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authorLinus Torvalds <torvalds@ppc970.osdl.org>2005-04-16 15:20:36 -0700
committerLinus Torvalds <torvalds@ppc970.osdl.org>2005-04-16 15:20:36 -0700
commit1da177e4c3f41524e886b7f1b8a0c1fc7321cac2 (patch)
tree0bba044c4ce775e45a88a51686b5d9f90697ea9d /sound/oss/ac97_codec.c
Linux-2.6.12-rc2
Initial git repository build. I'm not bothering with the full history, even though we have it. We can create a separate "historical" git archive of that later if we want to, and in the meantime it's about 3.2GB when imported into git - space that would just make the early git days unnecessarily complicated, when we don't have a lot of good infrastructure for it. Let it rip!
Diffstat (limited to 'sound/oss/ac97_codec.c')
-rw-r--r--sound/oss/ac97_codec.c1576
1 files changed, 1576 insertions, 0 deletions
diff --git a/sound/oss/ac97_codec.c b/sound/oss/ac97_codec.c
new file mode 100644
index 000000000000..124b1e10a13d
--- /dev/null
+++ b/sound/oss/ac97_codec.c
@@ -0,0 +1,1576 @@
+/*
+ * 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
+ * May 02, 2003 Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ * Removed non existant WM9700
+ * Added support for WM9705, WM9708, WM9709, WM9710, WM9711
+ * WM9712 and WM9717
+ * Mar 28, 2002 Randolph Bentson <bentson@holmsjoen.com>
+ * corrections to support WM9707 in ViewPad 1000
+ * 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/kernel.h>
+#include <linux/slab.h>
+#include <linux/string.h>
+#include <linux/errno.h>
+#include <linux/bitops.h>
+#include <linux/delay.h>
+#include <linux/pci.h>
+#include <linux/ac97_codec.h>
+#include <asm/uaccess.h>
+
+#define CODEC_ID_BUFSZ 14
+
+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_init03(struct ac97_codec * codec);
+static int wolfson_init04(struct ac97_codec * codec);
+static int wolfson_init05(struct ac97_codec * codec);
+static int wolfson_init11(struct ac97_codec * codec);
+static int wolfson_init13(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 ad1886_init(struct ac97_codec *codec);
+static int eapd_control(struct ac97_codec *codec, int);
+static int crystal_digital_control(struct ac97_codec *codec, int slots, int rate, int mode);
+static int cmedia_init(struct ac97_codec * codec);
+static int cmedia_digital_control(struct ac97_codec *codec, int slots, int rate, int mode);
+static int generic_digital_control(struct ac97_codec *codec, int slots, int rate, 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 default_digital_ops = { NULL, eapd_control, generic_digital_control};
+static struct ac97_ops wolfson_ops03 = { wolfson_init03, NULL, NULL };
+static struct ac97_ops wolfson_ops04 = { wolfson_init04, NULL, NULL };
+static struct ac97_ops wolfson_ops05 = { wolfson_init05, NULL, NULL };
+static struct ac97_ops wolfson_ops11 = { wolfson_init11, NULL, NULL };
+static struct ac97_ops wolfson_ops13 = { wolfson_init13, 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 };
+static struct ac97_ops ad1886_ops = { ad1886_init, eapd_control, NULL };
+static struct ac97_ops cmedia_ops = { NULL, eapd_control, NULL};
+static struct ac97_ops cmedia_digital_ops = { cmedia_init, eapd_control, cmedia_digital_control};
+
+/* sorted by vendor/device id */
+static const struct {
+ u32 id;
+ char *name;
+ struct ac97_ops *ops;
+ int flags;
+} 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},
+ {0x41445361, "Analog Devices AD1886", &ad1886_ops},
+ {0x41445370, "Analog Devices AD1981", &null_ops},
+ {0x41445372, "Analog Devices AD1981A", &null_ops},
+ {0x41445374, "Analog Devices AD1981B", &null_ops},
+ {0x41445460, "Analog Devices AD1885", &default_ops},
