on this chip is philips logo ??? I think yes.
A1334
this is Philips UDA1334 2 channel DAC user for rear channel. On 2003
this is missing,
then rear channels must go out throught other chip (I think through
AC97).
Last try,
Change this in alsa-driver/alsa-kernel/pci/emu10k1/emufx.c
if (emu->f
Did you try if oss free drivers works ???
Please try first. What dirstribution you use ??? Maybe it is
compiled in. Just try
/etc/init.d/alsasound stop
modprobe emu10k1
and play 5.1 sound through oss.
I think you can use osstest application from alsa-oss package
Peter Zubaj
I don't mean this, this is comercial binary only driver (You can
try).
This
http://prdownloads.sourceforge.net/emu10k1/emu10k1-v0.20a.tar.bz2?
download
or recompile kernel with OSS/free drivers enabled.
Peter Zubaj
http://www.pobox.sk/ - spolahliva a bezpecn
I don't mean alsa-oss-emulation, I mean oss driver from
opensource.creative.com or linux kernel. If it works, than something
is wrong with alsa (I think it will not work).
In SB Live up to 2002 was used AC97 codec for front and center/lfe
and another dac for rear channels. I think, that in SB L
Because Creative uses diffrent AC97 codec and this codec has same
controls as controls implemented in dsp. then on 9.8 it fails to
load.
Could you send me output of /proc/asound/card/codec97/ac97 <- this
is not exact name
/proc/asound/card/codec97/ac97reg
where card is card name
???
Peter Zub
In alsa-driver/alsa-kernel/emu10k1/emumixer there are two static
arrays:
1) static char *emu10k1_remove_ctls[]
remove from this array:
"Surround Playback Switch",
"Surround Playback Volume",
2)
static char *emu10k1_rename_ctls[]
remove from this array:
"Surround Digital Playback Volume", "Surrou
Hi,
What AC97 codec uses SB Live 2003 ???
Maybe for rear channels is used AC97 too and by default is rear
volume set to 0 and control is removed to avoid duplicite.
Peter Zubaj
== REKLAMA
Spolocnost SUN Microsystems uviedla na trh novy server Sun F
Hi,
Try to look at alsa-driver/alsa-kernel/Documentation/ControlNames.txt
But warning, this is not complete list, rather recomendation. Most
control names folows this recomendation, but there are some controls
for some soundcards, that have diffrent names.
Peter Zubaj
==
Hi,
My fault.
This is what is in datasheets:
>There are two option circuits for MIC to disable bias voltage. For
>ALC650 ver.E or later, there is Vrefout-disabled function, so
>bias voltage from Vrefout(pin28) is recommended as option2 circuit.
>When MIC is shared with Center/LFE, software sho
Hi,
My fault.
This is what is in datasheets:
>There are two option circuits for MIC to disable bias voltage. For
>ALC650 ver.E or later, there is Vrefout-disabled function, so
>bias voltage from Vrefout(pin28) is recommended as option2 circuit.
>When MIC is shared with Center/LFE, software sho
Hi,
My fault.
This is what is in datasheets:
>There are two option circuits for MIC to disable bias voltage. For
>ALC650 ver.E or later, there is Vrefout-disabled function, so
>bias voltage from Vrefout(pin28) is recommended as option2 circuit.
>When MIC is shared with Center/LFE, software sho
Hi,
>1) I've got 3 separate PCM 'mono' streams of data that I want to
>send to
>3 different outputs.
Should work.
> a) How many PCM inputs do these cards/drivers support? Under OSS,
>there were 2, /dev/dsp and /dev/dsp1. Ideally I would like each
>input
>stream to have a seperate sample rate.
>What exactly happens when I record at sample rate 44100 Hz?
>Does emu10k1 chip run at sample rate 44100 Hz as well?
>(It is claimed that emu10k1 runs only at rate 48000 Hz.)
>If emu10k1 runs at 48000 Hz, at what rate A/D converter runs?
>If emu10k1 runs at 48000 Hz and Alsa at rate 44100 Hz, wher
Hi,
When I remove control through hwdep layer on emu10k1
driver when in use by oss emulation I got oops.
This is because oss emulation holds pointer to this control.
Is there any api to disable an then enable oss emulation or
only proc interface ???
Jan 20 20:53:52 localhost kernel: Unable to
ld10k1 development was not stoped. Now I am rewriting some parts. I
hope, that I will finish this rewrite this week.
Peter Zubaj
http://www.logofun.pobox.sk - urobte radost svojmu telefonu
---
The SF.Net
Hi,
Try Threaded OSS (toss) instead of OSS driver in arts.
Peter Zubaj
http://www.logofun.pobox.sk - urobte radost svojmu telefonu
---
This SF.net email is sponsored by: Perforce Software.
Perforce is t
Hi,
This patch defaults initial center/lfe jack to analog mode (now it is
initialized to digital mode). This solves problem with noise from
center on SB Live 5.1.
