Whisper Transcription

Transcribe audio and video files using OpenAI's Whisper on CLORE.AI GPUs.

All examples can be run on GPU servers rented through CLORE.AI Marketplace.

Server Requirements

Parameter	Minimum	Recommended
RAM	8GB	16GB+
VRAM	4GB (small)	10GB+ (large-v3)
Network	200Mbps	500Mbps+
Startup Time	~1-2 minutes	-

What is Whisper?

OpenAI Whisper is a speech recognition model that can:

• Transcribe audio in 99 languages

• Translate to English

• Generate timestamps

• Handle noisy audio

Model Sizes

Model	VRAM	Speed	Quality
tiny	1GB	~32x realtime	Basic
base	1GB	~16x realtime	Good
small	2GB	~6x realtime	Better
medium	5GB	~2x realtime	Great
large-v3	10GB	~1x realtime	Best

Quick Deploy (Recommended)

Use the pre-built Faster-Whisper server for instant deployment:

Docker Image:

fedirz/faster-whisper-server:latest-cuda

Ports:

22/tcp
8000/http

No command needed - server starts automatically.

Verify It's Working

After deployment, find your http_pub URL in My Orders and test:

curl https://your-http-pub.clorecloud.net/

# Expected: server info JSON response

If you get HTTP 502, wait 1-2 minutes - the service is still starting.

Transcribe via API

# Transcribe audio file
curl -X POST https://your-http-pub.clorecloud.net/v1/audio/transcriptions \
  -F "[email protected]" \
  -F "model=Systran/faster-whisper-large-v3" \
  -F "response_format=json"

# With timestamps
curl -X POST https://your-http-pub.clorecloud.net/v1/audio/transcriptions \
  -F "[email protected]" \
  -F "model=Systran/faster-whisper-large-v3" \
  -F "response_format=verbose_json" \
  -F "timestamp_granularities[]=word"

Complete API Reference (Faster-Whisper-Server)

Endpoints

Endpoint	Method	Description
/v1/audio/transcriptions	POST	Transcribe audio (OpenAI-compatible)
/v1/audio/translations	POST	Translate audio to English
/v1/models	GET	List all available models
/v1/models/{model_name}	GET	Get specific model info
/api/ps	GET	List currently loaded models
/api/ps/{model_name}	GET	Check if specific model is loaded
/api/pull/{model_name}	POST	Download and load a model
/health	GET	Health check endpoint
/docs	GET	Swagger UI documentation
/openapi.json	GET	OpenAPI specification

List Available Models

curl https://your-http-pub.clorecloud.net/v1/models

Response:

{
  "data": [
    {"id": "Systran/faster-whisper-large-v3", "object": "model"},
    {"id": "Systran/faster-whisper-medium", "object": "model"}
  ]
}

Swagger Documentation

Open in browser for interactive API testing:

https://your-http-pub.clorecloud.net/docs

Transcription Options

Parameter	Type	Description
file	File	Audio file to transcribe
model	String	Model to use (default: Systran/faster-whisper-large-v3)
language	String	Force specific language (e.g., en, ja, ru)
response_format	String	json, text, srt, vtt, verbose_json
temperature	Float	Sampling temperature (0.0-1.0)
timestamp_granularities[]	Array	word or segment for timestamps

Response Formats

JSON (default):

{"text": "Transcribed text here..."}

Verbose JSON:

{
  "text": "Full transcription...",
  "segments": [
    {"start": 0.0, "end": 2.5, "text": "First segment"},
    {"start": 2.5, "end": 5.0, "text": "Second segment"}
  ],
  "language": "en"
}

SRT:

1
00:00:00,000 --> 00:00:02,500
First segment

2
00:00:02,500 --> 00:00:05,000
Second segment

Alternative: Manual Installation

If you need more control, deploy with manual installation:

Docker Image:

pytorch/pytorch:2.5.1-cuda12.4-cudnn9-runtime

Ports:

22/tcp

Command:

pip install openai-whisper faster-whisper

Manual installation takes 3-5 minutes. The pre-built image above is recommended for faster startup.

Basic Usage (SSH)

ssh -p <port> root@<proxy>

# Transcribe audio file
whisper audio.mp3 --model large-v3 --device cuda

# Output to specific format
whisper audio.mp3 --model large-v3 --output_format txt

# Specify language
whisper audio.mp3 --model large-v3 --language Japanese

Transcribe with Timestamps

whisper audio.mp3 --model large-v3 --word_timestamps True

Upload Audio Files

# Upload single file
scp -P <port> interview.mp3 root@<proxy>:/workspace/

# Upload folder
scp -P <port> -r ./audio_files/ root@<proxy>:/workspace/

Python API

import whisper

# Load model (downloads on first use)
model = whisper.load_model("large-v3", device="cuda")

# Transcribe
result = model.transcribe("audio.mp3")

# Print text
print(result["text"])

# Print with timestamps
for segment in result["segments"]:
    print(f"[{segment['start']:.2f}s -> {segment['end']:.2f}s] {segment['text']}")

Faster-Whisper (Recommended)

Faster-Whisper is 4x faster with lower VRAM usage:

pip install faster-whisper

from faster_whisper import WhisperModel

# Load model with optimizations
model = WhisperModel("large-v3", device="cuda", compute_type="float16")

# Transcribe
segments, info = model.transcribe("audio.mp3")

print(f"Detected language: {info.language}")
for segment in segments:
    print(f"[{segment.start:.2f}s -> {segment.end:.2f}s] {segment.text}")

