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606 lines
20 KiB
606 lines
20 KiB
# Copyright (c) 2022 PaddlePaddle Authors. All Rights Reserved.
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#
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# Licensed under the Apache License, Version 2.0 (the "License");
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# you may not use this file except in compliance with the License.
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# You may obtain a copy of the License at
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#
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# http://www.apache.org/licenses/LICENSE-2.0
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#
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# Unless required by applicable law or agreed to in writing, software
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# distributed under the License is distributed on an "AS IS" BASIS,
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# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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# See the License for the specific language governing permissions and
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# limitations under the License.
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import base64
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import json
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import logging
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import threading
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import time
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import numpy as np
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import requests
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import soundfile
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import websockets
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from paddlespeech.cli.log import logger
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from paddlespeech.server.utils.audio_process import save_audio
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from paddlespeech.server.utils.util import wav2base64
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class TextHttpHandler:
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def __init__(self, server_ip="127.0.0.1", port=8090):
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"""Text http client request
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Args:
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server_ip (str, optional): the text server ip. Defaults to "127.0.0.1".
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port (int, optional): the text server port. Defaults to 8090.
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"""
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super().__init__()
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self.server_ip = server_ip
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self.port = port
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if server_ip is None or port is None:
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self.url = None
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else:
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self.url = 'http://' + self.server_ip + ":" + str(
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self.port) + '/paddlespeech/text'
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logger.info(f"endpoint: {self.url}")
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def run(self, text):
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"""Call the text server to process the specific text
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Args:
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text (str): the text to be processed
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Returns:
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str: punctuation text
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"""
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if self.server_ip is None or self.port is None:
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return text
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request = {
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"text": text,
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}
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try:
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res = requests.post(url=self.url, data=json.dumps(request))
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response_dict = res.json()
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punc_text = response_dict["result"]["punc_text"]
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except Exception as e:
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logger.error(f"Call punctuation {self.url} occurs error")
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logger.error(e)
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punc_text = text
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return punc_text
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class ASRWsAudioHandler:
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def __init__(self,
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url=None,
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port=None,
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endpoint="/paddlespeech/asr/streaming",
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punc_server_ip=None,
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punc_server_port=None):
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"""PaddleSpeech Online ASR Server Client audio handler
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Online asr server use the websocket protocal
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Args:
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url (str, optional): the server ip. Defaults to None.
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port (int, optional): the server port. Defaults to None.
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endpoint(str, optional): to compatiable with python server and c++ server.
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punc_server_ip(str, optional): the punctuation server ip. Defaults to None.
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punc_server_port(int, optional): the punctuation port. Defaults to None
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"""
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self.url = url
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self.port = port
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if url is None or port is None or endpoint is None:
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self.url = None
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else:
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self.url = "ws://" + self.url + ":" + str(self.port) + endpoint
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self.punc_server = TextHttpHandler(punc_server_ip, punc_server_port)
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logger.info(f"endpoint: {self.url}")
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def read_wave(self, wavfile_path: str):
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"""read the audio file from specific wavfile path
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Args:
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wavfile_path (str): the audio wavfile,
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we assume that audio sample rate matches the model
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Yields:
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numpy.array: the samall package audio pcm data
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"""
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samples, sample_rate = soundfile.read(wavfile_path, dtype='int16')
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x_len = len(samples)
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assert sample_rate == 16000
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chunk_size = int(85 * sample_rate / 1000) # 85ms, sample_rate = 16kHz
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if x_len % chunk_size != 0:
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padding_len_x = chunk_size - x_len % chunk_size
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else:
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padding_len_x = 0
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padding = np.zeros((padding_len_x), dtype=samples.dtype)
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padded_x = np.concatenate([samples, padding], axis=0)
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assert (x_len + padding_len_x) % chunk_size == 0
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num_chunk = (x_len + padding_len_x) / chunk_size
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num_chunk = int(num_chunk)
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for i in range(0, num_chunk):
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start = i * chunk_size
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end = start + chunk_size
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x_chunk = padded_x[start:end]
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yield x_chunk
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async def run(self, wavfile_path: str):
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"""Send a audio file to online server
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Args:
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wavfile_path (str): audio path
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Returns:
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str: the final asr result
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"""
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logging.debug("send a message to the server")
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if self.url is None:
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logger.error("No asr server, please input valid ip and port")
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return ""
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# 1. send websocket handshake protocal
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start_time = time.time()
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async with websockets.connect(self.url) as ws:
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# 2. server has already received handshake protocal
