def play_song(fname):
faad='faad'
to_exec=faad + ' -i '+ fname #first, read header so we can report song info
aa=subprocess.check_output(to_exec,stderr=subprocess.STDOUT,shell=True)
nn=get_key(aa,'\ntitle:') #we'll get name of song
writer=get_key(aa,'\nwriter:') #and composer
myinfo=get_key(aa,'LC AAC') #we'll get some info about the file
tags=myinfo.split() #this splits strings into their pieces
rate=tags[-2]
nchan=tags[-4]
length=tags[0]
print 'playing ' + nn + ' by ' + writer + ' for ' + length + ' ' + tags[1]
to_exec=faad+ ' -f2 -w -b4 ' + fname #this magic command will have faad read data into float arrays
ff=open('/dev/null','w')
dd=subprocess.check_output(to_exec,stderr=ff,shell=True) #this calls faad to do the heavy lifting
ff.close()
dat=numpy.fromstring(dd,dtype='float32') #convert string output from faad into numpy array
nchan=numpy.int(nchan)
dat=numpy.reshape(dat,[len(dat)/nchan,nchan])
fs=numpy.int(rate) #get the sample rate the speakers are expecting
sd.play(dat,fs,blocking=False) #play through speakers using sounddevice
return numpy.float(length) #return length of song.
def play(self, file_path):
if self.convert:
self.convert_mp3_to_wav(file_path_mp3=file_path)
data, fs = sf.read(file_path)
sd.play(data, fs)
sd.wait()
def generateSound(num, time, fs):
n = time*fs
x = np.linspace(0, time, n)
if num == '0':
freq1 = 1336
freq2 = 941
elif num == '1':
freq1 = 1209
freq2 = 697
elif num == '2':
freq1 = 1336
freq2 = 697
elif num == '3':
freq1 = 1477
freq2 = 697
elif num == '4':
freq1 = 1209
freq2 = 770
elif num == '5':
freq1 = 1336
freq2 = 770
elif num == '6':
freq1 = 1477
freq2 = 770
elif num == '7':
freq1 = 1209
freq2 = 852
elif num == '8':
freq1 = 1336
freq2 = 852
elif num == '9':
freq1 = 1477
freq2 = 852
else:
freq1 = 0
freq2 = 0
s1 = (np.sin(freq1*x*2*np.pi))
s2 = (np.sin(freq2*x*2*np.pi))
s = s1 + s2
sd.play(s, 44100)
sd.wait()
# Plotting graphs
plt.figure(1)
plt.subplot(211)
plt.plot(s1)
plt.xlim(0, 250)
plt.subplot(211)
plt.plot(s2)
plt.xlim(0, 250)
plt.subplot(212)
plt.plot(s)
plt.xlim(0, 250)
plt.show()
return
def play_song(song, sample=SineWave):
"""
Play a song.
A song is a list of tuples of (frequency, duration, active time ratio).
"""
def frames(f):
return int(f * SAMPLE_RATE)
compiled = []
duration = 0
for freq, dur, active in song:
thissample = sample if freq else Sample
compiled.append((sample(freq), frames(dur), frames(dur*active)))
duration += frames(dur)
out = numpy.empty((duration, 1))
count = 0
for sample, dur, active in compiled:
for i in xrange(dur):
if i < active:
out[count + i] = sample.amplitude(i) * 0.8
else:
out[count + i] = 0
count += dur
print 'playing!'
