* via686/AD1886/Soundmax drivers
@ 2002-02-19 9:03 Peter Christy
2002-02-19 12:25 ` Vojtech Pavlik
0 siblings, 1 reply; 8+ messages in thread
From: Peter Christy @ 2002-02-19 9:03 UTC (permalink / raw)
To: linux-kernel
There was a thread running a month or so back concerning problems getting
the via686/AD1886 combination up and running (this chipset combination
seems to be mostly used on laptops). The original thread concerned Compaq
laptops, but I have a FIC A360+ which uses the same chipset and am having
the same problems. The original thread seemed to fizzle out with no real
solution found.
To summarize the issue: None of the available drivers work with this
chipset combination. The kernel driver fails to recognize the AD1886
correctly, the drivers on the Via website just don't work, and the Alsa
drivers recognize the mixer correctly, but still produce no sound.
The original thread seemed to suggest that the mixer was not being
initialized correctly, resulting in very faint or no sound output.
I am not a serious programmer (I peaked with 6502s and haven't done any
machine level stuff since!), but following hints in the earlier threads I
added the following lines to the kernel source ac97_codec.c file:
{0x41445361, "Analog Devices AD1886", &default_ops},
{0x41445461, "Analog Devices AD1886", &default_ops},
I added both lines as I had conflicting information as to the 0x4144
numbers.
After re-compiling the kernel, the kernel driver now correctly identifies
the mixer.
Using Kmix from KDE, if I open up the mic channel, I can get howl round. I
can play cds at normal volume. In other words, the mixer appears to be
working. However, I cannot get any wav or midi files to play. ie "computer
generated" sounds, as opposed to "peripheral" devices, don't get through.
This contradicts conclusions reached in earlier threads suggesting that
the mixer was muting the audio output.
I use the via686 kernel drivers on my desktop without problems. But my
investigations make me think it is the Via686 bit which is not working
rather than the AD1886. The only difference I can see is that the laptop
chipset is a via686b rather than a via686a.
I have now reached the limits of my knowledge on what to try next. Any
suggestions gratefully received.
Please CC to christy@attglobal.net
--
Pete
christy@attglobal.net
^ permalink raw reply [flat|nested] 8+ messages in thread* Re: via686/AD1886/Soundmax drivers
2002-02-19 9:03 via686/AD1886/Soundmax drivers Peter Christy
@ 2002-02-19 12:25 ` Vojtech Pavlik
2002-02-21 9:56 ` Peter Christy
0 siblings, 1 reply; 8+ messages in thread
From: Vojtech Pavlik @ 2002-02-19 12:25 UTC (permalink / raw)
To: Peter Christy; +Cc: linux-kernel
On Tue, Feb 19, 2002 at 09:03:03AM +0000, Peter Christy wrote:
> I am not a serious programmer (I peaked with 6502s and haven't done any
> machine level stuff since!), but following hints in the earlier threads I
> added the following lines to the kernel source ac97_codec.c file:
>
> {0x41445361, "Analog Devices AD1886", &default_ops},
> {0x41445461, "Analog Devices AD1886", &default_ops},
>
> I added both lines as I had conflicting information as to the 0x4144
> numbers.
It's simple, they're "ADS" ...
--
Vojtech Pavlik
SuSE Labs
^ permalink raw reply [flat|nested] 8+ messages in thread* Re: via686/AD1886/Soundmax drivers
2002-02-19 12:25 ` Vojtech Pavlik
@ 2002-02-21 9:56 ` Peter Christy
2002-02-21 10:15 ` Vojtech Pavlik
0 siblings, 1 reply; 8+ messages in thread
From: Peter Christy @ 2002-02-21 9:56 UTC (permalink / raw)
To: Vojtech Pavlik; +Cc: linux-kernel
On Tue, 19 Feb 2002 13:25:00 +0100
Vojtech Pavlik <vojtech@suse.cz> wrote:
> It's simple, they're "ADS" ...
????? Forgive my ignorance, but that doesn't mean much to me!
Have you had any success with these chips? SuSE Live Eval 7.3 correctly
identifies the chips, but still doesn't produce any sound!
--
Pete
christy@attglobal.net
^ permalink raw reply [flat|nested] 8+ messages in thread
* Re: via686/AD1886/Soundmax drivers
2002-02-21 9:56 ` Peter Christy
@ 2002-02-21 10:15 ` Vojtech Pavlik
2002-02-21 11:16 ` Peter Christy
0 siblings, 1 reply; 8+ messages in thread
From: Vojtech Pavlik @ 2002-02-21 10:15 UTC (permalink / raw)
To: Peter Christy; +Cc: linux-kernel
On Thu, Feb 21, 2002 at 09:56:53AM +0000, Peter Christy wrote:
> > It's simple, they're "ADS" ...
>
> ????? Forgive my ignorance, but that doesn't mean much to me!
Look up the ASCII codes for the letters A D and S, they'll be the same
as the hex numbers of the codec ID ...
> Have you had any success with these chips? SuSE Live Eval 7.3 correctly
> identifies the chips, but still doesn't produce any sound!
No, sorry.
--
Vojtech Pavlik
SuSE Labs
^ permalink raw reply [flat|nested] 8+ messages in thread
* via686/AD1886/Soundmax drivers
@ 2002-02-27 16:54 Peter Christy
2002-02-27 19:11 ` André Dahlqvist
0 siblings, 1 reply; 8+ messages in thread
From: Peter Christy @ 2002-02-27 16:54 UTC (permalink / raw)
To: linux-kernel
Ths saga continues..... (no sound on laptops using via686 and AD1886s -
mostly FIC A360+ and Compaq Presarios)
Following a trawl around the net, I found a patched version of
AC97_codec.c with alternative eapd control options. (apologies to whoever
originally submitted it - I can't find it again now!). After several
attempts, I got it to compile, and added an extra option to use the
default_ops as well. I've now tried it with all three options, and the
results are as shown below:
Error messages (dmesg)
### modified ac97_codec.c using default_ops ###
### this produces audio, dsp, dspW, mixer, sequencer & sequencer2 in
/dev/sound (devfs) ###
Via 686a audio driver 1.9.1
PCI: Enabling device 00:07.5 (0000 -> 0001)
PCI: Found IRQ 4 for device 00:07.5
IRQ routing conflict for 00:0c.1, have irq 10, want irq 4
ac97_codec: AC97 Audio codec, id: 0x4144:0x5361 (Analog Devices AD1886)
via82cxxx: board #1 at 0x1800, IRQ 4
.
.
.
.
via_audio: ignoring drain playback error -11
Assertion failed! chan->is_active ==
sg_active(chan->iobase),via82cxxx_audio.c,via_chan_maybe_start,line=1198
### modified ac97_codec.c, using eapd_init_ops ###
### note: this only produces dsp & mixer in /dev/sound (devfs) ###
Via 686a audio driver 1.9.1
PCI: Enabling device 00:07.5 (0000 -> 0001)
PCI: Found IRQ 4 for device 00:07.5
IRQ routing conflict for 00:0c.1, have irq 10, want irq 4
ac97_codec: AC97 Audio codec, id: 0x4144:0x5361 (Analog Devices AD1886)
via82cxxx: board #1 at 0x1800, IRQ 4
.
