Python pyaudio.paInt16() Examples
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code examples of pyaudio.paInt16().
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Example #1
Source File: proj2_yt_mvp.py From ai-makers-kit with MIT License | 13 votes |
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):
Example #2
Source File: vad.py From audio with BSD 2-Clause "Simplified" License | 11 votes |
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
Example #3
Source File: google_client.py From dialogflow_ros with MIT License | 9 votes |
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)
Example #4
Source File: smart_trash_can.py From ai-makers-kit with MIT License | 7 votes |
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):
Example #5
Source File: proj3_capital_game.py From ai-makers-kit with MIT License | 7 votes |
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):
Example #6
Source File: transcribe_streaming_mic.py From python-docs-samples with Apache License 2.0 | 6 votes |
def __enter__(self): 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
Example #7
Source File: microphone.py From audio-reactive-led-strip with MIT License | 6 votes |
def start_stream(callback): p = pyaudio.PyAudio() frames_per_buffer = int(config.MIC_RATE / config.FPS) stream = p.open(format=pyaudio.paInt16, channels=1, rate=config.MIC_RATE, input=True, frames_per_buffer=frames_per_buffer) overflows = 0 prev_ovf_time = time.time() while True: try: y = np.fromstring(stream.read(frames_per_buffer, exception_on_overflow=False), dtype=np.int16) y = y.astype(np.float32) stream.read(stream.get_read_available(), exception_on_overflow=False) callback(y) except IOError: overflows += 1 if time.time() > prev_ovf_time + 1: prev_ovf_time = time.time() print('Audio buffer has overflowed {} times'.format(overflows)) stream.stop_stream() stream.close() p.terminate()
Example #8
Source File: recorder.py From gif-music-visualizer with GNU General Public License v3.0 | 6 votes |
def setup(self): """initialize sound card.""" #TODO - windows detection vs. alsa or something for linux #TODO - try/except for sound card selection/initiation self.buffersToRecord=int(self.RATE*self.secToRecord/self.BUFFERSIZE) if self.buffersToRecord==0: self.buffersToRecord=1 self.samplesToRecord=int(self.BUFFERSIZE*self.buffersToRecord) self.chunksToRecord=int(self.samplesToRecord/self.BUFFERSIZE) self.secPerPoint=1.0/self.RATE self.p = pyaudio.PyAudio() self.inStream = self.p.open(format=pyaudio.paInt16,channels=1,rate=self.RATE,input=True,frames_per_buffer=self.BUFFERSIZE) self.xsBuffer=numpy.arange(self.BUFFERSIZE)*self.secPerPoint self.xs=numpy.arange(self.chunksToRecord*self.BUFFERSIZE)*self.secPerPoint self.audio=numpy.empty((self.chunksToRecord*self.BUFFERSIZE),dtype=numpy.int16)
Example #9
Source File: audio.py From Oe2sSLE with GNU General Public License v2.0 | 6 votes |
def __init__(self, data, fmt): if fmt.formatTag != RIFF.WAVE_fmt_.WAVE_FORMAT_PCM: raise Exception() if fmt.samplesPerSec < 1000 or fmt.samplesPerSec > 192000: data, fmt = wav_tools.wav_resample_preview(data, fmt, 1000, 192000) self.data = data self.fmt = fmt self._offset = 0 def callback(indata, frames, time, status): n_bytes=frames*fmt.blockAlign to_read=min(n_bytes, len(self.data) - self._offset) outdata = self.data[self._offset:self._offset+to_read] self._offset += to_read return (outdata,pa.paContinue) self.stream = audio.open(format=pa.paInt16, channels=fmt.channels, rate=fmt.samplesPerSec, output=True, stream_callback=callback)
Example #10
Source File: proj2_yt_mvp.py From ai-makers-kit with MIT License | 6 votes |
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):
Example #11
Source File: audiostream.py From rtmonoaudio2midi with GNU General Public License v3.0 | 6 votes |
def run(self): pya = PyAudio() self._stream = pya.open( format=paInt16, channels=1, rate=SAMPLE_RATE, input=True, frames_per_buffer=WINDOW_SIZE, stream_callback=self._process_frame, ) self._stream.start_stream() while self._stream.is_active() and not raw_input(): time.sleep(0.1) self._stream.stop_stream() self._stream.close() pya.terminate()
Example #12
Source File: smart_trash_can.py From ai-makers-kit with MIT License | 6 votes |
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):
Example #13
Source File: main.py From real-time-plot-microphone-kivy with MIT License | 6 votes |
def get_microphone_level(): """ source: http://stackoverflow.com/questions/26478315/getting-volume-levels-from-pyaudio-for-use-in-arduino audioop.max alternative to audioop.rms """ chunk = 1024 FORMAT = pyaudio.paInt16 CHANNELS = 1 RATE = 44100 p = pyaudio.PyAudio() s = p.open(format=FORMAT, channels=CHANNELS, rate=RATE, input=True, frames_per_buffer=chunk) global levels while True: data = s.read(chunk) mx = audioop.rms(data, 2) if len(levels) >= 100: levels = [] levels.append(mx)
