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PaddleSpeech/paddlespeech/server/utils/audio_handler.py

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20 KiB

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