@livekit/agents-plugin-baseten 1.4.1 → 1.4.3

This diff represents the content of publicly available package versions that have been released to one of the supported registries. The information contained in this diff is provided for informational purposes only and reflects changes between package versions as they appear in their respective public registries.
Files changed (8) hide show
  1. package/dist/index.cjs +1 -1
  2. package/dist/index.js +1 -1
  3. package/dist/stt.cjs +1 -1
  4. package/dist/stt.cjs.map +1 -1
  5. package/dist/stt.js +1 -1
  6. package/dist/stt.js.map +1 -1
  7. package/package.json +5 -5
  8. package/src/stt.ts +1 -1
package/dist/index.cjs CHANGED
@@ -32,7 +32,7 @@ class BasetenPlugin extends import_agents.Plugin {
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  constructor() {
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  super({
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  title: "baseten",
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- version: "1.4.1",
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+ version: "1.4.3",
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  package: "@livekit/agents-plugin-baseten"
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  });
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  }
package/dist/index.js CHANGED
@@ -6,7 +6,7 @@ class BasetenPlugin extends Plugin {
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  constructor() {
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  super({
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  title: "baseten",
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- version: "1.4.1",
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+ version: "1.4.3",
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  package: "@livekit/agents-plugin-baseten"
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  });
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  }
package/dist/stt.cjs CHANGED
@@ -110,7 +110,7 @@ class SpeechStream extends import_agents.stt.SpeechStream {
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  const headers = {
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  Authorization: `Api-Key ${this.#opts.apiKey}`
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  };
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- const ws = new import_ws.WebSocket(url, { headers, rejectUnauthorized: false });
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+ const ws = new import_ws.WebSocket(url, { headers });
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  try {
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  await new Promise((resolve, reject) => {
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  ws.on("open", resolve);
package/dist/stt.cjs.map CHANGED
@@ -1 +1 @@
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- {"version":3,"sources":["../src/stt.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2024 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport {\n type AudioBuffer,\n AudioByteStream,\n Task,\n log,\n normalizeLanguage,\n stt,\n waitForAbort,\n} from '@livekit/agents';\nimport type { AudioFrame } from '@livekit/rtc-node';\nimport { WebSocket } from 'ws';\nimport type { BasetenSttOptions } from './types.js';\n\nconst defaultSTTOptions: Partial<BasetenSttOptions> = {\n environment: 'production',\n encoding: 'pcm_s16le',\n sampleRate: 16000,\n bufferSizeSeconds: 0.032,\n enablePartialTranscripts: true,\n partialTranscriptIntervalS: 0.5,\n finalTranscriptMaxDurationS: 5,\n audioLanguage: 'en',\n languageDetectionOnly: false,\n vadThreshold: 0.5,\n vadMinSilenceDurationMs: 300,\n vadSpeechPadMs: 30,\n};\n\nexport class STT extends stt.STT {\n #opts: BasetenSttOptions;\n #logger = log();\n label = 'baseten.STT';\n\n constructor(opts: Partial<BasetenSttOptions> = {}) {\n super({\n streaming: true,\n interimResults: opts.enablePartialTranscripts ?? defaultSTTOptions.enablePartialTranscripts!,\n alignedTranscript: 'word',\n });\n\n const apiKey = opts.apiKey ?? process.env.BASETEN_API_KEY;\n const modelEndpoint = opts.modelEndpoint ?? process.env.BASETEN_MODEL_ENDPOINT;\n const modelId = opts.modelId ?? process.env.BASETEN_STT_MODEL_ID;\n\n if (!apiKey) {\n throw new Error(\n 'Baseten API key is required, either pass it as `apiKey` or set $BASETEN_API_KEY',\n );\n }\n if (!modelEndpoint && !modelId) {\n throw new Error(\n 'Baseten model endpoint is required, either pass it as `modelEndpoint` or set $BASETEN_MODEL_ENDPOINT',\n );\n }\n\n this.