Commit ·
62e8495
1
Parent(s): 1cdac6f
feat: add Orpheus TTS continuous batching benchmark
Browse files- Implement vLLM AsyncLLMEngine for parallel TTS processing
- Achieve 12x capacity improvement (12.6 real-time users vs 1 sequential)
- RTF improved from 0.85 to 0.079 with 16 concurrent requests
- Add stress test script validating 4/8/12/16 simultaneous users
Test results on RTX 4090:
| Requests | RTF | Real-time Users |
|----------|-------|-----------------|
| 4 | 0.253 | 4.0 |
| 8 | 0.129 | 7.8 |
| 12 | 0.092 | 10.9 |
| 16 | 0.079 | 12.6 |
🤖 Generated with [Claude Code](https://claude.com/claude-code)
Co-Authored-By: Claude Opus 4.5 <noreply@anthropic.com>
- docs/orpheus-tts-benchmark.md +84 -0
- scripts/orpheus_continuous_batching.py +322 -0
- scripts/orpheus_stress_test.py +195 -0
docs/orpheus-tts-benchmark.md
ADDED
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# Orpheus TTS - Benchmark com Continuous Batching
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**Data:** 2024-12-25
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**GPU:** NVIDIA RTX 4090 (24GB VRAM)
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**Modelo:** canopylabs/orpheus-3b-0.1-ft (3B parametros)
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## Resumo
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Implementamos e testamos **Continuous Batching** com vLLM AsyncLLMEngine para o Orpheus TTS, conseguindo **12x mais capacidade** comparado ao uso sequencial.
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## Resultados do Teste de Stress
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| Requests Simultaneos | Taxa Sucesso | Tempo Wall-Clock | Audio Gerado | RTF | Usuarios Real-Time |
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| 14 |
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|---------------------|--------------|------------------|--------------|-----|-------------------|
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| 4 | 100% | 2.93s | 11.61s | 0.253 | 4.0 |
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| 8 | 100% | 2.94s | 22.78s | 0.129 | 7.8 |
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| 12 | 100% | 3.13s | 34.05s | 0.092 | 10.9 |
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| 16 | 100% | 4.03s | 50.77s | 0.079 | **12.6** |
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## Comparacao
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| Metodo | RTF | Usuarios Real-Time | Melhoria |
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|--------|-----|-------------------|----------|
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| Sequencial (orpheus_tts lib) | 0.85 | ~1 | baseline |
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| Continuous Batching (4 req) | 0.253 | 4.0 | 4x |
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| Continuous Batching (16 req) | 0.079 | 12.6 | **12x** |
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## Otimizacoes Ativas (vLLM 0.13.0)
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- Flash Attention 2
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- Chunked Prefill
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- Prefix Caching
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- CUDA Graphs
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- torch.compile
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## Configuracao do Engine
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```python
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engine_args = AsyncEngineArgs(
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model="canopylabs/orpheus-3b-0.1-ft",
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dtype="bfloat16",
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max_model_len=4096,
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gpu_memory_utilization=0.9,
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max_num_seqs=16, # Continuous batching
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enable_chunked_prefill=True,
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enable_prefix_caching=True,
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enforce_eager=False,
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)
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```
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## Problema com a Biblioteca orpheus_tts
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A biblioteca oficial `orpheus_tts` tem um bug onde o engine vLLM morre apos a primeira inferencia (`EngineDeadError`). A solucao e usar o `AsyncLLMEngine` do vLLM diretamente.
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## Formato do Prompt Orpheus
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```python
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# Tokens especiais
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START_TOKEN = 128259
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END_TOKENS = [128009, 128260, 128261, 128257]
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STOP_TOKEN = 128258
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AUDIO_TOKEN_BASE = 128266
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# Formato do prompt
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prompt = f"{voice}: {text}"
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# Depois adiciona os tokens especiais
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```
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## Decodificacao de Audio (SNAC)
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Os tokens de audio sao organizados em frames de 7 tokens:
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- Layer 0: offset 0
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- Layer 1: offset 4096 (2 tokens por frame)
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- Layer 2: offset 8192, 12288, etc. (4 tokens por frame)
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## Arquivos
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- `scripts/orpheus_continuous_batching.py` - Script de teste
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- `scripts/orpheus_stress_test.py` - Teste de stress
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- `orpheus_demo_3_phrases.wav` - Audio de demonstracao
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## Conclusao
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Com Continuous Batching, uma unica RTX 4090 pode suportar **~12-13 usuarios simultaneos em tempo real** para TTS com Orpheus, uma melhoria de **12x** sobre o metodo sequencial.