Peter Zubaj
http://www.logofun.pobox.sk - urobte radost svojmu telefonu
C:\\live_ad.diff
D
Hi,
Probably user has normal soundcard handled by ALSA and TV card
(btaudio or saa7134) which provides only OSS drivers (warning - only
recording - not playback). TV card is default /dev/dsp (can by
changed by module option for btaudio, saa7134). For BT87x users there
is alsa driver in ALSA 1.
Hi,
In this mailing list was message that Line2 on LiveDrive is not
working. I think there is typo in driver (copy & paste typo). There
are created controls with same control names for line1 and line2 of
LiveDrive and then controls for Line2 are not created (maybe are
overwriten). Attached pat
Hi,
I think, there is bug, (there are controls with same names for line1
and line2)
Try find text "Line LiveDrive Playback Volume" in alsa-driver/alsa-
kernel/pci/emu10k1/emufx.c
There will be 2 ocurence. Change second to "Line2 LiveDrive Playback
Volume"
Do same thing for "Line LiveDrive Captur
Yes, Philips ADC doesn't help playback but recording yes. But ADC is
connected to emu10k2 by 24 bit, and I think that philips DAC (1328)
is connected by 24 bit too, this means that card internally work in
24 bit, only problem is how too get 24 bit data from to emu10k2 - one
solution - using TRA
Hi,
I have Audigy 1 (model SB0090)
Audigy 1 has UDA1361TS ADC too (look at card picture - just above
1328)
UDA1361 is connected on AC97 line out.
Why use UDA1361 ??? Because it is 24 bit (you get beter resolution)
and it has beter S/N as AC97.
If you have Audigy 2 you can use UDA1361 ADC witho
Hi,
TV card uses btaudio ??? If yes then try to add module options to
btaudio
dsp1=2 dsp2=3 mixer=2
or something similiar (then first oss device will be alsa soundcard
and next btaudio)
Peter Zubaj
== REKLAMA =
Vyrazne zlavy pocitacov a pr
Hi,
Last evening I found way how to use Philips ADC instead of AC97 on
Audigy 1.
Attached patch is tested only on Audigy 1 player (not tested on
Audigy 1 Platinium, Audigy 1 Platinium EX, Audigy 2, Audigy 2
Platinium, Audigy 2 Platinium EX), but I think it should work for all
Audigy 1 and Aud
>Good luck.
>When I told them my nForce board supported hardware mixing, based
upon
>nVidia's documentation, I was told, flat out, I was wrong...
>I hate to sound like a conspiracy theorist, but I think it was easy
for
>them to patch up the i810 module to support the nForce, and that
>driver's
Hi,
Realtek ALC650 is supported - but this is only analog part of sound
system, you need second part working - and this is soundstorm. But
Soundstorm is not supported very well. (Maybe i810 module is working
with it - I don't know, if yes, then only basic functions are
supported - no dolby dig
Hi,
These patches make some changes to mixer on Audigy 1 and Audigy 2.
Changes:
- new FX bus allocated for Wave Stereo playback ("Wave Playback
Volume")
- new "Master Playback Volume" control - controls output volume for
all channels (implemented using DSP code)
- new controls "PCM Front Playba
And use Philips DAC on Audigy 1 instead of AC97 for front channel (
Same volume levels as others channels, beter precision, ...).
AC97 will be then used only for analog inputs.
Peter Zubaj
http://www.pobox.sk/ - spolahliva a bezpecna prevadzka
-
Hi,
I have ALC650 rev. D and spdif is not working with 0.9.7.
Please delete this row from ac97_patch.c from function patch_alc650
ac97->ext_id &= ~AC97_EI_SPDIF; /* disable extended-id */
I think, that all ALC650 have spdif out and rev. E and later have
spdif in.
Peter Zubaj
__
>There exist some ac97 codecs capable of hw multiplexing?
>What are them?
I think that none. AC97 is only part of sound system. These part is
reponsible for A/D and D/A conversion + analog mixer + spdif in +
spdif out. Most capability is in second part. This second part is
responsible for playi
Hi,
This is taken from oss driver. It is not tested (I don't own
receiver). Can someone try it, maybe this works, I don't know.
Audigy.conf - copy this to /usr/share/alsa/cards/ and overwrite old
one (make backup).
emu10k1.h.patch - patch for file alsa-driver/alsa-
kernel/include/emu10k1.h
em
Hi,
Alsa mixer is not correctly working with controls with big range (for
example 0 - 20) (there is integer overflow).
Peter Zubaj
== REKLAMA =
Vyrazne zlavy pocitacov a prislusenstva
Digitalny fotoaparat Olympus µ300 v cenach uz od
Hi
Sorry, I was on holiday.
I have not enought free time - ld10k1 progress is only small (mostly
I was discovering how is TRAM working on Audigy).
On protocol:
There is such protokol, but mostly it is only one way (loader ->
linker) - linker -> loader part is missing (for now).
On lo10k1, ld10
Hi,
This patch is only my proposition, use at your own risk. If someone
is interested.
