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Audio Spectrogram Transformers Past the Lab

admin by admin
June 11, 2025
in Artificial Intelligence
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Audio Spectrogram Transformers Past the Lab
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Need to know what attracts me to soundscape evaluation?

It’s a subject that mixes science, creativity, and exploration in a manner few others do. To begin with, your laboratory is wherever your ft take you — a forest path, a metropolis park, or a distant mountain path can all turn out to be areas for scientific discovery and acoustic investigation. Secondly, monitoring a selected geographic space is all about creativity. Innovation is on the coronary heart of environmental audio analysis, whether or not it’s rigging up a customized system, hiding sensors in tree canopies, or utilizing solar energy for off-grid setups. Lastly, the sheer quantity of knowledge is actually unimaginable, and as we all know, in spatial evaluation, all strategies are truthful recreation. From hours of animal calls to the delicate hum of city equipment, the acoustic knowledge collected may be huge and complicated, and that opens the door to utilizing every part from deep studying to geographical info programs (GIS) in making sense of all of it.

After my earlier adventures with soundscape evaluation of one among Poland’s rivers, I made a decision to lift the bar and design and implement an answer able to analysing soundscapes in actual time. On this weblog put up, you’ll discover a description of the proposed methodology, together with some code that powers all the course of, primarily utilizing an Audio Spectrogram Transformer (AST) for sound classification.

Device prototype
Outside/City model of the sensor prototype (picture by writer)

Strategies

Setup

There are lots of the reason why, on this specific case, I selected to make use of a mix of Raspberry Pi 4 and AudioMoth. Imagine me, I examined a variety of units — from much less power-hungry fashions of the Raspberry Pi household, by numerous Arduino variations, together with the Portenta, all the best way to the Jetson Nano. And that was only the start. Selecting the best microphone turned out to be much more difficult.

In the end, I went with the Pi 4 B (4GB RAM) due to its stable efficiency and comparatively low energy consumption (~700mAh when working my code). Moreover, pairing it with the AudioMoth in USB microphone mode gave me lots of flexibility throughout prototyping. AudioMoth is a robust system with a wealth of configuration choices, e.g. sampling price from 8 kHz to gorgeous 384 kHz. I’ve a powerful feeling that — in the long term — it will show to be an ideal selection for my soundscape research.

AudioMoth USB Microphone configuration app. Bear in mind about flashing the system with the correct firmware earlier than configuring.

Capturing sound

Capturing audio from a USB microphone utilizing Python turned out to be surprisingly troublesome. After combating numerous libraries for some time, I made a decision to fall again on the great outdated Linux arecord. The entire sound seize mechanism is encapsulated with the next command:

arecord -d 1 -D plughw:0,7 -f S16_LE -r 16000 -c 1 -q /tmp/audio.wav

I’m intentionally utilizing a plug-in system to allow automated conversion in case I want to introduce any adjustments to the USB microphone configuration. AST is run on 16 kHz samples, so the recording and AudioMoth sampling are set to this worth.

Take note of the generator within the code. It’s vital that the system constantly captures audio on the time intervals I specify. I aimed to retailer solely the latest audio pattern on the system and discard it after the classification. This strategy will likely be particularly helpful later throughout larger-scale research in city areas, because it helps guarantee individuals’s privateness and aligns with GDPR compliance.

import asyncio
import re
import subprocess
from tempfile import TemporaryDirectory
from typing import Any, AsyncGenerator

import librosa
import numpy as np


class AudioDevice:
    def __init__(
        self,
        title: str,
        channels: int,
        sampling_rate: int,
        format: str,
    ):
        self.title = self._match_device(title)
        self.channels = channels
        self.sampling_rate = sampling_rate
        self.format = format