.
.
.
via_audio: ignoring drain playback error -11
Assertion failed! chan->is_active ==
sg_active(chan->iobase),via82cxxx_audio.c,via_chan_maybe_start,line=1198
via_audio: ignoring drain playback error -11
### modified ac97_codec.c,using eapd_init2_ops ###
### this produces audio, dsp, dspW, mixer, sequencer & sequencer2 in
/dev/sound (devfs) ###
Via 686a audio driver 1.9.1
PCI: Enabling device 00:07.5 (0000 -> 0001)
PCI: Found IRQ 4 for device 00:07.5
IRQ routing conflict for 00:0c.1, have irq 10, want irq 4
ac97_codec: AC97 Audio codec, id: 0x4144:0x5361 (Analog Devices AD1886)
via82cxxx: board #1 at 0x1800, IRQ 4
.
.
.
.
via_audio: ignoring drain playback error -11
Assertion failed! chan->is_active ==
sg_active(chan->iobase),via82cxxx_audio.c,via_chan_maybe_start,line=1198
via_audio: ignoring drain playback error -11
Obviously the eapd_init_ops doesn't properly initialize the chip, as it
doesn't produce a full set of devices in /dev/sound (I'm running devfs).
Both eapd_init2_ops and default_ops produce a full set of devices. All
three options allow the playing of CDs at normal volume, and bring the mic
up. NONE of them produce any sound from wav or midi files! (kmidi gives an
error reporting that it can't open /dev/sequncer - something else must be
using it - but there is nothing else using it and it does exist!)
dmesg seems to imply that there is a problem with the via82cxxx_audio
driver, but that same driver works fine on my desktop machine.
I'm rapidly coming to the conclusion that the problem lies NOT with the
AD1886 or ac97_codec (aside from the missing AD1886 lines), but that Via
have changed something in the 686 southbridge rendering the existing
driver useless. Unfortunately, I'm now struggling at the very limits of my
knowledge of programming! I can find where an error is occuring in the
via82cxxx_audio file, but I've no idea why or what to do about it! Any
suggestions gratefully received!
I'm appending the modified ac97_codec.c file. Apologies for not crediting
the original author!
Please CC to christy@attglobal.net
--
Pete
christy@attglobal.net
/*
* ac97_codec.c: Generic AC97 mixer/modem module
*
* Derived from ac97 mixer in maestro and trident driver.
*
* Copyright 2000 Silicon Integrated System Corporation
*
* This program is free software; you can redistribute it and/or
modify
* it under the terms of the GNU General Public License as published
by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*
**************************************************************************
*
* The Intel Audio Codec '97 specification is available at the Intel
* audio homepage: http://developer.intel.com/ial/scalableplatforms/audio/
*
* The specification itself is currently available at:
* ftp://download.intel.com/ial/scalableplatforms/ac97r22.pdf
*
**************************************************************************
*
* History
* v0.4 Mar 15 2000 Ollie Lho
* dual codecs support verified with 4 channels output
* v0.3 Feb 22 2000 Ollie Lho
* bug fix for record mask setting
* v0.2 Feb 10 2000 Ollie Lho
* add ac97_read_proc for /proc/driver/{vendor}/ac97
* v0.1 Jan 14 2000 Ollie Lho <ollie@sis.com.tw>
* Isolated from trident.c to support multiple ac97 codec
*/
#include <linux/module.h>
#include <linux/version.h>
#include <linux/kernel.h>
#include <linux/string.h>
#include <linux/errno.h>
#include <linux/bitops.h>
#include <linux/delay.h>
#include <linux/ac97_codec.h>
#include <asm/uaccess.h>
static int ac97_read_mixer(struct ac97_codec *codec, int oss_channel);
static void ac97_write_mixer(struct ac97_codec *codec, int oss_channel,
unsigned int left, unsigned int right);
static void ac97_set_mixer(struct ac97_codec *codec, unsigned int
oss_mixer, unsigned int val );
static int ac97_recmask_io(struct ac97_codec *codec, int rw, int mask);
static int ac97_mixer_ioctl(struct ac97_codec *codec, unsigned int cmd,
unsigned long arg);
static int ac97_init_mixer(struct ac97_codec *codec);
static int wolfson_init(struct ac97_codec * codec);
static int tritech_init(struct ac97_codec * codec);
static int tritech_maestro_init(struct ac97_codec * codec);
static int sigmatel_9708_init(struct ac97_codec *codec);
static int sigmatel_9721_init(struct ac97_codec *codec);
static int sigmatel_9744_init(struct ac97_codec *codec);
static int eapd_control(struct ac97_codec *codec, int);
static int eapd_on_control(struct ac97_codec *codec);
static int eapd_off_control(struct ac97_codec *codec);
static int crystal_digital_control(struct ac97_codec *codec, int mode);
/*
* AC97 operations.