sd.play(out, blocking=True)
def ellis_bpm(fname, start_bpm, hpss=True, hop_length=512, tightness=100.0, plot=False, sound=False):
y, sr = librosa.load(fname, sr=None)
log.debug(u'Estimating tempo: {}'.format(TERM.cyan(fname)))
if hpss:
log.debug(TERM.magenta("Getting percussive elements"))
y_harmonic, y_percussive = librosa.effects.hpss(y)
chunks = np.array_split(y_percussive, PLOT_SPLIT)
log.debug(TERM.magenta("Estimating beats per minute"))
bpm, beat_frames = librosa.beat.beat_track(y=y_percussive, sr=sr, start_bpm=start_bpm, hop_length=hop_length, tightness=tightness)
else:
log.debug(TERM.magenta("Estimating beats per minute"))
bpm, beat_frames = librosa.beat.beat_track(y=y, sr=sr, start_bpm=start_bpm, hop_length=hop_length, tightness=tightness)
chunks = np.array_split(y, PLOT_SPLIT)
log.debug(u'Tempo: {:6.2f} bpm'.format(bpm))
if plot:
plt.figure(figsize=(16,10))
curr_frame = 0
for i in range(PLOT_SPLIT):
plt.subplot(PLOT_SPLIT * 100 + 11 + i)
plt.plot(curr_frame + np.arange(len(chunks[i])), chunks[i], 'g')
for b in beat_frames:
plt.axvline(x=b*hop_length, color='k')
plt.xlim([curr_frame, len(chunks[i]) + curr_frame])
curr_frame += len(chunks[i])
plt.show(block=False)
if sound:
beat_times = librosa.frames_to_time(beat_frames, sr=sr, hop_length=hop_length)
clicks = mir_eval.sonify.clicks(beat_times, sr, length=len(y))
sd.play(y + clicks, sr)
input('Press Return key to stop sound')
sd.stop()
return bpm
def play(self): # play from current position if not debug: while self.pv.getSample(self.currentPosition) is not None: # use nsound to play the sample # make sure player is waiting correct amount of time before plating next sample pass else: sampleNumber = self.debugGetSample(self.currentPosition) while sampleNumber is not None: print sampleNumber self.currentPosition += 1.0 / float(self.SAMPLES_PER_SECOND) sampleNumber = self.debugGetSample(self.currentPosition) print len(self.debugSamples) #sd.play(self.debugSamples, self.SAMPLES_PER_SECOND) #plt.plot(self.debugSamples) #plt.show() #f, df = 440.0, 44100.0 #t = r_[0.0:1.0:1/df] #data = np.cos(2 * pi * f * t) #play(data) #data = np.random.uniform(-1, 1, 44100) data = self.debugSamples scaled = np.int16(data/np.max(np.abs(data)) * 32767) plt.plot(scaled) plt.show() sd.play(scaled, self.SAMPLES_PER_SECOND)
def run(self):
while not self.exitFlag:
# qsize() is unreliable
if not self.q.empty() and self.q.qsize() > self.minblocks:
# Get all values from all chunks and put in a list
data = []
n = 0
for n in range(self.minblocks):
# get minblocks amount of chunks
tmp = self.q.get()
n = n + 1
m = 0
for m in range(len(tmp)):
# append all values in chunk to data list
data.append(tmp[m])
m = m + 1
self.i = self.i + 1
print("Musicplayer: Playing data with length: ", len(data))
sd.play(data, self.Fs)
time.sleep(self.sleeptime)
def play(url):
f = urlopen(url)
chunk = f.read(4096)
bchunk = np.array(list(chunk), dtype=np.int16)
sd.play(bchunk, 44100)
def play_audio(self):
if self.play_button.isChecked():
data = preprocess_audio(self.data, spikes_only=self.play_spike_audio)
# Track progress
if settings.show_audio_line:
self.audio_line = self.addLine(x=0)
self.audio_timer = pg.QtCore.QTimer()
self.audio_timer.setInterval(AUDIO_TIMER_UPDATE)