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scripts/orpheus_continuous_batching.py
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| 1 |
+
"""
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| 2 |
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Orpheus TTS - Continuous Batching Test v3
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| 3 |
+
==========================================
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| 4 |
+
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| 5 |
+
Uses AsyncLLMEngine with correct token decoding based on Axolotl's preprocessing.
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| 6 |
+
Token format per 7-token frame:
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| 7 |
+
[layer0, layer1_a, layer2_a, layer2_b, layer1_b, layer2_c, layer2_d]
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| 8 |
+
Where:
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| 9 |
+
- layer0: 128266 + value
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| 10 |
+
- layer1_a: 128266 + 4096 + value
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| 11 |
+
- layer2_a: 128266 + 2*4096 + value
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| 12 |
+
- layer2_b: 128266 + 3*4096 + value
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| 13 |
+
- layer1_b: 128266 + 4*4096 + value
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| 14 |
+
- layer2_c: 128266 + 5*4096 + value
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| 15 |
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- layer2_d: 128266 + 6*4096 + value
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| 16 |
+
"""
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| 17 |
+
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| 18 |
+
import os
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| 19 |
+
import sys
|
| 20 |
+
import time
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| 21 |
+
import wave
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| 22 |
+
import asyncio
|
| 23 |
+
import numpy as np
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| 24 |
+
|
| 25 |
+
os.environ["VLLM_ATTENTION_BACKEND"] = "FLASH_ATTN"
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| 26 |
+
|
| 27 |
+
import torch
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| 28 |
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from transformers import AutoTokenizer
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| 29 |
+
from vllm import AsyncLLMEngine, AsyncEngineArgs, SamplingParams
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| 30 |
+
from snac import SNAC
|
| 31 |
+
|
| 32 |
+
# Orpheus special tokens
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| 33 |
+
START_TOKEN = 128259
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| 34 |
+
END_TOKENS = [128009, 128260, 128261, 128257]
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| 35 |
+
STOP_TOKEN = 128258
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| 36 |
+
AUDIO_TOKEN_BASE = 128266
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| 37 |
+
|
| 38 |
+
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| 39 |
+
def decode_tokens_to_audio(token_ids, snac_model):
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| 40 |
+
"""Decode Orpheus tokens to audio using SNAC with correct layer offsets."""
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| 41 |
+
# Filter audio tokens and decode by layer
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| 42 |
+
audio_frames = []
|
| 43 |
+
|
| 44 |
+
for t in token_ids:
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| 45 |
+
if isinstance(t, str):
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| 46 |
+
continue
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| 47 |
+
if t >= AUDIO_TOKEN_BASE:
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| 48 |
+
# Determine which layer this token belongs to
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| 49 |
+
offset = t - AUDIO_TOKEN_BASE
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| 50 |
+
layer = offset // 4096
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| 51 |
+
value = offset % 4096
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| 52 |
+
audio_frames.append((layer, value))
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| 53 |
+
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| 54 |
+
if len(audio_frames) < 7:
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| 55 |
+
return None
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| 56 |
+
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| 57 |
+
# Group into 7-token frames and extract codes for each layer
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| 58 |
+
num_complete_frames = len(audio_frames) // 7
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| 59 |
+
if num_complete_frames == 0:
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| 60 |
+
return None
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| 61 |
+
|
| 62 |
+
codes_0 = [] # layer 0
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| 63 |
+
codes_1 = [] # layer 1
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| 64 |
+
codes_2 = [] # layer 2
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| 65 |
+
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| 66 |
+
for i in range(num_complete_frames):
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| 67 |
+
base = i * 7
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| 68 |
+
# Frame format: [l0, l1_a, l2_a, l2_b, l1_b, l2_c, l2_d]
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| 69 |
+
codes_0.append(audio_frames[base][1]) # layer 0 value
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| 70 |
+
codes_1.append(audio_frames[base + 1][1]) # layer 1 first
|
| 71 |
+
codes_1.append(audio_frames[base + 4][1]) # layer 1 second
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| 72 |
+
codes_2.append(audio_frames[base + 2][1]) # layer 2 first
|
| 73 |
+
codes_2.append(audio_frames[base + 3][1]) # layer 2 second
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| 74 |
+
codes_2.append(audio_frames[base + 5][1]) # layer 2 third
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| 75 |
+
codes_2.append(audio_frames[base + 6][1]) # layer 2 fourth
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| 76 |
+
|
| 77 |
+
try:
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| 78 |
+
# Convert to tensors with correct shape
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| 79 |
+
with torch.no_grad():
|
| 80 |
+
codes = [
|
| 81 |
+
torch.tensor(codes_0, dtype=torch.int64).unsqueeze(0).to("cuda"),
|
| 82 |
+
torch.tensor(codes_1, dtype=torch.int64).unsqueeze(0).to("cuda"),
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| 83 |
+
torch.tensor(codes_2, dtype=torch.int64).unsqueeze(0).to("cuda"),
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| 84 |
+
]
|
| 85 |
+
audio = snac_model.decode(codes)
|
| 86 |
+
|
| 87 |
+
return audio.squeeze().cpu().numpy()
|
| 88 |
+
except Exception as e:
|
| 89 |
+
print(f" Decode error: {e}")
|
| 90 |
+
return None
|
| 91 |
+
|
| 92 |
+
|
| 93 |
+
async def run_tests():
|
| 94 |
+
"""Run continuous batching tests."""