*
* If you are adding a codec then you should be able to use
* eapd_ops - any codec that supports EAPD amp control (most)
* null_ops - any ancient codec that supports nothing
*
* The three functions are
* init - used for non AC97 standard initialisation
* amplifier - used to do amplifier control (1=on 0=off)
* digital - switch to digital modes (0 = analog)
*
* Not all codecs support all features, not all drivers use all the
* operations yet
*/
static struct ac97_ops null_ops = { NULL, NULL, NULL };
static struct ac97_ops default_ops = { NULL, eapd_control, NULL };
static struct ac97_ops eapd_init_ops = { eapd_on_control, NULL, NULL };
static struct ac97_ops eapd_init2_ops = { eapd_off_control, NULL, NULL };
static struct ac97_ops wolfson_ops = { wolfson_init, NULL, NULL };
static struct ac97_ops tritech_ops = { tritech_init, NULL, NULL };
static struct ac97_ops tritech_m_ops = { tritech_maestro_init, NULL, NULL
};
static struct ac97_ops sigmatel_9708_ops = { sigmatel_9708_init, NULL,
NULL };
static struct ac97_ops sigmatel_9721_ops = { sigmatel_9721_init, NULL,
NULL };
static struct ac97_ops sigmatel_9744_ops = { sigmatel_9744_init, NULL,
NULL };
static struct ac97_ops crystal_digital_ops = { NULL, eapd_control,
crystal_digital_control };
/* sorted by vendor/device id */
static const struct {
u32 id;
char *name;
struct ac97_ops *ops;
} ac97_codec_ids[] = {
{0x41445303, "Analog Devices AD1819", &null_ops},
{0x41445340, "Analog Devices AD1881", &null_ops},
{0x41445348, "Analog Devices AD1881A", &null_ops},
{0x41445360, "Analog Devices AD1885", &default_ops},
{0x41445460, "Analog Devices AD1885", &default_ops},
{0x41445361, "Analog Devices AD1886", &default_ops},
// {0x41445361, "Analog Devices AD1886", &eapd_init_ops},
// {0x41445361, "Analog Devices AD1886", &eapd_init2_ops},
{0x414B4D00, "Asahi Kasei AK4540", &null_ops},
{0x414B4D01, "Asahi Kasei AK4542", &null_ops},
{0x414B4D02, "Asahi Kasei AK4543", &null_ops},
{0x414C4710, "ALC200/200P", &null_ops},
{0x43525900, "Cirrus Logic CS4297", &default_ops},
{0x43525903, "Cirrus Logic CS4297", &default_ops},
{0x43525913, "Cirrus Logic CS4297A rev A", &default_ops},
{0x43525914, "Cirrus Logic CS4297A rev B", &default_ops},
{0x43525923, "Cirrus Logic CS4298", &null_ops},
{0x4352592B, "Cirrus Logic CS4294", &null_ops},
{0x4352592D, "Cirrus Logic CS4294", &null_ops},
{0x43525931, "Cirrus Logic CS4299 rev A", &crystal_digital_ops},
{0x43525933, "Cirrus Logic CS4299 rev C", &crystal_digital_ops},
{0x43525934, "Cirrus Logic CS4299 rev D", &crystal_digital_ops},
{0x45838308, "ESS Allegro ES1988", &null_ops},
{0x49434511, "ICE1232", &null_ops}, /* I hope --jk
*/
{0x4e534331, "National Semiconductor LM4549", &null_ops},
{0x53494c22, "Silicon Laboratory Si3036", &null_ops},
{0x53494c23, "Silicon Laboratory Si3038", &null_ops},
{0x545200FF, "TriTech TR?????", &tritech_m_ops},
{0x54524102, "TriTech TR28022", &null_ops},
{0x54524103, "TriTech TR28023", &null_ops},
{0x54524106, "TriTech TR28026", &null_ops},
{0x54524108, "TriTech TR28028", &tritech_ops},
{0x54524123, "TriTech TR A5", &null_ops},
{0x574D4C00, "Wolfson WM9704", &wolfson_ops},
{0x574D4C03, "Wolfson WM9703/9704", &wolfson_ops},
{0x574D4C04, "Wolfson WM9704 (quad)", &wolfson_ops},
{0x83847600, "SigmaTel STAC????", &null_ops},
{0x83847604, "SigmaTel STAC9701/3/4/5", &null_ops},
{0x83847605, "SigmaTel STAC9704", &null_ops},
{0x83847608, "SigmaTel STAC9708", &sigmatel_9708_ops},
{0x83847609, "SigmaTel STAC9721/23", &sigmatel_9721_ops},
{0x83847644, "SigmaTel STAC9744/45", &sigmatel_9744_ops},
{0x83847656, "SigmaTel STAC9756/57", &sigmatel_9744_ops},
{0x83847684, "SigmaTel STAC9783/84?", &null_ops},
{0x57454301, "Winbond 83971D", &null_ops},
};
static const char *ac97_stereo_enhancements[] =
{
/* 0 */ "No 3D Stereo Enhancement",
/* 1 */ "Analog Devices Phat Stereo",
/* 2 */ "Creative Stereo Enhancement",
/* 3 */ "National Semi 3D Stereo Enhancement",
/* 4 */ "YAMAHA Ymersion",
/* 5 */ "BBE 3D Stereo Enhancement",
/* 6 */ "Crystal Semi 3D Stereo Enhancement",
/* 7 */ "Qsound QXpander",
/* 8 */ "Spatializer 3D Stereo Enhancement",
/* 9 */ "SRS 3D Stereo Enhancement",
/* 10 */ "Platform Tech 3D Stereo Enhancement",
/* 11 */ "AKM 3D Audio",
/* 12 */ "Aureal Stereo Enhancement",
/* 13 */ "Aztech 3D Enhancement",
/* 14 */ "Binaura 3D Audio Enhancement",
/* 15 */ "ESS Technology Stereo Enhancement",
/* 16 */ "Harman International VMAx",
/* 17 */ "Nvidea 3D Stereo Enhancement",
/* 18 */ "Philips Incredible Sound",
/* 19 */ "Texas Instruments 3D Stereo Enhancement",
/* 20 */ "VLSI Technology 3D Stereo Enhancement",
/* 21 */ "TriTech 3D Stereo Enhancement",
/* 22 */ "Realtek 3D Stereo Enhancement",
/* 23 */ "Samsung 3D Stereo Enhancement",
/* 24 */ "Wolfson Microelectronics 3D Enhancement",
/* 25 */ "Delta Integration 3D Enhancement",
/* 26 */ "SigmaTel 3D Enhancement",
/* 27 */ "Winbond 3D Stereo Enhancement",
/* 28 */ "Rockwell 3D Stereo Enhancement",
/* 29 */ "Reserved 29",
/* 30 */ "Reserved 30",
/* 31 */ "Reserved 31"
};
/* this table has default mixer values for all OSS mixers. */
static struct mixer_defaults {
int mixer;
unsigned int value;
} mixer_defaults[SOUND_MIXER_NRDEVICES] = {
/* all values 0 -> 100 in bytes */
{SOUND_MIXER_VOLUME, 0x4343},
{SOUND_MIXER_BASS, 0x4343},
{SOUND_MIXER_TREBLE, 0x4343},
{SOUND_MIXER_PCM, 0x4343},
{SOUND_MIXER_SPEAKER, 0x4343},
{SOUND_MIXER_LINE, 0x4343},
