New out_asio is coming, stay tuned!

[M.J.]

Hi folks,

I've managed to, let's say it, "continue" Otachan's work on out_asio and it seems like there's some good progress already - I've been using this new version for about 3 weeks now on my Win7/Win8.1 PC and it works flawlessly.

So what's new in this version:

- Latest ASIO 2.3 support
- Windows 7/8.x support (settings are saved/restored correctly)
- Compiled with latest ICL (v15) + SSE3/SSE4.1/AVX/AVX2 support
- Larger buffer size support
- Revamped configuration dialog
- Minimalistic, no-resampler version is also available

In progress:

- Volume/pan control (although this is not bit-perfect at all)
- Various thread/playback optimizations

Webpage for this new version is currently being setup, so stay tuned.

Screenshots for now:





-M.J.

[ravermeister]

cool!!
hope it is a good replacement for the maiko wasapi plugin

[pbelkner]

Originally Posted by M.J.

I've managed to, let's say it, "continue" Otachan's work on out_asio and it seems like there's some good progress already

Very good news!

[Raj_09]

Very good news indeed. I am looking forward to trying the new version.

[ravermeister]

me too, is there yet any test version we could try?
sadly the much promising maiko plugin is dead,
and I really hope there will be a worthy replacement for it.

[ravermeister]

...updates...?

[bawheid]

I just signed up to register my interest. I'm glad development for Winamp is still alive!

[Branddy]

Originally Posted by bawheid

I just signed up to register my interest. I'm glad development for Winamp is still alive!

As I Did.

Glad To see someone Interrested in ASIO plugin Dev'

[Branddy]

any update from the author ?

[DrO]

give the guy a chance to get things to a point where it's ready (as it's only been a month since the initial post and people often have lives outside of these things which can lead to slow downs in things finally being released).

ravermeister: you do know that you're mixing up asio and wasapi which are two different systems and so are not truly comparable between this (or it's original version) and the maiko solution.

[ravermeister]

@DrO
thx for the hint,
yes you're right, maiko uses the WASAPI (new Windows Sound API since Vista)
and ASIO is another solution by Steinberg.

so you're right, they are not truly comparable.
to be honest, the maiko plugin in Combination with the ffsox plugin by pbelkner does the job really fine.
but I am always on the run for the ultimate sound quality, and hope the ASIO plugin
has some advantages in the Stereo upmixing, (as I don't like CMSS3D from creative and DTS Connect)
and switching the Samplerate
(I have to sync the Samplerate for maiko, to prevent the long delay when Samplerate changes)

and by the way, I cross the fingers that the ffsox plugin supports ID3 Tagging someday

but you're right good things take their time, but I think it is not
wrong to show the Developer that his work is appreciated

Kind regards
raver

[DrO]

it'd be simpler if ffsox just passed off metadata / artwork handling to the existing native input plug-ins rather than re-inventing even more of the wheel (i.e. sending any requests to the metadata exports it receives to those of the appropriate input plug-in) seeing as metadata is going to be the same whichever plug-in tries to do things - at least going that way would make things more compatible with what Winamp is expecting even if the audio playback aspect is different.

[M.J.]

Hello all,

it's been busy life, but I haven't forgotten :-)

First beta is released, feel free to download and give it a test ride until 31/1/2015. Please let me know how do you feel about it.

I want to continue development of this plugin and your feedback is absolutely needed!

Thanks.

http://bit.ly/1yqdpSr

-M.J.

[A11ectis]

Installed the plugin and ASIO4ALL. No real idea what either of them really do, but they seem to work as described.

[M.J.]

Roughly 30 downloads now. Please let me know if you have tried it and whether it works for you.

Thanks!

-M.J.

[Raj_09]

Cool. I am trying it now. What is the difference between the SSE2 version and the AVX version? I am testing it on my HP laptop with Realtek audio (HP says Beataudio) and Asio4all driver.