#opts = {\n ...defaultSTTOptions,\n ...opts,\n apiKey,\n modelEndpoint,\n modelId,\n audioLanguage: normalizeLanguage((opts.audioLanguage ?? defaultSTTOptions.audioLanguage)!),\n } as BasetenSttOptions;\n }\n\n // eslint-disable-next-line\n async _recognize(_: AudioBuffer): Promise<stt.SpeechEvent> {\n throw new Error('Recognize is not supported on Baseten STT');\n }\n\n updateOptions(opts: Partial<BasetenSttOptions>) {\n this.#opts = {\n ...this.#opts,\n ...opts,\n audioLanguage:\n opts.audioLanguage !== undefined\n ? normalizeLanguage(opts.audioLanguage)\n : this.#opts.audioLanguage,\n };\n }\n\n stream(): SpeechStream {\n return new SpeechStream(this, this.#opts);\n }\n}\n\nexport class SpeechStream extends stt.SpeechStream {\n #opts: BasetenSttOptions;\n #logger = log();\n #speaking = false;\n #requestId = '';\n label = 'baseten.SpeechStream';\n\n constructor(stt: STT, opts: BasetenSttOptions) {\n super(stt, opts.sampleRate);\n this.#opts = opts;\n this.closed = false;\n }\n\n private getWsUrl(): string {\n if (this.#opts.modelEndpoint) {\n return this.#opts.modelEndpoint;\n }\n // Fallback to constructing URL from modelId (deprecated)\n return `wss://model-${this.#opts.modelId}.api.baseten.co/environments/${this.#opts.environment}/websocket`;\n }\n\n protected async run() {\n const maxRetry = 32;\n let retries = 0;\n\n while (!this.input.closed && !this.closed) {\n const url = this.getWsUrl();\n const headers = {\n Authorization: `Api-Key ${this.#opts.apiKey}`,\n };\n\n const ws = new WebSocket(url, { headers, rejectUnauthorized: false });\n\n try {\n await new Promise((resolve, reject) => {\n ws.on('open', resolve);\n ws.on('error', (error) => reject(error));\n ws.on('close', (code) => reject(`WebSocket returned ${code}`));\n });\n\n await this.#runWS(ws);\n } catch (e) {\n if (!this.closed && !this.input.closed) {\n if (retries >= maxRetry) {\n throw new Error(`failed to connect to Baseten after ${retries} attempts: ${e}`);\n }\n\n const delay = Math.min(retries * 5, 10);\n retries++;\n\n this.#logger.warn(\n `failed to connect to Baseten, retrying in ${delay} seconds: ${e} (${retries}/${maxRetry})`,\n );\n await new Promise((resolve) => setTimeout(resolve, delay * 1000));\n } else {\n this.#logger.warn(\n `Baseten disconnected, connection is closed: ${e} (inputClosed: ${this.input.closed}, isClosed: ${this.closed})`,\n );\n }\n }\n }\n\n this.closed = true;\n }\n\n async #runWS(ws: WebSocket) {\n let closing = false;\n\n // Send initial metadata\n // Note: Baseten server expects 'vad_params' and 'streaming_whisper_params' field names\n // (not 'streaming_vad_config', 'streaming_params', 'whisper_params' as in older versions)\n const metadata = {\n vad_params: {\n threshold: this.#opts.vadThreshold,\n min_silence_duration_ms: this.#opts.vadMinSilenceDurationMs,\n speech_pad_ms: this.#opts.vadSpeechPadMs,\n },\n streaming_whisper_params: {\n encoding: this.#opts.encoding ?? 'pcm_s16le',\n sample_rate: this.#opts.sampleRate ?? 16000,\n enable_partial_transcripts: false,\n audio_language: this.#opts.audioLanguage ?? 'en',\n show_word_timestamps: true,\n },\n };\n\n ws.send(JSON.stringify(metadata));\n\n const sendTask = async () => {\n const sampleRate = this.#opts.sampleRate ?? 16000;\n const samplesPerChunk = sampleRate === 16000 ? 512 : 256;\n const audioByteStream = new AudioByteStream(sampleRate, 1, samplesPerChunk);\n\n try {\n while (!this.closed) {\n const result = await this.input.next();\n if (result.done) {\n break;\n }\n\n const data = result.value;\n\n let frames: AudioFrame[];\n if (data === SpeechStream.FLUSH_SENTINEL) {\n // Flush any remaining buffered audio\n frames = audioByteStream.flush();\n } else {\n if (data.sampleRate !== sampleRate || data.channels !== 1) {\n throw new Error(\n `sample rate or channel count mismatch: expected ${sampleRate}Hz/1ch, got ${data.sampleRate}Hz/${data.channels}ch`,\n );\n }\n frames = audioByteStream.write(data.data.buffer as ArrayBuffer);\n }\n\n for (const frame of frames) {\n const buffer = Buffer.from(\n frame.data.buffer,\n frame.data.byteOffset,\n frame.data.byteLength,\n );\n ws.send(buffer);\n }\n }\n } finally {\n closing = true;\n ws.close();\n }\n };\n\n const listenTask = Task.from(async (controller) => {\n const listenMessage = new Promise<void>((resolve, reject) => {\n ws.on('message', (data) => {\n try {\n let jsonString: string;\n\n if (typeof data === 'string') {\n jsonString = data;\n } else if (data instanceof Buffer) {\n jsonString = data.toString('utf-8');\n } else if (Array.isArray(data)) {\n jsonString = Buffer.concat(data).toString('utf-8');\n } else {\n return;\n }\n\n const msg = JSON.parse(jsonString);\n const isFinal = msg.is_final ?? true;\n const segments = msg.segments ?? [];\n const transcript = msg.transcript ?? '';\n const confidence = msg.confidence ?? 