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import io
import os
import queue
import re
import time
import traceback
import wave
from argparse import ArgumentParser
from http import HTTPStatus
from pathlib import Path
from typing import Annotated, Any
import librosa
import numpy as np
import ormsgpack
import pyrootutils
import soundfile as sf
import torch
import torchaudio
from baize.datastructures import ContentType
from kui.asgi import (
Body,
FactoryClass,
HTTPException,
HttpRequest,
HttpView,
JSONResponse,
Kui,
OpenAPI,
StreamResponse,
request,
)
from kui.asgi.routing import MultimethodRoutes
from loguru import logger
from transformers import AutoTokenizer
pyrootutils.setup_root(__file__, indicator=".project-root", pythonpath=True)
import struct
from threading import Lock
import httpx
from cachetools import LRUCache, cached
from funasr import AutoModel
from silero_vad import get_speech_timestamps, load_silero_vad
from fish_speech.conversation import IM_END_TOKEN, SEMANTIC_TOKEN
from fish_speech.models.text2semantic.llama import BaseModelArgs
# from fish_speech.models.vqgan.lit_module import VQGAN
from fish_speech.models.vqgan.modules.firefly import FireflyArchitecture
from fish_speech.text.chn_text_norm.text import Text as ChnNormedText
from fish_speech.utils import autocast_exclude_mps, set_seed
from tools.file import AUDIO_EXTENSIONS, audio_to_bytes, list_files, read_ref_text
from tools.llama.generate import (
GenerateRequest,
GenerateResponse,
WrappedGenerateResponse,
launch_thread_safe_queue,
launch_thread_safe_queue_agent,
)
from tools.schema import (
GLOBAL_NUM_SAMPLES,
ASRPackRequest,
ServeASRRequest,
ServeASRResponse,
ServeASRSegment,
ServeAudioPart,
ServeForwardMessage,
ServeMessage,
ServeRequest,
ServeResponse,
ServeStreamDelta,
ServeStreamResponse,
ServeTextPart,
ServeTimedASRResponse,
ServeTTSRequest,
ServeVQGANDecodeRequest,
ServeVQGANDecodeResponse,
ServeVQGANEncodeRequest,
ServeVQGANEncodeResponse,
ServeVQPart,
)
from tools.vqgan.inference import load_model as load_decoder_model
global_lock = Lock()
# Whether to disable keepalive (which is helpful if the server is in the same cluster)
DISABLE_KEEPALIVE = os.getenv("DISABLE_KEEPALIVE", "false").lower() == "true"
async_client = httpx.AsyncClient(
timeout=120, limits=httpx.Limits(keepalive_expiry=0 if DISABLE_KEEPALIVE else None)
)
backends = torchaudio.list_audio_backends()
if "ffmpeg" in backends:
backend = "ffmpeg"
else:
backend = "soundfile"
def wav_chunk_header(sample_rate=44100, bit_depth=16, channels=1):
buffer = io.BytesIO()
with wave.open(buffer, "wb") as wav_file:
wav_file.setnchannels(channels)
wav_file.setsampwidth(bit_depth // 8)
wav_file.setframerate(sample_rate)
wav_header_bytes = buffer.getvalue()
buffer.close()
return wav_header_bytes
# Define utils for web server
async def http_execption_handler(exc: HTTPException):
return JSONResponse(
dict(
statusCode=exc.status_code,
message=exc.content,
error=HTTPStatus(exc.status_code).phrase,
),
exc.status_code,
exc.headers,
)
async def other_exception_handler(exc: "Exception"):
traceback.print_exc()
status = HTTPStatus.INTERNAL_SERVER_ERROR
return JSONResponse(
dict(statusCode=status, message=str(exc), error=status.phrase),
status,
)
def load_audio(reference_audio, sr):
if len(reference_audio) > 255 or not Path(reference_audio).exists():
audio_data = reference_audio
reference_audio = io.BytesIO(audio_data)
waveform, original_sr = torchaudio.load(reference_audio, backend=backend)
if waveform.shape[0] > 1:
waveform = torch.mean(waveform, dim=0, keepdim=True)
if original_sr != sr:
resampler = torchaudio.transforms.Resample(orig_freq=original_sr, new_freq=sr)
waveform = resampler(waveform)
audio = waveform.squeeze().numpy()
return audio
def encode_reference(*, decoder_model, reference_audio, enable_reference_audio):
if enable_reference_audio and reference_audio is not None:
