本文整理匯總了Python中pyaudio.PyAudio方法的典型用法代碼示例。如果您正苦於以下問題：Python pyaudio.PyAudio方法的具體用法？Python pyaudio.PyAudio怎麽用？Python pyaudio.PyAudio使用的例子？那麽恭喜您, 這裏精選的方法代碼示例或許可以為您提供幫助。您也可以進一步了解該方法所在模塊pyaudio的用法示例。

在下文中一共展示了pyaudio.PyAudio方法的20個代碼示例，這些例子默認根據受歡迎程度排序。您可以為喜歡或者感覺有用的代碼點讚，您的評價將有助於我們的係統推薦出更棒的Python代碼示例。

示例1: __enter__

​點讚 7

​

# 需要導入模塊: import pyaudio [as 別名]

# 或者: from pyaudio import PyAudio [as 別名]

def __enter__(self):

self._audio_interface = pyaudio.PyAudio()

self._audio_stream = self._audio_interface.open(

# format=pyaudio.paInt16,

format=pyaudio.paFloat32,

# The API currently only supports 1-channel (mono) audio

# https://goo.gl/z757pE

channels=1,

rate=self._rate,

input=True,

frames_per_buffer=self._chunk,

input_device_index=self._device,

# Run the audio stream asynchronously to fill the buffer object.

# This is necessary so that the input device's buffer doesn't

# overflow while the calling thread makes network requests, etc.

stream_callback=self._fill_buffer,

)

self.closed = False

return self

開發者ID:pytorch，項目名稱:audio，代碼行數:23，

示例2: play_audio_file

​點讚 6

​

# 需要導入模塊: import pyaudio [as 別名]

# 或者: from pyaudio import PyAudio [as 別名]

def play_audio_file(fname=DETECT_DING):

"""Simple callback function to play a wave file. By default it plays

a Ding sound.

:param str fname: wave file name

:return: None

"""

ding_wav = wave.open(fname, 'rb')

ding_data = ding_wav.readframes(ding_wav.getnframes())

audio = pyaudio.PyAudio()

stream_out = audio.open(

format=audio.get_format_from_width(ding_wav.getsampwidth()),

channels=ding_wav.getnchannels(),

rate=ding_wav.getframerate(), input=False, output=True)

stream_out.start_stream()

stream_out.write(ding_data)

time.sleep(0.2)

stream_out.stop_stream()

stream_out.close()

audio.terminate()

開發者ID:warchildmd，項目名稱:google-assistant-hotword-raspi，代碼行數:22，

示例3: valid_input_devices

​點讚 6

​

# 需要導入模塊: import pyaudio [as 別名]

# 或者: from pyaudio import PyAudio [as 別名]

def valid_input_devices(self):

"""

See which devices can be opened for microphone input.

call this when no PyAudio object is loaded.

"""

mics=[]

for device in range(self.p.get_device_count()):

if self.valid_test(device):

mics.append(device)

if len(mics)==0:

print("no microphone devices found!")

else:

print("found %d microphone devices: %s"%(len(mics),mics))

return mics

### SETUP AND SHUTDOWN

開發者ID:swharden，項目名稱:Python-GUI-examples，代碼行數:18，

示例4: play_wav

​點讚 6

​

# 需要導入模塊: import pyaudio [as 別名]

# 或者: from pyaudio import PyAudio [as 別名]

def play_wav(fname, chunk=CHUNK):

# create an audio object

wf = wave.open(fname, 'rb')

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 len(data) > 0:

# writing to the stream is what *actually* plays the sound.

stream.write(data)

data = wf.readframes(chunk)

# cleanup stuff

stream.close()

p.terminate()

開發者ID:gigagenie，項目名稱:ai-makers-kit，代碼行數:25，

示例5: play_file

​點讚 6

​

# 需要導入模塊: import pyaudio [as 別名]

# 或者: from pyaudio import PyAudio [as 別名]

def play_file(fname):

# create an audio object

wf = wave.open(fname, 'rb')

p = pyaudio.PyAudio()

chunk = 1024

# 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 len(data) > 0:

# writing to the stream is what *actually* plays the sound.

stream.write(data)

data = wf.readframes(chunk)

# cleanup stuff.

stream.close()

p.terminate()

