def play(frames):
p=PyAudio()
stream=p.open(format=p.get_format_from_width(1),channels=1,rate=10000,output=True)
stream.write(frames)
stream.stop_stream()
stream.close()
p.terminate()
def Audio_play(filepath):
'''
play audio
'''
CHUNK = 1024
wf = wave.open(filepath, 'rb')
pa = PyAudio()
default_output = pa.get_default_host_api_info().get('defaultOutputDevice')
stream =pa.open(format = pa.get_format_from_width(wf.getsampwidth()),
channels = wf.getnchannels(),
rate = wf.getframerate(),
output = True,
output_device_index = default_output)
NUM = int(wf.getframerate()/CHUNK * 15)
logging.info(">> START TO PLAY AUDIO")
while NUM:
data = wf.readframes(CHUNK)
if data == " ": break
stream.write(data)
NUM -= 1
stream.stop_stream()
stream.close()
del data
pa.terminate()
def run(self): #定义每个线程要运行的函数
chunk = 2014
wf = wave.open(self.prams, 'rb')
p = PyAudio()
stream = p.open(format=p.get_format_from_width(wf.getsampwidth()),
channels=wf.getnchannels(),
rate=wf.getframerate(),
output=True)
print("开始播放")
while True:
data = wf.readframes(chunk)
if data == b"":
break
stream.write(data)
#print("正在读呢")
print("播放完毕")
stream.close()
p.terminate()
wf.close()
# main1函数是把录音转化为文字 如果是双击的话，那么把录音读出来，
# 如果是机器说话的话，把他说的文件删掉！以免过多
if self.type == 0:
main1(self.prams, 0)
else:
os.remove(self.prams)
def play(frequency, length):
print("Playing impulse at %sHz" % frequency);
frequency = float(frequency);
BITRATE = 16000
FREQUENCY = frequency
LENGTH = length
NUMBEROFFRAMES = int(BITRATE * LENGTH)
RESTFRAMES = NUMBEROFFRAMES % BITRATE
WAVEDATA = ''
for x in range(NUMBEROFFRAMES):
WAVEDATA += chr(int(math.sin(x / ((BITRATE / FREQUENCY) / math.pi)) * 127 + 128))
for x in range(RESTFRAMES):
WAVEDATA += chr(128)
p = PyAudio()
stream = p.open(
format=p.get_format_from_width(1),
channels=1,
rate=BITRATE,
output=True,
)
stream.write(WAVEDATA)
stream.stop_stream()
stream.close()
p.terminate()
class pybeeptone:
def __init__(self, rate=44100):
self.rate = 44100
self.pyaudio = PyAudio()
self.stream = self.pyaudio.open(
format = self.pyaudio.get_format_from_width(1),
channels = 1, rate = self.rate, output = True)
def play_tone(self, freq=1000, duration=0.3):
rate = self.rate
length = int(math.ceil(self.rate*duration))
data = ''.join( [chr(int(math.sin(x/((rate/freq)/math.pi))*127+128))
for x in xrange(length)] )
self.stream.write(data)
def play_rest(self, duration):
rate = self.rate
length = int(math.ceil(self.rate*duration))
data = ''.join( [chr(int(128)) for x in xrange(length)] )
self.stream.write(data)
def close(self):
self.stream.stop_stream()
self.stream.close()
self.pyaudio.terminate()
def Play_sound(self, File_name="resources/speak.wav"):
try:
wf = wave.open(File_name, 'rb')
except:
try:
os.system("mpg123 " + File_name) #这里的mpg123是一个程序插件，可以用来播放ＭＰ３文件
print("*" * 5, "播放成功！播放格式非wav！播放了本地文件！")
return True
except:
print("*" * 5, "播放失败！没有此目录或播放格式非wav ")
return False
p = PyAudio()
stream = p.open(format=p.get_format_from_width(wf.getsampwidth()),
channels=wf.getnchannels(),
rate=wf.getframerate(),
output=True)
# 读数据
data = wf.readframes(self.CHUNK)
print("*" * 5, "正在播放", "*" * 5)
# 播放流
while len(data) > 0:
stream.write(data)
data = wf.readframes(self.CHUNK)
print(".")
