1.0.10

* feat(murf): add Murf.ai TTS support to synthAudio

Add synthMurf() following the rimelabs/cartesia pattern:
- streaming path returns a say:{vendor=murf,...} filePath consumed by the
  FreeSWITCH mod_murf_tts module
- non-streaming path calls POST /v1/speech/stream (api-key header) and returns
  WAV audio for cache rendering

Register murf in the supported-vendor assert list and the synth switch.

Co-Authored-By: Claude Opus 4.8 (1M context) <noreply@anthropic.com>

* fix(murf): drop Accept: audio/basic header (caused 406 Not Acceptable)

Murf's /v1/speech/stream rejects an unmatched Accept header with 406; the
response container is chosen by the `format` body field instead. Verified a
WAV request now returns 200 (valid RIFF/WAVE).

Co-Authored-By: Claude Opus 4.8 (1M context) <noreply@anthropic.com>

---------

Co-authored-by: Claude Opus 4.8 (1M context) <noreply@anthropic.com>

lib/synth-audio.js

@@ -80,7 +80,8 @@ async function synthAudio(client, createHash, retrieveHash, logger, stats, { acc
   logger = logger || noopLogger;
 
   assert.ok(['google', 'aws', 'polly', 'microsoft', 'wellsaid', 'nvidia', 'elevenlabs',
-    'whisper', 'deepgram', 'rimelabs', 'cartesia', 'inworld', 'resemble'].includes(vendor) ||
+    'whisper', 'deepgram', 'deepgramflux', 'rimelabs', 'cartesia', 'inworld', 'resemble', 'murf', 'xai']
+    .includes(vendor) ||
   vendor.startsWith('custom'),
   `synthAudio supported vendors are google, aws, microsoft, nvidia and wellsaid ..etc, not ${vendor}`);
   if ('google' === vendor) {
@@ -124,9 +125,19 @@ async function synthAudio(client, createHash, retrieveHash, logger, stats, { acc
     if (!credentials.deepgram_tts_uri) {
       assert.ok(credentials.api_key, 'synthAudio requires api_key when deepgram is used');
     }
+  } else if ('deepgramflux' === vendor) {
+    // Deepgram Flux TTS (/v2/speak); the flux model rides on `voice`/`model`
+    if (!credentials.deepgram_tts_uri) {
+      assert.ok(credentials.api_key, 'synthAudio requires api_key when deepgramflux is used');
+    }
+  } else if ('xai' === vendor) {
+    assert.ok(credentials.api_key, 'synthAudio requires api_key when xai is used');
   } else if ('cartesia' === vendor) {
     assert.ok(credentials.api_key, 'synthAudio requires api_key when cartesia is used');
     assert.ok(credentials.model_id, 'synthAudio requires model_id when cartesia is used');
+  } else if ('murf' === vendor) {
+    assert.ok(voice, 'synthAudio requires voice when murf is used');
+    assert.ok(credentials.api_key, 'synthAudio requires api_key when murf is used');
   } else if (vendor === 'resemble') {
     assert.ok(voice, 'synthAudio requires voice when resemble is used');
     assert.ok(credentials.api_key, 'synthAudio requires api_key when resemble is used');
@@ -211,6 +222,11 @@ async function synthAudio(client, createHash, retrieveHash, logger, stats, { acc
           credentials, options, stats, language, voice, key, text, renderForCaching, disableTtsStreaming,
           disableTtsCache});
         break;
+      case 'murf':
+        audioData = await synthMurf(logger, {
+          credentials, options, stats, language, voice, key, text, renderForCaching, disableTtsStreaming,
+          disableTtsCache});
+        break;
       case 'whisper':
         audioData = await synthWhisper(logger, {
           credentials, stats, voice, key, text, instructions, renderForCaching, disableTtsStreaming,
@@ -220,6 +236,15 @@ async function synthAudio(client, createHash, retrieveHash, logger, stats, { acc
         audioData = await synthDeepgram(logger, {credentials, stats, model, key, text,
           renderForCaching, disableTtsStreaming, disableTtsCache});
         break;
+      case 'deepgramflux':
+        audioData = await synthDeepgramFlux(logger, {credentials, stats, model: model || voice, key, text,
+          renderForCaching, disableTtsStreaming, disableTtsCache});
+        break;
+      case 'xai':
+        audioData = await synthXai(logger, {
+          credentials, options, stats, language, voice, key, text, renderForCaching, disableTtsStreaming,
+          disableTtsCache});
+        break;
       case 'resemble':
         audioData = await synthResemble(logger, {
           credentials, stats, voice, key, text, options, renderForCaching, disableTtsStreaming, disableTtsCache});
@@ -358,6 +383,32 @@ const synthPolly = async(createHash, retrieveHash, logger,
   }
 };
 
