I tried pygame for playing wav file like this:
import pygame
pygame.init()
pygame.mixer.music.load("mysound.wav")
pygame.mixer.music.play()
pygame.event.wait()
but It change the voice and I don't know why!
I read this link solutions and can't solve my problem with playing wave file!
for this solution I dont know what should I import?
s = Sound()
s.read('sound.wav')
s.play()
and for this solution /dev/dsp dosen't exist in new version of linux :
from wave import open as waveOpen
from ossaudiodev import open as ossOpen
s = waveOpen('tada.wav','rb')
(nc,sw,fr,nf,comptype, compname) = s.getparams( )
dsp = ossOpen('/dev/dsp','w')
try:
from ossaudiodev import AFMT_S16_NE
except ImportError:
if byteorder == "little":
AFMT_S16_NE = ossaudiodev.AFMT_S16_LE
else:
AFMT_S16_NE = ossaudiodev.AFMT_S16_BE
dsp.setparameters(AFMT_S16_NE, nc, fr)
data = s.readframes(nf)
s.close()
dsp.write(data)
dsp.close()
and when I tried pyglet It give me this error:
import pyglet
music = pyglet.resource.media('mysound.wav')
music.play()
pyglet.app.run()
--------------------------
nima#ca005 Desktop]$ python play.py
Traceback (most recent call last):
File "play.py", line 4, in <module>
music = pyglet.resource.media('mysound.wav')
File "/usr/lib/python2.7/site-packages/pyglet/resource.py", line 587, in media
return media.load(path, streaming=streaming)
File "/usr/lib/python2.7/site-packages/pyglet/media/__init__.py", line 1386, in load
source = _source_class(filename, file)
File "/usr/lib/python2.7/site-packages/pyglet/media/riff.py", line 194, in __init__
format = wave_form.get_format_chunk()
File "/usr/lib/python2.7/site-packages/pyglet/media/riff.py", line 174, in get_format_chunk
for chunk in self.get_chunks():
File "/usr/lib/python2.7/site-packages/pyglet/media/riff.py", line 110, in get_chunks
chunk = cls(self.file, name, length, offset)
File "/usr/lib/python2.7/site-packages/pyglet/media/riff.py", line 155, in __init__
raise RIFFFormatException('Size of format chunk is incorrect.')
pyglet.media.riff.RIFFFormatException: Size of format chunk is incorrect.
AL lib: ReleaseALC: 1 device not closed
You can use PyAudio. An example here on my Linux it works:
#!usr/bin/env python
#coding=utf-8
import pyaudio
import wave
#define stream chunk
chunk = 1024
#open a wav format music
f = wave.open(r"/usr/share/sounds/alsa/Rear_Center.wav","rb")
#instantiate PyAudio
p = pyaudio.PyAudio()
#open stream
stream = p.open(format = p.get_format_from_width(f.getsampwidth()),
channels = f.getnchannels(),
rate = f.getframerate(),
output = True)
#read data
data = f.readframes(chunk)
#play stream
while data:
stream.write(data)
data = f.readframes(chunk)
#stop stream
stream.stop_stream()
stream.close()
#close PyAudio
p.terminate()
Works for me on Windows:
https://pypi.org/project/playsound/
>>> from playsound import playsound
>>> playsound('/path/to/a/sound/file/you/want/to/play.wav')
NOTE: This has a bug in Windows where it doesn't close the stream.
I've added a PR for a fix here:
https://github.com/TaylorSMarks/playsound/pull/53/commits/53240d970aef483b38fc6d364a0ae0ad6f8bf9a0
The reason pygame changes your audio is mixer defaults to a 22k sample rate:
initialize the mixer module
pygame.mixer.init(frequency=22050, size=-16, channels=2, buffer=4096): return None
Your wav is probably 8k. So when pygame plays it, it plays roughly twice as fast. So specify your wav frequency in the init.
