I'm trying to build an 'old radio' with raspberry pi 4 and python 3. I have a button which records my sound as long as I'm holding the button. When I release it, it replays the recorded sound and also plays an other wav file as background noise. My problem is that my voice is too clear and doesn't sound like an 'old radio' with noises and crackling.
I have tried to change CHUNK and nothing happened. Changing the RATE only makes my sound deeper/higher.
I'm using an USB converter for the microphone and the headset aswell.
Basically, I want to add 'noise' or any effect to my voice that makes it sound like it's from an old radio.
import time
import pyaudio
import wave
import sched
from pygame import mixer
import audiosegment
import numpy as np
from gpiozero import Button
CHUNK = 8192
FORMAT = pyaudio.paInt16
CHANNELS = 2
RATE = 45000
RECORD_SECONDS = 5
WAVE_OUTPUT_FILENAME = "output.wav"
button = Button(2)
p = pyaudio.PyAudio()
frames = []
def callback(in_data, frame_count, time_info, status):
frames.append(in_data)
return in_data, pyaudio.paContinue
started = False
stream = None
playing = None;
def recorder():
global started, p, stream, frames, button, playing
if button.is_pressed and not started and not playing:
# Start the recording
try:
stream = p.open(format=FORMAT,
channels=CHANNELS,
rate=RATE,
input=True,
frames_per_buffer=CHUNK,
stream_callback=callback)
print("Stream active:", stream.is_active())
started = True
print("start Stream")
except:
raise
elif not button.is_pressed and started:
try:
started = False
print("Stop recording")
stream.stop_stream()
stream.close()
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))
mixer.init()
mixer.Channel(0).set_volume(0.1)
tada = mixer.Sound('old_radio.wav')
channel = tada.play()
s = audiosegment.from_file(WAVE_OUTPUT_FILENAME)
main = mixer.Sound(WAVE_OUTPUT_FILENAME)
channel1 = main.play()
if 'channel1' in locals() and channel1 is not None:
while channel1.get_busy():
playing = True
channel.unpause()
channel.pause()
channel.stop()
channel1.stop()
playing = False
frames = []
except:
raise
# Reschedule the recorder function in 100 ms.
task.enter(0.1, 1, recorder, ())
print("Press and hold the button to begin recording")
print("Release the button to end recording")
task = sched.scheduler(time.time, time.sleep)
task.enter(0.1, 1, recorder, ())
task.run()
I'm not really that good with Python. Recently I've been toying around with pyAudio and using a simple script managed to record system sound. Thing is, the script I'm using is ran using a fixed timer. Instead, I want to end the recording whenever I want, from command line.
Does anyone know how to achieve this, if possible at all?
import pyaudio
import wave
import sys
DURATION = int(sys.argv[1])
FORMAT = pyaudio.paInt16
CHANNELS = 2
RATE = 44100
CHUNK = 1024
RECORD_SECONDS = DURATION
WAVE_OUTPUT_FILENAME = "file.wav"
print("hello")
p = pyaudio.PyAudio()
stream = p.open(format=FORMAT, channels=CHANNELS, rate=RATE, input=True, 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("Finished recording")
stream.stop_stream()
stream.close()
p.terminate()
waveFile = wave.open(WAVE_OUTPUT_FILENAME, 'wb')
waveFile.setnchannels(CHANNELS)
waveFile.setsampwidth(p.get_sample_size(FORMAT))
waveFile.setframerate(RATE)
waveFile.writeframes(b''.join(frames))
waveFile.close()
Same as in topic, i want to create timer while the recording function would run. I have tried a lot of ideas but all of them resulted as an error or just broke up program, maybe you have some ideas?