Language Options

from faster_whisper import WhisperModel

model = WhisperModel("large-v3", device="cuda")

# Auto-detect language
segments, info = model.transcribe("audio.mp3")
print(f"Language: {info.language} ({info.language_probability:.0%})")

# Force specific language
segments, _ = model.transcribe("audio.mp3", language="ja")

Translation to English

from faster_whisper import WhisperModel

model = WhisperModel("large-v3", device="cuda")
segments, _ = model.transcribe("japanese.mp3", task="translate")

for segment in segments:
    print(segment.text)

CLI:

whisper japanese.mp3 --model large-v3 --task translate

Subtitle Generation

SRT Format

from faster_whisper import WhisperModel

def format_timestamp(seconds):
    hours = int(seconds // 3600)
    minutes = int((seconds % 3600) // 60)
    secs = int(seconds % 60)
    millis = int((seconds - int(seconds)) * 1000)
    return f"{hours:02d}:{minutes:02d}:{secs:02d},{millis:03d}"

model = WhisperModel("large-v3", device="cuda")
segments, _ = model.transcribe("video.mp4")

with open("subtitles.srt", "w") as f:
    for i, segment in enumerate(segments, 1):
        f.write(f"{i}\n")
        f.write(f"{format_timestamp(segment.start)} --> {format_timestamp(segment.end)}\n")
        f.write(f"{segment.text.strip()}\n\n")

VTT Format

whisper video.mp4 --model large-v3 --output_format vtt

Word-Level Timestamps

from faster_whisper import WhisperModel

model = WhisperModel("large-v3", device="cuda")

segments, _ = model.transcribe("audio.mp3", word_timestamps=True)

for segment in segments:
    for word in segment.words:
        print(f"[{word.start:.2f}s] {word.word}")

Speaker Diarization

Who said what (requires pyannote):

pip install pyannote.audio

from faster_whisper import WhisperModel
from pyannote.audio import Pipeline

# Diarization
diarization = Pipeline.from_pretrained(
    "pyannote/speaker-diarization-3.1",
    use_auth_token="YOUR_HF_TOKEN"
)
diarization_result = diarization("audio.mp3")

# Transcription
whisper = WhisperModel("large-v3", device="cuda")
segments, _ = whisper.transcribe("audio.mp3")

# Combine
for segment in segments:
    # Find speaker at segment time
    speaker = None
    for turn, _, spk in diarization_result.itertracks(yield_label=True):
        if turn.start <= segment.start <= turn.end:
            speaker = spk
            break
    print(f"[{speaker}] {segment.text}")

REST API Server

Create a transcription API:

from fastapi import FastAPI, UploadFile
from faster_whisper import WhisperModel
import tempfile

app = FastAPI()
model = WhisperModel("large-v3", device="cuda", compute_type="float16")

@app.post("/transcribe")
async def transcribe(file: UploadFile):
    with tempfile.NamedTemporaryFile(delete=False) as tmp:
        tmp.write(await file.read())
        tmp_path = tmp.name

    segments, info = model.transcribe(tmp_path)

    return {
        "language": info.language,
        "text": " ".join([s.text for s in segments]),
        "segments": [
            {"start": s.start, "end": s.end, "text": s.text}
            for s in segments
        ]
    }

# Run: uvicorn server:app --host 0.0.0.0 --port 8000

Performance Benchmarks

Model	GPU	1hr Audio
large-v3	RTX 3090	~5 min
large-v3	RTX 4090	~3 min
large-v3	A100	~2 min
medium	RTX 3090	~2 min

Memory-Efficient Processing

For very long audio:

from faster_whisper import WhisperModel

model = WhisperModel("large-v3", device="cuda", compute_type="int8")

# Process in chunks
segments, _ = model.transcribe(
    "long_audio.mp3",
    vad_filter=True,  # Skip silence
    vad_parameters=dict(min_silence_duration_ms=500)
)

Download Results

# Download transcripts
scp -P <port> -r root@<proxy>:/workspace/transcripts/ ./

# Download subtitles
scp -P <port> root@<proxy>:/workspace/subtitles.srt ./

Troubleshooting

CUDA out of memory

• Use smaller model (medium instead of large)

• Use compute_type="int8" for faster-whisper

• Process shorter audio segments

HTTP 502 on http_pub URL

The service is still starting. Wait 1-2 minutes and retry:

curl https://your-http-pub.clorecloud.net/

Poor accuracy

• Use larger model

• Specify language: --language English

• Increase beam_size for faster-whisper

Slow processing

• Ensure GPU is used: nvidia-smi

• Use faster-whisper instead of original

• Enable VAD to skip silence

Cost Estimate

Typical CLORE.AI marketplace rates:

GPU	CLORE/day	Approx USD/hr	Good For
RTX 3060	~80	~$0.02	small/medium models
RTX 3090	~150	~$0.03	large-v3
RTX 4090	~200	~$0.04	large-v3, fast
A100 40GB	~400	~$0.08	batch processing

Prices vary by provider. Check CLORE.AI Marketplace for current rates. Pay with CLORE tokens for best value.