Pyglet has some problems correctly reading RIFF headers. If you have a very basic wav file (with exactly a 16 byte fmt block) with no other information in the fmt chunk (like 'fact' data), it works. But it makes no provision for additional data in the chunks, so it's really not adhering to the RIFF interface specification.
PyGame has 2 different modules for playing sound and music, the pygame.mixer module and the pygame.mixer.music module. This module contains classes for loading Sound objects and controlling playback. The difference is explained in the documentation:
The difference between the music playback and regular Sound playback is that the music is streamed, and never actually loaded all at once. The mixer system only supports a single music stream at once.
If you want to play a single wav file, you have to initialize the module and create a pygame.mixer.Sound() object from the file. Invoke play() to start playing the file. Finally, you have to wait for the file to play.
Use get_length() to get the length of the sound in seconds and wait for the sound to finish:
(The argument to pygame.time.wait() is in milliseconds)
import pygame
pygame.mixer.init()
my_sound = pygame.mixer.Sound('mysound.wav')
my_sound.play()
pygame.time.wait(int(my_sound.get_length() * 1000))
Alternatively you can use pygame.mixer.get_busy to test if a sound is being mixed. Query the status of the mixer continuously in a loop:
import pygame
pygame.init()
pygame.mixer.init()
my_sound = pygame.mixer.Sound('mysound.wav')
my_sound.play()
while pygame.mixer.get_busy():
pygame.time.delay(10)
pygame.event.poll()
Windows
winsound
If you are a Windows user,the easiest way is to use winsound.You don't even need to install it.
Not recommended, too few functions
import winsound
winsound.PlaySound("Wet Hands.wav", winsound.SND_FILENAME)
# add winsound.SND_ASYNC flag if you want to wait for it.
# like winsound.PlaySound("Wet Hands.wav", winsound.SND_FILENAME | winsound.SND_ASYNC)
mp3play
If you are looking for more advanced functions, you can try mp3play.
Unluckily,mp3play is only available in Python2 and Windows.
If you want to use it on other platforms,use playsound despite its poor functions.If you want to use it in Python3,I will give you the modified version which is available on Python 3.(at the bottom of the answer)
Also,mp3play is really good at playing wave files, and it gives you more choices.
import time
import mp3play
music = mp3play.load("Wet Hands.wav")
music.play()
time.sleep(music.seconds())
Cross-platform
playsound
Playsound is very easy to use,but it is not recommended because you can't pause or get some infomation of the music, and errors often occurs.Unless other ways doesn't work at all, you may try this.
import playsound
playsound.playsound("Wet Hands.wav", block=True)
pygame
I'm using this code and it works on Ubuntu 22.04 after my test.
If it doesn't work on your machine, consider updating your pygame lib.
import pygame
pygame.mixer.init()
pygame.mixer.music.load("Wet Hands.wav")
pygame.mixer.music.play()
while pygame.mixer.music.get_busy():
pass
pyglet
This works on Windows but it doesn't work on my Ubuntu, so I can do nothing.
import pyglet
import time
sound = pyglet.media.load("Wet Hands.wav", "Wet Hands.wav")
sound.play()
time.sleep(sound.duration)
Conclusion
It seems that you are using Linux,so playsound may be your choice.My code maybe cannot solve your problem by using pygame and pyglet,because I always use Windows.If none of the solutions work on your machine,I suggest you run the program on Windows...
To other users seeing my answer, I have done many tests among many libraries,so if you are using Windows,you may try mp3play which can play both mp3 and wave files, and mp3play is the most pythonic, easy, light-weight and functional library.
mp3play in Python3
just copy the code below and create a file named mp3play.py in your working directory and paste the content.
