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Audio Effects Processor (MATLAB)
Introduction

This project details the design and development of a real-time audio effects processor, "AudioLab," built using MATLAB. The project includes a graphical user interface (GUI) that allows users to load audio files, apply various audio effects, and revert changes. Audio effects implemented in the project include gain, fade, reverse playback, speed control, trim, and more.

The application also features real-time recording, de-effectifying capabilities, and intuitive controls inspired by modern audio editing software like VEED.io and AudioMass.co

AudioLab is an audio editing tool that allows users to upload audio files or record sound in realtime, apply several audio effects, and observe the results in real-time. The interface is userfriendly and designed to make audio manipulation accessible for beginners and professionals alike. AudioLab supports a range of effects, and includes a "de-effect" feature to reverse any changes made. The project is designed to work seamlessly across MATLAB desktop and online

versions, using a clean and modern user interface that guides users through the editing process.

The application is developed using MATLAB’s GUI design framework, ensuring ease of use and the ability to visually represent audio waveforms.

Description
Features in This Project
  1. Gain Control: This method increases or decreases the amplitude of the signal. If the gain factor is greater than 1, the volume is amplified. If it is less than 1, the signal is attenuated.

  2. Fade In/Out: In a fade-in, the volume starts at zero and gradually increases to full amplitude. For a fade-out, the volume starts at full amplitude and gradually decreases to zero. This is done by multiplying the signal with a linear ramp function over the duration of the fade.

  3. Speed Control: By changing the rate at which samples are played, the duration and pitch of the audio change. A higher sampling rate speeds up playback and raises the pitch, while a lower rate slows it down and lowers the pitch.

  4. Reverse Playback: The samples of the audio signal are flipped, playing them in reverse.

  5. Compressor: Compression adjusts the amplitude based on a threshold. If the amplitude exceeds the threshold, it’s reduced by a set ratio.

  6. Normalize: The audio is scaled by a factor that brings the highest peak to a specified target amplitude.

  7. Reverb: The original signal is combined with delayed and decayed copies of itself, creating the effect of reverberation.

  8. Echo: Delayed versions of the original signal are added back to the signal, spaced at regular intervals.

  9. Distortion: The signal is amplified until its peaks exceed a set threshold, where it is then clipped to prevent further increase.

  10. Equalization (EQ): Different frequency bands are either amplified or attenuated, allowing for finetuning of the audio's tonal balance.

For watch the final demonstration, click here

For download the project report, Download

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