mirror of
https://github.com/RVC-Boss/GPT-SoVITS.git
synced 2026-05-12 21:08:11 +08:00
1171 lines
45 KiB
Python
1171 lines
45 KiB
Python
"""
|
||
# WebAPI文档
|
||
|
||
` python api_v2.py -a 127.0.0.1 -p 9880 -c GPT_SoVITS/configs/tts_infer.yaml `
|
||
|
||
## 执行参数:
|
||
`-a` - `绑定地址, 默认"127.0.0.1"`
|
||
`-p` - `绑定端口, 默认9880`
|
||
`-c` - `TTS配置文件路径, 默认"GPT_SoVITS/configs/tts_infer.yaml"`
|
||
|
||
## 调用:
|
||
|
||
### 推理
|
||
|
||
endpoint: `/tts`
|
||
GET:
|
||
```
|
||
http://127.0.0.1:9880/tts?text=先帝创业未半而中道崩殂,今天下三分,益州疲弊,此诚危急存亡之秋也。&text_lang=zh&ref_audio_path=archive_jingyuan_1.wav&prompt_lang=zh&prompt_text=我是「罗浮」云骑将军景元。不必拘谨,「将军」只是一时的身份,你称呼我景元便可&text_split_method=cut5&batch_size=1&media_type=wav&streaming_mode=true
|
||
```
|
||
|
||
POST:
|
||
```json
|
||
{
|
||
"text": "", # str.(required) text to be synthesized
|
||
"text_lang: "", # str.(required) language of the text to be synthesized
|
||
"ref_audio_path": "", # str.(required) reference audio path
|
||
"aux_ref_audio_paths": [], # list.(optional) auxiliary reference audio paths for multi-speaker tone fusion
|
||
"prompt_text": "", # str.(optional) prompt text for the reference audio
|
||
"prompt_lang": "", # str.(required) language of the prompt text for the reference audio
|
||
"top_k": 15, # int. top k sampling
|
||
"top_p": 1, # float. top p sampling
|
||
"temperature": 1, # float. temperature for sampling
|
||
"text_split_method": "cut5", # str. text split method, see text_segmentation_method.py for details.
|
||
"batch_size": 1, # int. batch size for inference
|
||
"batch_threshold": 0.75, # float. threshold for batch splitting.
|
||
"split_bucket": True, # bool. whether to split the batch into multiple buckets.
|
||
"speed_factor":1.0, # float. control the speed of the synthesized audio.
|
||
"fragment_interval":0.3, # float. to control the interval of the audio fragment.
|
||
"seed": -1, # int. random seed for reproducibility.
|
||
"parallel_infer": True, # bool. whether to use parallel inference.
|
||
"repetition_penalty": 1.35, # float. repetition penalty for T2S model.
|
||
"sample_steps": 32, # int. number of sampling steps for VITS model V3.
|
||
"super_sampling": False, # bool. whether to use super-sampling for audio when using VITS model V3.
|
||
"streaming_mode": False, # bool or int. return audio chunk by chunk.T he available options are: 0,1,2,3 or True/False (0/False: Disabled | 1/True: Best Quality, Slowest response speed (old version streaming_mode) | 2: Medium Quality, Slow response speed | 3: Lower Quality, Faster response speed )
|
||
"overlap_length": 2, # int. overlap length of semantic tokens for streaming mode.
|
||
"min_chunk_length": 16, # int. The minimum chunk length of semantic tokens for streaming mode. (affects audio chunk size)
|
||
}
|
||
```
|
||
|
||
RESP:
|
||
成功: 直接返回 wav 音频流, http code 200
|
||
失败: 返回包含错误信息的 json, http code 400
|
||
|
||
### 命令控制
|
||
|
||
endpoint: `/control`
|
||
|
||
command:
|
||
"restart": 重新运行
|
||
"exit": 结束运行
|
||
|
||
GET:
|
||
```
|
||
http://127.0.0.1:9880/control?command=restart
|
||
```
|
||
POST:
|
||
```json
|
||
{
|
||
"command": "restart"
|
||
}
|
||
```
|
||
|
||
RESP: 无
|
||
|
||
|
||
### 切换GPT模型
|
||
|
||
endpoint: `/set_gpt_weights`
|
||
|
||
GET:
|
||
```
|
||
http://127.0.0.1:9880/set_gpt_weights?weights_path=GPT_SoVITS/pretrained_models/s1bert25hz-2kh-longer-epoch=68e-step=50232.ckpt
|
||
```
|
||
RESP:
|
||
成功: 返回"success", http code 200
|
||
失败: 返回包含错误信息的 json, http code 400
|
||
|
||
|
||
### 切换Sovits模型
|
||
|
||
endpoint: `/set_sovits_weights`
|
||
|
||
GET:
|
||
```
|
||
http://127.0.0.1:9880/set_sovits_weights?weights_path=GPT_SoVITS/pretrained_models/s2G488k.pth
|
||
```
|
||
|
||
RESP:
|
||
成功: 返回"success", http code 200
|
||
失败: 返回包含错误信息的 json, http code 400
|
||
|
||
"""
|
||
|
||
import asyncio
|
||
import os
|
||
import sys
|
||
import time
|
||
import traceback
|
||
import uuid
|
||
from dataclasses import dataclass
|
||
from pathlib import Path
|
||
from typing import Generator, List, Union
|
||
|
||
now_dir = os.getcwd()
|
||
sys.path.append(now_dir)
|
||
sys.path.append("%s/GPT_SoVITS" % (now_dir))
|
||
|
||
import argparse
|
||
import subprocess
|
||
import wave
|
||
import signal
|
||
import numpy as np
|
||
import soundfile as sf
|
||
import torch
|
||
from fastapi import FastAPI, Response
|
||
from fastapi.responses import StreamingResponse, JSONResponse
|
||
import uvicorn
|
||
from io import BytesIO
|
||
from tools.i18n.i18n import I18nAuto
|
||
from GPT_SoVITS.TTS_infer_pack.TTS import TTS, TTS_Config
|
||
from GPT_SoVITS.TTS_infer_pack.t2s_scheduler import (
|
||
SchedulerRequestSpec,
|
||
T2SFinishedItem,
|
||
T2SRunningRequest,
|
||
T2SRequestState,
|
||
prepare_request_state,
|
||
run_decode_step_for_running,
|
||
run_prefill_step,
|
||
run_scheduler_continuous,
|
||
)
|
||
from GPT_SoVITS.TTS_infer_pack.text_segmentation_method import get_method_names as get_cut_method_names
|
||
from pydantic import BaseModel
|
||
import threading
|
||
|
||
# print(sys.path)
|
||
i18n = I18nAuto()
|
||
cut_method_names = get_cut_method_names()
|
||
|
||
parser = argparse.ArgumentParser(description="GPT-SoVITS api")
|
||
parser.add_argument("-c", "--tts_config", type=str, default="GPT_SoVITS/configs/tts_infer.yaml", help="tts_infer路径")
|
||
parser.add_argument("-a", "--bind_addr", type=str, default="127.0.0.1", help="default: 127.0.0.1")