self.audio_timer.timeout.connect(self.update_audio_line)
sd.stop()
sd.play(data)
if settings.show_audio_line:
self.audio_timer.start()
# Notify all other audio components
self.guard = True
self.src.s.stop_audio.emit()
else:
sd.stop()
self.audio_timer.stop()
if settings.show_audio_line:
self.audio_line.hide()
self.audio_line = None
def preview_wave(message, wpm=WPM, frequency=FREQUENCY, framerate=FRAMERATE, amplitude=AMPLITUDE, word_ref=WORD):
"""
Listen (preview) wave
sounddevice is required http://python-sounddevice.readthedocs.org/
$ pip install sounddevice
"""
samp_nb = samples_nb(message=message, wpm=wpm, framerate=framerate, word_spaced=False)
import sounddevice as sd
lst_bin = _encode_binary(message)
amplitude = _limit_value(amplitude)
seconds_per_dot = _seconds_per_dot(word_ref) # 1.2
a = [calculate_wave(i, lst_bin, wpm, frequency, framerate, amplitude, seconds_per_dot)
for i in range(samp_nb)]
sd.play(a, framerate, blocking=True)
def offline():
print("Offline analysis")
duration = 5 # seconds
fs = 48000
sd.default.samplerate = fs
audio = sd.rec(duration * fs, channels=2, dtype="float64")
sd.wait()
print(" [{} channels, {:.2f} sec. {} Hz]".format(audio.shape[1], audio.shape[0] / fs, fs))
for pair in combinations(range(audio.shape[1]), 2):
print(" Ch. {} -> {}:".format(*pair), end='')
shift = _compute_shift(audio[:, pair[1]], audio[:, pair[0]])
print(" {:>5} sample(s) [{:>7,.3f} ms]".format(shift, shift / (fs / 1000)))
sd.play(audio)
sd.wait()
def get_hpss(filename):
# Get sound file (input sr = None, as we don't want to resample)
log.debug(u"Loading {}".format(TERM.green(filename)))
y, sr = librosa.load(filename, sr=None)
log.debug(TERM.magenta("Filtering high frequencies"))
y_filt = fir_lowpass(y, 5000, sr)
# Get the percussive and harmonic elements
log.debug(TERM.magenta("Splitting into harmonic and percussive"))
y_harmonic, y_percussive = librosa.effects.hpss(y_filt)
y_spss = {"harmonic":y_harmonic, "percussive":y_percussive}
# Play each in turn
while True:
component = get_response("Which component do you want to play?", ["harmonic", "percussive"], numbered=True)
if component is None:
return None
sd.play(y_spss[component], sr)
input('Press Return key to stop sound')
sd.stop()
def SoundCalOut(Rate, Freq, WaveDur):
""" Generate a sine wave from 1 to -1 """
Pulse = GenSineWave(Rate, Freq, 1, WaveDur)
SD.default.device = 'system'
SD.default.samplerate = Rate
SD.default.channels = 1
SD.default.blocksize = 0
SD.default.latency = 'low'
# Stream = SD.OutputStream()
#
# print('Playing... ', end='')
# with Stream:
# for PulseNo in range(SoundPulseNo):
# Stream.write(Pulse)
#
print('Playing...', end='')
SD.play(Pulse, blocking=True);
print('Done.')
return(None)
def sound(x, fs=44100, blocking = True):
"""
Play sound from x array at given sampling rate
"""
if x.ndim==1:
ch = 1
elif x.ndim==2:
if 1 in x.shape:
x = x.flatten()
ch = 1
elif x.shape[0]==2:
ch = 2
x = x.T
elif x.shape[1]==2:
ch = 2
elif x.ndim>2 or x.shape[1]>2:
print('Error: 1-D array for mono, or 2-D array where rows should be the number of channels, 1 (mono) or 2 (stereo)')
return
if audio_ok:
if x.ndim==2 and x.shape[0]==2:
x=x.T
play(x,fs)
if blocking:
pause(float(x.shape[0])/fs)
else:
warnings.warn('Cannot play sound, no sounddevice o audiolab module.')
