Decode and convert mobile voice recordings and MMS audio files (.amr) into standard 16-bit linear PCM WAV audio directly in your browser. Pure offline execution with zero server uploads.
Detailed side-by-side format architecture for AMR Sound Stream / Module (.amr) and Waveform Audio File Format (.wav). Every conversion executes strictly within isolated browser memory without network transit.
Drag and drop your .amr file onto the working surface or use the system file dialog. File magic bytes are verified locally on device.
Select a quality preset (Lossless, Visually Lossless, Balanced, Smallest) or adjust advanced encoding switches in the options panel.
Click CONVERT to execute local WebAssembly compilation. Save the finished .wav directly to local storage, or stage it for chaining.
Prefer local shell scripting? Run this standard POSIX command on your machine to achieve equivalent transcoding without a web interface:
# Convert AMR to WAV locally
ffmpeg -i input.amr -codec:a pcm_s16le output.wavWorking with files across multiple web pages? The official convrtr Chrome Extension adds native Side Panel docking, right-click context menu media extraction, and visible viewport capture. Powered by the same zero-upload client-side WebAssembly engine.
Compare 3GPP Adaptive Multi-Rate (AMR) cellular speech compression with uncompressed 16-bit RIFF WAV audio: bandwidth, compression ratios, and fidelity.
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An in-depth audio engineering comparison between WAV and MP3. Compare bitrates, file sizes, audio frequencies, and production workflows.
AMR (Adaptive Multi-Rate audio codec) is an international 3GPP speech standard widely used in 2G/3G/4G cellular telephony, MMS text attachments, mobile voice memos, and legacy VoIP streams. It adapts between eight distinct bitrates ranging from 4.75 kbps to 12.2 kbps to preserve speech intelligibility under varying network conditions.
Yes. convrtr automatically detects the magic header (#!AMR\n for narrowband 8kHz and #!AMR-WB\n for wideband 16kHz), unpacks the speech mode frames, and synthesizes 16-bit linear PCM WAV audio.
Completely private. All AMR frame parsing, LPC speech synthesis, and WAV encoding run locally in your browser sandbox. No audio data is ever transmitted over the network.
AMR Sound Stream / Module (AMR) and Waveform Audio File Format (WAV) serve distinct roles across the digital workflow. Converting to WAV provides optimized compatibility, modern compression efficiency, or standardized playback across web browsers, operating systems, and media production environments.
The output format (WAV) is inherently lossless or supports lossless operation. Your original visual or audio fidelity is mathematically preserved without quantization artifacts.
Never. convrtr is architected with a strict zero-upload model. All processing, decoding, decompression, and encoding happen 100% locally on your machine using sandboxed WebAssembly and Web Workers. Your data never touches an external server, database, or analytics collector.
You can execute this conversion locally via command-line utilities such as ffmpeg, cwebp, or ImageMagick. For example: see the developer terminal recipe provided in the technical specifications dossier above. convrtr provides the identical local-first capability directly inside your browser without installing CLI packages.
Yes. You can drop multiple .amr files into the instrument to process them concurrently in browser Web Workers, or click 'Open in Master Studio' to queue mixed formats, configure individual targets, and export as a single ZIP archive.
Learn how Adaptive Multi-Rate (AMR-NB and AMR-WB) speech codecs compress voice with ACELP and how convrtr synthesizes them into 16-bit linear PCM WAV without server uploads.