+ {0x41445461, "Analog Devices AD1886", &ad1886_ops},
+ {0x414B4D00, "Asahi Kasei AK4540", &null_ops},
+ {0x414B4D01, "Asahi Kasei AK4542", &null_ops},
+ {0x414B4D02, "Asahi Kasei AK4543", &null_ops},
+ {0x414C4326, "ALC100P", &null_ops},
+ {0x414C4710, "ALC200/200P", &null_ops},
+ {0x414C4720, "ALC650", &default_digital_ops},
+ {0x434D4941, "CMedia", &cmedia_ops, AC97_NO_PCM_VOLUME },
+ {0x434D4942, "CMedia", &cmedia_ops, AC97_NO_PCM_VOLUME },
+ {0x434D4961, "CMedia", &cmedia_digital_ops, AC97_NO_PCM_VOLUME },
+ {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},
+ {0x43585442, "CXT66", &default_ops, AC97_DELUDED_MODEM },
+ {0x44543031, "Diamond Technology DT0893", &default_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},
+ {0x574D4C03, "Wolfson WM9703/07/08/17", &wolfson_ops03},
+ {0x574D4C04, "Wolfson WM9704M/WM9704Q", &wolfson_ops04},
+ {0x574D4C05, "Wolfson WM9705/WM9710", &wolfson_ops05},
+ {0x574D4C09, "Wolfson WM9709", &null_ops},
+ {0x574D4C12, "Wolfson WM9711/9712", &wolfson_ops11},
+ {0x574D4C13, "Wolfson WM9713", &wolfson_ops13, AC97_DEFAULT_POWER_OFF},
+ {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},
+ {0x83847652, "SigmaTel STAC9752/53", &default_ops},
+ {0x83847656, "SigmaTel STAC9756/57", &sigmatel_9744_ops},
+ {0x83847666, "SigmaTel STAC9750T", &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
+};
+
+static LIST_HEAD(codecs);
+static LIST_HEAD(codec_drivers);
+static DECLARE_MUTEX(codec_sem);
+
+/* 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;
+ memset(&info, 0, sizeof(info));
+ strlcpy(info.id, codec->name, sizeof(info.id));
+ strlcpy(info.name, codec->name, sizeof(info.name));
+ info.modify_counter = codec->modcnt;
+ if (copy_to_user((void __user *)arg, &info, sizeof(info)))
+ return -EFAULT;
+ return 0;
+ }
+ if (cmd == SOUND_OLD_MIXER_INFO) {
+ _old_mixer_info info;
+ memset(&info, 0, sizeof(info));
+ strlcpy(info.id, codec->name, sizeof(info.id));
+ strlcpy(info.name, codec->name, sizeof(info.name));
+ if (copy_to_user((void __user *)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 __user *)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 __user *)arg);
+ }
+
+ if (_SIOC_DIR(cmd) == (_SIOC_WRITE|_SIOC_READ)) {
+ codec->modcnt++;
+ if (get_user(val, (int __user *)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;
+}
+
+/**
+ * codec_id - Turn id1/id2 into a PnP string
+ * @id1: Vendor ID1
+ * @id2: Vendor ID2
+ * @buf: CODEC_ID_BUFSZ byte buffer
+ *
+ * Fills buf with a zero terminated PnP ident string for the id1/id2
+ * pair. For convenience the return is the passed in buffer pointer.
+ */
+
+static char *codec_id(u16 id1, u16 id2, char *buf)
+{
+ if(id1&0x8080) {
+ snprintf(buf, CODEC_ID_BUFSZ, "0x%04x:0x%04x", id1, id2);
+ } else {
+ buf[0] = (id1 >> 8);
+ buf[1] = (id1 & 0xFF);
+ buf[2] = (id2 >> 8);
+ snprintf(buf+3, CODEC_ID_BUFSZ - 3, "%d", id2&0xFF);
+ }
+ return buf;
+}
+
+/**
+ * ac97_check_modem - Check if the Codec is a modem
+ * @codec: codec to check
+ *
+ * Return true if the device is an AC97 1.0 or AC97 2.0 modem
+ */
+
+static int ac97_check_modem(struct ac97_codec *codec)
+{
+ /* Check for an AC97 1.0 soft modem (ID1) */
+ if(codec->codec_read(codec, AC97_RESET) & 2)
+ return 1;
+ /* Check for an AC97 2.x soft modem */
+ codec->codec_write(codec, AC97_EXTENDED_MODEM_ID, 0L);
+ if(codec->codec_read(codec, AC97_EXTENDED_MODEM_ID) & 1)
+ return 1;
+ return 0;
+}
+
+
+/**
+ * ac97_alloc_codec - Allocate an AC97 codec
+ *
+ * Returns a new AC97 codec structure. AC97 codecs may become
+ * refcounted soon so this interface is needed. Returns with
+ * one reference taken.