import pygame
import wave
import sounddevice as sd
pygame.mixer.init()
file = wave.open("Alarm01.wav")
#s = pygame.mixer.Sound("Alarm01.wav")
#file.play()
sd.play(file)
random.seed()
wave_dict = []
# import Tkinter
# from Tkinter import *
# import tkMessageBox
#
# top = Tkinter.Tk()
# top.title("Fourier Synth")
# w = Canvas(top, width=500, height=500)
# w.grid(row=4, column=1,sticky=W)
# top.mainle
t=0
while True:
wave_dict = []
melody_speed = random.randint(12000,64000)
for i in range(0,random.randint(1,10)):
wave_dict.append({"freq":random.randint(1,2000),"melody_speed": ,"melody":[random.randint(50,200)/100.0],"amp":random.randint(0,100)/100.0,"phase":random.randint(1,314)/100.0,"LFO_freq": random.randint(1,1000)/100.0,"LFO_phase":random.randint(1,314)/100.0,"LFO_F":random.randint(1,100)/100.0,"LFO_A":random.randint(1,100)/100.0,"LFO_P":random.randint(1,314)/100.0})
input_array = []
for i in range(t,int(48000*20)+t):
amplitude = 0.0
for wave in wave_dict:
LFO_amp = math.sin((i*wave["LFO_freq"])/7000.0+wave["LFO_phase"])
amplitude += math.sin((i*(wave["freq"]+(LFO_amp*wave["LFO_F"])))/7000.0+(wave["phase"]+(LFO_amp*wave["LFO_P"])))*(wave["amp"]+(LFO_amp*wave["LFO_A"]))
input_array.append(amplitude)
array = np.array(input_array)
sd.wait()
sd.play(array,48000)
t+=480000*1
def play(self):
import sounddevice as sd
sd.default.channels = 2
sd.default.samplerate = self.sample_rate
sd.play(self.play_data, self.sample_rate)
sd.wait()
def reproducir(onda):
play(to_wav(onda))
sd.wait()
um = s697 + s1209
dois = s697 + s1336
tres = s697 + s1477
quatro = s770 + s1209
cinco = s770 + s1336
seis = s770 + s1477
sete = s852 + s1209
oito = s852 + s1336
nove = s852 + s1477
zero = s941 + s1336
while True:
if keyboard.is_pressed('1'):
sd.play(um)
sd.wait()
signal.plotFFT(um, fs)
print("1")
sd.stop()
if keyboard.is_pressed('2'):
sd.play(dois)
sd.wait()
signal.plotFFT(dois, fs)
print("2")
sd.stop()
if keyboard.is_pressed('3'):
sd.play(tres)
sd.wait()
def play_note_sounddevice(notePath, duration):
data, fs = sf.read(notePath, dtype='float32')
sd.play(data, fs)
time.sleep(duration)
return True
def play_stereo_tone(left_frequency, right_frequency, duration):
sd.play(gen_stereo(left_frequency, right_frequency, duration), fs)
sd.wait()
def play(self):
sounddevice.play(np.array(self.wav, dtype=np.int16),
samplerate=self.f_ech,
blocking=True)
def sound_once(self):
sounddevice.play(self.sound, self.sound_fs)
x, fs = librosa.load('../data/f1_005.wav', dtype='double', sr=None)
_f0, t = pw.dio(x, fs) # raw pitch extractor
f0 = pw.stonemask(x, _f0, t, fs) # pitch refinement
sp = pw.cheaptrick(x, f0, t, fs) # extract smoothed spectrogram
ap = pw.d4c(x, f0, t, fs) # extract aperiodicity
#y = pw.synthesize(f0*2**(3/12), sp, ap, fs)
#mix = y[0:len(x)-len(y)] + x
#sd.play(mix, fs)
