Split View: torchaudio 완전 가이드 — 오디오 처리부터 음성인식, TTS, 음악 분석까지

pip install torch torchaudio

import torch
import torchaudio

# Load audio
waveform, sample_rate = torchaudio.load("speech.wav")
print(f"Shape: {waveform.shape}")    # [channels, samples]
print(f"Sample Rate: {sample_rate}")  # 16000
print(f"Duration: {waveform.shape[1] / sample_rate:.2f}s")

# Channels: mono (1) vs stereo (2)
if waveform.shape[0] == 2:
    mono = waveform.mean(dim=0, keepdim=True)  # Stereo -> Mono

# Resampling (44100Hz -> 16000Hz)
resampler = torchaudio.transforms.Resample(
    orig_freq=44100, new_freq=16000
)
waveform_16k = resampler(waveform)

# Save
torchaudio.save("output.wav", waveform_16k, 16000)

# Supported formats: wav, flac, mp3, ogg, opus, sphere
# Backends: sox, soundfile, ffmpeg
print(torchaudio.list_audio_backends())

import matplotlib.pyplot as plt

# Waveform
fig, axes = plt.subplots(3, 1, figsize=(12, 8))

# 1. Time domain (waveform)
time_axis = torch.arange(0, waveform.shape[1]) / sample_rate
axes[0].plot(time_axis, waveform[0])
axes[0].set_title("Waveform")
axes[0].set_xlabel("Time (s)")
axes[0].set_ylabel("Amplitude")

# 2. Spectrogram
spectrogram = torchaudio.transforms.Spectrogram(n_fft=1024)(waveform)
axes[1].imshow(
    spectrogram[0].log2().numpy(),
    aspect='auto', origin='lower', cmap='magma'
)
axes[1].set_title("Spectrogram")

# 3. Mel Spectrogram
mel_spec = torchaudio.transforms.MelSpectrogram(
    sample_rate=sample_rate, n_fft=1024, n_mels=80
)(waveform)
axes[2].imshow(
    mel_spec[0].log2().numpy(),
    aspect='auto', origin='lower', cmap='magma'
)
axes[2].set_title("Mel Spectrogram")

plt.tight_layout()
plt.savefig("audio_analysis.png", dpi=150)

# STFT (Short-Time Fourier Transform)
# Converts from time domain to time+frequency domain
spectrogram_transform = torchaudio.transforms.Spectrogram(
    n_fft=1024,       # FFT window size (frequency resolution)
    hop_length=256,    # Window hop interval (time resolution)
    win_length=1024,   # Window length
    power=2.0,         # 2.0=power, 1.0=amplitude
)

spec = spectrogram_transform(waveform)
# shape: [channels, n_freq_bins, time_frames]
# n_freq_bins = n_fft // 2 + 1 = 513

Trade-off between n_fft and hop_length:

n_fft up -> frequency resolution up, time resolution down
n_fft down -> frequency resolution down, time resolution up

Common settings:
+-- Speech: n_fft=400~512, hop=160 (at 16kHz)
+-- Music: n_fft=2048, hop=512 (at 44.1kHz)
+-- General: n_fft=1024, hop=256

# The human ear is sensitive to low frequencies, less so to high frequencies.
# Mel scale = a frequency scale that reflects human auditory perception.

mel_transform = torchaudio.transforms.MelSpectrogram(
    sample_rate=16000,
    n_fft=1024,
    hop_length=256,
    n_mels=80,          # Number of Mel filters (typically 40~128)
    f_min=0,            # Minimum frequency
    f_max=8000,         # Maximum frequency (Nyquist)
)

mel_spec = mel_transform(waveform)
# shape: [1, 80, time_frames]

# Convert to dB scale (log compression)
amplitude_to_db = torchaudio.transforms.AmplitudeToDB(stype='power', top_db=80)
mel_spec_db = amplitude_to_db(mel_spec)

Mel frequency conversion formula:
  mel = 2595 * log10(1 + freq / 700)

Frequency -> Mel:
  100 Hz  ->  150 mel   (low freq: dense)
  1000 Hz ->  1000 mel
  4000 Hz ->  2146 mel
  8000 Hz ->  2840 mel  (high freq: sparse)

-> Low frequencies are analyzed finely, high frequencies are coarsely grouped
-> A representation similar to how humans hear!

