Re: via82xx 1.0.0rc1 experience

Re: via82xx 1.0.0rc1 experience

[Takashi Iwai]

At Thu, 4 Dec 2003 13:47:07 +0100 (CET),
Tim Goetze wrote:
> 
> i have a new box with an ASRock K7VM2 board, it comes with an on-board
> audio interface:
> 
> <lspci -vv>
> 00:11.5 Multimedia audio controller: VIA Technologies, Inc. AC97 Audio
> Controller (rev 50)
>         Subsystem: Unknown device 1849:3059
>         Control: I/O+ Mem- BusMaster- SpecCycle- MemWINV- VGASnoop-
> ParErr- Stepping- SERR- FastB2B-
>         Status: Cap+ 66Mhz- UDF- FastB2B- ParErr- DEVSEL=medium
> >TAbort- <TAbort- <MAbort- >SERR- <PERR-
>         Interrupt: pin C routed to IRQ 22
>         Region 0: I/O ports at cc00 [size=256]
>         Capabilities: [c0] Power Management version 2
>                 Flags: PMEClk- DSI- D1+ D2+ AuxCurrent=0mA
> PME(D0-,D1-,D2-,D3hot-,D3cold-)
>                 Status: D0 PME-Enable- DSel=0 DScale=0 PME-
> </lspci>
> 
> it works fine on 2.4.22 with alsa-driver-1.0.0rc1 and these module
> options:
> 
> <modules.conf>
> options snd-via82xx index=0 id=ac97 ac97_clock=48000 dxs_support=2
> </modules.conf>
> 
> other dxs_support values cause block-size *and* erratic distortion in
> the output sound. ac97_clock=44100 causes slowed-down playback.

ok, i'll add dxs_support=2 as default for your board.
(still i'm wondering dxs_support=1 or 4 might work.)

as default, ac97_clock is 48000, so you don't need to specify it.
a different ac97_clock shouldn't be given unless the chip is really
driven in that clock.

> however, to enable analog output (only one present on the board) the
> mixer controls for 'headphone' not 'master' must be unmuted (volume
> and playback switch). since they aren't accessible from alsamixer this
> had me puzzling for a while.

please show /proc/asound/card0/codec97#0/* files.
the mixer assignment depends on ac97 codec, not the sound DMA chip
itself.



> one question remains: iiutc, the hw DACs run at a fixed rate of 48kHz.

no, it's in the case dxs_support=3 (and 0 if no pre-definition).

thanks for the feedback.


ciao,

Takashi


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via82xx 1.0.0rc1 experience

[Tim Goetze]

i have a new box with an ASRock K7VM2 board, it comes with an on-board
audio interface:

<lspci -vv>
00:11.5 Multimedia audio controller: VIA Technologies, Inc. AC97 Audio
Controller (rev 50)
        Subsystem: Unknown device 1849:3059
        Control: I/O+ Mem- BusMaster- SpecCycle- MemWINV- VGASnoop-
ParErr- Stepping- SERR- FastB2B-
        Status: Cap+ 66Mhz- UDF- FastB2B- ParErr- DEVSEL=medium
>TAbort- <TAbort- <MAbort- >SERR- <PERR-
        Interrupt: pin C routed to IRQ 22
        Region 0: I/O ports at cc00 [size=256]
        Capabilities: [c0] Power Management version 2
                Flags: PMEClk- DSI- D1+ D2+ AuxCurrent=0mA
PME(D0-,D1-,D2-,D3hot-,D3cold-)
                Status: D0 PME-Enable- DSel=0 DScale=0 PME-
</lspci>

it works fine on 2.4.22 with alsa-driver-1.0.0rc1 and these module
options:

<modules.conf>
options snd-via82xx index=0 id=ac97 ac97_clock=48000 dxs_support=2
</modules.conf>

other dxs_support values cause block-size *and* erratic distortion in
the output sound. ac97_clock=44100 causes slowed-down playback.

however, to enable analog output (only one present on the board) the
mixer controls for 'headphone' not 'master' must be unmuted (volume
and playback switch). since they aren't accessible from alsamixer this
had me puzzling for a while.

before i turned on CONFIG_X86_UP_IOAPIC, the relevant IRQ was shared
between eth0, usb-uhci and snd-via82xx. as a result, when the PCM was
playing, it stopped when there was eth0 traffic (IRQ dropped?) and
resumed playing when the mouse was moved (usb-uhci IRQ?). the oss
driver (via82cxxx_audio) played steadily under the same circumstances.
now things work perfectly however, another welcome opportunity for me
to thank you for your fine efforts.

