Initializing... drag & drop files here
Supports: AVCHD
This walks you through pulling the audio track out of an AVCHD camcorder clip (the .mts or .m2ts files Sony and Panasonic cameras record) and decoding it to an uncompressed WAV file you can drop straight into a DAW or video editor. You'll end up with a standalone PCM .wav — no re-encoding of the video, just the soundtrack.
.mts or .m2ts clip onto the page, or click "+ Add Files" to browse. You can queue several clips and process them with the same settings.The single decision that matters here is the bit depth under Audio Codec, because WAV stores raw PCM samples and the profile sets how many bits represent each one:
Leaving Audio Sample Rate on Original avoids an unnecessary resample — AVCHD audio is generally captured at 48 kHz, which is already the standard for video work.
Converting to WAV changes the container and decodes the audio — it does not recover quality. If the source was Dolby AC-3, the WAV is a faithful copy of the already-compressed signal, not a restored uncompressed master; no tool can rebuild detail that AC-3 discarded at capture. The conversion also can't help with copy-protected or corrupted recordings, or clips whose audio stream is damaged on the camera's card. If you only need the soundtrack from a standalone Dolby Digital file (not a full AVCHD clip), AC3 to WAV is the more direct route. Files are uploaded over an encrypted connection, processed on our servers, and deleted automatically a few hours after conversion — no sign-up, no watermark, never shared or made public.
No. WAV is an uncompressed container, but it can only hold whatever the source already contained. If your AVCHD audio was Dolby AC-3 (the default on most consumer camcorders), the WAV preserves that audio exactly as decoded — it cannot restore frequencies or detail that AC-3 removed during the original recording. You get a lossless copy of a lossy source, which is ideal for editing but not a quality upgrade.
AVCHD camcorders record audio as either Dolby Digital (AC-3) or Linear PCM. AC-3 is the common choice on consumer models, running at 64 to 640 kbit/s across 1 to 5.1 channels. Some professional bodies record uncompressed Linear PCM (around 1.5 Mbit/s for stereo, supporting up to 7.1 channels). Either way, this converter decodes that track to PCM and wraps it in a WAV file.
WAV stores raw, uncompressed samples, so size scales with sample rate, bit depth, and channel count rather than content. A 48 kHz, 16-bit stereo track is about 11 MB per minute regardless of how quiet or busy the audio is. The original .mts is smaller because its audio (and video) were compressed. In our testing, a one-minute clip with 48 kHz 16-bit stereo audio produced a WAV close to 11 MB.
Match the source. If the recording was AC-3, 16-bit is the right target — higher bit depths only enlarge the file without adding information. If it was Linear PCM, 24-bit preserves the full resolution for mixing. Reach for 32-bit only when a specific tool requests high-bit or float input.
PCM WAV is one of the most broadly supported audio formats. It opens in Audacity, Adobe Audition, Pro Tools, Logic, and other editors, and plays in standard media players on Windows, macOS, Linux, Android, and iOS. That universal compatibility is the main reason to extract to WAV instead of leaving audio locked inside the AVCHD container.
Yes. The Trim control lets you set a start point and duration so the WAV contains just the segment you need, which is handy for grabbing a single line of dialogue or a sound effect from a long recording without exporting and cutting the full track afterward.