stream.stop_stream() # 暂停播放/录制
stream.close() # 终止播放
p.terminate() # 终止portaudio会话
return True
def sine_tone(frequency, duration, volume=1, sample_rate=22050):
n_samples = int(sample_rate * duration)
restframes = n_samples % sample_rate
p = PyAudio()
stream = p.open(format=p.get_format_from_width(2), # 16 bit
channels=2,
rate=sample_rate,
output=True)
for i in xrange(0, 10):
if i % 2 == 0:
frequency = ZERO_FREQUENCY
else:
frequency = ONE_FREQUENCY
s = lambda t: volume * math.sin(2 * math.pi * frequency * t / sample_rate)
samples = (int(s(t) * 0x7f + 0x80) for t in xrange(n_samples))
for buf in izip(*[samples]*sample_rate): # write several samples at a time
stream.write(bytes(bytearray(buf)))
# fill remainder of frameset with silence
stream.write(b'\x80' * restframes)
stream.stop_stream()
stream.close()
p.terminate()
def play():
wf = wave.open(filename, 'rb')
p = PyAudio()
stream = p.open(format=p.get_format_from_width(wf.getsampwidth()),
channels=wf.getnchannels(),
rate=wf.getframerate(),
output=True)
data = wf.readframes(chunk)
print('Playing your record')
while len(data) > 0:
stream.write(data)
data = wf.readframes(chunk)
stream.stop_stream()
stream.close()
p.terminate()
# gender classifier
if deploy_gender(filename) == 1:
print('You are man')
else:
print('You are woman')
# age classifier
age_list = ['teens', 'twenties', 'thirties', 'fourties']
age = deploy_age(filename)
print('Your age:', age_list[age])
def sine_tone(frequencies, amplitudes, duration, volume=1.0, sample_rate=22050):
n_samples = int(sample_rate * duration)
restframes = n_samples % sample_rate
p = PyAudio()
stream = p.open(format=p.get_format_from_width(1), # 8bit
channels=1, # mono
rate=sample_rate,
output=True)
def s(t):
r = 0
for i in range(0, len(frequencies)):
r += volume * amplitudes[i] * math.sin(2 * math.pi * frequencies[i] * t / sample_rate)
return r
samples = (int(s(t) * 0x7f + 0x80) for t in range(n_samples))
for buf in zip(*[samples]*sample_rate): # write several samples at a time
stream.write(bytes(bytearray(buf)))
# fill remainder of frameset with silence
stream.write(b'\x80' * restframes)
stream.stop_stream()
stream.close()
p.terminate()
def play(): wavName = 'test.wav' print "play %s" % (wavName) wf = wave.open(wavName, 'rb') pa = PyAudio() stream = pa.open(format=pa.get_format_from_width(wf.getsampwidth()), channels=wf.getnchannels(), rate=wf.getframerate(), output=True) data = wf.readframes(CHUNK) td = threading.Thread(target=startGame) td.start() while data != '': stream.write(data) data = wf.readframes(CHUNK) audio_data = np.fromstring(data, dtype=np.short) print data stream.stop_stream() stream.close() pa.terminate()
def play_voice(audio_file):
chunk = 1024 # 2014kb
wf = wave.open(audio_file, 'rb')
p = PyAudio()
stream = p.open(format=p.get_format_from_width(wf.getsampwidth()),
channels=wf.getnchannels(),
rate=wf.getframerate(),
output=True)
data = wf.readframes(chunk) # 读取数据
while True:
data = wf.readframes(chunk)
if signal1.is_set():
# 点击结束后这个变为true
break
if data == "":
break
stream.write(data)
stream.stop_stream() # 停止数据流
stream.close()
p.terminate() # 关闭 PyAudio
print('play函数结束！')
def _playback(self):
"""
Playback function for thread. Plays a segment until running is set to
false or end is reached
"""
# open stream
p = PyAudio()
stream = p.open(
format=p.get_format_from_width(self._segment.sample_width),
channels=self._segment.channels,
rate=self._segment.frame_rate,
output=True,
output_device_index=self._audio_device,
)
# output the file in millisecond steps
while self._pos < len(self._segment) and self._running:
stream.write(self._segment[self._pos]._data)
self._pos += 1
# close stream
time.sleep(stream.get_output_latency())
stream.stop_stream()
stream.close()
p.terminate()
def get_microphone_working(
self,
device_name: str,
device_index: int,
audio: pyaudio.PyAudio,
chunk_size: int = 1024,
) -> bool:
"""Record some audio from a microphone and check its energy."""
try:
# read audio
pyaudio_stream = audio.open(
input_device_index=device_index,
channels=self.channels,
format=audio.get_format_from_width(self.sample_width),
rate=self.sample_rate,
input=True,
)
try:
audio_data = pyaudio_stream.read(chunk_size)
if not pyaudio_stream.is_stopped():
pyaudio_stream.stop_stream()
finally:
pyaudio_stream.close()
debiased_energy = AudioSummary.get_debiased_energy(audio_data)
# probably actually audio
return debiased_energy > 30
except Exception:
_LOGGER.exception("get_microphone_working ({device_name})")
pass
return False
def run(self): # 定义每个线程要运行的函数
chunk = 2014
try:
wf = wave.open(self.prams, 'rb')
p = PyAudio()
stream = p.open(format=p.get_format_from_width(wf.getsampwidth()),
channels=wf.getnchannels(),
rate=wf.getframerate(),
output=True)
# print("开始播放")
except FileNotFoundError:
print("录音文件未找到！")
return
while True:
data = wf.readframes(chunk)
if data == b"":
break
stream.write(data)
# print("正在读呢")
# print("播放完毕")
stream.close()
p.terminate()
wf.close()
# main1函数是把录音转化为文字
# 1 代表提示音 不处理
# 2 代表精灵的录音 听完直接删除掉 以免过多
if self.typ == 1:
pass
elif self.typ == 2:
os.remove(self.prams)
def ps(lng, fq):
# See http://en.wikipedia.org/wiki/Bit_rate#Audio
BITRATE = 16000 #number of frames per second/frameset.
#See http://www.phy.mtu.edu/~suits/notefreqs.html
FREQUENCY = fq #Hz, waves per second, 261.63=C4-note.