class rec(object):
def __init__(self):
FORMAT = pyaudio.paInt16
CHANNELS = 2
RATE = 44100
CHUNK = 1024
WAVE_OUTPUT_FILENAME = str(e2.get() + '.wav')
try:
RECORD_SECONDS = int(e1.get())
except ValueError:
tkMessageBox.showinfo("Error", "Please Enter number of seconds")
i = 0
while os.path.exists(WAVE_OUTPUT_FILENAME):
WAVE_OUTPUT_FILENAME = str(e2.get() + '%d.wav'%i)
i += 1
audio = pyaudio.PyAudio()
stream = audio.open(format=FORMAT, channels=CHANNELS,
rate=RATE, input=True,
frames_per_buffer=CHUNK)
frames = []
print "recording...\n\n"
for i in range(0, int(RATE / CHUNK * RECORD_SECONDS)):
sys.stdout.write('\r%i' % i)
data = stream.read(CHUNK)
frames.append(data)
print "\nfinished recording\n"
# stop Recording
stream.stop_stream()
stream.close()
audio.terminate()
waveFile = wave.open(WAVE_OUTPUT_FILENAME, 'wb')
waveFile.setnchannels(CHANNELS)
waveFile.setsampwidth(audio.get_sample_size(FORMAT))
waveFile.setframerate(RATE)
waveFile.writeframes(b''.join(frames))
# waveFile.Wave_read.getnframes()
waveFile.close()
tkMessageBox.showinfo("Recorded", "Track:%s"%WAVE_OUTPUT_FILENAME)
I tried to make other function with timer and initiate that function, when i press button, but if i pressed button firstly program ran rec function, and after gone timer function, also i tried to initiate those function by pipe but it does the same, do i have to make that with threads?
I've been tinkering around with pyaudio for a while now, trying to reverse a simple wave file with no success.
In (my) theory I would only have to iterate from end to beginning through the file with every callback of pyaudio (1024 frames) fetch the audio data from the according index in the file, reverse the resulting string and play it.
Here is my code (only pyaudio callback and file handling, the rest is untouched from the example code):
import pyaudio
import wave
import time
import sys
if len(sys.argv) < 2:
print("Plays a wave file.\n\nUsage: %s filename.wav" % sys.argv[0])
sys.exit(-1)
index = 40*1024
wf = wave.open(sys.argv[1], 'rb')
wf.setpos(index)
p = pyaudio.PyAudio()
def callback(in_data, frame_count, time_info, status):
global index
data = wf.readframes(frame_count)
data = data[::-1]
index-=1024
wf.setpos(index)
return (data, pyaudio.paContinue)
stream = p.open(format=p.get_format_from_width(wf.getsampwidth()),
channels=wf.getnchannels(),
rate=wf.getframerate(),
output=True,
stream_callback=callback)
stream.start_stream()
while stream.is_active():
time.sleep(0.1)
stream.stop_stream()
stream.close()
wf.close()
p.terminate()
I know this will crash when it reaches the file beginning, but it should play 40 × 1024 frames of reversed audio...
If the file you want reversed is small enough to fit in memory, your best bet would be loading it entirely and reversing the data, then streaming it:
import pyaudio
import wave
wavefile_name = 'wavefile.wav'
wf = wave.open(wavefile_name, 'rb')
p = pyaudio.PyAudio()
stream = p.open(format =
p.get_format_from_width(wf.getsampwidth()),
channels = wf.getnchannels(),
rate = wf.getframerate(),
output = True)
full_data = []
data = wf.readframes(1024)
while data:
full_data.append(data)
data = wf.readframes(1024)
data = ''.join(full_data)[::-1]
for i in range(0, len(data), 1024):
stream.write(data[i:i+1024])
However, if the file is too big to fit in memory, you'll need some way of reading the file backwards and feed into the sound system. That's an entirely different problem, because it involves some low level I/O programming to handle the backward reading.
