I am trying to write a script that records audio when certain keys are pressed and stops recording when the keys are released, and saves the audio to a .wav.
The program starts and activates my microphone, but it never seems to actually record and removing my fingers from the Control + Shift keys does not seem to have any affect. At any rate, I do not end up with a .wav file.
import keyboard
import pyaudio
import wave
import time
CHUNK = 1024
FORMAT = pyaudio.paInt16
CHANNELS = 1
RATE = 44100
RECORD_SECONDS = 5
WAVE_OUTPUT_FILENAME = "output.wav"
p = pyaudio.PyAudio()
stream = p.open(format=FORMAT,
channels=CHANNELS,
rate=RATE,
input=True,
frames_per_buffer=CHUNK)
print("* recording")
frames = []
while True:
if keyboard.is_pressed('ctrl+shift'):
print('You Pressed A Key!')
break
for i in range(0, int(RATE / CHUNK * RECORD_SECONDS)):
data = stream.read(CHUNK)
frames.append(data)
print("* done recording")
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()
Related
i'm trying to build an app with pyaudio that records speaker and microphone sound but i just don't know how to record them both.I tried enabling stereo mixer but it didn't work out because i could only listen the sound from the speakers.
This code that i'm using records audio from the default microphone in a background thread using pyaudio:
import pyaudio
import wave
import threading
import time
import subprocess
CHUNK = 1024
FORMAT = pyaudio.paInt16
CHANNELS = 2
RATE = 44100
RECORD_SECONDS = 5
WAVE_OUTPUT_FILENAME = "tmp/tmp.wav"
class recorder:
def __init__(self):
self.going = False
self.process = None
self.filename = "ScreenCapture.mpg"
def record(self,filename):
try:
if self.process.is_alive():
self.going = False
except AttributeError:
print("test")
self.process = threading.Thread(target=self._record)
self.process.start()
self.filename = filename
def _record(self):
p = pyaudio.PyAudio()
stream = p.open(format=FORMAT,
channels=CHANNELS,
rate=RATE,
input=True,
frames_per_buffer=CHUNK)
print("* recording")
frames = []
self.going = True
while self.going:
data = stream.read(CHUNK)
frames.append(data)
print("* done recording")
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()
def stop_recording(self):
self.going = False
I'm using Windows and I can also use other library, not only PyAudio. I just need to get this work.
Edit : I found this code to record the output from speaker but i couldn't make it work in my application:
import pyaudio
import wave
CHUNK = 1024
FORMAT = pyaudio.paInt16
CHANNELS = 2
RATE = 44100
RECORD_SECONDS = 5
WAVE_OUTPUT_FILENAME = "output.wav"
p = pyaudio.PyAudio()
SPEAKERS = p.,get_default_output_device_info()["hostApi"]
stream = p.open(format=FORMAT,
channels=CHANNELS,
rate=RATE,
input=True,
frames_per_buffer=CHUNK,
input_host_api_specific_stream_info=SPEAKERS,
as_loopback=True)
print("* recording")
frames = []
for i in range(0, int(RATE / CHUNK * RECORD_SECONDS)):
data = stream.read(CHUNK)
frames.append(data)
print("* done recording")
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()
UPDATE :
I could record both my speaker and my microphone with this:
import pyaudio
import wave
import numpy as np
CHUNK = 1024
FORMAT = pyaudio.paInt16
RATE = 44100
RECORD_SECONDS = 2
WAVE_OUTPUT_FILENAME = "tmp.wav"
p = pyaudio.PyAudio()
for i in range(0, p.get_device_count()):
print(i, p.get_device_info_by_index(i)['name'])
#stream using as_loopback to get sound from OS
stream = p.open(
format = FORMAT,
channels = 2,
rate = RATE,
input=True,
frames_per_buffer=CHUNK,
input_device_index=2,
as_loopback=True)
##stream using my Microphone's input device
stream2 = p.open(
format = FORMAT,
channels = 1,
rate = RATE,
input=True,
frames_per_buffer=CHUNK,
input_device_index=1)
#as_loopback=False)
frames = []
frames2 = []
for i in range(0, int(RATE / CHUNK * RECORD_SECONDS)):
data = stream.read(CHUNK)
data2 = stream2.read(CHUNK)
frames.append(data)
frames2.append(data2)
#frames = as_loopback sound data (Speakers)
frames= b''.join(frames);
#frames2 = sound data of Microphone
frames2= b''.join(frames2);
#decoding Speaker data
Sdecoded = np.frombuffer(frames, 'int16')
#decoding the microphone data
Mdecoded = np.frombuffer(frames2, 'int16')