Extends HWDEP of emu10k1 driver:
Enables TRAM on audigy (for now only external tram tested)
Enables 0x200 GPRs - (two times more as SB Live) - tested, should work
Enables 0x400 instructions (two times more as
Following information was obtained by trial-and-error.
Use it at your own risk!
TRAM setup:
TCBS (0x44) and TCB (0x41) has same meaning as on SB Live
Internal TRAM size is 0x4000 (16 bits words)
Max external TRAM size is 0x10 (16 bits words) - as on SB Live
Register description:
0xdb - Int
I know about this, but for me is simpler to add 20 DSP instruction
than change 100 lines of driver code.
ALSA emu10k1 driver uses only TRAM for AC3 passthrough, other method
is not implemented.
And I don't need (I don't have receiver) AC3 passthrough (this will
by bonus for others), what I nee
Hi,
I wanted to add support for TRAM on Audigy to emu10k1 driver.
I look (tryed) at OSS driver and found that TRAM is not working
too. :-(
Then I use trial and error method and found how to setup TRAM on
Audigy. I know how to read from and write to TRAM in EMU10k2 DSP
program but not more.
Is
Thanks,
I don't need TRAM_POKE (TRAM_PEEK) now (in as10k1 is no way to tell
init values for tram), maybe later.
Peter Zubaj
== REKLAMA =
Vyrazne zlavy pocitacov a prislusenstva
Digitalny fotoaparat Olympus ľ300 v cenach uz od 16.100,- Sk bez DP
And ones more:
IOCTL SNDRV_EMU10K1_IOCTL_PCM_PEEK in snd_emu10k1_fx8010_ioctl
allocate structure to ipcm which is then passed to
snd_emu10k1_ipcm_peek which then uses one of member (ipcm->substream)
which is not initialized (may come from user space ???).
Peter Zubaj
==
You can downoad it from http://pzwz.wz.cz
No TRAM support yet (I am working on it).
Peter Zubaj
== REKLAMA =
Vyrazne zlavy pocitacov a prislusenstva
Sun Fire V210 server v cenach uz od 125.000,- Sk bez DPH
Navstivte nas na adrese http://www.
Hi,
Ioctl SNDRV_EMU10K1_IOCTL_PCM_PEEK in snd_emu10k1_fx8010_ioctl
allocate structure to ipcm which is then passed to
snd_emu10k1_ipcm_peek which then uses one of member (ipcm->substream)
which is not initialized (may come from user space ???).
Peter Zubaj
== REKLAMA ===
Hi,
I am working on SB Live dsp patch loader an found, that driver will
segfault when I try to delete control in function
snd_emu10k1_del_controls in emufx.c on line
list_del(&ctl->list);
this is because control is removed from list in function before
snd_emu10k1_ctl_private_free in emufx.c
Pet
Try this:
Find in file
alsa-driver/alsa-kernel/pci/emu10k1/emufx.c
this
/* analog speakers */
//A_PUT_STEREO_OUTPUT(A_EXTOUT_AFRONT_L, A_EXTOUT_AFRONT_R,
playback + SND_EMU10K1_PLAYBACK_CHANNELS);
A_PUT_STEREO_OUTPUT(A_EXTOUT_AC97_L, A_EXTOUT_AC97_R,
playback + SND_EMU10K1_PLAYBA
CVS 26.02.2003 works OK.
Thanks
Peter Zubaj
http://www.pobox.sk/ - urcujeme trendy
---
This SF.net email is sponsored by: Scholarships for Techies!
Can't afford IT training? All 2003 ictp students recei
This weekend i tested via8233a + alc650.
Surround output jack is swaped with center/LFE jack (but not controls
in mixer - surround control controls volume of center/LFE, ..)
Please change condition in via82xx.c
unsigned short val;
if (runtime->channels > 4)
/* slot mapping: 3,4,7,8 */
val
I tink, there is still small bug (I think) - via82xx.c
else
/* slot mapping: 3,4,6,9,7,8 */
val = 0x4000;
snd_ac97_update_bits(chip->ac97, AC97_ALC650_MULTICH, 0xc000, val);
is this correct ???
val = 0x4000; (I think, that there should be 0xc000)
try change this
Peter Zubaj
_
I played with 8233A ALC650 and:
1) this part of code is not working:
switch (runtime->channels) {
case 1: slots = (1<<0) | (1<<4); break;
case 2: slots = (1<<0) | (2<<4); break;
case 3: slots = (1<<0) | (2<<4) | (5<<8); break;
case 4: slots = (1<<0) | (2<<4) | (3<<
This add bass and treble control to SB Audigy. It's modified copy
from SB Live code. It works for me, but I'm not sure if it is good.
Peter Zubaj
http://www.pobox.sk/ - urcujeme trendy
emu_tone.diff
Description: Binary data
Hi,
There is bug in via82xx.c. VIA8233A is wrongly detected as VIA8233
and then not working.
in function snd_via82xx_probe is this
case TYPE_CARD_VIA8233:
chip_type = TYPE_VIA8233;
sprintf(card->shortname, "VIA 823x rev%d", revision);
for (i = 0;
49 matches
Mail list logo
