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Deepgram performance tuning

Skill ComeOnOliver/skillshub/skills/jeremylongshore/claude-code-plugins-plus-skills/deepgram-performance-tuning

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npx -y skills add ComeOnOliver/skillshub --skill deepgram-performance-tuning

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Optimize Deepgram API performance for faster transcription and lower latency. Use when improving transcription speed, reducing latency, or optimizing audio processing pipelines. Trigger: "deepgram performance", "speed up deepgram", "optimize transcription", "deepgram latency", "deepgram faster", "deepgram throughput".

The file declares its own license as MIT. That is the author’s claim about this one file, and it is not the same thing as the license GitHub reports for the repository, which is listed with the other numbers below.

SKILL.md

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Deepgram Performance Tuning

Overview

Optimize Deepgram transcription performance through audio preprocessing with ffmpeg, model selection for speed vs accuracy, streaming for large files, parallel processing, result caching, and connection reuse. Targets: <2s latency for short files, 100+ files/minute batch throughput.

Performance Levers

FactorImpactDefaultOptimized
Audio formatHighAny format16kHz mono WAV
ModelHighnova-3base (speed) or nova-3 (accuracy)
File sizeHighFull file syncStream >60s, callback >5min
ConcurrencyMediumSequential50 parallel (p-limit)
CachingMediumNoneRedis hash by audio+options
FeaturesMediumAll enabledDisable unused (diarize, utterances)

Instructions

Step 1: Audio Preprocessing with ffmpeg

# Optimal format for Deepgram: 16kHz, 16-bit, mono, WAV
ffmpeg -i input.mp3 \
  -ar 16000 \          # 16kHz sample rate (ideal for speech)
  -ac 1 \              # Mono channel
  -acodec pcm_s16le \  # 16-bit signed LE PCM
  -f wav \
  output.wav

# Remove silence (saves API cost + processing time)
ffmpeg -i input.wav \
  -af "silenceremove=stop_periods=-1:stop_duration=0.5:stop_threshold=-30dB" \
  -ar 16000 -ac 1 -acodec pcm_s16le \
  trimmed.wav

# Noise reduction + normalization
ffmpeg -i input.wav \
  -af "highpass=f=200,lowpass=f=3000,loudnorm=I=-16:TP=-1.5:LRA=11" \
  -ar 16000 -ac 1 -acodec pcm_s16le \
  clean.wav
import { execSync } from 'child_process';
import { statSync } from 'fs';

function preprocessAudio(inputPath: string, outputPath: string): {
  originalSize: number;
  optimizedSize: number;
  savings: string;
} {
  const originalSize = statSync(inputPath).size;

  execSync(`ffmpeg -y -i "${inputPath}" \
    -af "silenceremove=stop_periods=-1:stop_duration=0.5:stop_threshold=-30dB,\
    highpass=f=200,lowpass=f=3000" \
    -ar 16000 -ac 1 -acodec pcm_s16le \
    "${outputPath}" 2>/dev/null`);

  const optimizedSize = statSync(outputPath).size;
  const savings = ((1 - optimizedSize / originalSize) * 100).toFixed(1);

  console.log(`Preprocessed: ${inputPath}`);
  console.log(`  Original: ${(originalSize / 1024).toFixed(0)}KB`);
  console.log(`  Optimized: ${(optimizedSize / 1024).toFixed(0)}KB (${savings}% smaller)`);

  return { originalSize, optimizedSize, savings };
}

Step 2: Model Selection Strategy

import { createClient } from '@deepgram/sdk';

type Priority = 'accuracy' | 'speed' | 'cost';

function selectModel(priority: Priority, audioDuration: number): string {
  // Nova-3: Best accuracy, fast, $0.0043/min (STT)
  // Nova-2: Proven stable, fast, $0.0043/min
  // Base:   Fastest, lower accuracy, $0.0048/min
  // Whisper: Multilingual (100+ langs), slower, $0.0048/min

  switch (priority) {
    case 'accuracy':
      return 'nova-3';
    case 'speed':
      return audioDuration > 300 ? 'base' : 'nova-2';  // Base for long files
    case 'cost':
      return 'nova-2';  // Same price as Nova-3, slightly faster
    default:
      return 'nova-3';
  }
}

// Feature cost: disable what you don't need
function optimizedOptions(priority: Priority) {
  return {
    model: selectModel(priority, 0),
    smart_format: true,      // Free — always enable
    punctuate: true,         // Free — always enable
    // These add processing time:
    diarize: priority === 'accuracy',   // Adds latency
    utterances: priority === 'accuracy',
    paragraphs: priority === 'accuracy',
    summarize: false,        // Only when needed
    detect_topics: false,    // Only when needed
    sentiment: false,        // Only when needed
  };
}