+ */
+
+struct ac97_codec *ac97_alloc_codec(void)
+{
+ struct ac97_codec *codec = kmalloc(sizeof(struct ac97_codec), GFP_KERNEL);
+ if(!codec)
+ return NULL;
+
+ memset(codec, 0, sizeof(*codec));
+ spin_lock_init(&codec->lock);
+ INIT_LIST_HEAD(&codec->list);
+ return codec;
+}
+
+EXPORT_SYMBOL(ac97_alloc_codec);
+
+/**
+ * ac97_release_codec - Release an AC97 codec
+ * @codec: codec to release
+ *
+ * Release an allocated AC97 codec. This will be refcounted in
+ * time but for the moment is trivial. Calls the unregister
+ * handler if the codec is now defunct.
+ */
+
+void ac97_release_codec(struct ac97_codec *codec)
+{
+ /* Remove from the list first, we don't want to be
+ "rediscovered" */
+ down(&codec_sem);
+ list_del(&codec->list);
+ up(&codec_sem);
+ /*
+ * The driver needs to deal with internal
+ * locking to avoid accidents here.
+ */
+ if(codec->driver)
+ codec->driver->remove(codec, codec->driver);
+ kfree(codec);
+}
+
+EXPORT_SYMBOL(ac97_release_codec);
+
+/**
+ * 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.
+ *
+ * Some codecs will power down when a register reset is
+ * performed. We now check for such codecs.
+ *
+ * Returns 1 (true) on success, or 0 (false) on failure.
+ */
+
+int ac97_probe_codec(struct ac97_codec *codec)
+{
+ u16 id1, id2;
+ u16 audio;
+ int i;
+ char cidbuf[CODEC_ID_BUFSZ];
+ u16 f;
+ struct list_head *l;
+ struct ac97_driver *d;
+
+ /* wait for codec-ready state */
+ if (codec->codec_wait)
+ codec->codec_wait(codec);
+ else
+ udelay(10);
+
+ /* will the codec power down if register reset ? */
+ id1 = codec->codec_read(codec, AC97_VENDOR_ID1);
+ id2 = codec->codec_read(codec, AC97_VENDOR_ID2);
+ codec->name = NULL;
+ codec->codec_ops = &null_ops;
+ 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;
+ codec->flags = ac97_codec_ids[i].flags;
+ break;
+ }
+ }
+
+ codec->model = (id1 << 16) | id2;
+ if ((codec->flags & AC97_DEFAULT_POWER_OFF) == 0) {
+ /* reset codec and wait for the ready bit before we continue */
+ codec->codec_write(codec, AC97_RESET, 0L);
+ if (codec->codec_wait)
+ codec->codec_wait(codec);
+ else
+ udelay(10);
+ }
+
+ /* 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 !!
+ */
+ if ((audio = codec->codec_read(codec, AC97_RESET)) & 0x8000) {
+ printk(KERN_ERR "ac97_codec: %s ac97 codec not present\n",
+ (codec->id & 0x2) ? (codec->id&1 ? "4th" : "Tertiary")
+ : (codec->id&1 ? "Secondary": "Primary"));
+ return 0;
+ }
+
+ /* probe for Modem Codec */
+ codec->modem = ac97_check_modem(codec);
+
+ /* enable SPDIF */
+ f = codec->codec_read(codec, AC97_EXTENDED_STATUS);
+ if((codec->codec_ops == &null_ops) && (f & 4))
+ codec->codec_ops = &default_digital_ops;
+
+ /* A device which thinks its a modem but isnt */
+ if(codec->flags & AC97_DELUDED_MODEM)
+ codec->modem = 0;
+
+ if (codec->name == NULL)
+ codec->name = "Unknown";
+ printk(KERN_INFO "ac97_codec: AC97 %s codec, id: %s (%s)\n",
+ codec->modem ? "Modem" : (audio ? "Audio" : ""),
+ codec_id(id1, id2, cidbuf), codec->name);
+
+ if(!ac97_init_mixer(codec))
+ return 0;
+
+ /*
+ * Attach last so the caller can override the mixer
+ * callbacks.
+ */
+
+ down(&codec_sem);
+ list_add(&codec->list, &codecs);
+
+ list_for_each(l, &codec_drivers) {
+ d = list_entry(l, struct ac97_driver, list);
+ if ((codec->model ^ d->codec_id) & d->codec_mask)
+ continue;
+ if(d->probe(codec, d) == 0)
+ {
+ codec->driver = d;
+ break;
+ }
+ }
+
+ up(&codec_sem);
+ return 1;
+}
+
+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) == 0x2020)
+ codec->bit_resolution = 6;
+ else
+ codec->bit_resolution = 5;
+
+ /* 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;
+
+ /* 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);
+ }
+
+ /* codec specific initialization for 4-6 channel output or secondary codec stuff */
+ if (codec->codec_ops->init != NULL) {
+ codec->codec_ops->init(codec);
+ }
+
+ /*
+ * Volume is MUTE only on this device. We have to initialise
+ * it but its useless beyond that.
