commit dbe3f7cede3212637c177c202b6ef17b7c93520f Author: jan Date: Fri Jan 5 11:22:10 2018 -0800 initial commit diff --git a/main.py b/main.py new file mode 100644 index 0000000..4d947bf --- /dev/null +++ b/main.py @@ -0,0 +1,111 @@ +import logging + +import pyaudio +import numpy +from numpy import pi + +logging.basicConfig(level=logging.INFO) +logger = logging.getLogger(__name__) + + +standard_sample_rates = 1000 * numpy.array([ + 8, 9.6, 11.025, 12, 16, 22.05, 24, 32, + 44.1, 48, 88.2, 96, 192]) + +def monitor_pitch(device: int = 5, + max_freq: float = 6000, + min_freq: float = 10, + samples_per_buffer: int = 1024, + audio: pyaudio.PyAudio = None, + ): + if audio is None: + audio = pyaudio.PyAudio() + + supported_sample_rates = [] + devinfo = audio.get_device_info_by_index(device) + for rate in standard_sample_rates: + try: + if audio.is_format_supported(rate, + input_device=device, + input_channels=devinfo['maxInputChannels'], + input_format=pyaudio.paInt16): + supported_sample_rates.append(rate) + except ValueError: + pass + supported_sample_rates = numpy.array(supported_sample_rates) + logger.info('Supported rates: {}'.format(supported_sample_rates)) + + ''' + max_freq < 2 * sample_rate + min_freq * 2**(1/12) > freq_resolution (for discrimination), more for accuracy... + freq_resolution <= sample_rate / (samples_per_buffer * num_buffers) + ''' + freq_resolution = min_freq * 2**(1/12) / 10 + + rate_is_acceptable = supported_sample_rates >= 2 * max_freq + sample_rate = int(numpy.min(supported_sample_rates[rate_is_acceptable])) + num_buffers = int(numpy.ceil(sample_rate / (samples_per_buffer * freq_resolution))) + samples_per_fft = samples_per_buffer * num_buffers + + logger.info('Running on device {} with {} buffers,'.format(device, num_buffers) + + ' {} sample rate, {} samples per buffer'.format( + device, num_buffers, sample_rate, samples_per_buffer)) + logger.info('Buffers take {:.3g} sec to fully clear'.format(samples_per_fft / sample_rate)) + + stream = audio.open(format=pyaudio.paInt16, + channels=1, + rate=sample_rate, + input=True, + frames_per_buffer=samples_per_buffer) + stream.start_stream() + + + # Hanning window + window = (1 - numpy.cos(numpy.linspace(0, 2 * pi, samples_per_fft, False))) / 2 + + freqs = numpy.fft.fftfreq(samples_per_fft, 1 / sample_rate) + + buf = numpy.zeros(num_buffers * samples_per_buffer, dtype=numpy.float32) + note_names = ['C', 'C#', 'D', 'D#', 'E', 'F', 'F#', 'G', 'G#', 'A', 'A#', 'B'] + + while stream.is_active(): + # Shift the buffer down and new data in + buf[:-samples_per_buffer] = buf[samples_per_buffer:] + buf[-samples_per_buffer:] = numpy.fromstring(stream.read(samples_per_buffer), numpy.int16) + + fft = numpy.fft.rfft(buf * window) + + # Get frequency of maximum response in range + ind = numpy.abs(fft[1:]).argmax() + 1 + freq = freqs[ind] + mag = numpy.abs(fft[ind]) + + # Get note number and nearest note + q = numpy.log2(freq/440) + n = 12 * q + 69 + n0 = int(round(n)) + + delta = n - n0 + logger.info('freq: {:7.2f} Hz mag:{:7.2f} note: {:>3s} {:+.2f}'.format( + freq, numpy.log10(mag), note_names[n0 % 12] + str(n0//12 - 1), delta)) + + delta_part = int(delta // 0.1) + if delta_part > 0: + signal = ' ' * 6 + '+' * delta_part + elif delta_part == 0: + signal = ' ' * 5 + '|' + elif delta_part < 0: + signal = ' ' * (5 + delta_part) + '-' * delta_part + logger.info(' {}'.format(signal)) + + +if __name__ == '__main__': + audio = pyaudio.PyAudio() + logger.info("Available devices:") + for device in range(audio.get_device_count()): + devinfo = audio.get_device_info_by_index(device) + if devinfo['maxInputChannels'] > 0: + logger.info('{}: {}'.format(device, devinfo['name'])) + + monitor_pitch(device=5, min_freq=20) +