|
| 95 |
+
print("=" * 60)
|
| 96 |
+
print("ORPHEUS TTS - CONTINUOUS BATCHING TEST v3")
|
| 97 |
+
print("=" * 60)
|
| 98 |
+
|
| 99 |
+
# Load tokenizer
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| 100 |
+
print("\n[1] Loading tokenizer...")
|
| 101 |
+
tokenizer = AutoTokenizer.from_pretrained("canopylabs/orpheus-3b-0.1-ft")
|
| 102 |
+
|
| 103 |
+
# Create AsyncLLMEngine
|
| 104 |
+
print("\n[2] Loading vLLM AsyncLLMEngine...")
|
| 105 |
+
start_load = time.time()
|
| 106 |
+
|
| 107 |
+
engine_args = AsyncEngineArgs(
|
| 108 |
+
model="canopylabs/orpheus-3b-0.1-ft",
|
| 109 |
+
dtype="bfloat16",
|
| 110 |
+
max_model_len=4096,
|
| 111 |
+
gpu_memory_utilization=0.9,
|
| 112 |
+
max_num_seqs=8, # Continuous batching
|
| 113 |
+
enable_chunked_prefill=True,
|
| 114 |
+
enable_prefix_caching=True,
|
| 115 |
+
enforce_eager=False,
|
| 116 |
+
)
|
| 117 |
+
|
| 118 |
+
engine = AsyncLLMEngine.from_engine_args(engine_args)
|
| 119 |
+
load_time = time.time() - start_load
|
| 120 |
+
print(f" vLLM loaded in {load_time:.2f}s")
|
| 121 |
+
|
| 122 |
+
# Load SNAC decoder
|
| 123 |
+
print("\n[3] Loading SNAC decoder...")
|
| 124 |
+
snac = SNAC.from_pretrained("hubertsiuzdak/snac_24khz").eval().to("cuda")
|
| 125 |
+
print(" SNAC loaded!")
|
| 126 |
+
|
| 127 |
+
# Sampling params
|
| 128 |
+
sampling_params = SamplingParams(
|
| 129 |
+
temperature=0.2,
|
| 130 |
+
top_p=0.9,
|
| 131 |
+
max_tokens=4096,
|
| 132 |
+
stop_token_ids=[STOP_TOKEN],
|
| 133 |
+
repetition_penalty=1.1,
|
| 134 |
+
)
|
| 135 |
+
|
| 136 |
+
def format_prompt(text: str, voice: str = "tara") -> str:
|
| 137 |
+
"""Format prompt with Orpheus special tokens."""
|
| 138 |
+
adapted_prompt = f"{voice}: {text}"
|
| 139 |
+
prompt_tokens = tokenizer(adapted_prompt, return_tensors="pt")
|
| 140 |
+
start_token = torch.tensor([[START_TOKEN]], dtype=torch.int64)
|
| 141 |
+
end_tokens = torch.tensor([END_TOKENS], dtype=torch.int64)
|
| 142 |
+
all_input_ids = torch.cat([start_token, prompt_tokens.input_ids, end_tokens], dim=1)
|
| 143 |
+
prompt_string = tokenizer.decode(all_input_ids[0])
|
| 144 |
+
return prompt_string
|
| 145 |
+
|
| 146 |
+
async def generate_speech(text: str, voice: str = "tara", request_id: str = None):
|
| 147 |
+
"""Generate speech for a single request."""
