This repo is developed for aliyun tts websocket. For more information, please see: https://help.aliyun.com/document_detail/324262.html?spm=a2c4g.324262.0.0.7dd87d63BjFUFh
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$ npm install aliyun-tts-websocket --saveWith webpack, vite or other build tools:
import AliyunTtsWebSocket from 'aliyun-tts-websocket'
const service = new AliyunTtsWebSocket({
ws: '', // A websocket address with your token.
ttsHeaders: {
appkey: '' // Your aliyun cloud appkey
}
})
service.start()
service.stop()| name | type | default | description |
|---|---|---|---|
| ws | string | '' | Websocket url. |
| mediaStreamHandler | (stream: MediaStream, send: Function) => void | null | A function which can make you change audio stream and send to websocket. |
| ttsConfig | object | tts default config | Same as the payload of StartTranscription directive. |
| ttsHeaders | object | tts default header | Same as the header of each directive. |
This function make you convert audio format by your self. PCM is used by default.
{
enable_intermediate_result: true,
enable_punctuation_prediction: true
}This params is used for StartTranscription directive's payload. For more information, see: https://help.aliyun.com/document_detail/324262.html?spm=a2c4g.324262.0.0.7dd87d63BjFUFh#sectiondiv-rz2-i36-2gv
{
appkey: ''
}This params is used for each directive. For more information, see: https://help.aliyun.com/document_detail/324262.html?spm=a2c4g.324262.0.0.7dd87d63BjFUFh#sectiondiv-mkk-cs4-5kh
Notice that only appkey field is required, other fields is not recommended to pass.
| name | description |
|---|---|
| start | Open a websocket connection, send StartTranscription directive to server and weakup microphone device. |
| stop | Disconnect the microphone devece and send StopTranscription directive to serve. The websocket connection will be closed after receive the TranscriptionCompleted event. |
| on | Alias of addEventListener. |
Aliyun-tts-websocket extends EventTarget, you can listen events as same as DOM events.
const service = new AliyunTtsWebSocket({
// ...
})
service.addEventListener('ws:open', function () {
// do something.
})
service.addEventListener('tts:change', function (event) {
console.log(event.detail); // get event data.
})| name | event data | description |
|---|---|---|
| ws:open | undefined | Will be called after websocket connected. |
| ws:message | Websocket MessageEvent | Will be called after receive websocket messages. |
| ws:close | undefined | Will be called after websocket disconnected. |
| tts:begin | same as SentenceBegin event |
Will be called after receive SentenceBegin event. |
| tts:change | same as TranscriptionResultChanged event |
Will be called after receive TranscriptionResultChanged event. |
| tts:end | same as SentenceEnd event |
Will be called after receive SentenceEnd event. |
| tts:stop | same as TranscriptionCompleted event |
Will be called after receive TranscriptionCompleted event. |
| ready | undefined | Will be called after weakup microphone device. |
Some code is referenced from this article: https://juejin.cn/post/7028003824912564231, thanks!