import random
from ctypes import windll, c_buffer
class _mci:
def __init__(self):
self.w32mci = windll.winmm.mciSendStringA
self.w32mcierror = windll.winmm.mciGetErrorStringA
def send(self, command):
buffer = c_buffer(255)
command = command.encode(encoding="utf-8")
errorcode = self.w32mci(command, buffer, 254, 0)
if errorcode:
return errorcode, self.get_error(errorcode)
else:
return errorcode, buffer.value
def get_error(self, error):
error = int(error)
buffer = c_buffer(255)
self.w32mcierror(error, buffer, 254)
return buffer.value
def directsend(self, txt):
(err, buf) = self.send(txt)
# if err != 0:
# print('Error %s for "%s": %s' % (str(err), txt, buf))
return err, buf
class _AudioClip(object):
def __init__(self, filename):
filename = filename.replace('/', '\\')
self.filename = filename
self._alias = 'mp3_%s' % str(random.random())
self._mci = _mci()
self._mci.directsend(r'open "%s" alias %s' % (filename, self._alias))
self._mci.directsend('set %s time format milliseconds' % self._alias)
err, buf = self._mci.directsend('status %s length' % self._alias)
self._length_ms = int(buf)
def volume(self, level):
"""Sets the volume between 0 and 100."""
self._mci.directsend('setaudio %s volume to %d' %
(self._alias, level * 10))
def play(self, start_ms=None, end_ms=None):
start_ms = 0 if not start_ms else start_ms
end_ms = self.milliseconds() if not end_ms else end_ms
err, buf = self._mci.directsend('play %s from %d to %d'
% (self._alias, start_ms, end_ms))
def isplaying(self):
return self._mode() == 'playing'
def _mode(self):
err, buf = self._mci.directsend('status %s mode' % self._alias)
return buf
def pause(self):
self._mci.directsend('pause %s' % self._alias)
def unpause(self):
self._mci.directsend('resume %s' % self._alias)
def ispaused(self):
return self._mode() == 'paused'
def stop(self):
self._mci.directsend('stop %s' % self._alias)
self._mci.directsend('seek %s to start' % self._alias)
def milliseconds(self):
return self._length_ms
def __del__(self):
self._mci.directsend('close %s' % self._alias)
_PlatformSpecificAudioClip = _AudioClip
class AudioClip(object):
__slots__ = ['_clip']
def __init__(self, filename):
self._clip = _PlatformSpecificAudioClip(filename)
def play(self, start_ms=None, end_ms=None):
if end_ms is not None and end_ms < start_ms:
return
else:
return self._clip.play(start_ms, end_ms)
def volume(self, level):
assert 0 <= level <= 100
return self._clip.volume(level)
def isplaying(self):
return self._clip.isplaying()
def pause(self):
return self._clip.pause()
def unpause(self):
return self._clip.unpause()
def ispaused(self):
return self._clip.ispaused()
def stop(self):
return self._clip.stop()
def seconds(self):
return int(round(float(self.milliseconds()) / 1000))
def milliseconds(self):
return self._clip.milliseconds()
def load(filename):
"""Return an AudioClip for the given filename."""
return AudioClip(filename)
Related
First of all I'm pretty new to this library and I don't really understand everything. With the help of the internet I managed to get this code snippet working. This code basically plays an audio file(.wav to be specific). The problem is that it only plays once; I want the audio file to loop until I set the is_looping variable to False.
import pyaudio
import wave
class AudioFile:
chunk = 1024
def __init__(self, file_dir):
""" Init audio stream """
self.wf = wave.open(file_dir, 'rb')
self.p = pyaudio.PyAudio()
self.stream = self.p.open(
format=self.p.get_format_from_width(self.wf.getsampwidth()),
channels=self.wf.getnchannels(),
rate=self.wf.getframerate(),
output=True
)
def play(self):
""" Play entire file """
data = self.wf.readframes(self.chunk)
while data != '':
self.stream.write(data)
data = self.wf.readframes(self.chunk)
def close(self):
""" Graceful shutdown """
self.stream.close()
self.p.terminate()
is_looping = True
audio = AudioFile("___.wav")
audio.play()
audio.close()
I tried doing something like this, but it still didn't work:
is_looping = True
audio = AudioFile("___.wav")
while is_looping:
audio.play()
audio.close()
I couldn't find a way to loop the audio using my code, but I found a code in the internet that does exactly what I wanted it to do. Here's the link: https://gist.github.com/THeK3nger/3624478
And here is the code from that link:
import os
import wave
import threading
import sys
# PyAudio Library
import pyaudio
class WavePlayerLoop(threading.Thread):
CHUNK = 1024
def __init__(self, filepath, loop=True):
"""
Initialize `WavePlayerLoop` class.