chorus = np.zeros(f0.size)
phonetic = [16.352, 18.354, 20.602, 21.827, 24.5, 27.5, 30.868]
for k, freq_f0 in enumerate(f0):
if freq_f0 == 0:
continue
temp = freq_f0 / phonetic
log2temp = [math.log2(i) for i in temp]
diff = list(map(sub, log2temp, [round(i) for i in log2temp]))
diff = [abs(i) for i in diff]
idx = diff.index(min(diff))
if idx == 0 or idx == 3 or idx == 4:
chorus[k] = freq_f0 * 2**(4 / 12)
else:
chorus[k] = freq_f0 * 2**(3 / 12)
y = pw.synthesize(chorus, sp, ap, fs)
mix = y[0:len(x) - len(y)] * 0.6 + x
sd.play(mix, fs)
write('f1_005_chorus_up3.wav', fs, mix)
def playSound2(self, n):
sd.play(self.inverse[n], self.rate)
def playSound(self):
sd.play(self.ywav1[1], self.fs_rate)
figure(6)
stem(k,abs(fft(x))/M,'r')
grid(True);
ylim([-.01, 1.25]);
xlim([-21 ,21])
xlabel('FFT bins');
figure(7)
stem(k*fs/M,abs(fft(x))/M,'r')
grid(True);
ylim([-.01, 1.25]);
xlabel('Frequency (Hz)');
xlim([-55 ,55])
fs = 22050.0;
n = arange(4*fs);
t = n/fs;
x = cos(2000*pi*t*(t+2))
import sounddevice
#sounddevice.play(x,samplerate=22050)
fs = 8000.0;
n = arange(4*fs);
t = n/fs;
x = cos(2000*pi*t*(t+2))
sounddevice.play(x,samplerate=8000)
figure(8)
specgram(x, NFFT = 1000, Fs= 8000, noverlap=500)
xlabel('Time (seconds)')
ylabel('Frequency (Hz)')
#def playFinished(client, userdata, msg):
# print(msg.topic)
def stop():
global _RUNNING
_RUNNING = False
if __name__ == '__main__':
print('Starting hermes audio player')
# init GPIO if not done yet
initGpio()
# load empty wave file to prevent lag on playing first message
data, fs = soundfile.read('void.wav', dtype='float32')
sounddevice.play(data, fs, device=args.outputDevice)
status = sounddevice.wait()
if status:
print('Error during playback: ' + str(status))
lmsServer = LMSServer(lmsHost[0], lmsHost[1])
lmsPlayer = LMSPlayer(args.macAddress, lmsServer)
mqttClient = mqtt.Client()
mqttClient.on_connect = onConnect
mqttClient.message_callback_add("hermes/audioServer/{}/playBytes/#".format(args.siteId), playBytes)
# mqttClient.message_callback_add("hermes/audioServer/{}/playFinished/#".format(args.siteId), playFinished)
mqttClient.connect(mqttHost[0], int(mqttHost[1]))
mqttClient.loop_start()
try:
def noise(duration):
data = np.random.uniform(-1, 1, fs * duration)
sd.play(data, blocking=True)
def play_audio(signal, samplerate, seconds=5):
sd.play(signal, samplerate)
sd.sleep(int(seconds) * 1_000)
sd.stop()
import io
import math
from pprint import pprint
from urllib.request import urlopen
import soundfile as sf
url = "http://tinyurl.com/shepard-risset"
data, fs = sf.read(io.BytesIO(urlopen(url).read()))
import sounddevice as sd
sd.play(data, fs)
fs, duration = 44100, 1.5
data = sd.rec(duration * fs, samplerate=fs, channels=2)
sf.write('file.mp3', data, fs)
fs = 44100/20
data = []
msg = 'Hello World!'