+/* google AudioConfig settings we support, as [google camelCase name, freeswitch param name] */
+const GOOGLE_AUDIO_SETTINGS = [
+  ['speakingRate', 'speaking_rate'],
+  ['pitch', 'pitch'],
+  ['volumeGainDb', 'volume_gain_db']
+];
+
+/**
+ * Extract google AudioConfig settings from the synthesizer options.  They may be supplied
+ * nested under an audioConfig property (mirroring google's AudioConfig object) or flat at
+ * the top level, and either google's camelCase or snake_case names are accepted.
+ * @see https://cloud.google.com/text-to-speech/docs/reference/rest/v1/text/synthesize#AudioConfig
+ * @returns object keyed by google's camelCase names, holding only valid numeric settings
+ */
+const googleAudioConfig = (options) => {
+  const provided = {...options, ...(options?.audioConfig || {})};
+  const audioConfig = {};
+  for (const [name, snakeName] of GOOGLE_AUDIO_SETTINGS) {
+    const value = provided[name] ?? provided[snakeName];
+    /* note: 0 is meaningful for pitch and volumeGainDb, so check for absence explicitly */
+    if (value === undefined || value === null || value === '') continue;
+    const num = Number(value);
+    if (Number.isFinite(num)) audioConfig[name] = num;
+  }
+  return audioConfig;
+};
 
 const synthGoogle = async(logger, {
   credentials, stats, language, voice, gender, key, text, model, options, instructions,
@@ -393,6 +444,15 @@ const synthGoogle = async(logger, {
     // comma is used to separate parameters in freeswitch tts module
     const prompt = options?.prompt || instructions;
     if (prompt) params += `,prompt=${prompt.replace(/\n/g, ' ').replace(/,/g, ';')}`;
+    /**
+     * AudioConfig settings.  Note google only honors these in some api modes:
+     * tts applies all of them, live (HD voices) applies speakingRate only,
+     * and gemini ignores them entirely (use prompt instead for style control).
+     */
+    const audioSettings = googleAudioConfig(options);
+    for (const [name, snakeName] of GOOGLE_AUDIO_SETTINGS) {
+      if (name in audioSettings) params += `,${snakeName}=${audioSettings[name]}`;
+    }
     params += '}';
 
     return {
@@ -461,6 +521,9 @@ const synthGoogle = async(logger, {
     sampleRate = 8000;
   }
 
+  /* gemini voices do not support the AudioConfig settings; they use prompt for style control */
+  if (!isGemini) Object.assign(audioConfig, googleAudioConfig(options));
+
   const opts = { input, voice: voiceParams, audioConfig };
 