one question remains: iiutc, the hw DACs run at a fixed rate of 48kHz.
then again when playing 44.1 kHz audio i read this:

<hw_params>
access: RW_INTERLEAVED
format: S16_LE
subformat: STD
channels: 2
rate: 44100 (44100/1)
period_size: 2048
buffer_size: 16384
tick_time: 10000
OSS format: S16_LE
OSS channels: 2
OSS rate: 44100
OSS period bytes: 8192
OSS periods: 8
</hw_params>

so is my understanding about fixed-rate DACs incorrect, or is there
resampling going on? the output audio quality is quite good fwiw.

tim


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Re: via82xx 1.0.0rc1 experience

[Tim Goetze]

Takashi Iwai wrote:

>> <modules.conf>
>> options snd-via82xx index=0 id=ac97 ac97_clock=48000 dxs_support=2
>> </modules.conf>
>>
>> other dxs_support values cause block-size *and* erratic distortion in
>> the output sound. ac97_clock=44100 causes slowed-down playback.
>
>ok, i'll add dxs_support=2 as default for your board.
>(still i'm wondering dxs_support=1 or 4 might work.)

dxs_support=1 produces garbled sound.
dxs_support=4 works ok!

iirc it didn't with the shared interrupt line, want me to try again?
i'm not in the mood for rebooting right now if it needn't be.

tried playing three 16-bit stereo 44.1 kHz streams via OSS-emulation
in parallel, it works. amazing!

>as default, ac97_clock is 48000, so you don't need to specify it.
>a different ac97_clock shouldn't be given unless the chip is really
>driven in that clock.

ok, i was suspecting so.

>> however, to enable analog output (only one present on the board) the
>> mixer controls for 'headphone' not 'master' must be unmuted (volume
>> and playback switch). since they aren't accessible from alsamixer this
>> had me puzzling for a while.
>
>please show /proc/asound/card0/codec97#0/* files.

follows at the end of this mail (module loaded with dxs_support=4).

>thanks for the feedback.

thanks for the quick answer. :)

tim

ps: `cat /proc/asound/card0/codec97#0/*`
0-0/0: VIA Technologies VIA1612A

Capabilities     : -headphone out-
DAC resolution   : 18-bit
ADC resolution   : 18-bit
3D enhancement   : IC Ensemble/KS Waves

Current setup
Mic gain         : +0dB [+0dB]
POP path         : pre 3D
Sim. stereo      : off
3D enhancement   : off
Loudness         : off
Mono output      : MIX
Mic select       : Mic1
ADC/DAC loopback : off
Extended ID      : codec=0 rev=0 AMAP DSA=0 VRA
Extended status  : VRA
PCM front DAC    : 48000Hz
PCM ADC          : 48000Hz
0:00 = 6d50
0:02 = 8808
0:04 = 0808
0:06 = 801f
0:08 = 0000
0:0a = 801e
0:0c = 801f
0:0e = 801f
0:10 = 9f1f
0:12 = 9f1f
0:14 = 9f1f
0:16 = 9f1f
0:18 = 0808
0:1a = 0000
0:1c = 8000
0:1e = 0000
0:20 = 0000
0:22 = 0000
0:24 = 0000
0:26 = 000f
0:28 = 0201
0:2a = 0001
0:2c = bb80
0:2e = 0000
0:30 = 0000
0:32 = bb80
0:34 = 0000
0:36 = 0000
0:38 = 0000
0:3a = 0000
0:3c = 0000
0:3e = 0000
0:40 = 0000
0:42 = 0000
0:44 = 0000
0:46 = 0000
0:48 = 0000
0:4a = 0000
0:4c = 0000
0:4e = 0000
0:50 = 0000
0:52 = 0000
0:54 = 0000
0:56 = 0000
0:58 = 0000
0:5a = 8230
0:5c = 0010
0:5e = f7ff
0:60 = 0000
0:62 = 0000
0:64 = 0000
0:66 = 0000
0:68 = 0000
0:6a = 0000
0:6c = 0000
0:6e = 0000
0:70 = 0000
0:72 = 0000
0:74 = 0000
0:76 = 0000
0:78 = 0000
0:7a = 0000
0:7c = 5649
0:7e = 4161



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Can't set 88.2/96kHz samplerate with Hammerfall

[Arve Knudsen]

Hi

Whenever I try to activate double speed (88.2/96kHz) mode with my RME 
Hammerfall Lite (DIGI 9636), snd_pcm_hw_params fails with a 'Device or 
resource busy' message. This is with the number of channels set to 10, 
since the number of ADAT channels is halved. Is this a known defect of the 
Hammerfall driver?