0.0;\n const languageCode = normalizeLanguage(msg.language_code ?? this.#opts.audioLanguage);\n\n // Skip if no transcript text\n if (!transcript) {\n this.#logger.debug('Received non-transcript message:', msg);\n return;\n }\n\n // Emit START_OF_SPEECH if not already speaking (only for interim or first final)\n if (!this.#speaking && !isFinal) {\n this.#speaking = true;\n this.queue.put({ type: stt.SpeechEventType.START_OF_SPEECH });\n }\n\n // Note: Baseten uses 'start_time' and 'end_time' field names (with underscores)\n const startTime =\n segments.length > 0\n ? (segments[0].start_time ?? 0.0) + this.startTimeOffset\n : this.startTimeOffset;\n const endTime =\n segments.length > 0\n ? (segments[segments.length - 1].end_time ?? 0.0) + this.startTimeOffset\n : this.startTimeOffset;\n\n // Note: Baseten returns segments (chunks) which we treat as words for aligned transcripts\n const words = segments.map(\n (segment: { text?: string; start_time?: number; end_time?: number }) => ({\n text: segment.text ?? '',\n startTime: (segment.start_time ?? 0.0) + this.startTimeOffset,\n endTime: (segment.end_time ?? 0.0) + this.startTimeOffset,\n startTimeOffset: this.startTimeOffset,\n confidence: confidence,\n }),\n );\n\n const speechData: stt.SpeechData = {\n language: languageCode,\n text: transcript,\n startTime,\n endTime,\n confidence,\n words: words.length > 0 ? words : undefined,\n };\n\n // Handle interim vs final transcripts (matching Python implementation)\n if (!isFinal) {\n // Interim transcript\n this.queue.put({\n type: stt.SpeechEventType.INTERIM_TRANSCRIPT,\n alternatives: [speechData],\n });\n } else {\n // Final transcript\n this.queue.put({\n type: stt.SpeechEventType.FINAL_TRANSCRIPT,\n alternatives: [speechData],\n });\n\n // Emit END_OF_SPEECH after final transcript\n if (this.#speaking) {\n this.#speaking = false;\n this.queue.put({ type: stt.SpeechEventType.END_OF_SPEECH });\n }\n }\n\n if (this.closed || closing) {\n resolve();\n }\n } catch (err) {\n this.#logger.error(`STT: Error processing message: ${data}`);\n reject(err);\n }\n });\n\n ws.on('error', (err) => {\n if (!closing) {\n reject(err);\n }\n });\n\n ws.on('close', () => {\n if (!closing) {\n resolve();\n }\n });\n });\n\n await Promise.race([listenMessage, waitForAbort(controller.signal)]);\n }, this.abortController);\n\n await Promise.all([sendTask(), listenTask.result]);\n closing = true;\n ws.close();\n 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+ {"version":3,"sources":["../src/stt.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2024 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport {\n type AudioBuffer,\n AudioByteStream,\n Task,\n log,\n normalizeLanguage,\n stt,\n waitForAbort,\n} from '@livekit/agents';\nimport type { AudioFrame } from '@livekit/rtc-node';\nimport { WebSocket } from 'ws';\nimport type { BasetenSttOptions } from './types.js';\n\nconst defaultSTTOptions: Partial<BasetenSttOptions> = {\n environment: 'production',\n encoding: 'pcm_s16le',\n sampleRate: 16000,\n bufferSizeSeconds: 0.032,\n enablePartialTranscripts: true,\n partialTranscriptIntervalS: 0.5,\n finalTranscriptMaxDurationS: 5,\n audioLanguage: 'en',\n languageDetectionOnly: false,\n vadThreshold: 0.5,\n vadMinSilenceDurationMs: 300,\n vadSpeechPadMs: 30,\n};\n\nexport class STT extends stt.STT {\n #opts: BasetenSttOptions;\n #logger = log();\n label = 'baseten.STT';\n\n constructor(opts: Partial<BasetenSttOptions> = {}) {\n super({\n streaming: true,\n interimResults: opts.enablePartialTranscripts ?? defaultSTTOptions.enablePartialTranscripts!,\n alignedTranscript: 'word',\n });\n\n const apiKey = opts.apiKey ?? process.env.BASETEN_API_KEY;\n const modelEndpoint = opts.modelEndpoint ?? process.env.BASETEN_MODEL_ENDPOINT;\n const modelId = opts.modelId ?? process.env.BASETEN_STT_MODEL_ID;\n\n if (!apiKey) {\n throw new Error(\n 'Baseten API key is required, either pass it as `apiKey` or set $BASETEN_API_KEY',\n );\n }\n if (!modelEndpoint && !modelId) {\n throw new Error(\n 'Baseten model endpoint is required, either pass it as `modelEndpoint` or set $BASETEN_MODEL_ENDPOINT',\n );\n }\n\n this.