Audio Quality Enhancer

Background noise is one of the most common audio quality issues in recordings. Our advanced noise reduction algorithm analyzes your audio to identify and remove unwanted ambient sounds while preserving the clarity of speech.

The technology uses spectral analysis to distinguish between voice frequencies and background noise, ensuring that your primary content remains crystal clear. Whether it's air conditioning hum, computer fans, or distant traffic, our tool can significantly reduce these distractions.

This feature works best when applied moderately. Over-processing can sometimes create artifacts, so we provide adjustable strength controls to help you find the perfect balance for your specific recording.

Echo and reverb occur when sound bounces off hard surfaces in a room, creating a washed-out or distant sound quality. This is particularly common in rooms with hardwood floors, glass windows, or minimal furnishings.

Our echo reduction technology analyzes the reverb pattern in your audio and intelligently removes these reflections. This results in a more intimate, professional sound that makes speech easier to understand and transcribe.

The algorithm preserves the natural timbre of voices while removing room reflections, making it ideal for improving recordings made in less-than-ideal acoustic environments.

Electrical hum is a persistent low-frequency noise caused by electrical interference from power lines, lighting, or other electronic equipment. It typically manifests as a constant 50 Hz or 60 Hz tone (depending on your region).

Our hum removal tool uses notch filtering to precisely target and eliminate these specific frequencies along with their harmonics, without affecting the rest of your audio spectrum. This results in cleaner audio without any loss of voice quality.

This feature is particularly useful for recordings made near electrical equipment, in offices with fluorescent lighting, or when using equipment with ground loop issues.

Clicks and pops are sharp, transient noises that can occur from various sources: mouth sounds, microphone handling, connection issues, or interference. These distractions make audio less professional and can interfere with transcription accuracy.

Our click removal algorithm detects these brief impulses and intelligently repairs the waveform, replacing problematic sections with appropriate audio based on the surrounding context. The process is transparent and maintains the natural flow of speech.

This tool is especially valuable for cleaning up audio from handheld recording devices, wireless microphone systems, or recordings with digital interference.

Wind noise and low-frequency rumble can overpower dialogue in outdoor recordings or when using sensitive microphones. These low-frequency disturbances often obscure speech and create an unprofessional listening experience.

Our wind and rumble reduction uses sophisticated high-pass filtering combined with spectral analysis to remove these low-frequency intrusions. The algorithm adapts to the characteristics of your recording to preserve bass content in voices while removing unwanted rumble.

This feature is essential for outdoor interviews, travel vlogs, documentary footage, or any recording made in less controlled environments.

When enhancing audio, it's important to maintain the natural character and timbre of the speaker's voice. Our preservation technology ensures that noise reduction and other processing don't create an artificial or over-processed sound.

The "Natural to Aggressive" slider allows you to balance between maximum noise reduction and natural voice quality. The natural setting applies minimal processing for a transparent sound, while the aggressive setting prioritizes noise removal for challenging recordings.

This control gives you creative flexibility to match the processing to your specific needs, whether you're preparing a recording for critical listening or simply need to improve intelligibility for transcription.

Speech clarity enhancement uses carefully tuned equalization to emphasize the frequency ranges most important for understanding human speech. This makes dialogue more intelligible without making it sound harsh or unnatural.

We offer three preset curves: Warm (emphasizes lower frequencies for fuller sound), Neutral (balanced enhancement), and Bright (emphasizes upper frequencies for crisp articulation). Each preset is designed for different voice types and recording scenarios.

This feature is particularly valuable when recordings have been made with equipment that doesn't accurately capture the full range of speech, or when speakers are soft-spoken or speak with less articulation.

Sibilance refers to harsh "S", "T", and "Z" sounds that can become overly prominent in recordings, especially when using certain microphones or with speakers who naturally have strong sibilance. These sounds can be distracting and fatiguing to listeners.

Our de-esser intelligently detects sibilant frequencies (typically 4-8 kHz) and applies gentle compression only when needed. This reduces the harshness while maintaining natural speech patterns and not affecting other parts of the audio.

