I wrote a DTS encoder

Let me announce a piece of software that I have published a week ago: dcaenc, an open-source DTS encoder. The package contains sources for a shared library, a command-line tool and an ALSA plugin.

DTS is one of the compressed formats that allow transfer of multichannel (e.g., 5.1) audio over SPDIF connections. The other common format is AC3. The SPDIF standard does not define a method for passing more than two channels of uncompressed PCM audio, so compression has to be used. Both AC3 and DTS are also used in DVD sound tracks.

Open-source decoders for both AC3 and DTS already exist: liba52 and libdca (side note: please don't use libdca, it is a security risk, there are some files that crash it or are decoded improperly). FFmpeg can also decode these formats. However, useful open-source encoders existed only for AC3: one in FFmpeg, and the other one (aften) based on it. The DTS "encoder" in FFmpeg was ported by someone else from my old proof-of-concept code that served as a tool to understand the DTS subband transform. It could only encode stereo PCM files into a valid DTS bitstream of the same bitrate, which is useless for any practical purpose. Now dcaenc provides a useful encoder that accepts multichannel sound and encodes it to the bitrate specified by the command line parameter.

As already mentioned, there are the following use cases for my encoder:

• On-the-fly encoding of multichannel PCM audio produced by arbitrary ALSA applications (e.g. games) for transmission via SPDIF
• Creation of DVD soundtracks and DTS CDs.

Some people ask me why I didn't integrate my encoder into FFmpeg instead of releasing it as a standalone package. Indeed, there are faster implementations of the basic DSP building blocks in FFmpeg, and the criticism that I reinvented a lot of wheels is valid. Integration with FFmpeg is indeed a desired long-term goal.

There are still several reasons why I decided not to integrate right from the beginning. First, I don't think that my work is in the necessary shape for integration yet. E.g., in FFmpeg, floating-point codecs are preferred, while my library currently uses fixed-point (I thought it would be beneficial for porting to the ARM architecture). Second and the most important reason: when the encoder is standalone, users can get it immediately and use it, without the hassle of replacing the distribution-provided FFmpeg package and potentially breaking distribution-provided software such as VLC that depends on it. Third, if I know that I wrote all the code myself, I can sell LGPL exceptions.

While dcaenc already produces "transparent" output at 1411 or 1536 kilobits per second, there is still room for quality improvement at lower bitrates. This is because the library does not yet use all compression possibilities offered by the DTS standard. I am going to implement at least linear prediction (incorrectly called ADPCM in the specification) and channel coupling in the future versions. Stay tuned!