LENGTH = lng #seconds to play sound
NUMBEROFFRAMES = int(BITRATE * LENGTH)
RESTFRAMES = NUMBEROFFRAMES % BITRATE
WAVEDATA = ''
for x in xrange(NUMBEROFFRAMES):
WAVEDATA += chr(
int(math.sin(x / ((BITRATE / FREQUENCY) / math.pi)) * 127 + 128))
#fill remainder of frameset with silence
for x in xrange(RESTFRAMES):
WAVEDATA += chr(128)
p = PyAudio()
stream = p.open(
format=p.get_format_from_width(1),
channels=1,
rate=BITRATE,
output=True,
)
stream.write(WAVEDATA)
stream.stop_stream()
stream.close()
p.terminate()
def main():
# read in some block data from pyaudio
RATE=44100
INPUT_BLOCK_TIME=0.2
INPUT_FRAMES_PER_BLOCK=int(RATE*INPUT_BLOCK_TIME)
pa=PyAudio()
data=True
fmt="%dh"%INPUT_FRAMES_PER_BLOCK
total_rms=0
total_blocks=0
while data:
for dr,subdr,fnames in os.walk(path):
for filename in fnames:
try:
print filename
wf=wave.open("%s/%s"%(path,filename),'rb')
strm=pa.open(format=pa.get_format_from_width(wf.getsampwidth()),
channels=wf.getnchannels(),
rate=wf.getframerate(),
input=True)
strm.stop_stream()
strm.close()
d=wf.readframes(INPUT_FRAMES_PER_BLOCK)
d=struct.unpack(fmt,d)
wf.close()
total_rms+=calc_rms(d)
total_blocks+=1
except:
#print e
print "*** ERROR ***"
data=False
avg=total_rms/total_blocks
print "The average is %f"%avg
class Stream(Thread):
def __init__(self, f, on_terminated):
self.__active = True
self.__path = f
self.__paused = True
self.on_terminated = on_terminated
self.__position = 0
self.__chunks = []
self.__pyaudio = PyAudio()
Thread.__init__(self)
self.start()
def play(self):
self.__paused = False
def seek(self, seconds):
self.__position = int(seconds * 10)
def is_playing(self):
return self.__active and not self.__paused
def get_position(self):
return int(self.__position / 10)
def get_duration(self):
return int(len(self.__chunks) / 10)
def pause(self):
self.__paused = True
def kill(self):
self.__active = False
def __get_stream(self):
self.__segment = AudioSegment.from_file(self.__path)
self.__chunks = make_chunks(self.__segment, 100)
return self.__pyaudio.open(format=self.__pyaudio.get_format_from_width(self.__segment.sample_width),
channels=self.__segment.channels,
rate=self.__segment.frame_rate,
output=True)
def run(self):
stream = self.__get_stream()
while self.__position < len(self.__chunks):
if not self.__active:
break
if not self.__paused:
# noinspection PyProtectedMember
data = self.__chunks[self.__position]._data
self.__position += 1
else:
free = stream.get_write_available()
data = chr(0) * free
stream.write(data)
stream.stop_stream()
self.__pyaudio.terminate()
if self.__active:
self.on_terminated()
def worker():
p = PyAudio()
stream = p.open(format=p.get_format_from_width(2),
channels=1, rate=44100, output=True)
while True:
self.lock.acquire()
stream.write(self.wavdata.tostring())
self.lock.release()
def play(filepath):
fp = open(filepath, 'rb')
data = fp.read()
fp.close()
aud = BytesIO(data)
sound = AudioSegment.from_file(aud, format='mp3') # LONG
global isPlaying
p = PyAudio()
stream = p.open(format=p.get_format_from_width(sound.sample_width),
channels=sound.channels,
rate=sound.frame_rate,
output=True)
print("[PLAYAUDIO INTERVAL]%f" % (time.time()))
datas = sound.raw_data
isPlaying = True
stream.write(datas) # ------------这里，接住！
'''
来来来兄弟们，来看看网上大神的代码↓（稍加修改，但是思想融会贯通）
# framewidth = sound.frame_width
# i = 0
# while True:
# data = datas[i*framewidth:(i+1)*framewidth]
# if data == b"":
# break
# i += 1
# stream.write(data)
这个人教我一帧一帧的去写，我一想好像没什么毛病。
当然后果就是CPU必须时刻保持有一定空闲，让出来给这一线程。
总之就是说CPU必须不能跑高了，稍微高一点就爆音。
然而我TM发现这东西可以一次全写进去！(如上一句话↑)
让我们看看官方的注释：
def write(self, frames, num_frames=None,
exception_on_underflow=False):
"""
Write samples to the stream. Do not call when using
*non-blocking* mode.
:param frames:#音频的数据帧
The frames of data.
:param num_frames:#帧的数量
The number of frames to write.