#converting Speaker data into a Numpy vector (making life easier when picking up audio channels)
Sdecoded= np.array(Sdecoded, dtype='int16')
#getting the data on the right side
direito=Sdecoded[1::2]
#getting the data on the left side
esquerdo=Sdecoded[::2]
#mixing everything to mono = add right side + left side + Microphone decoded data that is already mono
mix=(direito+esquerdo+Mdecoded)
#ensuring no value goes beyond the limits of short int
signal=np.clip(mix, -32767, 32766)
#encode the data again
encodecoded = wave.struct.pack("%dh"%(len(signal)), *list(signal))
#stop all streams and terminate pyaudio
stream.stop_stream()
stream.close()
stream2.stop_stream()
stream2.close()
p.terminate()
#recording mixed audio in mono
wf = wave.open(WAVE_OUTPUT_FILENAME, 'wb')
wf.setnchannels(1)
wf.setsampwidth(p.get_sample_size(FORMAT))
wf.setframerate(RATE)
wf.writeframes((encodecoded))
wf.close()
But when i try to mix with my code it doesn't work.What is going on?I think i'm almost solving it
import numpy as np
import pyaudio
import wave
import threading
import time
import subprocess
CHUNK = 1024
FORMAT = pyaudio.paInt16
RATE = 44100
RECORD_SECONDS = 5
WAVE_OUTPUT_FILENAME = "tmp/tmp.wav"
p = pyaudio.PyAudio()
for i in range(0, p.get_device_count()):
print(i, p.get_device_info_by_index(i)['name'])
class recorder:
def __init__(self):
self.going = False
self.process = None
self.filename = "ScreenCapture.mpg"
def record(self,filename):
try:
if self.process.is_alive():
self.going = False
except AttributeError:
print("test")
self.process = threading.Thread(target=self._record)
self.process.start()
self.filename = filename
def _record(self):
p = pyaudio.PyAudio()
#stream using as_loopback to get sound from OS
stream = p.open(
format=FORMAT,
channels=2,
rate=RATE,
input=True,
frames_per_buffer=CHUNK,
input_device_index=2,
as_loopback=True)
##stream using my Microphone's input device
stream2 = p.open(
format=FORMAT,
channels=1,
rate=RATE,
input=True,
frames_per_buffer=CHUNK,
input_device_index=1)
# as_loopback=False)
#print("* recording")
frames = []
frames2= []
self.going = True
while self.going:
data = stream.read(CHUNK)
data2 = stream2.read(CHUNK)
frames.append(data)
frames2.append(data2)
# frames = as_loopback sound data (Speakers)
frames = b''.join(frames);
# frames2 = sound data of Microphone
frames2 = b''.join(frames2);
# decoding Speaker data
Sdecoded = np.frombuffer(frames, 'int16')
# decoding the microphone data
Mdecoded = np.frombuffer(frames2, 'int16')
# converting Speaker data into a Numpy vector (making life easier when picking up audio channels)
Sdecoded = np.array(Sdecoded, dtype='int16')
# getting the data on the right side
direito = Sdecoded[1::2]
# getting the data on the left side
esquerdo = Sdecoded[::2]
# mixing everything to mono = add right side + left side + Microphone decoded data that is already mono
mix = (direito + esquerdo + Mdecoded)
# ensuring no value goes beyond the limits of short int
signal = np.clip(mix, -32767, 32766)
# encode the data again
encodecoded = wave.struct.pack("%dh" % (len(signal)), *list(signal))
# print("* done recording")
stream.stop_stream()
stream.close()
stream2.stop_stream()
stream2.close()
p.terminate()
wf = wave.open(WAVE_OUTPUT_FILENAME, 'wb')
wf.setnchannels(1)
wf.setsampwidth(p.get_sample_size(FORMAT))
wf.setframerate(RATE)
wf.writeframes(encodecoded)
wf.close()
def stop_recording(self):
self.going = False
I made the code very clean and commented each part so you understand what is going on.I made a for loop in the beginning for Pyaudio show me what are the interfaces I have in my OS:
0 Mapeador de som da Microsoft - Input
1 Microfone (Realtek(R) Audio)
2 Mixagem estéreo (Realtek(R) Aud
3 Mapeador de som da Microsoft - Output
4 Alto-falantes (Realtek(R) Audio
5 Alto-falantes (Realtek(R) Audio)
6 Microfone (Realtek(R) Audio)
7 Mixagem estéreo (Realtek(R) Audio)
8 Speakers 1 (Realtek HD Audio output with SST)
9 Speakers 2 (Realtek HD Audio output with SST)
10 Alto-falante (Realtek HD Audio output with SST)
11 Microfone (Realtek HD Audio Mic input)
12 Mixagem estéreo (Realtek HD Audio Stereo input)
You can use 2 separate threads to record from 2 different devices ( providing separate device Index) into separate Wav files.