   try {
@@ -969,6 +1032,72 @@ const synthRimelabs = async(logger, {
     throw err;
   }
 };
+
+const synthMurf = async(logger, {
+  credentials, options, stats, language, voice, key, text, renderForCaching, disableTtsStreaming, disableTtsCache
+}) => {
+  const {api_key, model_id, api_uri, options: credOpts} = credentials;
+  const opts = !!options && Object.keys(options).length !== 0 ? options : JSON.parse(credOpts || '{}');
+
+  /* default to using the streaming interface, unless disabled by env var OR we want just a cache file */
+  if (!JAMBONES_DISABLE_TTS_STREAMING && !renderForCaching && !disableTtsStreaming) {
+    /* param keys here must match mod_murf_tts's text_param handler */
+    let params = '{';
+    params += `api_key=${api_key}`;
+    params += `,playback_id=${key}`;
+    params += ',vendor=murf';
+    params += `,voice=${voice}`;
+    if (model_id) params += `,model_id=${model_id}`;
+    if (language) params += `,language=${language}`;
+    if (api_uri) params += `,api_uri=${api_uri}`;
+    if (opts.style) params += `,style=${opts.style}`;
+    if (opts.rate !== undefined && opts.rate !== null) params += `,rate=${opts.rate}`;
+    if (opts.pitch !== undefined && opts.pitch !== null) params += `,pitch=${opts.pitch}`;
+    if (opts.variation !== undefined && opts.variation !== null) params += `,variation=${opts.variation}`;
+    params += `,write_cache_file=${disableTtsCache ? 0 : 1}`;
+    params += '}';
+
+    return {
+      filePath: `say:${params}${text.replace(/\n/g, ' ').replace(/\r/g, ' ')}`,
+      servedFromCache: false,
+      rtt: 0
+    };
+  }
+
+  try {
+    const sampleRate = 8000;
+    /* no Accept header: murf returns 406 if it doesn't match; the response
+       container is selected by the `format` field in the body instead */
+    const post = bent(api_uri || 'https://global.api.murf.ai', 'POST', 'buffer', {
+      'api-key': api_key,
+      'Content-Type': 'application/json'
+    });
+    /* murf REST schema is documented loosely; field names follow the SDK params
+       (voice_id/model/format/sample_rate) plus the websocket voice fields. */
+    const audioContent = await post('/v1/speech/stream', {
+      text,
+      voice_id: voice,
+      ...(model_id && {model: model_id}),
+      ...(language && {locale: language}),
+      ...(opts.style && {style: opts.style}),
+      ...(opts.rate !== undefined && opts.rate !== null && {rate: opts.rate}),
+      ...(opts.pitch !== undefined && opts.pitch !== null && {pitch: opts.pitch}),
+      ...(opts.variation !== undefined && opts.variation !== null && {variation: opts.variation}),
+      format: 'WAV',
+      sample_rate: sampleRate,
+      channel_type: 'MONO'
+    });
+    return {
+      audioContent,
+      extension: 'wav',
+      sampleRate
+    };
+  } catch (err) {
+    logger.info({err}, 'synth murf returned error');
+    stats.increment('tts.count', ['vendor:murf', 'accepted:no']);
+    throw err;
+  }
+};
 const synthWhisper = async(logger, {credentials, stats, voice, key, text, instructions,
   renderForCaching, disableTtsStreaming, disableTtsCache}) => {
   const {api_key, model_id, baseURL, timeout, speed} = credentials;
@@ -1059,6 +1188,116 @@ const synthDeepgram = async(logger, {credentials, stats, model, key, text, rende
   }
 };
 