|
||
parser.add_argument("-p", "--port", type=int, default="9880", help="default: 9880")
|
||
args = parser.parse_args()
|
||
config_path = args.tts_config
|
||
# device = args.device
|
||
port = args.port
|
||
host = args.bind_addr
|
||
argv = sys.argv
|
||
|
||
if config_path in [None, ""]:
|
||
config_path = "GPT-SoVITS/configs/tts_infer.yaml"
|
||
|
||
tts_config = TTS_Config(config_path)
|
||
print(tts_config)
|
||
tts_pipeline = TTS(tts_config)
|
||
|
||
APP = FastAPI()
|
||
|
||
|
||
class TTS_Request(BaseModel):
|
||
text: str = None
|
||
text_lang: str = None
|
||
ref_audio_path: str = None
|
||
aux_ref_audio_paths: list = None
|
||
prompt_lang: str = None
|
||
prompt_text: str = ""
|
||
top_k: int = 15
|
||
top_p: float = 1
|
||
temperature: float = 1
|
||
text_split_method: str = "cut5"
|
||
batch_size: int = 1
|
||
batch_threshold: float = 0.75
|
||
split_bucket: bool = True
|
||
speed_factor: float = 1.0
|
||
fragment_interval: float = 0.3
|
||
seed: int = -1
|
||
media_type: str = "wav"
|
||
streaming_mode: Union[bool, int] = False
|
||
parallel_infer: bool = True
|
||
repetition_penalty: float = 1.35
|
||
sample_steps: int = 32
|
||
super_sampling: bool = False
|
||
overlap_length: int = 2
|
||
min_chunk_length: int = 16
|
||
|
||
|
||
class Scheduler_Debug_Request_Item(BaseModel):
|
||
request_id: str | None = None
|
||
text: str
|
||
text_lang: str
|
||
ref_audio_path: str
|
||
prompt_lang: str
|
||
prompt_text: str = ""
|
||
top_k: int = 15
|
||
top_p: float = 1
|
||
temperature: float = 1
|
||
repetition_penalty: float = 1.35
|
||
early_stop_num: int = -1
|
||
ready_step: int = 0
|
||
|
||
|
||
class Scheduler_Debug_Request(BaseModel):
|
||
requests: List[Scheduler_Debug_Request_Item]
|
||
max_steps: int = 1500
|
||
seed: int = -1
|
||
|
||
|
||
class Scheduler_Submit_Request(BaseModel):
|
||
request_id: str | None = None
|
||
text: str
|
||
text_lang: str
|
||
ref_audio_path: str
|
||
prompt_lang: str
|
||
prompt_text: str = ""
|
||
top_k: int = 15
|
||
top_p: float = 1
|
||
temperature: float = 1
|
||
repetition_penalty: float = 1.35
|
||
early_stop_num: int = -1
|
||
speed_factor: float = 1.0
|
||
sample_steps: int = 32
|
||
media_type: str = "wav"
|
||
timeout_sec: float = 30.0
|
||
|
||
|
||
@dataclass
|
||
class SchedulerPendingJob:
|
||
request_id: str
|
||
state: T2SRequestState
|
||
done_event: threading.Event
|
||
enqueue_time: float
|
||
speed_factor: float
|
||
sample_steps: int
|
||
media_type: str
|
||
prepare_ms: float = 0.0
|
||
prepare_wall_ms: float = 0.0
|
||
first_schedule_time: float | None = None
|
||
prefill_ms: float = 0.0
|
||
decode_ms: float = 0.0
|
||
synth_ms: float = 0.0
|
||
pack_ms: float = 0.0
|
||
decode_steps: int = 0
|
||
result: dict | None = None
|
||
sample_rate: int | None = None
|
||
audio_data: np.ndarray | None = None
|
||
error: str | None = None
|
||
|
||
|
||
class SchedulerDebugWorker:
|
||
def __init__(self, tts: TTS, max_steps: int = 1500, micro_batch_wait_ms: int = 5):
|
||
self.tts = tts
|
||
self.max_steps = max_steps
|
||
self.micro_batch_wait_s = micro_batch_wait_ms / 1000.0
|
||
self.prepare_lock = threading.Lock()
|
||
self.condition = threading.Condition()
|
||
self.pending_jobs: List[SchedulerPendingJob] = []
|
||
self.running_requests: List[T2SRunningRequest] = []
|
||
self.job_map: dict[str, SchedulerPendingJob] = {}
|
||
self.total_finished = 0
|
||
self.total_submitted = 0
|
||
self.worker_thread = threading.Thread(target=self._run_loop, name="t2s-scheduler-debug-worker", daemon=True)
|
||
self.worker_thread.start()
|
||
|
||
def _sync_device(self) -> None:
|
||
try:
|
||
device_str = str(self.tts.configs.device)
|
||
if device_str.startswith("cuda") and torch.cuda.is_available():
|
||
torch.cuda.synchronize(self.tts.configs.device)
|
||
elif device_str == "mps" and hasattr(torch, "mps") and hasattr(torch.mps, "synchronize"):
|
||
torch.mps.synchronize()
|
||
except Exception:
|
||
pass
|
||
|
||
def prepare_state(self, spec: SchedulerRequestSpec) -> T2SRequestState:
|
||
with self.prepare_lock:
|
||
return prepare_request_state(self.tts, spec)
|
||
|
||
def submit(
|
||
self,
|
||
state: T2SRequestState,
|
||
speed_factor: float,
|
||
sample_steps: int,
|
||
media_type: str,
|
||
prepare_ms: float,
|
||
prepare_wall_ms: float,
|
||
) -> SchedulerPendingJob:
|
||
job = SchedulerPendingJob(
|
||
request_id=state.request_id,
|
||
state=state,
|
||
done_event=threading.Event(),
|
||
enqueue_time=time.perf_counter(),
|
||
speed_factor=float(speed_factor),
|
||
sample_steps=int(sample_steps),
|
||
media_type=media_type,
|
||
prepare_ms=float(prepare_ms),
|
||
prepare_wall_ms=float(prepare_wall_ms),
|
||
)
|
||
with self.condition:
|
||
self.pending_jobs.append(job)
|
||
self.job_map[job.request_id] = job
|
||
self.total_submitted += 1
|
||
self.condition.notify_all()
|
||
return job
|
||
|
||
def _mark_prefill_started(self, jobs: List[SchedulerPendingJob], started_at: float) -> None:
|
||
with self.condition:
|
||
for job in jobs:
|
||
tracked_job = self.job_map.get(job.request_id)
|
||
if tracked_job is not None and tracked_job.first_schedule_time is None:
|
||
tracked_job.first_schedule_time = started_at
|
||
|
||
def _add_prefill_time(self, jobs: List[SchedulerPendingJob], elapsed_s: float) -> None:
|
||
elapsed_ms = elapsed_s * 1000.0
|
||
with self.condition:
|
||
for job in jobs:
|
||
tracked_job = self.job_map.get(job.request_id)
|
||
if tracked_job is not None:
|
||
tracked_job.prefill_ms += elapsed_ms
|
||
|
||
def _add_decode_time(self, request_ids: List[str], elapsed_s: float) -> None:
|
||
elapsed_ms = elapsed_s * 1000.0
|
||
with self.condition:
|
||
for request_id in request_ids:
|
||
job = self.job_map.get(request_id)
|
||
if job is not None:
|
||
job.decode_ms += elapsed_ms
|
||
job.decode_steps += 1
|
||
|
||
def _synthesize_finished_audio(self, job: SchedulerPendingJob, item: T2SFinishedItem) -> tuple[int, np.ndarray]:
|
||
semantic_tokens = item.semantic_tokens.unsqueeze(0).unsqueeze(0).to(self.tts.configs.device)
|
||
phones = job.state.phones.unsqueeze(0).to(self.tts.configs.device)
|
||
audio_fragment = self.tts.synthesize_audio_request_local(
|
||
semantic_tokens=semantic_tokens,
|
||
phones=phones,
|
||
prompt_semantic=job.state.prompt_semantic,
|
||
prompt_phones=job.state.prompt_phones,
|
||
refer_spec=job.state.refer_spec,
|
||
raw_audio=job.state.raw_audio,
|
||
raw_sr=job.state.raw_sr,
|
||
speed=float(job.speed_factor),
|
||
sample_steps=int(job.sample_steps),
|
||
)
|
||
output_sr = self.tts.configs.sampling_rate if not self.tts.configs.use_vocoder else self.tts.vocoder_configs["sr"]
|
||
return self.tts.audio_postprocess(
|
||
audio=[[audio_fragment]],
|
||
sr=int(output_sr),
|
||
batch_index_list=None,
|
||
speed_factor=float(job.speed_factor),
|
||
split_bucket=False,
|
||
fragment_interval=0.0,
|
||
super_sampling=False,
|
||
)
|
||
|
||
def get_state(self) -> dict:
|
||
with self.condition:
|
||
return {
|
||
"pending_jobs": len(self.pending_jobs),
|
||
"running_requests": len(self.running_requests),
|
||
"tracked_jobs": len(self.job_map),
|
||
"total_submitted": self.total_submitted,
|
||
"total_finished": self.total_finished,
|
||
"max_steps": self.max_steps,
|
||
"micro_batch_wait_ms": int(self.micro_batch_wait_s * 1000),
|
||
}
|
||
|
||
def _finalize_finished(self, items: List[T2SFinishedItem]) -> None:
|
||
if not items:
|
||
return
|
||
jobs_to_finalize: List[tuple[SchedulerPendingJob, T2SFinishedItem]] = []
|
||
with self.condition:
|
||
for item in items:
|
||
job = self.job_map.get(item.request_id)
|
||
if job is not None:
|
||
jobs_to_finalize.append((job, item))
|
||
|
||
for job, item in jobs_to_finalize:
|
||
try:
|
||
self._sync_device()
|
||
synth_start = time.perf_counter()
|
||
sample_rate, audio_data = self._synthesize_finished_audio(job, item)
|
||
self._sync_device()
|
||
synth_ms = (time.perf_counter() - synth_start) * 1000.0
|
||
except Exception as exc:
|
||
self._finalize_error([item.request_id], str(exc))
|
||
continue
|
||
|
||
finished_at = time.perf_counter()
|
||
with self.condition:
|
||
if self.job_map.get(item.request_id) is not job:
|
||
continue
|
||
queue_wait_ms = 0.0
|
||
if job.first_schedule_time is not None:
|
||
queue_wait_ms = max(0.0, (job.first_schedule_time - job.enqueue_time) * 1000.0)
|
||
worker_total_ms = max(0.0, (finished_at - job.enqueue_time) * 1000.0)
|
||
job.synth_ms += synth_ms
|
||
job.sample_rate = int(sample_rate)
|
||
job.audio_data = audio_data
|
||
prepare_profile = dict(job.state.prepare_profile)
|
||
job.result = {
|
||
"request_id": item.request_id,
|
||
"semantic_len": int(item.semantic_tokens.shape[0]),
|
||
"finish_idx": int(item.finish_idx),
|
||
"finish_reason": item.finish_reason,
|
||
"prepare_ms": job.prepare_ms,
|
||
"prepare_wall_ms": job.prepare_wall_ms,
|
||
"prepare_profile": prepare_profile,
|
||
"queue_wait_ms": queue_wait_ms,
|
||
"prefill_ms": job.prefill_ms,
|
||
"decode_ms": job.decode_ms,
|
||
"synth_ms": job.synth_ms,
|
||
"worker_total_ms": worker_total_ms,
|
||
"decode_steps": int(job.decode_steps),
|
||
"sample_rate": int(sample_rate),
|
||
"media_type": job.media_type,
|
||
}
|
||
job.done_event.set()
|
||
self.job_map.pop(item.request_id, None)
|
||
self.total_finished += 1
|
||
|
||
def _finalize_error(self, request_ids: List[str], error: str) -> None:
|
||
if not request_ids:
|
||
return
|
||
with self.condition:
|
||
for request_id in request_ids:
|
||
job = self.job_map.get(request_id)
|
||
if job is None:
|
||
continue
|
||
job.error = error
|
||
job.done_event.set()
|
||
self.job_map.pop(request_id, None)
|
||
self.total_finished += 1
|
||
|
||
def _take_pending_snapshot(self, wait_for_batch: bool) -> List[SchedulerPendingJob]:
|
||
with self.condition:
|
||
if not self.pending_jobs and not self.running_requests:
|
||
self.condition.wait(timeout=self.micro_batch_wait_s)
|
||
elif wait_for_batch and self.pending_jobs:
|
||
self.condition.wait(timeout=self.micro_batch_wait_s)
|
||
if not self.pending_jobs:
|
||
return []
|
||
pending = list(self.pending_jobs)
|
||
self.pending_jobs.clear()
|
||
return pending
|
||
|
||
def _run_loop(self) -> None:
|
||
while True:
|
||
wait_for_batch = len(self.running_requests) == 0
|
||
pending_jobs = self._take_pending_snapshot(wait_for_batch=wait_for_batch)
|
||
|
||
if pending_jobs:
|
||
try:
|
||
self._sync_device()
|
||
prefill_start = time.perf_counter()
|
||
self._mark_prefill_started(pending_jobs, prefill_start)
|
||
admitted_running, admitted_finished = run_prefill_step(
|
||
self.tts.t2s_model.model,
|
||
[job.state for job in pending_jobs],
|
||
max_steps=self.max_steps,
|
||
)
|
||
self._sync_device()
|
||