+ */
+ if(codec->flags & AC97_NO_PCM_VOLUME)
+ {
+ codec->supported_mixers &= ~SOUND_MASK_PCM;
+ printk(KERN_WARNING "AC97 codec does not have proper volume support.\n");
+ }
+ 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 cmedia_init(struct ac97_codec *codec)
+{
+ /* Initialise the CMedia 9739 */
+ /*
+ We could set various options here
+ Register 0x20 bit 0x100 sets mic as center bass
+ Also do multi_channel_ctrl &=~0x3000 |=0x1000
+
+ For now we set up the GPIO and PC beep
+ */
+
+ u16 v;
+
+ /* MIC */
+ codec->codec_write(codec, 0x64, 0x3000);
+ v = codec->codec_read(codec, 0x64);
+ v &= ~0x8000;
+ codec->codec_write(codec, 0x64, v);
+ codec->codec_write(codec, 0x70, 0x0100);
+ codec->codec_write(codec, 0x72, 0x0020);
+ return 0;
+}
+
+#define AC97_WM97XX_FMIXER_VOL 0x72
+#define AC97_WM97XX_RMIXER_VOL 0x74
+#define AC97_WM97XX_TEST 0x5a
+#define AC97_WM9704_RPCM_VOL 0x70
+#define AC97_WM9711_OUT3VOL 0x16
+
+static int wolfson_init03(struct ac97_codec * codec)
+{
+ /* this is known to work for the ViewSonic ViewPad 1000 */
+ codec->codec_write(codec, AC97_WM97XX_FMIXER_VOL, 0x0808);
+ codec->codec_write(codec, AC97_GENERAL_PURPOSE, 0x8000);
+ return 0;
+}
+
+static int wolfson_init04(struct ac97_codec * codec)
+{
+ codec->codec_write(codec, AC97_WM97XX_FMIXER_VOL, 0x0808);
+ codec->codec_write(codec, AC97_WM97XX_RMIXER_VOL, 0x0808);
+
+ // patch for DVD noise
+ codec->codec_write(codec, AC97_WM97XX_TEST, 0x0200);
+
+ // init vol as PCM vol
+ codec->codec_write(codec, AC97_WM9704_RPCM_VOL,
+ codec->codec_read(codec, AC97_PCMOUT_VOL));
+
+ /* set rear surround volume */
+ codec->codec_write(codec, AC97_SURROUND_MASTER, 0x0000);
+ return 0;
+}
+
+/* WM9705, WM9710 */
+static int wolfson_init05(struct ac97_codec * codec)
+{
+ /* set front mixer volume */
+ codec->codec_write(codec, AC97_WM97XX_FMIXER_VOL, 0x0808);
+ return 0;
+}
+
+/* WM9711, WM9712 */
+static int wolfson_init11(struct ac97_codec * codec)
+{
+ /* stop pop's during suspend/resume */
+ codec->codec_write(codec, AC97_WM97XX_TEST,
+ codec->codec_read(codec, AC97_WM97XX_TEST) & 0xffbf);
+
+ /* set out3 volume */
+ codec->codec_write(codec, AC97_WM9711_OUT3VOL, 0x0808);
+ return 0;
+}
+
+/* WM9713 */
+static int wolfson_init13(struct ac97_codec * codec)
+{
+ codec->codec_write(codec, AC97_RECORD_GAIN, 0x00a0);
+ codec->codec_write(codec, AC97_POWER_CONTROL, 0x0000);
+ codec->codec_write(codec, AC97_EXTENDED_MODEM_ID, 0xDA00);
+ codec->codec_write(codec, AC97_EXTEND_MODEM_STAT, 0x3810);
+ codec->codec_write(codec, AC97_PHONE_VOL, 0x0808);
+ codec->codec_write(codec, AC97_PCBEEP_VOL, 0x0808);
+
+ 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;
+}
+
+
+
+/*
+ * Presario700 workaround
+ * for Jack Sense/SPDIF Register mis-setting causing
+ * no audible output
+ * by Santiago Nullo 04/05/2002
+ */
+
+#define AC97_AD1886_JACK_SENSE 0x72
+
