def write_wav(data, filename, framerate, amplitude, length):
wavfile = wave.open(filename,'w')
nchannels = 1
sampwidth = 2
framerate = framerate
nframes = length
comptype = "NONE"
compname = "not compressed"
wavfile.setparams((nchannels,
sampwidth,
framerate,
nframes,
comptype,
compname))
frames = []
for s in data:
mul = int(s * amplitude)
print("I am here")
print(mul)
frames.append(struct.pack('h', mul))
frames = ''.join(frames)
wavfile.writeframes(frames)
wavfile.close()
print("%s written" %(filename))
def save_file(filename):
wavfile = wave.open(filename, "w")
# wav parameters
# Number of channels
nchannels = 1
# Sample width in bytes
sampwidth = 2
# Number of frames or samples
nframes = len(data)
comptype = "NONE"
compname = "not compressed"
wavfile.setparams(
(nchannels, sampwidth, samples, nframes, comptype, compname))
for sample in data:
"""
struct - Takes the data from my "data" list and packs it as binary data or else
the file is not readable by an audio player.
h - Means 16 bit number
32767.0 - The floating point numbers are not represented right and will not work
when writing to the wave file. Convert the floating point numbers to fixed point
to get full scale audio.
"""
wavfile.writeframes(struct.pack('h', int(sample * 32767.0)))
wavfile.close()
return
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 load(self, originalname):
# Load + convert audio
print("Loading sound file <{}>...".format(originalname))
name = originalname.split(".")[0]
ext = originalname.split(".")[-1].lower()
audiofile = originalname
if ext == MP3:
newfilename = name + ".wav"
old = AudioSegment.from_mp3(audiofile)
old.export(newfilename, format=WAV)
self.filename = newfilename
elif ext == PCM:
print "This is a PCM file!"
pcmfile = open(audiofile, 'rb')
pcmdata = pcmfile.read()
pcmfile.close()
wavfile = wave.open(name + '.wav', 'wb')
wavfile.setparams((1, 2, 44100, 16, 'NONE', 'NONE'))
wavfile.writeframes(pcmdata)
wavfile.close()
self.filename = name + ".wav"
else:
self.filename = audiofile
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
def write_wav(data, filename, framerate, amplitude):
wavfile = wave.open(filename,'w')
nchannels = 1
sampwidth = 2
framerate = framerate
nframes = len(data)
comptype = "NONE"
compname = "not compressed"
wavfile.setparams((nchannels,
sampwidth,
framerate,
nframes,
comptype,
compname))
frames = []
for s in data:
mul = int(s * amplitude)
frames.append(struct.pack('h', mul))
frames = ''.join(frames)
wavfile.writeframes(frames)
wavfile.close()
print("%s written" %(filename))
)
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
import numpy as np
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'
sys.exit(0)