Example #14
Source File: mic_client.py From dialogflow_ros with MIT License | 6 votes |
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 text_topic = rospy.get_param('/text_topic', '/dialogflow_text') self.text_pub = rospy.Publisher(text_topic, String, queue_size=10)
Example #15
Source File: cross_record.py From nyumaya_audio_recognition with Apache License 2.0 | 6 votes |
def run(self): self.mic_stream = self.p.open(format=pyaudio.paInt16, channels=self.channels, rate=self.sample_rate, input=True, frames_per_buffer=self.blocksize) self.running = True while self.mic_stream and self.running: input_data = self.mic_stream.read(self.blocksize) if input_data: self.audio_buffer.write(input_data) #Shutdown record command if(self.mic_stream): self.mic_stream.close() self.mic_stream = None
Example #16
Source File: demo_client.py From ZASR_tensorflow with Apache License 2.0 | 6 votes |
def main(): # prepare audio recorder p = pyaudio.PyAudio() stream = p.open( format=pyaudio.paInt16, channels=1, rate=16000, input=True, stream_callback=callback) stream.start_stream() # prepare keyboard listener with keyboard.Listener( on_press=on_press, on_release=on_release) as listener: listener.join() # close up stream.stop_stream() stream.close() p.terminate()
Example #17
Source File: offline_ser.py From LIVE_SER with Apache License 2.0 | 6 votes |
def predict_file(dec, pyaudio, path, frames, args, rate = 16000, format = pyaudio.paInt16, save = False): wf = wave.open(path, 'wb') wf.setnchannels(1) wf.setsampwidth(pyaudio.get_sample_size(format)) wf.setframerate(rate) #this code works for only for pulseaudio #wf.writeframes(b''.join(frames)) wf.writeframes(frames) wf.close() results = dec.predict_file(path, feat_mode = args.feat_mode, feat_dim = args.feat_dim, three_d = args.three_d) if save == False: os.remove(path) if args.predict_mode == 0: task_outputs = dec.returnDiff(results) elif args.predict_mode == 1: task_outputs = dec.returnLabel(results) else: task_outputs = dec.returnClassDist(results) return task_outputs #main loop for speech emotion recognition
Example #18
Source File: AVrecordeR.py From AVrecordeR with GNU General Public License v2.0 | 6 votes |
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
Example #19
Source File: core_recorder.py From rcaudio with MIT License | 6 votes |
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()
Example #20
Source File: core_recorder.py From rcaudio with MIT License | 6 votes |
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()
Example #21
Source File: audio_server.py From dialogflow_ros with MIT License | 6 votes |
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!")
Example #22
Source File: main.py From Systematic-LEDs with MIT License | 6 votes |
def startStream(self): self.stream = py_audio.open(format = pyaudio.paInt16, channels = 1, rate = self.device_rate, input = True, input_device_index = self.device_id, frames_per_buffer = self.frames_per_buffer) # overflows = 0 # prev_ovf_time = time.time() while True: try: y = np.fromstring(self.stream.read(self.frames_per_buffer), dtype=np.int16) y = y.astype(np.float32) self.callback_func(y) except IOError: pass # overflows += 1 # if time.time() > prev_ovf_time + 1: # prev_ovf_time = time.time() # if config.settings["configuration"]["USE_GUI"]: # gui.label_error.setText('Audio buffer has overflowed {} times'.format(overflows)) # else: # print('Audio buffer has overflowed {} times'.format(overflows))
Example #23
Source File: audio-gate-level.py From silvius with BSD 2-Clause "Simplified" License | 5 votes |
def run_test(self): import pyaudio import audioop pa = pyaudio.PyAudio() sample_rate = self.byterate stream = None while stream is None: try: # try adjusting this if you want fewer network packets self.chunk = 2048 * 2 * sample_rate / self.byterate mic = self.mic if mic == -1: mic = pa.get_default_input_device_info()['index'] print >> sys.stderr, "Selecting default mic" print >> sys.stderr, "Using mic #", mic stream = pa.open( rate = sample_rate, format = pyaudio.paInt16, channels = 1, input = True, input_device_index = mic, frames_per_buffer = self.chunk) except IOError, e: if(e.errno == -9997 or e.errno == 'Invalid sample rate'): new_sample_rate = int(pa.get_device_info_by_index(mic)['defaultSampleRate']) if(sample_rate != new_sample_rate): sample_rate = new_sample_rate continue print >> sys.stderr, "\n", e print >> sys.stderr, "\nCould not open microphone. Please try a different device." global fatal_error fatal_error = True sys.exit(0)
Example #24