#opts = {\n ...defaultSTTOptions,\n ...opts,\n apiKey,\n modelEndpoint,\n modelId,\n audioLanguage: normalizeLanguage((opts.audioLanguage ?? defaultSTTOptions.audioLanguage)!),\n } as BasetenSttOptions;\n }\n\n // eslint-disable-next-line\n async _recognize(_: AudioBuffer): Promise<stt.SpeechEvent> {\n throw new Error('Recognize is not supported on Baseten STT');\n }\n\n updateOptions(opts: Partial<BasetenSttOptions>) {\n this.#opts = {\n ...this.#opts,\n ...opts,\n audioLanguage:\n opts.audioLanguage !== undefined\n ? normalizeLanguage(opts.audioLanguage)\n : this.#opts.audioLanguage,\n };\n }\n\n stream(): SpeechStream {\n return new SpeechStream(this, this.#opts);\n }\n}\n\nexport class SpeechStream extends stt.SpeechStream {\n #opts: BasetenSttOptions;\n #logger = log();\n #speaking = false;\n #requestId = '';\n label = 'baseten.SpeechStream';\n\n constructor(stt: STT, opts: BasetenSttOptions) {\n super(stt, opts.sampleRate);\n this.#opts = opts;\n this.closed = false;\n }\n\n private getWsUrl(): string {\n if (this.#opts.modelEndpoint) {\n return this.#opts.modelEndpoint;\n }\n // Fallback to constructing URL from modelId (deprecated)\n return `wss://model-${this.#opts.modelId}.api.baseten.co/environments/${this.#opts.environment}/websocket`;\n }\n\n protected async run() {\n const maxRetry = 32;\n let retries = 0;\n\n while (!this.input.closed && !this.closed) {\n const url = this.getWsUrl();\n const headers = {\n Authorization: `Api-Key ${this.#opts.apiKey}`,\n };\n\n const ws = new WebSocket(url, { headers });\n\n try {\n await new Promise((resolve, reject) => {\n ws.on('open', resolve);\n ws.on('error', (error) => reject(error));\n ws.on('close', (code) => reject(`WebSocket returned ${code}`));\n });\n\n await this.#runWS(ws);\n } catch (e) {\n if (!this.closed && !this.input.closed) {\n if (retries >= maxRetry) {\n throw new Error(`failed to connect to Baseten after ${retries} attempts: ${e}`);\n }\n\n const delay = Math.min(retries * 5, 10);\n retries++;\n\n this.#logger.warn(\n `failed to connect to Baseten, retrying in ${delay} seconds: ${e} (${retries}/${maxRetry})`,\n );\n await new Promise((resolve) => setTimeout(resolve, delay * 1000));\n } else {\n this.#logger.warn(\n `Baseten disconnected, connection is closed: ${e} (inputClosed: ${this.input.closed}, isClosed: ${this.closed})`,\n );\n }\n }\n }\n\n this.closed = true;\n }\n\n async #runWS(ws: WebSocket) {\n let closing = false;\n\n // Send initial metadata\n // Note: Baseten server expects 'vad_params' and 'streaming_whisper_params' field names\n // (not 'streaming_vad_config', 'streaming_params', 'whisper_params' as in older versions)\n const metadata = {\n vad_params: {\n threshold: this.#opts.vadThreshold,\n min_silence_duration_ms: this.#opts.vadMinSilenceDurationMs,\n speech_pad_ms: this.#opts.vadSpeechPadMs,\n },\n streaming_whisper_params: {\n encoding: this.#opts.encoding ?? 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'en',\n show_word_timestamps: true,\n },\n };\n\n ws.send(JSON.stringify(metadata));\n\n const sendTask = async () => {\n const sampleRate = this.#opts.sampleRate ?? 16000;\n const samplesPerChunk = sampleRate === 16000 ? 512 : 256;\n const audioByteStream = new AudioByteStream(sampleRate, 1, samplesPerChunk);\n\n try {\n while (!this.closed) {\n const result = await this.input.next();\n if (result.done) {\n break;\n }\n\n const data = result.value;\n\n let frames: AudioFrame[];\n if (data === SpeechStream.FLUSH_SENTINEL) {\n // Flush any remaining buffered audio\n frames = audioByteStream.flush();\n } else {\n if (data.sampleRate !== sampleRate || data.channels !== 1) {\n throw new Error(\n `sample rate or channel count mismatch: expected ${sampleRate}Hz/1ch, got ${data.sampleRate}Hz/${data.channels}ch`,\n );\n }\n frames = audioByteStream.write(data.data.buffer as ArrayBuffer);\n }\n\n for (const frame of frames) {\n const buffer = Buffer.from(\n frame.data.buffer,\n frame.data.byteOffset,\n frame.data.byteLength,\n );\n ws.send(buffer);\n }\n }\n } finally {\n closing = true;\n ws.close();\n }\n };\n\n const listenTask = Task.from(async (controller) => {\n const listenMessage = new Promise<void>((resolve, reject) => {\n ws.on('message', (data) => {\n try {\n let jsonString: string;\n\n if (typeof data === 'string') {\n jsonString = data;\n } else if (data instanceof Buffer) {\n jsonString = data.toString('utf-8');\n } else if (Array.isArray(data)) {\n jsonString = Buffer.concat(data).toString('utf-8');\n } else {\n return;\n }\n\n const msg = JSON.parse(jsonString);\n const isFinal = msg.is_final ?? true;\n const segments = msg.segments ?? 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package/dist/stt.js CHANGED
