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Allow developers to control the audio codecs (#1190)
Co-authored-by: Ammar Ansari <[email protected]>
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-331
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‎.changeset/cute-zebras-grin.md‎

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@@ -0,0 +1,10 @@
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---
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'@signalwire/webrtc': minor
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'@signalwire/js': minor
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---
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Added dial params:
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- audioCodecs
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The new parameter allows developers to select the audio codec to use with the desired configuration

‎internal/e2e-js/tests/roomSessionStreamingAPI.spec.ts‎

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@@ -52,7 +52,6 @@ test.describe('Room Streaming from REST API', () => {
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const STREAM_CHECK_URL = process.env.STREAM_CHECK_URL!
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await pageTwo.goto(STREAM_CHECK_URL, { waitUntil: 'domcontentloaded' })
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await pageTwo.waitForSelector(`text=${streamName}`, { timeout: 10_000 })
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await deleteRoom(roomData.id)
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})
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})

‎internal/playground-js/src/fabric/index.html‎

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@@ -26,6 +26,7 @@
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href="https://signalwire.com/assets/images/favicon.ico"
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/>
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</head>
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<body class="bg-light">
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<div class="container">
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<div class="row pt-5">
@@ -104,7 +105,23 @@ <h5>Connect</h5>
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</label>
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</div>
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</div>
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<div class="d-grid gap-2">
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<div class="card">
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<div class="card-body">
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<div class="form-group">
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<label for="targetCodec">Audio Codec Override</label>
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<textarea
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type="text"
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rows="3"
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class="form-control"
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id=""
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placeholder="opus:maxplaybackrate=8000;maxaveragebitrate=4000\nPCMU:"
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onchange="saveInLocalStorage(event)"
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></textarea>
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</div>
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</div>
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</div>
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<div class="d-grid gap-2 mt-3">
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<button
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id="btnConnect"
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class="btn btn-primary"

‎internal/playground-js/src/fabric/index.js‎

Lines changed: 14 additions & 2 deletions
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@@ -307,14 +307,20 @@ window.dial = async ({ reattach = false } = {}) => {
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const dialer = reattach ? client.reattach : client.dial
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310-
const call = await dialer({
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const dialOptions = {
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nodeId: steeringId,
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to: document.getElementById('destination').value,
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rootElement: document.getElementById('rootElement'),
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fromFabricAddressId: document.getElementById('fromFabricAddressId').value,
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video: document.getElementById('video').checked,
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audio: document.getElementById('audio').checked,
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})
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}
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if (document.getElementById('audioCodec').value.trim().length) {
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const userInput = document.getElementById('audioCodec').value.trim()
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dialOptions.audioCodecs = userInput.split('\n').map((codec) => codec.trim())
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}
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const call = await dialer(dialOptions)
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window.__call = call
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roomObj = call
@@ -910,6 +916,12 @@ window.ready(async function () {
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document.getElementById('audio').checked = true
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document.getElementById('video').checked =
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localStorage.getItem('fabric.ws.video') === 'true'
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document.getElementById('targetCodec').value =
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localStorage.getItem('fabric.ws.targetCodec') || ''
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document.getElementById('maxPlaybackRate').value =
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localStorage.getItem('fabric.ws.maxPlaybackRate') || ''
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document.getElementById('maxAverageBitrate').value =
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localStorage.getItem('fabric.ws.maxAverageBitrate') || ''
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const urlParams = new URLSearchParams(window.location.search)
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const room = urlParams.get('room')

‎package-lock.json‎

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‎packages/core/src/types/utils.ts‎

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@@ -147,3 +147,11 @@ export type Promisify<T> = {
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export type Prettify<T> = NonNullable<unknown> & {
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[K in keyof T]: Prettify<T[K]>
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} & {}
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export type AudioCodecOptions =
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| `opus:${string}`
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| `red:${string}`
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| `G722:${string}`
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| `PCMU:${string}`
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| `PCMA:${string}`
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| `telephone-event:${string}`

‎packages/js/src/fabric/WSClient.ts‎

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@@ -195,6 +195,7 @@ export class WSClient extends BaseClient<{}> implements WSClientContract {
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disableUdpIceServers: params.disableUdpIceServers || false,
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userVariables: params.userVariables || this.wsClientOptions.userVariables,
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fromFabricAddressId: params.fromFabricAddressId,
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audioCodecs: params.audioCodecs,
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})
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// WebRTC connection left the room.
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prevCallId: payload.callID,
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disableUdpIceServers: params.disableUdpIceServers || false,
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userVariables: params.userVariables || this.wsClientOptions.userVariables,
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audioCodecs: params.audioCodecs
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})
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// WebRTC connection left the room.