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# client start to send the command
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audio_info = json.dumps(
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{
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"name": "test.wav",
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"signal": "start",
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"nbest": 1
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},
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sort_keys=True,
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indent=4,
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separators=(',', ': '))
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await ws.send(audio_info)
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msg = await ws.recv()
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logger.info("client receive msg={}".format(msg))
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# 3. send chunk audio data to engine
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for chunk_data in self.read_wave(wavfile_path):
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await ws.send(chunk_data.tobytes())
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msg = await ws.recv()
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msg = json.loads(msg)
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if self.punc_server and len(msg["result"]) > 0:
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msg["result"] = self.punc_server.run(msg["result"])
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logger.info("client receive msg={}".format(msg))
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# 4. we must send finished signal to the server
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audio_info = json.dumps(
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{
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"name": "test.wav",
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"signal": "end",
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"nbest": 1
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},
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sort_keys=True,
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indent=4,
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separators=(',', ': '))
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await ws.send(audio_info)
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msg = await ws.recv()
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# 5. decode the bytes to str
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msg = json.loads(msg)
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if self.punc_server:
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msg["result"] = self.punc_server.run(msg["result"])
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# 6. logging the final result and comptute the statstics
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elapsed_time = time.time() - start_time
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audio_info = soundfile.info(wavfile_path)
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logger.info("client final receive msg={}".format(msg))
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logger.info(
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f"audio duration: {audio_info.duration}, elapsed time: {elapsed_time}, RTF={elapsed_time/audio_info.duration}"
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)
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result = msg
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return result
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class ASRHttpHandler:
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def __init__(self, server_ip=None, port=None, endpoint="/paddlespeech/asr"):
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"""The ASR client http request
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Args:
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server_ip (str, optional): the http asr server ip. Defaults to "127.0.0.1".
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port (int, optional): the http asr server port. Defaults to 8090.
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"""
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super().__init__()
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self.server_ip = server_ip
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self.port = port
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if server_ip is None or port is None:
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self.url = None
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else:
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self.url = 'http://' + self.server_ip + ":" + str(
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self.port) + endpoint
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logger.info(f"endpoint: {self.url}")
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def run(self, input, audio_format, sample_rate, lang):
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"""Call the http asr to process the audio
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Args:
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input (str): the audio file path
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audio_format (str): the audio format
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sample_rate (str): the audio sample rate
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lang (str): the audio language type
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Returns:
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str: the final asr result
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"""
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if self.url is None:
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logger.error(
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"No punctuation server, please input valid ip and port")
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return ""
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audio = wav2base64(input)
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data = {
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"audio": audio,
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"audio_format": audio_format,
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"sample_rate": sample_rate,
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"lang": lang,
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}
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res = requests.post(url=self.url, data=json.dumps(data))
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return res.json()
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class TTSWsHandler:
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def __init__(self, server="127.0.0.1", port=8092, play: bool=False):
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"""PaddleSpeech Online TTS Server Client audio handler
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Online tts server use the websocket protocal
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Args:
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server (str, optional): the server ip. Defaults to "127.0.0.1".
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port (int, optional): the server port. Defaults to 8092.
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play (bool, optional): whether to play audio. Defaults False
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"""
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self.server = server
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self.port = port
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self.url = "ws://" + self.server + ":" + str(
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self.port) + "/paddlespeech/tts/streaming"
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self.play = play
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# get model sample rate
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self.url_get_sr = "http://" + str(self.server) + ":" + str(
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self.port) + "/paddlespeech/tts/streaming/samplerate"
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self.sample_rate = requests.get(self.url_get_sr).json()["sample_rate"]
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if self.play:
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import pyaudio
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self.buffer = b''
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self.p = pyaudio.PyAudio()
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self.stream = self.p.open(
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format=self.p.get_format_from_width(2),
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channels=1,
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rate=self.sample_rate,
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output=True)
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self.mutex = threading.Lock()
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self.start_play = True
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self.t = threading.Thread(target=self.play_audio)
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self.max_fail = 50
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logger.info(f"endpoint: {self.url}")
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def play_audio(self):
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while True:
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if not self.buffer:
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self.max_fail -= 1
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time.sleep(0.05)
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if self.max_fail < 0:
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break
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self.mutex.acquire()
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self.stream.write(self.buffer)
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self.buffer = b''
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self.mutex.release()
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async def run(self, text: str, spk_id=0, output: str=None):
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"""Send a text to online server
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Args:
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text (str): sentence to be synthesized
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spk_id (int, optional): speaker id. Defaults to 0.