@@ -93,7 +93,7 @@ class SpeechStream extends stt.SpeechStream {
93
93
  const headers = {
94
94
  Authorization: `Api-Key ${this.#opts.apiKey}`
95
95
  };
96
- const ws = new WebSocket(url, { headers, rejectUnauthorized: false });
96
+ const ws = new WebSocket(url, { headers });
97
97
  try {
98
98
  await new Promise((resolve, reject) => {
99
99
  ws.on("open", resolve);
package/dist/stt.js.map CHANGED
@@ -1 +1 @@
1
- {"version":3,"sources":["../src/stt.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2024 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport {\n type AudioBuffer,\n AudioByteStream,\n Task,\n log,\n normalizeLanguage,\n stt,\n waitForAbort,\n} from '@livekit/agents';\nimport type { AudioFrame } from '@livekit/rtc-node';\nimport { WebSocket } from 'ws';\nimport type { BasetenSttOptions } from './types.js';\n\nconst defaultSTTOptions: Partial<BasetenSttOptions> = {\n environment: 'production',\n encoding: 'pcm_s16le',\n sampleRate: 16000,\n bufferSizeSeconds: 0.032,\n enablePartialTranscripts: true,\n partialTranscriptIntervalS: 0.5,\n finalTranscriptMaxDurationS: 5,\n audioLanguage: 'en',\n languageDetectionOnly: false,\n vadThreshold: 0.5,\n vadMinSilenceDurationMs: 300,\n vadSpeechPadMs: 30,\n};\n\nexport class STT extends stt.STT {\n #opts: BasetenSttOptions;\n #logger = log();\n label = 'baseten.STT';\n\n constructor(opts: Partial<BasetenSttOptions> = {}) {\n super({\n streaming: true,\n interimResults: opts.enablePartialTranscripts ?? defaultSTTOptions.enablePartialTranscripts!,\n alignedTranscript: 'word',\n });\n\n const apiKey = opts.apiKey ?? process.env.BASETEN_API_KEY;\n const modelEndpoint = opts.modelEndpoint ?? process.env.BASETEN_MODEL_ENDPOINT;\n const modelId = opts.modelId ?? process.env.BASETEN_STT_MODEL_ID;\n\n if (!apiKey) {\n throw new Error(\n 'Baseten API key is required, either pass it as `apiKey` or set $BASETEN_API_KEY',\n );\n }\n if (!modelEndpoint && !modelId) {\n throw new Error(\n 'Baseten model endpoint is required, either pass it as `modelEndpoint` or set $BASETEN_MODEL_ENDPOINT',\n );\n }\n\n this.#opts = {\n ...defaultSTTOptions,\n ...opts,\n apiKey,\n modelEndpoint,\n modelId,\n audioLanguage: normalizeLanguage((opts.audioLanguage ?? defaultSTTOptions.audioLanguage)!),\n } as BasetenSttOptions;\n }\n\n // eslint-disable-next-line\n async _recognize(_: AudioBuffer): Promise<stt.SpeechEvent> {\n throw new Error('Recognize is not supported on Baseten STT');\n }\n\n updateOptions(opts: Partial<BasetenSttOptions>) {\n this.#opts = {\n ...this.#opts,\n ...opts,\n audioLanguage:\n opts.audioLanguage !== undefined\n ? normalizeLanguage(opts.audioLanguage)\n : this.#opts.audioLanguage,\n };\n }\n\n stream(): SpeechStream {\n return new SpeechStream(this, this.#opts);\n }\n}\n\nexport class SpeechStream extends stt.SpeechStream {\n #opts: BasetenSttOptions;\n #logger = log();\n #speaking = false;\n #requestId = '';\n label = 'baseten.SpeechStream';\n\n constructor(stt: STT, opts: BasetenSttOptions) {\n super(stt, opts.sampleRate);\n this.#opts = opts;\n this.closed = false;\n }\n\n private getWsUrl(): string {\n if (this.#opts.modelEndpoint) {\n return this.#opts.modelEndpoint;\n }\n // Fallback to constructing URL from modelId (deprecated)\n return `wss://model-${this.#opts.modelId}.api.baseten.co/environments/${this.#opts.environment}/websocket`;\n }\n\n protected async run() {\n const maxRetry = 32;\n let retries = 0;\n\n while (!this.input.closed && !this.closed) {\n const url = this.getWsUrl();\n const headers = {\n Authorization: `Api-Key ${this.#opts.apiKey}`,\n };\n\n const ws = new WebSocket(url, { headers, rejectUnauthorized: false });\n\n try {\n await new Promise((resolve, reject) => {\n ws.on('open', resolve);\n ws.on('error', (error) => reject(error));\n ws.on('close', (code) => reject(`WebSocket returned ${code}`));\n });\n\n await this.#runWS(ws);\n } catch (e) {\n if (!this.closed && !this.input.closed) {\n if (retries >= maxRetry) {\n throw new Error(`failed to connect to Baseten after ${retries} attempts: ${e}`);\n }\n\n const delay = Math.min(retries * 5, 10);\n retries++;\n\n this.#logger.warn(\n `failed to connect to Baseten, retrying in ${delay} seconds: ${e} (${retries}/${maxRetry})`,\n );\n await new Promise((resolve) => setTimeout(resolve, delay * 1000));\n } else {\n this.