+// Deepgram Flux TTS — the conversation-native model served from /v2/speak.
+// Streaming rides the mediajam deepgramflux dialect via a say: filePath; the
+// batch/cache path POSTs to /v2/speak (mp3 is batch-only for Flux).
+const synthDeepgramFlux = async(logger, {credentials, stats, model, key, text, renderForCaching,
+  disableTtsStreaming, disableTtsCache}) => {
+  const {api_key, deepgram_tts_uri} = credentials;
+  if (!JAMBONES_DISABLE_TTS_STREAMING && !renderForCaching && !disableTtsStreaming) {
+    let params = '{';
+    params += `api_key=${api_key}`;
+    params += `,playback_id=${key}`;
+    params += ',vendor=deepgramflux';
+    params += `,voice=${model}`;
+    params += `,write_cache_file=${disableTtsCache ? 0 : 1}`;
+    if (deepgram_tts_uri) params += `,endpoint=${deepgram_tts_uri}`;
+    params += '}';
+
+    return {
+      filePath: `say:${params}${text.replace(/\n/g, ' ')}`,
+      servedFromCache: false,
+      rtt: 0
+    };
+  }
+  try {
+    const post = bent(deepgram_tts_uri || 'https://api.deepgram.com', 'POST', 'buffer', {
+      // on-premise deepgram does not require to have api_key
+      ...(api_key && {'Authorization': `Token ${api_key}`}),
+      'Accept': 'audio/mpeg',
+      'Content-Type': 'application/json'
+    });
+    const audioContent = await post(`/v2/speak?model=${model}`, {
+      text
+    });
+    return {
+      audioContent,
+      extension: 'mp3',
+      sampleRate: 8000
+    };
+  } catch (err) {
+    logger.info({err}, 'synth Deepgram Flux returned error');
+    stats.increment('tts.count', ['vendor:deepgramflux', 'accepted:no']);
+    throw err;
+  }
+};
+
+const synthXai = async(logger, {
+  credentials, options, stats, language, voice, key, text, renderForCaching, disableTtsStreaming, disableTtsCache
+}) => {
+  const {api_key, api_uri, options: credOpts} = credentials;
+  const opts = !!options && Object.keys(options).length !== 0 ? options : JSON.parse(credOpts || '{}');
+  const speed = opts.speed;
+  if (!JAMBONES_DISABLE_TTS_STREAMING && !renderForCaching && !disableTtsStreaming) {
+    let params = '{';
+    params += `api_key=${api_key}`;
+    params += `,playback_id=${key}`;
+    params += ',vendor=xai';
+    if (voice) params += `,voice=${voice}`;
+    if (language) params += `,language=${language}`;
+    if (speed !== null && speed !== undefined) params += `,speed=${speed}`;
+    if (opts.optimize_streaming_latency != null) {
+      params += `,optimize_streaming_latency=${opts.optimize_streaming_latency}`;
+    }
+    if (opts.text_normalization != null) params += `,text_normalization=${opts.text_normalization}`;
+    params += `,write_cache_file=${disableTtsCache ? 0 : 1}`;
+    if (api_uri) params += `,endpoint=${api_uri}`;
+    params += '}';
+
+    return {
+      filePath: `say:${params}${text.replace(/\n/g, ' ').replace(/\r/g, ' ')}`,
+      servedFromCache: false,
+      rtt: 0
+    };
+  }
+  try {
+    const post = bent(`https://${api_uri || 'api.x.ai'}`, 'POST', 'buffer', {
+      'Authorization': `Bearer ${api_key}`,
+      'Content-Type': 'application/json'
+    });
+    const audioContent = await post('/v1/tts', {
+      text,
+      language: language || 'auto',
+      ...(voice && {voice_id: voice}),
+      ...(speed !== null && speed !== undefined && {speed}),
+      ...(opts.optimize_streaming_latency != null && {optimize_streaming_latency: opts.optimize_streaming_latency}),
+      ...(opts.text_normalization != null && {text_normalization: opts.text_normalization}),
+      output_format: {
+        codec: 'wav',
+        sample_rate: 8000
+      }
+    });
+    return {
+      audioContent,
+      extension: 'wav',
+      sampleRate: 8000
+    };
+  } catch (err) {
+    // xAI errors are JSON {code, error} - read the body so the surfaced message isn't 'undefined'
+    if (err.name === 'StatusError' && typeof err.text === 'function') {
+      try {
+        const body = await err.text();
+        if (body) err.message = body;
+      } catch (readErr) {
+        logger.info({readErr}, 'synth xai: failed to read error response body');
+      }
+    }
+    logger.info({err}, 'synth xai returned error');
+    stats.increment('tts.count', ['vendor:xai', 'accepted:no']);
+    throw err;
+  }
+};
+
 const synthCartesia = async(logger, {
   credentials, options, stats, voice, language, key, text, renderForCaching, disableTtsStreaming, disableTtsCache
 }) => {

package-lock.json

@@ -1,12 +1,12 @@
 {
   "name": "@jambonz/speech-utils",
-  "version": "1.0.6",
+  "version": "1.0.10",
   "lockfileVersion": 2,
   "requires": true,
   "packages": {
     "": {
       "name": "@jambonz/speech-utils",
-      "version": "1.0.6",
+      "version": "1.0.10",
       "license": "MIT",
       "dependencies": {
         "23": "^0.0.0",

package.json

@@ -1,6 +1,6 @@
 {
   "name": "@jambonz/speech-utils",
-  "version": "1.0.6",
+  "version": "1.0.10",
   "description": "TTS-related speech utilities for jambonz",
   "main": "index.js",
   "author": "Dave Horton",

test/synth.js

@@ -442,6 +442,95 @@ test('Google TTS streaming tests (!JAMBONES_DISABLE_TTS_STREAMING)', async(t) =>
     });
     t.ok(result.filePath.includes('api_mode=tts'), 'options.apiMode=tts overrides HD voice default');
 
