JLing
JLing是一個可以工作在Linux的自定義中文語音對話機器人
(csdn :https://blog.csdn.net/weixin_40490238)
(github: https://github.com/Kingzhoudk/JLing)
上了代碼就應該差不多了把,代碼中有部分註釋:
# -*- coding: utf-8 -*-
import webrtcvad
import collections
import sys
import signal
import pyaudio
from array import array
from struct import pack
import wave
import time
class VCAD:
def __init__(self):
# 對音頻數據進行定義
self.FORMAT = pyaudio.paInt16
self.CHANNELS = 1
self.RATE = 16000
self.CHUNK_DURATION_MS = 30 # supports 10, 20 and 30 (ms)
self.PADDING_DURATION_MS = 1500 # 1 sec jugement
self.CHUNK_SIZE = int(self.RATE * self.CHUNK_DURATION_MS / 1000) # chunk to read
self.CHUNK_BYTES = self.CHUNK_SIZE * 2 # 16bit = 2 bytes, PCM
self.NUM_PADDING_CHUNKS = int(self.PADDING_DURATION_MS / self.CHUNK_DURATION_MS)
# self.NUM_WINDOW_CHUNKS = int(240 / self.CHUNK_DURATION_MS)
self.NUM_WINDOW_CHUNKS = int(400 / self.CHUNK_DURATION_MS) # 400 ms/ 30ms ge
self.NUM_WINDOW_CHUNKS_END = self.NUM_WINDOW_CHUNKS * 2
self.START_OFFSET = int(self.NUM_WINDOW_CHUNKS * self.CHUNK_DURATION_MS * 0.5 * self.RATE)
def handle_int(self, sig, chunk):
global leave, got_a_sentence
leave = True
got_a_sentence = True
def record_to_file(self, path, data, sample_width):
"來自麥克風的記錄並將結果數據輸出到'path'"
# sample_width, data = record()
data = pack('<' + ('h' * len(data)), *data)
wf = wave.open(path, 'wb')
wf.setnchannels(1)
wf.setsampwidth(sample_width)
wf.setframerate(self.RATE)
wf.writeframes(data)
wf.close()
def normalize(self, snd_data):
"平均輸出量"
MAXIMUM = 32767 # 16384
times = float(MAXIMUM) / max(abs(i) for i in snd_data)
r = array('h')
for i in snd_data:
r.append(int(i * times))
return r
def start(self):
pa = pyaudio.PyAudio()
vad = webrtcvad.Vad(1)
signal.signal(signal.SIGINT, self.handle_int)
got_a_sentence = False
leave = False
stream = pa.open(format=self.FORMAT,
channels=self.CHANNELS,
rate=self.RATE,
input=True,
start=False,
# input_device_index=2,
frames_per_buffer=self.CHUNK_SIZE)
while not leave:
ring_buffer = collections.deque(maxlen=self.NUM_PADDING_CHUNKS)
triggered = False
voiced_frames = []
ring_buffer_flags = [0] * self.NUM_WINDOW_CHUNKS
ring_buffer_index = 0
ring_buffer_flags_end = [0] * self.NUM_WINDOW_CHUNKS_END
ring_buffer_index_end = 0
buffer_in = ''
# WangS
raw_data = array('h')
index = 0
start_point = 0
StartTime = time.time()
print("* recording: ")
stream.start_stream()
while not got_a_sentence and not leave:
chunk = stream.read(self.CHUNK_SIZE)
# 增加 WangS
raw_data.extend(array('h', chunk))
index += self.CHUNK_SIZE
TimeUse = time.time() - StartTime
active = vad.is_speech(chunk, self.RATE)
sys.stdout.write('1' if active else '_')
ring_buffer_flags[ring_buffer_index] = 1 if active else 0
ring_buffer_index += 1
ring_buffer_index %= self.NUM_WINDOW_CHUNKS
ring_buffer_flags_end[ring_buffer_index_end] = 1 if active else 0
ring_buffer_index_end += 1
ring_buffer_index_end %= self.NUM_WINDOW_CHUNKS_END
# 起始點檢測
if not triggered:
ring_buffer.append(chunk)
num_voiced = sum(ring_buffer_flags)
if num_voiced > 0.65 * self.NUM_WINDOW_CHUNKS:
sys.stdout.write(' Open ')
triggered = True
start_point = index - self.CHUNK_SIZE * 20 # 開始
# voiced_frames.extend(ring_buffer)
ring_buffer.clear()
# 結束點檢測
else:
# voiced_frames.append(chunk)
ring_buffer.append(chunk)
num_unvoiced = self.NUM_WINDOW_CHUNKS_END - sum(ring_buffer_flags_end)
if num_unvoiced > 0.65 * self.NUM_WINDOW_CHUNKS_END or TimeUse > 10:
sys.stdout.write(' Close ')
triggered = False
got_a_sentence = True
sys.stdout.flush()
sys.stdout.write('\n')
# data = b''.join(voiced_frames)
stream.stop_stream()
print("* done recording")
got_a_sentence = False
# write to file
raw_data.reverse()
for index in range(start_point):
raw_data.pop()
raw_data.reverse()
raw_data = self.normalize(raw_data)
self.record_to_file("input.wav", raw_data, 2)
leave = True
stream.close()
此句代碼中的0.65爲控制參數:(實際作用嘛,試試就曉得了)
if num_unvoiced > 0.65 * self.NUM_WINDOW_CHUNKS_END or TimeUse > 10: