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Compress AAC and M4A audio files by lowering bitrate, sample rate, or channels to significantly reduce file size while keeping speech and music clear for web playback.
M4A Compressor helps you reduce the size of an M4A audio file by re‑encoding it to AAC in an M4A container with a lower target bitrate. Instead of guessing export presets in a full audio editor, you upload a file, choose a bitrate, and optionally set sample rate and mono/stereo channels to match the content type. For podcasts and voice notes, mono at 64–96 kbps often keeps speech clear while cutting size dramatically. For music, keeping stereo and using higher bitrates (128–256 kbps) usually preserves detail better. The tool runs the actual compression on the backend using FFmpeg so your browser stays responsive for larger files. An optional AI Assistant (triggered only when you click) can suggest safe settings based on whether your audio is speech, podcast, music, or mixed, and whether you care most about smallest file size or highest quality.
M4A Converter is a focused audio conversion tool for people who regularly work with M4A files and need them in other everyday formats without learning a full audio editor. You upload a single file—typically an M4A track exported from a phone, recorder, or editing app—choose an output such as MP3, M4A, WAV, OGG, or FLAC, and adjust simple controls for bitrate, sample rate, and mono or stereo channels. The backend uses FFmpeg to decode the source and re-encode it into the chosen format while reporting both the original and converted sizes so you can see the impact of your settings. This makes it easy to create small MP3 copies for sharing, lossless WAV or FLAC files for editing and archiving, or compatible M4A versions for mobile players, all from the same interface. When you are unsure which path to take, an optional AI Assistant can recommend a target format and quality based on your goal (sharing, speech, archiving, or compatibility) without exposing any model details in the browser.
AAC Converter lets you turn AAC and other common audio files into web-friendly formats like MP3, M4A, WAV, and OGG without guessing which export preset to use in a full audio editor. You upload a track once, choose an output format, adjust bitrate, sample rate, and channel layout, and immediately download a new file that is easier to share, host, or embed. The interface is tuned for spoken-word and everyday listening scenarios where you want to keep speech natural and music clear while avoiding oversized, high-bitrate masters that waste bandwidth. For users who are unsure which settings to pick, the optional AI Assistant can suggest safe defaults based on file length and use-case without exposing any model details in the browser.
AAC to MP3 focuses on a single, common task: converting AAC and related audio files into MP3 copies that are easier to share, host, and embed across a wide range of players and platforms. Instead of exposing every possible output codec, the interface fixes the output format to MP3 and gives you direct control over bitrate, sample rate, and mono or stereo channels so you can tune quality for podcasts, voice notes, or music previews. Behind the scenes, a stateless backend endpoint uses FFmpeg to handle the actual re‑encoding, and the tool reports the original size, new size, and percentage reduction so you always see the trade‑off between quality and file size. When you are unsure what settings to choose, an optional AI Assistant can suggest MP3 parameters tailored for spoken‑word content without exposing any model details in the browser.
AAC to WAV is focused on a single, predictable outcome: taking AAC and other compatible audio inputs and turning them into uncompressed WAV files that are easier to edit, archive, and pass through professional audio workflows. Instead of exposing every codec and container combination, the UI fixes the output format to WAV and lets you choose a sensible sample rate and mono or stereo channels so you can prepare material for DAWs, transcription tools, or post‑production without guessing about compatibility. Behind the scenes, a stateless backend endpoint uses FFmpeg to decode the source audio and re‑render it into linear PCM WAV while reporting the original and converted file sizes and percentage change so you can anticipate storage impact. When you are not sure which sample rate or channel layout to pick for spoken‑word material, an optional AI Assistant can suggest settings and a short rationale without exposing any model details or keys in the browser.
AI Voice Generator turns short written scripts into playable audio clips that you can preview directly in the browser. Instead of asking you to manage raw audio tracks, the interface focuses on the essentials for explainers, intros, and walkthroughs: a text box for your script, simple options for speaking style and speed, and a single Generate voice button that produces an audio preview you can replay. Behind the scenes, a backend endpoint calls an AI-powered text-to-speech engine, returning an audio payload that the tool converts into a temporary URL without exposing any model configuration or keys. For writers who are unsure how their copy will sound when spoken aloud, an optional AI Assistant can refine the script and suggest style and pacing settings while keeping all processing on the server.
Audio Balance Adjuster lets you correct left-right channel balance on existing stereo audio files without opening a full audio editor. You upload a track once, use a single pan slider to nudge the stereo image slightly left, right, or back to center, and then download a new copy with the adjusted balance. The backend uses FFmpeg to apply a gentle per-channel gain change rather than collapsing everything to mono, so you can fix off-center dialogue, uneven interview recordings, or distracting channel imbalances while keeping a natural stereo field. For spoken content, an optional AI Assistant can suggest a subtle pan value based on which side currently dominates, returning a short explanation while keeping all model calls on the server.
Audio Bass Booster lets you add controlled low-end emphasis to existing audio files without opening a full mixing session. You upload a track, choose how many decibels of bass gain you want, and click a single Boost bass button to generate a new version that feels fuller on small speakers and headphones. Behind the scenes, a backend endpoint uses FFmpeg to apply a focused equalizer band around the low-frequency region and returns an encoded audio payload along with original and processed file sizes, so you can see how much the transformation changed the file. For users who are unsure how aggressive the boost should be, an optional AI Assistant can suggest a gain value tailored to music on laptop or phone speakers while keeping all model calls on the server.
Audio Volume Booster helps you make quiet recordings louder with a simple, controllable gain setting. Upload an audio file, choose a boost level in decibels, and the backend applies an FFmpeg volume filter to increase loudness, optionally followed by a safety limiter to reduce obvious clipping on peaks. You can export the boosted result as MP3 for easy sharing or WAV for an uncompressed download, and the tool returns a ready-to-download file plus original and processed sizes so you can confirm the change quickly. This is useful for voice memos, interviews, lectures, podcasts, and music exports that were recorded too quietly or delivered at inconsistent levels. Because boosting also raises background noise, the tool keeps the workflow fast and repeatable: try a moderate boost first, listen, then adjust. An optional AI Assistant can suggest a conservative gain level and whether to enable the limiter for your use case and playback target, with AI processing handled securely on the backend and only triggered when you click the button.
MP3 to WAV converts a compressed MP3 file into an uncompressed WAV (PCM) file so you can edit, archive, or import audio into DAWs and video editors that prefer WAV. Upload an MP3, choose a target sample rate and channel layout (mono or stereo), and download the resulting .wav file. Because WAV is uncompressed, the output is usually larger than the original MP3, but it can be easier to trim, normalize, or process further without introducing new generation-loss from repeated lossy re-encodes. Conversion runs on the backend using FFmpeg to keep the browser responsive for larger uploads, and the results panel shows original size, output size, and percent size change. A Sample input button lets you test the full flow instantly with a real MP3 file. An optional AI Assistant (runs only when you click) can recommend practical WAV settings based on your content type and goal, and all AI processing is handled securely on the backend.
Audio Volume Reducer helps you quickly decrease the loudness of an audio file when it is too hot, uncomfortable to listen to, or likely to clip when combined with other tracks. Upload a file, choose how many decibels (dB) to reduce the level by, and the backend applies an FFmpeg volume filter to lower the signal safely and consistently across the whole file. This is useful for recordings that were exported too loud, clips that need headroom before mixing, or audio that must be brought down before you normalize a batch to a shared target. You can export the reduced result as MP3 for broad compatibility or WAV for an uncompressed intermediate. Because volume reduction affects the entire signal, the workflow is intentionally fast and repeatable: make a moderate reduction first, listen on your target device, then adjust the dB value until the level feels comfortable. An optional AI Assistant can suggest a conservative reduction amount based on your scenario (too loud, clipping, or inconsistent levels), and it only runs when you click the button with all AI processing handled securely on the backend.