self._add_prefill_time(pending_jobs, time.perf_counter() - prefill_start)
|
||
self._finalize_finished(admitted_finished)
|
||
self.running_requests.extend(admitted_running)
|
||
except Exception as exc:
|
||
self._finalize_error([job.request_id for job in pending_jobs], str(exc))
|
||
|
||
if self.running_requests:
|
||
try:
|
||
active_request_ids = [item.state.request_id for item in self.running_requests]
|
||
self._sync_device()
|
||
decode_start = time.perf_counter()
|
||
self.running_requests, step_finished = run_decode_step_for_running(
|
||
self.tts.t2s_model.model,
|
||
self.running_requests,
|
||
max_steps=self.max_steps,
|
||
)
|
||
self._sync_device()
|
||
self._add_decode_time(active_request_ids, time.perf_counter() - decode_start)
|
||
self._finalize_finished(step_finished)
|
||
except Exception as exc:
|
||
self._finalize_error(active_request_ids, str(exc))
|
||
self.running_requests = []
|
||
continue
|
||
|
||
if not pending_jobs:
|
||
time.sleep(self.micro_batch_wait_s)
|
||
|
||
|
||
scheduler_debug_worker = SchedulerDebugWorker(tts_pipeline)
|
||
|
||
|
||
def pack_ogg(io_buffer: BytesIO, data: np.ndarray, rate: int):
|
||
# Author: AkagawaTsurunaki
|
||
# Issue:
|
||
# Stack overflow probabilistically occurs
|
||
# when the function `sf_writef_short` of `libsndfile_64bit.dll` is called
|
||
# using the Python library `soundfile`
|
||
# Note:
|
||
# This is an issue related to `libsndfile`, not this project itself.
|
||
# It happens when you generate a large audio tensor (about 499804 frames in my PC)
|
||
# and try to convert it to an ogg file.
|
||
# Related:
|
||
# https://github.com/RVC-Boss/GPT-SoVITS/issues/1199
|
||
# https://github.com/libsndfile/libsndfile/issues/1023
|
||
# https://github.com/bastibe/python-soundfile/issues/396
|
||
# Suggestion:
|
||
# Or split the whole audio data into smaller audio segment to avoid stack overflow?
|
||
|
||
def handle_pack_ogg():
|
||
with sf.SoundFile(io_buffer, mode="w", samplerate=rate, channels=1, format="ogg") as audio_file:
|
||
audio_file.write(data)
|
||
|
||
|
||
|
||
# See: https://docs.python.org/3/library/threading.html
|
||
# The stack size of this thread is at least 32768
|
||
# If stack overflow error still occurs, just modify the `stack_size`.
|
||
# stack_size = n * 4096, where n should be a positive integer.
|
||
# Here we chose n = 4096.
|
||
stack_size = 4096 * 4096
|
||
try:
|
||
threading.stack_size(stack_size)
|
||
pack_ogg_thread = threading.Thread(target=handle_pack_ogg)
|
||
pack_ogg_thread.start()
|
||
pack_ogg_thread.join()
|
||
except RuntimeError as e:
|
||
# If changing the thread stack size is unsupported, a RuntimeError is raised.
|
||
print("RuntimeError: {}".format(e))
|
||
print("Changing the thread stack size is unsupported.")
|
||
except ValueError as e:
|
||
# If the specified stack size is invalid, a ValueError is raised and the stack size is unmodified.
|
||
print("ValueError: {}".format(e))
|
||
print("The specified stack size is invalid.")
|
||
|
||
return io_buffer
|
||
|
||
|
||
def pack_raw(io_buffer: BytesIO, data: np.ndarray, rate: int):
|
||
io_buffer.write(data.tobytes())
|
||
return io_buffer
|
||
|
||
|
||
def pack_wav(io_buffer: BytesIO, data: np.ndarray, rate: int):
|
||
io_buffer = BytesIO()
|
||
sf.write(io_buffer, data, rate, format="wav")
|
||
return io_buffer
|
||
|
||
|
||
def pack_aac(io_buffer: BytesIO, data: np.ndarray, rate: int):
|
||
process = subprocess.Popen(
|
||
[
|
||
"ffmpeg",
|
||
"-f",
|
||
"s16le", # 输入16位有符号小端整数PCM
|
||
"-ar",
|
||
str(rate), # 设置采样率
|
||
"-ac",
|
||
"1", # 单声道
|
||
"-i",
|
||
"pipe:0", # 从管道读取输入
|
||
"-c:a",
|
||
"aac", # 音频编码器为AAC
|
||
"-b:a",
|
||
"192k", # 比特率
|
||
"-vn", # 不包含视频
|
||
"-f",
|
||
"adts", # 输出AAC数据流格式
|
||
"pipe:1", # 将输出写入管道
|
||
],
|
||
stdin=subprocess.PIPE,
|
||
stdout=subprocess.PIPE,
|
||
stderr=subprocess.PIPE,
|
||
)
|
||
out, _ = process.communicate(input=data.tobytes())
|
||
io_buffer.write(out)
|
||
return io_buffer
|
||
|
||
|
||
def pack_audio(io_buffer: BytesIO, data: np.ndarray, rate: int, media_type: str):
|
||
if media_type == "ogg":
|
||
io_buffer = pack_ogg(io_buffer, data, rate)
|
||
elif media_type == "aac":
|
||
io_buffer = pack_aac(io_buffer, data, rate)
|
||
elif media_type == "wav":
|
||
io_buffer = pack_wav(io_buffer, data, rate)
|
||
else:
|
||
io_buffer = pack_raw(io_buffer, data, rate)
|
||
io_buffer.seek(0)
|
||
return io_buffer
|
||
|
||
|
||
# from https://huggingface.co/spaces/coqui/voice-chat-with-mistral/blob/main/app.py
|
||
def wave_header_chunk(frame_input=b"", channels=1, sample_width=2, sample_rate=32000):
|
||
# This will create a wave header then append the frame input
|
||
# It should be first on a streaming wav file
|
||
# Other frames better should not have it (else you will hear some artifacts each chunk start)
|
||
wav_buf = BytesIO()
|
||
with wave.open(wav_buf, "wb") as vfout:
|
||
vfout.setnchannels(channels)
|
||
vfout.setsampwidth(sample_width)
|
||
vfout.setframerate(sample_rate)
|
||
vfout.writeframes(frame_input)
|
||
|
||
wav_buf.seek(0)
|
||
return wav_buf.read()
|
||
|
||
|
||
def handle_control(command: str):
|
||
if command == "restart":
|
||
os.execl(sys.executable, sys.executable, *argv)
|
||
elif command == "exit":
|
||
os.kill(os.getpid(), signal.SIGTERM)
|
||
exit(0)
|
||
|
||
|
||
def check_params(req: dict):
|
||
text: str = req.get("text", "")
