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[stealth82]

I never thought a software like Media Center 17 could ever exist, let alone be written. I wonder why, after some 5 years dedicated to HTPCs, I didn't give it a chance before. I can really tell you have a deep commitment for quality.
In fact what brought me here was the possibility of using ASIO, both for music and video files, along with DSP filters, above all DRC. Now, seeing that you have even added your own Convolver to the stack is giving me chills. Media Center is all what I've always wanted from a HTPC and much more.

Now, after the due praise, I'm asking for your help.
I'm still new to the forum, I searched a bit but couldn't find the answers I'm looking for to solve my current problems and I'm too excited to wait and see if they come out so I'm going to pose specific questions. If the questions have already been addressed somewhere else please forgive me and point me to them.

1. I love CoreAVC and saw I could include it into my configuration through the Custom Video Mode feature. I wrote a rule by making use of the condition field filled out as follows: codec(h264); codec(vc-1).
The problem is that CoreAVC doesn't kick in no matter how I specify the codec name, be it (h-264, h.264, H264 etc, I tried many combinations). The only way to make it work is to change those conditions in a more simple container(mkv). Obviously that condition doesn't seem so neat nor comprehensive. I would really prefer to specify the codecs there. What am I doing wrong? Is there a list somewhere of the supported codecs that can be typed into the condition field?

2. I have several hi-res music files (bought at HDtracks.com site) converted from FLAC to ALAC format - up until today I always used iTunes to play them. Most of them are even 192KHz/24bit but none of them play at all with Media Center. They just produce a sort of white noise when played. I don't understant why that happens. The Output Format filter is unchecked, I'm using the Asus Xonar ASIO driver, 32 bit-depth, 50ms latency. What's wrong with those files?

3.Is there any chance of seeing a routing option to front speakers under Room Correction filter when applying a crossover frequency? Now the available choices are: remove bass, move bass to sub, copy bass to sub.

Thanks in advance.
Guys you ROCK, hard.

[Matt]

First, thanks for the kind words.

1. I love CoreAVC and saw I could include it into my configuration through the Custom Video Mode feature. I wrote a rule by making use of the condition field filled out as follows: codec(h264); codec(vc-1).
The problem is that CoreAVC doesn't kick in no matter how I specify the codec name, be it (h-264, h.264, H264 etc, I tried many combinations). The only way to make it work is to change those conditions in a more simple container(mkv). Obviously that condition doesn't seem so neat nor comprehensive. I would really prefer to specify the codecs there. What am I doing wrong? Is there a list somewhere of the supported codecs that can be typed into the condition field?

It keys off the 'Compression' field.  What is in the compression field for the files in question (Action Window > Tag or details list with the 'Compression' column showing)?

2. I have several hi-res music files (bought at HDtracks.com site) converted from FLAC to ALAC format - up until today I always used iTunes to play them. Most of them are even 192KHz/24bit but none of them play at all with Media Center. They just produce a sort of white noise when played. I don't understant why that happens. The Output Format filter is unchecked, I'm using the Asus Xonar ASIO driver, 32 bit-depth, 50ms latency. What's wrong with those files?

I know the files are big, but could you provide a sample?  I'm matt at jriver dot com.

3.Is there any chance of seeing a routing option to front speakers under Room Correction filter when applying a crossover frequency? Now the available choices are: remove bass, move bass to sub, copy bass to sub.

Routing (and more) can be done with DSP Studio > Parametric Equalizer.  Remember you can drag effects to reorder them if you need to swap channels before or after DRC.

[JimH]

Thanks for the kind words and welcome to the forum.

1.  Did you try Red October HQ?  That would be your best bet.  But if you want to do your own Directshow filter setup, there is a wiki article.

2.  Are you using MC17?

3.  I don't know.

[stealth82]

It keys off the 'Compression' field.  What is in the compression field for the files in question (Action Window > Tag or details list with the 'Compression' column showing)?

You nailed it. Sorry I'm still testing out the whole application and I didn't setup avideo library yet so I was just opening video files directly. I tried to look for some information regarding the file but I just thought it had to be H264 after all. So, after importing the file I saw the app classified it as AVC1 instead
I now added AVC1 to the rules and everything works as expected . Great!

I know the files are big, but could you provide a sample?  I'm matt at jriver dot com.

I just sent you an email. Thanks.

Routing (and more) can be done with DSP Studio > Parametric Equalizer.  Remember you can drag effects to reorder them if you need to swap channels before or after DRC.