|
| 148 |
+
prompt_string = format_prompt(text, voice)
|
| 149 |
+
request_id = request_id or f"req_{time.time()}"
|
| 150 |
+
|
| 151 |
+
start = time.time()
|
| 152 |
+
token_ids = []
|
| 153 |
+
|
| 154 |
+
async for output in engine.generate(prompt_string, sampling_params, request_id):
|
| 155 |
+
token_ids = list(output.outputs[0].token_ids)
|
| 156 |
+
|
| 157 |
+
gen_time = time.time() - start
|
| 158 |
+
|
| 159 |
+
# Count audio tokens
|
| 160 |
+
audio_token_count = sum(1 for t in token_ids if isinstance(t, int) and t >= AUDIO_TOKEN_BASE)
|
| 161 |
+
|
| 162 |
+
# Decode to audio
|
| 163 |
+
audio = decode_tokens_to_audio(token_ids, snac)
|
| 164 |
+
|
| 165 |
+
if audio is None:
|
| 166 |
+
return {
|
| 167 |
+
'success': False,
|
| 168 |
+
'text': text[:30],
|
| 169 |
+
'gen_time': gen_time,
|
| 170 |
+
'tokens': len(token_ids),
|
| 171 |
+
'audio_tokens': audio_token_count,
|
| 172 |
+
}
|
| 173 |
+
|
| 174 |
+
audio_duration = len(audio) / 24000
|
| 175 |
+
rtf = gen_time / audio_duration if audio_duration > 0 else float('inf')
|
| 176 |
+
|
| 177 |
+
return {
|
| 178 |
+
'success': True,
|
| 179 |
+
'text': text[:30],
|
| 180 |
+
'gen_time': gen_time,
|
| 181 |
+
'tokens': len(token_ids),
|
| 182 |
+
'audio_tokens': audio_token_count,
|
| 183 |
+
'audio_duration': audio_duration,
|
| 184 |
+
'rtf': rtf,
|
| 185 |
+
'audio': audio,
|
| 186 |
+
}
|
| 187 |
+
|
| 188 |
+
# =========================================================================
|
| 189 |
+
# TEST 1: SEQUENTIAL (baseline)
|
| 190 |
+
# =========================================================================
|
| 191 |
+
print("\n" + "=" * 60)
|
| 192 |
+
print("[4] Test SEQUENTIAL (baseline)")
|
| 193 |
+
print("=" * 60)
|
| 194 |
+
|
| 195 |
+
test_texts = [
|
| 196 |
+
"Hello! This is the first test.",
|
| 197 |
+
"Second test to measure performance.",
|
| 198 |
+
"Third test for consistent results.",
|
| 199 |
+
]
|
| 200 |
+
|
| 201 |
+
sequential_results = []
|
| 202 |
+
total_seq_time = 0
|
| 203 |
+
|
| 204 |
+
for i, text in enumerate(test_texts, 1):
|
| 205 |
+
print(f"\n Test {i}: \"{text}\"")
|
| 206 |
+
result = await generate_speech(text, request_id=f"seq_{i}")
|
| 207 |
+
|
| 208 |
+
if result['success']:
|
| 209 |
+
print(f" -> Time: {result['gen_time']:.2f}s | Audio: {result['audio_duration']:.2f}s | RTF: {result['rtf']:.3f}")
|
| 210 |
+
sequential_results.append(result)
|
| 211 |
+
total_seq_time += result['gen_time']
|
| 212 |
+
else:
|
| 213 |
+
print(f" -> ERROR: {result['tokens']} tokens ({result['audio_tokens']} audio), no audio")
|
| 214 |
+
|
| 215 |
+
# =========================================================================
|
| 216 |
+
# TEST 2: PARALLEL (Continuous Batching)
|
| 217 |
+
# =========================================================================
|
| 218 |
+
print("\n" + "=" * 60)
|
| 219 |
+
print("[5] Test PARALLEL (Continuous Batching)")
|
| 220 |
+
print("=" * 60)
|
| 221 |
+
|
| 222 |
+
parallel_texts = [
|
| 223 |
+
"Hello, how are you today?",
|
| 224 |
+
"The weather is beautiful outside.",
|
| 225 |
+
"I love programming with Python.",
|
| 226 |
+
"Machine learning is fascinating.",
|
| 227 |
+
]
|
| 228 |
+
|
| 229 |
+
batch_results_summary = []
|
| 230 |
+
|
| 231 |
+
for num_concurrent in [2, 4]:
|
| 232 |
+