time_to_play = 0.5
time_of_char = int(fs*time_to_play/len(msg))
for char in msg:
asci = ord(char)
data.extend([[math.cos(_*asci), math.cos(_*asci)] for _ in range(time_of_char)])
sd.play(data, fs, blocking=False)
parser.add_argument('-l',
'--list-devices',
action='store_true',
help='show list of audio devices and exit')
args, remaining = parser.parse_known_args()
if args.list_devices:
print(sd.query_devices())
parser.exit(0)
parser = argparse.ArgumentParser(
description=__doc__,
formatter_class=argparse.RawDescriptionHelpFormatter,
parents=[parser])
parser.add_argument('filename',
metavar='FILENAME',
help='audio file to be played back')
parser.add_argument('-d',
'--device',
type=int_or_str,
help='output device (numeric ID or substring)')
args = parser.parse_args(remaining)
try:
data, fs = sf.read(args.filename, dtype='float32')
sd.play(data, fs, device=args.device)
status = sd.wait()
except KeyboardInterrupt:
parser.exit('\nInterrupted by user')
except Exception as e:
parser.exit(type(e).__name__ + ': ' + str(e))
if status:
parser.exit('Error during playback: ' + str(status))
import sounddevice as sd
(fs1, x) = read('Traffic_stereo.wav', 'rb')
sd.play(x, fs1)
try:
from matplotlib.animation import FuncAnimation
import matplotlib.pyplot as plt
import numpy as np
import sounddevice as sd
import soundfile as sf
with sf.SoundFile(wav_file) as f:
data = f.buffer_read(BLOCK * 50, ctype='float')
#block = sf.blocks(wav_file, blocksize=2048, overlap=512)
#soundfile = sf.SoundFile(wav_file)
#data = soundfile.read(2048)
print(data)
#print(soundfile)
sd.play(data)
#print(block)
while True:
pass
#outstream = sd.OutputStream(
# device=0, channels=max(f.channels),
# samplerate=f.samplerate, callback=audio_callback)
print(len(data))
#if args.list_devices:
# print(sd.query_devices())
# parser.exit(0)
def sound(track):
sd.play(track, 44000)
def play(self, wav, sample_rate):
sd.stop()
sd.play(wav, sample_rate)
sd.wait()
def play(self, event):
sd.play(self.data.ifft, self.data.fs)
time.sleep(len(self.data.input_sound) / self.data.fs)
sd.stop()
import sounddevice as sd sd.default.latency = 'low' sd.default.device = 4 sd.default.channels = 2 fs = 44100 sd.default.samplerate = fs duration = 5 print(1) myrecording = sd.rec(int(duration * fs)) print(2) sd.wait() print(3) print(myrecording) print(4) sd.play(myrecording) input() print(5)
if 'posix' == os.name:
#dataset_home_dir_path = '/home/sangwook/my_dataset'
dataset_home_dir_path = '/home/HDD1/sangwook/my_dataset'
else:
dataset_home_dir_path = 'D:/dataset'
data_dir_path = dataset_home_dir_path + '/failure_analysis/defect/knock_sound/500-1500Hz'
#--------------------------------------------------------------------
import numpy as np
import sounddevice as sd
fs = 44100
data = np.random.uniform(-1, 1, fs)
sd.play(data, fs)
sd.default.dtype
sd.query_devices()
sd.default.samplerate = 44100
# A single value sets both input and output at the same time.
#sd.default.device = 'digital output'
#sd.default.device = 7
# Different values for input and output.
sd.default.channels = 5, 7
sd.play(data)
sd.default.reset()
def play(wave, duration, sample_rate):
sd.play(wave, sample_rate)
time.sleep(duration)
sd.stop()
def play(self, id):
data, fs = self.audiofile_cache[id]
sd.play(data, fs)
carry = (np.sin(15000*x*2*np.pi))
data = data * carry
# Plot fourier transform
plotFFT(data, 'Fourier after modulation')
return data
############################################################
#################### DECODING ####################
############################################################
# Multiply by carry
data = readFile()
n = len(data)
x = np.linspace(0,20, n)
carry = (np.sin(15000*x*2*np.pi))
data = data * carry
# Appliyng low pass filter
data = lfilter(taps, 1.0, data)
# Plot fourier transform
plotFFT(data, 'Fourier after demodulation')
# Play demodulated wave
sd.play(data, 44100)
sd.wait()
def play_two_waves(wave_A, wave_B):
stereo_wave = np.column_stack((wave_A, wave_B))
sd.play(stereo_wave, fs)
sd.wait()
sd.wait()
fay=[]
yaf=[]
fred=math.ceil(fs*duration)-10
for p in range(1,fred):
fay.append(recording[p][0])
for elem in fay:
yaf.append(abs(elem))
a=a+1
krow.append(np.mean(yaf))
for elem in krow:
hire.append(abs(np.mean(krow)-elem))
if np.mean(krow)<np.mean(yaf)-np.mean(hire):
fp = wave.open('int.wav')
nchan = fp.getnchannels()
N = fp.getnframes()
dstr = fp.readframes(N*nchan)
data = numpy.fromstring(dstr, numpy.int16)
data = numpy.reshape(data, (-1,nchan))
fs= 30010
duration=2
sd.play(data,fs)
sd.wait()
print('intruder!!!')
