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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.file import audio_to_bytes, read_ref_text
- from tools.schema import ServeReferenceAudio, ServeTTSRequest
- def parse_args():
- parser = argparse.ArgumentParser(
- description="Send a WAV file and text to a server and receive synthesized audio.",
- formatter_class=argparse.RawTextHelpFormatter,
- )
- 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 to be used for the speech\n(Local: name of folder containing audios and files)",
- )
- parser.add_argument(
- "--reference_audio",
- "-ra",
- type=str,
- nargs="+",
- default=None,
- help="Path to the audio 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",
- action=argparse.BooleanOptionalAction,
- 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(
- "--latency",
- type=str,
- default="normal",
- choices=["normal", "balanced"],
- help="Used in api.fish.audio/v1/tts",
- )
- parser.add_argument(
- "--max_new_tokens",
- type=int,
- default=1024,
- help="Maximum new tokens to generate. \n0 means no limit.",
- )
- parser.add_argument(
- "--chunk_length", type=int, default=200, 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(
- "--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")
- parser.add_argument(
- "--use_memory_cache",
- type=str,
- default="off",
- choices=["on", "off"],
- help="Cache encoded references codes in memory.\n",
- )
- parser.add_argument(
- "--seed",
- type=int,
- default=None,
- help="`None` means randomized inference, otherwise deterministic.\n"
- "It can't be used for fixing a timbre.",
- )
- parser.add_argument(
- "--api_key",
- type=str,
- default="YOUR_API_KEY",
- help="API key for authentication",
- )
- 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 if ref_audio is not None else b"", text=ref_text
- )
- for ref_text, ref_audio in zip(ref_texts, byte_audios)
- ],
- "reference_id": idstr,
- "normalize": args.normalize,
- "format": args.format,
- "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,
- "streaming": args.streaming,
- "use_memory_cache": args.use_memory_cache,
- "seed": args.seed,
- }
- pydantic_data = ServeTTSRequest(**data)
- response = requests.post(
- args.url,
- data=ormsgpack.packb(pydantic_data, option=ormsgpack.OPT_SERIALIZE_PYDANTIC),
- stream=args.streaming,
- headers={
- "authorization": f"Bearer {args.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())
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