oss4 not compilable on gentoo

[igorzwx]

Delta 1010LT is crap

It does not support 192kHz sample rate
User Manual →_http://godzilla.kennedykrieger.org/fmri/Delta1010LT.pdf

Delta 1010LT – 24bit 96kHz

ESI Juli@ – 24bit 192kHz

[ossuserr]

Thanks for the data! By the way you have given some link http://manuals.opensound.com/usersguide/oss_sblive.html Judging from that manual audigy2 and all emu10k1 based cards also have hw mixer right? So how can i employ it to get rid of jackd? My stream will be mscore or denemo ---> linuxsampler ---> system:playback (let's suppose i have decided to sacrifice vstis, drumgizmo and some instruments since LS loads not all instruments with jackd1). Does that mixer mixes only pcm streams or also handles midi streams? Anyway I will have to use alsa modules described earlier here for midi input. I suppose the same modules are used for all midi streams right?
Also i forgot one thing denemo supports portaudio and portmidi. It would be interesting to check if that portmidi can be used to get rid of jackd for midi input. What drives me crazy is that mscore does not have internal synth supporting gigs and sfz, i.e. best soundfonts which urges using LS. I think denemo's internal synth is even worse. If they made a good internal synth and vst support like in ardour we could do without jackd... Linux exists for so many years and no such things were implemented. Who cares for sound now!?

delta 1010ls is indeed a crap! I was looking for 32 bit cards and found none ones! Even expensive rme cards are 24 bit. And i think all of them are fixed point. Why did not they make all cards float point if that float point is used in sound editors...? If i were a developer i would construct the OS to have one universal resampling layer with 64-bit floatpoint and all streams would connect to that layer. And a good resampler like Petrov's to serve that layer.

[igorzwx]

By the way you have given some link http://manuals.opensound.com/usersguide/oss_sblive.html Judging from that manual audigy2 and all emu10k1 based cards also have hw mixer right? So how can i employ it to get rid of jackd?

Your hw mixer seems to be a very simple one. You may need a "software mixer" for your card , such as PulseAudio, VMIX, and the like.
You may try to google "user manual" for your card.

[ossuserr]

I don't gugl. i use currently unbubble.eu and sometimes ixquick.com All the rest are rascals filtering torrent links. On alsa i don't even see those mixing capabilities. I will enquire about this. On Oss i saw programs which were playing in ossxmix, but it's software mix. I never heard about software gui for linux allowing to control hw mixer streams.

[ossuserr]

I think that hw mixing could work like that on linux if for example mscore could create separate output pcm devices for each channel as dynamic wav files and if internal mscore's synth could support sfz and gig. I could use a command "musescore -d/dev/oss/oss_sblive0/pcm0 <channel1_file1.wav> &
musescore -d/dev/oss/oss_sblive0/pcm1 <ch2_file2.wav> &
musescore -d /dev/oss/oss_sblive0/pcm2 <ch3_file3.wav> &"

Don't know if -d options is needed. The command was taken from example showing ossplay -d.

[igorzwx]

I think that hw mixing could work like that on linux if for example mscore could create separate output pcm devices for each channel as dynamic wav files and if internal mscore's synth could support sfz and gig. I could use a command "musescore -d/dev/oss/oss_sblive0/pcm0 <channel1_file1.wav> &
musescore -d/dev/oss/oss_sblive0/pcm1 <ch2_file2.wav> &
musescore -d /dev/oss/oss_sblive0/pcm2 <ch3_file3.wav> &"

Don't know if -d options is needed. The command was taken from example showing ossplay -d.

This seems to be MISUNDERSTANDING.

[ossuserr]

Well, i did not claim that i understand that cool mystical hw mixing. So far i don't even see any need in it since all work is tied to gackd.
I have talked to mscore developers and they told they are ready to implement support for hw mixing but they don't understand what it is. So if you can normally explain to me how we could implement it for mscore to avoid jackd i can tell to them.

[igorzwx]

Well, i did not claim that i understand that cool mystical hw mixing

What "mystical" is your way of thinking (that is why you can consume any "conspiracy theory").

