---
title: "Getting Started"
description: "Install phonometry and run your first fractional octave analysis."
---

## Installation

**Option 1: From PyPI (Recommended)**

```bash
pip install phonometry
```

Optional extras:

```bash
pip install phonometry[plot]   # matplotlib, for filter response plots and result .plot() methods
pip install phonometry[perf]  # numba, faster 'impulse' time weighting
pip install phonometry[full]  # both
```

**Option 2: Cloning and Installing**

```bash
git clone https://github.com/jmrplens/phonometry.git
cd phonometry
pip install .
```

**Option 3: Git Submodule**

```bash
git submodule add https://github.com/jmrplens/phonometry.git
# Then install in editable mode to use it from your project
pip install -e ./phonometry
```

## The processing chain at a glance

Every phonometry analysis is some subset of one pipeline — take the raw
signal, convert it to physical units, weight it in frequency, split it into
standardized bands, smooth it in time and reduce it to metrics:

<img class="light-only" src="https://raw.githubusercontent.com/jmrplens/phonometry/main/.github/images/diagram_signal_chain.svg" alt="phonometry processing chain: signal, calibration, frequency weighting, octave filter bank, time weighting and metrics, with the standard verified at each stage" style="width:92%"><img class="dark-only" src="https://raw.githubusercontent.com/jmrplens/phonometry/main/.github/images/diagram_signal_chain_dark.svg" alt="phonometry processing chain: signal, calibration, frequency weighting, octave filter bank, time weighting and metrics, with the standard verified at each stage" style="width:92%">

Each stage is an independent function or class you can use on its own; the
guides cover them left to right ([Calibration](/phonometry/guides/calibration/) →
[Frequency Weighting](/phonometry/guides/weighting/) → [Filter Banks](/phonometry/guides/filter-banks/) →
[Time Weighting](/phonometry/guides/time-weighting/) → [Levels](/phonometry/guides/levels/)).

## Basic Usage: 1/3 Octave Analysis

Analyze a signal and get the Sound Pressure Level (SPL) per frequency band.

```python
import numpy as np
from phonometry import octave_filter

fs = 48000
t = np.linspace(0, 1, fs, endpoint=False)
# Composite signal: 100Hz + 1000Hz
signal = np.sin(2 * np.pi * 100 * t) + np.sin(2 * np.pi * 1000 * t)

# Apply 1/3 octave filter bank
spl, freq = octave_filter(signal, fs=fs, fraction=3)

print(f"Bands: {freq}")
# Bands: [12.589254117941678, 15.848931924611138, ..., 19952.623149688785]  (33 bands)
print(f"SPL [dB]: {spl}")
# SPL [dB]: [46.88395351 47.96774897 49.04991279 ...]  — ~90.7 dB at 100 Hz and ~90.9 dB at 1 kHz
```

<img class="light-only" src="https://raw.githubusercontent.com/jmrplens/phonometry/main/.github/images/signal_response_fraction_3.svg" alt="One-third-octave spectrum analysis of a multi-tone signal with the raw PSD in the background" style="width:80%"><img class="dark-only" src="https://raw.githubusercontent.com/jmrplens/phonometry/main/.github/images/signal_response_fraction_3_dark.svg" alt="One-third-octave spectrum analysis of a multi-tone signal with the raw PSD in the background" style="width:80%">

*Example of a 1/3 Octave Band spectrum analysis of a complex signal.*

<details>
<summary>Show the code for this figure</summary>

```python
import matplotlib.pyplot as plt
import scipy.signal

# Uses `signal`, `fs`, `spl` and `freq` from the snippet above.
# Gray background: the raw-signal PSD (Welch), shifted to sit just below the
# band SPLs so both spectral shapes share one axis.
f_psd, psd = scipy.signal.welch(signal, fs, nperseg=8192)
psd_db = 10 * np.log10(psd + 1e-12)
psd_db += np.max(spl) - np.max(psd_db) - 5

fig, ax = plt.subplots()
ax.semilogx(f_psd, psd_db, color="gray", alpha=0.6, label="Raw signal PSD")
ax.semilogx(freq, spl, marker="o", markerfacecolor="white",
            label="1/3 octave bands")
ax.set_xlabel("Frequency [Hz]")
ax.set_ylabel("SPL [dB]")
ax.legend()
plt.show()
```

</details>

## Analyzing an audio file

```python
from scipy.io import wavfile
from phonometry import octave_filter

# Load standard WAV file
fs, signal = wavfile.read("measurement.wav")

# Analyze
# Note: To obtain real-world SPL values, you must calibrate the input.
# See the Calibration guide.
spl, freq = octave_filter(signal, fs=fs, fraction=3)
```

Integer audio (e.g. int16 WAV data) is converted to float64 internally, so it is
safe to pass `wavfile.read` output directly.

## Where to go next

- [Filter Banks](/phonometry/guides/filter-banks/) — choose an architecture and inspect responses
- [Calibration and dBFS](/phonometry/guides/calibration/) — get real-world SPL values
- [API Reference](/phonometry/reference/api/) — every parameter of every function