# MFCC = Mel Spectrogram + DCT (Discrete Cosine Transform)
# Key features representing the "shape" of speech

mfcc_transform = torchaudio.transforms.MFCC(
    sample_rate=16000,
    n_mfcc=13,          # Number of MFCC coefficients (typically 13~40)
    melkwargs={
        'n_fft': 1024,
        'n_mels': 80,
        'hop_length': 256,
    }
)

mfcc = mfcc_transform(waveform)
# shape: [1, 13, time_frames]

# Delta (1st derivative) + Delta-Delta (2nd derivative)
# -> Adds rate-of-change information for speech
delta = torchaudio.functional.compute_deltas(mfcc)
delta_delta = torchaudio.functional.compute_deltas(delta)

# Final features: concatenate [MFCC, Delta, Delta-Delta]
features = torch.cat([mfcc, delta, delta_delta], dim=1)
# shape: [1, 39, time_frames]

Where are these used?
+-- MFCC: Traditional speech recognition (HMM-GMM), speaker recognition
+-- Mel Spectrogram: Deep learning speech recognition (Wav2Vec2, Whisper)
+-- Spectrogram: Music analysis, environmental sound classification
+-- Raw Waveform: End-to-end models (latest trend)

# Time masking (SpecAugment)
time_masking = torchaudio.transforms.TimeMasking(
    time_mask_param=30   # Mask up to 30 frames
)

# Frequency masking (SpecAugment)
freq_masking = torchaudio.transforms.FrequencyMasking(
    freq_mask_param=15   # Mask up to 15 channels
)

# SpecAugment (significantly improves speech recognition accuracy!)
augmented_spec = time_masking(freq_masking(mel_spec))

# Time stretching
time_stretch = torchaudio.transforms.TimeStretch()
stretched = time_stretch(complex_spec, overriding_rate=1.2)  # 20% faster

# Pitch shifting
pitch_shift = torchaudio.transforms.PitchShift(
    sample_rate=16000, n_steps=4  # Shift up by 4 semitones
)
shifted = pitch_shift(waveform)

# Adding noise
def add_noise(waveform, snr_db=10):
    """Add white noise based on SNR in dB"""
    noise = torch.randn_like(waveform)
    signal_power = waveform.norm(p=2)
    noise_power = noise.norm(p=2)
    snr = 10 ** (snr_db / 20)
    scale = signal_power / (snr * noise_power)
    return waveform + scale * noise

import torchaudio
from torchaudio.pipelines import WAV2VEC2_ASR_BASE_960H

# Load pipeline
bundle = WAV2VEC2_ASR_BASE_960H
model = bundle.get_model()
labels = bundle.get_labels()  # Token list

# Inference
waveform, sr = torchaudio.load("speech.wav")
if sr != bundle.sample_rate:
    waveform = torchaudio.transforms.Resample(sr, bundle.sample_rate)(waveform)

with torch.no_grad():
    emissions, _ = model(waveform)

# CTC Decoding (Greedy)
def greedy_decode(emissions, labels):
    indices = torch.argmax(emissions, dim=-1)[0]
    tokens = []
    prev = -1
    for idx in indices:
        if idx != prev and idx != 0:  # 0 = blank
            tokens.append(labels[idx])
        prev = idx
    return "".join(tokens).replace("|", " ").strip()

text = greedy_decode(emissions, labels)
print(f"Recognition result: {text}")

from torchaudio.pipelines import HUBERT_BASE

bundle = HUBERT_BASE
model = bundle.get_model()

with torch.no_grad():
    features, _ = model(waveform)
# features: [1, time_frames, 768]
# -> Semantic representation vectors of speech
# -> Used for speaker recognition, emotion analysis, speech classification