#logger.warn(\n `Baseten disconnected, connection is closed: ${e} (inputClosed: ${this.input.closed}, isClosed: ${this.closed})`,\n );\n }\n }\n }\n\n this.closed = true;\n }\n\n async #runWS(ws: WebSocket) {\n let closing = false;\n\n // Send initial metadata\n // Note: Baseten server expects 'vad_params' and 'streaming_whisper_params' field names\n // (not 'streaming_vad_config', 'streaming_params', 'whisper_params' as in older versions)\n const metadata = {\n vad_params: {\n threshold: this.#opts.vadThreshold,\n min_silence_duration_ms: this.#opts.vadMinSilenceDurationMs,\n speech_pad_ms: this.#opts.vadSpeechPadMs,\n },\n streaming_whisper_params: {\n encoding: this.#opts.encoding ?? 'pcm_s16le',\n sample_rate: this.#opts.sampleRate ?? 16000,\n enable_partial_transcripts: false,\n audio_language: this.#opts.audioLanguage ?? 'en',\n show_word_timestamps: true,\n },\n };\n\n ws.send(JSON.stringify(metadata));\n\n const sendTask = async () => {\n const sampleRate = this.#opts.sampleRate ?? 16000;\n const samplesPerChunk = sampleRate === 16000 ? 512 : 256;\n const audioByteStream = new AudioByteStream(sampleRate, 1, samplesPerChunk);\n\n try {\n while (!this.closed) {\n const result = await this.input.next();\n if (result.done) {\n break;\n }\n\n const data = result.value;\n\n let frames: AudioFrame[];\n if (data === SpeechStream.FLUSH_SENTINEL) {\n // Flush any remaining buffered audio\n frames = audioByteStream.flush();\n } else {\n if (data.sampleRate !== sampleRate || data.channels !== 1) {\n throw new Error(\n `sample rate or channel count mismatch: expected ${sampleRate}Hz/1ch, got ${data.sampleRate}Hz/${data.channels}ch`,\n );\n }\n frames = audioByteStream.write(data.data.buffer as ArrayBuffer);\n }\n\n for (const frame of frames) {\n const buffer = Buffer.from(\n frame.data.buffer,\n frame.data.byteOffset,\n frame.data.byteLength,\n );\n ws.send(buffer);\n }\n }\n } finally {\n closing = true;\n ws.close();\n }\n };\n\n const listenTask = Task.from(async (controller) => {\n const listenMessage = new Promise<void>((resolve, reject) => {\n ws.on('message', (data) => {\n try {\n let jsonString: string;\n\n if (typeof data === 'string') {\n jsonString = data;\n } else if (data instanceof Buffer) {\n jsonString = data.toString('utf-8');\n } else if (Array.isArray(data)) {\n jsonString = Buffer.concat(data).toString('utf-8');\n } else {\n return;\n }\n\n const msg = JSON.parse(jsonString);\n const isFinal = msg.is_final ?? true;\n const segments = msg.segments ?? [];\n const transcript = msg.transcript ?? '';\n const confidence = msg.confidence ?? 0.0;\n const languageCode = normalizeLanguage(msg.language_code ?? this.#opts.audioLanguage);\n\n // Skip if no transcript text\n if (!transcript) {\n this.#logger.debug('Received non-transcript message:', msg);\n return;\n }\n\n // Emit START_OF_SPEECH if not already speaking (only for interim or first final)\n if (!this.#speaking && !isFinal) {\n this.#speaking = true;\n this.queue.put({ type: stt.SpeechEventType.START_OF_SPEECH });\n }\n\n // Note: Baseten uses 'start_time' and 'end_time' field names (with underscores)\n const startTime =\n segments.length > 0\n ? (segments[0].start_time ?? 0.0) + this.startTimeOffset\n : this.startTimeOffset;\n const endTime =\n segments.length > 0\n ? (segments[segments.length - 1].end_time ?? 0.0) + this.startTimeOffset\n : this.startTimeOffset;\n\n // Note: Baseten returns segments (chunks) which we treat as words for aligned transcripts\n const words = segments.map(\n (segment: { text?: string; start_time?: number; end_time?: number }) => ({\n text: segment.text ?? '',\n startTime: (segment.start_time ?? 0.0) + this.startTimeOffset,\n endTime: (segment.end_time ?? 0.0) + this.startTimeOffset,\n startTimeOffset: this.startTimeOffset,\n confidence: confidence,\n }),\n );\n\n const speechData: stt.SpeechData = {\n language: languageCode,\n text: transcript,\n startTime,\n endTime,\n confidence,\n words: words.length > 0 ? words : undefined,\n };\n\n // Handle interim vs final transcripts (matching Python implementation)\n if (!isFinal) {\n // Interim transcript\n this.queue.put({\n type: stt.SpeechEventType.INTERIM_TRANSCRIPT,\n alternatives: [speechData],\n });\n } else {\n // Final transcript\n this.queue.put({\n type: stt.SpeechEventType.FINAL_TRANSCRIPT,\n alternatives: [speechData],\n });\n\n // Emit END_OF_SPEECH after final transcript\n if (this.#speaking) {\n this.