Step 1: set "spkmode" to "FRONT+SURR"

Step 2: Run three audio players simultaneously:

1. the first is playing file1.wav to /dev/dsp0
2. the second is playing file2.wav /dev/dsp1
3. the third is playing file3.wav to /dev/dsp2

You should hear all three wav files playing simultaneously.

[ossuserr]

Perhaps, you are a joker. Some time ago you proposed using hardware mixing instead of iackd. Now i ask HOW? Instead you tell something about running three players simultaneously. Ok suppose, your cool hw mixing works and no vmix is used and i can run 3 players simultaneously. How it replaces jackd? And Namely for mscore case how? If that hw mixer is needed to run several programs simultanously it does not mean that hw mixer can connect programs to each other or even join several streams-channels within one program though this joining can be done without hw mixer. If mscore can select directly alsa hw:x.y device then its internal sampler will convert all channels into 32bit float point, as i understand, do math operations to join into one streama and then will send to the hardware output which will do all sampling rate convertion to samplerate set somewhere, and i should be able to record via a driver function Record what you hear. I don't know if alsa has such function but it might be available. In order to connect streams from one program into another i don't see any possible application of hw mixing according to how you described it. Parallel playback without any software interfering is not equal to joining programs. So what you achieved you made me think about useless things. It seems that alsa + jackd suits my needs. Maybe i will manage to pursuade linux programmers to introduce better resamplers and 64-bit float for that jackov-d.

By the way presence of hw mixer in cards for linux IS a requirement according to their_pedia.com.... Cards without it even won't work...
Didn't i confuse something? You told lynx2 has something inside to replace jackd. Then you told esi juli is cool because it has hw mixer... Well ... all cards for linux MUST have hw mixer. I think even 5 usd cmedia cards have hw mixer since they work in linux. Thanks for chat anyway. That juli card seems good by the way according to specs.

[igorzwx]

Ok suppose, your cool hw mixing works and no vmix is used and i can run 3 players simultaneously.

Have you tried it, or not?

I do not have your card, you should do it yourself.
VMIX should be disabled completely.

You may try it with ALSA

/dev/dsp0 → hw 0
/dev/dsp1 → hw 1
/dev/dsp2 → hw 2

You should have a kind of "spkmode" in your ALSA mixer (it may have another name).

If "mixing" works, you may try to record the mix.

Perhaps, you are a joker.

If you want to know how to record the mix, you have to study your "ossmix" yourself.

[ossuserr]

/dev/dsps cannot correspond to hw devices because the card has only 4 devices but number of dsps is 7 by default and can be increased up to 32. I suspect those 32 dsps may correspond to subdevices of hw:0,3 because that device is called multiplayback. Perhaps that word 'multiplayback' implies that very hw mixing feature.

[igorzwx]

/dev/dsps cannot correspond to hw devices because the card has only 4 devices but number of dsps is 7 by default and can be increased up to 32. I suspect those 32 dsps may correspond to subdevices of hw:0,3 because that device is called multiplayback.

You need the true "hardware devices".
Some of ALSA "devices", or "subdevices" might be "virtual devices" of "ALSA software mixer", a king of "plugins" or other sort of "virtual crap".

Perhaps, you may want to learn ALSA, if you are going to use it.
There is a kind of "secret esoteric manual" _http://www.volkerschatz.com/noise/alsa.html

Perhaps that word 'multiplayback' implies that very hw mixing feature.

Yes. It might be "hw mixing".

Some days ago, you wanted to have a kind of "routing":

My next question is 'Is it possible to route everything from front channel (fronts output of soundcard) to rear channel (rear out) because i want to plug headphones into rear-out and hear the same what comes from speakers connected to front-out.
_http://www.opensound.com/forum/viewtopic.php?f=3&t=5807&start=30#p21266

If you want to get PCM playback from both front and rear speakers,
you may try something like this:

Code: Select all

$ ossmix spkmode FRONT+SURR

You can also use OSS Mixer GUI:

Code: Select all

$ ossxmix

_http://www.opensound.com/forum/viewtopic.php?f=3&t=5807&start=30#p21269

Didn't i confuse something? ... all cards for linux MUST have hw mixer.

I have a very old computer (of 2003) with AC'97 codec on matherbord. It has "spkmode", and it can can make the same "routing" as that of your soundcard.

