def savewav(self, filename):
wf = wave.open(filename, 'wb')
wf.setnchannels(1)
wf.setsampwidth(2)
wf.setframerate(self.SAMPLING_RATE)
wf.writeframes(np.array(self.Voice_String).tostring())
wf.close()
def capture_audio(callback, final_callback):
import pyaudio
import wave as wv
from array import array
FORMAT = pyaudio.paInt16
RATE = 16000
CHUNK = 1024
COUNTCHUNK = (RATE + CHUNK - 1) / CHUNK
RECORD_SECONDS = 1
FILE_NAME = "/tmp/mic_rec.wav"
audio = pyaudio.PyAudio()
stream = audio.open(format=FORMAT,
rate=RATE,
channels=1,
input=True,
frames_per_buffer=CHUNK)
#starting recording
wave = []
callback_free = [True]
print("started")
last = time.time()
lastloudsoundtime = time.time() - 100000
while True:
data = stream.read(CHUNK)
data_chunk = array('h', data)
vol = max(data_chunk)
# print("Volume {}".format(vol))
wave.append(data)
if len(wave) > COUNTCHUNK:
wave.pop(0)
if (vol >= 2500):
lastloudsoundtime = time.time()
if (time.time() - last > 0.05
and time.time() - lastloudsoundtime < 0.8):
last = time.time()
if len(wave) == COUNTCHUNK:
#writing to file
wavfile = wv.open(FILE_NAME, 'wb')
wavfile.setnchannels(1)
wavfile.setsampwidth(audio.get_sample_size(FORMAT))
wavfile.setframerate(RATE)
wavfile.writeframes(b''.join(wave))
wavfile.close()
if callback_free[0]:
callback_free[0] = False
callback(callback_free, FILE_NAME, final_callback)
# print("Something is said")
else:
pass
def save_wave_file(filename, data):
# define of params
framerate = 8000
channels = 1
sampwidth = 2
'''save the date to the wav file'''
wf = wave.open(filename, 'wb')
wf.setnchannels(channels)
wf.setsampwidth(sampwidth)
wf.setframerate(framerate)
wf.writeframes("".join(data))
wf.close()
return 0
def stop(self, path='output.wav'):
sd.stop()
duration = int(math.ceil(time.time() - self._start_time))
data = self._recording[:duration * sd.default.samplerate]
# save audio
audio = pyaudio.PyAudio()
format = pyaudio.paInt16
wavfile = wave.open(path, 'wb')
wavfile.setnchannels(CHANNELS)
wavfile.setsampwidth(audio.get_sample_size(format))
wavfile.setframerate(SAMPLERATE)
wavfile.writeframes(data.tostring())
wavfile.close()
def run(self):
audio = PyAudio()
wavfile = wave.open(self.audiofile, 'ab')
wavfile.setnchannels(self.channels)
wavfile.setsampwidth(audio.get_sample_size(self.format))
wavfile.setframerate(self.rate)
wavstream = audio.open(format=self.format,
channels=self.channels,
rate=self.rate,
input=True,
frames_per_buffer=self.chunk)
while self.bRecord:
wavfile.writeframes(wavstream.read(self.chunk))
wavstream.stop_stream()
wavstream.close()
audio.terminate()
def run(self):
audio = PyAudio()
print("Sound device:", self.dev_idx)
device_info = audio.get_device_info_by_index(self.dev_idx)
self.channels = device_info["maxInputChannels"] if (
device_info["maxOutputChannels"] < device_info["maxInputChannels"]
) else device_info["maxOutputChannels"]
self.rate = int(device_info["defaultSampleRate"])
print(color.yellow(str(device_info)))
wavstream = audio.open(format=self.format,
channels=self.channels,
rate=self.rate,
input=True,
frames_per_buffer=self.chunk,
input_device_index=device_info["index"],
as_loopback=True)
# wavstream = audio.open(format=self.format,
# channels=self.channels,
# rate=self.rate,
# input=True,
# frames_per_buffer=self.chunk)
# 如果没有外放的话，loopback会没有数据，造成阻塞
# 循环读取输入流
while self.bRecord:
data = wavstream.read(self.chunk)
self._frames.append(data)
self._status = 1
wavstream.stop_stream()
wavstream.close()
# 保存到文件
print("Saveing .... ", self.audiofile)
with wave.open(self.audiofile, 'wb') as wavfile:
wavfile.setnchannels(self.channels)
wavfile.setsampwidth(audio.get_sample_size(self.format))
wavfile.setframerate(self.rate)
wavfile.writeframes(b''.join(self._frames))
audio.terminate()
self._status = 2
frames_per_buffer=CHUNK
)
frames = []
for i in range(0, int(RATE / CHUNK * RECORD_SECONDS)):
data = stream.read(CHUNK)
frames.append(data)
stream.stop_stream()
stream.close()
audio.terminate()
wavfile = wave.open(FILE_NAME, 'w')
wavfile.setnchannels(CHANNELS)
wavfile.setsampwidth(audio.get_sample_size(FORMAT))
wavfile.setframerate(RATE)
wavfile.writeframes(b''.join(frames))
wavfile.close()
wavfile = wave.open(FILE_NAME, 'r')
samplingFrequency, signalData = scipy.io.wavfile.read('ses.wav')
plot.subplot(211)
plot.plot(signalData)
plot.subplot(212)
plot.specgram(signalData, Fs=samplingFrequency)
plot.show()
#3.Kısım - Real time mikrofon ses verileriyle sinyal işleme
import pyaudio
import struct
hexString = ""
hexString += (hex(binVal)[2:-1])
decryptMess = binascii.unhexlify(hexString)
print("original text: " +decryptMess)
stream.stop_stream()
stream.close()
p.terminate()
wf = wave.open(WAVE_OUTPUT_FILENAME, 'wb')
wf.setnchannels(CHANNELS)
wf.setsampwidth(p.get_sample_size(FORMAT))
wf.setframerate(RATE)
wf.writeframes(b''.join(frames))
wf.close()
#-------------------------- ----------------------------
#plotting sound below
#------------------------------------------------------
spf = wave.open('input.wav','r')
signal = spf.readframes(-1)
signal = np.fromstring(signal, 'Int16')
fs = spf.getframerate()
#If Stereo
if spf.getnchannels() == 2:
print 'Just mono files'