Well, I looked into those and, as a matter of fact, I've already set a low-pass filter for my subwoofer (40Hz) and an high-pass filter (60Hz) for my center channel. However I can't see a way to redirect everything below 60Hz for my center channel to my front speakers. Am I wrong?

[stealth82]

2.  Are you using MC17?

Yes I am.

[Matt]

Well, I looked into those and, as a matter of fact, I've already set a low-pass filter for my subwoofer (40Hz) and an high-pass filter (60Hz) for my center channel. However I can't see a way to redirect everything below 60Hz for my center channel to my front speakers. Am I wrong?

Bass Management is part of 'Room Correction'.  Normally this means moving bass to the subwoofer.

If you want to do custom routing, like moving the bass of the center to the sides, you could probably do it with Parametric Equalizer like:
Copy center to User 1
Low pass User 1
Add User 1 to Left
Add User 1 to Right

[stealth82]

Unbelievable. Thank you very much, I'll try that as soon as possible.

[Alex B]

2. I have several hi-res music files (bought at HDtracks.com site) converted from FLAC to ALAC format - up until today I always used iTunes to play them. Most of them are even 192KHz/24bit but none of them play at all with Media Center. They just produce a sort of white noise when played. I don't understant why that happens. The Output Format filter is unchecked, I'm using the Asus Xonar ASIO driver, 32 bit-depth, 50ms latency. What's wrong with those files?

I know the files are big, but could you provide a sample?  I'm matt at jriver dot com.

I reported a similar problem in April 2011 when the new M4A plugin was developed. The problem has not been fixed (I just re-tested MC16 and MC17).

Topic: MP4 Plugin (m4a/AAC & ALAC decoder) http://yabb.jriver.com/interact/index.php?topic=63215

...a new bug: My 24-bit high resolution QuickTime/qaac encoded ALAC sample files do not decode correctly (they produce only loud noise). My 24-bit ALAC samples that were encoded with dBpoweramp's ALAC encoder play correctly.

The header info may be wrong in the 24-bit QuickTime/qaac ALAC samples. Some tools report "16-bit". Despite this foobar2000 can play the files correctly. ...

A link to my QT/ALAC sample files is in the quoted post. (http://yabb.jriver.com/interact/index.php?topic=63215.msg425903#msg425903)

[Matt]

Regarding the ALAC issue, I got a sample "05 Wasted Time_ Reprise.m4a" from stealth82.  It plays correctly for me.

I then tried Alex's sample "ALACsamples.zip".  All 18 files sounded alright to me.

I'm stumped on this one.  Any other clues?

[Alex B]

I can reproduce the problem on two different machines (Windows 7 & XP). Only the 24-bit QuickTime ALAC samples play as static. As far as I can see, the output mode or DSP settings do not affect the behavior anyhow.

As a test, I tried the Disk Writer output mode. Here is a link to the output file, if you want to hear it:

http://cid-138ca589c542aeee.office.live.com/self.aspx/MC%20stuff/m4a%20samples/QT_ALAC_Disk_Writer_Output_ape.zip

I played the QT ALAC samples in this order:



The highlighted files play as static. As you can see, these 24-bit files show up as 16-bit.

The last file in the list is the output file. I configured the output to be 44.1/16 in DSP Studio to make possible to write a single output file.

A playback log of this test is attached.

[Matt]

I see the problem now.  I had M4A playing with a DirectShow filter instead of natively, due to some other testing I was doing.  Sorry about that.

Anyway, I'll follow up later once I know more about the actual problem.  I'm a little tied up right now, so it might not get fixed until tomorrow.

[Matt]

Next build:
Fixed: 24-bit ALAC files were not playing properly.

It's kind of silly how Apple implemented the ALAC MP4 headers.  There's already a standard sample size / bitdepth in the audio header.  But that number is wrong, and instead they overwrite it with a special ALAC atom.  That means the same information is in the header twice, and the values are different.

[stealth82]

Great, I can't wait to test out the new build!

In the meanwhile I've been able to setup the rerouting thing for my center channel successfully. Kudos to Matt, again!

[stealth82]

It seems that playback problems with hi-res music files have not been completely solved.

Playing 192KHz tracks seems to give different results... starting from the track I sent you some while ago I noticed that the recognized sample rate is 96KHz rather than 192KHz, this from the Audio Path. Also, as a cosmetic bug I think the display text shows the wrong bit-depth: 16-bit instead of 24-bit.