Defaults to None, in which this value will be
automatically computed.#你如果不输就是None，系统自动给你判断
结果网上的代码让我一帧一帧些，他每次都会判断一次帧的数量为1。
求CPU心理阴影面积。
不过感谢这位老哥指导我用了pydub这个库。
改完之后妈妈再也不用担心我音频卡顿了，哪怕我主线程写个while True：pass也不怕了！
'''
stream.stop_stream() # 停止数据流
stream.close()
p.terminate() # 关闭 PyAudio
def play_audio(self):
audio = PyAudio()
stream = audio.open(format=audio.get_format_from_width(self.sampwidth),
channels=self.nchannels,
rate=self.rate,
output=True)
stream.write(self.response)
stream.stop_stream()
stream.close()
audio.terminate()
def playWaveData(self, waveData):
p = PyAudio()
stream = p.open(format = p.get_format_from_width(1),
channels = 1,
rate = self.bitRate,
output = True)
stream.write(waveData)
stream.stop_stream()
stream.close()
p.terminate()
def play(self,filename):
#os.popen(self.locpla+filename )
wf=wave.open(self.locpla+filename,'rb')
p=PyAudio()
stream=p.open(format=p.get_format_from_width(wf.getsampwidth()),channels=wf.getnchannels(),rate=wf.getframerate(),output=True)
data = wf.readframes(self.chunk)
while data != b'':
stream.write(data)
data = wf.readframes(self.chunk)
stream.close()
p.terminate()
def play():
wf=wave.open(r"C:\Users\pengfy\PycharmProjects\smart-audio\record_files\2019-07-30-11-52-51.wav",'rb')
p=PyAudio()
stream=p.open(format=p.get_format_from_width(wf.getsampwidth()),channels=
wf.getnchannels(),rate=wf.getframerate(),output=True)
while True:
data=wf.readframes(chunk)
if data=="":break
stream.write(data)
stream.close()
p.terminate()
def play():
wf=wave.open(file,'r')
p=PyAudio()
stream=p.open(format=p.get_format_from_width(wf.getsampwidth()),channels=
wf.getnchannels(),rate=wf.getframerate(),output=True)
data=wf.readframes(8000)
while len(data)>0:
stream.write(data)
data=wf.readframes(8000)
stream.close()
p.terminate()
def play():
wf=wave.open(r"01.wav",'rb')
p=PyAudio()
stream=p.open(format=p.get_format_from_width(wf.getsampwidth()),channels=
wf.getnchannels(),rate=wf.getframerate(),output=True)
while True:
data=wf.readframes(chunk)
if data=="":break
stream.write(data)
stream.close()
p.terminate()
def play(self,event):
wf=wave.open(self.name,'rb')
p=PyAudio()
stream=p.open(format=p.get_format_from_width(wf.getsampwidth()),channels=
wf.getnchannels(),rate=wf.getframerate(),output=True)
while True:
data=wf.readframes(1024)
if not data:
break
stream.write(data)
stream.close()
p.terminate()
def play():
wf = wave.open(r"01.wav", 'rb') #以二进制的格式打开
p = PyAudio() #创建一个对象
stream = p.open(format=p.get_format_from_width(wf.getsampwidth()),
channels=wf.getnchannels(),
rate=wf.getframerate(),
output=True) #这里是output
while True:
data = wf.readframes(chunk) #进行读取
if data == "": break
stream.write(data) #进行输出
stream.close() #关闭输出流
p.terminate() # 关闭文件的输入流
def play():
wf=wave.open(SOUND_FILE,'rb')
p=PyAudio()
stream=p.open(format=p.get_format_from_width(wf.getsampwidth()),channels=
wf.getnchannels(),rate=wf.getframerate(),output=True)
data = wf.readframes(CHUNK)
while data != '' or data != " ":
stream.write(data)
data = wf.readframes(CHUNK)
stream.start_stream()
stream.close()
p.terminate()
def pyaudio_stream(sample_rate=22050):
p = PyAudio()
stream = p.open(
format=p.get_format_from_width(1), # 8bit
channels=1, # mono
rate=sample_rate,
output=True,
)
yield stream
stream.stop_stream()
stream.close()
p.terminate()
class PySine(object):
BITRATE = 96000.
def __init__(self):
self.pyaudio = PyAudio()
try:
self.stream = self.pyaudio.open(
format=self.pyaudio.get_format_from_width(1),
channels=1,
rate=int(self.BITRATE),
output=True)
except:
logger.error("No audio output is available. Mocking audio stream to simulate one...")
# output stream simulation with magicmock
try:
from mock import MagicMock
except: # python > 3.3
from unittest.mock import MagicMock
from time import sleep
self.stream = MagicMock()
def write(array):
duration = len(array)/float(self.BITRATE)
sleep(duration)
self.stream.write = write
def __del__(self):
self.stream.stop_stream()
self.stream.close()
self.pyaudio.terminate()
def sine(self, frequency=440.0, duration=1.0):
"""
This will generate a sine wave with minimum of 1 second duration.
"""
grain = round(self.BITRATE / frequency)
points = grain * round(self.BITRATE * duration / grain)
duration = points / self.BITRATE
data = np.zeros(int(self.BITRATE * max(duration, 1.0)))
try:
times = np.linspace(0, duration, points, endpoint=False)
data[:points] = np.sin(times * frequency * 2 * np.pi)
data = np.array((data + 1.0) * 127.5, dtype=np.int8).tostring()
except: # do it without numpy
data = ''
omega = 2.0*pi*frequency/self.BITRATE
for i in range(points):
data += chr(int(127.5*(1.0+sin(float(i)*omega))))
self.stream.write(data)
def play(file):
wf = wave.open(file, 'rb')
pa = PyAudio()
stream = pa.open(format=pa.get_format_from_width(wf.getsampwidth()),
channels=wf.getnchannels(),
rate=wf.getframerate(),
output=True)
while True:
data = wf.readframes(1024)
if data == '':
break
stream.write(data)
stream.close()
pa.terminate()
def out_audio(data):
from pyaudio import PyAudio
