Allow developers to control the audio codecs

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

@@ -307,13 +307,28 @@ window.dial = async ({ reattach = false } = {}) => {
 
   const dialer = reattach ? client.reattach : client.dial
 
-  const call = await dialer({
+  const dialOptions = {
     nodeId: steeringId,
     to: document.getElementById('destination').value,
     rootElement: document.getElementById('rootElement'),
     video: document.getElementById('video').checked,
     audio: document.getElementById('audio').checked,
-  })
+  }
+
+  if (document.getElementById('targetCodec').value.trim().length) {
+    const codecPrefix = document.getElementById('targetCodec').value
+    if (document.getElementById('maxPlaybackRate').value.trim().length) {
+      dialOptions[`${codecPrefix}MaxPlaybackRate`] = parseInt(
+        document.getElementById('maxPlaybackRate').value
+      )
+    }
+    if (document.getElementById('maxAverageBitrate').value.trim().length) {
+      dialOptions[`${codecPrefix}maxAverageBitrate`] = parseInt(
+        document.getElementById('maxAverageBitrate').value
+      )
+    }
+  }
+  const call = await dialer(dialOptions)
 
   window.__call = call
   roomObj = call
@@ -888,6 +903,12 @@ window.ready(async function () {
   document.getElementById('audio').checked = true
   document.getElementById('video').checked =
     localStorage.getItem('fabric.ws.video') === 'true'
+  document.getElementById('targetCodec').value =
+    localStorage.getItem('fabric.ws.targetCodec') || ''
+  document.getElementById('maxPlaybackRate').value =
+    localStorage.getItem('fabric.ws.maxPlaybackRate') || ''
+  document.getElementById('maxAverageBitrate').value =
+    localStorage.getItem('fabric.ws.maxAverageBitrate') || ''
 
   const urlParams = new URLSearchParams(window.location.search)
   const room = urlParams.get('room')

package.json

@@ -34,6 +34,7 @@
   },
   "devDependencies": {
     "@types/node": "^20.3.2",
+    "@types/sdp-transform": "^2.4.9",
     "dotenv": "^16.3.1",
     "jest-websocket-mock": "^2.5.0",
     "mock-socket": "^9.3.1",

packages/core/src/index.ts

@@ -23,7 +23,7 @@ import {
   asyncRetry,
   increasingDelay,
   decreasingDelay,
-  constDelay
+  constDelay,
 } from './utils'
 import { WEBRTC_EVENT_TYPES, isWebrtcEventType } from './utils/common'
 import { BaseSession } from './BaseSession'
@@ -81,7 +81,7 @@ export {
   asyncRetry,
   increasingDelay,
   decreasingDelay,
-  constDelay
+  constDelay,
 }
 
 export * from './redux/features/component/componentSlice'

packages/core/src/types/utils.ts

@@ -147,3 +147,15 @@ export type Promisify<T> = {
 export type Prettify<T> = NonNullable<unknown> & {
   [K in keyof T]: Prettify<T[K]>
 } & {}
+
+export enum PlaybackRate {
+  hz_8000 = 8000,
+  hz_12000 = 12000,
+  hz_16000 = 16000,
+  hz_24000 = 24000,
+  hz_48000 = 48000,
+}
+
+export const isPlaybackRate = (value: unknown): value is PlaybackRate => {
+  return Object.values(PlaybackRate).includes(value as PlaybackRate)
+}

packages/core/src/utils/index.ts

@@ -227,3 +227,4 @@ export const isJSONRPCResponse = (
 export const isSATAuth = (e?: Authorization): e is SATAuthorization => {
   return typeof e !== 'undefined' && 'jti' in e
 }
+

packages/js/src/fabric/WSClient.ts

@@ -5,6 +5,7 @@ import {
   VertoBye,
   VertoSubscribe,
   VideoRoomSubscribedEventParams,
+  isPlaybackRate
 } from '@signalwire/core'
 import { MakeRoomOptions } from '../video'
 import { createFabricRoomSessionObject } from './FabricRoomSession'
@@ -154,12 +155,39 @@ export class WSClient extends BaseClient<{}> implements WSClientContract {
     return room
   }
 
-  private buildOutboundCall(params: DialParams & { attach?: boolean }) {
-    const [pathname, query] = params.to.split('?')
+  private _validateDialParams(params: DialParams) {
+    const [pathname] = params.to.split('?')
+
     if (!pathname) {
       throw new Error('Invalid destination address')
     }
 