{SOUND_MIXER_MIC, 0x0000},
{SOUND_MIXER_CD, 0x4343},
{SOUND_MIXER_ALTPCM, 0x4343},
{SOUND_MIXER_IGAIN, 0x4343},
{SOUND_MIXER_LINE1, 0x4343},
{SOUND_MIXER_PHONEIN, 0x4343},
{SOUND_MIXER_PHONEOUT, 0x4343},
{SOUND_MIXER_VIDEO, 0x4343},
{-1,0}
};
/* table to scale scale from OSS mixer value to AC97 mixer register value
*/
static struct ac97_mixer_hw {
unsigned char offset;
int scale;
} ac97_hw[SOUND_MIXER_NRDEVICES]= {
[SOUND_MIXER_VOLUME] = {AC97_MASTER_VOL_STEREO,64},
[SOUND_MIXER_BASS] = {AC97_MASTER_TONE, 16},
[SOUND_MIXER_TREBLE] = {AC97_MASTER_TONE, 16},
[SOUND_MIXER_PCM] = {AC97_PCMOUT_VOL, 32},
[SOUND_MIXER_SPEAKER] = {AC97_PCBEEP_VOL, 16},
[SOUND_MIXER_LINE] = {AC97_LINEIN_VOL, 32},
[SOUND_MIXER_MIC] = {AC97_MIC_VOL, 32},
[SOUND_MIXER_CD] = {AC97_CD_VOL, 32},
[SOUND_MIXER_ALTPCM] = {AC97_HEADPHONE_VOL, 64},
[SOUND_MIXER_IGAIN] = {AC97_RECORD_GAIN, 16},
[SOUND_MIXER_LINE1] = {AC97_AUX_VOL, 32},
[SOUND_MIXER_PHONEIN] = {AC97_PHONE_VOL, 32},
[SOUND_MIXER_PHONEOUT] = {AC97_MASTER_VOL_MONO, 64},
[SOUND_MIXER_VIDEO] = {AC97_VIDEO_VOL, 32},
};
/* the following tables allow us to go from OSS <-> ac97 quickly. */
enum ac97_recsettings {
AC97_REC_MIC=0,
AC97_REC_CD,
AC97_REC_VIDEO,
AC97_REC_AUX,
AC97_REC_LINE,
AC97_REC_STEREO, /* combination of all enabled outputs.. */
AC97_REC_MONO, /*.. or the mono equivalent */
AC97_REC_PHONE
};
static const unsigned int ac97_rm2oss[] = {
[AC97_REC_MIC] = SOUND_MIXER_MIC,
[AC97_REC_CD] = SOUND_MIXER_CD,
[AC97_REC_VIDEO] = SOUND_MIXER_VIDEO,
[AC97_REC_AUX] = SOUND_MIXER_LINE1,
[AC97_REC_LINE] = SOUND_MIXER_LINE,
[AC97_REC_STEREO]= SOUND_MIXER_IGAIN,
[AC97_REC_PHONE] = SOUND_MIXER_PHONEIN
};
/* indexed by bit position */
static const unsigned int ac97_oss_rm[] = {
[SOUND_MIXER_MIC] = AC97_REC_MIC,
[SOUND_MIXER_CD] = AC97_REC_CD,
[SOUND_MIXER_VIDEO] = AC97_REC_VIDEO,
[SOUND_MIXER_LINE1] = AC97_REC_AUX,
[SOUND_MIXER_LINE] = AC97_REC_LINE,
[SOUND_MIXER_IGAIN] = AC97_REC_STEREO,
[SOUND_MIXER_PHONEIN] = AC97_REC_PHONE
};
/* reads the given OSS mixer from the ac97 the caller must have insured
that the ac97 knows
about that given mixer, and should be holding a spinlock for the card
*/
static int ac97_read_mixer(struct ac97_codec *codec, int oss_channel)
{
u16 val;
int ret = 0;
int scale;
struct ac97_mixer_hw *mh = &ac97_hw[oss_channel];
val = codec->codec_read(codec , mh->offset);
if (val & AC97_MUTE) {
ret = 0;
} else if (AC97_STEREO_MASK & (1 << oss_channel)) {
/* nice stereo mixers .. */
int left,right;
left = (val >> 8) & 0x7f;
right = val & 0x7f;
if (oss_channel == SOUND_MIXER_IGAIN) {
right = (right * 100) / mh->scale;
left = (left * 100) / mh->scale;
} else {
/* these may have 5 or 6 bit resolution */
if(oss_channel == SOUND_MIXER_VOLUME ||
oss_channel == SOUND_MIXER_ALTPCM)
scale = (1 << codec->bit_resolution);
else
scale = mh->scale;
right = 100 - ((right * 100) / scale);
left = 100 - ((left * 100) / scale);
}
ret = left | (right << 8);
} else if (oss_channel == SOUND_MIXER_SPEAKER) {
ret = 100 - ((((val & 0x1e)>>1) * 100) / mh->scale);
} else if (oss_channel == SOUND_MIXER_PHONEIN) {
ret = 100 - (((val & 0x1f) * 100) / mh->scale);
} else if (oss_channel == SOUND_MIXER_PHONEOUT) {
scale = (1 << codec->bit_resolution);
ret = 100 - (((val & 0x1f) * 100) / scale);
} else if (oss_channel == SOUND_MIXER_MIC) {
ret = 100 - (((val & 0x1f) * 100) / mh->scale);
/* the low bit is optional in the tone sliders and
masking
it lets us avoid the 0xf 'bypass'.. */
} else if (oss_channel == SOUND_MIXER_BASS) {
ret = 100 - ((((val >> 8) & 0xe) * 100) / mh->scale);
} else if (oss_channel == SOUND_MIXER_TREBLE) {
ret = 100 - (((val & 0xe) * 100) / mh->scale);
}
#ifdef DEBUG
printk("ac97_codec: read OSS mixer %2d (%s ac97 register 0x%02x),
"
"0x%04x -> 0x%04x\n",
oss_channel, codec->id ? "Secondary" : "Primary",
mh->offset, val, ret);
#endif
return ret;
}
/* write the OSS encoded volume to the given OSS encoded mixer, again
caller's job to
make sure all is well in arg land, call with spinlock held */
static void ac97_write_mixer(struct ac97_codec *codec, int oss_channel,
unsigned int left, unsigned int right)
{
u16 val = 0;
int scale;
struct ac97_mixer_hw *mh = &ac97_hw[oss_channel];
#ifdef DEBUG
printk("ac97_codec: wrote OSS mixer %2d (%s ac97 register 0x%02x),
"
"left vol:%2d, right vol:%2d:",
oss_channel, codec->id ? "Secondary" : "Primary",
mh->offset, left, right);
#endif
if (AC97_STEREO_MASK & (1 << oss_channel)) {
/* stereo mixers */
if (left == 0 && right == 0) {
val = AC97_MUTE;
} else {
if (oss_channel == SOUND_MIXER_IGAIN) {
right = (right * mh->scale) / 100;
left = (left * mh->scale) / 100;
if (right >= mh->scale)
right = mh->scale-1;
if (left >= mh->scale)
left = mh->scale-1;
} else {
/* these may have 5 or 6 bit resolution */
if (oss_channel == SOUND_MIXER_VOLUME ||
oss_channel == SOUND_MIXER_ALTPCM)
scale = (1 <<
codec->bit_resolution);
else
scale = mh->scale;
right = ((100 - right) * scale) / 100;
left = ((100 - left) * scale) / 100;
if (right >= scale)
right = scale-1;
if (left >= scale)
left = scale-1;
}
val = (left << 8) | right;
}
} else if (oss_channel == SOUND_MIXER_BASS) {
val = codec->codec_read(codec , mh->offset) & ~0x0f00;
left = ((100 - left) * mh->scale) / 100;
if (left >= mh->scale)
left = mh->scale-1;
val |= (left << 8) & 0x0e00;
} else if (oss_channel == SOUND_MIXER_TREBLE) {
val = codec->codec_read(codec , mh->offset) & ~0x000f;