Then Mix these 2 files Using the pydub library
from pydub import AudioSegment
speakersound = AudioSegment.from_file("/path/speaker.wav")
micsound = AudioSegment.from_file("/path/mic.wav")
mixsound = speakersound.overlay(micsound)
mixsound.export("/path/mixsound.wav", format='wav')
I am using pyaudio to record data from a two channel soundcard using this function
def record(self):
'''
Record Function reads from stream with configured soundcard and stores items in an array
uses callback function as can be told to stop recording during stream. After ending writes contents
to wav file
'''
wf = wave.open('audiooutput.wav', 'wb')
wf.setnchannels(2)
wf.setsampwidth(pyaudio.get_sample_size(pyaudio.paInt16))
wf.setframerate(44100)
p = pyaudio.PyAudio()
frames = []
# sub function checks the queue for a message to stop recording
def check_for_recordstop():
try:
message = self.my_queue.get(timeout = 0.1)
except:
return
if message == None:
pass
elif message.payload == "Stop":
self.confirm_message_recieved(message)
stream.stop_stream()
#subfunction callback
def callback(in_data, frame_count, time_info, status):
if stream.is_active():
frames.append(in_data)
return (in_data, pyaudio.paContinue)
else:
frames.append(in_data)
return (in_data, pyaudio.paComplete)
stream = p.open(format=pyaudio.get_sample_size(pyaudio.paInt16),
channels= 2,
rate=44100,
input=True,
frames_per_buffer=1024,
input_device_index=1,
stream_callback = callback)
self.logger.info("Recording")
stream.start_stream() # callback is run on a new thread when start_stream() is triggered
while stream.is_active(): #Loop to keep thread alive while callback is running
time.sleep(0.1)
check_for_recordstop()
print("done")
stream.close()
wf.writeframes(b''.join(frames))
wf.close()
p.terminate()
However when viewing this data in audacity I do not end up with 2 channel stream it looks like this
But When using a Function like this
CHUNK = 1024
FORMAT = pyaudio.paInt16
CHANNELS = 2
RATE = 44100
RECORD_SECONDS = 5
p = pyaudio.PyAudio()
stream = p.open(format=FORMAT,
channels=CHANNELS,
rate=RATE,
input=True,
input_device_index=1,
frames_per_buffer=CHUNK)
print("* recording")
frames = []
for i in range(0, int(RATE / CHUNK * RECORD_SECONDS)):
data = stream.read(CHUNK)
frames.append(data)
print("* done recording")
stream.stop_stream()
stream.close()
p.terminate()
wf = wave.open("pyaudoutput.wav", 'wb')
wf.setnchannels(CHANNELS)
wf.setsampwidth(p.get_sample_size(FORMAT))
wf.setframerate(RATE)
wf.writeframes(b''.join(frames))
wf.close()
the audio data comes out as expected see here
I cannot seem to get the first function to produce the desired results from the second!
I need to use the callback capability of pyaudio for my usecase but cannot get the two channels to separate. Any Advice would be great!