if isinstance(data, np.ndarray):
data = list(data.astype(int) + 0x80)
p = PyAudio()
stream = p.open(
format=p.get_format_from_width(1), # 8bit
channels=1, # mono
rate=Player.sample_rate,
output=True)
stream.write(bytes(data))
stream.stop_stream()
stream.close()
p.terminate()
class Voice2Midi(object):
def __init__(self):
self._synth = Synth()
self._onset_detector = OnsetDetector()
self._f0_detector = F0Detector()
self._wf = wave.open('audio_files/vocal2.wav', 'rb')
self._p = PyAudio()
self._e = threading.Event()
self._input_buffer = []
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,
frames_per_buffer=WINDOW_SIZE,
stream_callback=self._process_frame)
def run(self):
print(self._wf.getframerate())
self._synth.run() #main loop of the synth
self._stream.stop_stream()
self._stream.close()
self._p.terminate()
def _process_frame(self, in_data, frame_count, time_info, status_flag):
data = self._wf.readframes(frame_count)
data_array = np.frombuffer(data, dtype=np.int32)
if np.shape(data_array)[0] == WINDOW_SIZE and np.shape(
np.nonzero(data_array))[1] > 0:
# print(np.shape(np.nonzero(data_array))[1])
data_array = data_array.astype('float32')
data_array = data_array / np.max(data_array)
# onset = self._onset_detector.find_onset(data_array)
if self._onset_detector.find_onset(data_array):
freq0 = self._f0_detector.find_f0(data_array)
if freq0:
# Onset detected
print("Note detected; fundamental frequency: ", freq0)
midi_note_value = int(hz_to_midi(freq0)[0])
print("Midi note value: ", midi_note_value)
note = RTNote(midi_note_value, 100, 0.5)
self._synth.set_new_note(note)
self._synth.e.set()
return data, paContinue
else:
print("Sending over")
return data, paComplete
class AudioStream(object):
def __init__(self, sample_rate=44100, channels=1, width=2, chunk=1024,
input_device_index=None):
self.sample_rate = sample_rate
self.channels = channels
self.width = width
self.chunk = chunk
self.input_device_index = input_device_index
def __enter__(self):
self._pa = PyAudio()
if self.input_device_index is None:
self.input_device_index = \
self._pa.get_default_input_device_info()['index']
self._stream = self._pa.open(
format=self._pa.get_format_from_width(self.width),
channels=self.channels,
rate=self.sample_rate,
input=True,
frames_per_buffer=self.chunk,
input_device_index=self.input_device_index)
self._stream.start_stream()
return self
def read(self):
''' On a buffer overflow this returns 0 bytes. '''
try:
return self._stream.read(self.chunk)
except IOError:
return ''
except AttributeError:
raise Exception('Must be used as a context manager.')
def stream(self):
try:
while True:
bytes = self.read()
if bytes:
self.handle(bytes)
except (KeyboardInterrupt, SystemExit):
pass
def __exit__(self, type, value, traceback):
self._stream.stop_stream()
self._stream.close()
self._pa.terminate()
def handle(self, bytes):
pass
def play_wave_data(self):
p = PyAudio()
stream = p.open(
format=p.get_format_from_width(1),
channels=self.CHANNELS,
rate=self.BITRATE,
output=True,
)
stream.write(self.WAVE_DATA)
stream.stop_stream()
stream.close()
p.terminate()
def openWav(self):
chunk = 1024
wf = wave.open(r"result.wav", 'rb')
p = PyAudio()
stream = p.open(format = p.get_format_from_width(wf.getsampwidth()), channels = wf.getnchannels(), rate = wf.getframerate(), output = True)
while True:
data = wf.readframes(chunk)
if data == "":break
stream.write(data)
stream.close()
p.terminate()
def play(filename):
wf = wave.open(filename, 'rb')
p = PyAudio()
stream = p.open(format=p.get_format_from_width(wf.getsampwidth()),
channels=wf.getnchannels(),
rate=wf.getframerate(),
output=True)
while True:
data = wf.readframes(NUM_SAMPLES)
if data == b'':
break
stream.write(data)
stream.close()
p.terminate()
def play(wave_data):
chunk_size = BITRATE/10
p = PyAudio()
stream = p.open(format = p.get_format_from_width(1),
channels = 1,
rate = BITRATE,
output = True)
for chunk in itertools.islice(wave_data, chunk_size):
stream.write(chunk)
stream.stop_stream()
stream.close()
p.terminate()
def go():
filepath = "/home/pi/tmp/talkynotes/" + "resources/audio/" + file + ".wav"
wf = wave.open(filepath, "rb")
chunk = 1024
p = PyAudio()
stream = p.open(format=p.get_format_from_width(wf.getsampwidth()),
channels=wf.getnchannels(),
rate=wf.getframerate(),
output=True)
data = wf.readframes(chunk)
while data != b'':
stream.write(data)
data = wf.readframes(chunk)
stream.close()
p.terminate()
def play(wave_data):
chunk_size = BITRATE / 10
p = PyAudio()
stream = p.open(format=p.get_format_from_width(1),
channels=1,
rate=BITRATE,
output=True)
for chunk in itertools.islice(wave_data, chunk_size):
stream.write(chunk)
stream.stop_stream()
stream.close()
p.terminate()
def read_file():
wf = wave.open(filename, 'rb')
p = PyAudio()
stream = p.open(format = p.get_format_from_width(wf.getsampwidth()),
channels = wf.getnchannels(),rate = wf.getframerate(),output = True,)
time.sleep(2)
print('开始放音')
while True:
data = wf.readframes(chunk)
if data == b'':break
stream.write(data)
print('...')
wf.close()
stream.close()
p.terminate()
def Audio_play():
#play audio
CHUNK =8192
filename = current_directory + '/audio_lib/source1.wav'
wf = wave.open(filename, 'rb')
pa = PyAudio()
stream =pa.open(format = pa.get_format_from_width(wf.getsampwidth()), channels = wf.getnchannels(), rate = wf.getframerate(), output = True)
NUM = int(wf.getframerate()/CHUNK * 15)
logging.info("-> START PLAY AUDIO")
while NUM:
data = wf.readframes(CHUNK)
if data == " ": break
stream.write(data)
NUM -= 1
stream.stop_stream()
stream.close()
logging.info(">> PLAY DONE")
def getWavFeature(wavName): print "processing %s" % (wavName) wf = wave.open(wavName, 'rb') pa = PyAudio() stream = pa.open(format=pa.get_format_from_width(wf.getsampwidth()), channels=wf.getnchannels(), rate=wf.getframerate(), output=True) data = wf.readframes(CHUNK) while data != '': stream.write(data) data = wf.readframes(CHUNK) stream.stop_stream() stream.close() pa.terminate()
def play(wavName): print "play %s" % (wavName) wf = wave.open(wavName, 'rb') pa = PyAudio() stream = pa.open(format=pa.get_format_from_width(wf.getsampwidth()), channels=wf.getnchannels(), rate=wf.getframerate(), output=True) # td = threading.Thread(target=startGame) # td.start() data = wf.readframes(CHUNK) while data != '': stream.write(data) data = wf.readframes(CHUNK) stream.stop_stream() stream.close() pa.terminate()
def button_play_Click():
global state, wav_out, stream_out, data_out, wave_form, wavefile_name, script_count, script_list
state = 2 # change the state flag into "playing"
# update button states
button_stop.configure(state = NORMAL)
button_record.configure(state = DISABLED)
button_back.configure(state = DISABLED)
button_next.configure(state = DISABLED)
button_play.configure(state = DISABLED)
wav_out = PyAudio()
wave_form = wave.open(wavefile_name, 'rb')
stream_out = wav_out.open(format = wav_out.get_format_from_width(wave_form.getsampwidth()),
channels = wave_form.getnchannels(),
rate = wave_form.getframerate(),
output = True)
data_out = wave_form.readframes(CHUNK)
echo_text.configure(text = "Playing...", bg = 'blue', fg = 'white', font = ("Helvetica", 50))
play_wave()
def Play_WAV(self, filename):
chunk = 1024
f = wave.open(filename,"rb")
pa = PyAudio()
# open stream
stream = pa.open(format = pa.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
pa.terminate()
data = waves[instrument][pitch].readframes(frame_count)
raw = fromstring(data, dtype=int16)
# print(raw)
gainlvl = vectorize(lambda x: x * volume)
raw = gainlvl(raw)
data = raw.astype(int16).tostring()
if len(data) < 2048 * swidth : # If file is over then rewind.
waves[instrument][pitch].rewind()
data = waves[instrument][pitch].readframes(frame_count)
data_pos = waves[instrument][pitch].tell()
return (data, paContinue)
print "Hey, Parent Process, Opening Stream..."
stream = pa.open(
format = pa.get_format_from_width(waves[instrument][1].getsampwidth()),
channels = waves[instrument][1].getnchannels(),
rate = waves[instrument][1].getframerate(),
output = True,
stream_callback = loopaudio)
print "Hey, Parent Process1"
stream.start_stream()
print "Hey, Parent Process2"
while stream.is_active():
#print "Hey, Parent Process3"
data=r.readline()
if not data: break
str_vals = data.strip().split()
int_list = [int(i) for i in str_vals]
sc=[1,0,2,3,0,2,4,5,5,6,7,5,5,6,8,5,5,6,7,5,9,7,5,5,6,7,5,5,6,8,5,5,6,7,5,9,10,0,2,3,0,2,4,0,2,3,0,2,4,5,5,6,7,5,5,6,8,5,5,6,7,5,9,11,12,13,12,13,14,11,12,13,12,13,14,13,12,13,12,13,14,13,12,13,12,15]
gtN=[[g],[-k,-22,-20,-19],[-22,-20,-k],[-22,-20,-k,g,-13,-15],[-k,-20,-k,g,-k,-20,-k],[-k,-k,-15,g,-15,-13],[g,-15,-13,g,-k,-15],[g,-15,-13,g,-k,-15],[-12,-10,-12,-10,-12,-10],[-21,-19,-17]]
gtD=[[96],[8,4,2,2],[4,4,8],[4,4,4,2,1,1],[c]+[b]*6,[c,b,b,c,b,b],[c,b,b,c,b,b],[a]*6,[b]*6,[4,4,8]]
gtL=[[96],[8,4,2,2],[4,4,8],gtD[3],gtD[4],gtD[5],gtD[6],gtD[7],[b,c,b,c,b,b],[4,4,8]]
gt=[0,1,1]+[2]*7+[3,4,4,4,5,4,4,4,6,4,4,4,5,4,4,4,7]+[1]*4+[2]*4+[9]*4+[8]*8
sg["bd"]=[bd,bdN,bdD,bdL]
sg["mn"]=[mn,mnN,mnD,mnL]
sg["cr"]=[cr,crN,crD,crL]
sg["rc"]=[rc,rcN,rcD,rcL]
sg["bs"]=[bs,bsN,bsD,bsL]
sg["sy"]=[sy,syN,syD,syL]
sg["sc"]=[sc,scN,scD,scL]
sg["gt"]=[gt,gtN,gtD,gtL]
z=[[] for i in X(8)]
p=PyAudio()
st=p.open(format=p.get_format_from_width(1),channels=1,rate=R,output=True)
os.system("clear")
for i in X(R):sD.append(math.sin(tp*(float(i)/R)))
tP=0
for tk in sg:
A=z[tP].append
for pt in sg[tk][0]:
pP=nP=0
for n in sg[tk][1][pt]:
ss=int(qL*sg[tk][3][pt][nP]*R)
du=int(qL*sg[tk][2][pt][nP]*R)
if n!=g:
F=440*pow(1.059463094,n)
for x in X(int(du)):
if n==g:A(0.)
else:
## LOADING GLOBAL VARIABLES
# Pre-Loading Music data
names = ['500', '505', '510', '515', '520', '525', '530', '535', '540', '545', '550']
sounds = map(lambda i: wave.open("stimuli/" + names[i] + ".wav"), range(0,len(names)))
data = map(lambda x: x.readframes(x.getnframes()),sounds)
# Pre-loading yes/no data
ynnames = ['no_f', 'yes_m']
ynsounds = map(lambda i: wave.open("stimuli_yesno/" + ynnames[i] + ".wav"), range(0,len(ynnames)))
yndata = map(lambda x: x.readframes(x.getnframes()),ynsounds)
# Opening Audio Stream
print("tone width " + str(sounds[0].getsampwidth()) + " framerate " + str(sounds[0]._framerate) + " nch " + str(sounds[0].getnchannels()));
print("yesno width " + str(ynsounds[0].getsampwidth()) + " framerate " + str(ynsounds[0]._framerate) + " nch " + str(ynsounds[0].getnchannels()));
stream = p.open(format=p.get_format_from_width(sounds[0].getsampwidth()),
channels=sounds[0].getnchannels(),
rate=sounds[0].getframerate(),
output=True)
# set up the colors
BLACK = (0, 0, 0)
WHITE = (255, 255, 255)
# set up fonts
basicFont = pygame.font.SysFont(None, 48)
basicBigFont = pygame.font.SysFont(None, 48*2)
updateframe(["Welcome to the BCI Music Experiment"])
#
if options.link is not None:
config.set("messages", "encrypted_link", options.link)
if options.section is not None:
config.set("messages", "encrypted_section", options.section)
msg_decode = options.msgdecode
espeak_voice = options.voice
if espeak_voice is not None:
import struct
from pyaudio import PyAudio
import espeak
rate = espeak.initialize(espeak.AUDIO_OUTPUT_SYNCHRONOUS)
espeak.set_voice(espeak_voice)
pa = PyAudio()
snd = pa.open(format=pa.get_format_from_width(2), channels=1,
rate=rate, output=True)
snd.start_stream()
def synth_cb(samples, nsamples, events):
result = bytes()
for sample in samples[:nsamples]:
result += struct.pack("=h", sample)
snd.write(result)
return 0
espeak.set_synth_callback(synth_cb)
jid = config.get("xmpp", "jid")
try:
password = config.get("xmpp", "password")
except:
password = getpass("Password: ")
class AudioTool:
'''
This function include record and play, if you want to play and record,
please set the play is True.
The sample rate is 44100
Bit:16
'''
def __init__(self):
self.chunk = 1024
self.channels = 2
self.samplerate = 44100
self.format = paInt16
#open audio stream
self.pa = PyAudio()
self.save_buffer = []
def record_play(self,seconds,play=False,file_play_path=None,file_save_path=None):
NUM = int((self.samplerate/float(self.chunk)) * seconds)
if play is True:
swf = wave.open(file_play_path, 'rb')
stream = self.pa.open(
format = self.format,
channels = self.channels,
rate = self.samplerate,
input = True,
output = play,
frames_per_buffer = self.chunk
)
# wave_data = []
while NUM:
data = stream.read(self.chunk)
self.save_buffer.append(data)
wave_data=np.fromstring(data, dtype = np.short)
wave_data.shape = -1,2
wave_data = wave_data.T #transpose multiprocessing.Process
# print int(data)
print wave_data
NUM -= 1
if play is True:
data = swf.readframes(self.chunk)
stream.write(data)
if data == " ": break
if play is True:
swf.close()
#stop stream
stream.stop_stream()
stream.close()
# save wav file
def _save_wave_file(filename,data):
wf_save = wave.open(filename, 'wb')
wf_save.setnchannels(self.channels)
wf_save.setsampwidth(self.pa.get_sample_size(self.format))
wf_save.setframerate(self.samplerate)
wf_save.writeframes("".join(data))
wf_save.close()
_save_wave_file(file_save_path, self.save_buffer)
del self.save_buffer[:]
print file_save_path," Record Sucessful!"
def play(self,filepath):
wf = wave.open(filepath, 'rb')
stream =self.pa.open(
format = self.pa.get_format_from_width(wf.getsampwidth()),
channels = wf.getnchannels(),
rate = wf.getframerate(),
output = True,
)
NUM = int(wf.getframerate()/self.chunk * 15)
print "playing.."
