ASIO Music Player Plugin (2 Viewers)

[Hr. Kvist]

Hi Symphy

First, thank you for a truly great plugin.

I have a problem: When using newer versions of your plugin and trying to play a FLAC file, Patchmix DSP interrupts and complains that an application is trying to set a samplerate of 192khz witch does not match my current session.

This problem does not exsist in version 0.1.1.0.

I'm using E-MU 1820m, PatchMix 1.81.0

Any ideas?

 

[Symphy]

Hi Symphy

First, thank you for a truly great plugin.

I have a problem: When using newer versions of your plugin and trying to play a FLAC file, Patchmix DSP interrupts and complains that an application is trying to set a samplerate of 192khz witch does not match my current session.

This problem does not exsist in version 0.1.1.0.

I'm using E-MU 1820m, PatchMix 1.81.0

Any ideas?

The newer versions attempt to auto-detect the supported samplingrates. This is done by setting them all and see if it failes or not. Apparently your device does not like that. I'll work out a solution for the coming version.

Symphy

 

[appleyk]

Hi, and thanks for all your work on this awesome plugin.

I'm having a problem with it, but I'm 99.99% sure that its actually a problem with my soundcard/drivers rather than the plugin, but thought I'd give it a go posting here in case anyone has any insight.

I'm using an ASUS Xonar D2 card, which is supposed to support ASIO natively. When I select the card itself in the pureaudio plugin config, all appears to be well, but when I actually try and play any music through MP (have tried mp3, flac, wav, and dts encoded wav) all I hear is random hight pitched tones, changing once every second or so.

I then tried using the ASIO4ALL program, when selecting this, with it pointing at the Xonar card, I can play MP3, flac and wav files with no problem (and there is a definite difference in the quality of the lossless files like this), however when playing dts encoded wav files, I get just a static noise. I assume this is because the ASIO4ALL is just passing the stream to the card, which is converting to PCM, rather than passing through as a bitstream.

I'm assuming that the problem is something to do with the cards ASIO implementation, hence it not working when selecting the cards ASIO drivers directly.

Incidentally, dts passthrough etc works fine for films.

I've had a search around on the net, and cant find any widely reported problems with Xonar ASIO, so I'm a little stuck as to what else I can look at. Oh, I tried the spdif program you posted to test playback of dts files, and get an unsupported format message, if that helps at all.

Any suggestions would be gladly recieved, and I apologise for posting this here when its almost certainly not a problem with your plugin.

Cheers

 

[Symphy]

I'm using an ASUS Xonar D2 card, which is supposed to support ASIO natively. When I select the card itself in the pureaudio plugin config, all appears to be well, but when I actually try and play any music through MP (have tried mp3, flac, wav, and dts encoded wav) all I hear is random hight pitched tones, changing once every second or so.

Sounds like a very weird problem. Could you post a debuglevel-logfile? Maybe that contains some clues.

I then tried using the ASIO4ALL program, when selecting this, with it pointing at the Xonar card, I can play MP3, flac and wav files with no problem (and there is a definite difference in the quality of the lossless files like this), however when playing dts encoded wav files, I get just a static noise. I assume this is because the ASIO4ALL is just passing the stream to the card, which is converting to PCM, rather than passing through as a bitstream.

Does your card include a DTS decoder?
The DTS encoded wav file format is originally designed to play from a standard audio CD and to be fed into an external decoder (receiver) over S/PDIF. It's basically DTS in a PCM package.

Oh, I tried the spdif program you posted to test playback of dts files, and get an unsupported format message, if that helps at all.

The data-mode used in that program is used for DVD playback (AC3 and DTS). However, because DVD is always 48 kHz, most devices only support 48 kHz when used that way, while the DTS encoded wavs are all 44.1....

In general I would try a program like foobar and see if you get the same problems.

Symphy

 

[Spragleknas]

What functionality would you like to be added?

Missing option: Crossfading (I just love it)

 

[appleyk]

I'm using an ASUS Xonar D2 card, which is supposed to support ASIO natively. When I select the card itself in the pureaudio plugin config, all appears to be well, but when I actually try and play any music through MP (have tried mp3, flac, wav, and dts encoded wav) all I hear is random hight pitched tones, changing once every second or so.

Sounds like a very weird problem. Could you post a debuglevel-logfile? Maybe that contains some clues.

I then tried using the ASIO4ALL program, when selecting this, with it pointing at the Xonar card, I can play MP3, flac and wav files with no problem (and there is a definite difference in the quality of the lossless files like this), however when playing dts encoded wav files, I get just a static noise. I assume this is because the ASIO4ALL is just passing the stream to the card, which is converting to PCM, rather than passing through as a bitstream.

Does your card include a DTS decoder?
The DTS encoded wav file format is originally designed to play from a standard audio CD and to be fed into an external decoder (receiver) over S/PDIF. It's basically DTS in a PCM package.

Oh, I tried the spdif program you posted to test playback of dts files, and get an unsupported format message, if that helps at all.

