Old 24th October 2014, 13:14   #1
M.J.
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New out_asio is coming, stay tuned!

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.
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Old 24th October 2014, 15:17   #2
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cool!!
hope it is a good replacement for the maiko wasapi plugin

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Old 24th October 2014, 19:51   #3
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Quote:
Originally Posted by M.J. View Post
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!
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Old 25th October 2014, 03:58   #4
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Very good news indeed. I am looking forward to trying the new version.
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Old 3rd November 2014, 13:45   #5
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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.

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Old 20th November 2014, 08:48   #6
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...updates...?

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Old 20th November 2014, 23:30   #7
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I just signed up to register my interest. I'm glad development for Winamp is still alive!
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Old 21st November 2014, 11:02   #8
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Originally Posted by bawheid View Post
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'
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Old 25th November 2014, 22:18   #9
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any update from the author ?
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Old 25th November 2014, 23:05   #10
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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.
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Old 26th November 2014, 09:00   #11
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@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

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Old 26th November 2014, 11:43   #12
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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.
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Old 30th November 2014, 19:23   #13
M.J.
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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.
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Old 30th November 2014, 21:20   #14
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Installed the plugin and ASIO4ALL. No real idea what either of them really do, but they seem to work as described.
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Old 6th December 2014, 21:04   #15
M.J.
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Roughly 30 downloads now. Please let me know if you have tried it and whether it works for you.

Thanks!

-M.J.
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Old 7th December 2014, 10:18   #16
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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.
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Old 7th December 2014, 10:34   #17
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@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.
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Old 9th December 2014, 12:35   #18
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Originally Posted by Raj_09 View Post
@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.
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Old 10th December 2014, 08:10   #19
M.J.
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Quote:
Originally Posted by Raj_09 View Post
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.
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Old 10th December 2014, 08:11   #20
M.J.
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Quote:
Originally Posted by Raj_09 View Post
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?
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Old 10th December 2014, 14:03   #21
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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.
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Old 10th December 2014, 14:15   #22
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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.
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Old 11th December 2014, 13:37   #23
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I've downloaded and will test over the next few days.

I intend throwing some mp3's and FLAC's at it (a mixture of standard 16-bit/44.1 kHz, 16-bit/48 kHz, 24-bit/96 kHz, 24-bit/192kHz, and some 5.1 SACD rips at ~19,000kbps/~170 kHz.) I'll let you know the results!
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Old 14th December 2014, 02:30   #24
Eyeof Astranger
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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
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Old 16th December 2014, 20:26   #25
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Quote:
Originally Posted by Raj_09 View Post
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.
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Old 17th December 2014, 13:19   #26
M.J.
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Hello all,

Quote:
Originally Posted by Raj_09 View Post
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.
Quote:
Originally Posted by Eyeof Astranger View Post
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.

Quote:
Originally Posted by Branddy View Post
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.
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Old 17th December 2014, 23:28   #27
Branddy
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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.
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Old 27th December 2014, 19:01   #28
Eyeof Astranger
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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.
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Old 2nd January 2015, 11:48   #29
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As above, mixed results

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!
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Old 3rd January 2015, 01:20   #30
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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.
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Old 10th January 2015, 18:15   #31
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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.
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Old 11th January 2015, 09:07   #32
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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.
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Old 11th January 2015, 09:47   #33
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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.
Yes please disregard the comment about flac, I think I was mistaken and tested using the wrong file. At the moment I can't use this output because of the nasty noise when there is a change of sample rate between tracks. So far I have only overcome this by stopping and restarting play but I will go through other settings to see if there is a work round.
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Old 14th January 2015, 16:46   #34
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Yes please disregard the comment about flac, I think I was mistaken and tested using the wrong file. At the moment I can't use this output because of the nasty noise when there is a change of sample rate between tracks. So far I have only overcome this by stopping and restarting play but I will go through other settings to see if there is a work round.
Strange, as I do have the same audio interface as you do (RME Babyface), and I don't experience this behavior when, for example, listening to 44.1kHz tracks followed by 48kHz tracks.

I tend to use lowest buffer size (48 samples) within RME's driver, and OUT_ASIO is set to 63 buffers size / Time Critical thread-priority.

I suggest playing with these combinations.

Regards,
-M.J.
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Old 15th January 2015, 18:40   #35
roadster.wa
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Strange, as I do have the same audio interface as you do (RME Babyface), and I don't experience this behavior when, for example, listening to 44.1kHz tracks followed by 48kHz tracks.

I tend to use lowest buffer size (48 samples) within RME's driver, and OUT_ASIO is set to 63 buffers size / Time Critical thread-priority.

I suggest playing with these combinations.

Regards,
-M.J.
Just tried those settings and created a 48KHz file to test too.

There is no problem switching between 44.1 and 48KHz but if you go to 96 or higher the difficulty is as I described. This may be because the Babyface driver reconfigures the total number of input and output sockets. It doesn't need to do this to achieve 48KHz. Incidentally I always have the initial BF setting at 192KHz. I have also noticed that a small buffer size tends to cause pops or crackles if there is any other processor activity so I tend to run it at the default.

You may already know that the Foobar2K wasapi driver has similar problem ( actually worse because it doesn't recover at the second attempt) but the F2K ASIO works OK.
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Old 15th January 2015, 22:05   #36
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Just tried those settings and created a 48KHz file to test too.

There is no problem switching between 44.1 and 48KHz but if you go to 96 or higher the difficulty is as I described. This may be because the Babyface driver reconfigures the total number of input and output sockets. It doesn't need to do this to achieve 48KHz. Incidentally I always have the initial BF setting at 192KHz. I have also noticed that a small buffer size tends to cause pops or crackles if there is any other processor activity so I tend to run it at the default.

You may already know that the Foobar2K wasapi driver has similar problem ( actually worse because it doesn't recover at the second attempt) but the F2K ASIO works OK.
Thanks for the feedback, I will try with higher resolution files.

Feedback like this is very helpful, because I've started to think about implementing dynamic buffer settings with regards to ASIO driver's buffer size - then there would be no such option for setting buffer size in OUT_ASIO, because it'd be set in realtime. I'm not sure if this gets to the next version that is planned for release at the end of January, but who knows :-)

Regards,
-M.J.
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Old 18th January 2015, 15:02   #37
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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).
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Old 21st January 2015, 14:17   #38
M.J.
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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.
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Old 21st January 2015, 14:37   #39
M.J.
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!!! 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.
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Old 21st January 2015, 18:04   #40
A11ectis
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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
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