|
||
text_lang: str = req.get("text_lang", "")
|
||
ref_audio_path: str = req.get("ref_audio_path", "")
|
||
streaming_mode: bool = req.get("streaming_mode", False)
|
||
media_type: str = req.get("media_type", "wav")
|
||
prompt_lang: str = req.get("prompt_lang", "")
|
||
text_split_method: str = req.get("text_split_method", "cut5")
|
||
|
||
if ref_audio_path in [None, ""]:
|
||
return JSONResponse(status_code=400, content={"message": "ref_audio_path is required"})
|
||
if text in [None, ""]:
|
||
return JSONResponse(status_code=400, content={"message": "text is required"})
|
||
if text_lang in [None, ""]:
|
||
return JSONResponse(status_code=400, content={"message": "text_lang is required"})
|
||
elif text_lang.lower() not in tts_config.languages:
|
||
return JSONResponse(
|
||
status_code=400,
|
||
content={"message": f"text_lang: {text_lang} is not supported in version {tts_config.version}"},
|
||
)
|
||
if prompt_lang in [None, ""]:
|
||
return JSONResponse(status_code=400, content={"message": "prompt_lang is required"})
|
||
elif prompt_lang.lower() not in tts_config.languages:
|
||
return JSONResponse(
|
||
status_code=400,
|
||
content={"message": f"prompt_lang: {prompt_lang} is not supported in version {tts_config.version}"},
|
||
)
|
||
if media_type not in ["wav", "raw", "ogg", "aac"]:
|
||
return JSONResponse(status_code=400, content={"message": f"media_type: {media_type} is not supported"})
|
||
# elif media_type == "ogg" and not streaming_mode:
|
||
# return JSONResponse(status_code=400, content={"message": "ogg format is not supported in non-streaming mode"})
|
||
|
||
if text_split_method not in cut_method_names:
|
||
return JSONResponse(
|
||
status_code=400, content={"message": f"text_split_method:{text_split_method} is not supported"}
|
||
)
|
||
|
||
return None
|
||
|
||
|
||
def set_scheduler_seed(seed: int):
|
||
if seed in ["", None]:
|
||
return
|
||
seed = int(seed)
|
||
if seed < 0:
|
||
return
|
||
np.random.seed(seed)
|
||
torch.manual_seed(seed)
|
||
if torch.cuda.is_available():
|
||
torch.cuda.manual_seed(seed)
|
||
torch.cuda.manual_seed_all(seed)
|
||
|
||
|
||
def build_scheduler_request_specs(request_items: List[Scheduler_Debug_Request_Item]) -> List[SchedulerRequestSpec]:
|
||
specs: List[SchedulerRequestSpec] = []
|
||
for index, item in enumerate(request_items):
|
||
payload = item.dict()
|
||
req = {
|
||
"text": payload["text"],
|
||
"text_lang": payload["text_lang"].lower(),
|
||
"ref_audio_path": payload["ref_audio_path"],
|
||
"aux_ref_audio_paths": None,
|
||
"prompt_text": payload["prompt_text"],
|
||
"prompt_lang": payload["prompt_lang"].lower(),
|
||
"top_k": payload["top_k"],
|
||
"top_p": payload["top_p"],
|
||
"temperature": payload["temperature"],
|
||
"text_split_method": "cut5",
|
||
"batch_size": 1,
|
||
"batch_threshold": 0.75,
|
||
"speed_factor": 1.0,
|
||
"split_bucket": False,
|
||
"fragment_interval": 0.3,
|
||
"seed": -1,
|
||
"media_type": "wav",
|
||
"streaming_mode": False,
|
||
"parallel_infer": False,
|
||
"repetition_penalty": payload["repetition_penalty"],
|
||
"sample_steps": 32,
|
||
"super_sampling": False,
|
||
"overlap_length": 2,
|
||
"min_chunk_length": 16,
|
||
}
|
||
check_res = check_params(req)
|
||
if check_res is not None:
|
||
detail = check_res.body.decode("utf-8") if hasattr(check_res, "body") else str(check_res)
|
||
raise ValueError(f"request[{index}] 参数非法: {detail}")
|
||
specs.append(
|
||
SchedulerRequestSpec(
|
||
request_id=payload["request_id"] or f"req_{index:03d}",
|
||
ref_audio_path=Path(payload["ref_audio_path"]),
|
||
prompt_text=payload["prompt_text"],
|
||
prompt_lang=payload["prompt_lang"].lower(),
|
||
text=payload["text"],
|
||
text_lang=payload["text_lang"].lower(),
|
||
top_k=int(payload["top_k"]),
|
||
top_p=float(payload["top_p"]),
|
||
temperature=float(payload["temperature"]),
|
||
repetition_penalty=float(payload["repetition_penalty"]),
|
||
early_stop_num=int(payload["early_stop_num"]),
|
||
ready_step=int(payload["ready_step"]),
|
||
)
|
||
)
|
||
return specs
|
||
|
||
|
||
def summarize_scheduler_states(states: List[T2SRequestState]) -> List[dict]:
|
||
return [
|
||
{
|
||
"request_id": state.request_id,
|
||
"ready_step": int(state.ready_step),
|
||
"ref_audio_path": str(state.ref_audio_path),
|
||
"prompt_semantic_len": int(state.prompt_semantic.shape[0]),
|
||
"all_phone_len": int(state.all_phones.shape[0]),
|
||
"bert_len": int(state.all_bert_features.shape[-1]),
|
||
"norm_text": state.norm_text,
|
||
}
|
||
for state in states
|
||
]
|
||
|
||
|
||
def summarize_scheduler_finished(items: List[T2SFinishedItem]) -> List[dict]:
|
||
return [
|
||
{
|
||
"request_id": item.request_id,
|
||
"semantic_len": int(item.semantic_tokens.shape[0]),
|
||
"finish_idx": int(item.finish_idx),
|
||
"finish_reason": item.finish_reason,
|
||
}
|
||
for item in items
|
||
]
|
||
|
||
|
||
def prepare_scheduler_states_batch(specs: List[SchedulerRequestSpec]) -> List[T2SRequestState]:
|
||
return [scheduler_debug_worker.prepare_state(spec) for spec in specs]
|
||
|
||
|
||
def build_scheduler_submit_spec(request: Scheduler_Submit_Request) -> SchedulerRequestSpec:
|
||
payload = request.dict()
|
||
request_id = payload["request_id"] or f"job_{uuid.uuid4().hex[:12]}"
|
||
req = {
|
||
"text": payload["text"],
|
||
"text_lang": payload["text_lang"].lower(),
|