開發者ID:gigagenie，項目名稱:ai-makers-kit，代碼行數:26，

示例6: __enter__

​點讚 6

​

# 需要導入模塊: import pyaudio [as 別名]

# 或者: from pyaudio import PyAudio [as 別名]

def __enter__(self):

self._audio_interface = pyaudio.PyAudio()

self._audio_stream = self._audio_interface.open(

format=pyaudio.paInt16,

channels=1, rate=self._rate,

input=True, frames_per_buffer=self._chunk,

# Run the audio stream asynchronously to fill the buffer object.

# This is necessary so that the input device's buffer doesn't

# overflow while the calling thread makes network requests, etc.

stream_callback=self._fill_buffer,

)

self.closed = False

return self

#def __exit__(self, type, value, traceback):

開發者ID:gigagenie，項目名稱:ai-makers-kit，代碼行數:19，

示例7: play_file

​點讚 6

​

# 需要導入模塊: import pyaudio [as 別名]

# 或者: from pyaudio import PyAudio [as 別名]

def play_file(fname):

# create an audio object

wf = wave.open(fname, 'rb')

p = pyaudio.PyAudio()

chunk = 1024

# 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 len(data) > 0:

# writing to the stream is what *actually* plays the sound.

stream.write(data)

data = wf.readframes(chunk)

# cleanup stuff.

stream.close()

p.terminate()

開發者ID:gigagenie，項目名稱:ai-makers-kit，代碼行數:26，

示例8: _play_audio

​點讚 6

​

# 需要導入模塊: import pyaudio [as 別名]

# 或者: from pyaudio import PyAudio [as 別名]

def _play_audio(path, delay):

try:

time.sleep(delay)

wf = wave.open(path, '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

開發者ID:junzew，項目名稱:HanTTS，代碼行數:25，

示例9: play

​點讚 6

​

# 需要導入模塊: import pyaudio [as 別名]

# 或者: from pyaudio import PyAudio [as 別名]

def play(self, file_):

wf = wave.open(file_, '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(self.CHUNK)

while data != '':

stream.write(data)

data = wf.readframes(self.CHUNK)

stream.stop_stream()

stream.close()

p.terminate()

開發者ID:namco1992，項目名稱:voicetools，代碼行數:20，

示例10: __init__

​點讚 6

​

# 需要導入模塊: import pyaudio [as 別名]

# 或者: from pyaudio import PyAudio [as 別名]

def __init__(self, config, verbose, logger):

"""Initialize an MQTT client.

Args:

config (:class:`.ServerConfig`): The configuration of

the MQTT client.

verbose (bool): Whether or not the MQTT client runs in verbose

mode.

logger (:class:`logging.Logger`): The Logger object for logging

messages.

"""

self.config = config

self.verbose = verbose

self.logger = logger

self.mqtt = Client()

self.logger.debug('Using %s', pyaudio.get_portaudio_version_text())

self.logger.debug('Creating PyAudio object...')

self.audio = pyaudio.PyAudio()

self.initialize()

self.mqtt.on_connect = self.on_connect

self.mqtt.on_disconnect = self.on_disconnect

self.connect()

開發者ID:koenvervloesem，項目名稱:hermes-audio-server，代碼行數:26，

示例11: __init__

​點讚 6

​

# 需要導入模塊: import pyaudio [as 別名]

# 或者: from pyaudio import PyAudio [as 別名]

def __init__(self):

self.interrupted = False

self.detector = None

rpack = RosPack()

# UMDL or PMDL file paths along with audio files

pkg_path = rpack.get_path('dialogflow_ros')

self.model_path = pkg_path + '/scripts/snowboy/resources/jarvis.umdl'

ding_path = pkg_path + '/scripts/snowboy/resources/ding.wav'

# Setup df

self.df_client = None

# Setup audio output

ding = wave.open(ding_path, 'rb')

self.ding_data = ding.readframes(ding.getnframes())

self.audio = pyaudio.PyAudio()

self.stream_out = self.audio.open(

format=self.audio.get_format_from_width(ding.getsampwidth()),

channels=ding.getnchannels(), rate=ding.getframerate(),

input=False, output=True)

self.last_contexts = []

rospy.loginfo("HOTWORD_CLIENT: Ready!")