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output (str, optional): client save audio path. Defaults to None.
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"""
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all_bytes = b''
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receive_time_list = []
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chunk_duration_list = []
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# 1. Send websocket handshake request
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async with websockets.connect(self.url) as ws:
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# 2. Server has already received handshake response, send start request
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start_request = json.dumps({"task": "tts", "signal": "start"})
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await ws.send(start_request)
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msg = await ws.recv()
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logger.info(f"client receive msg={msg}")
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msg = json.loads(msg)
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session = msg["session"]
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# 3. send speech synthesis request
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#text_base64 = str(base64.b64encode((text).encode('utf-8')), "UTF8")
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params = {
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"text": text,
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"spk_id": spk_id,
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}
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request = json.dumps(params)
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st = time.time()
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await ws.send(request)
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logging.debug("send a message to the server")
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# 4. Process the received response
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message = await ws.recv()
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first_response = time.time() - st
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message = json.loads(message)
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status = message["status"]
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while True:
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# When throw an exception
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if status == -1:
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# send end request
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end_request = json.dumps({
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"task": "tts",
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"signal": "end",
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"session": session
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})
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await ws.send(end_request)
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break
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# Rerutn last packet normally, no audio information
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elif status == 2:
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final_response = time.time() - st
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duration = len(all_bytes) / 2.0 / self.sample_rate
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if output is not None:
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save_audio_success = save_audio(all_bytes, output,
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self.sample_rate)
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else:
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save_audio_success = False
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# send end request
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end_request = json.dumps({
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"task": "tts",
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"signal": "end",
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"session": session
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})
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await ws.send(end_request)
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break
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# Return the audio stream normally
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elif status == 1:
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receive_time_list.append(time.time())
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audio = message["audio"]
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audio = base64.b64decode(audio) # bytes
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chunk_duration_list.append(
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len(audio) / 2.0 / self.sample_rate)
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all_bytes += audio
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if self.play:
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self.mutex.acquire()
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self.buffer += audio
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self.mutex.release()
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if self.start_play:
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self.t.start()
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self.start_play = False
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message = await ws.recv()
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message = json.loads(message)
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status = message["status"]
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else:
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logger.error("infer error, return status is invalid.")
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if self.play:
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self.t.join()
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self.stream.stop_stream()
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self.stream.close()
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self.p.terminate()
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return first_response, final_response, duration, save_audio_success, receive_time_list, chunk_duration_list
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class TTSHttpHandler:
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def __init__(self, server="127.0.0.1", port=8092, play: bool=False):
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"""PaddleSpeech Online TTS Server Client audio handler
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Online tts server use the websocket protocal
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Args:
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server (str, optional): the server ip. Defaults to "127.0.0.1".
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port (int, optional): the server port. Defaults to 8092.
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play (bool, optional): whether to play audio. Defaults False
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"""
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self.server = server
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self.port = port
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self.url = "http://" + str(self.server) + ":" + str(
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self.port) + "/paddlespeech/tts/streaming"
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self.play = play
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# get model sample rate
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self.url_get_sr = "http://" + str(self.server) + ":" + str(
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self.port) + "/paddlespeech/tts/streaming/samplerate"
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self.sample_rate = requests.get(self.url_get_sr).json()["sample_rate"]
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if self.play:
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import pyaudio
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self.buffer = b''
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self.p = pyaudio.PyAudio()
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self.start_play = True
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self.max_fail = 50
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self.stream = self.p.open(
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format=self.p.get_format_from_width(2),
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channels=1,
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rate=self.sample_rate,
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output=True)
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self.mutex = threading.Lock()
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self.t = threading.Thread(target=self.play_audio)
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logger.info(f"endpoint: {self.url}")
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def play_audio(self):
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while True:
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if not self.buffer:
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self.max_fail -= 1
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time.sleep(0.05)
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if self.max_fail < 0:
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break
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self.mutex.acquire()
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self.stream.write(self.buffer)
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self.buffer = b''
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self.mutex.release()
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def run(self, text: str, spk_id=0, output: str=None):
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"""Send a text to tts online server
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Args:
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text (str): sentence to be synthesized.
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spk_id (int, optional): speaker id. Defaults to 0.
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output (str, optional): client save audio path. Defaults to None.