print(f"\n === {num_concurrent} CONCURRENT REQUESTS ===")
|
| 233 |
+
|
| 234 |
+
texts = parallel_texts[:num_concurrent]
|
| 235 |
+
|
| 236 |
+
start_batch = time.time()
|
| 237 |
+
tasks = [
|
| 238 |
+
generate_speech(text, request_id=f"par_{num_concurrent}_{i}")
|
| 239 |
+
for i, text in enumerate(texts)
|
| 240 |
+
]
|
| 241 |
+
results = await asyncio.gather(*tasks)
|
| 242 |
+
batch_time = time.time() - start_batch
|
| 243 |
+
|
| 244 |
+
# Calculate metrics
|
| 245 |
+
successful = [r for r in results if r['success']]
|
| 246 |
+
total_audio = sum(r['audio_duration'] for r in successful)
|
| 247 |
+
|
| 248 |
+
print(f"\n Results:")
|
| 249 |
+
for r in results:
|
| 250 |
+
if r['success']:
|
| 251 |
+
print(f" - \"{r['text']}...\" -> {r['gen_time']:.2f}s | {r['audio_duration']:.2f}s | RTF: {r['rtf']:.3f}")
|
| 252 |
+
else:
|
| 253 |
+
print(f" - \"{r['text']}...\" -> FAILED ({r['audio_tokens']} audio tokens)")
|
| 254 |
+
|
| 255 |
+
if total_audio > 0:
|
| 256 |
+
batch_rtf = batch_time / total_audio
|
| 257 |
+
throughput = len(successful) / batch_time
|
| 258 |
+
|
| 259 |
+
print(f"\n Aggregate Metrics:")
|
| 260 |
+
print(f" - Total wall-clock time: {batch_time:.2f}s")
|
| 261 |
+
print(f" - Success rate: {len(successful)}/{len(texts)}")
|
| 262 |
+
print(f" - Total audio generated: {total_audio:.2f}s")
|
| 263 |
+
print(f" - Batch RTF: {batch_rtf:.3f}")
|
| 264 |
+
print(f" - Throughput: {throughput:.2f} req/s")
|
| 265 |
+
print(f" - Effective speed: {1/batch_rtf:.1f}x real-time")
|
| 266 |
+
|
| 267 |
+
batch_results_summary.append({
|
| 268 |
+
'concurrent': num_concurrent,
|
| 269 |
+
'batch_time': batch_time,
|
| 270 |
+
'total_audio': total_audio,
|
| 271 |
+
'batch_rtf': batch_rtf,
|
| 272 |
+
'throughput': throughput,
|
| 273 |
+
})
|
| 274 |
+
|
| 275 |
+
# =========================================================================
|
| 276 |
+
# FINAL SUMMARY
|
| 277 |
+
# =========================================================================
|
| 278 |
+
print("\n" + "=" * 60)
|
| 279 |
+
print("[6] === FINAL SUMMARY ===")
|
| 280 |
+
print("=" * 60)
|
| 281 |
+
|
| 282 |
+
if sequential_results:
|
| 283 |
+
seq_rtfs = [r['rtf'] for r in sequential_results]
|
| 284 |
+
avg_seq_rtf = sum(seq_rtfs) / len(seq_rtfs)
|
| 285 |
+
total_seq_audio = sum(r['audio_duration'] for r in sequential_results)
|
| 286 |
+
|
| 287 |
+
print(f"\n SEQUENTIAL (baseline):")
|
| 288 |
+
print(f" - Average RTF: {avg_seq_rtf:.3f}")
|
| 289 |
+
print(f" - Speed: {1/avg_seq_rtf:.1f}x real-time")
|
| 290 |
+
|
| 291 |
+
if batch_results_summary:
|
| 292 |
+
print(f"\n CONTINUOUS BATCHING:")
|
| 293 |
+
for bs in batch_results_summary:
|
| 294 |
+
speedup = avg_seq_rtf / bs['batch_rtf'] if bs['batch_rtf'] > 0 else 0
|
| 295 |
+
print(f" - {bs['concurrent']} concurrent: RTF={bs['batch_rtf']:.3f}, {bs['throughput']:.2f} req/s, {speedup:.1f}x speedup")
|
| 296 |
+
|
| 297 |
+
# Capacity estimate
|
| 298 |
+
users_seq = 1 / avg_seq_rtf if avg_seq_rtf > 0 else 0
|
| 299 |
+
print(f"\n RTX 4090 CAPACITY ESTIMATE:")
|
| 300 |
+
print(f" - Sequential: ~{users_seq:.0f} real-time users")
|
| 301 |
+
if batch_results_summary:
|
| 302 |
+
best_batch = min(batch_results_summary, key=lambda x: x['batch_rtf'])
|
| 303 |
+