Edit: after seeing your full code, I don't see any errors. The code runs properly in my computer, only to fail at the ending of the playback. However, you can do two things. First, you can go to the end of the wavefile to play the entire sound backwards, using this:
wf = wave.open(sys.argv[1], 'rb')
index = wf.getnframes() - 1024
wf.setpos(index)
Second, you have to modify the callback so it doesn't fail when the seek head goes beyond the beginning of the file:
def callback(in_data, frame_count, time_info, status):
global index
data = wf.readframes(frame_count)
data = data[::-1]
index-=1024
if index < 0:
return (data, pyaudio.paAbort)
else:
wf.setpos(max(index,0))
return (data, pyaudio.paContinue)
Other than that, it works pretty ok.
I'm trying to make real-time plotting sound in python. I need to get chunks from my microphone.
Using PyAudio, try to use
import pyaudio
import wave
import sys
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"
all = []
for i in range(0, RATE / chunk * RECORD_SECONDS):
data = stream.read(chunk)
all.append(data)
print "* done recording"
stream.close()
p.terminate()
After, I getting the followin error:
* recording
Traceback (most recent call last):
File "gg.py", line 23, in <module>
data = stream.read(chunk)
File "/usr/lib64/python2.7/site-packages/pyaudio.py", line 564, in read
return pa.read_stream(self._stream, num_frames)
IOError: [Errno Input overflowed] -9981
I can't understand this buffer. I want, to use blocking IO mode, so if chunks not available, i want to wait for those chunks. But when I creating try except segment or sleep(0.1), i hear clicks, so this is not what i want.
Please suggest the best solution for my ploblem?
pyaudio.Stream.read() has a keyword parameter exception_on_overflow, set this to False.
For your sample code that would look like:
import pyaudio
import wave
import sys
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"
all = []
for i in range(0, RATE / chunk * RECORD_SECONDS):
data = stream.read(chunk, exception_on_overflow = False)
all.append(data)
print "* done recording"
stream.close()
p.terminate()
See the PyAudio documentation for more details.
It seems like a lot of people are encountering this issue. I dug a bit into it and I think it means that between the previous call to stream.read() and this current call, data from the stream was lost (i.e. the buffer filled up faster than you cleared it).
From the doc for Pa_ReadStream() (the PortAudio function that stream.read() eventually ends up calling):
#return On success PaNoError will be returned, or PaInputOverflowed if
input data was discarded by PortAudio after the previous call and
before this call.
(PaInputOverflowed then causes an IOError in the pyaudio wrapper).
If it's OK for you to not capture every single frame, then you may ignore this error. If it's absolutely critical for you to have every frame, then you'll need to find a way to increase the priority of your application. I'm not familiar enough with Python to know a pythonic way to do this, but it's worth trying a simple nice command, or changing the scheduling policy to SCHED_DEADLINE.
Edit:
One issue right now is that when IOError is thrown, you lose all the frames collected in that call. To instead ignore the overflow and just return what we have, you can apply the patch below, which will cause stream.read() to ignore output underrun and input overflow errors from PortAudio (but still throw something if a different error occurred). A better way would be to make this behaviour (throw/no throw) customizable depending on your needs.
diff --git a/src/_portaudiomodule.c b/src/_portaudiomodule.c
index a8f053d..0878e74 100644
--- a/src/_portaudiomodule.c
+++ b/src/_portaudiomodule.c
## -2484,15 +2484,15 ## pa_read_stream(PyObject *self, PyObject *args)
} else {
/* clean up */
_cleanup_Stream_object(streamObject);
+
+ /* free the string buffer */
+ Py_XDECREF(rv);
+
+ PyErr_SetObject(PyExc_IOError,
+ Py_BuildValue("(s,i)",
+ Pa_GetErrorText(err), err));
+ return NULL;
}
-
- /* free the string buffer */
- Py_XDECREF(rv);
-
- PyErr_SetObject(PyExc_IOError,
- Py_BuildValue("(s,i)",
- Pa_GetErrorText(err), err));
- return NULL;
}
return rv;
I got the same error when I ran your code. I looked at the default sample rate of my default audio device, my macbook's internal microphone, it was 48000Hz not 44100Hz.
p.get_device_info_by_index(0)['defaultSampleRate']
Out[12]: 48000.0
When I changed RATE to this value, it worked.
