def play_audio(wf:wave.Wave_read):
CHUNK = 1024
# instantiate PyAudio (1)
p = pyaudio.PyAudio()
# open stream (2)
stream = p.open(format=p.get_format_from_width(wf.getsampwidth()),
channels=wf.getnchannels(),
rate=wf.getframerate(),
output=True)
# read data
data = wf.readframes(CHUNK)
# play stream (3)
while len(data) > 0:
stream.write(data)
data = wf.readframes(CHUNK)
stream.stop_stream()
stream.close()
p.terminate()
def fourier(audio: wave.Wave_read) -> Tuple[Optional[int], Optional[int]]:
"""Fourierova analýza vstupních dat, vracející (nejnižší, nejvyšší) frekvenci."""
# data
length = audio.getnframes()
sample_rate = audio.getframerate()
windows_count = length // sample_rate
channels = 1 if audio.getnchannels() == 1 else 2 # Stereo (2) vs. Mono (1)
frames = sample_rate * windows_count
data = np.array(unpack(f"{channels * frames}h", audio.readframes(frames)))
if channels == 2:
data = merge_channels(data)
# amplitudy
low, high = None, None
for i in range(windows_count):
bounds = (i * sample_rate, i * sample_rate + sample_rate)
window = data[bounds[0]:bounds[1]]
amplitudes = np.abs(np.fft.rfft(window))
average = np.average(amplitudes)
# peaks
peak = lambda amp: amp >= 20 * average # ze zadání
for j in range(len(amplitudes)):
amplitude = amplitudes[j]
if not peak(amplitude):
continue
if not low:
low = j
high = j
else:
high = j
if not any((low, high)):
return None, None
return (high, low) if high < low else (low, high) # Může být totiž prohozené
def __get_wav_stats(self, audio: wave.Wave_read):
return {
"waveform": audio,
"frameRate": audio.getframerate(),
"nChannels": audio.getnchannels(),
"sampWidth": audio.getsampwidth()
}
def filter_lowpass(wav: Wave_read, cutoff: int):
signal = wav.data
signal = np.fromstring(signal, "Int16")
index = -1
frames = []
for frame in signal:
index += 1
if abs(frame) < cutoff:
frames.append(10)
pass
else:
frames.append(frame)
wav.close()
filtered: wave.Wave_write = wave.open(join(const.AUDIO_DIR, 'temp.wav'),
'w')
filtered.setframerate(wav.getframerate())
filtered.setsampwidth(wav.getsampwidth())
filtered.setnchannels(wav.getnchannels())
for frame in frames:
data = struct.pack('<h', frame)
filtered.writeframesraw(data)
filtered.close()
return wave.open(join(const.AUDIO_DIR, 'temp.wav'), 'r')
def get_bitrate(wave_obj:wave.Wave_read):
framerate = wave_obj.getframerate()
num_channels = wave_obj.getnchannels()
sample_width = wave_obj.getsampwidth()
bitrate = (framerate * num_channels * sample_width) / 1000
return bitrate
def filter_lowpassTest(wav: Wave_read, cutoff: int):
signal = wav.readframes(-1)
signal = np.fromstring(signal, "Int16")
filtered: wave.Wave_write = wave.open(join(const.AUDIO_DIR, 'temp.wav'),
'w')
filtered.setframerate(wav.getframerate())
filtered.setsampwidth(wav.getsampwidth())
filtered.setnchannels(wav.getnchannels())
for frame in frames:
data = struct.pack('<h', frame)
filtered.writeframesraw(data)
filtered.close()
return wave.open(join(const.AUDIO_DIR, 'temp.wav'), 'r')