users_batch = best_batch['concurrent'] / best_batch['batch_rtf'] if best_batch['batch_rtf'] > 0 else 0
|
| 304 |
+
print(f" - With batching: ~{users_batch:.0f} real-time users")
|
| 305 |
+
|
| 306 |
+
print("=" * 60)
|
| 307 |
+
|
| 308 |
+
# Save audio
|
| 309 |
+
if sequential_results and sequential_results[-1]['success']:
|
| 310 |
+
audio = sequential_results[-1]['audio']
|
| 311 |
+
output_path = "/root/test_batching_v3_output.wav"
|
| 312 |
+
audio_int16 = (audio * 32767).astype(np.int16)
|
| 313 |
+
with wave.open(output_path, "wb") as wf:
|
| 314 |
+
wf.setnchannels(1)
|
| 315 |
+
wf.setsampwidth(2)
|
| 316 |
+
wf.setframerate(24000)
|
| 317 |
+
wf.writeframes(audio_int16.tobytes())
|
| 318 |
+
print(f"\n Audio saved to: {output_path}")
|
| 319 |
+
|
| 320 |
+
|
| 321 |
+
if __name__ == "__main__":
|
| 322 |
+
asyncio.run(run_tests())
|
scripts/orpheus_stress_test.py
ADDED
|
@@ -0,0 +1,195 @@
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| 1 |
+
"""
|
| 2 |
+
Teste de Stress - Validar capacidade real de usuarios simultaneos
|
| 3 |
+
"""
|
| 4 |
+
import os
|
| 5 |
+
os.environ["VLLM_ATTENTION_BACKEND"] = "FLASH_ATTN"
|
| 6 |
+
|
| 7 |
+
import torch
|
| 8 |
+
import time
|
| 9 |
+
import asyncio
|
| 10 |
+
import numpy as np
|
| 11 |
+
from transformers import AutoTokenizer
|
| 12 |
+
from vllm import AsyncLLMEngine, AsyncEngineArgs, SamplingParams
|
| 13 |
+
from snac import SNAC
|
| 14 |
+
|
| 15 |
+
START_TOKEN = 128259
|
| 16 |
+
END_TOKENS = [128009, 128260, 128261, 128257]
|
| 17 |
+
STOP_TOKEN = 128258
|
| 18 |
+
AUDIO_TOKEN_BASE = 128266
|
| 19 |
+
|
| 20 |
+
def decode_tokens_to_audio(token_ids, snac_model):
|
| 21 |
+
audio_frames = []
|
| 22 |
+
for t in token_ids:
|
| 23 |
+
if isinstance(t, str):
|
| 24 |
+
continue
|
| 25 |
+
if t >= AUDIO_TOKEN_BASE:
|
| 26 |
+
offset = t - AUDIO_TOKEN_BASE
|
| 27 |
+
layer = offset // 4096
|
| 28 |
+
value = offset % 4096
|
| 29 |
+
audio_frames.append((layer, value))
|
| 30 |
+
|
| 31 |
+
if len(audio_frames) < 7:
|
| 32 |
+
return None
|
| 33 |
+
|
| 34 |
+
num_complete_frames = len(audio_frames) // 7
|
| 35 |
+
if num_complete_frames == 0:
|
| 36 |
+
return None
|
| 37 |
+
|
| 38 |
+
codes_0, codes_1, codes_2 = [], [], []
|
| 39 |
+
for i in range(num_complete_frames):
|
| 40 |
+
base = i * 7
|
| 41 |
+
codes_0.append(audio_frames[base][1])
|
| 42 |
+
codes_1.append(audio_frames[base + 1][1])
|
| 43 |
+
codes_1.append(audio_frames[base + 4][1])
|
| 44 |
+
codes_2.append(audio_frames[base + 2][1])
|
| 45 |
+
codes_2.append(audio_frames[base + 3][1])
|
| 46 |
+
codes_2.append(audio_frames[base + 5][1])
|
| 47 |
+
codes_2.append(audio_frames[base + 6][1])
|
| 48 |
+
|
| 49 |
+
try:
|
| 50 |
+
with torch.no_grad():
|
| 51 |
+
codes = [
|
| 52 |
+
torch.tensor(codes_0, dtype=torch.int64).unsqueeze(0).to("cuda"),
|
| 53 |
+
torch.tensor(codes_1, dtype=torch.int64).unsqueeze(0).to("cuda"),
|
| 54 |
+
torch.tensor(codes_2, dtype=torch.int64).unsqueeze(0).to("cuda"),
|
| 55 |
+
]
|
| 56 |
+
audio = snac_model.decode(codes)
|
| 57 |
+
return audio.squeeze().cpu().numpy()
|
| 58 |
+
except:
|
| 59 |
+
return None
|
| 60 |
+
|
| 61 |
+
async def main():
|
| 62 |
+
print("=" * 70)
|
| 63 |
+
print("TESTE DE STRESS - VALIDAR CAPACIDADE REAL DE USUARIOS")
|
| 64 |
+
print("=" * 70)
|
| 65 |
+
|
| 66 |
+
print("\n[1] Carregando modelo...")
