Guide: Bitstreaming HDMI HD Audio Formats from your HTPC

I have almost everything in ISOs or BDMV format as well and I use a mix of PDVD10 and TMT3 (one other application but I'll mention that separately).  For the few MKVs I do have, typically lower quality Blu-ray content that I'll rip using MakeMKV, I am content with LPCM for now.

It is great that ffdshow tryouts supports DTS-HD and Dolby TrueHD bitstreaming but I find it too unstable with W7 x86-64 to be used reliably by my family.  I'm looking forward to a more stable build but until then I'll choose reliability.

@captain_video Glad Mount Image is still helpful

Mike, this is a great resource.  Thanks for putting it together.  I have a quick comment though.  I'm not sure it was totally clear from the beginning what the different audio paths are and their advantages and limitations.  Specifically, LPCM and Spdif didn't really get an introduction early on and I am worried that a newbie could get confused.  I'm gonna try my hand here so it will help newbies.

The different audio paths for audio from a PC to a Receiver

Stereo analog audio:
If you don't know what this is stop reading now and go back to the Apple store and buy something shinny that catches your eye.   I thought I would include it here to be complete.  

Spdif
Supports bitstreaming DTS, Dolby Digital, and Stereo (via stereo PCM).  Comes in either Optical form or digital coax.  Quality of both connection types are equivalent.  This is the go to way to connect your PC to your receiver if you do not wish to take advantage of the HD audio.  These formats are lossy (except for Stereo PCM) but still sound great.  When watching a Blu-ray you will get the highest quality of each codec.  For DTS you get 1.5Mbit/s, and for Dolby Digital (DD) you get 640Kbit/s.  This quality is higher than DVDs which use a 448k DD track and DTS at 754 Kbit/s.  At the bitrates DD and DTS use on BDs the differences between the lossy and the HD versions are small and many experts agree that full bitrate DD and DTS are indistinguishable to the uncoded source in a double blind test.  (I had to throw that in  :) )

MultiChannel Analog Audio
Mike did a great job of explaining this so the only thing I thought I would add is that you are relying on the audio card in your PC to do the decoding.  If your sound on your PC isn't very good, quality will suffer.  Also and perhaps more importantly for the average listener, is that your receiver will no longer be able to use its post processing.  You will have to do your calibrations in the PC's software, meaning you will have to set the channel levels, crossover frequencies, phases, and subwoofer level in your PC.   Using analog audio you will have to set your levels all over again and won't be able to use the auto calibration settings on your receiver.  That's why I personally suggest Spdif if you can't get HDMI to your receiver.  I guarantee you will be able to hear setup problems introduced by an improperly calibrated setup, while the differences between lossy Spdif and and their HD counterparts are much harder to differentiate.

LPCM
LPCM is like an unpacked zip file of audio.  DTS HD and TrueHD use lossless compression to make their files sizes smaller, just like a zip file can take your documents and pack them into a smaller package without losing any of your data.  LPCM is simply an unpacked version of whatever audio format you give it.  Whether you play a DTS track, a DD track, or a TrueHD track, LPCM will pass that signal like an unpacked zip file.  Hence, LPCM can either pass lossy data, or lossless audio.  It is more like a transport that a format.  LPCM is supported by HDMI versions 1.1 and later and as long as the PC doesn't muck with the data as Mike describes, will be exactly the same data as their DTS HD and TrueHD counterparts.  LPCM is actually the way that Blu-ray engineers intended you to get HD audio from BDs and that is why early players launched with HDMI 1.1 and 1.2 connections.  It wasn't until HDMI 1.3 that bitstreaming of DTS HD and TrueHD became possible at all.  An advantage to LPCM is that the player software will mux together multiple audio streams like the commentary and beeps from navigating the menus in your BD plater.  If you go bitstream, you can only get one audio stream at a time.  

Bitstreaming HD audio.  
Supported by HDMI 1.3 and later versions.  Advantages are it can't be screwed with by your player or Windows, and the pretty HD audio light on your receiver impresses your friends with your audio greatness.  The disadvantage is that you miss out on beeps and some other audio that may be muxed by your player.  I've never missed those but you may come across a situation that you want LPCM for two audio streams to be muxed together.  Coupled with a supporting video card, motherboard, or sound card, HD bitstreaming seems to just work, and works well.

Anyways, I hope that helps a bit for those who are unfamiliar with some of the terms used here.  Audio is pretty complex because of all the differnt formats, and all the settings you need to set properly to get it to work.