# Load audios, and prepare basic info here
reference_audio_content = load_audio(
reference_audio, decoder_model.spec_transform.sample_rate
)
audios = torch.from_numpy(reference_audio_content).to(decoder_model.device)[
None, None, :
]
audio_lengths = torch.tensor(
[audios.shape[2]], device=decoder_model.device, dtype=torch.long
)
logger.info(
f"Loaded audio with {audios.shape[2] / decoder_model.spec_transform.sample_rate:.2f} seconds"
)
# VQ Encoder
if isinstance(decoder_model, FireflyArchitecture):
prompt_tokens = decoder_model.encode(audios, audio_lengths)[0][0]
logger.info(f"Encoded prompt: {prompt_tokens.shape}")
else:
prompt_tokens = None
logger.info("No reference audio provided")
return prompt_tokens
def decode_vq_tokens(
*,
decoder_model,
codes,
):
feature_lengths = torch.tensor([codes.shape[1]], device=decoder_model.device)
logger.info(f"VQ features: {codes.shape}")
if isinstance(decoder_model, FireflyArchitecture):
# VQGAN Inference
return decoder_model.decode(
indices=codes[None],
feature_lengths=feature_lengths,
)[0].squeeze()
raise ValueError(f"Unknown model type: {type(decoder_model)}")
routes = MultimethodRoutes(base_class=HttpView)
def get_content_type(audio_format):
if audio_format == "wav":
return "audio/wav"
elif audio_format == "flac":
return "audio/flac"
elif audio_format == "mp3":
return "audio/mpeg"
else:
return "application/octet-stream"
@torch.no_grad()
@torch.autocast(device_type="cuda", dtype=torch.half)
def batch_encode(model, audios: list[bytes | torch.Tensor]):
audios = [
(
torch.from_numpy(
librosa.load(io.BytesIO(audio), sr=model.spec_transform.sample_rate)[0]
)[None]
if isinstance(audio, bytes)
else audio
)
for audio in audios
]
# if any(audio.shape[-1] > model.spec_transform.sample_rate * 120 for audio in audios):
# raise ValueError("Single audio length is too long (>120s)")
max_length = max(audio.shape[-1] for audio in audios)
print(f"Encode max length: {max_length / model.spec_transform.sample_rate:.2f}s")
lengths = torch.tensor([audio.shape[-1] for audio in audios], device=model.device)
max_length = lengths.max().item()
padded = torch.stack(
[
torch.nn.functional.pad(audio, (0, max_length - audio.shape[-1]))
for audio in audios
]
).to(model.device)
features, feature_lengths = model.encode(padded, audio_lengths=lengths)
features, feature_lengths = features.cpu(), feature_lengths.cpu()
return [feature[..., :length] for feature, length in zip(features, feature_lengths)]
@cached(
cache=LRUCache(maxsize=10000),
key=lambda model, audios: (model.device, tuple(audios)),
)
def cached_vqgan_batch_encode(model, audios: list[bytes]):
return batch_encode(model, audios)
@routes.http.post("/v1/vqgan/encode")
def api_vqgan_encode(payload: Annotated[ServeVQGANEncodeRequest, Body(exclusive=True)]):
start_time = time.time()
tokens = cached_vqgan_batch_encode(decoder_model, payload.audios)
logger.info(f"[EXEC] VQGAN encode time: {(time.time() - start_time) * 1000:.2f}ms")
return ormsgpack.packb(
ServeVQGANEncodeResponse(tokens=[i.tolist() for i in tokens]),
option=ormsgpack.OPT_SERIALIZE_PYDANTIC,
)
@torch.no_grad()
@torch.autocast(device_type="cuda", dtype=torch.half)
def vqgan_decode(model, features):
lengths = torch.tensor(
[feature.shape[-1] for feature in features], device=model.device
)
max_length = lengths.max().item()
padded = torch.stack(
[
torch.nn.functional.pad(feature, (0, max_length - feature.shape[-1]))
for feature in features
]
).to(model.device)
# If bs too large, we do micro batch decode
audios, audio_lengths = [], []
for i in range(0, padded.shape[0], 8):
audio, audio_length = model.decode(
padded[i : i + 8], feature_lengths=lengths[i : i + 8]
)
audios.append(audio)
audio_lengths.append(audio_length)
audios = torch.cat(audios, dim=0)
audio_lengths = torch.cat(audio_lengths, dim=0)
audios, audio_lengths = audios.cpu(), audio_lengths.cpu()
