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import argparse | |
import base64 | |
import wave | |
import ormsgpack | |
import pyaudio | |
import requests | |
from pydub import AudioSegment | |
from pydub.playback import play | |
from tools.commons import ServeReferenceAudio, ServeTTSRequest | |
from tools.file import audio_to_bytes, read_ref_text | |
def parse_args(): | |
parser = argparse.ArgumentParser( | |
description="Send a WAV file and text to a server and receive synthesized audio." | |
) | |
parser.add_argument( | |
"--url", | |
"-u", | |
type=str, | |
default="http://127.0.0.1:8080/v1/tts", | |
help="URL of the server", | |
) | |
parser.add_argument( | |
"--text", "-t", type=str, required=True, help="Text to be synthesized" | |
) | |
parser.add_argument( | |
"--reference_id", | |
"-id", | |
type=str, | |
default=None, | |
help="ID of the reference model o be used for the speech", | |
) | |
parser.add_argument( | |
"--reference_audio", | |
"-ra", | |
type=str, | |
nargs="+", | |
default=None, | |
help="Path to the WAV file", | |
) | |
parser.add_argument( | |
"--reference_text", | |
"-rt", | |
type=str, | |
nargs="+", | |
default=None, | |
help="Reference text for voice synthesis", | |
) | |
parser.add_argument( | |
"--output", | |
"-o", | |
type=str, | |
default="generated_audio", | |
help="Output audio file name", | |
) | |
parser.add_argument( | |
"--play", | |
type=bool, | |
default=True, | |
help="Whether to play audio after receiving data", | |
) | |
parser.add_argument("--normalize", type=bool, default=True) | |
parser.add_argument( | |
"--format", type=str, choices=["wav", "mp3", "flac"], default="wav" | |
) | |
parser.add_argument("--mp3_bitrate", type=int, default=64) | |
parser.add_argument("--opus_bitrate", type=int, default=-1000) | |
parser.add_argument("--latency", type=str, default="normal", help="延迟选项") | |
parser.add_argument( | |
"--max_new_tokens", | |
type=int, | |
default=1024, | |
help="Maximum new tokens to generate", | |
) | |
parser.add_argument( | |
"--chunk_length", type=int, default=100, help="Chunk length for synthesis" | |
) | |
parser.add_argument( | |
"--top_p", type=float, default=0.7, help="Top-p sampling for synthesis" | |
) | |
parser.add_argument( | |
"--repetition_penalty", | |
type=float, | |
default=1.2, | |
help="Repetition penalty for synthesis", | |
) | |
parser.add_argument( | |
"--temperature", type=float, default=0.7, help="Temperature for sampling" | |
) | |
parser.add_argument( | |
"--speaker", type=str, default=None, help="Speaker ID for voice synthesis" | |
) | |
parser.add_argument("--emotion", type=str, default=None, help="Speaker's Emotion") | |
parser.add_argument( | |
"--streaming", type=bool, default=False, help="Enable streaming response" | |
) | |
parser.add_argument( | |
"--channels", type=int, default=1, help="Number of audio channels" | |
) | |
parser.add_argument("--rate", type=int, default=44100, help="Sample rate for audio") | |
return parser.parse_args() | |
if __name__ == "__main__": | |
args = parse_args() | |
idstr: str | None = args.reference_id | |
# priority: ref_id > [{text, audio},...] | |
if idstr is None: | |
ref_audios = args.reference_audio | |
ref_texts = args.reference_text | |
if ref_audios is None: | |
byte_audios = [] | |
else: | |
byte_audios = [audio_to_bytes(ref_audio) for ref_audio in ref_audios] | |
if ref_texts is None: | |
ref_texts = [] | |
else: | |
ref_texts = [read_ref_text(ref_text) for ref_text in ref_texts] | |
else: | |
byte_audios = [] | |
ref_texts = [] | |
pass # in api.py | |
data = { | |
"text": args.text, | |
"references": [ | |
ServeReferenceAudio(audio=ref_audio, text=ref_text) | |
for ref_text, ref_audio in zip(ref_texts, byte_audios) | |
], | |
"reference_id": idstr, | |
"normalize": args.normalize, | |
"format": args.format, | |
"mp3_bitrate": args.mp3_bitrate, | |
"opus_bitrate": args.opus_bitrate, | |
"max_new_tokens": args.max_new_tokens, | |
"chunk_length": args.chunk_length, | |
"top_p": args.top_p, | |
"repetition_penalty": args.repetition_penalty, | |
"temperature": args.temperature, | |
"speaker": args.speaker, | |
"emotion": args.emotion, | |
"streaming": args.streaming, | |
} | |
pydantic_data = ServeTTSRequest(**data) | |
response = requests.post( | |
args.url, | |
data=ormsgpack.packb(pydantic_data, option=ormsgpack.OPT_SERIALIZE_PYDANTIC), | |
stream=args.streaming, | |
headers={ | |
"authorization": "Bearer YOUR_API_KEY", | |
"content-type": "application/msgpack", | |
}, | |
) | |
if response.status_code == 200: | |
if args.streaming: | |
p = pyaudio.PyAudio() | |
audio_format = pyaudio.paInt16 # Assuming 16-bit PCM format | |
stream = p.open( | |
format=audio_format, channels=args.channels, rate=args.rate, output=True | |
) | |
wf = wave.open(f"{args.output}.wav", "wb") | |
wf.setnchannels(args.channels) | |
wf.setsampwidth(p.get_sample_size(audio_format)) | |
wf.setframerate(args.rate) | |
stream_stopped_flag = False | |
try: | |
for chunk in response.iter_content(chunk_size=1024): | |
if chunk: | |
stream.write(chunk) | |
wf.writeframesraw(chunk) | |
else: | |
if not stream_stopped_flag: | |
stream.stop_stream() | |
stream_stopped_flag = True | |
finally: | |
stream.close() | |
p.terminate() | |
wf.close() | |
else: | |
audio_content = response.content | |
audio_path = f"{args.output}.{args.format}" | |
with open(audio_path, "wb") as audio_file: | |
audio_file.write(audio_content) | |
audio = AudioSegment.from_file(audio_path, format=args.format) | |
if args.play: | |
play(audio) | |
print(f"Audio has been saved to '{audio_path}'.") | |
else: | |
print(f"Request failed with status code {response.status_code}") | |
print(response.json()) | |