left = ((100 - left) * mh->scale) / 100;
if (left >= mh->scale)
left = mh->scale-1;
val |= left & 0x000e;
} else if(left == 0) {
val = AC97_MUTE;
} else if (oss_channel == SOUND_MIXER_SPEAKER) {
left = ((100 - left) * mh->scale) / 100;
if (left >= mh->scale)
left = mh->scale-1;
val = left << 1;
} else if (oss_channel == SOUND_MIXER_PHONEIN) {
left = ((100 - left) * mh->scale) / 100;
if (left >= mh->scale)
left = mh->scale-1;
val = left;
} else if (oss_channel == SOUND_MIXER_PHONEOUT) {
scale = (1 << codec->bit_resolution);
left = ((100 - left) * scale) / 100;
if (left >= mh->scale)
left = mh->scale-1;
val = left;
} else if (oss_channel == SOUND_MIXER_MIC) {
val = codec->codec_read(codec , mh->offset) & ~0x801f;
left = ((100 - left) * mh->scale) / 100;
if (left >= mh->scale)
left = mh->scale-1;
val |= left;
/* the low bit is optional in the tone sliders and
masking
it lets us avoid the 0xf 'bypass'.. */
}
#ifdef DEBUG
printk(" 0x%04x", val);
#endif
codec->codec_write(codec, mh->offset, val);
#ifdef DEBUG
val = codec->codec_read(codec, mh->offset);
printk(" -> 0x%04x\n", val);
#endif
}
/* a thin wrapper for write_mixer */
static void ac97_set_mixer(struct ac97_codec *codec, unsigned int
oss_mixer, unsigned int val )
{
unsigned int left,right;
/* cleanse input a little */
right = ((val >> 8) & 0xff) ;
left = (val & 0xff) ;
if (right > 100) right = 100;
if (left > 100) left = 100;
codec->mixer_state[oss_mixer] = (right << 8) | left;
codec->write_mixer(codec, oss_mixer, left, right);
}
/* read or write the recmask, the ac97 can really have left and right
recording
inputs independantly set, but OSS doesn't seem to want us to express
that to
the user. the caller guarantees that we have a supported bit set, and
they
must be holding the card's spinlock */
static int ac97_recmask_io(struct ac97_codec *codec, int rw, int mask)
{
unsigned int val;
if (rw) {
/* read it from the card */
val = codec->codec_read(codec, AC97_RECORD_SELECT);
#ifdef DEBUG
printk("ac97_codec: ac97 recmask to set to 0x%04x\n",
val);
#endif
return (1 << ac97_rm2oss[val & 0x07]);
}
/* else, write the first set in the mask as the
output */
/* clear out current set value first (AC97 supports only 1 input!)
*/
val = (1 << ac97_rm2oss[codec->codec_read(codec,
AC97_RECORD_SELECT) & 0x07]);
if (mask != val)
mask &= ~val;
val = ffs(mask);
val = ac97_oss_rm[val-1];
val |= val << 8; /* set both channels */
#ifdef DEBUG
printk("ac97_codec: setting ac97 recmask to 0x%04x\n", val);
#endif
codec->codec_write(codec, AC97_RECORD_SELECT, val);
return 0;
};
static int ac97_mixer_ioctl(struct ac97_codec *codec, unsigned int cmd,
unsigned long arg)
{
int i, val = 0;
if (cmd == SOUND_MIXER_INFO) {
mixer_info info;
strncpy(info.id, codec->name, sizeof(info.id));
strncpy(info.name, codec->name, sizeof(info.name));
info.modify_counter = codec->modcnt;
if (copy_to_user((void *)arg, &info, sizeof(info)))
return -EFAULT;
return 0;
}
if (cmd == SOUND_OLD_MIXER_INFO) {
_old_mixer_info info;
strncpy(info.id, codec->name, sizeof(info.id));
strncpy(info.name, codec->name, sizeof(info.name));
if (copy_to_user((void *)arg, &info, sizeof(info)))
return -EFAULT;
return 0;
}
if (_IOC_TYPE(cmd) != 'M' || _SIOC_SIZE(cmd) != sizeof(int))
return -EINVAL;
if (cmd == OSS_GETVERSION)
return put_user(SOUND_VERSION, (int *)arg);
if (_SIOC_DIR(cmd) == _SIOC_READ) {
switch (_IOC_NR(cmd)) {
case SOUND_MIXER_RECSRC: /* give them the current record
source */
if (!codec->recmask_io) {
val = 0;
} else {
val = codec->recmask_io(codec, 1, 0);
}
break;
case SOUND_MIXER_DEVMASK: /* give them the supported
mixers */
val = codec->supported_mixers;
break;
case SOUND_MIXER_RECMASK: /* Arg contains a bit for each
supported recording source */
val = codec->record_sources;
break;
case SOUND_MIXER_STEREODEVS: /* Mixer channels supporting
stereo */
val = codec->stereo_mixers;
break;
case SOUND_MIXER_CAPS:
val = SOUND_CAP_EXCL_INPUT;
break;
default: /* read a specific mixer */
i = _IOC_NR(cmd);
if (!supported_mixer(codec, i))
return -EINVAL;
/* do we ever want to touch the hardware? */
/* val = codec->read_mixer(codec, i); */
val = codec->mixer_state[i];
break;
}
return put_user(val, (int *)arg);
}
if (_SIOC_DIR(cmd) == (_SIOC_WRITE|_SIOC_READ)) {
codec->modcnt++;
if (get_user(val, (int *)arg))
return -EFAULT;
switch (_IOC_NR(cmd)) {
case SOUND_MIXER_RECSRC: /* Arg contains a bit for each
recording source */
if (!codec->recmask_io) return -EINVAL;
if (!val) return 0;
if (!(val &= codec->record_sources)) return
-EINVAL;
codec->recmask_io(codec, 0, val);
return 0;
default: /* write a specific mixer */
i = _IOC_NR(cmd);
if (!supported_mixer(codec, i))
return -EINVAL;
ac97_set_mixer(codec, i, val);
return 0;
}
}
return -EINVAL;
}
/* entry point for /proc/driver/controller_vendor/ac97/%d */
int ac97_read_proc (char *page, char **start, off_t off,
int count, int *eof, void *data)
{
int len = 0, cap, extid, val, id1, id2;
struct ac97_codec *codec;
int is_ac97_20 = 0;
if ((codec = data) == NULL)
return -ENODEV;
id1 = codec->codec_read(codec, AC97_VENDOR_ID1);
id2 = codec->codec_read(codec, AC97_VENDOR_ID2);
len += sprintf (page+len, "Vendor name : %s\n", codec->name);
len += sprintf (page+len, "Vendor id : %04X %04X\n", id1,
id2);
extid = codec->codec_read(codec, AC97_EXTENDED_ID);
extid &= ~((1<<2)|(1<<4)|(1<<5)|(1<<10)|(1<<11)|(1<<12)|(1<<13));