Remember that the experience you get from any audio source, whether it is a lossy DD stream on a DVD, or a TrueHD stream on a Blu-ray, is only as good as your audio system and its calibration.  Before you even start to worry about HD audio on your system, it is ABSOLUTELY required that you take the time to properly calibrate your speaker placement, speaker levels, subwoofer levels, phase, and speaker crossovers.  Even the best audio stream on a $5000 system will sound like absolute garbage on an improperly setup system, and even low bitrate lossy Dolby Digital on a DVD can sound absolutely amazing if you have taken the time to setup your system properly.

Another reason would be if you're like me and have an older receiver that has only analog inputs.

autoboy, thanks for the additional information.  You are absolutely correct, properly setup and calibration is incredibly important for both audio and video quality.  But then again if you don't know what is "good", sadly most people don't, then it is hard to calibrate.

Calibration shouldn't be done by ear so it really shouldn't matter what someone thinks sounds good.  Thankfully most new receivers come with an autocalibration mic and they tend to actually work pretty well.  The only limitation I have found is their settings for the crossover.  My Axiom m60s were set as large and pretty much no speakers should ever be set for large if you have a subwoofer.  Then, my center channel was set at 150hz which is far too high.  I set mine at 90hz and used test tones to verify that it was correct.  150hz is far too high and with a center channel especially, you run the risk of getting boomy voices localized to the sub.

While overpriced I don't consider the cost of the Auzentech to be that outrageous considering the high quality analog output it offers.  I've been a fan of Auzentech since XP MCE 2005 for two reasons: 1- Their sound cards were one of the first and few to offer bit-perfect audio over SPDIF; 2- They design their sound cards with swappable op-amps for all channels.

What is a little absurd is that PDVD is not bundled with the card.

To my knowledge, there is a huge difference in the ease of getting forced subtitles to work.  With TMT3 as a player for a Blu-ray image (optionally, a reduced size image using ClownBD), forced subtitles just plain work with no tinkering.  The last time I checked, you can get forced subtitles to work with MPC HC or WMP, but it was a major hack job.  I understand that most people try to minimize cost and thus avoid the commercial players whenever possible, but I finally gave up and paid the price for TMT3 to have forced subtitles work with no problems.

I don't think bit streaming introduces any new issues to the difficulty getting subtitles (forced or otherwise) to work properly in many players.

Retail BD solutions do offer a better experience for the feature (as they should), but now that the ffdshow DXVA video filter supports BD subs it should work in players that can leverage that (provided you're willing to deal with ffdshow's nuances).

I use PDVD for most of my BD viewing, but Sage V7 works pretty well if menus aren't important - it supports external SRT so for the ISO that get more use, and require subs, that is a good way to make it work.

FLAC is an excellent option if you can convert losslessly and get the audio off w/o it getting messed with - it's hard to go wrong.

That said, it's a lot more effort to do that each time then setuping up bit streaming once and just using that.  I didn't expect it to happen, but I've deleted most of my mkvs and just store the ISO now.

SageTV works fine with Reclock, so that's not a big deal.

Reclock works with WMC as well, although I think it may break something in WMC (possibly live tv, haven't tested reclock in a while so not 100% sure)

Last I checked it breaks LiveTV in WMC - so I wouldn't recommend it to most WMC users.

I could be misunderstanding something, since I'm still new to this whole HD audio thing, so bear with me, but the big caveat in my mind is the DRM (aka protected path).

Maybe the best way to get an understanding would be to ask a more direct question about my own setup?  Here's what I have:

1.  A/V receiver (no HDMI, no HD audio decoding)
2.  Analog audio from mobo to receiver
3.  HDMI (video) to TV via ATI 4650
4.  SageTV (v6, so far)
5.  Windows 7

Without using an external player (i.e.  PDVD, TMT3, etc.), what are my options for playing back HD audio via SageTV?

Analog is your only option.

1) Configure ffdshow for PCM output (i.e. disable all the bit streaming options), then setup the mixer options how you see fit (I disable it).
2) Configure Windows to use the analog outs with the proper number of speakers.
3) Done.

FWIW, I tried the mobo based analog route a while back and got much better sound quality with AC3/DTS over S/PDIF.  Integrated analog audio has gotten better since then (esp. on some boards), so your experience might be different - just make sure to test it out w/ the same track in both configs.

1) Configure ffdshow for PCM output (i.e. disable all the bit streaming options), then setup the mixer options how you see fit (I disable it).
2) Configure Windows to use the analog outs with the proper number of speakers.
3) Done.

FWIW, I tried the mobo based analog route a while back and got much better sound quality with AC3/DTS over S/PDIF.  Integrated analog audio has gotten better since then (esp. on some boards), so your experience might be different - just make sure to test it out w/ the same track in both configs.