PARAM:
-- filepath (String) : File Path to wave file.
-- loop (boolean) : True if you want loop playback.
False otherwise.
"""
super(WavePlayerLoop, self).__init__()
self.filepath = os.path.abspath(filepath)
self.loop = loop
def run(self):
# Open Wave File and start play!
wf = wave.open(self.filepath, 'rb')
player = pyaudio.PyAudio()
# Open Output Stream (based on PyAudio tutorial)
stream = player.open(format=player.get_format_from_width(wf.getsampwidth()),
channels=wf.getnchannels(),
rate=wf.getframerate(),
output=True)
# PLAYBACK LOOP
data = wf.readframes(self.CHUNK)
while self.loop:
stream.write(data)
data = wf.readframes(self.CHUNK)
if data == b'': # If file is over then rewind.
wf.rewind()
data = wf.readframes(self.CHUNK)
stream.close()
player.terminate()
def play(self):
"""
Just another name for self.start()
"""
self.start()
def stop(self):
"""
Stop playback.
"""
self.loop = False
You just need to add something like this outside the class and it should work:
player = WavePlayerLoop("sounds/1.wav")
player.play()
I am working on speech interface with python. I am having trouble with audio playback.
What do you use to black back simple mp3 files on the raspberry pi?
I need to play audio and 2 seconds before the end of the playback I need to start another task (opening the stream of the microphone)
How can I archive this? May problem is that I haven't found a way to read the current seconds of the playback yet. If I could read this, I would just start a new thread when the currenttime is audiolength - 2 seconds.
I hope you can help me or have any experience with this.
I found a solution to this.
PyAudio is providing a way to play audio chunk by chunk. Through that you can read the current chunk and compare it to the overall size of the audio.
class AudioPlayer():
"""AudioPlayer class"""
def __init__(self):
self.chunk = 1024
self.audio = pyaudio.PyAudio()
self._running = True
def play(self, audiopath):
self._running = True
#storing how much we have read already
self.chunktotal = 0
wf = wave.open(audiopath, 'rb')
stream = self.audio.open(format =self.audio.get_format_from_width(wf.getsampwidth()),channels = wf.getnchannels(),rate = wf.getframerate(),output = True)
print(wf.getframerate())
# read data (based on the chunk size)
data = wf.readframes(self.chunk)
#THIS IS THE TOTAL LENGTH OF THE AUDIO
audiolength = wf.getnframes() / float(wf.getframerate())
while self._running:
if data != '':
stream.write(data)
self.chunktotal = self.chunktotal + self.chunk
#calculating the percentage
percentage = (self.chunktotal/wf.getnframes())*100
#calculating the current seconds
current_seconds = self.chunktotal/float(wf.getframerate())
data = wf.readframes(self.chunk)
if data == b'':
break
# cleanup stream
stream.close()
def stop(self):
self._running = False
Hope it helps someone,
Alex
Try just_playback. It's a wrapper I wrote around miniaudio that provides playback control functionality like pausing, resuming, seeking, getting the current playback positions and setting the playback volume.