Background Noise Remover helps you reduce steady noise like hiss, fan/AC noise, and room tone so speech and instruments sound clearer without requiring a full audio workstation. Upload an audio file, choose a mode (speech or music) and a strength level (light, medium, strong), and the backend applies an FFmpeg denoise filter tuned for fast, single-pass cleanup. The tool works best on constant background noise and is designed to be repeatable: start with medium, listen for artifacts, then switch to light if the result sounds watery or robotic, or to strong if noise remains too obvious. You can export the cleaned audio as MP3 for sharing or WAV for an uncompressed intermediate, and the output panel shows original and processed sizes plus the applied noise-reduction setting so you can verify what was done. An optional AI Assistant can recommend a safe preset based on your use case and noise type, but it only runs when you click the button and all AI processing is handled securely on the backend.
Chipmunk Voice Generator lets you create a classic high-pitched “chipmunk” effect from any audio recording by shifting pitch up in semitones. Upload a file, choose how many semitones to raise the pitch (from subtle to extreme), and optionally decide whether to keep the original duration. When duration is preserved, the tool applies a pitch-shift chain that compensates tempo so the timing stays similar; when duration is not preserved, the pitch shift also speeds up playback for a more traditional chipmunk feel. Processing is done on the backend with FFmpeg, so you can use it on mobile or desktop without installing software. Free plans support audio uploads up to 20MB, while paid plans unlock uploads up to 50MB. Download the result as MP3 for easy sharing or as WAV for an uncompressed intermediate you can edit further. An optional AI Assistant can recommend a conservative pitch setting and whether to keep duration based on your content type and desired intensity, and it only runs when you click the button with all AI processing handled securely on the backend.
Deep Voice Generator helps you make any voice recording sound deeper, richer, and more commanding by lowering pitch in controlled semitone steps and optionally adding a gentle bass emphasis. Upload an audio file, pick a depth setting from -2 to -14 semitones, decide whether to keep the original duration for a more natural result, and choose an output format (MP3 or WAV). The backend processes your upload with an FFmpeg pitch-shift pipeline designed to preserve timing when Keep duration is enabled; if you disable it, the output will slow down along with the pitch change for a heavier effect. You can also add a conservative bass boost (0–12 dB) to increase low-end presence for narration, gaming personas, voiceovers, or dramatic character lines. A Sample input button provides a real audio example so you can test the workflow instantly. An optional AI Assistant can recommend safe settings (depth, duration preference, and bass boost) based on your content type and desired vibe, but it only runs when you click and all AI processing is handled securely on the backend.
Echo Effect Generator adds a classic repeated echo to any audio file by delaying the signal and feeding part of the delayed sound back into the effect so repeats decay over time. Upload an audio file, set a delay time (35–2000 ms) to control how far apart the echoes are, adjust feedback to control how long the echoes keep repeating, and set a wet mix to control how loud the echoes are relative to the original. This is useful for voiceovers, podcast clips, dramatic narration, sound design, and creative edits where you want clear, rhythmic repeats without opening a full audio workstation. Processing runs on the backend with FFmpeg’s audio filters, so the tool works reliably on mobile and desktop without local installs. A Sample input button provides a real audio example so you can test the flow instantly, and the output panel shows the exact delay/feedback/mix settings used so you can reproduce the sound. An optional AI Assistant can suggest safe echo settings for common use cases (ambience, podcast, dramatic, metallic), but it runs only when you click and all AI processing is handled securely on the backend.
FLAC Compressor helps you reduce FLAC file size by re-encoding to a different FLAC compression level without changing audio quality. FLAC is lossless at every compression level, so the decoded audio stays identical; the trade-off is encoding speed versus file size. Upload a FLAC file, choose a compression level from 0 (fastest, larger files) to 12 (slowest, slightly smaller files), and optionally enable a verification pass for archival workflows. The backend processes your upload with FFmpeg’s FLAC encoder and returns a downloadable .flac file plus a size comparison so you can see whether recompression helped. This is useful when you have FLACs encoded with a low compression level, when you want to standardize a large library to a consistent level, or when you need smaller lossless files for storage and transfer without switching to a lossy format. A Sample input button provides a real FLAC example to test the full pipeline. An optional AI Assistant can recommend a sensible compression level based on whether you prioritize speed, smallest size, or long-term archiving, but it runs only when you click and all AI processing is handled securely on the backend.