Thanks

Arve Knudsen


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RE: Can't set 88.2/96kHz samplerate with Hammerfall

[Mark Knecht]

> 
> Whenever I try to activate double speed (88.2/96kHz) mode with my RME 
> Hammerfall Lite (DIGI 9636), snd_pcm_hw_params fails with a 'Device or 
> resource busy' message. This is with the number of channels set to 10, 
> since the number of ADAT channels is halved. Is this a known 
> defect of the 
> Hammerfall driver?
> 
> Thanks
> 
> Arve Knudsen

>From the Hammerfall Light Alsa sound card page:

Known bugs

- 96kHz and 88.2kHz not accessible via PCM interface 




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RE: Can't set 88.2/96kHz samplerate with Hammerfall

[Mark Knecht]

> >
> > From the Hammerfall Light Alsa sound card page:
> >
> > Known bugs
> >
> > - 96kHz and 88.2kHz not accessible via PCM interface
> Ok, thanks. I suppose I should've investigated a little better.
> Too bad, I
> would like to see this card working as well as under Windows.
>
> Regards
>
> Arve Knudsen

Me too. I own the same card in one of my systems. This is really the
downside to Linux audio. Without dedicated developers that own the cards and
continue working on improvements support stalls out. The best I've been able
to determine is that this card last got a real facelift over a year ago and
since then nothing. I've grown to accept that it will never happen, and will
be overjoyed that was if and when it finally does. Don't expect anything
soon, unless you're willing to be that developer.

BTW - if you get a chance, take a look at /proc/asound/card0/hdsp  (modify
the card number if required and I'm guessing on the hdsp part) and see if
it's listing spdif under the status section? Mine doesn't, or didn't, the
last time I looked. It indicated 3 ADAT ports which this card does not have.

- Mark




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Re: Can't set 88.2/96kHz samplerate with Hammerfall

[Paul Davis]

>On Thu, 4 Dec 2003 06:41:56 -0800, Mark Knecht <mknecht@controlnet.com>=20
>wrote:
>
>>>
>>> Whenever I try to activate double speed (88.2/96kHz) mode with my RME
>>> Hammerfall Lite (DIGI 9636), snd_pcm_hw_params fails with a 'Device or
>>> resource busy' message. This is with the number of channels set to 10,
>>> since the number of ADAT channels is halved. Is this a known
>>> defect of the
>>> Hammerfall driver?
>>>
>>> Thanks
>>>
>>> Arve Knudsen
>>
>> From the Hammerfall Light Alsa sound card page:
>>
>> Known bugs
>>
>> - 96kHz and 88.2kHz not accessible via PCM interface
>Ok, thanks. I suppose I should've investigated a little better. Too bad, =
>I=20
>would like to see this card working as well as under Windows.

to be clear: you can't get DS mode using the PCM interface but you CAN
(and several people have) do this, you just have to use the control
interface in addition.

--p


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Re: Can't set 88.2/96kHz samplerate with Hammerfall

[Arve Knudsen]

On Thu, 4 Dec 2003 06:41:56 -0800, Mark Knecht <mknecht@controlnet.com> 
wrote:

>>
>> Whenever I try to activate double speed (88.2/96kHz) mode with my RME
>> Hammerfall Lite (DIGI 9636), snd_pcm_hw_params fails with a 'Device or
>> resource busy' message. This is with the number of channels set to 10,
>> since the number of ADAT channels is halved. Is this a known
>> defect of the
>> Hammerfall driver?
>>
>> Thanks
>>
>> Arve Knudsen
>
> From the Hammerfall Light Alsa sound card page:
>
> Known bugs
>
> - 96kHz and 88.2kHz not accessible via PCM interface
Ok, thanks. I suppose I should've investigated a little better. Too bad, I 
would like to see this card working as well as under Windows.

Regards

Arve Knudsen


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Re: Can't set 88.2/96kHz samplerate with Hammerfall

[Arve Knudsen]

On Thu, 4 Dec 2003 09:43:36 -0800, Mark Knecht <mknecht@controlnet.com> 
wrote:

>> >
>> > From the Hammerfall Light Alsa sound card page:
>> >
>> > Known bugs
>> >
>> > - 96kHz and 88.2kHz not accessible via PCM interface
>> Ok, thanks. I suppose I should've investigated a little better.
>> Too bad, I
>> would like to see this card working as well as under Windows.
>>
>> Regards
>>
>> Arve Knudsen
>
> Me too. I own the same card in one of my systems. This is really the
> downside to Linux audio. Without dedicated developers that own the cards 
> and
> continue working on improvements support stalls out. The best I've been 
> able
> to determine is that this card last got a real facelift over a year ago 
> and
> since then nothing. I've grown to accept that it will never happen, and 
> will
> be overjoyed that was if and when it finally does. Don't expect anything
> soon, unless you're willing to be that developer.
Actually, I'm not completely opposed to the idea. But I am totally 
clueless when it comes to writing device drivers, and would have no idea 
where to start. Do you know if RME were forthcoming with specs?