AC'97 (Audio Codec '97; also MC'97 for Modem Codec '97) is an audio codec standard developed by Intel Architecture Labs in 1997.
_https://en.wikipedia.org/wiki/AC%2797

It seems that the hardware mixer of you soundcard is the hw mixer of the ancient AC'97 codec.
Such a "professional soundcard" might be an ideal choice for mass production of "digital crap" for semi-deaf ALSA users.

That is why, perhaps, the OSS4 developers are not going to fix the OSS driver for this card.

[ossuserr]

Your substantiation sounds sound. That audigy2 and all those cards since sblive 5.1 were a marketed crap and impudent deceipt. The marketeers assured innocent people that those cards would function as a kind of home studios and knowing that at that time pro sound cards costed much money they assured like sblive and the followers are kind of charity, a step towards our needs. They created abundance of features, wrote fat manuals to make that shit look more attractive. But mud is mud. And instead of making high-quality p16v as the main dsp in the card, they left emu on audigy2 cards. Whatever i tried to make that p16v work with jackd, it's not possible. I guess the reason is that p16v addresses both jackd and alsamixer simultanously which locks it till reboot... They claim jackd is kind of indepenent from alsa. In fact it's half-truth because you can still change sound volume in alsamixer when card is running jackd. So alsa is still employed and jackd is not indepenent in this aspect though its internal sound volume levels remain independent - you may have loud recording with jack_capture but low volume from speakers due to setting in alsa mixer. Still this renders p16v unusable. And i will have to change the card because i can't stand the dirty sound of emu10k1 it makes me vomit.
It sounds like music is played in a tomb among deadmen.

[igorzwx]

The second codec of your souncard provides several PCM hw devices of 96kHz (if I am not mistaken), but it does not have own hw mixer.
Hence, the hw mixer of AC'97 codec is to be used, but it might be incapable of mixing "digital crap" at 96kHz.
Therefore, your ALSA driver may not support such hw mixing.
Your OSS4 driver may support hw mixing, but 96kHz may not be supported.
To make donwsampling before hw mixing, ALSA does need a "sofware resampler".

Sergey Petrov claims that the "secret resampler" of ALSA cannot be disabled.
He seems to be right. ALSA is so designed that it cannot work without "software resamplers".

That is why Petrov created a special plugin to disable the "secret resampler" of ALSA.
The only way to disable this crappy resampler, is to replace it with another one.
This is exactly what Petrov did. He replaced the "secret resampler" with his own resampler.
However, his plugin needs the ALSA "software mixer" to work.
This crappy "software mixer" produces sound distortions.

The ALSA "software mixer" cannot be disabled, because ALSA is a crappy "software mixer" (+ crappy drivers),
as the "secret esoteric manual" states: _http://www.volkerschatz.com/noise/alsa.html

You may try "hw mixing" with Windows and ASIO driver, if available.
If it works, you may try it with OSS4 and winasio.

[ossuserr]

That is interesting info.
On gen2 you may disable libsamplerate for the entire system easily. In /etc/portage/make.conf you should put USE="libsoxr -libsamplerate" It will disable libsamplerate completely and enable libsoxr. As OS's info says you can enable libsoxr only for severl things so far:
libsoxr
media-sound/audacity:libsoxr - Uses media-libs/soxr as audio resampling library: Better quality than the included resampler and much faster than libsamplerate while keeping almost the same quality.
media-sound/mpd:libsoxr - Enable the libsoxr resampler
media-video/ffmpeg:libsoxr - Enables audio resampling through media-libs/soxr.

Does Petrov mean libsamplerate under alsa secret resamapler or some other resampler? If the first, then it can be disabled... And devices/subdevices can be addressed directly as hw:xy devices easily from programs supporting it without any resampling. If Petrov means some other resampling on the level of the driver and not as external lib, then it's bad. Of course i would prefer oss driver but i need dssi-vst and linux sampler fully working. Why there is no a single programer to fix those things? Shit, shit, shit everywhere. They work for money and like Gayrope and don't care for musicians. As for audigys and sblives they are complete crap i agree and creative team are marketeer bustards.