I worked this on OS X 10.10, Got the same error while trying to get audio from the microphone in a SYBA USB card (C Media chipset), and process it in real time with fft's and more:
IOError: [Errno Input overflowed] -9981
The overflow was completely solved when using a Callback Mode, instead of the Blocking Mode, as written by libbkmz.(https://www.python.org/dev/peps/pep-0263/)
Based on that, the bit of the working code looked like this:
"""
Creating the audio stream from our mic
"""
rate=48000
self.chunk=2**12
width = 2
p = pyaudio.PyAudio()
# callback function to stream audio, another thread.
def callback(in_data,frame_count, time_info, status):
self.audio = numpy.fromstring(in_data,dtype=numpy.int16)
return (self.audio, pyaudio.paContinue)
#create a pyaudio object
self.inStream = p.open(format = p.get_format_from_width(width, unsigned=False),
channels=1,
rate=rate,
input=True,
frames_per_buffer=self.chunk,
stream_callback = callback)
"""
Setting up the array that will handle the timeseries of audio data from our input
"""
self.audio = numpy.empty((self.buffersize),dtype="int16")
self.inStream.start_stream()
while True:
try:
self.ANY_FUNCTION() #any function to run parallel to the audio thread, running forever, until ctrl+C is pressed.
except KeyboardInterrupt:
self.inStream.stop_stream()
self.inStream.close()
p.terminate()
print("* Killed Process")
quit()
This code will create a callback function, then create a stream object, start it and then loop in any function. A separate thread streams audio, and that stream is closed when the main loop is stopped. self.audio is used in any function. I also had problems with the thread running forever if not terminated.
Since Pyaudio runs this stream in a separate thread, and this made the audio stream stable, the Blocking mode might have been saturating depending on the speed or timing of the rest of the processes in the script.
Note that the chunk size is 2^12, but smaller chunks work just as well. There are other parameters I considered and played around with to make sure they all made sense:
Chunk size larger or smaller(no effect)
Number and format of bits for the words in the buffer, signed 16 bit in this case.
signedness of variables(tried with unsigned and got saturation patterns)
Nature of mic input, and selection as default in the system, gain etc.
Hope that works for someone!
My other answer solved the problem in most cases. However sometimes the error still occurs.
That was the reason why I scrapped pyaudio and switched to pyalsaaudio. My Raspy now smoothly records any sound.
import alsaaudio
import numpy as np
import array
# constants
CHANNELS = 1
INFORMAT = alsaaudio.PCM_FORMAT_FLOAT_LE
RATE = 44100
FRAMESIZE = 1024
# set up audio input
recorder=alsaaudio.PCM(type=alsaaudio.PCM_CAPTURE)
recorder.setchannels(CHANNELS)
recorder.setrate(RATE)
recorder.setformat(INFORMAT)
recorder.setperiodsize(FRAMESIZE)
buffer = array.array('f')
while <some condition>:
buffer.fromstring(recorder.read()[1])
data = np.array(buffer, dtype='f')
FORMAT = pyaudio.paInt16
Make sure to set the correct format, my internal microphone was set to 24 Bit (see Audio-Midi-Setup application).
I had the same issue on the really slow raspberry pi, but I was able to solve it (for most cases) by using the faster array module for storing the data.
import array
import pyaudio
FORMAT = pyaudio.paInt16
CHANNELS = 1
INPUT_CHANNEL=2
RATE = 48000
CHUNK = 512
p = pyaudio.PyAudio()
stream = p.open(format=FORMAT,
channels=CHANNELS,
rate=RATE,
input=INPUT_CHANNEL,
frames_per_buffer =CHUNK)
print("* recording")
try:
data = array.array('h')
for i in range(0, int(RATE / CHUNK * RECORD_SECONDS)):
data.fromstring(stream.read(CHUNK))
finally:
stream.stop_stream()
stream.close()
p.terminate()
print("* done recording")
The content of data is rather binary afterwards.