len += sprintf (page+len, "AC97 Version : %s\n",
extid ? "2.0 or later" : "1.0");
if (extid) is_ac97_20 = 1;
cap = codec->codec_read(codec, AC97_RESET);
len += sprintf (page+len, "Capabilities :%s%s%s%s%s%s\n",
cap & 0x0001 ? " -dedicated MIC PCM IN channel-" :
"",
cap & 0x0002 ? " -reserved1-" : "",
cap & 0x0004 ? " -bass & treble-" : "",
cap & 0x0008 ? " -simulated stereo-" : "",
cap & 0x0010 ? " -headphone out-" : "",
cap & 0x0020 ? " -loudness-" : "");
val = cap & 0x00c0;
len += sprintf (page+len, "DAC resolutions :%s%s%s\n",
" -16-bit-",
val & 0x0040 ? " -18-bit-" : "",
val & 0x0080 ? " -20-bit-" : "");
val = cap & 0x0300;
len += sprintf (page+len, "ADC resolutions :%s%s%s\n",
" -16-bit-",
val & 0x0100 ? " -18-bit-" : "",
val & 0x0200 ? " -20-bit-" : "");
len += sprintf (page+len, "3D enhancement : %s\n",
ac97_stereo_enhancements[(cap >> 10) & 0x1f]);
val = codec->codec_read(codec, AC97_GENERAL_PURPOSE);
len += sprintf (page+len, "POP path : %s 3D\n"
"Sim. stereo : %s\n"
"3D enhancement : %s\n"
"Loudness : %s\n"
"Mono output : %s\n"
"MIC select : %s\n"
"ADC/DAC loopback : %s\n",
val & 0x8000 ? "post" : "pre",
val & 0x4000 ? "on" : "off",
val & 0x2000 ? "on" : "off",
val & 0x1000 ? "on" : "off",
val & 0x0200 ? "MIC" : "MIX",
val & 0x0100 ? "MIC2" : "MIC1",
val & 0x0080 ? "on" : "off");
extid = codec->codec_read(codec, AC97_EXTENDED_ID);
cap = extid;
len += sprintf (page+len, "Ext Capabilities :%s%s%s%s%s%s%s\n",
cap & 0x0001 ? " -var rate PCM audio-" : "",
cap & 0x0002 ? " -2x PCM audio out-" : "",
cap & 0x0008 ? " -var rate MIC in-" : "",
cap & 0x0040 ? " -PCM center DAC-" : "",
cap & 0x0080 ? " -PCM surround DAC-" : "",
cap & 0x0100 ? " -PCM LFE DAC-" : "",
cap & 0x0200 ? " -slot/DAC mappings-" : "");
if (is_ac97_20) {
len += sprintf (page+len, "Front DAC rate : %d\n",
codec->codec_read(codec,
AC97_PCM_FRONT_DAC_RATE));
}
return len;
}
/**
* ac97_probe_codec - Initialize and setup AC97-compatible codec
* @codec: (in/out) Kernel info for a single AC97 codec
*
* Reset the AC97 codec, then initialize the mixer and
* the rest of the @codec structure.
*
* The codec_read and codec_write fields of @codec are
* required to be setup and working when this function
* is called. All other fields are set by this function.
*
* codec_wait field of @codec can optionally be provided
* when calling this function. If codec_wait is not %NULL,
* this function will call codec_wait any time it is
* necessary to wait for the audio chip to reach the
* codec-ready state. If codec_wait is %NULL, then
* the default behavior is to call schedule_timeout.
* Currently codec_wait is used to wait for AC97 codec
* reset to complete.
*
* Returns 1 (true) on success, or 0 (false) on failure.
*/
int ac97_probe_codec(struct ac97_codec *codec)
{
u16 id1, id2;
u16 audio, modem;
int i;
/* probing AC97 codec, AC97 2.0 says that bit 15 of register 0x00
(reset) should
* be read zero.
*
* FIXME: is the following comment outdated? -jgarzik
* Probing of AC97 in this way is not reliable, it is not even
SAFE !!
*/
codec->codec_write(codec, AC97_RESET, 0L);
/* also according to spec, we wait for codec-ready state */
if (codec->codec_wait)
codec->codec_wait(codec);
else
udelay(10);
if ((audio = codec->codec_read(codec, AC97_RESET)) & 0x8000) {
printk(KERN_ERR "ac97_codec: %s ac97 codec not present\n",
codec->id ? "Secondary" : "Primary");
return 0;
}
/* probe for Modem Codec */
codec->codec_write(codec, AC97_EXTENDED_MODEM_ID, 0L);
modem = codec->codec_read(codec, AC97_EXTENDED_MODEM_ID);
codec->name = NULL;
codec->codec_ops = &null_ops;
id1 = codec->codec_read(codec, AC97_VENDOR_ID1);
id2 = codec->codec_read(codec, AC97_VENDOR_ID2);
for (i = 0; i < ARRAY_SIZE(ac97_codec_ids); i++) {
if (ac97_codec_ids[i].id == ((id1 << 16) | id2)) {
codec->type = ac97_codec_ids[i].id;
codec->name = ac97_codec_ids[i].name;
codec->codec_ops = ac97_codec_ids[i].ops;
break;
}
}
if (codec->name == NULL)
codec->name = "Unknown";
printk(KERN_INFO "ac97_codec: AC97 %s codec, id: 0x%04x:"
"0x%04x (%s)\n", audio ? "Audio" : (modem ? "Modem" : ""),
id1, id2, codec->name);
return ac97_init_mixer(codec);
}
static int ac97_init_mixer(struct ac97_codec *codec)
{
u16 cap;
int i;
cap = codec->codec_read(codec, AC97_RESET);
/* mixer masks */
codec->supported_mixers = AC97_SUPPORTED_MASK;
codec->stereo_mixers = AC97_STEREO_MASK;
codec->record_sources = AC97_RECORD_MASK;
if (!(cap & 0x04))
codec->supported_mixers &=
~(SOUND_MASK_BASS|SOUND_MASK_TREBLE);
if (!(cap & 0x10))
codec->supported_mixers &= ~SOUND_MASK_ALTPCM;
/* detect bit resolution */
codec->codec_write(codec, AC97_MASTER_VOL_STEREO, 0x2020);
if(codec->codec_read(codec, AC97_MASTER_VOL_STEREO) == 0x1f1f)
codec->bit_resolution = 5;
else
codec->bit_resolution = 6;
/* generic OSS to AC97 wrapper */
codec->read_mixer = ac97_read_mixer;
codec->write_mixer = ac97_write_mixer;
codec->recmask_io = ac97_recmask_io;
codec->mixer_ioctl = ac97_mixer_ioctl;
/* codec specific initialization for 4-6 channel output or
secondary codec stuff */
if (codec->codec_ops->init != NULL) {
codec->codec_ops->init(codec);
}
/* initialize mixer channel volumes */
for (i = 0; i < SOUND_MIXER_NRDEVICES; i++) {
struct mixer_defaults *md = &mixer_defaults[i];
if (md->mixer == -1)
break;
if (!supported_mixer(codec, md->mixer))
continue;
ac97_set_mixer(codec, md->mixer, md->value);
}
return 1;
}
#define AC97_SIGMATEL_ANALOG 0x6c /* Analog Special */