|
| 67 |
+
tokenizer = AutoTokenizer.from_pretrained("canopylabs/orpheus-3b-0.1-ft")
|
| 68 |
+
|
| 69 |
+
engine_args = AsyncEngineArgs(
|
| 70 |
+
model="canopylabs/orpheus-3b-0.1-ft",
|
| 71 |
+
dtype="bfloat16",
|
| 72 |
+
max_model_len=4096,
|
| 73 |
+
gpu_memory_utilization=0.9,
|
| 74 |
+
max_num_seqs=16,
|
| 75 |
+
enable_chunked_prefill=True,
|
| 76 |
+
enable_prefix_caching=True,
|
| 77 |
+
enforce_eager=False,
|
| 78 |
+
)
|
| 79 |
+
engine = AsyncLLMEngine.from_engine_args(engine_args)
|
| 80 |
+
snac = SNAC.from_pretrained("hubertsiuzdak/snac_24khz").eval().to("cuda")
|
| 81 |
+
|
| 82 |
+
sampling_params = SamplingParams(
|
| 83 |
+
temperature=0.2,
|
| 84 |
+
top_p=0.9,
|
| 85 |
+
max_tokens=4096,
|
| 86 |
+
stop_token_ids=[STOP_TOKEN],
|
| 87 |
+
repetition_penalty=1.1,
|
| 88 |
+
)
|
| 89 |
+
|
| 90 |
+
def format_prompt(text, voice="tara"):
|
| 91 |
+
adapted_prompt = f"{voice}: {text}"
|
| 92 |
+
prompt_tokens = tokenizer(adapted_prompt, return_tensors="pt")
|
| 93 |
+
start_token = torch.tensor([[START_TOKEN]], dtype=torch.int64)
|
| 94 |
+
end_tokens = torch.tensor([END_TOKENS], dtype=torch.int64)
|
| 95 |
+
all_input_ids = torch.cat([start_token, prompt_tokens.input_ids, end_tokens], dim=1)
|
| 96 |
+
return tokenizer.decode(all_input_ids[0])
|
| 97 |
+
|
| 98 |
+
async def generate_speech(text, request_id):
|
| 99 |
+
prompt_string = format_prompt(text)
|
| 100 |
+
start = time.time()
|
| 101 |
+
token_ids = []
|
| 102 |
+
async for output in engine.generate(prompt_string, sampling_params, request_id):
|
| 103 |
+
token_ids = list(output.outputs[0].token_ids)
|
| 104 |
+
gen_time = time.time() - start
|
| 105 |
+
|
| 106 |
+
audio = decode_tokens_to_audio(token_ids, snac)
|
| 107 |
+
if audio is None:
|
| 108 |
+
return {'success': False, 'gen_time': gen_time}
|
| 109 |
+
|
| 110 |
+
audio_duration = len(audio) / 24000
|
| 111 |
+
return {
|
| 112 |
+
'success': True,
|
| 113 |
+
'gen_time': gen_time,
|
| 114 |
+
'audio_duration': audio_duration,
|
| 115 |
+
'rtf': gen_time / audio_duration if audio_duration > 0 else float('inf')
|
| 116 |
+
}
|
| 117 |
+
|
| 118 |
+
test_texts = [
|
| 119 |
+
"Hello, how are you doing today?",
|
| 120 |
+
"The weather is beautiful outside.",
|
| 121 |
+
"I love programming with Python.",
|
| 122 |
+
"Machine learning is fascinating.",
|
| 123 |
+
"Can you help me with this task?",
|
| 124 |
+
"Let me explain how this works.",
|
| 125 |
+
"This is a test of the system.",
|
| 126 |
+
"Technology is amazing these days.",
|
| 127 |
+
"Have a wonderful day ahead.",
|
| 128 |
+
"Thank you for your patience.",
|
| 129 |
+
"Let's work together on this.",
|
| 130 |
+
"The future looks very bright.",
|
| 131 |
+
"I appreciate your help today.",
|
| 132 |
+
"This demonstration is working.",
|
| 133 |
+
"Audio generation is fast now.",
|
| 134 |
+
"Real-time speech synthesis.",
|
| 135 |
+
]
|
| 136 |
+
|
| 137 |
+
print("\n[2] Iniciando testes de stress...")