return [audio[..., :length].numpy() for audio, length in zip(audios, audio_lengths)]
@routes.http.post("/v1/vqgan/decode")
def api_vqgan_decode(payload: Annotated[ServeVQGANDecodeRequest, Body(exclusive=True)]):
tokens = [torch.tensor(token, dtype=torch.int) for token in payload.tokens]
start_time = time.time()
audios = vqgan_decode(decoder_model, tokens)
logger.info(f"[EXEC] VQGAN decode time: {(time.time() - start_time) * 1000:.2f}ms")
audios = [audio.astype(np.float16).tobytes() for audio in audios]
return ormsgpack.packb(
ServeVQGANDecodeResponse(audios=audios), option=ormsgpack.OPT_SERIALIZE_PYDANTIC
)
@torch.no_grad()
def batch_asr(model, audios, sr, language="auto"):
resampled_audios = []
for audio in audios:
audio = torchaudio.functional.resample(audio, sr, 16000)
assert audio.ndim == 1
resampled_audios.append(audio)
with global_lock:
res = model.generate(
input=resampled_audios,
batch_size=len(resampled_audios),
language=language,
use_itn=True,
)
results = []
for r, audio in zip(res, audios):
text = r["text"]
text = re.sub(r"<\|.*?\|>", "", text)
duration = len(audio) / sr * 1000
huge_gap = False
if "timestamp" in r and len(r["timestamp"]) > 2:
for timestamp_a, timestamp_b in zip(
r["timestamp"][:-1], r["timestamp"][1:]
):
# If there is a gap of more than 5 seconds, we consider it as a huge gap
if timestamp_b[0] - timestamp_a[1] > 5000:
huge_gap = True
break
# Doesn't make sense to have a huge gap at the end
if duration - r["timestamp"][-1][1] > 3000:
huge_gap = True
results.append(
{
"text": text,
"duration": duration,
"huge_gap": huge_gap,
}
)
return results
@routes.http.post("/v1/asr")
def api_invoke_asr(payload: Annotated[ServeASRRequest, Body(exclusive=True)]):
start_time = time.time()
audios = [np.frombuffer(audio, dtype=np.float16) for audio in payload.audios]
audios = [torch.from_numpy(audio).float() for audio in audios]
if any(audios.shape[-1] >= 30 * payload.sample_rate for audios in audios):
raise HTTPException(status_code=400, detail="Audio length is too long")
transcriptions = batch_asr(
asr_model, audios=audios, sr=payload.sample_rate, language=payload.language
)
logger.info(f"[EXEC] ASR time: {(time.time() - start_time) * 1000:.2f}ms")
return ormsgpack.packb(
ServeASRResponse(transcriptions=transcriptions),
option=ormsgpack.OPT_SERIALIZE_PYDANTIC,
)
from fish_speech.conversation import Conversation, Message
def execute_request(
input_queue: queue.Queue,
tokenizer: AutoTokenizer,
config: BaseModelArgs,
request: ServeRequest,
device: str = "cuda:0",
):
semantic_id, im_end_id = tokenizer.convert_tokens_to_ids(
[SEMANTIC_TOKEN, IM_END_TOKEN]
)
messages = []
for message in request.messages:
messages.append(message.to_conversation_message())
assert len(messages) >= 1, "At least one message is required"
# assert messages[-1].role == "user", "The last message must be from the user"
if messages[-1].role == "user":
messages.append(Message(role="assistant", parts=[], add_im_end=False))
else:
assert (
messages[-1].role == "assistant"
), "The last message must be from the assistant"
messages[-1].add_im_end = False
conv = Conversation(messages=messages)
prompt = conv.encode_for_inference(
tokenizer=tokenizer, num_codebooks=config.num_codebooks
).to(device)
if request.streaming:
for i in range(request.num_samples):
yield ServeStreamResponse(
sample_id=i,
delta=ServeStreamDelta(
role="assistant",
),
)
req = {
"prompt": prompt,
"max_new_tokens": request.max_new_tokens,
"im_end_id": im_end_id,
"semantic_id": semantic_id,
"temperature": request.temperature,
"top_p": request.top_p,
"repetition_penalty": request.repetition_penalty,
"num_samples": request.num_samples,
"early_stop_threshold": request.early_stop_threshold,
}
start = time.time()
response_queue = queue.Queue()
input_queue.put(GenerateRequest(req, response_queue))
# Decoding
decode_buffer = [[] for _ in range(request.num_samples)]