||
"ref_audio_path": payload["ref_audio_path"],
|
||
"aux_ref_audio_paths": None,
|
||
"prompt_text": payload["prompt_text"],
|
||
"prompt_lang": payload["prompt_lang"].lower(),
|
||
"top_k": payload["top_k"],
|
||
"top_p": payload["top_p"],
|
||
"temperature": payload["temperature"],
|
||
"text_split_method": "cut5",
|
||
"batch_size": 1,
|
||
"batch_threshold": 0.75,
|
||
"speed_factor": float(payload["speed_factor"]),
|
||
"split_bucket": False,
|
||
"fragment_interval": 0.3,
|
||
"seed": -1,
|
||
"media_type": payload["media_type"],
|
||
"streaming_mode": False,
|
||
"parallel_infer": False,
|
||
"repetition_penalty": payload["repetition_penalty"],
|
||
"sample_steps": int(payload["sample_steps"]),
|
||
"super_sampling": False,
|
||
"overlap_length": 2,
|
||
"min_chunk_length": 16,
|
||
}
|
||
check_res = check_params(req)
|
||
if check_res is not None:
|
||
detail = check_res.body.decode("utf-8") if hasattr(check_res, "body") else str(check_res)
|
||
raise ValueError(f"request 参数非法: {detail}")
|
||
return SchedulerRequestSpec(
|
||
request_id=request_id,
|
||
ref_audio_path=Path(payload["ref_audio_path"]),
|
||
prompt_text=payload["prompt_text"],
|
||
prompt_lang=payload["prompt_lang"].lower(),
|
||
text=payload["text"],
|
||
text_lang=payload["text_lang"].lower(),
|
||
top_k=int(payload["top_k"]),
|
||
top_p=float(payload["top_p"]),
|
||
temperature=float(payload["temperature"]),
|
||
repetition_penalty=float(payload["repetition_penalty"]),
|
||
early_stop_num=int(payload["early_stop_num"]),
|
||
ready_step=0,
|
||
)
|
||
|
||
|
||
async def tts_scheduler_debug_handle(request: Scheduler_Debug_Request):
|
||
try:
|
||
set_scheduler_seed(request.seed)
|
||
specs = build_scheduler_request_specs(request.requests)
|
||
states = await asyncio.to_thread(prepare_scheduler_states_batch, specs)
|
||
finished = run_scheduler_continuous(tts_pipeline.t2s_model.model, states, max_steps=int(request.max_steps))
|
||
return JSONResponse(
|
||
status_code=200,
|
||
content={
|
||
"message": "success",
|
||
"request_count": len(states),
|
||
"max_steps": int(request.max_steps),
|
||
"requests": summarize_scheduler_states(states),
|
||
"finished": summarize_scheduler_finished(finished),
|
||
},
|
||
)
|
||
except Exception as e:
|
||
return JSONResponse(
|
||
status_code=400,
|
||
content={"message": "scheduler debug failed", "Exception": str(e)},
|
||
)
|
||
|
||
|
||
async def tts_scheduler_submit_handle(request: Scheduler_Submit_Request):
|
||
try:
|
||
request_start = time.perf_counter()
|
||
spec = build_scheduler_submit_spec(request)
|
||
prepare_start = time.perf_counter()
|
||
state = await asyncio.to_thread(scheduler_debug_worker.prepare_state, spec)
|
||
prepare_wall_ms = (time.perf_counter() - prepare_start) * 1000.0
|
||
prepare_ms = float(state.prepare_profile.get("total_ms", prepare_wall_ms))
|
||
job = scheduler_debug_worker.submit(
|
||
state,
|
||
speed_factor=float(request.speed_factor),
|
||
sample_steps=int(request.sample_steps),
|
||
media_type=request.media_type,
|
||
prepare_ms=prepare_ms,
|
||
prepare_wall_ms=prepare_wall_ms,
|
||
)
|
||
timeout_ok = await asyncio.to_thread(job.done_event.wait, float(request.timeout_sec))
|
||
if not timeout_ok:
|
||
return JSONResponse(
|
||
status_code=202,
|
||
content={
|
||
"message": "queued",
|
||
"request_id": job.request_id,
|
||
"timings": {
|
||
"prepare_ms": prepare_ms,
|
||
"prepare_wall_ms": prepare_wall_ms,
|
||
"request_elapsed_ms": max(0.0, (time.perf_counter() - request_start) * 1000.0),
|
||
},
|
||
"worker_state": scheduler_debug_worker.get_state(),
|
||
},
|
||
)
|
||
if job.error is not None:
|
||
return JSONResponse(
|
||
status_code=400,
|
||
content={"message": "scheduler submit failed", "request_id": job.request_id, "Exception": job.error},
|
||
)
|
||
if job.audio_data is None or job.sample_rate is None:
|
||
return JSONResponse(
|
||
status_code=500,
|
||
content={
|
||
"message": "scheduler submit failed",
|
||
"request_id": job.request_id,
|
||
"Exception": "job finished without audio payload",
|
||
},
|
||
)
|
||
pack_start = time.perf_counter()
|
||
audio_data = pack_audio(BytesIO(), job.audio_data, int(job.sample_rate), job.media_type).getvalue()
|
||
pack_ms = (time.perf_counter() - pack_start) * 1000.0
|
||
job.pack_ms = pack_ms
|
||
request_total_ms = max(0.0, (time.perf_counter() - request_start) * 1000.0)
|
||
headers = {
|
||
"X-Request-Id": job.request_id,
|
||
"X-Semantic-Len": str(job.result["semantic_len"]) if job.result is not None else "0",
|
||
"X-Finish-Reason": job.result["finish_reason"] if job.result is not None else "unknown",
|
||
"X-Queue-Wait-Ms": (
|
||
f"{float(job.result['queue_wait_ms']):.3f}" if job.result is not None else "0.000"
|
||
),
|
||
"X-Prepare-Ms": f"{prepare_ms:.3f}",
|
||
"X-Prepare-Wall-Ms": f"{prepare_wall_ms:.3f}",
|
||
"X-Prefill-Ms": f"{float(job.result['prefill_ms']):.3f}" if job.result is not None else "0.000",
|
||
"X-Decode-Ms": f"{float(job.result['decode_ms']):.3f}" if job.result is not None else "0.000",
|
||
"X-Synth-Ms": f"{float(job.result['synth_ms']):.3f}" if job.result is not None else "0.000",
|
||
"X-Pack-Ms": f"{pack_ms:.3f}",
|
||
"X-Worker-Total-Ms": (
|
||
f"{float(job.result['worker_total_ms']):.3f}" if job.result is not None else "0.000"
|
||
),
|
||
"X-Request-Total-Ms": f"{request_total_ms:.3f}",
|
||
