WAV gives our speech model the cleanest possible input — no compression artifacts to work around. But 96kHz sample rates and 24-bit depth don't improve speech transcription. We'll explain why, and transcribe your WAV accurately regardless.
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WAV files contain uncompressed audio — no lossy encoding, no compression artifacts, no frequency cutoffs. This gives speech recognition models a cleaner signal to work with compared to MP3 or AAC. But there are persistent myths: recording at 96kHz instead of 44.1kHz does not improve transcription accuracy for speech, and 24-bit depth offers no advantage over 16-bit for voice. Human speech lives below 8kHz and has about 50dB of dynamic range — well within 16-bit/44.1kHz capabilities. What matters is that WAV preserves what was captured without adding compression damage.
Drag and drop any WAV file — PCM, IEEE float, any sample rate, any bit depth. We handle 16-bit/44.1kHz studio recordings and 32-bit float DAW exports equally.
The uncompressed audio goes directly to our speech model with no decoder artifacts added. Internal processing resamples to the optimal rate for speech recognition.
Edit the transcript in the browser, then export as plain text, subtitles, or documents. Timestamps are synced to the original WAV timeline.
The real benefit of WAV: no lossy encoding artifacts. MP3 introduces pre-echo, bandwidth limiting, and stereo imaging artifacts. AAC introduces different but similar artifacts. WAV has none of these. For difficult audio (quiet speech, heavy accents, overlapping voices), this cleaner signal genuinely helps accuracy.
Human speech is concentrated below 8kHz. A 44.1kHz WAV captures frequencies up to 22kHz — well beyond what matters for speech. Recording at 96kHz or 192kHz captures ultrasonic frequencies that speech models ignore entirely. We resample high-rate files internally, so a 96kHz WAV and a 44.1kHz WAV of the same recording produce identical transcripts.
16-bit audio has 96dB of dynamic range. Human speech typically has 40-50dB of dynamic range. 24-bit provides 144dB of dynamic range — useful for music mastering, irrelevant for speech recognition. Your 16-bit recordings will transcribe just as accurately as 24-bit ones.
DAWs like Pro Tools, Logic, Ableton, and Reaper export 32-bit float WAV files by default. We handle these without issues — the float samples are processed directly without clipping or precision loss during internal conversion.
Broadcast and studio WAV files sometimes contain more than two channels — surround sound mixes, isolated microphone feeds, or multi-track bounces. We process all channels to capture speech from wherever it appears in the mix.
Radio broadcasts, voice-over sessions, and studio recordings are typically archived as WAV. Upload them directly and get the highest accuracy transcription from the highest quality source material.
Ethnographers, linguists, and oral historians recording with field recorders (Zoom H6, Tascam DR-40) typically capture WAV. These lossless recordings give the best possible input for transcribing challenging field conditions.
When you bounce a podcast, voice-over, or narration from your DAW, the export is usually 32-bit float WAV. Upload it directly — no need to convert to MP3 first. You'll get better results from the lossless source.
Oral histories, rare interviews, and historical recordings preserved as WAV represent irreplaceable audio. Converting them to searchable text creates a backup of the content that can be indexed, cited, and referenced without replaying the audio.
Transcribe WAV recordings from studio sessions, voice-over work, and broadcast productions. Your high-quality source material translates directly into higher transcription accuracy.
Convert WAV field recordings of interviews, focus groups, and oral histories into text for qualitative coding and analysis. Lossless audio preserves details that help with difficult-to-hear passages.
Transcribe the WAV master before compressing to MP3 for distribution. Get better accuracy from the lossless source, and use the transcript for show notes and content repurposing.
Convert WAV archives of historical recordings, oral histories, and institutional audio into searchable text. Make decades of audio content discoverable without replaying every file.
No. Human speech is concentrated below 8kHz, and a 44.1kHz sample rate captures frequencies up to 22kHz — more than enough. Recording at 96kHz captures ultrasonic frequencies that speech recognition models don't use. Internally, we resample high-rate audio to the model's optimal rate. A 96kHz and a 44.1kHz WAV of the same recording will produce identical transcripts.
Not for speech. 16-bit audio provides 96dB of dynamic range, while human speech typically uses 40-50dB. The extra dynamic range of 24-bit (144dB) benefits music production but provides no practical advantage for speech recognition. Save disk space and use 16-bit for speech recordings.
Absolutely not. Converting a lossy MP3 to WAV just wraps the already-damaged audio in a much larger uncompressed container. The frequency information and detail lost during MP3 encoding cannot be recovered by changing the container format. Upload the MP3 directly — it's smaller, faster to upload, and produces identical results.
Yes. DAWs like Pro Tools, Logic, Ableton, and Reaper often export 32-bit IEEE float WAV by default. We process float samples natively without clipping or precision loss. There's no need to bounce to 16-bit or 24-bit before uploading.
WAV stores uncompressed audio samples. A minute of 16-bit/44.1kHz stereo WAV is about 10 MB, while the same audio as a 128kbps MP3 is about 1 MB. The file is larger because no data is discarded — which is precisely why WAV gives speech models a cleaner signal to work with.
Yes. We process multi-channel WAV files by analyzing all channels for speech content. For surround sound files (5.1, 7.1), the center channel typically carries dialogue. For multi-track bounces, we process all channels to ensure no speech is missed regardless of which track it's on.
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