# Temporal alignment between speech and text!
# -> Essential for subtitle generation and lyrics synchronization

from torchaudio.pipelines import MMS_FA  # Multilingual!

bundle = MMS_FA
model = bundle.get_model()
tokenizer = bundle.get_tokenizer()
aligner = bundle.get_aligner()

transcript = "hello nice to meet you"
tokens = tokenizer(transcript)

with torch.no_grad():
    emissions, _ = model(waveform)

token_spans = aligner(emissions[0], tokens)
# Returns start/end times for each token in frame units!

for span, token in zip(token_spans, transcript):
    start_time = span.start * model.hop_length / sample_rate
    end_time = span.end * model.hop_length / sample_rate
    print(f"  '{token}': {start_time:.3f}s ~ {end_time:.3f}s")

# torchaudio.functional — GPU-accelerated audio processing

import torchaudio.functional as F

# Volume adjustment
loud = F.gain(waveform, gain_db=6.0)    # +6dB
quiet = F.gain(waveform, gain_db=-6.0)  # -6dB

# High-pass / Low-pass filter
highpass = F.highpass_biquad(waveform, sample_rate, cutoff_freq=300)
lowpass = F.lowpass_biquad(waveform, sample_rate, cutoff_freq=3000)

# Equalizer
eq = F.equalizer_biquad(
    waveform, sample_rate,
    center_freq=1000,  # Around 1kHz
    gain=5.0,          # +5dB boost
    Q=0.707
)

# Reverb
rir, _ = torchaudio.load("room_impulse_response.wav")  # RIR file
reverb = F.fftconvolve(waveform, rir)

# Fade in/out
fade = torchaudio.transforms.Fade(
    fade_in_len=sample_rate,      # 1 second fade in
    fade_out_len=sample_rate * 3  # 3 second fade out
)
faded = fade(waveform)

# VAD (Voice Activity Detection)
vad = torchaudio.transforms.Vad(sample_rate=16000)
speech_only = vad(waveform)  # Remove silent segments

import torch.nn as nn

class AudioClassifier(nn.Module):
    def __init__(self, n_classes=10):
        super().__init__()
        self.mel = torchaudio.transforms.MelSpectrogram(
            sample_rate=16000, n_fft=1024, n_mels=64
        )
        self.db = torchaudio.transforms.AmplitudeToDB()

        # Feed the Mel spectrogram into a CNN as if it were an "image"!
        self.cnn = nn.Sequential(
            nn.Conv2d(1, 32, 3, padding=1), nn.ReLU(), nn.MaxPool2d(2),
            nn.Conv2d(32, 64, 3, padding=1), nn.ReLU(), nn.MaxPool2d(2),
            nn.Conv2d(64, 128, 3, padding=1), nn.ReLU(),
            nn.AdaptiveAvgPool2d((1, 1)),
        )
        self.fc = nn.Linear(128, n_classes)

    def forward(self, waveform):
        # [B, 1, samples] -> [B, 1, n_mels, time]
        x = self.mel(waveform)
        x = self.db(x)
        x = self.cnn(x)
        x = x.view(x.size(0), -1)
        return self.fc(x)

# Mel Spectrogram = an "image" of audio
# -> Can be classified with CNNs (ResNet, EfficientNet)!

from torchaudio.io import StreamReader

# Real-time microphone input processing
reader = StreamReader(src=":0", format="avfoundation")  # macOS
reader.add_basic_audio_stream(
    frames_per_chunk=16000,  # 1 second chunks
    sample_rate=16000,
)

for (chunk,) in reader.stream():
    # chunk: [1, 16000]
    mel = mel_transform(chunk)
    with torch.no_grad():
        prediction = model(mel)
    print(f"Detected: {labels[prediction.argmax()]}")