The data-mode used in that program is used for DVD playback (AC3 and DTS). However, because DVD is always 48 kHz, most devices only support 48 kHz when used that way, while the DTS encoded wavs are all 44.1....

In general I would try a program like foobar and see if you get the same problems.

Symphy

Thanks for your response. I'm attaching a debug logfile, it looks ok to me, but you'll know what you're looking for better than I can.

I just tried the same things using foobar, and basically it exhibits the same behaviour, which pretty much confirms its not your plugin at fault. Just for completeness, here are the steps followed with foobar.

Using latest foobar version, with ASIO and SPDIF plugins loaded.

When using ASIO, pointed directly at the XONAR, I get the same high pitched sounds as with pureaudio, no matter what format played.

When using ASIO pointed at ASIO4ALL, I get static when playing the dts wav file, other formats work ok. Definitely seems as though the card is outputting the bitstream in PCM format, rather than untouched. Its not decoding it, just outputting the bits as PCM.

When using SPDIF passthrough in Foobar (with the file renamed to have .dts extension as required by foobar) when foobar set to upsample to 48k if necessary, it works, sends the bitstream directly over the SPDIF interface, and my receiver decodes it succesfully. If no upsampling is done, then foobar reports that the SPDIF refused the stream at 44.1k.

So it definitely looks like an ASIO problem with the XONAR drivers to me. I'll try and moan at ASUS about it, although I dont expect to get anywhere, I'm sure if it was a widespread issue, someone else would have noticed by now.

I'm going to keep using the pureaudio plugin pointed at ASIO4ALL though, because although it wont play dts files, it does seem to make quite a difference in playback quality for flac files. Now if only it would drive the equaliser display on my IMON LCD, like the BASS player does

Thanks a lot for looking at the problem.

Cheers

Oh bother, I've found another problem. I play my music directly from the Shares view (Have test, and found the same problem with a playlist). When I select a song in a directory, when that song is finished, it should move onto, and play, the next song in order. However, it seems to hang the display for about a minute or so after finishing the song, then just stops and doesnt play any further. While its hanging, I can hear my remote commands being received by the PC, but nothing happens. After its stopped haging, I can move on and select the next song manually, but it only seems to play 1 song each time before stopping. Hmm, I though it was ok now, because after selecting the 3rd song manually, it went on and played the 4th ok, but then stopped again.

Is this a known problem, or should I post another log with this happening?

Cheers again

 

[johns11]

Hi all

Back in Nov there was some interesting discussion in this thread about merits of upsampling/oversampling music in either software or the DAC. I have found the following document, entitled "The art of building computer transports" v0.3, to be very interesting on this subject and I have certainly gained a little more understanding from it. It can be accessed at either of the following:

http://imageevent.com/cics/v03thear...q474t2.kizo_s?p=0&n=1&m=1&c=1&l=0&w=4&s=0&z=2

http://www.audioasylum.com/cgi/vt.mpl?f=pcaudio&m=23886

One suggestion offered is to avoid DACs or upsamplers which work in multiples of the original sampling rate. This is where I was reminded of the postings in this thread.

I am currently experimenting with software upsampling and doing sound comparisons. I’m tending towards recapturing my CD collection (currently in lossless .wmv) from the original CDs using EAC, and then storing 24 bit/96khz upsampled data on my disk (approx 1.5 Gb per CD). All made viable by the use of this plugin!

Just thought I'd just share this with people interested in this thread. Hope you get something from it also.

 

[Arn01805]

I am currently experimenting with software upsampling and doing sound comparisons. I’m tending towards recapturing my CD collection (currently in lossless .wmv) from the original CDs using EAC, and then storing 24 bit/96khz upsampled data on my disk (approx 1.5 Gb per CD).

What's the point in storing the upsampled data. Isn't it possible to do the upsampling 'on-the-fly'. This way you can store the double amount of CD's.

 

[GWI]

UD-10

Hi,


I installed a Trends Audio UD-10 today. I think there is a conflict between the UD-10 and Media Portal. I use the Pure Audio plugin with ASIO4all. When I stop a radio stream or a song, I get an error that says: Media Portal has encountered a problem and needs to be closed. However I still can continue playing. I attached the log files. Can someone help me?

 

[Symphy]

When using SPDIF passthrough in Foobar (with the file renamed to have .dts extension as required by foobar) when foobar set to upsample to 48k if necessary, it works, sends the bitstream directly over the SPDIF interface, and my receiver decodes it succesfully. If no upsampling is done, then foobar reports that the SPDIF refused the stream at 44.1k.

If i'm not mistaken, in this setup it is actually the foobar plugin decoding the DTS, not your external receiver. The foo_input_dts.dll is actually a decoder. What ends up in your receiver is either a stereo downmix (without any surround effects) or just only the front-left/front-right channels. DTS data can never be upsampled without decoding it first or without destroying it completely.
To have your receiver decode it, you would have to circumvent the dts plugin by NOT renaming files to .dts and make sure foobar doesn't change the data in any way. (no resampling, dithering, replaygain or anything).

I'll get back on the log and your other problem shortly

Symphy