FLAC Converter lets you convert audio files to FLAC for lossless archiving or convert FLAC to other common formats for compatibility and sharing. Upload an audio file, choose an output format (FLAC, WAV, MP3, AAC, or OGG), then download the converted result. When exporting FLAC, you can set the FLAC compression level (0–12) to trade encoding time for smaller files while keeping audio lossless. When exporting MP3/AAC/OGG, you can choose a target bitrate (64–320 kbps) to balance file size and quality for your use case. The backend runs the conversion with FFmpeg and preserves stream metadata when possible, so tags are carried through when supported by the target container. A Sample input button provides a real audio example so you can test the full pipeline instantly. An optional AI Assistant can recommend an output format and safe settings based on your goal (archive, editing workflow, sharing, or compatibility), but it runs only when you click and all AI processing is handled securely on the backend.
FLAC to MP3 converts a lossless FLAC audio file into a widely compatible MP3 you can share, upload, or play on devices that don’t support FLAC. Upload a FLAC file, choose an encoding mode, and download the result. The tool supports two practical workflows: VBR (variable bitrate) for efficient, quality-first encoding and CBR (constant bitrate) for predictable file sizes. In VBR mode, you select a quality scale (0–9, where 0 is the highest quality) and the encoder allocates bits where they matter most. In CBR mode, you select a target bitrate (64–320 kbps) for straightforward size control. Conversion runs on the backend with FFmpeg’s MP3 encoder and attempts to carry metadata through to the output when possible. A Sample input button provides a real FLAC example to test the flow instantly, and the result panel shows the exact settings used. An optional AI Assistant can recommend safe MP3 settings for music, speech, or podcasts based on your priority (balanced, smallest file, highest quality, or predictable size), but it only runs when you click and all AI processing is handled securely on the backend.
MP3 to OGG converts an MP3 audio file into an OGG file encoded with the Vorbis codec so you can use it in workflows that prefer open formats, games, or web playback. Upload a single MP3, choose a Vorbis quality value (0–10), and download the resulting .ogg file. Lower values prioritize smaller file size, while higher values prioritize fewer compression artifacts. Conversion is performed on the backend using FFmpeg to keep the browser responsive for larger uploads, and the result panel shows the original size, output size, and percent change so you can quickly compare quality settings. A Sample input button provides a real MP3 example to test the flow instantly. An optional AI Assistant (triggered only when you click) can recommend a conservative quality value based on your use case (music, speech, podcast, game audio, or web) and your priority (balanced, smallest file, highest quality, or compatibility), and all AI processing is handled securely on the backend.
OGG to MP3 converts an OGG audio file into a widely compatible MP3 you can play on virtually any device, upload to platforms that prefer MP3, or share with people who don’t have OGG support. You upload a single .ogg file, choose a target MP3 bitrate (64–320 kbps), then run the conversion and download the resulting .mp3. Lower bitrates reduce file size for faster sharing and smaller downloads, while higher bitrates preserve more detail and reduce compression artifacts for music and complex audio. Conversion runs on the backend using FFmpeg so large files don’t slow down your browser, and the results panel shows the original size, output size, and percent change so you can quickly see the tradeoff you selected. A Sample input button provides a real-world OGG example so you can test the full flow instantly. If you’re unsure which bitrate is appropriate, an optional AI Assistant (triggered only when you click) can recommend a conservative bitrate based on your use case (music, speech, podcast, game audio, or web) and your priority (balanced, smallest file, highest quality, or compatibility), with all AI processing handled securely on the backend.
Audio Bitrate Analyzer helps you quickly inspect the technical properties of an audio file before you compress, convert, or redistribute it. You upload a track once, the backend runs a lightweight probe, and the tool reports average bitrate, sample rate, channel count, codec, container format, duration, and file size in a compact panel. This saves you from guessing whether a file was exported at 64 kbps or 256 kbps, and it makes it easier to decide if re-encoding will meaningfully reduce size or simply waste time. For people who are unsure what bitrate they should target for spoken-word exports, an optional AI Assistant can suggest a range based on the current metrics and a podcast-style use case, while keeping all model calls on the server.