I followed a tutorial on Youtube on how to do TextToSpeech with python, and I am getting the following error
import re
import wave
import pyaudio
import _thread
import time
class TextToSpeech:
CHUNK = 1024
def __init__(self, words_pron_dict:str = 'cmudict-0.7b.txt'):
self._l = {}//Error right here ^
self._load_words(words_pron_dict)
def _load_words(self, words_pron_dict:str):
with open(words_pron_dict, 'r') as file:
for line in file:
if not line.startswith(';;;'):
key, val = line.split(' ',2)
self._l[key] = re.findall(r"[A-Z]+",val)
def get_pronunciation(self, str_input):
list_pron = []
for word in re.findall(r"[\w']+",str_input.upper()):
if word in self._l:
list_pron += self._l[word]
print(list_pron)
delay=0
for pron in list_pron:
_thread.start_new_thread( TextToSpeech._play_audio, (pron,delay,))
delay += 0.145
def _play_audio(sound, delay):
try:
time.sleep(delay)
wf = wave.open("sounds/"+sound+".wav", 'rb')
p = pyaudio.PyAudio()
stream = p.open(format=p.get_format_from_width(wf.getsampwidth()),
channels=wf.getnchannels(),
rate=wf.getframerate(),
output=True)
data = wf.readframes(TextToSpeech.CHUNK)
while data:
stream.write(data)
data = wf.readframes(TextToSpeech.CHUNK)
stream.stop_stream()
stream.close()
p.terminate()
return
except:
pass
if __name__ == '__main__':
tts = TextToSpeech()
while True:
tts.get_pronunciation(input('Enter a word or phrase: '))
The error is "Invalid Sytanx" right where the colon is right before "str" at the top. I'm not sure what I am doing wrong. I am using IDLE for the editor, this script requires pyaudio, which I have installed, and it also requires the document "cmudict-0.7b.text" which I also have.
I've tried copying the name of the file directly to the code, adding parenthesis changing the ' to a " where the txt file name is, to no prevail. I would appreciate it if someone could help me on this and give me some insight on what I'm doing wrong.
I'm using Python 2.7.
Thanks.
I'm trying to do a simple GUI application that has only one button: Record.
You press the button and the recording begins. When you release the button the recording is stopped and the recording is saved.
However, I get the following error when I click the button:
Traceback (most recent call last):
...
data = self.stream.read(self.CHUNK)
File (...), line 608, in read
return pa.read_stream(self._stream, num_frames, exception_on_overflow)
IOError: [Errno -9981] Input overflowed
Exception in Tkinter callback
However I do not have problems with recording a simple audio without the button and Tkinter (the code example they give here).
This is the code:
import Tkinter as tk
import pyaudio, wave
class AppRecording:
def __init__(self, root):
self.root = root
self.mouse_pressed = False
recordingButton = tk.Button(root, text = "Record")
recordingButton.pack()
recordingButton.bind("<ButtonPress-1>", self.OnMouseDown)
recordingButton.bind("<ButtonRelease-1>", self.OnMouseUp)
self.CHUNK = 1024
self.FORMAT = pyaudio.paInt16
self.CHANNELS = 2
self.RATE = 44100
self.WAVE_OUTPUT_FILENAME = "output.wav"
self.p = pyaudio.PyAudio()
try: self.stream = self.p.open(format=self.FORMAT,
channels=self.CHANNELS,
rate=self.RATE,
input=True,
frames_per_buffer=self.CHUNK)
except:
raise Exception("There is no connected microphone. Check that you connect to the left hole if you have a PC.")
return None
self.frames = []
def recordFrame(self):
try:
data = self.stream.read(self.CHUNK)
print "after try"
except IOError as ex:
print "inside except"
if ex[1] != pyaudio.paInputOverflowed:
print "before raise"
raise
print "after raise"
data = '\x00' * self.CHUNK # or however you choose to handle it, e.g. return None
self.frames.append(data)
def finishRecording(self):
self.stream.stop_stream()
self.stream.close()
self.p.terminate()
wf = wave.open(self.WAVE_OUTPUT_FILENAME, 'wb')
wf.setnchannels(self.CHANNELS)
wf.setsampwidth(self.p.get_sample_size(self.FORMAT))
wf.setframerate(self.RATE)
wf.writeframes(b''.join(self.frames))
wf.close()
def OnMouseDown(self, event):
self.mouse_pressed = True
self.poll()
def OnMouseUp(self, event):
self.root.after_cancel(self.after_id)
print "Finished recording!"
self.finishRecording()
def poll(self):
if self.mouse_pressed:
self.recordFrame()
self.after_id = self.root.after(1, self.poll)
root=tk.Tk()
app = AppRecording(root)
root.mainloop()
I tried to change the self.CHUNK and self.RATE. The internal microphone of my iMac says that the rate is 44100. In some places I read that I should change the chunk or rate value, tried both but no one helped. Another place told me to add the except IOError as ex: (...)