開發者ID:piraka9011，項目名稱:dialogflow_ros，代碼行數:22，

示例12: __init__

​點讚 6

​

# 需要導入模塊: import pyaudio [as 別名]

# 或者: from pyaudio import PyAudio [as 別名]

def __init__(self):

# Audio stream input setup

FORMAT = pyaudio.paInt16

CHANNELS = 1

RATE = 16000

self.CHUNK = 4096

self.audio = pyaudio.PyAudio()

self.stream = self.audio.open(format=FORMAT, channels=CHANNELS,

rate=RATE, input=True,

frames_per_buffer=self.CHUNK,

stream_callback=self.get_data)

self._buff = Queue.Queue() # Buffer to hold audio data

self.closed = False

# ROS Text Publisher

self.text_pub = rospy.Publisher('/google_client/text', String, queue_size=10)

# Context clues in yaml file

rospack = rospkg.RosPack()

yamlFileDir = rospack.get_path('dialogflow_ros') + '/config/context.yaml'

with open(yamlFileDir, 'r') as f:

self.context = yaml.load(f)

開發者ID:piraka9011，項目名稱:dialogflow_ros，代碼行數:24，

示例13: __init__

​點讚 6

​

# 需要導入模塊: import pyaudio [as 別名]

# 或者: from pyaudio import PyAudio [as 別名]

def __init__(self):

FORMAT = pyaudio.paInt16

CHANNELS = 1

RATE = 16000

CHUNK = 4096

self.audio = pyaudio.PyAudio()

self.stream = self.audio.open(format=FORMAT, channels=CHANNELS, rate=RATE,

input=True, frames_per_buffer=CHUNK,

stream_callback=self._callback)

self.serversocket = socket.socket(socket.AF_INET, socket.SOCK_STREAM)

self.read_list = [self.serversocket]

self._server_name = rospy.get_param('/dialogflow_client/server_name',

'127.0.0.1')

self._port = rospy.get_param('/dialogflow_client/port', 4444)

rospy.loginfo("DF_CLIENT: Audio Server Started!")

開發者ID:piraka9011，項目名稱:dialogflow_ros，代碼行數:19，

示例14: _record

​點讚 6

​

# 需要導入模塊: import pyaudio [as 別名]

# 或者: from pyaudio import PyAudio [as 別名]

def _record(self):

# Start recording audio on the current thread until stop() is

# called.

p = pyaudio.PyAudio()

channels, rate = self.config.CHANNELS, self.config.RATE

frames_per_buffer = self.config.FRAMES_PER_BUFFER

pa_format = pyaudio.get_format_from_width(self.config.SAMPLE_WIDTH)

stream = p.open(input=True, format=pa_format, channels=channels,

rate=rate, frames_per_buffer=frames_per_buffer)

# Start recognising in a loop

stream.start_stream()

while self._recording:

with self._condition:

self._buffers.append(stream.read(frames_per_buffer))

# Notify waiting threads (if any).

self._condition.notifyAll()

# This improves the performance; we don't need to process as

# much audio as the device can read.

time.sleep(self.read_interval)

stream.close()

p.terminate()

開發者ID:dictation-toolbox，項目名稱:dragonfly，代碼行數:27，

示例15: _get_pa_instance

​點讚 6

​

# 需要導入模塊: import pyaudio [as 別名]

# 或者: from pyaudio import PyAudio [as 別名]

def _get_pa_instance():

# Suppress initial ALSA messages if using ALSA.

# Got this from: https://stackoverflow.com/a/17673011/12157649

try:

asound = cdll.LoadLibrary('libasound.so')

c_error_handler = ERROR_HANDLER_FUNC(

lambda filename, line, function, err, fmt: None

)

asound.snd_lib_error_set_handler(c_error_handler)

except:

# We'll most likely get here if the Port Audio host API isn't ALSA.

asound = None

# Create the pa instance.

pa = pyaudio.PyAudio()

# If necessary, restore the original error handler.

if asound:

asound.snd_lib_error_set_handler(None)

return pa

開發者ID:dictation-toolbox，項目名稱:dragonfly，代碼行數:22，

示例16: __init__

​點讚 6

​

# 需要導入模塊: import pyaudio [as 別名]

# 或者: from pyaudio import PyAudio [as 別名]

def __init__(self, speaker, passive_stt_engine, active_stt_engine):

"""

Initiates the pocketsphinx instance.

