Spaces:
Sleeping
Sleeping
from fastapi import File, Form, HTTPException, Body, UploadFile | |
from fastapi.responses import StreamingResponse | |
import io | |
from numpy import clip | |
import soundfile as sf | |
from pydantic import BaseModel, Field | |
from fastapi.responses import FileResponse | |
from modules.synthesize_audio import synthesize_audio | |
from modules.normalization import text_normalize | |
from modules import generate_audio as generate | |
from typing import List, Literal, Optional, Union | |
import pyrubberband as pyrb | |
from modules.api import utils as api_utils | |
from modules.api.Api import APIManager | |
from modules.speaker import speaker_mgr | |
from modules.data import styles_mgr | |
import numpy as np | |
class AudioSpeechRequest(BaseModel): | |
input: str # 需要合成的文本 | |
model: str = "chattts-4w" | |
voice: str = "female2" | |
response_format: Literal["mp3", "wav"] = "mp3" | |
speed: float = Field(1, ge=0.1, le=10, description="Speed of the audio") | |
seed: int = 42 | |
temperature: float = 0.3 | |
style: str = "" | |
# 是否开启batch合成,小于等于1表示不适用batch | |
# 开启batch合成会自动分割句子 | |
batch_size: int = Field(1, ge=1, le=20, description="Batch size") | |
spliter_threshold: float = Field( | |
100, ge=10, le=1024, description="Threshold for sentence spliter" | |
) | |
async def openai_speech_api( | |
request: AudioSpeechRequest = Body( | |
..., description="JSON body with model, input text, and voice" | |
) | |
): | |
model = request.model | |
input_text = request.input | |
voice = request.voice | |
style = request.style | |
response_format = request.response_format | |
batch_size = request.batch_size | |
spliter_threshold = request.spliter_threshold | |
speed = request.speed | |
speed = clip(speed, 0.1, 10) | |
if not input_text: | |
raise HTTPException(status_code=400, detail="Input text is required.") | |
if speaker_mgr.get_speaker(voice) is None: | |
raise HTTPException(status_code=400, detail="Invalid voice.") | |
try: | |
if style: | |
styles_mgr.find_item_by_name(style) | |
except: | |
raise HTTPException(status_code=400, detail="Invalid style.") | |
try: | |
# Normalize the text | |
text = text_normalize(input_text, is_end=True) | |
# Calculate speaker and style based on input voice | |
params = api_utils.calc_spk_style(spk=voice, style=style) | |
spk = params.get("spk", -1) | |
seed = params.get("seed", request.seed or 42) | |
temperature = params.get("temperature", request.temperature or 0.3) | |
prompt1 = params.get("prompt1", "") | |
prompt2 = params.get("prompt2", "") | |
prefix = params.get("prefix", "") | |
# Generate audio | |
sample_rate, audio_data = synthesize_audio( | |
text, | |
temperature=temperature, | |
top_P=0.7, | |
top_K=20, | |
spk=spk, | |
infer_seed=seed, | |
batch_size=batch_size, | |
spliter_threshold=spliter_threshold, | |
prompt1=prompt1, | |
prompt2=prompt2, | |
prefix=prefix, | |
) | |
if speed != 1: | |
audio_data = pyrb.time_stretch(audio_data, sample_rate, speed) | |
# Convert audio data to wav format | |
buffer = io.BytesIO() | |
sf.write(buffer, audio_data, sample_rate, format="wav") | |
buffer.seek(0) | |
if response_format == "mp3": | |
# Convert wav to mp3 | |
buffer = api_utils.wav_to_mp3(buffer) | |
return StreamingResponse(buffer, media_type="audio/mp3") | |
except Exception as e: | |
import logging | |
logging.exception(e) | |
if isinstance(e, HTTPException): | |
raise e | |
else: | |
raise HTTPException(status_code=500, detail=str(e)) | |
class TranscribeSegment(BaseModel): | |
id: int | |
seek: float | |
start: float | |
end: float | |
text: str | |
tokens: list[int] | |
temperature: float | |
avg_logprob: float | |
compression_ratio: float | |
no_speech_prob: float | |
class TranscriptionsVerboseResponse(BaseModel): | |
task: str | |
language: str | |
duration: float | |
text: str | |
segments: list[TranscribeSegment] | |
def setup(app: APIManager): | |
app.post( | |
"/v1/audio/speech", | |
response_class=FileResponse, | |
description=""" | |
openai api document: | |
[https://platform.openai.com/docs/guides/text-to-speech](https://platform.openai.com/docs/guides/text-to-speech) | |
以下属性为本系统自定义属性,不在openai文档中: | |
- batch_size: 是否开启batch合成,小于等于1表示不使用batch (不推荐) | |
- spliter_threshold: 开启batch合成时,句子分割的阈值 | |
- style: 风格 | |
> model 可填任意值 | |
""", | |
)(openai_speech_api) | |
async def transcribe( | |
file: UploadFile = File(...), | |
model: str = Form(...), | |
language: Optional[str] = Form(None), | |
prompt: Optional[str] = Form(None), | |
response_format: str = Form("json"), | |
temperature: float = Form(0), | |
timestamp_granularities: List[str] = Form(["segment"]), | |
): | |
# TODO: Implement transcribe | |
return api_utils.success_response("not implemented yet") | |