Step 3: Streaming for Large Files

import { createClient, LiveTranscriptionEvents } from '@deepgram/sdk';
import { createReadStream } from 'fs';

async function streamLargeFile(filePath: string): Promise<string> {
  const deepgram = createClient(process.env.DEEPGRAM_API_KEY!);
  const transcripts: string[] = [];

  return new Promise((resolve, reject) => {
    const connection = deepgram.listen.live({
      model: 'nova-3',
      smart_format: true,
      encoding: 'linear16',
      sample_rate: 16000,
      channels: 1,
    });

    connection.on(LiveTranscriptionEvents.Open, () => {
      // Stream file in 32KB chunks
      const stream = createReadStream(filePath, { highWaterMark: 32 * 1024 });

      stream.on('data', (chunk: Buffer) => {
        connection.send(chunk);
      });

      stream.on('end', () => {
        // Signal end of audio
        connection.finish();
      });

      stream.on('error', reject);
    });

    connection.on(LiveTranscriptionEvents.Transcript, (data) => {
      if (data.is_final) {
        const text = data.channel.alternatives[0]?.transcript;
        if (text) transcripts.push(text);
      }
    });

    connection.on(LiveTranscriptionEvents.Close, () => {
      resolve(transcripts.join(' '));
    });

    connection.on(LiveTranscriptionEvents.Error, reject);
  });
}

Step 4: Parallel Batch Processing

import pLimit from 'p-limit';
import { createClient } from '@deepgram/sdk';

async function batchTranscribe(
  files: string[],
  concurrency = 50,   // Stay under your plan's concurrency limit
  model = 'nova-3'
) {
  const client = createClient(process.env.DEEPGRAM_API_KEY!);
  const limit = pLimit(concurrency);
  const startTime = Date.now();

  const results = await Promise.allSettled(
    files.map((file, i) =>
      limit(async () => {
        const fileStart = Date.now();
        const { result, error } = await client.listen.prerecorded.transcribeFile(
          require('fs').readFileSync(file),
          { model, smart_format: true, mimetype: 'audio/wav' }
        );
        if (error) throw error;

        const elapsed = Date.now() - fileStart;
        console.log(`[${i + 1}/${files.length}] ${file} — ${elapsed}ms (${result.metadata.duration}s audio)`);
        return { file, result, elapsed };
      })
    )
  );

  const totalTime = Date.now() - startTime;
  const succeeded = results.filter(r => r.status === 'fulfilled').length;
  console.log(`\nBatch: ${succeeded}/${files.length} in ${totalTime}ms`);
  console.log(`Throughput: ${(files.length / (totalTime / 60000)).toFixed(1)} files/min`);

  return results;
}

Step 5: Result Caching

import { createHash } from 'crypto';
import Redis from 'ioredis';

const redis = new Redis(process.env.REDIS_URL ?? 'redis://localhost:6379');

function cacheKey(audioUrl: string, options: Record<string, any>): string {
  const hash = createHash('sha256')
    .update(audioUrl + JSON.stringify(options))
    .digest('hex');
  return `dg:cache:${hash}`;
}

async function cachedTranscribe(
  client: ReturnType<typeof createClient>,
  url: string,
  options: Record<string, any>,
  ttlSeconds = 3600  // 1 hour default
) {
  const key = cacheKey(url, options);

  // Check cache
  const cached = await redis.get(key);
  if (cached) {
    console.log('Cache hit:', url.substring(0, 60));
    return JSON.parse(cached);
  }

  // Transcribe and cache
  const { result, error } = await client.listen.prerecorded.transcribeUrl(
    { url }, options
  );
  if (error) throw error;

  await redis.setex(key, ttlSeconds, JSON.stringify(result));
  console.log('Cached result:', url.substring(0, 60));
  return result;
}

Step 6: Performance Benchmarking

async function benchmark(audioUrl: string) {
  const client = createClient(process.env.DEEPGRAM_API_KEY!);
  const models = ['nova-3', 'nova-2', 'base'] as const;

  console.log('Performance Benchmark');
  console.log('='.repeat(60));

  for (const model of models) {
    const times: number[] = [];
    for (let i = 0; i < 3; i++) {
      const start = Date.now();
      const { result, error } = await client.listen.prerecorded.transcribeUrl(
        { url: audioUrl }, { model, smart_format: true }
      );
      times.push(Date.now() - start);
      if (error) { console.error(`${model} error:`, error.message); break; }
    }
    const avg = times.reduce((a, b) => a + b, 0) / times.length;
    console.log(`${model}: avg ${avg.toFixed(0)}ms (${times.map(t => `${t}ms`).join(', ')})`);
  }
}

Output

  • Audio preprocessing pipeline (16kHz mono, silence removal, noise reduction)
  • Model selection strategy by priority (accuracy/speed/cost)
  • Streaming transcription for large files (>60s)
  • Parallel batch processing with configurable concurrency
  • Redis-backed result caching with TTL
  • Performance benchmarking script

Error Handling

IssueCauseSolution
Slow transcriptionUnoptimized audio formatPreprocess to 16kHz mono WAV
429 in batchConcurrency too highReduce p-limit to 50% of plan limit
ffmpeg not foundNot installedapt install ffmpeg / brew install ffmpeg
Cache staleAudio changed at same URLInclude hash of audio content in cache key

Resources

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