parts = [[] for _ in range(request.num_samples)]
def send_reset_buffer(sample_id):
nonlocal decode_buffer
if len(decode_buffer[sample_id]) == 0:
return
decoded = tokenizer.decode(decode_buffer[sample_id])
part = ServeTextPart(text=decoded)
if request.streaming:
yield ServeStreamResponse(delta=ServeStreamDelta(part=part))
else:
parts[sample_id].append(part)
decode_buffer[sample_id] = []
# Decode process
finished = [False for _ in range(request.num_samples)]
stats = {}
idx = 0
while True:
response = response_queue.get()
if response in ["stop", "error"]:
break
for sample_id, tokens in enumerate(response):
if finished[sample_id]:
continue
if tokens[0] == im_end_id:
finished[sample_id] = True
if request.streaming:
yield from send_reset_buffer(sample_id)
yield ServeStreamResponse(
sample_id=sample_id,
finish_reason="stop",
stats=stats,
)
continue
if tokens[0] == semantic_id and request.streaming:
yield from send_reset_buffer(sample_id)
# Streaming vq
_tokens = tokens[1:].clone() - 1
if config.share_codebook_embeddings is False:
for i in range(len(_tokens)):
_tokens[i] -= config.codebook_size * i
yield ServeStreamResponse(
sample_id=sample_id,
delta=ServeStreamDelta(part=ServeVQPart(codes=_tokens.tolist())),
)
continue
# Not streaming vq
if tokens[0] == semantic_id:
yield from send_reset_buffer(sample_id)
# None streaming vq
if len(parts[sample_id]) == 0 or not isinstance(
parts[sample_id][-1], ServeVQPart
):
_tokens = tokens[1:].clone() - 1
if config.share_codebook_embeddings is False:
for i in range(len(_tokens)):
_tokens[i] -= config.codebook_size * i
parts[sample_id].append(ServeVQPart(codes=_tokens.tolist()))
else:
for codebook_id, value in enumerate(tokens[1:, :]):
val = value.item() - 1
if config.share_codebook_embeddings is False:
val -= config.codebook_size * codebook_id
parts[sample_id][-1].codes[codebook_id].append(val)
continue
if tokens[0] != semantic_id:
# Stream text decode is not supported now
decode_buffer[sample_id].append(tokens[0, 0])
if idx == 0:
stats["time_to_first_token"] = (time.time() - start) * 1000
idx += 1
for sample_id in range(request.num_samples):
yield from send_reset_buffer(sample_id)
stats["total_time"] = (time.time() - start) * 1000
stats["total_tokens"] = idx
if request.streaming:
for sample_id in range(request.num_samples):
if finished[sample_id]:
continue
yield ServeStreamResponse(
finish_reason=response, stats=stats, sample_id=sample_id
)
return
yield ServeResponse(
messages=[
ServeMessage(role="assistant", parts=parts[i])
for i in range(request.num_samples)
],
finish_reason=response,
stats=stats,
)
@routes.http.post("/v1/chat")
def api_invoke_chat(
req: Annotated[ServeRequest, Body(exclusive=True)],
):
"""
Invoke model and generate audio
"""
# This makes torch compile happy
assert (
req.num_samples == GLOBAL_NUM_SAMPLES
), f"num_samples must be {GLOBAL_NUM_SAMPLES}"
content_type = request.headers.get("Content-Type", "application/json")
json_mode = "application/json" in content_type
async def wrapped_generator():
generator = execute_request(llama_queue, tokenizer, config, req, args.device)
for i in generator:
if json_mode:
body = i.model_dump_json().encode("utf-8")
yield b"data: " + body + b"\n\n"
else:
body = ormsgpack.packb(i, option=ormsgpack.OPT_SERIALIZE_PYDANTIC)
yield struct.pack("I", len(body)) + body
# Naive mode
if req.streaming is False:
result = next(execute_request(llama_queue, tokenizer, config, req, args.device))
if json_mode:
return JSONResponse(result.model_dump())
else:
return ormsgpack.packb(result, option=ormsgpack.OPT_SERIALIZE_PYDANTIC)
return StreamResponse(
iterable=wrapped_generator(), content_type="text/event-stream"
)
@torch.inference_mode()
def inference(req: ServeTTSRequest):
global prompt_tokens, prompt_texts
idstr: str | None = req.reference_id
if idstr is not None:
ref_folder = Path("references") / idstr