Audio Sample Rate Checker helps you confirm the sample rate of any audio file and quickly understand whether it matches a common delivery target. Upload a track once and the backend reads its technical stream info to report the detected sample rate in Hz/kHz, plus supporting context such as channels, duration, codec, and container format. The tool then compares the result against practical targets like music/streaming (typically 44.1 kHz), video/broadcast (typically 48 kHz), and high-resolution workflows (often 96 kHz or 192 kHz), so you can spot mismatches before importing into a session or exporting for a platform. This is especially useful when collaborating across DAWs or moving between audio-only and video timelines where sample-rate mismatches can trigger resampling or timing issues. An optional AI Assistant can generate a conservative export checklist and recommended settings for your selected target workflow, with all AI processing handled securely on the backend.
Audio Silence Remover helps you clean up recordings by removing silent sections based on a simple, controllable definition of silence. Upload an audio file and choose whether you want to trim silence only at the beginning and end (a safe default for voice memos, interviews, and exports with dead air) or remove long silent gaps throughout the entire file to tighten speech-heavy content. You can tune the silence threshold in dB, the minimum duration a quiet section must last before it is treated as silence, and a small padding value so cuts don’t clip the start of words. The backend processes your file with a fast FFmpeg filter pipeline and returns a downloadable MP3 or WAV along with original/new duration and time removed, so you can confirm the edit immediately. An optional AI Assistant can recommend conservative settings for common voice workflows, but it only runs when you click the button and is executed securely on the backend.
Audio Cutter lets you trim a single segment from an audio file by specifying start and end times in seconds. You upload a track, set the start and end range (or use the sample preset for the first minute), and click Cut audio to receive a new file containing only that segment, re-encoded as MP3. The backend uses FFmpeg to perform a precise time-based cut and returns the trimmed audio plus original and cut sizes and total duration. An optional AI Assistant can suggest a trim range (for example an intro, outro, or middle clip) based on the file duration; the AI only proposes start and end values and does not modify your file until you run the cut yourself.
AVIF Compressor reduces AVIF image file size while keeping controllable visual quality for web performance, SEO image optimization, and lightweight app delivery. You upload an AVIF image, set compression quality and encoder effort, optionally keep metadata, and export a compressed AVIF file from backend Sharp processing. This solves a common pain point where modern AVIF assets are still heavier than required for target use contexts like thumbnails, product grids, and responsive image sets. The tool includes a sample input action so users can quickly understand compression behavior before uploading production assets. For users unsure about optimal settings, an optional AI Assistant can suggest quality/effort tradeoffs based on whether size, quality, or balanced output is the priority. The workflow is explicit, fast, and suitable for repeated production optimization passes.
Audio Duration Calculator helps you quickly find the exact length of an audio file so you can plan edits, uploads, and publishing constraints without guessing. You upload an audio file and click Calculate duration to read its duration from media metadata on the backend using FFmpeg’s probe tools. The results are shown as both a readable timecode (minutes/seconds or hours/minutes/seconds for long files) and precise seconds with millisecond-level precision, along with the detected container/format label and file size for context. This is useful when you need to check whether a clip fits a time limit, estimate how much content you recorded, or verify that an export matches the expected length. For users who want quick guidance on what a given length is best suited for, an optional AI Assistant can interpret the duration and suggest practical next steps (such as using it as a short preview or trimming a highlight), while keeping all AI processing on the server.
Audio Fade In Tool adds a smooth fade-in to the beginning of an audio file so playback starts naturally instead of abruptly. You upload a track, choose a fade duration in seconds, and click Apply fade-in to generate a new MP3 where volume ramps from silence at time 0 to full level over the selected interval. The backend uses FFmpeg’s audio filters to apply the fade reliably across common formats and then returns the processed audio along with original and processed sizes so you can see how the output changed. A built-in Sample input button loads a short real audio example to help you understand the effect immediately. For users unsure what fade length to pick, an optional AI Assistant can suggest a fade duration based on the clip length and a typical content type, while keeping all AI execution on the server and requiring an explicit user action.