#define AC97_SIGMATEL_DAC2INVERT 0x6e
#define AC97_SIGMATEL_BIAS1 0x70
#define AC97_SIGMATEL_BIAS2 0x72
#define AC97_SIGMATEL_MULTICHN 0x74 /* Multi-Channel programming */
#define AC97_SIGMATEL_CIC1 0x76
#define AC97_SIGMATEL_CIC2 0x78
static int sigmatel_9708_init(struct ac97_codec * codec)
{
u16 codec72, codec6c;
codec72 = codec->codec_read(codec, AC97_SIGMATEL_BIAS2) & 0x8000;
codec6c = codec->codec_read(codec, AC97_SIGMATEL_ANALOG);
if ((codec72==0) && (codec6c==0)) {
codec->codec_write(codec, AC97_SIGMATEL_CIC1, 0xabba);
codec->codec_write(codec, AC97_SIGMATEL_CIC2, 0x1000);
codec->codec_write(codec, AC97_SIGMATEL_BIAS1, 0xabba);
codec->codec_write(codec, AC97_SIGMATEL_BIAS2, 0x0007);
} else if ((codec72==0x8000) && (codec6c==0)) {
codec->codec_write(codec, AC97_SIGMATEL_CIC1, 0xabba);
codec->codec_write(codec, AC97_SIGMATEL_CIC2, 0x1001);
codec->codec_write(codec, AC97_SIGMATEL_DAC2INVERT,
0x0008);
} else if ((codec72==0x8000) && (codec6c==0x0080)) {
/* nothing */
}
codec->codec_write(codec, AC97_SIGMATEL_MULTICHN, 0x0000);
return 0;
}
static int sigmatel_9721_init(struct ac97_codec * codec)
{
/* Only set up secondary codec */
if (codec->id == 0)
return 0;
codec->codec_write(codec, AC97_SURROUND_MASTER, 0L);
/* initialize SigmaTel STAC9721/23 as secondary codec, decoding AC
link
sloc 3,4 = 0x01, slot 7,8 = 0x00, */
codec->codec_write(codec, AC97_SIGMATEL_MULTICHN, 0x00);
/* we don't have the crystal when we are on an AMR card, so use
BIT_CLK as our clock source. Write the magic word ABBA and read
back to enable register 0x78 */
codec->codec_write(codec, AC97_SIGMATEL_CIC1, 0xabba);
codec->codec_read(codec, AC97_SIGMATEL_CIC1);
/* sync all the clocks*/
codec->codec_write(codec, AC97_SIGMATEL_CIC2, 0x3802);
return 0;
}
static int sigmatel_9744_init(struct ac97_codec * codec)
{
// patch for SigmaTel
codec->codec_write(codec, AC97_SIGMATEL_CIC1, 0xabba);
codec->codec_write(codec, AC97_SIGMATEL_CIC2, 0x0000); // is this
correct? --jk
codec->codec_write(codec, AC97_SIGMATEL_BIAS1, 0xabba);
codec->codec_write(codec, AC97_SIGMATEL_BIAS2, 0x0002);
codec->codec_write(codec, AC97_SIGMATEL_MULTICHN, 0x0000);
return 0;
}
static int wolfson_init(struct ac97_codec * codec)
{
codec->codec_write(codec, 0x72, 0x0808);
codec->codec_write(codec, 0x74, 0x0808);
// patch for DVD noise
codec->codec_write(codec, 0x5a, 0x0200);
// init vol as PCM vol
codec->codec_write(codec, 0x70,
codec->codec_read(codec, AC97_PCMOUT_VOL));
codec->codec_write(codec, AC97_SURROUND_MASTER, 0x0000);
return 0;
}
static int tritech_init(struct ac97_codec * codec)
{
codec->codec_write(codec, 0x26, 0x0300);
codec->codec_write(codec, 0x26, 0x0000);
codec->codec_write(codec, AC97_SURROUND_MASTER, 0x0000);
codec->codec_write(codec, AC97_RESERVED_3A, 0x0000);
return 0;
}
/* copied from drivers/sound/maestro.c */
static int tritech_maestro_init(struct ac97_codec * codec)
{
/* no idea what this does */
codec->codec_write(codec, 0x2A, 0x0001);
codec->codec_write(codec, 0x2C, 0x0000);
codec->codec_write(codec, 0x2C, 0XFFFF);
return 0;
}
/*
* This is basically standard AC97. It should work as a default for
* almost all modern codecs. Note that some cards wire EAPD
*backwards*
* That side of it is up to the card driver not us to cope with.
*
*/
static int eapd_control(struct ac97_codec * codec, int on)
{
if(on)
codec->codec_write(codec, AC97_POWER_CONTROL,
codec->codec_read(codec,
AC97_POWER_CONTROL)|0x8000);
else
codec->codec_write(codec, AC97_POWER_CONTROL,
codec->codec_read(codec,
AC97_POWER_CONTROL)&~0x8000);
return 0;
}
static int eapd_on_control(struct ac97_codec * codec)
{
eapd_control(codec,0);
return 0;
}
static int eapd_off_control(struct ac97_codec * codec)
{
eapd_control(codec,1);
return 0;
}
/*
* Crystal digital audio control (CS4299
*/
static int crystal_digital_control(struct ac97_codec *codec, int mode)
{
u16 cv;
switch(mode)
{
case 0: cv = 0x0; break; /* SPEN off */
case 1: cv = 0x8004; break; /* 48KHz digital */
case 2: cv = 0x8104; break; /* 44.1KHz digital */
default:
return -1; /* Not supported yet(eg
AC3) */
}
codec->codec_write(codec, 0x68, cv);
return 0;
}
/* copied from drivers/sound/maestro.c */
#if 0 /* there has been 1 person on the planet with a pt101 that we
know of. If they care, they can put this back in :) */
static int pt101_init(struct ac97_codec * codec)
{
printk(KERN_INFO "ac97_codec: PT101 Codec detected, initializing
but _not_ installing mixer device.\n");
/* who knows.. */
codec->codec_write(codec, 0x2A, 0x0001);
codec->codec_write(codec, 0x2C, 0x0000);
codec->codec_write(codec, 0x2C, 0xFFFF);
codec->codec_write(codec, 0x10, 0x9F1F);
codec->codec_write(codec, 0x12, 0x0808);
codec->codec_write(codec, 0x14, 0x9F1F);
codec->codec_write(codec, 0x16, 0x9F1F);
codec->codec_write(codec, 0x18, 0x0404);
codec->codec_write(codec, 0x1A, 0x0000);
codec->codec_write(codec, 0x1C, 0x0000);
codec->codec_write(codec, 0x02, 0x0404);
codec->codec_write(codec, 0x04, 0x0808);
codec->codec_write(codec, 0x0C, 0x801F);
codec->codec_write(codec, 0x0E, 0x801F);
return 0;
}
#endif
EXPORT_SYMBOL(ac97_read_proc);
EXPORT_SYMBOL(ac97_probe_codec);
/*
* AC97 library support routines
*/
/**
* ac97_set_dac_rate - set codec rate adaption
* @codec: ac97 code
* @rate: rate in hertz
*
* Set the DAC rate. Assumes the codec supports VRA. The caller is
* expected to have checked this little detail.
