Convert HEVC to AU — Free Online Tool
Extract and convert the audio track from an HEVC/H.265 video file into Sun AU format, encoding it as 16-bit big-endian PCM audio (pcm_s16be). AU is a headerless-style legacy format native to Unix systems, making this conversion useful for Unix-based audio pipelines, legacy application compatibility, or extracting raw PCM audio from high-efficiency video files.
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FFmpeg Command
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Estimated output:
Conversion Complete!
DownloadHow It Works
HEVC (H.265) is a video-only container-agnostic codec, and raw .hevc files typically carry a compressed video stream with an accompanying audio track. During this conversion, FFmpeg discards the video stream entirely and extracts only the audio, re-encoding it into 16-bit signed big-endian PCM (pcm_s16be) wrapped in a Sun AU container. There is no video in the output — AU is a purely audio format. The audio is decoded from whatever codec was used in the source and re-encoded into uncompressed PCM, so the output will be significantly larger than the source audio but perfectly lossless in terms of audible quality at the 16-bit depth.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg program, the open-source multimedia processing engine that powers this conversion. In the browser-based tool, this runs via FFmpeg.wasm compiled to WebAssembly — no server involved. |
-i input.hevc
|
Specifies the input file — a raw HEVC/H.265 bitstream. FFmpeg will probe this file to detect the video stream and any accompanying audio tracks before processing. |
-c:a pcm_s16be
|
Sets the audio codec to 16-bit signed big-endian PCM, the default and most widely compatible audio encoding for the AU format. This produces uncompressed audio data in the byte order expected by Sun/Unix systems and Java's native audio API. |
output.au
|
Defines the output file with a .au extension, which tells FFmpeg to wrap the encoded PCM audio in a Sun AU container. The video stream from the HEVC input is automatically omitted because AU has no capacity for video data. |
Common Use Cases
- Extracting the audio commentary or soundtrack from an H.265-encoded video to feed into a Unix-based audio processing pipeline that expects AU-formatted input
- Archiving the audio portion of an HEVC broadcast recording in a simple, headerless PCM format for long-term storage on Unix/Linux servers without dependency on complex codecs
- Supplying raw PCM audio from an H.265 video to legacy Sun Solaris or early Unix workstation software that only reads AU files
- Preparing audio samples originally recorded as HEVC video (e.g., from a modern smartphone) for use in older Java applications, which have native AU playback support via the javax.sound API
- Stripping and converting audio from an H.265 surveillance or dashcam clip into AU format for ingestion into a Unix-based audio analysis or forensics tool
- Testing or debugging an audio pipeline that requires a simple, uncompressed big-endian PCM stream by sourcing audio from an existing HEVC video file
Frequently Asked Questions
No. AU is a purely audio format with no support for video streams, so FFmpeg automatically drops the video during this conversion. Only the audio track from your HEVC source is extracted and re-encoded into the AU file. If you need to retain the video, you would need to choose a different output format that supports both audio and video streams.
It depends on what audio codec was used in the source HEVC file. If the source audio was lossy (e.g., AAC or AC-3), that lossy quality is already baked in — converting it to PCM in AU format does not recover the lost information, but it also does not degrade it further. The output pcm_s16be encoding is itself lossless at 16-bit depth, so no additional quality is lost during the FFmpeg conversion step itself.
This is expected. The AU output uses uncompressed 16-bit PCM audio (pcm_s16be), which stores every audio sample as raw data with no compression whatsoever. Modern video files typically compress audio with codecs like AAC or Opus at very low bitrates. Uncompressed 16-bit stereo audio at 44.1 kHz consumes roughly 10 MB per minute, compared to a few hundred kilobytes per minute for compressed audio.
The AU format supports several PCM variants: pcm_s8 (8-bit signed), pcm_u8 (8-bit unsigned), pcm_alaw (G.711 A-law, a lossy telephony codec), and pcm_mulaw (G.711 mu-law, also lossy telephony). To use one, replace '-c:a pcm_s16be' with '-c:a pcm_mulaw' for example. The pcm_alaw and pcm_mulaw options produce much smaller files but at significantly reduced audio quality, typical of telephone-grade audio.
You can add '-ar 44100' to set the sample rate (e.g., 44100 Hz or 8000 Hz for telephony) and '-ac 1' to force mono or '-ac 2' for stereo. For example: 'ffmpeg -i input.hevc -c:a pcm_s16be -ar 8000 -ac 1 output.au' would produce a mono, 8 kHz AU file, which is a common configuration for legacy Unix telephony applications using mu-law or A-law codecs.
Yes. On Linux or macOS, you can use a shell loop: 'for f in *.hevc; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.hevc}.au"; done'. On Windows Command Prompt, use: 'for %f in (*.hevc) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.au"'. The browser-based tool processes one file at a time, so the FFmpeg command is particularly valuable for batch processing large collections locally.
Technical Notes
Raw .hevc files are bitstream containers for H.265-encoded video and may carry audio in various codecs depending on the muxer that produced them. The AU format, standardized by Sun Microsystems, uses a minimal 24-byte header followed by raw audio data, making it one of the simplest audio formats in existence. The default codec chosen here, pcm_s16be, uses 16-bit signed samples in big-endian byte order — matching the big-endian architecture of the SPARC systems where AU originated. On modern little-endian x86 systems, byte order is automatically handled by FFmpeg. AU files do not support metadata tags, album art, chapter markers, or multiple audio tracks, so any such information present in the HEVC source is silently discarded. The format also lacks the concept of a bitrate setting in the traditional sense — audio quality is determined entirely by bit depth and sample rate, not a compression bitrate. If your HEVC source contains multiple audio tracks, FFmpeg will select the first (default) track by default; use '-map 0:a:1' to select the second track explicitly.