I'm currently creating a small Program with Python and pyo that should use the microphone input as a source and then add several effects and filters provided by pyo. I couldn't find anything in the docs, is there a way to use the microphone input as a source, are there any alternatives to pyo?
Here's the basic example I have so far:
from pyo import *
s = Server().boot()
s.start()
s.amp = 0.1
# use microphone input here
sf = Sig(1).out()
# Passes the sine wave through an harmonizer.
h1 = Harmonizer(sf).out()
s.gui(locals())
I know there's a function to set the input device, like
s.setInputDevice(5), but I can not figure out how to actually use it.
Thanks for the help!
It looks like you aren't actually creating an input stream. Something like this (no gui) will output your microphone input:
from pyo import *
s = Server().boot()
s.start()
miccheck = Input().play().out()
Or, to modify the harmonizer default example:
from pyo import *
s = Server().boot()
mic = Input().play().out()
env = WinTable(8)
wsize = .1
trans = -7
ratio = pow(2., trans/12.)
rate = -(ratio-1) / wsize
ind = Phasor(freq=rate, phase=[0,0.5])
win = Pointer(table=env, index=ind, mul=.7)
snd = Delay(mic, delay=ind*wsize, mul=win).mix(1).out(1)
s.gui(locals())
Related
I am using the soundcard library to record my microphone input, it records in a NumPy array and I want to grab that audio and save it as an mp3 file.
Code:
import soundcard as sc
import numpy
import threading
speakers = sc.all_speakers() # Gets a list of the systems speakers
default_speaker = sc.default_speaker() # Gets the default speaker
mics = sc.all_microphones() # Gets a list of all the microphones
default_mic = sc.get_microphone('Headset Microphone (Arctis 7 Chat)') # Gets the default microphone
# Records the default microphone
def record_mic():
print('Recording...')
with default_mic.recorder(samplerate=48000) as mic, default_speaker.player(samplerate=48000) as sp:
for _ in range(1000000000000):
data = mic.record(numframes=None) # 'None' creates zero latency
sp.play(data)
# Save the mp3 file here
recordThread = threading.Thread(target=record_mic)
recordThread.start()
With Scipy (to wav file)
You can easily convert to wav and then separately convert wav to mp3. More details here.
from scipy.io.wavfile import write
samplerate = 44100; fs = 100
t = np.linspace(0., 1., samplerate)
amplitude = np.iinfo(np.int16).max
data = amplitude * np.sin(2. * np.pi * fs * t)
write("example.wav", samplerate, data.astype(np.int16))
With pydub (to mp3)
Try this function from this excellent thread -
import pydub
import numpy as np
def write(f, sr, x, normalized=False):
"""numpy array to MP3"""
channels = 2 if (x.ndim == 2 and x.shape[1] == 2) else 1
if normalized: # normalized array - each item should be a float in [-1, 1)
y = np.int16(x * 2 ** 15)
else:
y = np.int16(x)
song = pydub.AudioSegment(y.tobytes(), frame_rate=sr, sample_width=2, channels=channels)
song.export(f, format="mp3", bitrate="320k")
#[[-225 707]
# [-234 782]
# [-205 755]
# ...,
# [ 303 89]
# [ 337 69]
# [ 274 89]]
write('out2.mp3', sr, x)
Note: Output MP3 will of cause be 16-bit, because MP3s are always 16 bit. However, you can set sample_width=3 as suggested by #Arty for 24-bit input.
As of now the accepted answer produces extremely distorted sound atleast in my case so here is the improved version :
#librosa read
y,sr=librosa.load(dir+file,sr=None)
y=librosa.util.normalize(y)
#pydub read
sound=AudioSegment.from_file(dir+file)
channel_sounds = sound.split_to_mono()
samples = [s.get_array_of_samples() for s in channel_sounds]
fp_arr = np.array(samples).T.astype(np.float32)
fp_arr /= np.iinfo(samples[0].typecode).max
fp_arr=np.array([x[0] for x in fp_arr])
#i normalize the pydub waveform with librosa for comparison purposes
fp_arr=librosa.util.normalize(fp_arr)
so you read the audiofile from any library and you have a waveform then you can export it to any pydub supported codec with this code below, i also used librosa read waveform and it works perfect.
wav_io = io.BytesIO()
scipy.io.wavfile.write(wav_io, sample_rate, waveform)
wav_io.seek(0)
sound = AudioSegment.from_wav(wav_io)
with open("file_exported_by_pydub.mp3",'wb') as af:
sound.export(
af,
format='mp3',
codec='mp3',
bitrate='160000',
)
I am new learner of audio editing libs - Pydub. I want to change some audio files' playback speed using Pydub(say .wav/mp3 format files), but I don't know how to make it. The only module I saw that could possibly deal with this problem is speedup module in effect.py. However, there is no explanation about how I am supposed to call it.