But you can use numpy.array(data, dtype='i') to get a numpy array of intergers.
Instead of
chunk = 1024
use:
chunk = 4096
It worked for me on a USB microphone.
This was helpful for me:
input_ = stream.read(chunk, exception_on_overflow=False)
exception_on_overflow = False
For me this helped: https://stackoverflow.com/a/46787874/5047984
I used multiprocessing to write the file in parallel to recording audio. This is my code:
recordAudioSamples.py
import pyaudio
import wave
import datetime
import signal
import ftplib
import sys
import os
# configuration for assos_listen
import config
# run the audio capture and send sound sample processes
# in parallel
from multiprocessing import Process
# CONFIG
CHUNK = config.chunkSize
FORMAT = pyaudio.paInt16
CHANNELS = 1
RATE = config.samplingRate
RECORD_SECONDS = config.sampleLength
# HELPER FUNCTIONS
# write to ftp
def uploadFile(filename):
print("start uploading file: " + filename)
# connect to container
ftp = ftplib.FTP(config.ftp_server_ip, config.username, config.password)
# write file
ftp.storbinary('STOR '+filename, open(filename, 'rb'))
# close connection
ftp.quit()
print("finished uploading: " +filename)
# write to sd-card
def storeFile(filename,frames):
print("start writing file: " + filename)
wf = wave.open(filename, 'wb')
wf.setnchannels(CHANNELS)
wf.setsampwidth(p.get_sample_size(FORMAT))
wf.setframerate(RATE)
wf.writeframes(b''.join(frames))
wf.close()
print(filename + " written")
# abort the sampling process
def signal_handler(signal, frame):
print('You pressed Ctrl+C!')
# close stream and pyAudio
stream.stop_stream()
stream.close()
p.terminate()
sys.exit(0)
# MAIN FUNCTION
def recordAudio(p, stream):
sampleNumber = 0
while (True):
print("* recording")
sampleNumber = sampleNumber +1
frames = []
startDateTimeStr = datetime.datetime.now().strftime("%Y_%m_%d_%I_%M_%S_%f")
for i in range(0, int(RATE / CHUNK * RECORD_SECONDS)):
data = stream.read(CHUNK)
frames.append(data)
fileName = str(config.sensorID) + "_" + startDateTimeStr + ".wav"
# create a store process to write the file in parallel
storeProcess = Process(target=storeFile, args=(fileName,frames))
storeProcess.start()
if (config.upload == True):
# since waiting for the upload to finish will take some time
# and we do not want to have gaps in our sample
# we start the upload process in parallel
print("start uploading...")
uploadProcess = Process(target=uploadFile, args=(fileName,))
uploadProcess.start()
# ENTRYPOINT FROM CONSOLE
if __name__ == '__main__':
p = pyaudio.PyAudio()
stream = p.open(format=FORMAT,
channels=CHANNELS,
rate=RATE,
input=True,
frames_per_buffer=CHUNK)
# directory to write and read files from
os.chdir(config.storagePath)
# abort by pressing C
signal.signal(signal.SIGINT, signal_handler)
print('\n\n--------------------------\npress Ctrl+C to stop the recording')
# start recording
recordAudio(p, stream)
config.py
### configuration file for assos_listen
# upload
upload = False
# config for this sensor
sensorID = "al_01"
# sampling rate & chunk size
chunkSize = 8192
samplingRate = 44100 # 44100 needed for Aves sampling
# choices=[4000, 8000, 16000, 32000, 44100] :: default 16000
# sample length in seconds
sampleLength = 10
# configuration for assos_store container
ftp_server_ip = "192.168.0.157"
username = "sensor"
password = "sensor"
# storage on assos_listen device
storagePath = "/home/pi/assos_listen_pi/storage/"