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from copy import copy
import numpy as np
import time
import whisper
print('Loaded audio.py')
CHUNK_LENGTH = 24000 # 48000 Hz * 0.5 s
def process_pcm(audio_chunks, data):
# pymumble PCM is 16-bit 48000 Hz
start = time.time()
audio_chunks.append(data)
if len(audio_chunks) > 75:
audio_chunks.pop(0)
#print('finished chunk in', time.time() - start, 's')
def process_stream(audio_chunks, model):
if len(audio_chunks) != 75:
print('Skipping, bad length.')
time.sleep(0.5)
return
start = time.time()
a = copy(audio_chunks)
b = b''.join(a)
c = np.frombuffer(b, np.int16)
# Define a low-pass filter kernel
fs = 48000
cutoff_freq = fs / 6
nyquist_freq = fs / 2
num_taps = 101
taps = np.sinc(2 * cutoff_freq / fs * (np.arange(num_taps) - (num_taps - 1) / 2))
taps *= np.blackman(num_taps)
taps /= np.sum(taps)
# Apply the filter kernel to audio_data using convolution
filtered_audio_data = np.convolve(c, taps, mode='same')
# Downsample filtered_audio_data by a factor of 3 using take
downsampled_audio_data = filtered_audio_data.take(np.arange(0, len(filtered_audio_data), 3)).flatten()
norm_audio = downsampled_audio_data.astype(np.float32) / 32768.0
#abs_mean = np.mean(np.abs(downsampled_audio_data ** 3))
#print('abs mean:', abs_mean)
#if abs_mean < 0.0:
# print('silence detected, skipping')
# time.sleep(1)
# return
d = whisper.pad_or_trim(norm_audio)
#print('processed audio in', time.time() - start, 's')
start = time.time()
e = model.transcribe(d, language='en')
print('transcribed audio in', time.time() - start, 's')
if time.time() - start > 10:
with open('downsampled.pcm', 'wb') as f:
f.write(downsampled_audio_data.astype(np.int16).tobytes())
print('wrote file, sleeping')
#breakpoint()
time.sleep(100)
print(' ', e['text'])