‎packages/js/src/fabric/interfaces/wsClient.ts‎

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@@ -1,4 +1,8 @@
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import { SessionOptions, UserOptions } from '@signalwire/core'
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import {
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AudioCodecOptions,
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SessionOptions,
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UserOptions,
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} from '@signalwire/core'
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import {
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IncomingCallHandler,
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IncomingCallHandlers,
@@ -117,6 +121,8 @@ export interface CallParams extends DefaultCallParams {
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stopMicrophoneWhileMuted?: boolean
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/** Fabric address ID matching one of the subscriber’s addresses to attribute conversation API events in the INVITE. */
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fromFabricAddressId?: string
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/** Audio codec with fmtp params */
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audioCodecs?: AudioCodecOptions[]
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}
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export interface DialParams extends CallParams {

‎packages/webrtc/package.json‎

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@@ -38,7 +38,10 @@
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},
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"dependencies": {
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"@signalwire/core": "4.3.0-dev.202506111650.ad2f5be.0",
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"sdp": "^3.2.0"
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"sdp-transform": "2.15.0"
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},
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"types": "dist/cjs/webrtc/src/index.d.ts"
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"types": "dist/cjs/webrtc/src/index.d.ts",
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"devDependencies": {
45+
"@types/sdp-transform": "^2.4.9"
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}
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}

‎packages/webrtc/src/RTCPeer.ts‎

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@@ -5,10 +5,11 @@ import {
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filterIceServers,
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} from './utils/helpers'
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import {
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sdpStereoHack,
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sdpBitrateHack,
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sdpMediaOrderHack,
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sdpHasValidCandidates,
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useAudioCodecs,
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sdpStereoHack,
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} from './utils/sdpHelpers'
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import { BaseConnection } from './BaseConnection'
1415
import {
@@ -71,6 +72,8 @@ export default class RTCPeer<EventTypes extends EventEmitter.ValidEventTypes> {
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this.options
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)
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75+
this._validateOptions()
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this._onIce = this._onIce.bind(this)
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this._onEndedTrackHandler = this._onEndedTrackHandler.bind(this)
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@@ -86,6 +89,16 @@ export default class RTCPeer<EventTypes extends EventEmitter.ValidEventTypes> {
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this.rtcConfigPolyfill = this.config
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}
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92+
private _validateOptions() {
93+
if (
94+
this.options.useStereo === true &&
95+
typeof this.options.audio === 'object' &&
96+
(this.options.audio.channelCount ?? 2) !== 2
97+
) {
98+
throw new Error('Mismatch params: useStereo, audio.channelCount')
99+
}
100+
}
101+
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get options() {
90103
return this.call.options
91104
}
@@ -820,6 +833,7 @@ export default class RTCPeer<EventTypes extends EventEmitter.ValidEventTypes> {
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googleMaxBitrate,
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googleMinBitrate,
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googleStartBitrate,
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audioCodecs,
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} = this.options
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if (localDescription.sdp && useStereo) {
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localDescription.sdp = sdpStereoHack(localDescription.sdp)
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googleStartBitrate
838852
)
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}
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// this.logger.debug(
841-
// 'LOCAL SDP \n',
842-
// `Type: ${localDescription.type}`,
843-
// '\n\n',
844-
// localDescription.sdp
845-
// )
854+
if (localDescription.sdp && audioCodecs) {
855+
localDescription.sdp = useAudioCodecs(localDescription.sdp, audioCodecs)
856+
}
846857
return this.instance.setLocalDescription(localDescription)
847858
}
848859

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private _setRemoteDescription(remoteDescription: RTCSessionDescriptionInit) {
850-
if (remoteDescription.sdp && this.options.useStereo) {
861+
const { useStereo } = this.options
862+
if (remoteDescription.sdp && useStereo) {
851863
remoteDescription.sdp = sdpStereoHack(remoteDescription.sdp)
852864
}
853865
if (remoteDescription.sdp && this.instance.localDescription) {

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