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ttsreal.py
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ttsreal.py
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import time
import numpy as np
import soundfile as sf
import resampy
import asyncio
import edge_tts
from typing import Iterator
import requests
import queue
from queue import Queue
from io import BytesIO
from threading import Thread, Event
from enum import Enum
class State(Enum):
RUNNING=0
PAUSE=1
class BaseTTS:
def __init__(self, opt, parent):
self.opt=opt
self.parent = parent
self.fps = opt.fps # 20 ms per frame
self.sample_rate = 16000
self.chunk = self.sample_rate // self.fps # 320 samples per chunk (20ms * 16000 / 1000)
self.input_stream = BytesIO()
self.msgqueue = Queue()
self.state = State.RUNNING
def pause_talk(self):
self.msgqueue.queue.clear()
self.state = State.PAUSE
def put_msg_txt(self,msg):
if len(msg)>0:
self.msgqueue.put(msg)
def render(self,quit_event):
process_thread = Thread(target=self.process_tts, args=(quit_event,))
process_thread.start()
def process_tts(self,quit_event):
while not quit_event.is_set():
try:
msg = self.msgqueue.get(block=True, timeout=1)
self.state=State.RUNNING
except queue.Empty:
continue
self.txt_to_audio(msg)
print('ttsreal thread stop')
def txt_to_audio(self,msg):
pass
###########################################################################################
class EdgeTTS(BaseTTS):
def txt_to_audio(self,msg):
voicename = "zh-CN-YunxiaNeural"
text = msg
t = time.time()
asyncio.new_event_loop().run_until_complete(self.__main(voicename,text))
print(f'-------edge tts time:{time.time()-t:.4f}s')
if self.input_stream.getbuffer().nbytes<=0: #edgetts err
print('edgetts err!!!!!')
return
self.input_stream.seek(0)
stream = self.__create_bytes_stream(self.input_stream)
streamlen = stream.shape[0]
idx=0
while streamlen >= self.chunk and self.state==State.RUNNING:
self.parent.put_audio_frame(stream[idx:idx+self.chunk])
streamlen -= self.chunk
idx += self.chunk
#if streamlen>0: #skip last frame(not 20ms)
# self.queue.put(stream[idx:])
self.input_stream.seek(0)
self.input_stream.truncate()
def __create_bytes_stream(self,byte_stream):
#byte_stream=BytesIO(buffer)
stream, sample_rate = sf.read(byte_stream) # [T*sample_rate,] float64
print(f'[INFO]tts audio stream {sample_rate}: {stream.shape}')
stream = stream.astype(np.float32)
if stream.ndim > 1:
print(f'[WARN] audio has {stream.shape[1]} channels, only use the first.')
stream = stream[:, 0]
if sample_rate != self.sample_rate and stream.shape[0]>0:
print(f'[WARN] audio sample rate is {sample_rate}, resampling into {self.sample_rate}.')
stream = resampy.resample(x=stream, sr_orig=sample_rate, sr_new=self.sample_rate)
return stream
async def __main(self,voicename: str, text: str):
try:
communicate = edge_tts.Communicate(text, voicename)
#with open(OUTPUT_FILE, "wb") as file:
first = True
async for chunk in communicate.stream():
if first:
first = False
if chunk["type"] == "audio" and self.state==State.RUNNING:
#self.push_audio(chunk["data"])
self.input_stream.write(chunk["data"])
#file.write(chunk["data"])
elif chunk["type"] == "WordBoundary":
pass
except Exception as e:
print(e)
###########################################################################################
class VoitsTTS(BaseTTS):
def txt_to_audio(self,msg):
self.stream_tts(
self.gpt_sovits(
msg,
self.opt.REF_FILE,
self.opt.REF_TEXT,
"zh", #en args.language,
self.opt.TTS_SERVER, #"http://127.0.0.1:5000", #args.server_url,
)
)
def gpt_sovits(self, text, reffile, reftext,language, server_url) -> Iterator[bytes]:
start = time.perf_counter()
req={
'text':text,
'text_lang':language,
'ref_audio_path':reffile,
'prompt_text':reftext,
'prompt_lang':language,
'media_type':'raw',
'streaming_mode':True
}
# req["text"] = text
# req["text_language"] = language
# req["character"] = character
# req["emotion"] = emotion
# #req["stream_chunk_size"] = stream_chunk_size # you can reduce it to get faster response, but degrade quality
# req["streaming_mode"] = True
try:
res = requests.post(
f"{server_url}/tts",
json=req,
stream=True,
)
end = time.perf_counter()
print(f"gpt_sovits Time to make POST: {end-start}s")
if res.status_code != 200:
print("Error:", res.text)
return
first = True