    @staticmethod
    def _match_device(title: str):
        strains = subprocess.check_output(['arecord', '-l'], textual content=True).splitlines()
        units = [
            f'plughw:{m.group(1)},{m.group(2)}'
            for line in lines
            if name.lower() in line.lower()
            if (m := re.search(r'card (d+):.*device (d+):', line))
        ]

        if len(units) == 0:
            elevate ValueError(f'No units discovered matching `{title}`')
        if len(units) > 1:
            elevate ValueError(f'A number of units discovered matching `{title}` -> {units}')
        return units[0]

    async def continuous_capture(
        self,
        sample_duration: int = 1,
        capture_delay: int = 0,
    ) -> AsyncGenerator[np.ndarray, Any]:
        with TemporaryDirectory() as temp_dir:
            temp_file = f'{temp_dir}/audio.wav'
            command = (
                f'arecord '
                f'-d {sample_duration} '
                f'-D {self.title} '
                f'-f {self.format} '
                f'-r {self.sampling_rate} '
                f'-c {self.channels} '
                f'-q '
                f'{temp_file}'
            )

            whereas True:
                subprocess.check_call(command, shell=True)
                knowledge, sr = librosa.load(
                    temp_file,
                    sr=self.sampling_rate,
                )
                await asyncio.sleep(capture_delay)
                yield knowledge

Classification

Now for essentially the most thrilling half.

Utilizing the Audio Spectrogram Transformer (AST) and the wonderful HuggingFace ecosystem, we will effectively analyse audio and classify detected segments into over 500 classes.
Word that I’ve ready the system to help numerous pre-trained fashions. By default, I take advantage of MIT/ast-finetuned-audioset-10–10–0.4593, because it delivers one of the best outcomes and runs properly on the Raspberry Pi 4. Nonetheless, onnx-community/ast-finetuned-audioset-10–10–0.4593-ONNX can be value exploring — particularly its quantised model, which requires much less reminiscence and serves the inference outcomes faster.

You might discover that I’m not limiting the mannequin to a single classification label, and that’s intentional. As a substitute of assuming that just one sound supply is current at any given time, I apply a sigmoid operate to the mannequin’s logits to acquire impartial chances for every class. This enables the mannequin to precise confidence in a number of labels concurrently, which is essential for real-world soundscapes the place overlapping sources — like birds, wind, and distant visitors — typically happen collectively. Taking the high 5 outcomes ensures that the system captures the more than likely sound occasions within the pattern with out forcing a winner-takes-all resolution.

from pathlib import Path
from typing import Non-obligatory

import numpy as np
import pandas as pd
import torch
from optimum.onnxruntime import ORTModelForAudioClassification
from transformers import AutoFeatureExtractor, ASTForAudioClassification


class AudioClassifier:
    def __init__(self, pretrained_ast: str, pretrained_ast_file_name: Non-obligatory[str] = None):
        if pretrained_ast_file_name and Path(pretrained_ast_file_name).suffix == '.onnx':
            self.mannequin = ORTModelForAudioClassification.from_pretrained(
                pretrained_ast,
                subfolder='onnx',
                file_name=pretrained_ast_file_name,
            )
            self.feature_extractor = AutoFeatureExtractor.from_pretrained(
                pretrained_ast,
                file_name=pretrained_ast_file_name,
            )
        else:
            self.mannequin = ASTForAudioClassification.from_pretrained(pretrained_ast)
            self.feature_extractor = AutoFeatureExtractor.from_pretrained(pretrained_ast)

        self.sampling_rate = self.feature_extractor.sampling_rate

    async def predict(
        self,
        audio: np.array,
        top_k: int = 5,
    ) -> pd.DataFrame:
        with torch.no_grad():
            inputs = self.feature_extractor(
                audio,
                sampling_rate=self.sampling_rate,
                return_tensors='pt',
            )
            logits = self.mannequin(**inputs).logits[0]
            proba = torch.sigmoid(logits)
            top_k_indices = torch.argsort(proba)[-top_k:].flip(dims=(0,)).tolist()

            return pd.DataFrame(
                {
                    'label': [self.model.config.id2label[i] for i in top_k_indices],
                    'rating': proba[top_k_indices],
                }
            )

To run the ONNX model of the mannequin, it’s good to add Optimum to your dependencies.