Audio Fade Out Tool adds a smooth fade-out to the end of an audio file so your track finishes cleanly instead of cutting off abruptly. Upload an audio file, set a fade duration in seconds, and click Apply fade-out to generate a new MP3 where the final segment gradually ramps down from full volume to silence. The tool processes audio on the backend using FFmpeg’s `afade` filter so it works reliably across common audio formats and avoids browser-specific quirks. You’ll get a ready-to-download result plus original vs processed file sizes to quickly validate the output. A Sample input button provides a real, short audio example so you can see how fade-out timing feels before using your own file. If you’re unsure what fade length fits your content, an optional AI Assistant can suggest a fade duration based on the clip length and a typical use case, and it only runs when you click the button.
Audio Joiner combines multiple audio files into a single continuous track with a simple upload → order → join workflow. Add two or more audio files, arrange them in the exact order you want using quick move controls, and click Join files to generate a downloadable MP3. The backend uses FFmpeg to concatenate audio reliably across common formats, then re-encodes the result into a consistent output that works well across devices and editors. The tool reports total input size, output size, percent size change, and the final joined duration so you can quickly confirm the merge completed as expected. A Sample input button loads a real multi-file example (intro, main, outro) to demonstrate the join behavior instantly. For planning and organization, an optional AI Assistant can suggest a joined title and clear labels for each segment based on filenames and durations, and it only runs when you explicitly trigger it.
Audio Loudness Analyzer measures how loud your audio actually sounds, using loudness standards designed for real listening rather than simple peak meters. Upload an audio file and run a server-side loudness pass to get Integrated loudness in LUFS, True Peak in dBTP, Loudness Range (LRA), and the total duration. These measurements are useful when you are preparing a podcast episode, music preview, video voiceover, or broadcast-style deliverable and want consistent playback volume across platforms. The tool also shows a quick target comparison for common use cases, so you can see whether your file is likely to be normalized up or down and whether true-peak headroom may be an issue after encoding. For users who want actionable guidance, an optional AI Assistant can recommend a sensible loudness target and true-peak ceiling for the selected use case and explain why, while keeping AI processing entirely on the backend.
Audio Loop Maker turns a single audio clip into a longer, repeatable loop in one simple step. Upload a file, choose how many times to repeat it, optionally add a small crossfade at the seam, and download a single MP3 output that plays continuously. This is useful for ambient beds, meditation backgrounds, game audio loops, short intros, and any sound that needs to run longer without manual editing. The backend uses FFmpeg concatenation to repeat your clip reliably across common input formats, and when crossfade is enabled it blends the end and start of repeats to reduce audible clicks. The tool reports estimated total input size, output size, percent size change, and final duration so you can confirm the loop length. An optional AI Assistant can suggest a practical repeat count and crossfade amount for your use case, but it only runs when you explicitly request it and it never changes your audio automatically.
Audio Merger mixes multiple audio files together into a single track, which is ideal when you need to overlay voice on background music, combine music layers, or blend sound effects into a bed. Upload two or more files, adjust each track’s volume using simple sliders, and click Merge audio to get one downloadable MP3. Unlike a join tool that plays files one after another, the merger overlays tracks at the same time and outputs a combined mix that follows the longest track duration. The backend uses FFmpeg’s mixing filters to apply per-track gain and combine streams into one consistent MP3 output. The tool also probes track durations so you can quickly see how long each file is and what the longest layer will be. For faster setup, an optional AI Assistant can suggest starting volume levels for common scenarios such as voice-over-music, layered music, or an effects bed, while keeping AI processing fully on the backend and only running when you request it.
Audio Metadata Editor lets you view and update the most common audio tag fields in one quick upload → edit → download flow. After you upload a file, the tool probes the container and codec, reads existing tags, and displays editable fields such as title, artist, album, album artist, genre, date/year, track number, disc number, and comment. When you click Update metadata, the backend writes your updated tag values into a new MP3 file for consistent playback compatibility and easy sharing. This is helpful when your music library has wrong titles, when podcast exports are missing artist/episode info, or when you want cleaner metadata for car stereos and media players. The tool includes a Sample input button so you can instantly see how tag detection and writing works. For power users, an optional AI Assistant can suggest clean, consistent tag values based on the filename and detected metadata, but it only runs when you explicitly request it and it never changes your file automatically.