else:
print('Nothing')
except:
pass
def sin(frequency, duration):
samples = np.arange(fs * duration) / fs
waveform = np.sin(2 * np.pi * frequency * samples)
sd.play(waveform, blocking=True)
print("Created the mel spectrogram")
## Generating the waveform
print("Synthesizing the waveform:")
# Synthesizing the waveform is fairly straightforward. Remember that the longer the
# spectrogram, the more time-efficient the vocoder.
generated_wav = vocoder.infer_waveform(spec)
## Post-generation
# There's a bug with sounddevice that makes the audio cut one second earlier, so we
# pad it.
generated_wav = np.pad(generated_wav, (0, synthesizer.sample_rate),
mode="constant")
# Play the audio (non-blocking)
if not args.no_sound:
sd.stop()
sd.play(generated_wav, synthesizer.sample_rate)
# Save it on the disk
fpath = "demo_output_%02d.wav" % num_generated
print(generated_wav.dtype)
librosa.output.write_wav(fpath, generated_wav.astype(np.float32),
synthesizer.sample_rate)
num_generated += 1
print("\nSaved output as %s\n\n" % fpath)
except Exception as e:
print("Caught exception: %s" % repr(e))
print("Restarting\n")
#!/usr/bin/env python3
"""Load an audio file and play its contents.
PySoundFile (https://github.com/bastibe/PySoundFile/) has to be installed!
"""
import argparse
import logging
parser = argparse.ArgumentParser(description=__doc__)
parser.add_argument("filename", help="audio file to be played back")
parser.add_argument("-d", "--device", type=int, help="device ID")
args = parser.parse_args()
try:
import sounddevice as sd
import soundfile as sf
data, fs = sf.read(args.filename, dtype='float32')
sd.play(data, fs, device=args.device, blocking=True)
status = sd.get_status()
if status:
logging.warning(str(status))
except BaseException as e:
# This avoids printing the traceback, especially if Ctrl-C is used.
raise SystemExit(str(e))
# combine all waveforms
def combineWaveforms(waveforms):
waveform = waveforms[0]
n_waveforms = len(waveforms)
for i in range(1, n_waveforms):
waveform += waveforms[i]
waveform = waveform / n_waveforms
return waveform
waveform = combineWaveforms(waveforms)
attinuated_waveform = waveform * volume
sd.play(attinuated_waveform, sps, loop=True)
key_sensitivity = 0.03
while True:
for event in pygame.event.get():
if event.type == pygame.QUIT:
sys.exit()
if event.type == pygame.KEYDOWN:
if event.key == pygame.K_UP:
if volume + key_sensitivity > 1:
volume = 1
else:
volume += key_sensitivity
if event.key == pygame.K_DOWN:
if volume - key_sensitivity < 0:
def play_sample(sample,samplerate=22050,blocking=True):
sd.play(sample, samplerate=samplerate,blocking=blocking)
def play(data: np.array, rate: int=FS) -> None:
""" Plays a song to the default speaker. """
sd.play(data, rate)
sd.wait()
import sounddevice as sd
import numpy as np
fs=44100
duration = 5 # seconds
myrecording = sd.rec(duration * fs, samplerate=fs, channels=2,dtype='float64')
print("Recording Audio")
sd.wait()
print("Audio recording complete , Play Audio")
sd.play(myrecording, fs)
sd.wait()
print("Play Audio Complete")
"B": [1, 3],
"7": [2, 0],
"8": [2, 1],
"9": [2, 2],
"C": [2, 3],
"*": [3, 0],
"0": [3, 1],
"#": [3, 2],
"D": [3, 3]
}
valid_digits = "0123456789*#ABCD"
duration = 0.5
volume = 0.5
fs = 44100 # sampling rate
while True:
digits = list(input("Enter the digits: ").replace(" ", ""))
if (all(d in valid_digits for d in digits)):
for i in digits:
f1 = low_freq[digits_tones[i][0]]
f2 = high_freq[digits_tones[i][1]]
s1 = np.sin(2 * np.pi * np.arange(fs * duration) * f1 / fs)
s2 = np.sin(2 * np.pi * np.arange(fs * duration) * f2 / fs)
sd.play(s1 * volume + s2 * volume)
time.sleep(1)
else:
print("wrong input..")
