prosody_model

Prosody Predictor

A small (682K param) convolutional model that predicts pitch (F0) and volume (RMS) contours from text at 100ms resolution.

Model Architecture

Text -> CharEncoder (4x Conv1d) -> DurationPredictor (2x Conv1d, detached)
                                -> LengthRegulator (repeat by durations)
                                -> FrameDecoder (3x Conv1d) -> [F0, RMS]

• CharEncoder: Char embedding (51 -> 128) + sinusoidal positional encoding + 4x Conv1d(128, k=5) + ReLU + LayerNorm + Dropout
• DurationPredictor: Detached encoder input + 2x Conv1d(128, k=3) + ReLU + LN + Drop -> linear(1)
• LengthRegulator: Repeats encoder output per-character by predicted durations
• FrameDecoder: 3x Conv1d(128, k=5) + ReLU + LN + Drop -> linear(2) for [F0, RMS]

Quickstart

import torch
from model_prosody import ProsodyPredictor
from infer_prosody import predict_prosody

ckpt = torch.load("final_model.pt", map_location="cpu", weights_only=False)
model = ProsodyPredictor(vocab_size=ckpt["vocab_size"], d_model=128, dropout=0.0)
model.load_state_dict(ckpt["model"])
model.eval()

result = predict_prosody("Hello, I am Kobi AI", model, ckpt["norm_stats"])
# result["f0_hz"]     - pitch in Hz per 100ms frame
# result["rms"]       - volume per 100ms frame
# result["duration_s"] - total duration in seconds

Synthesize as Sine Wave

import numpy as np
import soundfile as sf
from scipy.interpolate import CubicSpline

f0 = result["f0_hz"]
rms = result["rms"]
sr = 24000
frame_dur = 0.1
n_frames = len(f0)
total_samples = int(n_frames * frame_dur * sr)

# Smooth interpolation between frames
frame_times = (np.arange(n_frames) + 0.5) * frame_dur
sample_times = np.arange(total_samples) / sr
f0_smooth = np.clip(CubicSpline(frame_times, f0, bc_type='clamped')(sample_times), 50, 300)
rms_smooth = np.clip(CubicSpline(frame_times, rms, bc_type='clamped')(sample_times), 0, None)

# Generate with continuous phase
phase = np.cumsum(2 * np.pi * f0_smooth / sr)
audio = (rms_smooth * np.sin(phase)).astype(np.float32)
audio = audio / (np.abs(audio).max() + 1e-8) * 0.8
sf.write("output.wav", audio, sr)

Files

Training Details

• Data: 2000 TTS WAV samples (24kHz mono) with text transcripts
• Features: F0 via librosa pyin (50-300 Hz), RMS, z-score normalized
• Split: 95/5 train/val, seed=42
• Optimizer: AdamW, lr=1e-3 -> 1e-5 cosine annealing, 200-step warmup
• Loss: MSE(pitch, voiced only) + MSE(volume, all frames) + 0.1 * MSE(log duration)
• Batch size: 48, Epochs: 200, Grad clip: 1.0

Limitations

• Duration prediction uses proportional alignment (frames / chars), not forced alignment. The model learns positional averages rather than phoneme-specific timing.
• Deterministic output -- no sampling or variance prediction. Same text always produces the same contour.
• Trained on a single TTS voice, so prosody patterns reflect that speaker's style.