+    if (params.opusMaxPlaybackRate) {
+      if (!isPlaybackRate(params.opusMaxPlaybackRate)) {
+        throw new Error('Invalid opusMaxPlaybackRate')
+      }
+      if (typeof params.audio === 'object') {
+        if (
+          params.audio?.sampleRate &&
+          params.audio?.sampleRate !== params.opusMaxPlaybackRate
+        ) {
+          throw new Error(
+            'Mismatching parameters: opusMaxPlaybackRate, audio.sampleRate'
+          )
+        }
+      }
+    }
+
+    if (params.opusMaxAverageBitrate && params.opusMaxAverageBitrate <= 0) {
+      throw new Error('Invalid opusMaxPlaybackRate')
+    }
+  }
+
+  private buildOutboundCall(params: DialParams & { attach?: boolean }) {
+    this._validateDialParams(params)
+
+    const [, query] = params.to.split('?')
+
     let video = false
     let negotiateVideo = false
 
@@ -186,6 +214,8 @@ export class WSClient extends BaseClient<{}> implements WSClientContract {
       attach: params.attach ?? false,
       disableUdpIceServers: params.disableUdpIceServers || false,
       userVariables: params.userVariables || this.wsClientOptions.userVariables,
+      opusMaxPlaybackRate: params.opusMaxPlaybackRate,
+      opusMaxAverageBitrate: params.opusMaxAverageBitrate,
     })
 
     // WebRTC connection left the room.

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

@@ -1,4 +1,4 @@
-import { UserOptions } from '@signalwire/core'
+import { PlaybackRate, UserOptions } from '@signalwire/core'
 import { IncomingCallHandlers } from './incomingCallManager'
 import { FabricRoomSession } from '../FabricRoomSession'
 import { ApiRequestRetriesOptions } from '../SATSession'
@@ -94,6 +94,10 @@ export interface CallParams {
   negotiateVideo?: boolean
   /** User & UserAgent metadata */
   userVariables?: WSClientOptions['userVariables']
+  /** OPUS audio codec max playback rate in Hz */
+  opusMaxPlaybackRate?: PlaybackRate
+  /** OPUS audio codec max average bitrate in Hz */
+  opusMaxAverageBitrate?: number
 }
 
 export interface DialParams extends CallParams {

packages/webrtc/package.json

@@ -38,7 +38,7 @@
   },
   "dependencies": {
     "@signalwire/core": "4.2.1",
-    "sdp": "^3.2.0"
+    "sdp-transform": "2.15.0"
   },
   "types": "dist/cjs/webrtc/src/index.d.ts"
 }

packages/webrtc/src/RTCPeer.ts

@@ -5,10 +5,10 @@ import {
   filterIceServers,
 } from './utils/helpers'
 import {
-  sdpStereoHack,
   sdpBitrateHack,
   sdpMediaOrderHack,
   sdpHasValidCandidates,
+  updateSDPForOpus,
 } from './utils/sdpHelpers'
 import { BaseConnection } from './BaseConnection'
 import {
@@ -71,6 +71,8 @@ export default class RTCPeer<EventTypes extends EventEmitter.ValidEventTypes> {
       this.options
     )
 
+    this._validateOptions()
+
     this._onIce = this._onIce.bind(this)
     this._onEndedTrackHandler = this._onEndedTrackHandler.bind(this)
 
@@ -86,6 +88,16 @@ export default class RTCPeer<EventTypes extends EventEmitter.ValidEventTypes> {
     this.rtcConfigPolyfill = this.config
   }
 
+  private _validateOptions() {
+    if (
+      this.options.useStereo === true &&
+      typeof this.options.audio === 'object' &&
+      (this.options.audio.channelCount ?? 2) != 2
+    ) {
+      throw new Error('Mismatch params: useStereo, audio.channelCount')
+    }
+  }
+
   get options() {
     return this.call.options
   }
@@ -800,14 +812,13 @@ export default class RTCPeer<EventTypes extends EventEmitter.ValidEventTypes> {
   }
 
   private _setLocalDescription(localDescription: RTCSessionDescriptionInit) {
-    const {
-      useStereo,
-      googleMaxBitrate,
-      googleMinBitrate,
-      googleStartBitrate,
-    } = this.options
-    if (localDescription.sdp && useStereo) {
-      localDescription.sdp = sdpStereoHack(localDescription.sdp)
+    const { googleMaxBitrate, googleMinBitrate, googleStartBitrate } =
+      this.options
+    if (localDescription.sdp) {
+      localDescription.sdp = updateSDPForOpus(
+        localDescription.sdp,
+        this.options
+      )
     }
     if (
       localDescription.sdp &&
@@ -832,8 +843,11 @@ export default class RTCPeer<EventTypes extends EventEmitter.ValidEventTypes> {
   }
 
   private _setRemoteDescription(remoteDescription: RTCSessionDescriptionInit) {
-    if (remoteDescription.sdp && this.options.useStereo) {
-      remoteDescription.sdp = sdpStereoHack(remoteDescription.sdp)
+    if (remoteDescription.sdp) {
+      remoteDescription.sdp = updateSDPForOpus(
+        remoteDescription.sdp,
+        this.options
+      )
     }
     if (remoteDescription.sdp && this.instance.localDescription) {
       remoteDescription.sdp = sdpMediaOrderHack(