while NUM:
data = wf.readframes(self.chunk)
if data == " ": break
stream.write(data)
NUM -= 1
stream.stop_stream()
del data
stream.close()
def close(self):
self.pa.terminate()
#plot.add_img_plot(mag_vec, yok, linspace(0, len(anr)-1, len(anr)))
#zplot.add_img_plot(zname="blah", zdata=mag_vec)#z, ydata=linspace(0, len(anr)-1, len(anr)), xdata=linspace(0, len(yok)-1, len(yok)))
#plot.add_plot("cross_sec", yname="Macvec1", ydata=c)
# plot.add_plot("cross_se2", yname="Macvec2", ydata=mag_vec[:, 75])
plot.show()
for x in xrange(NUMBEROFFRAMES):
WAVEDATA = WAVEDATA+chr(c[x])
#fill remainder of frameset with silence
for x in xrange(RESTFRAMES):
WAVEDATA = WAVEDATA+chr(128)
p = PyAudio()
FORMAT=p.get_format_from_width(1)
stream = p.open(format = p.get_format_from_width(1),
channels = 1,
rate = BITRATE,
output = True)
stream.write(WAVEDATA)
stream.stop_stream()
stream.close()
p.terminate()
if 0:
import wave
wf = wave.open('short_pulse.wav', 'wb')
wf.setnchannels(1)
wf.setsampwidth(p.get_sample_size(FORMAT))
wf.setframerate(BITRATE)
class InterfaceAport(Thread):
def __init__(self, bitrate):
Thread.__init__(self)
# Stream parameters
self.bitrate_ = bitrate
self.channels_ = 1
# create an audio object
self.p_ = PyAudio()
self.stream_ = None
self.createStream()
self.q = Queue.Queue()
self.lastq = [None,0]
self.play_ = True
self.stop_ = False
def __del__(self):
self.p_.terminate()
self.stream_.close()
@property
def bitrate(self):
return self.bitrate_
@bitrate.setter
def bitrate(self, x):
self.bitrate_=x
self.createStream()
@property
def channels(self):
return self.channels_
@channels.setter
def channels(self, x):
self.channels_=x
self.createStream()
def createStream(self):
if self.stream_ != None:
self.stream_.close()
self.stream_ = self.p_.open(
format = self.p_.get_format_from_width(1),
channels = self.channels_,
rate = self.bitrate_,
output = True)
def playData(self, dataIn, repetitions=None):
# Encodage
data=''
for x in xrange(int(len(dataIn))):
data+=chr(int(dataIn[x]*127+128))
self.q.put([data,repetitions])
def run(self):
while (self.stop_ == False): # stop = True on arrete tout
if (self.q.qsize()!=0 and self.play_ == True): # Si il a a à jouer et que cest pas pause
self.lastq = self.q.get()
if self.lastq[1] == None: # Si un nombre de repetition nest pas None
while self.lastq[1] == None and self.q.qsize()==0:
self.stream_.write(self.lastq[0]) # on joue a linfini
else :
for x in xrange(int(self.lastq[1])):
self.stream_.write(self.lastq[0])
def stopLast(self,repetitions=0):
self.lastq[1] = repetitions
def play(self):
self.play_ = True
self.stop_ = False
def pause(self):
self.play_ = False
self.stop_ = False
def stop(self):
self.play_ = False
self.stop_ = True
xrange = range
volume=0.1
sample_rate = 22050
duration = 0.1
print(sample_rate*duration)
# Set the root
root = 37
n_samples = int(sample_rate*duration)
restframes = n_samples % sample_rate
p = PyAudio()
stream = p.open(format=p.get_format_from_width(1), # 8bit
channels=1, # mono
rate = sample_rate,
output = True)
for i in range(0, length):
if melody[i] == 1:
play = 0
if melody[i] == 2:
play = 2
if melody[i] == 3:
play = 4
if melody[i] == 4:
play = 5
if melody[i] == 5:
play = 7
gtN=[[g],[-k,-22,-20,-19],[-22,-20,-k],[-22,-20,-k,g,-13,-15],[-k,-20,-k,g,-k,-20,-k],[-k,-k,-15,g,-15,-13],[g,-15,-13,g,-k,-15],[g,-15,-13,g,-k,-15],[-12,-10,-12,-10,-12,-10],[-21,-19,-17]]
gtD=[[96],[8,4,2,2],[4,4,8],[4,4,4,2,1,1],[c,b,b,b,b,b,b],[c,b,b,c,b,b],[c,b,b,c,b,b],[a,a,a,a,a,a],[b,b,b,b,b,b],[4,4,8]]
gtL=[[96],[8,4,2,2],[4,4,8],gtD[3],gtD[4],gtD[5],gtD[6],gtD[7],[b,c,b,c,b,b],[4,4,8]]
gt=[0,1,1,2,2,2,2,2,2,2,3,4,4,4,5,4,4,4,6,4,4,4,5,4,4,4,7,1,1,1,1,2,2,2,2,9,9,9,9,8,8,8,8,8,8,8,8]
sg["bd"]=[bd,bdN,bdD,bdL]
sg["mn"]=[mn,mnN,mnD,mnL]
sg["cr"]=[cr,crN,crD,crL]
sg["rc"]=[rc,rcN,rcD,rcL]
sg["bs"]=[bs,bsN,bsD,bsL]
sg["sy"]=[sy,syN,syD,syL]
sg["sc"]=[sc,scN,scD,scL]
sg["gt"]=[gt,gtN,gtD,gtL]
z=[[] for i in X(8)]
p=PyAudio()
st=p.open(format =
p.get_format_from_width(1),
channels=1,
rate=R,
output=True)
os.system("clear")
for i in X(R):
sD.append(math.sin(tp*(float(i)/R)))
tP=0
for tk in sg:
A=z[tP].append
for pt in sg[tk][0]:
pP=0
nP=0
for n in sg[tk][1][pt]:
ss=int(qL*sg[tk][3][pt][nP]*R);
du=int(qL*sg[tk][2][pt][nP]*R);
import wave
import sys
from pyaudio import PyAudio
CHUNK = 1024
wf = wave.open(sys.argv[1], 'rb')
p = PyAudio()
stream = p.open(format=p.get_format_from_width(wf.getsampwidth()), channels=wf.getnchannels(), rate=wf.getframerate(),
output=True)
data = wf.readframes(CHUNK)
while data != '':
stream.write(data)
data = wf.readframes(CHUNK)
stream.stop_stream()
stream.close()
p.terminate()