[M.J.]

Originally Posted by Raj_09

Cool. I am trying it now. What is the difference between the SSE2 version and the AVX version? I am testing it on my HP laptop with Realtek audio (HP says Beataudio) and Asio4all driver.

AVX build is better suited for Intel Core i7 CPUs (>= Sandy Bridge), since the code is compiled & optimized for this architecture. If it doesn't run, please use SSE2 version instead. Standard x86 build ("normal") is included for convenience.

[Raj_09]

@M.J: Fantastic work. Nothing to note for the time being. I will be on the lookout for issues. I can't believe I have actually switched from Maiko after all this time.

[Raj_09]

Originally Posted by Raj_09

@M.J: Fantastic work. Nothing to note for the time being. I will be on the lookout for issues. I can't believe I have actually switched from Maiko after all this time.

Edit @ 09 Dec:

I can't play some mp3 files. There is no way I can identify what is different - whether it is the ID3 tags or cover art embedded. They simply would not play. Maiko plugin plays them fine.

[M.J.]

Originally Posted by Raj_09

Edit @ 09 Dec:

I can't play some mp3 files. There is no way I can identify what is different - whether it is the ID3 tags or cover art embedded. They simply would not play. Maiko plugin plays them fine.

Thanks for reporting. Can you try changing buffer size, both under plugin and/or under ASIO4ALL, to see if it works with some combination?

[Raj_09]

Originally Posted by M.J.

Thanks for reporting. Can you try changing buffer size, both under plugin and/or under ASIO4ALL, to see if it works with some combination?

Thanks I have tried maxing the buffer within Winamp and Asio4all but it still would not play. I will try different combinations and see.

[Raj_09]

Originally Posted by Raj_09

Thanks I have tried maxing the buffer within Winamp and Asio4all but it still would not play. I will try different combinations and see.

Edit @11 Dec

I seem to have found it. When I disable a DSP plugin Enhancer, it plays the file correctly. I can put the effect back while playing the song and it kicks in with a delay sometime mid song.

That is strange as I did not experience the same issue with other files and they played fine on out asio with enhancer dsp plugin. The same file also plays with the dsp on maiko exclusive mode. That song is a vbr encoded song. Most other songs are cbr encoded.

Well, MJ the following combination works: max 2mb in the out_asio plugin buffer and default 512 samples in ASIO4ALL.

[Branddy]

Originally Posted by Raj_09

I can't play some mp3 files. There is no way I can identify what is different - whether it is the ID3 tags or cover art embedded. They simply would not play. Maiko plugin plays them fine.

Same Here with ASUS XONAR Essence STX II.

I think it's related to the implementation of ASIO ASUS in their (crappy) Driver, because others ASIO plugins have the same problem but maiko WASAPI Plugin play find all kind of files. (Directsound)

Config:
Win8.1 x64
ASUS DRIVER 8.1.11.5
ASIO 0.99b (SSE2) Buffer size to 2047, Thread Priority to Norrmal.
ASIO Driver lantency 20ms, Depth 24bit.
Winamp v5.666 Allowed to play 24bit / surround.

File details:
Size payload: 2858004 bytes
Header found: 2541 bytes
Encoding time: 576, completion zero 1152
Duration: 178 seconds
MPEG-2 Layer 3
128 kbit, 6838 frames
22050Hz Stereo

No DSP and/or Input plugins

For the Rest, the sound is clearer with your ASIO plugin than WASAPI.

Good Work.

[M.J.]

Hello all,

Originally Posted by Raj_09

Edit @11 Dec

I seem to have found it. When I disable a DSP plugin Enhancer, it plays the file correctly. I can put the effect back while playing the song and it kicks in with a delay sometime mid song.

That is strange as I did not experience the same issue with other files and they played fine on out asio with enhancer dsp plugin. The same file also plays with the dsp on maiko exclusive mode. That song is a vbr encoded song. Most other songs are cbr encoded.