+    /* AudioConfig settings (speakingRate, pitch, volumeGainDb) */
+    const googleCreds = {
+      credentials: {
+        client_email: creds.client_email,
+        private_key: creds.private_key,
+      },
+    };
+
+    // Test 9: AudioConfig nested under options.audioConfig, as in google's API docs
+    result = await synthAudio(stats, {
+      vendor: 'google',
+      credentials: googleCreds,
+      language: 'en-US',
+      voice: 'en-US-Wavenet-D',
+      text: 'Testing nested audioConfig settings.',
+      options: { audioConfig: { speakingRate: 1.4, pitch: -2.5, volumeGainDb: 6 } },
+      disableTtsCache: true
+    });
+    t.ok(result.filePath.includes(',speaking_rate=1.4'), 'nested audioConfig sets speaking_rate');
+    t.ok(result.filePath.includes(',pitch=-2.5'), 'nested audioConfig sets pitch');
+    t.ok(result.filePath.includes(',volume_gain_db=6'), 'nested audioConfig sets volume_gain_db');
+
+    // Test 10: AudioConfig flat at the top level of options, in camelCase or snake_case
+    result = await synthAudio(stats, {
+      vendor: 'google',
+      credentials: googleCreds,
+      language: 'en-US',
+      voice: 'en-US-Wavenet-D',
+      text: 'Testing flat audioConfig settings.',
+      options: { speaking_rate: '0.8', volumeGainDb: 3 },
+      disableTtsCache: true
+    });
+    t.ok(result.filePath.includes(',speaking_rate=0.8'), 'flat snake_case speaking_rate is honored');
+    t.ok(result.filePath.includes(',volume_gain_db=3'), 'flat camelCase volumeGainDb is honored');
+    t.ok(!result.filePath.includes(',pitch='), 'unspecified audioConfig setting is omitted');
+
+    // Test 11: zero is a meaningful value for pitch and volumeGainDb, not an absent one
+    result = await synthAudio(stats, {
+      vendor: 'google',
+      credentials: googleCreds,
+      language: 'en-US',
+      voice: 'en-US-Wavenet-D',
+      text: 'Testing zero audioConfig settings.',
+      options: { audioConfig: { pitch: 0, volumeGainDb: 0 } },
+      disableTtsCache: true
+    });
+    t.ok(result.filePath.includes(',pitch=0'), 'pitch=0 is passed through rather than dropped');
+    t.ok(result.filePath.includes(',volume_gain_db=0'), 'volume_gain_db=0 is passed through rather than dropped');
+
+    // Test 12: non-numeric values are ignored, so they cannot corrupt the freeswitch param string
+    result = await synthAudio(stats, {
+      vendor: 'google',
+      credentials: googleCreds,
+      language: 'en-US',
+      voice: 'en-US-Wavenet-D',
+      text: 'Testing invalid audioConfig settings.',
+      options: { audioConfig: { speakingRate: 'fast,evil=1', pitch: null, volumeGainDb: '' } },
+      disableTtsCache: true
+    });
+    t.ok(!result.filePath.includes('speaking_rate'), 'non-numeric speakingRate is ignored');
+    t.ok(!result.filePath.includes('evil=1'), 'non-numeric value cannot inject extra params');
+    t.ok(!result.filePath.includes('pitch='), 'null pitch is ignored');
+    t.ok(!result.filePath.includes('volume_gain_db'), 'empty volumeGainDb is ignored');
+
+    // Test 13: no audioConfig supplied leaves the param string untouched
+    result = await synthAudio(stats, {
+      vendor: 'google',
+      credentials: googleCreds,
+      language: 'en-US',
+      voice: 'en-US-Wavenet-D',
+      text: 'Testing absent audioConfig settings.',
+      disableTtsCache: true
+    });
+    t.ok(!/speaking_rate|pitch=|volume_gain_db/.test(result.filePath),
+      'no audioConfig params are added when none are supplied');
+
+    // Test 14: HD voice (api_mode=live) also carries speaking_rate, the one setting google streams support
+    result = await synthAudio(stats, {
+      vendor: 'google',
+      credentials: googleCreds,
+      language: 'en-US',
+      voice: 'en-US-Chirp3-HD-Charon',
+      text: 'Testing audioConfig on an HD voice.',
+      options: { audioConfig: { speakingRate: 1.25 } },
+      disableTtsCache: true
+    });
+    t.ok(result.filePath.includes('api_mode=live'), 'HD voice with audioConfig still uses api_mode=live');
+    t.ok(result.filePath.includes(',speaking_rate=1.25'), 'HD voice streaming path carries speaking_rate');
+
   } catch (err) {
     console.error(err);
     t.end(err);
@@ -526,6 +615,42 @@ test('Google TTS non-streaming tests (JAMBONES_DISABLE_TTS_STREAMING=true)', asy
     t.ok(!result.filePath.startsWith('say:'), 'Gemini TTS does NOT return streaming say: path when disabled');
     t.ok(result.filePath.endsWith('.mp3'), 'Gemini TTS returns mp3 file path');
 
