Working speaker detection with advanced LLM identifying. Improved LLM json function
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This commit is contained in:
2026-02-14 09:39:17 -05:00
parent 0360f2493d
commit 4143d00de7
4 changed files with 90 additions and 56 deletions

View File

@@ -1,14 +1,17 @@
import { pipeline } from '@xenova/transformers';
import { parentPort } from 'worker_threads';
import * as fs from 'node:fs';
import wavefile from 'wavefile';
import { spawn } from 'node:child_process';
import { execSync } from 'node:child_process';
import { mkdtempSync, rmSync, readFileSync } from 'node:fs';
import { join } from 'node:path';
import { tmpdir } from 'node:os';
import wavefile from 'wavefile';
let whisperPipeline: any;
export async function canDiarization(): Promise<boolean> {
return new Promise((resolve) => {
const proc = spawn('python3', ['-c', 'import pyannote.audio']);
const proc = spawn('python', ['-c', 'import pyannote.audio']);
proc.on('close', (code: number) => resolve(code === 0));
proc.on('error', () => resolve(false));
});
@@ -21,25 +24,20 @@ import json
import os
from pyannote.audio import Pipeline
os.environ['TORCH_HOME'] = "${dir}"
os.environ['HF_TOKEN'] = "${token}"
pipeline = Pipeline.from_pretrained("pyannote/speaker-diarization-3.1")
diarization = pipeline(sys.argv[1])
os.environ['TORCH_HOME'] = r"${dir}"
pipeline = Pipeline.from_pretrained("pyannote/speaker-diarization-3.1", token="${token}")
output = pipeline(sys.argv[1])
segments = []
for turn, _, speaker in diarization.itertracks(yield_label=True):
segments.append({
"start": turn.start,
"end": turn.end,
"speaker": speaker
})
for turn, speaker in output.speaker_diarization:
segments.append({"start": turn.start, "end": turn.end, "speaker": speaker})
print(json.dumps(segments))
`;
return new Promise((resolve, reject) => {
let output = '';
const proc = spawn('python3', ['-c', script, audioPath]);
const proc = spawn('python', ['-c', script, audioPath]);
proc.stdout.on('data', (data: Buffer) => output += data.toString());
proc.stderr.on('data', (data: Buffer) => console.error(data.toString()));
proc.on('close', (code: number) => {
@@ -72,55 +70,65 @@ function combineSpeakerTranscript(chunks: any[], speakers: any[]): string {
const speaker = speakers.find((s: any) => time >= s.start && time <= s.end);
const speakerNum = speaker ? speakerMap.get(speaker.speaker) : 1;
if (speakerNum !== currentSpeaker) {
if(currentText) lines.push(`[speaker ${currentSpeaker}]: ${currentText.trim()}`);
if(currentText) lines.push(`[Speaker ${currentSpeaker}]: ${currentText.trim()}`);
currentSpeaker = speakerNum;
currentText = chunk.text;
} else {
currentText += chunk.text;
}
});
if(currentText) lines.push(`[speaker ${currentSpeaker}]: ${currentText.trim()}`);
if(currentText) lines.push(`[Speaker ${currentSpeaker}]: ${currentText.trim()}`);
return lines.join('\n');
}
function prepareAudioBuffer(file: string): [string, Float32Array] {
let wav: any, tmp;
try {
wav = new wavefile.WaveFile(readFileSync(file));
} catch(err) {
tmp = join(mkdtempSync(join(tmpdir(), 'audio-')), 'converted.wav');
execSync(`ffmpeg -i "${file}" -ar 16000 -ac 1 -f wav "${tmp}"`, { stdio: 'ignore' });
wav = new wavefile.WaveFile(readFileSync(tmp));
} finally {
wav.toBitDepth('32f');
wav.toSampleRate(16000);
const samples = wav.getSamples();
if(Array.isArray(samples)) {
const left = samples[0];
const right = samples[1];
const buffer = new Float32Array(left.length);
for (let i = 0; i < left.length; i++) buffer[i] = (left[i] + right[i]) / 2;
return [tmp || file, buffer];
}
return [tmp || file, samples];
}
}
parentPort?.on('message', async ({ file, speaker, model, modelDir, token }) => {
try {
console.log('worker', file);
if(!whisperPipeline) whisperPipeline = await pipeline('automatic-speech-recognition', `Xenova/${model}`, {cache_dir: modelDir, quantized: true});
// Prepare audio file (convert to mono channel wave)
const wav = new wavefile.WaveFile(fs.readFileSync(file));
wav.toBitDepth('32f');
wav.toSampleRate(16000);
const samples = wav.getSamples();
let buffer;
if(Array.isArray(samples)) { // stereo to mono - average the channels
const left = samples[0];
const right = samples[1];
buffer = new Float32Array(left.length);
for (let i = 0; i < left.length; i++) buffer[i] = (left[i] + right[i]) / 2;
} else {
buffer = samples;
}
// Prepare audio file
const [f, buffer] = prepareAudioBuffer(file);
// Transcribe
const transcriptResult = await whisperPipeline(buffer, {return_timestamps: speaker ? 'word' : false});
if(!speaker) {
parentPort?.postMessage({ text: transcriptResult.text?.trim() || null });
return;
}
// Fetch transcript and speakers
const hasDiarization = speaker && await canDiarization();
const [transcript, speakers] = await Promise.all([
whisperPipeline(buffer, {return_timestamps: speaker ? 'word' : false}),
(!speaker || !token || !hasDiarization) ? Promise.resolve(): runDiarization(f, modelDir, token),
]);
if(file != f) rmSync(f, { recursive: true, force: true });
// Speaker Diarization
const hasDiarization = await canDiarization();
if(!token || !hasDiarization) {
parentPort?.postMessage({ text: transcriptResult.text?.trim() || null, error: 'Speaker diarization unavailable' });
return;
}
// Return any results / errors if no more processing required
const text = transcript.text?.trim() || null;
if(!speaker) return parentPort?.postMessage({ text });
if(!token) return parentPort?.postMessage({ text, error: 'HuggingFace token required' });
if(!hasDiarization) return parentPort?.postMessage({ text, error: 'Speaker diarization unavailable' });
const speakers = await runDiarization(file, modelDir, token);
const combined = combineSpeakerTranscript(transcriptResult.chunks || [], speakers);
// Combine transcript and speakers
const combined = combineSpeakerTranscript(transcript.chunks || [], speakers || []);
parentPort?.postMessage({ text: combined });
} catch (err) {
parentPort?.postMessage({ error: (err as Error).message });
} catch (err: any) {
parentPort?.postMessage({ error: err.stack || err.message });
}
});