+static int ad1886_init(struct ac97_codec * codec)
+{
+ /* from AD1886 Specs */
+ codec->codec_write(codec, AC97_AD1886_JACK_SENSE, 0x0010);
+ 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 generic_digital_control(struct ac97_codec *codec, int slots, int rate, int mode)
+{
+ u16 reg;
+
+ reg = codec->codec_read(codec, AC97_SPDIF_CONTROL);
+
+ switch(rate)
+ {
+ /* Off by default */
+ default:
+ case 0:
+ reg = codec->codec_read(codec, AC97_EXTENDED_STATUS);
+ codec->codec_write(codec, AC97_EXTENDED_STATUS, (reg & ~AC97_EA_SPDIF));
+ if(rate == 0)
+ return 0;
+ return -EINVAL;
+ case 1:
+ reg = (reg & AC97_SC_SPSR_MASK) | AC97_SC_SPSR_48K;
+ break;
+ case 2:
+ reg = (reg & AC97_SC_SPSR_MASK) | AC97_SC_SPSR_44K;
+ break;
+ case 3:
+ reg = (reg & AC97_SC_SPSR_MASK) | AC97_SC_SPSR_32K;
+ break;
+ }
+
+ reg &= ~AC97_SC_CC_MASK;
+ reg |= (mode & AUDIO_CCMASK) << 6;
+
+ if(mode & AUDIO_DIGITAL)
+ reg |= 2;
+ if(mode & AUDIO_PRO)
+ reg |= 1;
+ if(mode & AUDIO_DRS)
+ reg |= 0x4000;
+
+ codec->codec_write(codec, AC97_SPDIF_CONTROL, reg);
+
+ reg = codec->codec_read(codec, AC97_EXTENDED_STATUS);
+ reg &= (AC97_EA_SLOT_MASK);
+ reg |= AC97_EA_VRA | AC97_EA_SPDIF | slots;
+ codec->codec_write(codec, AC97_EXTENDED_STATUS, reg);
+
+ reg = codec->codec_read(codec, AC97_EXTENDED_STATUS);
+ if(!(reg & 0x0400))
+ {
+ codec->codec_write(codec, AC97_EXTENDED_STATUS, reg & ~ AC97_EA_SPDIF);
+ return -EINVAL;
+ }
+ return 0;
+}
+
+/*
+ * Crystal digital audio control (CS4299)
+ */
+
+static int crystal_digital_control(struct ac97_codec *codec, int slots, int rate, int mode)
+{
+ u16 cv;
+
+ if(mode & AUDIO_DIGITAL)
+ return -EINVAL;
+
+ switch(rate)
+ {
+ case 0: cv = 0x0; break; /* SPEN off */
+ case 48000: cv = 0x8004; break; /* 48KHz digital */
+ case 44100: cv = 0x8104; break; /* 44.1KHz digital */
+ case 32768: /* 32Khz */
+ default:
+ return -EINVAL;
+ }
+ codec->codec_write(codec, 0x68, cv);
+ return 0;
+}
+
+/*
+ * CMedia digital audio control
+ * Needs more work.
+ */
+
+static int cmedia_digital_control(struct ac97_codec *codec, int slots, int rate, int mode)
+{
+ u16 cv;
+
+ if(mode & AUDIO_DIGITAL)
+ return -EINVAL;
+
+ switch(rate)
+ {
+ case 0: cv = 0x0001; break; /* SPEN off */
+ case 48000: cv = 0x0009; break; /* 48KHz digital */
+ default:
+ return -EINVAL;
+ }
+ codec->codec_write(codec, 0x2A, 0x05c4);
+ codec->codec_write(codec, 0x6C, cv);
+
+ /* Switch on mix to surround */
+ cv = codec->codec_read(codec, 0x64);
+ cv &= ~0x0200;
+ if(mode)
+ cv |= 0x0200;
+ codec->codec_write(codec, 0x64, 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);
+
+/**
+ * ac97_register_driver - register a codec helper
+ * @driver: Driver handler
+ *
+ * Register a handler for codecs matching the codec id. The handler
+ * attach function is called for all present codecs and will be
+ * called when new codecs are discovered.
