Whats wrong with my ASIO
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What version of asio2 plugin do you use ? Did you try with the last one ?
the reference to wasap2 in the log is a mispelling of the log message.
foo_out_asio2_1.1-beta2-patch-1. Peter's foo_out_asio works fine. I ran a new one with same bursts of 'hiss'.
Endpoint buffer size (sample) = 64
Sample cache factor = 128
I have a question. Why is the
a)Renderer::init_asio_static_data - ASIOGetBufferSize (min:176, max:176, preferred:176, granularity:0)
always the same as
b)ASIOGetLatencies (input: 176, output: 176.
In my previous post it was 144 and before that 488. Do we have any control at all over it.?
Thanks
may be something wrong because of the type of Asio Buffer returned by your driver: ASIOSTint24LSB (24 bits). Normally it is supported by asio2 plugin but I have not used it for a long time since this not my current config (my driver returns 32 bits buffers). May be you could try to change some parameter in your driver ?
I don't know why your latencies are identical to your asio buffer size. these infos are returned by your asio driver. it looks strange. Normally the allowed buffer sizes returned by your driver should be the same provided that your tracks have the same sample rate. In the other hand, allowed buffer size may be higher when playing HD tracks, depending on your driver.
You are right about the buffersize. All 44.1khz has same value and all 96khz same value. This ofcourse depended on the latency set in the asio drivers control panel. I am going to download another driver to try.
One more thing.Was the file that you sent to Anton B (ticket 6) very different from the latest. I get very faint music from that one with white noise. The latest I get only white noise
I don't have the source code of the patch attached in ticket 6 (I don't use any VCS and keep only source for the published releases), so it is difficult to say. As far as I remember, the latest release also added some change on the sample cache management. May be there is a side effect here. The problem is that as I said in ticket 6, I cannot test by myself asio drivers supporting only ASIOSTInt24LSB buffer type.
Thanks for your patience. I left foobar on pause for a while and it crashed. I have here the clash report.
crash seems to occur in rthdasio, not in foo_out_asio2:
Crash location:
Module: rthdasio
Offset: 18F6Dh
Symbol: "DllUnregisterServer" (+103CDh)
Concerning your initial problem (white noise instead of music), is is possible to increase the asio buffer size in your Realtek Asio control panel ? I have tried with 172 samples for a 44.1Kz/16bits track with ASIO4ALL on top a Realtek driver (not asio) and music is there but with some noise .
You've completely lost me. You installed a Realtek driver without asio. You installed ASIO4ALL. How does the Realtek Asio control panel affect this?. Could you explain it a little more what you did? I can get clear music with the Asio2:ASIO4ALL. Is the Asio2 file you are using the one in the Attachment?.
Sorry if I was not clear. I simply mean that even with ASIO4ALL, when using a very low buffer size (you can configure it in foobar advanced options for ASIO2, as ASIO4ALL permits to use different buffer sizes), sound may be crappy. Of course, with a decent buffer size (e.g 400 or more), ASOI4ALL sound is ok.
My suggestion concerned the realtek Asio output: trying to increase, if possible, the realtek asio buffer size in the realtek asio control panel I have seen in one of your screenshot above (I re-attached the screenshot it here)
And yes, your version of the asio2 plugin is the right one
Sorry for the delay, I was testing all the Asio2 drivers in the tickets. The only one that came close to working was the one in ticket6 (Anton B), as I posted earlier.
Do you necessarily have to use the same type the ASIO buffer that was RETURNED, or you can SET it on your own. This is my thinking_____
When query to interrogate what type the ASIO buffer is returns ASIOSTInt24LSB, then set it to
ASIOSTInt32LSB. I will be willing to test it. Attached is the console log from the file in ticket6 (Anton B). The track is 16/44.1. I dont think Foobar allows ASIOSTInt24LSB. Thanks
There is no way to force the channel type (e.g. ASIOSTInt32LSB) by means of the ASIO api. I can change the code to write 32 bits samples (instead of packed 24bits ones) in the output buffer. is it what you mean ? I doubt that it will work, but I can send you a patch like this (I will need to find a moment in the next days)
Thanks, that will be great. I will be waiting for it
Here is the patch.
I have also fixed a bug related to 24 bits buffer type processing. So give a try with default configuration before trying to force buffer type to 32 bits.
In order to do this, go to the plugin advanced options and enable the "Force 32 bits asio buf type" flag.!
Thanks . I will let you know as soon as
These are the results of the test so far.
Force 32bit buf type is not working
The default 24bit is working with one main exception. The volume level is way too high. If you look in the attachment, the volume set in Foobar(about 1/10th) is the same volume from, for example, foo_asio_out at set full level.