#speaking = false;\n this.queue.put({ type: stt.SpeechEventType.END_OF_SPEECH });\n }\n }\n\n if (this.closed || closing) {\n resolve();\n }\n } catch (err) {\n this.#logger.error(`STT: Error processing message: ${data}`);\n reject(err);\n }\n });\n\n ws.on('error', (err) => {\n if (!closing) {\n reject(err);\n }\n });\n\n ws.on('close', () => {\n if (!closing) {\n resolve();\n }\n });\n });\n\n await Promise.race([listenMessage, waitForAbort(controller.signal)]);\n }, this.abortController);\n\n await Promise.all([sendTask(), listenTask.result]);\n closing = true;\n ws.close();\n 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+ {"version":3,"sources":["../src/stt.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2024 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport {\n type AudioBuffer,\n AudioByteStream,\n Task,\n log,\n normalizeLanguage,\n stt,\n waitForAbort,\n} from '@livekit/agents';\nimport type { AudioFrame } from '@livekit/rtc-node';\nimport { WebSocket } from 'ws';\nimport type { BasetenSttOptions } from './types.js';\n\nconst defaultSTTOptions: Partial<BasetenSttOptions> = {\n environment: 'production',\n encoding: 'pcm_s16le',\n sampleRate: 16000,\n bufferSizeSeconds: 0.032,\n enablePartialTranscripts: true,\n partialTranscriptIntervalS: 0.5,\n finalTranscriptMaxDurationS: 5,\n audioLanguage: 'en',\n languageDetectionOnly: false,\n vadThreshold: 0.5,\n vadMinSilenceDurationMs: 300,\n vadSpeechPadMs: 30,\n};\n\nexport class STT extends stt.STT {\n #opts: BasetenSttOptions;\n #logger = log();\n label = 'baseten.STT';\n\n constructor(opts: Partial<BasetenSttOptions> = {}) {\n super({\n streaming: true,\n interimResults: opts.enablePartialTranscripts ?? defaultSTTOptions.enablePartialTranscripts!,\n alignedTranscript: 'word',\n });\n\n const apiKey = opts.apiKey ?? process.env.BASETEN_API_KEY;\n const modelEndpoint = opts.modelEndpoint ?? process.env.BASETEN_MODEL_ENDPOINT;\n const modelId = opts.modelId ?? process.env.BASETEN_STT_MODEL_ID;\n\n if (!apiKey) {\n throw new Error(\n 'Baseten API key is required, either pass it as `apiKey` or set $BASETEN_API_KEY',\n );\n }\n if (!modelEndpoint && !modelId) {\n throw new Error(\n 'Baseten model endpoint is required, either pass it as `modelEndpoint` or set $BASETEN_MODEL_ENDPOINT',\n );\n }\n\n this.#opts = {\n ...defaultSTTOptions,\n ...opts,\n apiKey,\n modelEndpoint,\n modelId,\n audioLanguage: normalizeLanguage((opts.audioLanguage ?? defaultSTTOptions.audioLanguage)!),\n } as BasetenSttOptions;\n }\n\n // eslint-disable-next-line\n async _recognize(_: AudioBuffer): Promise<stt.SpeechEvent> {\n throw new Error('Recognize is not supported on Baseten STT');\n }\n\n updateOptions(opts: Partial<BasetenSttOptions>) {\n this.#opts = {\n ...this.#opts,\n ...opts,\n audioLanguage:\n opts.audioLanguage !== undefined\n ? normalizeLanguage(opts.audioLanguage)\n : this.#opts.audioLanguage,\n };\n }\n\n stream(): SpeechStream {\n return new SpeechStream(this, this.#opts);\n }\n}\n\nexport class SpeechStream extends stt.SpeechStream {\n #opts: BasetenSttOptions;\n #logger = log();\n #speaking = false;\n #requestId = '';\n label = 'baseten.SpeechStream';\n\n constructor(stt: STT, opts: BasetenSttOptions) {\n super(stt, opts.sampleRate);\n this.#opts = opts;\n this.closed = false;\n }\n\n private getWsUrl(): string {\n if (this.#opts.modelEndpoint) {\n return this.#opts.modelEndpoint;\n }\n // Fallback to constructing URL from modelId (deprecated)\n return `wss://model-${this.#opts.modelId}.api.baseten.co/environments/${this.#opts.environment}/websocket`;\n }\n\n protected async run() {\n const maxRetry = 32;\n let retries = 0;\n\n while (!this.input.closed && !this.closed) {\n const url = this.getWsUrl();\n const headers = {\n Authorization: `Api-Key ${this.#opts.apiKey}`,\n };\n\n const ws = new WebSocket(url, { headers });\n\n try {\n await new Promise((resolve, reject) => {\n ws.on('open', resolve);\n ws.on('error', (error) => reject(error));\n ws.on('close', (code) => reject(`WebSocket returned ${code}`));\n });\n\n await this.#runWS(ws);\n } catch (e) {\n if (!this.closed && !this.input.closed) {\n if (retries >= maxRetry) {\n throw new Error(`failed to connect to Baseten after ${retries} attempts: ${e}`);\n }\n\n const delay = Math.min(retries * 5, 10);\n retries++;\n\n this.#logger.warn(\n `failed to connect to Baseten, retrying in ${delay} seconds: ${e} (${retries}/${maxRetry})`,\n );\n await new Promise((resolve) => setTimeout(resolve, delay * 1000));\n } else {\n this.