Well, MJ the following combination works: max 2mb in the out_asio plugin buffer and default 512 samples in ASIO4ALL.

Originally Posted by Eyeof Astranger

M.J., I've been using the v0.99b-AVX for a week with only 1 issue:

There is a perceptible gap in-between tracks that should be gapless.

When I play live tracks (i.e. Metallica's Through the never) or songs that fade into each other (e.g. Foo Fighters' Subterranean and I am a river from their last album), I ear what seems to be a 1 second gap when Winamp switch from track A to track B. Any such cd ripped has the same problem with the AVX driver.

This problem doesn't happen with either the DirectSound or Waveout drivers.

In case you need details on my computer, here it is:

Processor: Intel Core i7-3960X
Mobo: ASUS P9X79 WS
Ram: Kingston HyperX Genesis, Quad Channel 16Gb DDR3 1866Mhz CL9 DIMM
Soundboard: Edirol FA-66, connected through Firewire port
Windows 7, 64-bit

Thanks for trying it out, I'll see what I can do about it.

Originally Posted by Branddy

Same Here with ASUS XONAR Essence STX II.

I think it's related to the implementation of ASIO ASUS in their (crappy) Driver, because others ASIO plugins have the same problem but maiko WASAPI Plugin play find all kind of files. (Directsound)

Config:
Win8.1 x64
ASUS DRIVER 8.1.11.5
ASIO 0.99b (SSE2) Buffer size to 2047, Thread Priority to Norrmal.
ASIO Driver lantency 20ms, Depth 24bit.
Winamp v5.666 Allowed to play 24bit / surround.

File details:
Size payload: 2858004 bytes
Header found: 2541 bytes
Encoding time: 576, completion zero 1152
Duration: 178 seconds
MPEG-2 Layer 3
128 kbit, 6838 frames
22050Hz Stereo

No DSP and/or Input plugins

For the Rest, the sound is clearer with your ASIO plugin than WASAPI.

Good Work.

Here you're simply trying to play an MP3 file with 22050Hz sample rate, and although Xonar Essence STX's II DAC chip (TI PCM1792) natively supports sample rates from 10kHz, Xonar's ASIO driver seems to support native sample rates only from 44.1kHz and upwards - see http://www.asus.com/us/Sound_Cards_a...pecifications/ - so this is definitely a limitation of Xonar's ASIO driver.

-M.J.

[Eyeof Astranger]

M.J., I've been using the v0.99b-AVX for a week with only 1 issue:

There is a perceptible gap in-between tracks that should be gapless.

When I play live tracks (i.e. Metallica's Through the never) or songs that fade into each other (e.g. Foo Fighters' Subterranean and I am a river from their last album), I ear what seems to be a 1 second gap when Winamp switch from track A to track B. Any such cd ripped has the same problem with the AVX driver.

This problem doesn't happen with either the DirectSound or Waveout drivers.

In case you need details on my computer, here it is:

Processor: Intel Core i7-3960X
Mobo: ASUS P9X79 WS
Ram: Kingston HyperX Genesis, Quad Channel 16Gb DDR3 1866Mhz CL9 DIMM
Soundboard: Edirol FA-66, connected through Firewire port
Windows 7, 64-bit

[Branddy]

HI MJ,

Yes, you're right.

I've tried some files lower than 44.1Khz and all of them cannot be played.
Winamp get stuck by these files and every time I have to manually start playing the next (44.1khz) track.

Your plugin is definitively adopted.

Thx a lot for your work.

[Eyeof Astranger]

M.J.

Following the small gaps heard in gapless recordings with the AVX version of the driver, I tried the SSE2 version. Same exact result with small gaps in gapless recordings.

I tried changing thread priorities and buffer size, but neither changed anything.

BTW, great sound coming out of both driver versions. Well done.

Thanks for your work.