for chunk in res.iter_content(chunk_size=12800): # 1280 32K*20ms*2
if first:
end = time.perf_counter()
print(f"gpt_sovits Time to first chunk: {end-start}s")
first = False
if chunk and self.state==State.RUNNING:
yield chunk
#print("gpt_sovits response.elapsed:", res.elapsed)
except Exception as e:
print(e)
def stream_tts(self,audio_stream):
for chunk in audio_stream:
if chunk is not None and len(chunk)>0:
stream = np.frombuffer(chunk, dtype=np.int16).astype(np.float32) / 32767
stream = resampy.resample(x=stream, sr_orig=32000, sr_new=self.sample_rate)
#byte_stream=BytesIO(buffer)
#stream = self.__create_bytes_stream(byte_stream)
streamlen = stream.shape[0]
idx=0
while streamlen >= self.chunk:
self.parent.put_audio_frame(stream[idx:idx+self.chunk])
streamlen -= self.chunk
idx += self.chunk
###########################################################################################
class CosyVoiceTTS(BaseTTS):
def txt_to_audio(self,msg):
self.stream_tts(
self.cosy_voice(
msg,
self.opt.REF_FILE,
self.opt.REF_TEXT,
"zh", #en args.language,
self.opt.TTS_SERVER, #"http://127.0.0.1:5000", #args.server_url,
)
)
def cosy_voice(self, text, reffile, reftext,language, server_url) -> Iterator[bytes]:
start = time.perf_counter()
payload = {
'tts_text': text,
'prompt_text': reftext
}
try:
files = [('prompt_wav', ('prompt_wav', open(reffile, 'rb'), 'application/octet-stream'))]
res = requests.request("GET", f"{server_url}/inference_zero_shot", data=payload, files=files, stream=True)
end = time.perf_counter()
print(f"cosy_voice Time to make POST: {end-start}s")
if res.status_code != 200:
print("Error:", res.text)
return
first = True
for chunk in res.iter_content(chunk_size=8820): # 882 22.05K*20ms*2
if first:
end = time.perf_counter()
print(f"cosy_voice Time to first chunk: {end-start}s")
first = False
if chunk and self.state==State.RUNNING:
yield chunk
except Exception as e:
print(e)
def stream_tts(self,audio_stream):
for chunk in audio_stream:
if chunk is not None and len(chunk)>0:
stream = np.frombuffer(chunk, dtype=np.int16).astype(np.float32) / 32767
stream = resampy.resample(x=stream, sr_orig=22050, sr_new=self.sample_rate)
#byte_stream=BytesIO(buffer)
#stream = self.__create_bytes_stream(byte_stream)
streamlen = stream.shape[0]
idx=0
while streamlen >= self.chunk:
self.parent.put_audio_frame(stream[idx:idx+self.chunk])
streamlen -= self.chunk
idx += self.chunk
###########################################################################################
class XTTS(BaseTTS):
def __init__(self, opt, parent):
super().__init__(opt,parent)
self.speaker = self.get_speaker(opt.REF_FILE, opt.TTS_SERVER)
def txt_to_audio(self,msg):
self.stream_tts(
self.xtts(
msg,
self.speaker,
"zh-cn", #en args.language,
self.opt.TTS_SERVER, #"http://localhost:9000", #args.server_url,
"20" #args.stream_chunk_size
)
)
def get_speaker(self,ref_audio,server_url):
files = {"wav_file": ("reference.wav", open(ref_audio, "rb"))}
response = requests.post(f"{server_url}/clone_speaker", files=files)
return response.json()
def xtts(self,text, speaker, language, server_url, stream_chunk_size) -> Iterator[bytes]:
start = time.perf_counter()
speaker["text"] = text
speaker["language"] = language
speaker["stream_chunk_size"] = stream_chunk_size # you can reduce it to get faster response, but degrade quality
try:
res = requests.post(
f"{server_url}/tts_stream",
json=speaker,
stream=True,
)
end = time.perf_counter()
print(f"xtts Time to make POST: {end-start}s")
if res.status_code != 200:
print("Error:", res.text)
return
first = True
for chunk in res.iter_content(chunk_size=9600): #24K*20ms*2
if first:
end = time.perf_counter()
print(f"xtts Time to first chunk: {end-start}s")
first = False
if chunk:
yield chunk
except Exception as e:
print(e)
def stream_tts(self,audio_stream):
for chunk in audio_stream:
if chunk is not None and len(chunk)>0:
stream = np.frombuffer(chunk, dtype=np.int16).astype(np.float32) / 32767
stream = resampy.resample(x=stream, sr_orig=24000, sr_new=self.sample_rate)
#byte_stream=BytesIO(buffer)
#stream = self.__create_bytes_stream(byte_stream)
streamlen = stream.shape[0]
idx=0
while streamlen >= self.chunk:
self.parent.put_audio_frame(stream[idx:idx+self.chunk])
streamlen -= self.chunk
idx += self.chunk