#logger.warn(\n `Baseten disconnected, connection is closed: ${e} (inputClosed: ${this.input.closed}, isClosed: ${this.closed})`,\n );\n }\n }\n }\n\n this.closed = true;\n }\n\n async #runWS(ws: WebSocket) {\n let closing = false;\n\n // Send initial metadata\n // Note: Baseten server expects 'vad_params' and 'streaming_whisper_params' field names\n // (not 'streaming_vad_config', 'streaming_params', 'whisper_params' as in older versions)\n const metadata = {\n vad_params: {\n threshold: this.#opts.vadThreshold,\n min_silence_duration_ms: this.#opts.vadMinSilenceDurationMs,\n speech_pad_ms: this.#opts.vadSpeechPadMs,\n },\n streaming_whisper_params: {\n encoding: this.#opts.encoding ?? 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'en',\n show_word_timestamps: true,\n },\n };\n\n ws.send(JSON.stringify(metadata));\n\n const sendTask = async () => {\n const sampleRate = this.#opts.sampleRate ?? 16000;\n const samplesPerChunk = sampleRate === 16000 ? 512 : 256;\n const audioByteStream = new AudioByteStream(sampleRate, 1, samplesPerChunk);\n\n try {\n while (!this.closed) {\n const result = await this.input.next();\n if (result.done) {\n break;\n }\n\n const data = result.value;\n\n let frames: AudioFrame[];\n if (data === SpeechStream.FLUSH_SENTINEL) {\n // Flush any remaining buffered audio\n frames = audioByteStream.flush();\n } else {\n if (data.sampleRate !== sampleRate || data.channels !== 1) {\n throw new Error(\n `sample rate or channel count mismatch: expected ${sampleRate}Hz/1ch, got ${data.sampleRate}Hz/${data.channels}ch`,\n );\n }\n frames = audioByteStream.write(data.data.buffer as ArrayBuffer);\n }\n\n for (const frame of frames) {\n const buffer = Buffer.from(\n frame.data.buffer,\n frame.data.byteOffset,\n frame.data.byteLength,\n );\n ws.send(buffer);\n }\n }\n } finally {\n closing = true;\n ws.close();\n }\n };\n\n const listenTask = Task.from(async (controller) => {\n const listenMessage = new Promise<void>((resolve, reject) => {\n ws.on('message', (data) => {\n try {\n let jsonString: string;\n\n if (typeof data === 'string') {\n jsonString = data;\n } else if (data instanceof Buffer) {\n jsonString = data.toString('utf-8');\n } else if (Array.isArray(data)) {\n jsonString = Buffer.concat(data).toString('utf-8');\n } else {\n return;\n }\n\n const msg = JSON.parse(jsonString);\n const isFinal = msg.is_final ?? true;\n const segments = msg.segments ?? [];\n const transcript = msg.transcript ?? '';\n const confidence = msg.confidence ?? 0.0;\n const languageCode = normalizeLanguage(msg.language_code ?? this.#opts.audioLanguage);\n\n // Skip if no transcript text\n if (!transcript) {\n this.#logger.debug('Received non-transcript message:', msg);\n return;\n }\n\n // Emit START_OF_SPEECH if not already speaking (only for interim or first final)\n if (!this.#speaking && !isFinal) {\n this.#speaking = true;\n this.queue.put({ type: stt.SpeechEventType.START_OF_SPEECH });\n }\n\n // Note: Baseten uses 'start_time' and 'end_time' field names (with underscores)\n const startTime =\n segments.length > 0\n ? (segments[0].start_time ?? 0.0) + this.startTimeOffset\n : this.startTimeOffset;\n const endTime =\n segments.length > 0\n ? 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package/package.json CHANGED
@@ -1,6 +1,6 @@
1
1
  {
2
2
  "name": "@livekit/agents-plugin-baseten",
3
- "version": "1.4.1",
3
+ "version": "1.4.3",
4
4
  "description": "Baseten plugin for LiveKit Node Agents",
5
5
  "main": "dist/index.js",
6
6
  "require": "dist/index.cjs",
@@ -40,9 +40,9 @@
40
40
  "@types/ws": "^8.5.10",
41
41
  "tsx": "^4.7.0",
42
42
  "typescript": "^5.9.3",
43
- "@livekit/agents": "1.4.1",
44
- "@livekit/agents-plugin-silero": "1.4.1",
45
- "@livekit/agents-plugins-test": "1.4.1"
43
+ "@livekit/agents": "1.4.3",
44
+ "@livekit/agents-plugin-silero": "1.4.3",
45
+ "@livekit/agents-plugins-test": "1.4.3"
46
46
  },
47
47
  "dependencies": {
48
48
  "dotenv": "^17.2.3",
@@ -51,7 +51,7 @@
51
51
  },
52
52
  "peerDependencies": {
53
53
  "@livekit/rtc-node": "^0.13.27",
54
- "@livekit/agents": "1.4.1"
54
+ "@livekit/agents": "1.4.3"
55
55
  },
56
56
  "scripts": {
57
57
  "build": "tsup --onSuccess \"pnpm build:types\"",
package/src/stt.ts CHANGED
@@ -118,7 +118,7 @@ export class SpeechStream extends stt.SpeechStream {
118
118
  Authorization: `Api-Key ${this.#opts.apiKey}`,
119
119
  };
120
120
 
121
- const ws = new WebSocket(url, { headers, rejectUnauthorized: false });
121
+ const ws = new WebSocket(url, { headers });
122
122
 
123
123
  try {
124
124
  await new Promise((resolve, reject) => {