Mind walking me through the setup for ffdshow a little bit?  I'm installing ffdshow-tryouts and deselected all the video options during install.  What should I be selecting for audio?  Just LPCM?  Any other setup after install?

Windows is already set up to use the analog outputs, so I'm good there.

I had to tweak the merits in order to get ffdshow to be used over Haali.  Once I had that, though, when I change to the HD audio stream, I get severe stuttering.  Selecting LPCM causes a horrific sound from the speakers.  I couldn't find any setting which didn't cause this.  Any idea what the cause might be?

You were right.  AC3filter is intervening.  I tried lowering the merit of it and raising ffdshow, but that actually caused the audio to cut out entirely.  Am I perhaps missing a codec or something?

On a better note, I re-ripped a Blu-ray to FLAC and that plays back just fine.  If it takes too much more time to troubleshoot this, I may just go ahead and re-rip everything to FLAC.  Hmph...  I was halfway done ripping.  ;D

Yep... I did that.

But, what's this?  What exactly needs editing?  I know my way around that file pretty well, but I have no clue what needs editing for this part.

Awesome!  Thanks for that!  I'll try those two entries when I get home tonight.  I honestly don't think those entries currently exist, which would be why I didn't know what to change.

Lately, I've noticed a high pitch whine from my speakers every once in a while.  I'm guessing that's due to the low quality of the onboard sound.  I've got an X-Meridian I never even opened, so I'm going to give that a shot.

On a somewhat related note, is there anything out there which can adjust the speaker levels automatically using a microphone?  My receiver is capable of that, but now that I'm using analog, that feature doesn't work.  I'm doing some Googling, but I'm stuck at what terms I should be searching for.

Thanks.  I found the room correction and set those.  They sure don't make it easy to find, though!

That whine was getting annoying, so I installed the Auzentech X-Meridian yesterday.  I was pleasantly surprised to find that Auzentech released 64 bit drivers for Windows 7 for this--fairly old--card.  My sole complaint is that their software doesn't have a room correction, which goes by distance.  Instead, they use an interactive map, where you drag the speakers and it changes the decibel level automatically.  But, since it doesn't tell you how far you've moved the speaker.  In the end, I found that the default layout seemed to sound best.

Another interesting point was that I found that my subwoofer wasn't set to the proper crossover frequency and corrected that.  The bass sounds much more accurate now.

Best advice I can give is to find a A/V store with a good listening room and try out AVR from a bunch of different vendors at different tiers in their lineup.  Every AVR has a slightly different sound, find the one that sounds the best to you (and fits w/ in the budget). 

I'm really lazy when it comes to setting up room correction, so personally I'd opt for an AVR that has Audyssey MultEQ, or something similar, to do it for you (I didn't check to see if the VSX-820 has that feature - it might).

For bit perfect audio, yes.

You don't need the X-Meridian - just the 4650, the Realtek ATI HDMI driver, and an HDMI cable.

You can use ffdshow to decode TrueHD to LPCM losslessly during playback, but AFAIK there aren't any free/OSS lossless DTS-MA decoders so if you don't want to convert to FLAC prior to muxing the MKV (or they already exist) your best option is to bitstream the core for DTS-MA tracks - the difference is slight.

I did plan to use ffdshow, as I currently am, but the DTS-MA bothers me.  (Mainly because I didn't keep the core in my rips.)  I have a license for TMT3 and it's installed, but I know that their decoders aren't available to 3rd party apps, by default.  However, this post seems to have found a way to make them available.  Assuming that works, I guess I wouldn't need to use ffdshow for any of the decoding, since I could use the TMT3 decoders for everything.

Even you can get the TMT3 decoders to work properly (I wasn't able to) the audio filter won't work the way you want.  It defaults to stereo output and expects the application to configure it for bit streaming each time it's used.  A while back I wrote a wrapper filter that could work around this, but it stopped working with one of their updates and I haven't had the need to revisit it.

If you mean DTS-MA decoded to LPCM during playback, AFAIK the answer is no.  I converted all of my DTS-MA tracks to FLAC prior to muxing to workaround it.

If you mean DTS-MA decoded to LPCM during playback, AFAIK the answer is no.  I converted all of my DTS-MA tracks to FLAC prior to muxing to workaround it.

I wanted to avoid as many conversions as possible when ripping, so I tried (and am still trying) to keep everything as original as possible.  I didn't want to convert to FLAC for fear of something else replacing it down the line.  I also wanted to minimize storage by removing all the stuff I didn't want (or see a need for), which was why I removed the core audio.  Oh well.