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())