import soundfile as sf # Use this package
import sounddevice as sd # and this one
kiss, samplerate = sf.read('Kiss.aiff') # load Kiss.aiff into kiss variable
sd.play(kiss, samplerate*.8) # play the music at 80% speed
sd.wait() # wait until music finishes before exiting
def PlayBack(Signal, Fs):
sd.play(Signal, Fs, blocking=True)
def playback(y, sr):
print("Playback samplerate:", sr)
sd.play(y, sr) # play audio
def play_speech(rec):
print('Playing')
sd.play(rec)
import numpy import sounddevice as sd fs = 44100 # sampling frequency T = 1.5 # seconds t = numpy.linspace(0, T, int(T*fs), endpoint=False) # time variable y = numpy.sin(2*numpy.pi*1000*t) # pure sine wave at 1000 Hz sd.play(y, fs) sd.wait()
plt.tight_layout(0)
plt.show()
# plot transformations
n = 1
for xx in m, m2:
for x in xx:
plt.subplot(2, len(s), n)
specshow(x, x_axis='linear')
n += 1
plt.tight_layout(0)
plt.show()
# play reconstructions
for x, x2 in zip(w, w2):
sd.play(x, sr, blocking=True)
sd.play(x2, sr, blocking=True)
from util_np import r2c
from util_io import save, save_boxcar
plot = lambda x: specshow(np.log(1e-8 + np.abs(x)))
wav, sr = load(path('LJ001-0008'), sr=None)
frame = librosa.stft(wav, 512, 256)
plot(frame[:-1])
plt.savefig("../docs/presentation/image/original.pdf",
bbox_inches='tight',
pad_inches=0)
# wav = librosa.istft(frame, 256)
#!/usr/bin/env python
# MagPy - Magstripe reader
import sounddevice as sd
fs = 44100
duration = 5 # seconds
sd.default.samplerate = fs
sd.default.channels = 2
#sd.default.device = 0
print('RECORDING')
rec = sd.rec(duration*fs,device=1)
sd.wait()
#for r in rec:
# print r
print('PLAYING')
sd.play(rec, fs, device=2)
sd.wait()
numSamples = fs * recLength
times = np.linspace(0, recLength, numSamplesRec)
rec = sd.rec(numSamplesRec, samplerate=fs, channels=2)
rec = rec[:, 0]
sd.wait()
#%% plot and play audio
plt.figure(3)
plt.plot(times, rec)
plt.xlabel("time")
plt.ylabel("signal value")
plt.title("og recording")
sd.play(rec, fs)
sd.wait()
#%% shorten audio
startSampleShort = 0
timesShort = times[startSampleShort:startSampleShort + numSamplesAnalyze]
signal = rec[startSampleShort:startSampleShort + numSamplesAnalyze]
#signal = sampleAudio
sd.play(signal, fs)
sd.wait()
plt.figure(3)
plt.plot(timesShort, signal)
plt.xlabel("time")
plt.ylabel("signal value")