Could anyone kindly explain how to do this task in Pydub? Many thanks!
(A related question: Pydub - How to change frame rate without changing playback speed, but what I want to do is to change the playback speed without changing the audio quality.)
This can be done using pyrubberband package which requires rubberband library that can stretch audio while keeping the pitch and high quality. I was able to install the library on MacOS using brew, and same on Ubuntu with apt install. For extreme stretching, look into PaulStretch
brew install rubberband
This works simply with librosa package
import librosa
import pyrubberband
import soundfile as sf
y, sr = librosa.load(filepath, sr=None)
y_stretched = pyrubberband.time_stretch(y, sr, 1.5)
sf.write(analyzed_filepath, y_stretched, sr, format='wav')
To make pyrubberband work directly with AudioSegment from pydub without librosa I fiddled this function:
def change_audioseg_tempo(audiosegment, tempo, new_tempo):
y = np.array(audiosegment.get_array_of_samples())
if audiosegment.channels == 2:
y = y.reshape((-1, 2))
sample_rate = audiosegment.frame_rate
tempo_ratio = new_tempo / tempo
print(tempo_ratio)
y_fast = pyrb.time_stretch(y, sample_rate, tempo_ratio)
channels = 2 if (y_fast.ndim == 2 and y_fast.shape[1] == 2) else 1
y = np.int16(y_fast * 2 ** 15)
new_seg = pydub.AudioSegment(y.tobytes(), frame_rate=sample_rate, sample_width=2, channels=channels)
return new_seg
from pydub import AudioSegment
from pydub import effects
root = r'audio.wav'
velocidad_X = 1.5 # No puede estar por debajo de 1.0
sound = AudioSegment.from_file(root)
so = sound.speedup(velocidad_X, 150, 25)
so.export(root[:-4] + '_Out.mp3', format = 'mp3')
I know it's late but I wrote a program to convert mp3 to different playback speed.
First, Convert the .MP3 -> .Wav because PYRubberBand supports only .wav format. Then streach the time and pitch at the same time to avoid chipmunk effect.
import wave
import sys
from pydub import AudioSegment
#sound = AudioSegment.from_file("deviprasadgharpehai.mp3")
sound = AudioSegment.from_mp3(sys.argv[1])
sound.export("file.wav", format="wav")
print(sys.argv[1])
import soundfile as sf
import pyrubberband as pyrb
y, sr = sf.read("file.wav")
# Play back at extra low speed
y_stretch = pyrb.time_stretch(y, sr, 0.5)
# Play back extra low tones
y_shift = pyrb.pitch_shift(y, sr, 0.5)
sf.write("analyzed_filepathX5.wav", y_stretch, sr, format='wav')
sound = AudioSegment.from_wav("analyzed_filepathX5.wav")
sound.export("analyzed_filepathX5.mp3", format="mp3")
# Play back at low speed
y_stretch = pyrb.time_stretch(y, sr, 0.75)
# Play back at low tones
y_shift = pyrb.pitch_shift(y, sr, 0.75)
sf.write("analyzed_filepathX75.wav", y_stretch, sr, format='wav')
sound = AudioSegment.from_wav("analyzed_filepathX75.wav")
sound.export("analyzed_filepathX75.mp3", format="mp3")
# Play back at 1.5X speed
y_stretch = pyrb.time_stretch(y, sr, 1.5)
# Play back two 1.5x tones
y_shift = pyrb.pitch_shift(y, sr, 1.5)
sf.write("analyzed_filepathX105.wav", y_stretch, sr, format='wav')
sound = AudioSegment.from_wav("analyzed_filepathX105.wav")
sound.export("analyzed_filepathX105.mp3", format="mp3")
# Play back at same speed
y_stretch = pyrb.time_stretch(y, sr, 1)
# Play back two smae-tones
y_shift = pyrb.pitch_shift(y, sr, 1)
sf.write("analyzed_filepathXnormal.wav", y_stretch, sr, format='wav')
sound = AudioSegment.from_wav("analyzed_filepathXnormal.wav")
sound.export("analyzed_filepathXnormal.mp3", format="mp3")
**Make Sure to install **
Wave, AudioSegment, FFmpeg, PYRubberBand, Soundfile
To use this Run,
python3 filename.py mp3filename.mp3
To change the speed of the audio without changing the pitch (or creating chipmunk effect). You can use below code.
