Deepgram Performance Tuning
SkillDev tools'Optimize Deepgram API performance for faster transcription and lower
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Details
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What this skill tells your AI
The instructions your AI receives, as published by jeremylongshore/tons-of-skills-marketplace in skills/.curated/deepgram-performance-tuning/SKILL.md and read by Ahel’s review.
Prerequisites
- A baseline for latency, throughput, quality, error/throttle rate, and a named service owner.
- Licensed non-sensitive fixtures, approved load window, and a rollback threshold.
Examples
Measure the development/staging transcription baseline with short fixtures, change one concurrency, streaming, or model parameter, and compare aggregate quality/latency/error results. Keep the change only within the signed threshold; revert on regression and never use customer recordings as performance fixtures.
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
| Factor | Impact | Default | Optimized |
|---|---|---|---|
| Audio format | High | Any format | 16kHz mono WAV |
| Model | High | nova-3 | base (speed) or nova-3 (accuracy) |
| File size | High | Full file sync | Stream >60s, callback >5min |
| Concurrency | Medium | Sequential | 50 parallel (p-limit) |
| Caching | Medium | None | Redis hash by audio+options |
| Features | Medium | All enabled | Disable 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
| Issue | Cause | Solution |
|---|---|---|
| Slow transcription | Unoptimized audio format | Preprocess to 16kHz mono WAV |
| 429 in batch | Concurrency too high | Reduce p-limit to 50% of plan limit |
| ffmpeg not found | Not installed | apt install ffmpeg / brew install ffmpeg |
| Cache stale | Audio changed at same URL | Include hash of audio content in cache key |
Resources
- Audio Best Practices
- Model Options
- Concurrency Limits
Signals
- GitHub stars
- 3k
- Forks
- 415
- Last commit
- Oct 2026
Ahel review
K1binfo
installs-packages
Automated review, not a security audit. Ruleset v1+k2.
Advanced
- Item type
- skill
- Key
deepgram-performance-tuning- Source
- github.com/jeremylongshore/tons-of-skills-marketplace
github.com/jeremylongshore/tons-of-skills-marketplace
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