[CodeLurker]

Some mp3's I could not play, which were at 22050 Hz sampling rate, and others, at 44100 Hz and above, I could. This suggests that you need to be able to upsample to make some files play. I noticed you removed some sample rate conversion options from the ASIO plugin's dialog. The versions that could resample did work for me, as well; when I either resampled to a sample rate my driver could use, or had a file I could use. I was not able to get your version, which has had the resampling ripped from it, to work on 22050 Hz files, whatsoever; although, as I say, I could get older versions which could resample to 44.1 kHz to work on those files.

It appears that is what is needed, is to bring back the resampling options; but to have an additional option to say, "Upsample Lower Sample Rates Only". That way, you could set it to your lowest supported sampling rate, say 44.1 kHz, and anything that is slower will get upsampled so it would play; but things at higher sample rates would play at their superior quality rates as well. Better yet, might be a user-editable list of supported sample rates, which anything that wasn't on it would get upsampled if possible, or downsampled if required, so it would play. Maybe you could query the ASIO driver for such a list of supported rates; and resample accordingly, without additional user GUI.

As to the volume control, it would be nice if you could get the main one running. As a next best thing, I noticed that the "Preamp" vertical slider on the graphic equalizer DOES work. The downside of that, is that it only goes up or down +/- 12 dB, so you might not be able to get something as loud or as quiet as you'd like.

I'm using the native ASIO driver for a Presonus FP10, a low end professional audio unit. It can play DVD quality audio, which is to say, 96 kHz, and it does not choke on 192 kHz BlueRay audio. It must downsample that on the fly; although it can really play and record at 96 kHz. Your plugin works great, when it does work, as above. I'm really grateful that somebody is still working on it. Keep up the good work!

[CodeLurker]

I've found a 44.1 kHz stereo file it won't play, but most that are stereo 44.1 kHz it will. To look at their attributes in file info, I can't see any substantial difference. I do see that if you move to another position in the file, it starts playing right away. The "out_asio(exe)_070" version half the time freezes up, and half the time plays after a looooong delay when you seek to a new position. You seem to have fixed this problem.

[roadster.wa]

Thanks for putting your efforts into this. I have just tested using source files from minidlna. The sound quality is excellent and output appears to use exclusive mode as other asio tools are disabled.

Main problem for me seems to be that switching sample rates doesn't work at the first attempt. Incorrect output is sent ( to spdif in my case) then if the player is stopped and restarted the correct output is sent at the second attempt. The sample rate generally then matches the source which is exactly what I require. The exception to this is with flac files which always seem to be upsampled to 192KHz. They do play well but I would rather the upsampling was done in my external DAC/amplifier.

I had some doubt about the correct output channels being used but this seemed to correct itself and is now using the SPDIF outputs from my device ( which is an RME Babyface usb ADAC).

I look forward to the full release version and thanks again.

[M.J.]

Originally Posted by roadster.wa

Thanks for putting your efforts into this. I have just tested using source files from minidlna. The sound quality is excellent and output appears to use exclusive mode as other asio tools are disabled.

Main problem for me seems to be that switching sample rates doesn't work at the first attempt. Incorrect output is sent ( to spdif in my case) then if the player is stopped and restarted the correct output is sent at the second attempt. The sample rate generally then matches the source which is exactly what I require. The exception to this is with flac files which always seem to be upsampled to 192KHz. They do play well but I would rather the upsampling was done in my external DAC/amplifier.

I had some doubt about the correct output channels being used but this seemed to correct itself and is now using the SPDIF outputs from my device ( which is an RME Babyface usb ADAC).

I look forward to the full release version and thanks again.

Hello roadster.wa, thanks for your feedback!

In this version there's absolutely zero sample rate conversion performed (I've stripped that functionality for the moment). Files are played exactly at their sample rate and only if your soundcard supports that sample rate. Neither are FLAC files nor any other media file upsampled to 192kHz as you're describing.

Regards,
-M.J.