I was opening the stream using the sample size not the sample width
format=pyaudio.get_sample_size(pyaudio.paInt16),
should be
format=pyaudio.get_format_from_width(wf.getsampwidth()),
using the sample width from the file I writing to solved the issue
I am using the following code to record audio in python, it works perfectly. But when I set the microphone input volume to high, the audio thus recorded using this code has a lot of noise. How to get rid of this noise??
import pyaudio
import wave
def record(filename):
FORMAT = pyaudio.paInt16
CHANNELS = 1
RATE = 16000
CHUNK = 1024
RECORD_SECONDS = 15
WAVE_OUTPUT_FILENAME = filename
audio = pyaudio.PyAudio()
# start Recording
stream = audio.open(format=FORMAT, channels=CHANNELS,
rate=RATE, input=True,
frames_per_buffer=CHUNK)
print("recording...")
print('---------------------------------')
print(int(RATE / CHUNK * RECORD_SECONDS))
print('*********************************')
frames = []
for i in range(0, int(RATE / CHUNK * RECORD_SECONDS)):
data = stream.read(CHUNK)
frames.append(data)
print("Recording finished. . .")
# stop Recording
stream.stop_stream()
stream.close()
audio.terminate()
waveFile = wave.open(WAVE_OUTPUT_FILENAME, 'wb')
waveFile.setnchannels(CHANNELS)
waveFile.setsampwidth(2)
audio.get_sample_size(FORMAT)
waveFile.setframerate(RATE)
waveFile.writeframes(b''.join(frames))
waveFile.close()
I am using python to process the audio signal. I would like to record a signal and keep its SPL level. I am using pyaudio Library to do that. Here is the example I found. However, I find the signal is around 18db larger then it should be. Is there a way to change the recorded signal amplitude in the file wav? Second question is for the wav file, data are coded in signed 16 bits. I have the impression that the negative values are coded using Two's complement can anyone confirm this please?
Thank you very much!
enter code here def record(output_file_name, rec_device_id=None):
# Sampling = Data is recorded per batches of 1024 bits
CHUNK = 1024
FORMAT = pyaudio.paInt16
CHANNELS = 1
# Sampling = 48kHz
RATE = 48000
RECORD_SECONDS = 5
WAVE_OUTPUT_FILENAME = output_file_name
p = pyaudio.PyAudio()
stream = p.open(format=FORMAT,
channels=CHANNELS,
rate=RATE,
input=True,
frames_per_buffer=CHUNK,
input_device_index=rec_device_id)
# Sampling = 48kHz
print("*** recording start ***")
frames = []
rms=np.zeros(RATE / CHUNK * RECORD_SECONDS+1)
for i in range(0, int(RATE / CHUNK * RECORD_SECONDS)):
data = stream.read(CHUNK)
#print("*** data ***",data)
#decoded = struct.unpack('h',data)
print("*** decoded ***",decoded)
frames.append(data)
rms[i] = audioop.rms(data, 2)
print("*** recording done ***")
print("*** rms ***",rms)
stream.stop_stream()
stream.close()
p.terminate()
wf = wave.open(output_file_name, 'wb')
wf.setnchannels(CHANNELS)
wf.setsampwidth(p.get_sample_size(FORMAT))
wf.setframerate(RATE)
wf.writeframes(b''.join(frames))
wf.close()
I am writing a program in Python, to create and record audio file by using pyaudio. Audio file creation is done successfully but in recording, i am facing trouble because it is capturing nothing.
import pyaudio
import wave
import time
import sys
from datetime import datetime
CHUNK = 1024
FORMAT = pyaudio.paInt16
CHANNELS = 1
RATE = 44100
RECORD_SECONDS = 5
WAVE_OUTPUT_FILENAME = 'Audio.wav'
p = pyaudio.PyAudio()
stream = p.open(format=FORMAT,
channels=CHANNELS,
rate=RATE,
input=True,
frames_per_buffer=CHUNK)
frames = []
print('* done recording')
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()
How can I add duration in audio recording?
I got my answer, which I am sharing here
import pyaudio, wave, sys
CHUNK = 8192
FORMAT = pyaudio.paInt16
CHANNELS = 1
RATE = 44100
RECORD_SECONDS = 10
WAVE_OUTPUT_FILENAME = 'Audio_.wav'
p = pyaudio.PyAudio()
stream = p.open(format=FORMAT,
channels = CHANNELS,
rate = RATE,
input = True,
input_device_index = 0,
frames_per_buffer = CHUNK)
print("* recording")
frames = []
for i in range(0, int(RATE / CHUNK * RECORD_SECONDS)):
data = stream.read(CHUNK)
frames.append(data)
print("* done recording")
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()
Actually you had to call stream.read to start the capture,seems you figured it out haha