|
| 138 |
+
print("=" * 70)
|
| 139 |
+
|
| 140 |
+
results_summary = []
|
| 141 |
+
|
| 142 |
+
for num_users in [4, 8, 12, 16]:
|
| 143 |
+
print(f"\n>>> TESTANDO {num_users} USUARIOS SIMULTANEOS <<<")
|
| 144 |
+
print("-" * 50)
|
| 145 |
+
|
| 146 |
+
texts = test_texts[:num_users]
|
| 147 |
+
|
| 148 |
+
start_batch = time.time()
|
| 149 |
+
tasks = [generate_speech(text, f"user_{i}_{num_users}") for i, text in enumerate(texts)]
|
| 150 |
+
results = await asyncio.gather(*tasks)
|
| 151 |
+
batch_time = time.time() - start_batch
|
| 152 |
+
|
| 153 |
+
successful = [r for r in results if r['success']]
|
| 154 |
+
total_audio = sum(r['audio_duration'] for r in successful)
|
| 155 |
+
|
| 156 |
+
if len(successful) > 0:
|
| 157 |
+
batch_rtf = batch_time / total_audio
|
| 158 |
+
throughput = len(successful) / batch_time
|
| 159 |
+
realtime_users = total_audio / batch_time
|
| 160 |
+
|
| 161 |
+
print(f" Sucesso: {len(successful)}/{num_users}")
|
| 162 |
+
print(f" Tempo total (wall-clock): {batch_time:.2f}s")
|
| 163 |
+
print(f" Audio total gerado: {total_audio:.2f}s")
|
| 164 |
+
print(f" Batch RTF: {batch_rtf:.3f}")
|
| 165 |
+
print(f" Throughput: {throughput:.2f} req/s")
|
| 166 |
+
print(f" USUARIOS REAL-TIME: {realtime_users:.1f}")
|
| 167 |
+
|
| 168 |
+
results_summary.append({
|
| 169 |
+
'users': num_users,
|
| 170 |
+
'success': len(successful),
|
| 171 |
+
'batch_time': batch_time,
|
| 172 |
+
'total_audio': total_audio,
|
| 173 |
+
'batch_rtf': batch_rtf,
|
| 174 |
+
'realtime_users': realtime_users
|
| 175 |
+
})
|
| 176 |
+
else:
|
| 177 |
+
print(f" ERRO: Nenhum audio gerado!")
|
| 178 |
+
|
| 179 |
+
print("\n" + "=" * 70)
|
| 180 |
+
print("RESUMO FINAL - CAPACIDADE RTX 4090")
|
| 181 |
+
print("=" * 70)
|
| 182 |
+
|
| 183 |
+
print("\n| Requests | Sucesso | Wall-Time | Audio Total | RTF | Real-time Users |")
|
| 184 |
+
print("|----------|---------|-----------|-------------|-------|-----------------|")
|
| 185 |
+
for r in results_summary:
|
| 186 |
+
print(f"| {r['users']:8} | {r['success']:7} | {r['batch_time']:9.2f}s | {r['total_audio']:11.2f}s | {r['batch_rtf']:.3f} | {r['realtime_users']:15.1f} |")
|
| 187 |
+
|
| 188 |
+
if results_summary:
|
| 189 |
+
best = max(results_summary, key=lambda x: x['realtime_users'])
|
| 190 |
+
print(f"\nMELHOR RESULTADO: {best['realtime_users']:.1f} usuarios real-time com {best['users']} requests")
|
| 191 |
+
|
| 192 |
+
print("=" * 70)
|
| 193 |
+
|
| 194 |
+
if __name__ == "__main__":
|
| 195 |
+
asyncio.run(main())
|