[m_lesiu]

Hi M.J.
Is there a plan to bring back gapless mode (hopefully, which will work correctly)?

As for bug reports -I'm using v0.99b and have no problems with it but I blindly assumed, no conversion/scaling is done. I'm using original Asus ASIO drivers with Essence STX (which then outputs analog signal to my Denon AV receiver).

[M.J.]

Originally Posted by m_lesiu

Hi M.J.
Is there a plan to bring back gapless mode (hopefully, which will work correctly)?

As for bug reports -I'm using v0.99b and have no problems with it but I blindly assumed, no conversion/scaling is done. I'm using original Asus ASIO drivers with Essence STX (which then outputs analog signal to my Denon AV receiver).

Gapless mode will be back.

-M.J.

[M.J.]

!!! PLEASE READ !!!

Can anybody who's using the current version (0.99b) please post:

- ASIO buffer sizes / latencies (the settings in your ASIO control panel)

- Buffer Size in OUT_ASIO configuration dialog

- Sample rate in kHz (44.1, 48, 88.2, 96, 176.4, 192, etc...)

- Soundcard / ASIO driver you're using

- Plugin version (SSE, AVX, normal) + your CPU + RAM + Windows OS version

- Is this combination working for you 99.5% of the time? (i.e. no playback stuttering, dropouts, etc... aside the mentioned problems reported above - gapless playback, occasional playback stop on VBR files, etc...)

- (please also try playing with min./max. ASIO buffer size vs. Buffer Size in OUT_ASIO configuration)

I'd be extremely grateful for this information, as this will be used for dynamic buffer size adjustment. There's tendency of inverse relationship between ASIO buffer size in your control panel and OUT_ASIO buffer size - the more ASIO buffer size, the less OUT_ASIO buffer size needed for "correct" playback, but I need to verify, as not all ASIO drivers are the same, and I have even experienced buggy ASIO drivers which don't behave like this.

Thank you very much!

-M.J.

[pbelkner]

Originally Posted by M.J.

Can anybody who's using the current version (0.99b) please post:

The device is "InLine® USB HD Audio Adapter, USB Hi-Fi (24-bit 192kHz) zu Digital Coax / Toslink / I2S Audio Konverter, 33053I", working 100% of the time.

[M.J.]

Originally Posted by pbelkner

The device is "InLine® USB HD Audio Adapter, USB Hi-Fi (24-bit 192kHz) zu Digital Coax / Toslink / I2S Audio Konverter, 33053I", working 100% of the time.

This IS screenshot from original OUT_ASIO :-)

And what about buffer settings from ASIO control panel?

-M.J.

[pbelkner]

Originally Posted by M.J.

This IS screenshot from original OUT_ASIO :-)

And what about buffer settings from ASIO control panel?

-M.J.

I don't know about any other control panel ... What do you mean by that?

[M.J.]

Originally Posted by pbelkner

I don't know about any other control panel ... What do you mean by that?

It's the control panel of your soundcard - somewhere where you select your ASIO settings (mostly buffer size / latency)...

[A11ectis]

ASIO buffer sizes: 512 Samples
Buffer size in OUT_ASIO: 512
Sample rate: 44.1kHz
Soundcard/ASIO driver: Onboard Realtek HD Audio using ASIO4ALL v2.12
Plugin version/CPU/RAM/OS: x86, Intel Core i5 750, 8GB RAM, Windows 8.1
Issues: Currently playing through my FLAC library with no problems; <1sec delay before audio plays

[gkar2012]

no problems here using

RME Fireface 400
latest driver and firmware 3.110 & 1.71

ASIO Buffer in device control panel is dynamic and changes with bit rate(2048 for 24bit etc)
sample is set by rate of music(works well)

Running 99b SSE
win 7 x64, quad q9550 4GB

changes in device buffer to 48 causes drop outs, increasing buffer in ASIO stops them.

many thanks