Regards

Arve Knudsen


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Re: Can't set 88.2/96kHz samplerate with Hammerfall

[Paul Davis]

>> be overjoyed that was if and when it finally does. Don't expect anythin=
>g
>> soon, unless you're willing to be that developer.
>Actually, I'm not completely opposed to the idea. But I am totally=20
>clueless when it comes to writing device drivers, and would have no idea=20
>where to start. Do you know if RME were forthcoming with specs?

they were. but you can also offer to pay someone to fix the
driver. the correct model already exists in the RME HDSP driver, and
it wouldn't be that hard to implement for the digi9652. perhaps 1-3
hours work overall.

--p


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Re: Can't set 88.2/96kHz samplerate with Hammerfall

[Arve Knudsen]

On Thu, 04 Dec 2003 17:25:53 -0500, Paul Davis 
<paul@linuxaudiosystems.com> wrote:

>> On Thu, 4 Dec 2003 06:41:56 -0800, Mark Knecht 
>> <mknecht@controlnet.com>=20
>> wrote:
>>
>>>>
>>>> Whenever I try to activate double speed (88.2/96kHz) mode with my RME
>>>> Hammerfall Lite (DIGI 9636), snd_pcm_hw_params fails with a 'Device or
>>>> resource busy' message. This is with the number of channels set to 10,
>>>> since the number of ADAT channels is halved. Is this a known
>>>> defect of the
>>>> Hammerfall driver?
>>>>
>>>> Thanks
>>>>
>>>> Arve Knudsen
>>>
>>> From the Hammerfall Light Alsa sound card page:
>>>
>>> Known bugs
>>>
>>> - 96kHz and 88.2kHz not accessible via PCM interface
>> Ok, thanks. I suppose I should've investigated a little better. Too 
>> bad, =
>> I=20
>> would like to see this card working as well as under Windows.
>
> to be clear: you can't get DS mode using the PCM interface but you CAN
> (and several people have) do this, you just have to use the control
> interface in addition.
Ah, didn't see your mail untill now. For some reason, you wind up under 
spam (Opera mail client filter) :\ I suppose I should investigate this 
possibility.

Thanks

Arve Knudsen


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Re: Can't set 88.2/96kHz samplerate with Hammerfall

[Giuliano Pochini]

> Actually, I'm not completely opposed to the idea. But I am totally
> clueless when it comes to writing device drivers, and would have no
> idea where to start.

Start here: http://www.alsa-project.org/documentation.php3#Driver

You don't have to start writing the driver from scratch. You only
have to fix the current driver so you can also select 96KHz from
the PCM interface. The API is not difficult to learn.

> Do you know if RME were forthcoming with specs?

I asked for infos one year ago, but they just said no.


--
Giuliano.


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Re: Can't set 88.2/96kHz samplerate with Hammerfall

[Paul Davis]

>
>> Actually, I'm not completely opposed to the idea. But I am totally
>> clueless when it comes to writing device drivers, and would have no
>> idea where to start.
>
>Start here: http://www.alsa-project.org/documentation.php3#Driver
>
>You don't have to start writing the driver from scratch. You only
>have to fix the current driver so you can also select 96KHz from
>the PCM interface. The API is not difficult to learn.

there's almost no API work to do. the problem is really simple and
already solved in a closely related driver.

when you change between an SS (44.1/48) and DS rate (88.2/96), it
changes the channel count. this would be a bad thing if the PCM device
is open, because it alters all the addresses of the channel buffers
etc. etc. so the driver prevents you from doing this if the PCM device
is open. this creates a paradox, because you've already got the device
open.

the hdsp driver takes a different approach to this, even though it
faces an identical problem. its a better design, one based on our
experience with the digi9652 driver and its shortcomings. i'd love to
go in and fix the digi9652, but short of some compelling reason for me
to do so, with the work on ardour+jack at the level it is right now, i
just don't have time.

>> Do you know if RME were forthcoming with specs?
>
>I asked for infos one year ago, but they just said no.

they don't want to get involved with lots of people being sent
specs. the driver itself is the best source of documentation - its
more accurate than the information they sent me :)

--p


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