Convert DTS Audio to FLAC

Before you start

A DTS to FLAC converter changes the audio container without changing the content of the recording. You need the source file in DTS, WAV, or another container that holds DTS audio. The conversion usually runs locally on your computer with a tool such as FFmpeg. It does not require uploading the file to a service unless you choose a browser-based converter.

Check that the source has at least one audio stream. For files recorded from a camera, microphone, or video call, the audio may be mono, stereo, or multichannel. The output FLAC will keep the same number of channels unless you explicitly remix or downmix it.

Step-by-step instructions

Use the command line to control the sample rate and avoid changing the original timing. Open a terminal in the folder containing the source file.

• 1. Identify the source audio format and sample rate with ffprobe -i input.dts
• 2. Convert the file with ffmpeg -i input.dts output.flac
• 3. If the source is DTS inside a video file, use ffmpeg -i input.mkv -map 0:a -c:a flac output.flac
• 4. To keep a specific sample rate, add -ar 48000 before output.flac
• 5. To reduce loud background noise before encoding, add a filter such as -af highpass=f=80
• 6. For a stereo downmix from multichannel audio, add -ac 2 before output.flac

Modern FFmpeg versions can decode the DTS core and encode FLAC without a separate codec download. Older builds may require libdca or a DTS decoder enabled at compile time. If the command reports that the decoder is missing, update FFmpeg or use a build that includes DTS support.

Keep the original file during the first conversion. When you add filters, use a new output name instead of overwriting the source. This keeps the change reversible.

Check the output

After conversion, inspect the FLAC with ffprobe -i output.flac. Confirm that the codec is FLAC, the sample rate matches the source unless you changed it intentionally, and the duration is nearly the same as the original.

Duration can differ by a few milliseconds because of container padding. A large duration change indicates a decoding or sync problem, not normal conversion behavior.

FLAC is lossless for the decoded PCM audio, so the encoded result should not add artifacts. If the source DTS is lossy, the FLAC cannot restore detail that the DTS encoder removed.

Common problems and next steps

If the output FLAC is silent or noisy, check whether the DTS stream uses a variant that FFmpeg cannot fully decode. DTS-HD or DTS Express streams may require a different build or a dedicated extraction step.

When the source is quiet, do not raise the volume by adding excessive gain before fixing the input level. Normalize with a limiter or gain filter, then listen to a short section before converting the whole file.

A file from an input device such as a USB microphone may need a different sample rate than the recording software reports. Use the sample rate shown by ffprobe as the source value unless you have a reason to resample.

If you are preparing audio for an output device that expects a fixed rate, choose 44100 Hz or 48000 Hz early. Avoid converting later with repeated resampling, which can soften transients without changing the stated sample rate.