Cannot get MPC to transcode to DD/DTS

Howdy y'all, it seems my Windows 10 HTPC may be a bit of an epic FAIL. I've been having no success trying to get either MPC-HC or Kodi to properly transcode audio formats to DD/DTS for accurate playback of movies, etc. on my older AVR (JVC 8020VBK). My receiver only supports DD/DTS via Toslink (S/PDIF) and analog PCM. Obviously, analog isn't much of an option moving forward, so that leaves me with DD/DTS. I've tried changing just about every setting I can find in my filters\decoders and cannot get the transcode to work, I only get stereo to the AVR using MPC/Kodi for the newer signal formats. However, if I use WMP or the Movies & TV app, I do get some transcoding to work as hoped (for instance DD+).

Also, another problem that may/may not be related, I have a sample media file that has 3 audio layers, one of which is DD; but when I switch it to the DD layer, all I get is a flat, mid-frequency tone. Any ideas as to what's happening? I'm running the latest versions of MPC and the K-Lite filters/codecs (ffdshow/LAV).

I'm usually pretty savvy at tech troubleshooting and solving problems, but I'm sorta new to the codec experience, and one thing I'm struggling with is understanding how MPC uses the codecs in the first place. For instance, how do I know which one will be (or is being) used for any particular media file? It seems there are just too many variables involved for me to successfully troubleshoot. Perhaps someone can direct me to a guide on the logic behind the way that MPC interacts with the codecs so I can better understand how the software works.

UPDATE: I just found the guide for enabling passthrough in the FAQ and followed it for the various filters, still no luck. I also found a thread where someone suggested setting output to the internal device rather than the default, tried that also, again no luck...Here's the info from the MKV file I'm using to test with...
Video: MPEG4 Video (H264) 1920x1080 29.97fps [V: English [eng] (h264 high L4.1, yuv420p, 1920x1080) [default]]
Audio: DD+ 48000Hz 6ch 9216kbps [A: English [eng] (eac3, 48000 Hz, 6 channels) [default]]

First of all, you should only enable passthrough for AC3 and DTS, as those are the only formats that are supported when using S/PDIF connection. A common mistake is that people select all formats, which causes problems.
Once AC3/DTS work I can teach you how to transcode other formats to AC3.

Your sample file contains E-AC3 (DD+). That usually contains a AC3 (Dolby Digital) core that can be used for passthrough. But not always. So it is best to first test with a video file that contains normal AC3 or DTS.

Second thing to check is if the correct audio device is set as default in the Windows Control Panel.
Alternatively, you can explicitly select the desired device in MPC settings:
options > playback > output > audio renderer
(select the one with DirectSound prefix)

Thanks Admin...I did/do know about the AC3/DTS limitations of S/PDIF and have all my filters/decoders set accordingly. I'll let you know what I find out from your other tips. BTW, I wonder why there is a voting option for "Option to convert to AC3 for S/PDIF passthrough" on the K-lite website. Maybe there is a limitation with these filters?

I tested things a bit more, and I am able to get DD/DTS to pass-through fine. No transcoding is happening w/MPC though. It doesn't seem to matter what I start with (EAC3, AC3+, AAC, etc. etc.). I do get proper transcodes from Windows w/WMP and Movies & TV.

As I mentioned, the sample file located at http://s1.demo-world.eu/hd_trailers.php?file=dolby_amaze_lossless-DWEU.m2ts has a DD track but when I switch to it, it just renders a solid tonal signal, but it does show as DD on the AVR. I wonder if it's a problem with the file/track?

I have tried various audio renderers and nothing seems to change.

I should mention that I'm using a HDMI to Toslink switch b/w my PC and AVR. Windows seems to navigate it just fine, but maybe MPC is getting confused somehow?

well I finally figured it out, thanks to your post in another thread...your steps there really should be more prominent somewhere on the website. Thanks again.

now I'd be even happier if I could figure out how to adjust the volume!

nevermind...found it...can only use AVR I guess

Yes, the AVR is now responsible for the decoding, and thus also volume.

The FAQ has been updated:
http://codecguide.com/faq_configuration.htm#item7
Please have a look to see if the text is clear to you. Suggestions to improve are always welcome.

An option in the installer for automatic configuration is on the ToDo list.

The AC3 track you see with that sample file is actually the legacy core of the E-AC3 track. The modern audio formats have such cores for backwards compatibility.

There is an option in LAV Splitter to show such substreams as a separate track.

I suspect that the tonal signal is intentional so you can easily distinguish what is being decoded.

Thank you again for the help/info. When I have time I'll check the FAQ. One last remaining issue I am having involves testing AAC audio samples taken from the Fraunhofer website. Audio rendering is a bit whacked out. Audio playback seems to stutter/skip and mostly not output any sound at all. Any ideas?

You mean stutter with specific AAC files, or in general?

Pretty much all the files from the Fraunhofer site. Otherwise AAC files do seem to work okay.

Links?

I would not worry much about it. Might be files with unusual AAC features that aren't commonly used in real-life files.

Also check that you haven't forgotten to enable Resample (48000 Hz) in ffdshow settings.

I read through your updated FAQ, and it's pretty good/clear, but I do have one suggestion to make it a little clearer. When you first mention realtime AC3 conversion under 1) General info...you might cross-reference to item #3 in the list below.

Thanks again for all your help/work on this excellent software.

Thanks for the suggestion.