These are the codecs I've tried so far at 3ms and 11ms
DSD64, DSD128, DSD256
DST(6 channel >crashed), DST64(2channel)
MP3(cbr,vbr)
PCM_BLURAY(m2ts)
PCM(wav)
FLAC(16/44.1, 24/96, 24/192)
MONKEY AUDIO( 24/96 @3ms to 11ms> crashed , 20ms>OK)
(16/44,1@3ms to 11ms> crashed, skips lightly)
WAVEPACK( Most problematic. Goes faster with increased buber size and slower with reduced buffer size
All in all very good, except the volume and the 2 codecs. Here is to hopping and wishing it gets fixed.
Thanks
Last edit: mo phillips 2021-03-22
new patch attached.
Crashes should be fixed now.
I cannot test the fix for the loud volume, but it should be ok.
I cannot reproduce your problem with wavpack tracks. May you test it again, and send me the log ? Thanks.
Volume is fixed. Crashes are fixed.
DSD64, DSD128, DSD256, DST(6 channel), DST64(2channel), MP3(cbr,vbr), PCM_BLURAY(m2ts),
PCM(wav), FLAC(16/44.1, 24/96, 24/192) are fixed
Monkey audio(.ape) and Wavpack still skipping or playing fast or both with foo_out_asio2. They dont with foo_out_asio.
Wavpack skipping in ASIO2:ASIO4ALL>Realtek
Monkey audio does not skip or play fast in ASIO2:ASIO4ALL>Realtek.
Finally I have 1(single) channel mono tracks that play through the centre channel ( not the FL orFR). With ASIO2 the play life 2x the speed. With ASIO and ASIO4ALL, they play normal speed.
Ihave attached some logs
Last edit: mo phillips 2021-03-22
I am working on it.
The crashes will be fixed and I can reproduce the bug with the 1 channel tracks. but I am a bit confused with tracks which play fast or skip. you are saying:
"Monkey audio(.ape) and Wavpack still skipping or playing fast or both with foo_out_asio2. They dont with foo_out_asio.
Wavpack skipping in ASIO2:ASIO4ALL>Realtek
Monkey audio does not skip or play fast in ASIO2:ASIO4ALL>Realtek."
Do you mean that with foo_out_asio2, Wavpack tracks skip every time, either with the realtek driver or aiso4all, whereas monkey audio ones skip only with the realtek driver, not with asio4all ?
May you tell me what is your asio buffer size when you encounter the problem ? By now, I have tried both codecs with buf size = 488 and I cannot reproduce it. May you try also with a larger sample cache factor in the advanced options (e.g. 4096. see screenshot) ?
Correction > In ASIO2:ASIO4ALL Monkey audio and wavpack skip, they dont play faster
24/96 and 24/192
11ms>16384>Long skip (plays about 2s > skips 1-3min > plays 2s etc)
14ms>16384>OK
20ms>16384>OK
11ms>4096>Long skip
14ms>4096>OK
20ms>4096>OK
11ms>512>Long skip
14ms>512>OK
20ms>512>OK
16/44 11ms to 20ms @ 4096 to 16384 > Long skip
WavPack with ASIO2:Realtek
11ms to 20ms @ 4096 to 16384 > Long skip
No crashes
Thanks, These bugs should be fixed now in the attached patch, I think.
WavPack Ok
Mono 1Channel Ok
Monkey Audio 24/96 OK
Monkey Audio 16/44 skips ( Skipping has no particular pattern ) from 3ms to
20 ms @ cache ratio 0, 64-1024
No crashes
I cannot reproduce it. Anyway, i have changed something that could be the source of the problem. let me know.
By the way, I have also fixed a delay problem with pause ou volume change when using a big sample factor. may you check if changing foobar volume works, as I cannot do it myself for 24bits buffers ?
In then latest Asio2 the volume behaves as follows:
When the volume is decreased in foobar from full, we get very high distorted sound first and then normal sound. If the volume is increased to full again, we get high distorted sound and it stays distorted.
In the previous Asio2, a change in volume is preceded by a delay but no distortion.
In the new Asio2, ALACand FLAC codecs are now clicking and stuttering but not skipping. There is general clicking in the other codecs too.
So I am backto the previous Asio2 and this is additional info:
I had a look at the .ape files on the disk rather than in foobar and found that almost all of them are cuesheets(.cue) with large .ape files
Now back in foobar I noticed this about the cuesheets
1. All 1st track on the album plays without skipping and if the 1st track proceded to the 2nd without my interference, the second track also plays without skipping. 2nd to 3rd on its own also plays without skipping. If during playing the 1st track, I skipped to about few seconds to the end of the track and let the 2nd track start on its own, the 2nd track plays without skipping. For all to work I have to start from the 1st track.
2. If I start the album from the 2nd track or 3rd or 4th, it skips.
3. If I play only the large .ape file associated with the cuesheet, it plays without skipping. starting at any position I choose.
4. The plain .ape files with no cuesheet(probably thats the ones you were testing) didnt skip however I play them.
Hope this helps.