De-essing is a standard technique in professional audio production and is particularly important for recordings that will be heard on headphones or earbuds, where sibilance is most noticeable.

Voice isolation uses advanced machine learning to separate speech from all other sounds in a recording. This is our most aggressive noise reduction tool and can dramatically improve challenging recordings with multiple sound sources.

The technology has been trained on thousands of hours of speech and can distinguish human voice from music, background conversations, machinery, and other complex noise sources. However, being in beta, it may occasionally produce artifacts with unusual vocal styles or extreme noise conditions.

Use this feature when standard noise reduction isn't sufficient, such as recordings from noisy cafes, busy streets, or environments with music playing. Be aware that the process may slightly affect voice quality in exchange for superior noise removal.

Volume normalization ensures your audio reaches a consistent loudness level according to industry standards. This is essential for professional content delivery and ensures compatibility across different playback platforms.

We offer several LUFS (Loudness Units Full Scale) targets: Podcast (-16 LUFS, the podcast industry standard), Streaming (-14 LUFS, for YouTube and streaming platforms), Broadcast (-23 LUFS, for TV and radio), and Custom (set your own target).

Proper loudness normalization prevents your audio from being too quiet or too loud compared to other content, ensuring a comfortable listening experience and meeting platform requirements.

In multi-speaker recordings, different speakers often have varying volumes due to microphone distance, vocal projection, or recording levels. This creates an inconsistent listening experience and can make transcription difficult.

Our speaker leveling technology analyzes your audio to identify different speakers and automatically adjusts their relative volumes to create a balanced mix. Each speaker maintains their natural dynamics while achieving overall consistency.

This feature is invaluable for interviews, panel discussions, meetings, and any recording with multiple participants. It ensures that all speakers can be heard clearly without constant volume adjustments.

Audio clipping occurs when recording levels are too high, causing the waveform to be cut off and resulting in harsh distortion. This is one of the most challenging audio problems to fix, but our tool can help recover some clarity.

Our clipping repair algorithm uses interpolation and harmonic reconstruction to estimate what the audio should have sounded like before clipping occurred. While it can't completely restore severely clipped audio, it can significantly reduce harshness and improve intelligibility.

The repair slider allows you to control the strength of the correction. Higher settings provide more aggressive repair but may introduce subtle artifacts. Finding the right balance is key to optimal results.

Audio dynamics refer to the natural variation in volume throughout a recording – the soft and loud moments that give speech its natural expressiveness. Some processing can flatten these dynamics, making audio sound lifeless.

Enabling this option reduces the amount of compression applied during loudness normalization and other processing, preserving the natural ebb and flow of the speaker's delivery. This results in more engaging, natural-sounding audio.

While reduced compression may mean slightly more variation in volume, it maintains the emotional impact and natural communication style of the speakers, which is important for storytelling, persuasive content, and intimate recordings.

This preset automatically configures multiple enhancement options to maximize transcription accuracy. It prioritizes speech intelligibility over aesthetic audio quality, making it ideal when accurate text output is your primary goal.

When enabled, the tool applies moderate noise reduction, speech clarity enhancement, volume normalization, and converts to mono. These settings are specifically chosen to improve word recognition by both AI and human transcribers.

This option is recommended whenever you plan to have your audio transcribed, as it can significantly reduce errors and improve turnaround time. The enhancements make it easier to distinguish words, reducing ambiguity in the transcription process.

Stereo audio contains two channels (left and right), which is useful for music but often unnecessary for speech recordings. Converting to mono combines both channels into a single channel, reducing file size and improving compatibility.

Mono conversion is particularly useful for transcription, as it ensures all audio is equally balanced and prevents issues where dialogue might be panned to one side. It also reduces file size by approximately 50%, making uploads and downloads faster.

Many speech applications, including most transcription services and phone systems, work better with mono audio. Unless you have a specific reason to maintain stereo (such as spatial audio or music), mono is the better choice for speech content.