PyAudio version: 0.2.10
pyaudio.get_portaudio_version(): 1246720
pyaudio.get_portaudio_version_text(): PortAudio V19.6.0-devel, revision 396fe4b6699ae929d3a685b3ef8a7e97396139a4
Tkinter.__version__: $Revision: 81008 $
I would appreciate your help, thanks!
Which python/tk/portaudio/pyaudio version ?
I confirm that your code is good (no issue) under Ubuntu 14.04 LTS x64 (Python 2.7 with portaudio19-dev and PyAudio-0.2.10) so I assume that issue maybe relative to your python, tk, pyaudio or portaudio version...
Are you sure that you have the last portaudio & tk version installed on your computer ?
I do not understand the example material for pyaudio. It seems they had written an entire small program and it threw me off.
How do I just play a single audio file?
Format is not an issue, I just want to know the bare minimum code I need to play an audio file.
May be this small wrapper (warning: created on knees) of their example will help you to understand the meaning of code they wrote.
import pyaudio
import wave
import sys
class AudioFile:
chunk = 1024
def __init__(self, file):
""" Init audio stream """
self.wf = wave.open(file, 'rb')
self.p = pyaudio.PyAudio()
self.stream = self.p.open(
format = self.p.get_format_from_width(self.wf.getsampwidth()),
channels = self.wf.getnchannels(),
rate = self.wf.getframerate(),
output = True
)
def play(self):
""" Play entire file """
data = self.wf.readframes(self.chunk)
while data != b'':
self.stream.write(data)
data = self.wf.readframes(self.chunk)
def close(self):
""" Graceful shutdown """
self.stream.close()
self.p.terminate()
# Usage example for pyaudio
a = AudioFile("1.wav")
a.play()
a.close()
The example seems pretty clear to me. You simply save the example as playwav.py call:
python playwav.py my_fav_wav.wav
The wave example with some extra comments:
import pyaudio
import wave
import sys
# length of data to read.
chunk = 1024
# validation. If a wave file hasn't been specified, exit.
if len(sys.argv) < 2:
print "Plays a wave file.\n\n" +\
"Usage: %s filename.wav" % sys.argv[0]
sys.exit(-1)
'''
************************************************************************
This is the start of the "minimum needed to read a wave"
************************************************************************
'''
# open the file for reading.
wf = wave.open(sys.argv[1], 'rb')
# create an audio object
p = pyaudio.PyAudio()
# open stream based on the wave object which has been input.
stream = p.open(format =
p.get_format_from_width(wf.getsampwidth()),
channels = wf.getnchannels(),
rate = wf.getframerate(),
output = True)
# read data (based on the chunk size)
data = wf.readframes(chunk)
# play stream (looping from beginning of file to the end)
while data:
# writing to the stream is what *actually* plays the sound.
stream.write(data)
data = wf.readframes(chunk)
# cleanup stuff.
wf.close()
stream.close()
p.terminate()
This way requires ffmpeg for pydub, but can play not only wave files:
import pyaudio
import sys
from pydub import AudioSegment
if len(sys.argv) <= 1:
print('No File Name!')
sys.exit(1)
chunk = 1024
fn = ' '.join(sys.argv[1:])
pd = AudioSegment.from_file(fn)
p = pyaudio.PyAudio()
stream = p.open(format =
p.get_format_from_width(pd.sample_width),
channels = pd.channels,
rate = pd.frame_rate,
output = True)
i = 0
data = pd[:chunk]._data
while data:
stream.write(data)
i += chunk
data = pd[i:i + chunk]._data
stream.close()
p.terminate()
sys.exit(0)