Sound strain degree

Together with the audio classification, I seize info on sound strain degree. This strategy not solely identifies what made the sound but in addition positive factors perception into how strongly every sound was current. In that manner, the mannequin captures a richer, extra lifelike illustration of the acoustic scene and may ultimately be used to detect finer-grained noise air pollution info.

import numpy as np
from maad.spl import wav2dBSPL
from maad.util import mean_dB


async def calculate_sound_pressure_level(audio: np.ndarray, achieve=10 + 15, sensitivity=-18) -> np.ndarray:
    x = wav2dBSPL(audio, achieve=achieve, sensitivity=sensitivity, Vadc=1.25)
    return mean_dB(x, axis=0)

The achieve (preamp + amp), sensitivity (dB/V), and Vadc (V) are set primarily for AudioMoth and confirmed experimentally. In case you are utilizing a distinct system, you could establish these values by referring to the technical specification.

Storage

Knowledge from every sensor is synchronised with a PostgreSQL database each 30 seconds. The present city soundscape monitor prototype makes use of an Ethernet connection; subsequently, I’m not restricted by way of community load. The system for extra distant areas will synchronise the information every hour utilizing a GSM connection.

label           rating        system   sync_id                                sync_time
Hum             0.43894055   yor      9531b89a-4b38-4a43-946b-43ae2f704961   2025-05-26 14:57:49.104271
Mains hum       0.3894045    yor      9531b89a-4b38-4a43-946b-43ae2f704961   2025-05-26 14:57:49.104271
Static          0.06389702   yor      9531b89a-4b38-4a43-946b-43ae2f704961   2025-05-26 14:57:49.104271
Buzz            0.047603738  yor      9531b89a-4b38-4a43-946b-43ae2f704961   2025-05-26 14:57:49.104271
White noise     0.03204195   yor      9531b89a-4b38-4a43-946b-43ae2f704961   2025-05-26 14:57:49.104271
Bee, wasp, and so forth. 0.40881288   yor      8477e05c-0b52-41b2-b5e9-727a01b9ec87   2025-05-26 14:58:40.641071
Fly, housefly   0.38868183   yor      8477e05c-0b52-41b2-b5e9-727a01b9ec87   2025-05-26 14:58:40.641071
Insect          0.35616025   yor      8477e05c-0b52-41b2-b5e9-727a01b9ec87   2025-05-26 14:58:40.641071
Speech          0.23579548   yor      8477e05c-0b52-41b2-b5e9-727a01b9ec87   2025-05-26 14:58:40.641071
Buzz            0.105577625  yor      8477e05c-0b52-41b2-b5e9-727a01b9ec87   2025-05-26 14:58:40.641071

Outcomes

A separate software, constructed utilizing Streamlit and Plotly, accesses this knowledge. Presently, it shows details about the system’s location, temporal SPL (sound strain degree), recognized sound courses, and a variety of acoustic indices.

Dashboard
Streamit analytical dashboard (picture by writer)

And now we’re good to go. The plan is to increase the sensor community and attain round 20 units scattered round a number of locations in my metropolis. Extra details about a bigger space sensor deployment will likely be obtainable quickly.

Furthermore, I’m amassing knowledge from a deployed sensor and plan to share the information package deal, dashboard, and evaluation in an upcoming weblog put up. I’ll use an fascinating strategy that warrants a deeper dive into audio classification. The primary thought is to match totally different sound strain ranges to the detected audio courses. I hope to discover a higher manner of describing noise air pollution. So keep tuned for a extra detailed breakdown quickly.

Within the meantime, you may learn the preliminary paper on my soundscapes research (headphones are compulsory).


This put up was proofread and edited utilizing Grammarly to enhance grammar and readability.

Tags: audioLabSpectrogramTransformers
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