Source File: view_with_band_pass_filter.py From tdoa with Apache License 2.0 | 5 votes |
def start(self, quit_event=None, show=None): stream = self.pyaudio_instance.open( rate=RATE, frames_per_buffer=FRAMES, format=pyaudio.paInt16, channels=2, input=True, # output_device_index=1, stream_callback=self._callback) self.event.clear() if not quit_event: quit_event = threading.Event() phat = [0] * (2 * direction_n + 1) while not (quit_event.is_set() or self.event.is_set()): try: data = self.queue.get() buf = np.fromstring(data, dtype='int16') tau, cc = gcc_phat(buf[0::2] * window, buf[1::2] * window, fs=RATE, max_tau=max_tau, interp=1) theta = math.asin(tau / max_tau) * 180 / math.pi print('\ntheta: {}'.format(int(theta))) for i, v in enumerate(cc): phat[i] = int(v * 512) if show: show(phat) # print [l for l in level] except KeyboardInterrupt: break stream.close()
Example #25
Source File: __init__.py From PyBaiduYuyin with MIT License | 5 votes |
def __init__(self, device_index = None): assert device_index is None or isinstance(device_index, int), "Device index must be None or an integer" if device_index is not None: # ensure device index is in range audio = pyaudio.PyAudio(); count = audio.get_device_count(); audio.terminate() # obtain device count assert 0 <= device_index < count, "Device index out of range" self.device_index = device_index self.format = pyaudio.paInt16 # 16-bit int sampling self.SAMPLE_WIDTH = pyaudio.get_sample_size(self.format) self.RATE = 16000 # sampling rate in Hertz self.CHANNELS = 1 # mono audio self.CHUNK = 1024 # number of frames stored in each buffer self.audio = None self.stream = None
Example #26
Source File: mic.py From silvius with BSD 2-Clause "Simplified" License | 5 votes |
def opened(self): import pyaudio import audioop pa = pyaudio.PyAudio() sample_rate = self.byterate stream = None while stream is None: try: # try adjusting this if you want fewer network packets self.chunk = 2048 * 2 * sample_rate / self.byterate mic = self.mic if mic == -1: mic = pa.get_default_input_device_info()['index'] print >> sys.stderr, "Selecting default mic" print >> sys.stderr, "Using mic #", mic stream = pa.open( rate = sample_rate, format = pyaudio.paInt16, channels = 1, input = True, input_device_index = mic, frames_per_buffer = self.chunk) except IOError, e: if(e.errno == -9997 or e.errno == 'Invalid sample rate'): new_sample_rate = int(pa.get_device_info_by_index(mic)['defaultSampleRate']) if(sample_rate != new_sample_rate): sample_rate = new_sample_rate continue print >> sys.stderr, "\n", e print >> sys.stderr, "\nCould not open microphone. Please try a different device." global fatal_error fatal_error = True sys.exit(0)
Example #27
Source File: translate_from_mic.py From python-docs-samples with Apache License 2.0 | 5 votes |
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
Example #28
Source File: mic_array.py From mic_array with Apache License 2.0 | 5 votes |
def __init__(self, rate=16000, channels=8, chunk_size=None): self.pyaudio_instance = pyaudio.PyAudio() self.queue = Queue.Queue() self.quit_event = threading.Event() self.channels = channels self.sample_rate = rate self.chunk_size = chunk_size if chunk_size else rate / 100 device_index = None for i in range(self.pyaudio_instance.get_device_count()): dev = self.pyaudio_instance.get_device_info_by_index(i) name = dev['name'].encode('utf-8') print(i, name, dev['maxInputChannels'], dev['maxOutputChannels']) if dev['maxInputChannels'] == self.channels: print('Use {}'.format(name)) device_index = i break if device_index is None: raise Exception('can not find input device with {} channel(s)'.format(self.channels)) self.stream = self.pyaudio_instance.open( input=True, start=False, format=pyaudio.paInt16, channels=self.channels, rate=int(self.sample_rate), frames_per_buffer=int(self.chunk_size), stream_callback=self._callback, input_device_index=device_index, )
Example #29
Source File: alexa_audio_device_pyaduio.py From AlexaDevice with MIT License | 5 votes |
def __init__(self): self.pa = pyaudio.PyAudio() self.in_stream = self.pa.open(format=pyaudio.paInt16, channels=1, rate=16000, input=True) self.in_stream.start_stream() self.out_stream = self.pa.open(format=pyaudio.paInt16, channels=1, rate=16000, output=True) self.out_stream.start_stream()
Example #30
Source File: system_eq.py From BiblioPixelAnimations with MIT License | 5 votes |
def __init__(self, rate=8000, chunksize=128): self.rate = rate self.chunksize = chunksize self.p = pyaudio.PyAudio() self.stream = self.p.open(format=pyaudio.paInt16, channels=1, rate=self.rate, input=True, frames_per_buffer=self.chunksize, stream_callback=self.new_frame) self.lock = threading.Lock() self.stop = False self.frames = []