+ */
+
+int ac97_register_driver(struct ac97_driver *driver)
+{
+ struct list_head *l;
+ struct ac97_codec *c;
+
+ down(&codec_sem);
+ INIT_LIST_HEAD(&driver->list);
+ list_add(&driver->list, &codec_drivers);
+
+ list_for_each(l, &codecs)
+ {
+ c = list_entry(l, struct ac97_codec, list);
+ if(c->driver != NULL || ((c->model ^ driver->codec_id) & driver->codec_mask))
+ continue;
+ if(driver->probe(c, driver))
+ continue;
+ c->driver = driver;
+ }
+ up(&codec_sem);
+ return 0;
+}
+
+EXPORT_SYMBOL_GPL(ac97_register_driver);
+
+/**
+ * ac97_unregister_driver - unregister a codec helper
+ * @driver: Driver handler
+ *
+ * Unregister a handler for codecs matching the codec id. The handler
+ * remove function is called for all matching codecs.
+ */
+
+void ac97_unregister_driver(struct ac97_driver *driver)
+{
+ struct list_head *l;
+ struct ac97_codec *c;
+
+ down(&codec_sem);
+ list_del_init(&driver->list);
+
+ list_for_each(l, &codecs)
+ {
+ c = list_entry(l, struct ac97_codec, list);
+ if (c->driver == driver) {
+ driver->remove(c, driver);
+ c->driver = NULL;
+ }
+ }
+
+ up(&codec_sem);
+}
+
+EXPORT_SYMBOL_GPL(ac97_unregister_driver);
+
+static int swap_headphone(int remove_master)
+{
+ struct list_head *l;
+ struct ac97_codec *c;
+
+ if (remove_master) {
+ down(&codec_sem);
+ list_for_each(l, &codecs)
+ {
+ c = list_entry(l, struct ac97_codec, list);
+ if (supported_mixer(c, SOUND_MIXER_PHONEOUT))
+ c->supported_mixers &= ~SOUND_MASK_PHONEOUT;
+ }
+ up(&codec_sem);
+ } else
+ ac97_hw[SOUND_MIXER_PHONEOUT].offset = AC97_MASTER_VOL_STEREO;
+
+ /* Scale values already match */
+ ac97_hw[SOUND_MIXER_VOLUME].offset = AC97_MASTER_VOL_MONO;
+ return 0;
+}
+
+static int apply_quirk(int quirk)
+{
+ switch (quirk) {
+ case AC97_TUNE_NONE:
+ return 0;
+ case AC97_TUNE_HP_ONLY:
+ return swap_headphone(1);
+ case AC97_TUNE_SWAP_HP:
+ return swap_headphone(0);
+ case AC97_TUNE_SWAP_SURROUND:
+ return -ENOSYS; /* not yet implemented */
+ case AC97_TUNE_AD_SHARING:
+ return -ENOSYS; /* not yet implemented */
+ case AC97_TUNE_ALC_JACK:
+ return -ENOSYS; /* not yet implemented */
+ }
+ return -EINVAL;
+}
+
+/**
+ * ac97_tune_hardware - tune up the hardware
+ * @pdev: pci_dev pointer
+ * @quirk: quirk list
+ * @override: explicit quirk value (overrides if not AC97_TUNE_DEFAULT)
+ *
+ * Do some workaround for each pci device, such as renaming of the
+ * headphone (true line-out) control as "Master".
+ * The quirk-list must be terminated with a zero-filled entry.
+ *
+ * Returns zero if successful, or a negative error code on failure.
+ */
+
+int ac97_tune_hardware(struct pci_dev *pdev, struct ac97_quirk *quirk, int override)
+{
+ int result;
+
+ if (!quirk)
+ return -EINVAL;
+
+ if (override != AC97_TUNE_DEFAULT) {
+ result = apply_quirk(override);
+ if (result < 0)
+ printk(KERN_ERR "applying quirk type %d failed (%d)\n", override, result);
+ return result;
+ }
+
+ for (; quirk->vendor; quirk++) {
+ if (quirk->vendor != pdev->subsystem_vendor)
+ continue;
+ if ((! quirk->mask && quirk->device == pdev->subsystem_device) ||
+ quirk->device == (quirk->mask & pdev->subsystem_device)) {
+#ifdef DEBUG
+ printk("ac97 quirk for %s (%04x:%04x)\n", quirk->name, ac97->subsystem_vendor, pdev->subsystem_device);
+#endif
+ result = apply_quirk(quirk->type);
+ if (result < 0)
+ printk(KERN_ERR "applying quirk type %d for %s failed (%d)\n", quirk->type, quirk->name, result);
+ return result;
+ }
+ }
+ return 0;
+}
+
+EXPORT_SYMBOL_GPL(ac97_tune_hardware);
+
+MODULE_LICENSE("GPL");