Python sounddevice.InputStream方法代碼示例

# 需要導入模塊: import sounddevice [as 別名]
# 或者: from sounddevice import InputStream [as 別名]
def record_buffer(buffer, **kwargs):
    loop = asyncio.get_event_loop()
    event = asyncio.Event()
    idx = 0

    def callback(indata, frame_count, time_info, status):
        nonlocal idx
        if status:
            print(status)
        remainder = len(buffer) - idx
        if remainder == 0:
            loop.call_soon_threadsafe(event.set)
            raise sd.CallbackStop
        indata = indata[:remainder]
        buffer[idx:idx + len(indata)] = indata
        idx += len(indata)

    stream = sd.InputStream(callback=callback, dtype=buffer.dtype,
                            channels=buffer.shape[1], **kwargs)
    with stream:
        await event.wait() 

# 需要導入模塊: import sounddevice [as 別名]
# 或者: from sounddevice import InputStream [as 別名]
def __init__(self,
                 input_device: Optional[Union[int, str]] = None,
                 hotword: Optional[str] = None,
                 hotwords: Optional[List[str]] = None,
                 conversation_timeout: Optional[float] = 10.0,
                 block_duration: float = 1.0):
        """
        :param input_device: PortAudio device index or name that will be used for recording speech (default: default
            system audio input device).
        :param hotword: When this word is detected, the plugin will trigger a
            :class:`platypush.message.event.stt.HotwordDetectedEvent` instead of a
            :class:`platypush.message.event.stt.SpeechDetectedEvent` event. You can use these events for hooking other
            assistants.
        :param hotwords: Use a list of hotwords instead of a single one.
        :param conversation_timeout: If ``hotword`` or ``hotwords`` are set and ``conversation_timeout`` is set,
            the next speech detected event will trigger a :class:`platypush.message.event.stt.ConversationDetectedEvent`
            instead of a :class:`platypush.message.event.stt.SpeechDetectedEvent` event. You can hook custom hooks
            here to run any logic depending on the detected speech - it can emulate a kind of
            "OK, Google. Turn on the lights" interaction without using an external assistant (default: 10 seconds).
        :param block_duration: Duration of the acquired audio blocks (default: 1 second).
        """

        super().__init__()
        self.input_device = input_device
        self.conversation_timeout = conversation_timeout
        self.block_duration = block_duration

        self.hotwords = set(hotwords or [])
        if hotword:
            self.hotwords = {hotword}

        self._conversation_event = threading.Event()
        self._input_stream: Optional[sd.InputStream] = None
        self._recording_thread: Optional[threading.Thread] = None
        self._detection_thread: Optional[threading.Thread] = None
        self._audio_queue: Optional[queue.Queue] = None
        self._current_text = '' 

# 需要導入模塊: import sounddevice [as 別名]
# 或者: from sounddevice import InputStream [as 別名]
def recording_thread(self, block_duration: Optional[float] = None, block_size: Optional[int] = None,
                         input_device: Optional[str] = None) -> None:
        """
        Recording thread. It reads raw frames from the audio device and dispatches them to ``detection_thread``.

        :param block_duration: Audio blocks duration. Specify either ``block_duration`` or ``block_size``.
        :param block_size: Size of the audio blocks. Specify either ``block_duration`` or ``block_size``.
        :param input_device: Input device
        """
        assert (block_duration or block_size) and not (block_duration and block_size), \
            'Please specify either block_duration or block_size'

        if not block_size:
            block_size = int(self.rate * self.channels * block_duration)

        self.before_recording()
        self.logger.debug('Recording thread started')
        device = self._get_input_device(input_device)
        self._input_stream = sd.InputStream(samplerate=self.rate, device=device,
                                            channels=self.channels, dtype='int16', latency=0,
                                            blocksize=block_size)
        self._input_stream.start()
        self.on_recording_started()
        get_bus().post(SpeechDetectionStartedEvent())

        while self._input_stream:
            try:
                frames = self._input_stream.read(block_size)[0]
            except Exception as e:
                self.logger.warning('Error while reading from the audio input: {}'.format(str(e)))
                continue

            self._audio_queue.put(frames)

        get_bus().post(SpeechDetectionStoppedEvent())
        self.on_recording_ended()
        self.logger.debug('Recording thread terminated') 

# 需要導入模塊: import sounddevice [as 別名]
# 或者: from sounddevice import InputStream [as 別名]
def create_stream(self, device=None):
        if self.stream is not None:
            self.stream.close()
        self.stream = sd.InputStream(
            device=device, channels=1, callback=self.audio_callback)
        self.stream.start() 

# 需要導入模塊: import sounddevice [as 別名]
# 或者: from sounddevice import InputStream [as 別名]
def inputstream_generator(channels=1, **kwargs):
    """Generator that yields blocks of input data as NumPy arrays."""
    q_in = asyncio.Queue()
    loop = asyncio.get_event_loop()

    def callback(indata, frame_count, time_info, status):
        loop.call_soon_threadsafe(q_in.put_nowait, (indata.copy(), status))

    stream = sd.InputStream(callback=callback, channels=channels, **kwargs)
    with stream:
        while True:
            indata, status = await q_in.get()
            yield indata, status 

# 需要導入模塊: import sounddevice [as 別名]
# 或者: from sounddevice import InputStream [as 別名]
def __init__(self):
        self.stream = sd.InputStream(
            samplerate=SAMPLE_RATE,
            blocksize=BLOCK_SIZE,
            channels=1,
            callback=self.process_frames
        )
        self.stream.start()

        self.is_held = True
        self.times_pressed = 0 

# 需要導入模塊: import sounddevice [as 別名]
# 或者: from sounddevice import InputStream [as 別名]
def test_sounddevice_lib():
    import time

    import numpy as np
    from sounddevice import InputStream, OutputStream, sleep as sd_sleep
    """ 
    if no portaudio installed:
    Traceback (most recent call last):
  File "TestSoundCard.py", line 42, in <module>
    test_sounddevice_lib()
  File "TestSoundCard.py", line 5, in test_sounddevice_lib
    import sounddevice as sd
  File "/usr/lib/python3.6/site-packages/sounddevice.py", line 64, in <module>
    raise OSError('PortAudio library not found')
  OSError: PortAudio library not found

    """

    duration = 2.5  # seconds

    rx_buffer = np.ones((10 ** 6, 2), dtype=np.float32)
    global current_rx, current_tx
    current_rx = 0
    current_tx = 0

    def rx_callback(indata: np.ndarray, frames: int, time, status):
        global current_rx
        if status:
            print(status)

        rx_buffer[current_rx:current_rx + frames] = indata
        current_rx += frames

    def tx_callback(outdata: np.ndarray, frames: int, time, status):
        global current_tx
        if status:
            print(status)

        outdata[:] = rx_buffer[current_tx:current_tx + frames]
        current_tx += frames

    with InputStream(channels=2, callback=rx_callback):
        sd_sleep(int(duration * 1000))

    print("Current rx", current_rx)

    with OutputStream(channels=2, callback=tx_callback):
        sd_sleep(int(duration * 1000))

    print("Current tx", current_tx)