+    const googleCreds = {
+      credentials: {
+        client_email: creds.client_email,
+        private_key: creds.private_key,
+      },
+    };
+
+    /**
+     * Test 4: AudioConfig settings are accepted by the synthesize API.
+     * Google rejects out-of-range values with a 400, so a successful render also confirms
+     * the settings reached the request rather than being silently dropped.
+     */
+    result = await synthAudio(stats, {
+      vendor: 'google',
+      credentials: googleCreds,
+      language: 'en-US',
+      voice: 'en-US-Wavenet-D',
+      text: 'This is a test of audioConfig with streaming disabled.',
+      options: { audioConfig: { speakingRate: 1.4, pitch: -2.5, volumeGainDb: 6 } },
+      disableTtsCache: true
+    });
+    t.ok(result.filePath.endsWith('.mp3'), 'standard voice renders mp3 with audioConfig settings applied');
+
+    /* Test 5: gemini voices ignore the AudioConfig settings rather than failing on them */
+    result = await synthAudio(stats, {
+      vendor: 'google',
+      credentials: googleCreds,
+      language: 'en-US',
+      voice: 'Kore',
+      model: geminiModel,
+      text: 'This is a test of audioConfig on Gemini TTS.',
+      options: { audioConfig: { speakingRate: 1.4, pitch: -2.5, volumeGainDb: 6 } },
+      disableTtsCache: true
+    });
+    t.ok(result.filePath.endsWith('.mp3'), 'gemini voice renders mp3 with audioConfig settings skipped');
+
   } catch (err) {
     console.error(err);
     t.end(err);
@@ -1118,6 +1243,69 @@ test('Deepgram speech synth tests', async(t) => {
   client.quit();
 })
 
+test('Deepgram Flux speech synth tests', async(t) => {
+  const fn = require('..');
+  const {synthAudio, client} = fn(opts, logger);
+
+  if (!process.env.DEEPGRAM_API_KEY) {
+    t.pass('skipping Deepgram Flux speech synth tests since DEEPGRAM_API_KEY');
+    return t.end();
+  }
+  const text = 'Hi there and welcome to jambones!';
+  try {
+    const opts = await synthAudio(stats, {
+      vendor: 'deepgramflux',
+      credentials: {
+        api_key: process.env.DEEPGRAM_API_KEY
+      },
+      model: process.env.DEEPGRAM_FLUX_MODEL || 'flux-alexis-en',
+      text,
+      renderForCaching: true
+    });
+    t.ok(!opts.servedFromCache, `successfully synthesized deepgramflux audio to ${opts.filePath}`);
+
+  } catch (err) {
+    console.error(JSON.stringify(err));
+    t.end(err);
+  }
+  client.quit();
+});
+
+test('xai speech synth tests', async(t) => {
+  const fn = require('..');
+  const {synthAudio, client} = fn(opts, logger);
+
+  if (!process.env.XAI_API_KEY) {
+    t.pass('skipping xai speech synth tests - no XAI_API_KEY');
+    return t.end();
+  }
+  const text = 'Hi there and welcome to jambones!';
+  try {
+    const opts = await synthAudio(stats, {
+      vendor: 'xai',
+      credentials: {
+        api_key: process.env.XAI_API_KEY,
+        options: JSON.stringify({
+          voice: process.env.XAI_VOICE || 'eve',
+          speed: 1.0,
+          optimize_streaming_latency: 1,
+          text_normalization: true
+        })
+      },
+      language: 'en',
+      voice: process.env.XAI_VOICE || 'eve',
+      text,
+      renderForCaching: true
+    });
+    t.ok(!opts.servedFromCache && opts.filePath, `successfully synthesized xai audio to ${opts.filePath}`);
+
+  } catch (err) {
+    console.error(JSON.stringify(err));
+    t.end(err);
+  }
+  client.quit();
+});
+
 test('TTS Cache tests', async(t) => {
   const fn = require('..');
   const {purgeTtsCache, getTtsSize, client} = fn(opts, logger);