from pydub import AudioSegment
from pydub.effects import speedup
audio = AudioSegment.from_mp3(song.mp3)
new_file = speedup(audio,1.5,150)
new_file.export("file.mp3", format="mp3")
I want to calculate the single channel data (in order to calculate the audio cross correlation between the channel 1 and channel 4) of this code:
import time
import numpy as np
import pyaudio
import scipy
from scipy import signal, fftpack
pyaud = pyaudio.PyAudio()
#open the stream
stream = pyaud.open(
format = pyaudio.paInt16,
channels = 4,
rate = 16000,
input_device_index = 4,
output = False,
input = True,
frames_per_buffer=2048,)
while True:
rawsamps = stream.read(2048)
samps = np.fromstring(rawsamps, dtype=np.int16)
frames_per_buffer_length = len(samps) / 4 #(channels)
assert frames_per_buffer_length == int(frames_per_buffer_length)
samps = np.reshape(samps, (frames_per_buffer_length, 4)) #4 channels
Assuming that the raw data is interleaved.
This is the function i need to use :
signal.correlate(n1, n2, mode='full')
how can I create an array of data for each channel in order to use the correlate function? are the last lines of the code correct?
Thank you
I found the answer, using print loudness(samps[:,0]), loudness(samps[:,3]). It print in the shell " mic 1 loudness , mic 4 loudness"
I need to apply audio to video at certain time with certain duration, but some audio duration is bigger(or smaller) then needed. How to change speed of audio without changing pitch? I tried to change fps(by multiplying to division of needed duration to audio duration) but it is not work as I want.
original = VideoFileClip("orig.mp4")
clips = [orig.audio.volumex(0.3)]
subs = [] #some array
i = 0
for sub in subs:
clip = AudioFileClip("\\temp{}.mp3")
mult = clip.duration / (sub.end - sub.start) + 0.00001
clip = AudioArrayClip(clip.to_soundarray(buffersize=500, fps=24000/mult), fps=24000).set_start(sub.start).set_end(sub.end)
clips.append(clip)
i += 1
final = CompositeAudioClip(clips)
final.write_audiofile("final.mp3")
you can use librosa module:
from scipy.io import wavfile
import librosa, numpy as np
song, fs = librosa.load("song.wav")
song_2_times_faster = librosa.effects.time_stretch(song, 2)
scipy.io.wavfile.write("song_2_times_faster.wav", fs, song_2_times_faster) # save the song
Using wave: Change the sampling rate
import wave
CHANNELS = 1
swidth = 2
Change_RATE = 2
spf = wave.open('VOZ.wav', 'rb')
RATE=spf.getframerate()
signal = spf.readframes(-1)
wf = wave.open('changed.wav', 'wb')
wf.setnchannels(CHANNELS)
wf.setsampwidth(swidth)
wf.setframerate(RATE*Change_RATE)
wf.writeframes(signal)
wf.close()
I'm trying to test a Python script that hopefully produces an audio spectrogram from a wav file. I assume I need to input a path to a wav, but I am getting an error: IndexError: list index out of range when I tried it by entering it here:
sr,x = scipy.io.wavfile.read('mySoundFile.wav')
I also tried the path as an argument in the command line, but I am not getting it right. Any help?
http://mail.python.org/pipermail/chicago/2010-December/007314.html
"""
Compute and display a spectrogram.
Give WAV file as input
"""
import matplotlib.pyplot as plt
import scipy.io.wavfile
import numpy as np
import sys
wavfile = sys.argv[1]
sr,x = scipy.io.wavfile.read('BeatBoy01.wav')
## Parameters: 10ms step, 30ms window
nstep = int(sr * 0.01)
nwin = int(sr * 0.03)
nfft = nwin
window = np.hamming(nwin)
## will take windows x[n1:n2]. generate
## and loop over n2 such that all frames
## fit within the waveform
nn = range(nwin, len(x), nstep)
X = np.zeros( (len(nn), nfft/2) )
for i,n in enumerate(nn):
xseg = x[n-nwin:n]
z = np.fft.fft(window * xseg, nfft)
X[i,:] = np.log(np.abs(z[:nfft/2]))
plt.imshow(X.T, interpolation='nearest',
origin='lower',
aspect='auto')
plt.show()
You can use this try/except to get around the IndexError:
try:
wavefile = sys.argv[1]
except IndexError:
wavfile = 'BeatBoy01.wav'
sr,x = scipy.io.wavfile.read(wavfile)
This effectively sets the default file to BeatBoy01.wav if no argument is passed to the script. Keep in mind that BeatBoy01.wav should be in the same directory from where the script is executed for this to work.
For easier argument parsing, have a look at the OptParse library.
Dont use "wavfile" as a variable, it is the name of the library.
Try this:
inputFile = sys.argv[1]
sr,x = scipy.io.wavfile.read(inputFile)