ref_folder.mkdir(parents=True, exist_ok=True)
ref_audios = list_files(
ref_folder, AUDIO_EXTENSIONS, recursive=True, sort=False
)
if req.use_memory_cache == "never" or (
req.use_memory_cache == "on-demand" and len(prompt_tokens) == 0
):
prompt_tokens = [
encode_reference(
decoder_model=decoder_model,
reference_audio=audio_to_bytes(str(ref_audio)),
enable_reference_audio=True,
)
for ref_audio in ref_audios
]
prompt_texts = [
read_ref_text(str(ref_audio.with_suffix(".lab")))
for ref_audio in ref_audios
]
else:
logger.info("Use same references")
else:
# Parse reference audio aka prompt
refs = req.references
if req.use_memory_cache == "never" or (
req.use_memory_cache == "on-demand" and len(prompt_tokens) == 0
):
prompt_tokens = [
encode_reference(
decoder_model=decoder_model,
reference_audio=ref.audio,
enable_reference_audio=True,
)
for ref in refs
]
prompt_texts = [ref.text for ref in refs]
else:
logger.info("Use same references")
if req.seed is not None:
set_seed(req.seed)
logger.warning(f"set seed: {req.seed}")
# LLAMA Inference
request = dict(
device=decoder_model.device,
max_new_tokens=req.max_new_tokens,
text=(
req.text
if not req.normalize
else ChnNormedText(raw_text=req.text).normalize()
),
top_p=req.top_p,
repetition_penalty=req.repetition_penalty,
temperature=req.temperature,
compile=args.compile,
iterative_prompt=req.chunk_length > 0,
chunk_length=req.chunk_length,
max_length=4096,
prompt_tokens=prompt_tokens,
prompt_text=prompt_texts,
)
response_queue = queue.Queue()
llama_queue.put(
GenerateRequest(
request=request,
response_queue=response_queue,
)
)
if req.streaming:
yield wav_chunk_header()
segments = []
while True:
result: WrappedGenerateResponse = response_queue.get()
if result.status == "error":
raise result.response
break
result: GenerateResponse = result.response
if result.action == "next":
break
with autocast_exclude_mps(
device_type=decoder_model.device.type, dtype=args.precision
):
fake_audios = decode_vq_tokens(
decoder_model=decoder_model,
codes=result.codes,
)
fake_audios = fake_audios.float().cpu().numpy()
if req.streaming:
yield (fake_audios * 32768).astype(np.int16).tobytes()
else:
segments.append(fake_audios)
if req.streaming:
return
if len(segments) == 0:
raise HTTPException(
HTTPStatus.INTERNAL_SERVER_ERROR,
content="No audio generated, please check the input text.",
)
fake_audios = np.concatenate(segments, axis=0)
yield fake_audios
async def inference_async(req: ServeTTSRequest):
for chunk in inference(req):
yield chunk
async def buffer_to_async_generator(buffer):
yield buffer
@routes.http.post("/v1/tts")
async def api_invoke_model(
req: Annotated[ServeTTSRequest, Body(exclusive=True)],
):
"""
Invoke model and generate audio
"""
if args.max_text_length > 0 and len(req.text) > args.max_text_length:
raise HTTPException(
HTTPStatus.BAD_REQUEST,
content=f"Text is too long, max length is {args.max_text_length}",
)
if req.streaming and req.format != "wav":
raise HTTPException(
HTTPStatus.BAD_REQUEST,
content="Streaming only supports WAV format",
)
if req.streaming:
return StreamResponse(
iterable=inference_async(req),
headers={
"Content-Disposition": f"attachment; filename=audio.{req.format}",
},
content_type=get_content_type(req.format),
)
else:
fake_audios = next(inference(req))
buffer = io.BytesIO()
sf.write(
buffer,
fake_audios,
decoder_model.spec_transform.sample_rate,
format=req.format,
)
return StreamResponse(
iterable=buffer_to_async_generator(buffer.getvalue()),
headers={
"Content-Disposition": f"attachment; filename=audio.{req.format}",
},
content_type=get_content_type(req.format),
)
@routes.http.post("/v1/health")
async def api_health():
"""
Health check
"""
return JSONResponse({"status": "ok"})
def parse_args():
parser = ArgumentParser()
parser.add_argument("--mode", type=str, choices=["agent", "tts"], default="agent")
parser.add_argument("--load-asr-model", action="store_true")