From another 192KHz album I have I can't play any of its tracks, the music doesn't produce a sort of pink noise as before but stutters badly. I'm not sure it's MC dependant though, because I tried the pink noise 192KHz 8ch test tone and it does the same thing... I tried different audio outputs - ASIO, WASAPI, DirectSound - with no effect.

[Matt]

Playing 192KHz tracks seems to give different results... starting from the track I sent you some while ago I noticed that the recognized sample rate is 96KHz rather than 192KHz, this from the Audio Path. Also, as a cosmetic bug I think the display text shows the wrong bit-depth: 16-bit instead of 24-bit.

From another 192KHz album I have I can't play any of its tracks, the music doesn't produce a sort of pink noise as before but stutters badly. I'm not sure it's MC dependant though, because I tried the pink noise 192KHz 8ch test tone and it does the same thing... I tried different audio outputs - ASIO, WASAPI, DirectSound - with no effect.

What happens if you pick a sample rate in DSP Studio > Output Format that you know works?

[stealth82]

If I downsample from 192KHz to 96KHz it works.

[Matt]

If I downsample from 192KHz to 96KHz it works.

Then I think that means your hardware (or driver) isn't handling 192 kHz well.

[stealth82]

That seems unlikely. iTunes can play the same files just fine. Unless iTunes downsamples those tracks to a lower sample rate - and it shouldn't because it handles 192KHz - I can't explain what is going on.

[Matt]

Unless iTunes downsamples those tracks to a lower sample rate - and it shouldn't because it handles 192KHz - I can't explain what is going on.

iTunes resamples everything using the Windows mixer.

[Alex B]

iTunes uses Direct Sound and the Windows XP mixer "helpfully" resamples everything if needed. This is normally not a bad thing because at least it can usually make things work, but the Windows resampler is not designed to be of the highest possible quality. It must be reasonably fast to work correctly on lesser computers.

However, the possible quality problems that are caused by the Windows XP resampler are often exaggerated. By default XP uses the so called high-end multipoint interpolation. Microsoft states that its signal to noise ratio is better than 90 dB. Something that is -90 dB is normally considered to be utterly inaudible.

From another 192KHz album I have I can't play any of its tracks, the music doesn't produce a sort of pink noise as before but stutters badly. I'm not sure it's MC dependant though, because I tried the pink noise 192KHz 8ch test tone and it does the same thing... I tried different audio outputs - ASIO, WASAPI, DirectSound - with no effect.

It is rather odd that Direct Sound does not work with MC. The situation should not differ from iTunes playback. Kernel Mixer should take care of any format incompatibilities.

Could you please try the samples that I linked earlier:

Possibly a new bug: My 24-bit high resolution QuickTime/qaac encoded ALAC sample files do not decode correctly (they produce only loud noise). My 24-bit ALAC samples that were encoded with dBpoweramp's ALAC encoder play correctly.

The header info may be wrong in the 24-bit QuickTime/qaac ALAC samples. Some tools report "16-bit". Despite this foobar2000 can play the files correctly.

The samples are available here: http://cid-138ca589c542aeee.office.live.com/self.aspx/MC%20stuff/m4a%20samples/ALACsamples.zip  (It is a "skydrive" webpage and the actual download link is there.)

I have retested these sample files. The latest MC17 build decodes them correctly and the sample rate & bit depth fields show correct values when the samples files are imported as new library entries.

[stealth82]

That is untrue. It's quite a while since iTunes started using WASAPI under Windows Vista/7.

[Matt]

That is untrue. It's quite a while since iTunes started using WASAPI under Windows Vista/7.

WASAPI resamples through the Windows mixer, unless exclusive mode is used.  iTunes does not use exclusive mode.

[Alex B]

That is untrue. It's quite a while since iTunes started using WASAPI under Windows Vista/7.

I actually wrote about XP because for some reason I thought you were using it. I had just quickly read though a few recent threads and probably someone in another thread was using XP. Of course I should have noticed that you mentioned about trying also WASAPI.

I am now checking how iTunes works on Windows 7. At least on XP iTunes does not have any means for selecting the output mode. On Windows 7 WASAPI would not work as an only option because if there is a sample rate mismatch resampling would not occur. I'll report my findings soon. WASAPI is an acronym of "Windows Audio Session API", but as an output mode I have seen it used only as a name of a mode that bypasses the Windows mixer.

