCVideoSink = typeof RTCVideoSinkT;
export type RTCVideoSink = typeof RTCVideoSinkT;
export interface RTCVideoSinkEvent extends Event {

@@ -5,0 +5,0 @@ frame: any;

2

dist/nodejs/videosource.d.ts
export declare const RTCVideoSource: any;
export declare type RTCVideoSource = typeof RTCVideoSourceT;
export type RTCVideoSource = typeof RTCVideoSourceT;
declare class RTCVideoSourceT {
}
export {};

2

package.json

@@ -12,3 +12,3 @@ {

},
"version": "0.6.1",
"version": "0.7.0",
"author": "Alan K <ack@modeswitch.org> (http://blog.modeswitch.org)",

@@ -15,0 +15,0 @@ "contributors": [

8

README.md

@@ -5,3 +5,3 @@ # @cubicleai/wrtc

Node.js bindings for `libwebrtc`, which implements [WebRTC M95](https://groups.google.com/g/discuss-webrtc/c/SfzpFc-dH-E/m/JHlMpLO1AAAJ). This project aims for spec-compliance and is tested using the W3C's [web-platform-tests](https://github.com/web-platform-tests/wpt) project. A number of [nonstandard APIs](docs/nonstandard-apis.md) for testing are also included.
Node.js bindings for `libwebrtc`, which implements [WebRTC M96](https://groups.google.com/g/discuss-webrtc/c/Bp8OzBzipSc/m/0AC4OGhdAgAJ). This project aims for spec-compliance and is tested using the W3C's [web-platform-tests](https://github.com/web-platform-tests/wpt) project. A number of [nonstandard APIs](docs/nonstandard-apis.md) for testing are also included.

@@ -62,4 +62,4 @@ # Install

<td align="center"></td>
<td align="center">✓</td>
<td align="center"></td>
<td align="center"></td>
<td align="center">✓</td>

@@ -72,4 +72,4 @@ <td align="center">✓</td>

<td align="center"></td>
<td align="center">✓</td>
<td align="center"></td>
<td align="center"></td>
<td align="center">✓</td>

@@ -82,4 +82,4 @@ <td align="center">✓</td>

<td align="center"></td>
<td align="center">✓</td>
<td align="center"></td>
<td align="center"></td>
<td align="center">✓</td>

@@ -86,0 +86,0 @@ <td align="center">✓</td>

.circleci/config.yml
src/node/async_context_releaser.cc
src/node/async_context_releaser.h
src/node/async_object_wrap_with_loop.h
src/node/async_object_wrap.h
src/node/deferrer.cc
src/node/deferrer.h
src/node/error_factory.cc
src/node/error_factory.h
src/node/event_loop.h
src/node/event_queue.h
src/node/events.h
src/node/promise.h
src/node/utility.h
src/node/wrap.h
src/test.cc
src/test.h
tsconfig.json
dist/nodejs/eventemitter.js.map

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dist/nodejs/eventtarget.js.map

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dist/nodejs/index.js.map

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dist/nodejs/peerconnection.js.map

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src/interfaces/rtc_audio_source.h

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src/interfaces/rtc_peer_connection/peer_connection_factory.cc

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src/interfaces/rtc_rtp_sender.cc

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src/interfaces/rtc_rtp_sender.h

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