parser.add_argument(
"--llama-checkpoint-path",
type=str,
default="checkpoints/fish-agent-v0.1-3b",
)
parser.add_argument(
"--decoder-checkpoint-path",
type=str,
default="checkpoints/fish-speech-1.4/firefly-gan-vq-fsq-8x1024-21hz-generator.pth",
)
parser.add_argument("--decoder-config-name", type=str, default="firefly_gan_vq")
parser.add_argument("--device", type=str, default="cuda")
parser.add_argument("--half", action="store_true")
parser.add_argument("--compile", action="store_true",default=False)
parser.add_argument("--max-text-length", type=int, default=0)
parser.add_argument("--listen", type=str, default="127.0.0.1:8080")
parser.add_argument("--workers", type=int, default=1)
return parser.parse_args()
# Define Kui app
openapi = OpenAPI(
{
"title": "Fish Speech API",
"version": "1.4.2",
},
).routes
class MsgPackRequest(HttpRequest):
async def data(
self,
) -> Annotated[
Any, ContentType("application/msgpack"), ContentType("application/json")
]:
if self.content_type == "application/msgpack":
return ormsgpack.unpackb(await self.body)
elif self.content_type == "application/json":
return await self.json
raise HTTPException(
HTTPStatus.UNSUPPORTED_MEDIA_TYPE,
headers={"Accept": "application/msgpack, application/json"},
)
app = Kui(
routes=routes + openapi[1:], # Remove the default route
exception_handlers={
HTTPException: http_execption_handler,
Exception: other_exception_handler,
},
factory_class=FactoryClass(http=MsgPackRequest),
cors_config={},
)
def load_asr_model(*, device="cuda", hub="ms"):
return AutoModel(
model="iic/SenseVoiceSmall",
device=device,
disable_pbar=True,
hub=hub,
)
# Each worker process created by Uvicorn has its own memory space,
# meaning that models and variables are not shared between processes.
# Therefore, any global variables (like `llama_queue` or `decoder_model`)
# will not be shared across workers.
# Multi-threading for deep learning can cause issues, such as inconsistent
# outputs if multiple threads access the same buffers simultaneously.
# Instead, it's better to use multiprocessing or independent models per thread.
@app.on_startup
def initialize_app(app: Kui):
global args, llama_queue, tokenizer, config, decoder_model, vad_model, asr_model, prompt_tokens, prompt_texts
prompt_tokens, prompt_texts = [], []
args = parse_args() # args same as ones in other processes
args.precision = torch.half if args.half else torch.bfloat16
if args.load_asr_model:
logger.info(f"Loading ASR model...")
asr_model = load_asr_model(device=args.device)
logger.info("Loading Llama model...")
if args.mode == "tts":
llama_queue = launch_thread_safe_queue(
checkpoint_path=args.llama_checkpoint_path,
device=args.device,
precision=args.precision,
compile=args.compile,
)
else:
llama_queue, tokenizer, config = launch_thread_safe_queue_agent(
checkpoint_path=args.llama_checkpoint_path,
device=args.device,
precision=args.precision,
compile=args.compile,
)
logger.info("Llama model loaded, loading VQ-GAN model...")
decoder_model = load_decoder_model(
config_name=args.decoder_config_name,
checkpoint_path=args.decoder_checkpoint_path,
device=args.device,
)
logger.info("VQ-GAN model loaded, warming up...")
vad_model = load_silero_vad()
logger.info("VAD model loaded, warming up...")
if args.mode == "tts":
# Dry run to ensure models work and avoid first-time latency
list(
inference(
ServeTTSRequest(
text="Hello world.",
references=[],
reference_id=None,
max_new_tokens=0,
chunk_length=200,
top_p=0.7,
repetition_penalty=1.2,
temperature=0.7,
emotion=None,
format="wav",
)
)
)
logger.info(f"Warming up done, starting server at http://{args.listen}")
if __name__ == "__main__":
import uvicorn
args = parse_args()
host, port = args.listen.split(":")
uvicorn.run(
"tools.api:app",
host=host,
port=int(port),
workers=args.workers,
log_level="info",
)
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