[Matt]

I now checking how iTunes works on Windows 7. At least on XP iTunes does not have any means for selecting the output mode. On Windows 7 WASAPI would not work as an only option because if there is a sample rate mismatch resampling would not occur. I'll report my findings soon. WASAPI is an acronym of "Windows Audio Session API", but as an output mode I have seen it used only as a name of a mode that bypasses the Windows mixer.

You can use WASAPI in shared or exclusive mode.

Shared mode runs through the Windows mixer, and it's possible iTunes uses this.

That would be functionally identical to DirectSound so it doesn't really change what you've said (DirectSound may just be a WASAPI shared wrapper on Vista and newer).

[Alex B]

My understanding has been that in the WASAPI shared mode the stream format must match the device's set format. I'll test this in a minute. I had to install iTunes on my Windows 7 machine and restart it.

[Matt]

My understanding has been that in the WASAPI shared mode the stream format must match the device's set format. I'll test this in a minute. I had to install iTunes on my Windows 7 machine and restart it.

You're right.  It's the clients job to deliver data in the mixing format in shared mode.

I was thinking you could ask WASAPI to resample, but that may not be possible.

[Alex B]

I have now run some tests. Here are my findings.

- iTunes does not have an option for setting the output mode on Windows 7.
- QuickTime player has a "WASAPI" playback mode option.
- A Google search found some forum discussions about setting the WASAPI output mode in QuickTime for allegedly enabling it also in iTunes. In the past QT was a required part of an iTunes installation and the QT WASAPI setting may have done something, but iTunes 10.5 does not include QT anymore.
- when the Windows advanced speaker setting was set to 24-bit/44.1 kHz iTunes played all test tracks from 48 kHz up to 192 kHz at 44.1 kHz. I verified this by monitoring the display on my Terratec sound card's control panel. Similarly when the Windows setting was 96 KHz all test files played at 96 kHz.
- The QuickTime player did exactly the same as iTunes even though its "WASAPI" mode was enabled.
- MC's WASAPI exclusive mode switched the sound card's displayed sample rate automatically and correctly according to the played file independently of the Windows sample rate setting. My 176.4 and 192 kHz test files played at 96 kHz because I had MC set to resample them to 96 kHz. (The tested sound device supports up to 96 kHz.)
- MC's WASAPI shared mode could play the files only if MC's output sample rate matched the Windows advanced speaker sample rate setting, i.e. the Windows mixer did not resample the files.

The sound card in my test was Terratec DMX 6fire 24/96.

[stealth82]

So, what's the conclusion? 

However, I have strong doubts about the fact my audio card can't handle 192KHz: it's an Asus Xonar ST. It definitely should.

[stealth82]

All right, I found the cause. Whenever I set the number of speakers - in the Output Format - from 2 channels to 2.1/4/5.1 any of the 192kHz tracks stutter. It doesn't matter if I disable JRSS or not. Just setting the number of speakers to 2.1/4/5.1 makes the song stutter.

Any ideas?

[Alex B]

A quick Google search revealed that ASUS Xonar Essence ST is a 2 channel device. I don't know what is supposed to happen when the set channel number exceeds the number of available hardware channels.

I also don't know why only the highest sample rate, 192 kHz, seems to cause problems when more than two channels is set (if I have understood you correctly).

In any case it is good that you found a cure.

[stealth82]

A Xonar ST can be paired with its H6 module to transform it into a 7.1 sound card.

Actually I haven't found a cure since I don't find it practical to switch from 5.1 to 2.0 every time I want to listen to a 192kHz track. And that would be less of a cure I think.

As I said I can play 192kHz tracks, even the very same MC 192kHz 6 channel test clip, with another player and it works perfectly fine. The problems just happens with MC.

[mojave]

I have an ST and H6 daughter card. I think I could play 192 kHz FLAC files from HDTracks in multichannel output. I haven't had the cards installed since last summer, but could test next week with the files converted to ALAC.

[stealth82]

Thanks, that would be very kind of you. Looking forward to hearing from you soon

[stealth82]

I shot a video showing my current problem if that can help.

It's available here: http://dl.dropbox.com/u/2777693/mc17.mkv

You might also glimpse how the OSD incorrectly shows the track's bit-depth: 16-bit instead of 24-bit.
This is what I put on the second row:

Code: [Select]

[Elapsed Time] // [Total Time] - [Bitrate (labeled)] - [Sample Rate] kHz [Bit Depth]-bit - [Channels] ch   [PN Position] of [PN Tracks]  [Zone]