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"""
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# 1. Create request
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params = {
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"text": text,
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"spk_id": spk_id,
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}
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all_bytes = b''
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first_flag = 1
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receive_time_list = []
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chunk_duration_list = []
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# 2. Send request
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st = time.time()
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html = requests.post(self.url, json.dumps(params), stream=True)
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# 3. Process the received response
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for chunk in html.iter_content(chunk_size=None):
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receive_time_list.append(time.time())
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audio = base64.b64decode(chunk) # bytes
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if first_flag:
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first_response = time.time() - st
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first_flag = 0
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if self.play:
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self.mutex.acquire()
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self.buffer += audio
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self.mutex.release()
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if self.start_play:
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self.t.start()
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self.start_play = False
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all_bytes += audio
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chunk_duration_list.append(len(audio) / 2.0 / self.sample_rate)
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final_response = time.time() - st
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duration = len(all_bytes) / 2.0 / self.sample_rate
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html.close() # when stream=True
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if output is not None:
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save_audio_success = save_audio(all_bytes, output, self.sample_rate)
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else:
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save_audio_success = False
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if self.play:
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self.t.join()
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self.stream.stop_stream()
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self.stream.close()
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self.p.terminate()
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return first_response, final_response, duration, save_audio_success, receive_time_list, chunk_duration_list
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class VectorHttpHandler:
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def __init__(self, server_ip=None, port=None):
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"""The Vector client http request
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Args:
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server_ip (str, optional): the http vector server ip. Defaults to "127.0.0.1".
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port (int, optional): the http vector server port. Defaults to 8090.
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"""
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super().__init__()
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self.server_ip = server_ip
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self.port = port
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if server_ip is None or port is None:
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self.url = None
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else:
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self.url = 'http://' + self.server_ip + ":" + str(
|
|
self.port) + '/paddlespeech/vector'
|
|
logger.info(f"endpoint: {self.url}")
|
|
|
|
def run(self, input, audio_format, sample_rate, task="spk"):
|
|
"""Call the http asr to process the audio
|
|
|
|
Args:
|
|
input (str): the audio file path
|
|
audio_format (str): the audio format
|
|
sample_rate (str): the audio sample rate
|
|
|
|
Returns:
|
|
list: the audio vector
|
|
"""
|
|
if self.url is None:
|
|
logger.error("No vector server, please input valid ip and port")
|
|
return ""
|
|
|
|
audio = wav2base64(input)
|
|
data = {
|
|
"audio": audio,
|
|
"task": task,
|
|
"audio_format": audio_format,
|
|
"sample_rate": sample_rate,
|
|
}
|
|
|
|
res = requests.post(url=self.url, data=json.dumps(data))
|
|
|
|
return res
|
|
|
|
|
|
class VectorScoreHttpHandler:
|
|
def __init__(self, server_ip=None, port=None):
|
|
"""The Vector score client http request
|
|
|
|
Args:
|
|
server_ip (str, optional): the http vector server ip. Defaults to "127.0.0.1".
|
|
port (int, optional): the http vector server port. Defaults to 8090.
|
|
"""
|
|
super().__init__()
|
|
self.server_ip = server_ip
|
|
self.port = port
|
|
if server_ip is None or port is None:
|
|
self.url = None
|
|
else:
|
|
self.url = 'http://' + self.server_ip + ":" + str(
|
|
self.port) + '/paddlespeech/vector/score'
|
|
logger.info(f"endpoint: {self.url}")
|
|
|
|
def run(self, enroll_audio, test_audio, audio_format, sample_rate):
|
|
"""Call the http asr to process the audio
|
|
|
|
Args:
|
|
input (str): the audio file path
|
|
audio_format (str): the audio format
|
|
sample_rate (str): the audio sample rate
|
|
|
|
Returns:
|
|
list: the audio vector
|
|
"""
|
|
if self.url is None:
|
|
logger.error("No vector server, please input valid ip and port")
|
|
return ""
|
|
|
|
enroll_audio = wav2base64(enroll_audio)
|
|
test_audio = wav2base64(test_audio)
|
|
data = {
|
|
"enroll_audio": enroll_audio,
|
|
"test_audio": test_audio,
|
|
"task": "score",
|
|
"audio_format": audio_format,
|
|
"sample_rate": sample_rate,
|
|
}
|
|
|
|
res = requests.post(url=self.url, data=json.dumps(data))
|
|
|
|
return res
|