*/
unsigned int ac97_set_dac_rate(struct ac97_codec *codec, unsigned int
rate)
{
unsigned int new_rate = rate;
u32 dacp;
u32 mast_vol, phone_vol, mono_vol, pcm_vol;
u32 mute_vol = 0x8000; /* The mute volume? */
if(rate != codec->codec_read(codec, AC97_PCM_FRONT_DAC_RATE))
{
/* Mute several registers */
mast_vol = codec->codec_read(codec,
AC97_MASTER_VOL_STEREO);
mono_vol = codec->codec_read(codec, AC97_MASTER_VOL_MONO);
phone_vol = codec->codec_read(codec, AC97_HEADPHONE_VOL);
pcm_vol = codec->codec_read(codec, AC97_PCMOUT_VOL);
codec->codec_write(codec, AC97_MASTER_VOL_STEREO,
mute_vol);
codec->codec_write(codec, AC97_MASTER_VOL_MONO, mute_vol);
codec->codec_write(codec, AC97_HEADPHONE_VOL, mute_vol);
codec->codec_write(codec, AC97_PCMOUT_VOL, mute_vol);
/* Power down the DAC */
dacp=codec->codec_read(codec, AC97_POWER_CONTROL);
codec->codec_write(codec, AC97_POWER_CONTROL,
dacp|0x0200);
/* Load the rate and read the effective rate */
codec->codec_write(codec, AC97_PCM_FRONT_DAC_RATE, rate);
new_rate=codec->codec_read(codec,
AC97_PCM_FRONT_DAC_RATE);
/* Power it back up */
codec->codec_write(codec, AC97_POWER_CONTROL, dacp);
/* Restore volumes */
codec->codec_write(codec, AC97_MASTER_VOL_STEREO,
mast_vol);
codec->codec_write(codec, AC97_MASTER_VOL_MONO, mono_vol);
codec->codec_write(codec, AC97_HEADPHONE_VOL, phone_vol);
codec->codec_write(codec, AC97_PCMOUT_VOL, pcm_vol);
}
return new_rate;
}
EXPORT_SYMBOL(ac97_set_dac_rate);
/**
* ac97_set_adc_rate - set codec rate adaption
* @codec: ac97 code
* @rate: rate in hertz
*
* Set the ADC rate. Assumes the codec supports VRA. The caller is
* expected to have checked this little detail.
*/
unsigned int ac97_set_adc_rate(struct ac97_codec *codec, unsigned int
rate)
{
unsigned int new_rate = rate;
u32 dacp;
if(rate != codec->codec_read(codec, AC97_PCM_LR_ADC_RATE))
{
/* Power down the ADC */
dacp=codec->codec_read(codec, AC97_POWER_CONTROL);
codec->codec_write(codec, AC97_POWER_CONTROL,
dacp|0x0100);
/* Load the rate and read the effective rate */
codec->codec_write(codec, AC97_PCM_LR_ADC_RATE, rate);
new_rate=codec->codec_read(codec, AC97_PCM_LR_ADC_RATE);
/* Power it back up */
codec->codec_write(codec, AC97_POWER_CONTROL, dacp);
}
return new_rate;
}
EXPORT_SYMBOL(ac97_set_adc_rate);
int ac97_save_state(struct ac97_codec *codec)
{
return 0;
}
EXPORT_SYMBOL(ac97_save_state);
int ac97_restore_state(struct ac97_codec *codec)
{
int i;
unsigned int left, right, val;
for (i = 0; i < SOUND_MIXER_NRDEVICES; i++) {
if (!supported_mixer(codec, i))
continue;
val = codec->mixer_state[i];
right = val >> 8;
left = val & 0xff;
codec->write_mixer(codec, i, left, right);
}
return 0;
}
EXPORT_SYMBOL(ac97_restore_state);
MODULE_LICENSE("GPL");
^ permalink raw reply [flat|nested] 8+ messages in thread* Re: via686/AD1886/Soundmax drivers
2002-02-27 16:54 Peter Christy
@ 2002-02-27 19:11 ` André Dahlqvist
2002-02-27 22:12 ` James Cassidy
0 siblings, 1 reply; 8+ messages in thread
From: André Dahlqvist @ 2002-02-27 19:11 UTC (permalink / raw)
To: linux-kernel
On Wed, Feb 27, 2002 at 04:54:12PM +0000, Peter Christy wrote:
> Ths saga continues..... (no sound on laptops using via686 and AD1886s -
> mostly FIC A360+ and Compaq Presarios)
This is interesting. Have you given the ALSA drivers a shot? Do you know
what Compaq Presario laptops this affects?
--
André Dahlqvist <andre.dahlqvist@telia.com>
^ permalink raw reply [flat|nested] 8+ messages in thread
* Re: via686/AD1886/Soundmax drivers
2002-02-27 19:11 ` André Dahlqvist
@ 2002-02-27 22:12 ` James Cassidy
0 siblings, 0 replies; 8+ messages in thread
From: James Cassidy @ 2002-02-27 22:12 UTC (permalink / raw)
To: linux-kernel
[-- Attachment #1: Type: text/plain, Size: 984 bytes --]
This affects Compaq's new Compaq Presario 700's that were released
last summer, which has been discussed on this list several times. Peter
has a FIC A360+ board which he has managed to atleast play peripherals like
CD-ROM to work, this is one the HP laptops I believe.
I haven't been able to do that, with any of the default_ops or
eapd settings suggested on this list, on the Compaq presario I'm sending
this e-mail from. Only that frustrating crosstalk sound I get by turning
the peripheral volumes all the way up. The master volume does nothing.
-- James Cassidy (QFire)
On Wed, Feb 27, 2002 at 08:11:48PM +0100, Andr? Dahlqvist wrote:
> On Wed, Feb 27, 2002 at 04:54:12PM +0000, Peter Christy wrote:
>
> > Ths saga continues..... (no sound on laptops using via686 and AD1886s -
> > mostly FIC A360+ and Compaq Presarios)
>
> This is interesting. Have you given the ALSA drivers a shot? Do you know
> what Compaq Presario laptops this affects?
[-- Attachment #2: Type: application/pgp-signature, Size: 232 bytes --]
^ permalink raw reply [flat|nested] 8+ messages in thread
end of thread, other threads:[~2002-02-27 22:13 UTC | newest]
Thread overview: 8+ messages (download: mbox.gz follow: Atom feed
-- links below jump to the message on this page --
2002-02-19 9:03 via686/AD1886/Soundmax drivers Peter Christy
2002-02-19 12:25 ` Vojtech Pavlik
2002-02-21 9:56 ` Peter Christy
2002-02-21 10:15 ` Vojtech Pavlik
2002-02-21 11:16 ` Peter Christy
-- strict thread matches above, loose matches on Subject: below --
2002-02-27 16:54 Peter Christy
2002-02-27 19:11 ` André Dahlqvist
2002-02-27 22:12 ` James Cassidy
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