Audio Metadata Viewer helps you quickly inspect what’s inside an audio file without opening a full editor. Upload a file and the tool probes technical details such as container format, audio codec, duration, and file size, then extracts common tag fields including title, artist, album, album artist, genre, date/year, track, disc, and comment. This is useful for troubleshooting “Unknown Artist” issues, verifying podcast exports before publishing, checking whether a download actually contains tags, and generating a clean metadata report for clients or archiving. The viewer includes a Sample input button so you can see real tags immediately, and it can export the detected metadata as a JSON report for documentation or workflow automation. For paid users, an optional AI Assistant can summarize missing fields and potential problems (for example, a title that looks like a filename) and suggest conservative tag values based on the filename and existing tags. AI runs only when you request it and is processed securely on the backend.
Audio Normalizer brings uneven audio to a consistent loudness target so your exports match common publishing expectations. Upload an audio file, choose a target preset such as Streaming/Video (-14 LUFS), Podcast (-16 LUFS), Voice/Speech (-19 LUFS), or Broadcast (-23 LUFS), then click Normalize audio to download a normalized MP3. The backend runs a two-pass loudness workflow with FFmpeg’s loudness normalization filter: it first measures Integrated loudness (LUFS), True Peak (dBTP), loudness range, and threshold, then applies the final normalization pass using those measured values for predictable results. The tool also returns the measured input loudness so you can see how far your file was from the selected target. This is useful for creators exporting from different editors, interviews with varying volume, or deliverables where perceived loudness needs to be consistent. An optional AI Assistant can recommend the best preset for your use case, but it only runs when you click and never changes your file automatically.
Audio Pitch Changer shifts the pitch of an audio file up or down in semitone steps while keeping the duration as close to the original as possible. Upload an audio file, choose a pitch shift from -12 to +12 semitones (one octave down or up), and click Change pitch to download a processed MP3. This is useful for transposing a backing track to a more comfortable key, adjusting a voice recording for creative effects, or matching reference tones during practice. The backend uses FFmpeg audio filters to perform a duration-aware pitch shift by resampling and then compensating tempo so playback length stays similar while pitch moves by the selected semitones. The tool includes a Sample input button so you can hear the effect immediately, and it reports the chosen semitone shift and pitch ratio in the result. An optional AI Assistant can suggest a starting semitone value for common goals (like “deeper voice,” “chipmunk effect,” or “match instrument key”), but it only runs when you click and never changes your file automatically.
Audio Recorder lets you capture microphone audio directly in your browser with a fast, no-install workflow. Pick a microphone, click Record, and the tool will handle permission requests and show a live timer while you record. When you stop, you can preview playback instantly and download the original recording file. For maximum compatibility, you can also convert the recording to MP3 using a public backend endpoint powered by FFmpeg, which accepts your audio upload and returns a downloadable MP3. This makes it easy to create voice notes, voiceovers, interview clips, or quick ideas that work reliably across devices. A Sample button provides a real audio example so you can test the end-to-end flow without using your mic. An optional AI Assistant can suggest a clean title and filename base for your recording based on your purpose and duration, but it runs only when you click and never records automatically.
Podcast Recorder helps you capture a complete podcast-ready episode in a simple browser flow: record your main audio, optionally add an intro clip and outro clip, and export a single stitched MP3 you can share anywhere. The tool uses your browser’s microphone recording for the main take, then sends your audio parts to a public backend endpoint powered by FFmpeg to concatenate the clips and encode a consistent MP3 output. You can also set optional MP3 metadata (title, artist, album, year, genre, and a short comment) so the exported file is labeled cleanly in players and file managers. A Sample Input button loads real audio clips so you can test the full record → stitch → download pipeline without touching your microphone. For creators who want a faster publishing workflow, an optional AI Assistant (runs only when you click) can turn your outline into structured show notes, chapters, and hashtags.