Arguments:

speaker -- handles platform-independent audio output

passive_stt_engine -- performs STT while Jasper is in passive listen

mode

acive_stt_engine -- performs STT while Jasper is in active listen mode

"""

self._logger = logging.getLogger(__name__)

self.speaker = speaker

self.passive_stt_engine = passive_stt_engine

self.active_stt_engine = active_stt_engine

self._logger.info("Initializing PyAudio. ALSA/Jack error messages " +

"that pop up during this process are normal and " +

"can usually be safely ignored.")

self._audio = pyaudio.PyAudio()

self._logger.info("Initialization of PyAudio completed.")

開發者ID:jjwang，項目名稱:laibot-client，代碼行數:21，

示例17: __init__

​點讚 6

​

# 需要導入模塊: import pyaudio [as 別名]

# 或者: from pyaudio import PyAudio [as 別名]

def __init__(self, model):

import speech_recognition # lazy loading

import pyaudio

device_count = pyaudio.PyAudio().get_device_count() - 1

print("\ndevice_count: " + str(device_count))

self._r = speech_recognition.Recognizer()

try:

mics = speech_recognition.Microphone.list_microphone_names()

print("mics: " + str(mics))

index = mics.index(registered_mic)

except:

index = -1

print("index: " + str(index))

try:

if index == -1:

self._m = speech_recognition.Microphone()

else:

self._m = speech_recognition.Microphone(device_index=index)

except:

self._m = None

print("SpeechRecognizer_Common, no mic detected!")

self._model = model

開發者ID:richmondu，項目名稱:libfaceid，代碼行數:24，

示例18: run

​點讚 6

​

# 需要導入模塊: import pyaudio [as 別名]

# 或者: from pyaudio import PyAudio [as 別名]

def run(self):

self.logger.debug("Start to recording...")

self.logger.debug(" Time = %s"%self.time)

self.logger.debug(" Sample Rate = %s"%self.sr)

self.start_time = time.time()

pa=PyAudio()

stream=pa.open(format = paInt16,channels=1, rate=self.sr,input=True, frames_per_buffer=self.frames_per_buffer)

my_buf=[]

count=0

if self.time is None:

total_count = 1e10

else:

total_count = self.time * self.sr / self.batch_num

while count< total_count and self.__running.isSet():

datawav = stream.read(self.batch_num, exception_on_overflow = True)

datause = np.fromstring(datawav,dtype = np.short)

for w in datause:

self.buffer.put(w)

count+=1

stream.close()

開發者ID:mhy12345，項目名稱:rcaudio，代碼行數:22，

示例19: __init__

​點讚 6

​

# 需要導入模塊: import pyaudio [as 別名]

# 或者: from pyaudio import PyAudio [as 別名]

def __init__(self):

self.open = True

self.rate = 44100

self.frames_per_buffer = 1024

self.channels = 2

self.format = pyaudio.paInt16

self.audio_filename = "temp_audio.wav"

self.audio = pyaudio.PyAudio()

self.stream = self.audio.open(format=self.format,

channels=self.channels,

rate=self.rate,

input=True,

frames_per_buffer = self.frames_per_buffer)

self.audio_frames = []

# Audio starts being recorded

開發者ID:JRodrigoF，項目名稱:AVrecordeR，代碼行數:20，

示例20: __enter__

​點讚 6

​

# 需要導入模塊: import pyaudio [as 別名]

# 或者: from pyaudio import PyAudio [as 別名]

def __enter__(self):

with SuperManager.getInstance().commons.shutUpAlsaFFS():

self._audio_interface = pyaudio.PyAudio()

self._audio_stream = self._audio_interface.open(

format=pyaudio.paInt16,

# The API currently only supports 1-channel (mono) audio

# https://goo.gl/z757pE

channels=1, rate=self._rate,

input=True, frames_per_buffer=self._chunk,

# Run the audio stream asynchronously to fill the buffer object.

# This is necessary so that the input device's buffer doesn't

# overflow while the calling thread makes network requests, etc.

stream_callback=self._fill_buffer,

)

self.closed = False

return self

開發者ID:project-alice-assistant，項目名稱:ProjectAlice，代碼行數:21，

注：本文中的pyaudio.PyAudio方法示例整理自Github/MSDocs等源碼及文檔管理平台，相關代碼片段篩選自各路編程大神貢獻的開源項目，源碼版權歸原作者所有，傳播和使用請參考對應項目的License；未經允許，請勿轉載。