"X-Decode-Steps": str(job.result["decode_steps"]) if job.result is not None else "0",
|
||
}
|
||
if job.result is not None:
|
||
prepare_profile = job.result.get("prepare_profile", {})
|
||
headers.update(
|
||
{
|
||
"X-Prepare-Prompt-Text-Ms": f"{float(prepare_profile.get('prompt_text_features_ms', 0.0)):.3f}",
|
||
"X-Prepare-Target-Text-Ms": f"{float(prepare_profile.get('text_features_ms', 0.0)):.3f}",
|
||
"X-Prepare-Prompt-Semantic-Ms": f"{float(prepare_profile.get('prompt_semantic_ms', 0.0)):.3f}",
|
||
"X-Prepare-Ref-Spec-Ms": f"{float(prepare_profile.get('ref_spec_ms', 0.0)):.3f}",
|
||
"X-Prepare-Tensorize-Ms": f"{float(prepare_profile.get('tensorize_ms', 0.0)):.3f}",
|
||
"X-Prepare-Profile-Wall-Ms": f"{float(prepare_profile.get('wall_total_ms', 0.0)):.3f}",
|
||
}
|
||
)
|
||
return Response(audio_data, media_type=f"audio/{job.media_type}", headers=headers)
|
||
except Exception as e:
|
||
return JSONResponse(
|
||
status_code=400,
|
||
content={"message": "scheduler submit failed", "Exception": str(e)},
|
||
)
|
||
|
||
|
||
async def tts_handle(req: dict):
|
||
"""
|
||
Text to speech handler.
|
||
|
||
Args:
|
||
req (dict):
|
||
{
|
||
"text": "", # str.(required) text to be synthesized
|
||
"text_lang: "", # str.(required) language of the text to be synthesized
|
||
"ref_audio_path": "", # str.(required) reference audio path
|
||
"aux_ref_audio_paths": [], # list.(optional) auxiliary reference audio paths for multi-speaker tone fusion
|
||
"prompt_text": "", # str.(optional) prompt text for the reference audio
|
||
"prompt_lang": "", # str.(required) language of the prompt text for the reference audio
|
||
"top_k": 15, # int. top k sampling
|
||
"top_p": 1, # float. top p sampling
|
||
"temperature": 1, # float. temperature for sampling
|
||
"text_split_method": "cut5", # str. text split method, see text_segmentation_method.py for details.
|
||
"batch_size": 1, # int. batch size for inference
|
||
"batch_threshold": 0.75, # float. threshold for batch splitting.
|
||
"split_bucket": True, # bool. whether to split the batch into multiple buckets.
|
||
"speed_factor":1.0, # float. control the speed of the synthesized audio.
|
||
"fragment_interval":0.3, # float. to control the interval of the audio fragment.
|
||
"seed": -1, # int. random seed for reproducibility.
|
||
"parallel_infer": True, # bool. whether to use parallel inference.
|
||
"repetition_penalty": 1.35, # float. repetition penalty for T2S model.
|
||
"sample_steps": 32, # int. number of sampling steps for VITS model V3.
|
||
"super_sampling": False, # bool. whether to use super-sampling for audio when using VITS model V3.
|
||
"streaming_mode": False, # bool or int. return audio chunk by chunk.T he available options are: 0,1,2,3 or True/False (0/False: Disabled | 1/True: Best Quality, Slowest response speed (old version streaming_mode) | 2: Medium Quality, Slow response speed | 3: Lower Quality, Faster response speed )
|
||
"overlap_length": 2, # int. overlap length of semantic tokens for streaming mode.
|
||
"min_chunk_length": 16, # int. The minimum chunk length of semantic tokens for streaming mode. (affects audio chunk size)
|
||
}
|
||
returns:
|
||
StreamingResponse: audio stream response.
|
||
"""
|
||
|
||
streaming_mode = req.get("streaming_mode", False)
|
||
return_fragment = req.get("return_fragment", False)
|
||
media_type = req.get("media_type", "wav")
|
||
|
||
check_res = check_params(req)
|
||
if check_res is not None:
|
||
return check_res
|
||
|
||
if streaming_mode == 0:
|
||
streaming_mode = False
|
||
return_fragment = False
|
||
fixed_length_chunk = False
|
||
elif streaming_mode == 1:
|
||
streaming_mode = False
|
||
return_fragment = True
|
||
fixed_length_chunk = False
|
||
elif streaming_mode == 2:
|
||
streaming_mode = True
|
||
return_fragment = False
|
||
fixed_length_chunk = False
|
||
elif streaming_mode == 3:
|
||
streaming_mode = True
|
||
return_fragment = False
|
||
fixed_length_chunk = True
|
||
|
||
else:
|
||
return JSONResponse(status_code=400, content={"message": f"the value of streaming_mode must be 0, 1, 2, 3(int) or true/false(bool)"})
|
||
|
||
req["streaming_mode"] = streaming_mode
|
||
req["return_fragment"] = return_fragment
|
||
req["fixed_length_chunk"] = fixed_length_chunk
|
||
|
||
print(f"{streaming_mode} {return_fragment} {fixed_length_chunk}")
|
||
|
||
streaming_mode = streaming_mode or return_fragment
|
||
|
||
|
||
try:
|
||
tts_generator = tts_pipeline.run(req)
|
||
|
||
if streaming_mode:
|
||
|
||
def streaming_generator(tts_generator: Generator, media_type: str):
|
||
if_frist_chunk = True
|
||
for sr, chunk in tts_generator:
|
||
if if_frist_chunk and media_type == "wav":
|
||
yield wave_header_chunk(sample_rate=sr)
|
||
media_type = "raw"
|
||
if_frist_chunk = False
|
||
yield pack_audio(BytesIO(), chunk, sr, media_type).getvalue()
|
||
|
||
# _media_type = f"audio/{media_type}" if not (streaming_mode and media_type in ["wav", "raw"]) else f"audio/x-{media_type}"
|
||
return StreamingResponse(
|
||
streaming_generator(
|
||
tts_generator,
|
||
media_type,
|
||
),
|
||
media_type=f"audio/{media_type}",
|
||
)
|
||
|
||
else:
|
||
sr, audio_data = next(tts_generator)
|
||
audio_data = pack_audio(BytesIO(), audio_data, sr, media_type).getvalue()
|
||
return Response(audio_data, media_type=f"audio/{media_type}")
|
||
except Exception as e:
|
||
return JSONResponse(status_code=400, content={"message": "tts failed", "Exception": str(e)})
|
||
|
||
|
||
@APP.get("/control")
|
||
async def control(command: str = None):
|
||
if command is None:
|
||
return JSONResponse(status_code=400, content={"message": "command is required"})
|
||
handle_control(command)
|
||
|
||
|
||
@APP.get("/tts")
|
||
async def tts_get_endpoint(
|
||
text: str = None,
|
||
text_lang: str = None,
|
||
ref_audio_path: str = None,
|
||
aux_ref_audio_paths: list = None,
|
||
prompt_lang: str = None,
|
||
prompt_text: str = "",
|
||
top_k: int = 15,
|
||
top_p: float = 1,
|
||
temperature: float = 1,
|
||
text_split_method: str = "cut5",
|
||
batch_size: int = 1,
|
||
batch_threshold: float = 0.75,
|
||
split_bucket: bool = True,
|
||
speed_factor: float = 1.0,
|
||
fragment_interval: float = 0.3,
|
||
seed: int = -1,
|
||
media_type: str = "wav",
|
||
parallel_infer: bool = True,
|
||
repetition_penalty: float = 1.35,
|
||
sample_steps: int = 32,
|
||
super_sampling: bool = False,
|
||
streaming_mode: Union[bool, int] = False,
|
||
overlap_length: int = 2,
|
||
min_chunk_length: int = 16,
|
||
):
|
||
req = {
|
||
"text": text,
|
||
"text_lang": text_lang.lower(),
|
||
"ref_audio_path": ref_audio_path,
|
||
"aux_ref_audio_paths": aux_ref_audio_paths,
|
||
"prompt_text": prompt_text,
|
||
"prompt_lang": prompt_lang.lower(),
|
||
"top_k": top_k,
|
||
"top_p": top_p,
|
||
"temperature": temperature,
|
||
"text_split_method": text_split_method,
|
||
"batch_size": int(batch_size),
|
||
"batch_threshold": float(batch_threshold),
|
||
"speed_factor": float(speed_factor),
|
||
"split_bucket": split_bucket,
|
||
"fragment_interval": fragment_interval,
|
||
"seed": seed,
|
||
"media_type": media_type,
|
||
"streaming_mode": streaming_mode,
|
||
"parallel_infer": parallel_infer,
|
||
"repetition_penalty": float(repetition_penalty),
|
||
"sample_steps": int(sample_steps),
|
||
"super_sampling": super_sampling,
|
||
"overlap_length": int(overlap_length),
|
||
"min_chunk_length": int(min_chunk_length),
|
||
}
|
||
return await tts_handle(req)
|
||
|
||
|
||
@APP.post("/tts")
|
||
async def tts_post_endpoint(request: TTS_Request):
|
||
req = request.dict()
|
||
return await tts_handle(req)
|
||
|
||
|
||
@APP.post("/tts_scheduler_debug")
|
||
async def tts_scheduler_debug_endpoint(request: Scheduler_Debug_Request):
|
||
return await tts_scheduler_debug_handle(request)
|
||
|
||
|
||
@APP.post("/tts_scheduler_submit")
|
||
async def tts_scheduler_submit_endpoint(request: Scheduler_Submit_Request):
|
||
return await tts_scheduler_submit_handle(request)
|
||
|
||
|
||
@APP.get("/tts_scheduler_state")
|
||
async def tts_scheduler_state_endpoint():
|
||
return JSONResponse(status_code=200, content={"message": "success", "worker_state": scheduler_debug_worker.get_state()})
|
||
|
||
|
||
@APP.get("/set_refer_audio")
|
||
async def set_refer_aduio(refer_audio_path: str = None):
|
||
try:
|
||
tts_pipeline.set_ref_audio(refer_audio_path)
|
||
except Exception as e:
|
||
return JSONResponse(status_code=400, content={"message": "set refer audio failed", "Exception": str(e)})
|
||
return JSONResponse(status_code=200, content={"message": "success"})
|
||
|
||
|
||
# @APP.post("/set_refer_audio")
|
||
# async def set_refer_aduio_post(audio_file: UploadFile = File(...)):
|
||
# try:
|
||
# # 检查文件类型,确保是音频文件
|
||
# if not audio_file.content_type.startswith("audio/"):
|
||
# return JSONResponse(status_code=400, content={"message": "file type is not supported"})
|
||
|
||
# os.makedirs("uploaded_audio", exist_ok=True)
|
||
# save_path = os.path.join("uploaded_audio", audio_file.filename)
|
||
# # 保存音频文件到服务器上的一个目录
|
||
# with open(save_path , "wb") as buffer:
|
||
# buffer.write(await audio_file.read())
|
||
|
||
# tts_pipeline.set_ref_audio(save_path)
|
||
# except Exception as e:
|
||
# return JSONResponse(status_code=400, content={"message": f"set refer audio failed", "Exception": str(e)})
|
||
# return JSONResponse(status_code=200, content={"message": "success"})
|
||
|
||
|
||
@APP.get("/set_gpt_weights")
|
||
async def set_gpt_weights(weights_path: str = None):
|
||
try:
|
||
if weights_path in ["", None]:
|
||
return JSONResponse(status_code=400, content={"message": "gpt weight path is required"})
|
||
tts_pipeline.init_t2s_weights(weights_path)
|
||
except Exception as e:
|
||
return JSONResponse(status_code=400, content={"message": "change gpt weight failed", "Exception": str(e)})
|
||
|
||
return JSONResponse(status_code=200, content={"message": "success"})
|
||
|
||
|
||
@APP.get("/set_sovits_weights")
|
||
async def set_sovits_weights(weights_path: str = None):
|
||
try:
|
||
if weights_path in ["", None]:
|
||
return JSONResponse(status_code=400, content={"message": "sovits weight path is required"})
|
||
tts_pipeline.init_vits_weights(weights_path)
|
||
except Exception as e:
|
||
return JSONResponse(status_code=400, content={"message": "change sovits weight failed", "Exception": str(e)})
|
||
return JSONResponse(status_code=200, content={"message": "success"})
|
||
|
||
|
||
if __name__ == "__main__":
|
||
try:
|
||
if host == "None": # 在调用时使用 -a None 参数,可以让api监听双栈
|
||
host = None
|
||
uvicorn.run(app=APP, host=host, port=port, workers=1)
|
||
except Exception:
|
||
traceback.print_exc()
|
||
os.kill(os.getpid(), signal.SIGTERM)
|
||
exit(0)
|