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author | Linus Torvalds <torvalds@ppc970.osdl.org> | 2005-04-17 02:20:36 +0400 |
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committer | Linus Torvalds <torvalds@ppc970.osdl.org> | 2005-04-17 02:20:36 +0400 |
commit | 1da177e4c3f41524e886b7f1b8a0c1fc7321cac2 (patch) | |
tree | 0bba044c4ce775e45a88a51686b5d9f90697ea9d /sound/oss/ac97_codec.c | |
download | linux-1da177e4c3f41524e886b7f1b8a0c1fc7321cac2.tar.xz |
Linux-2.6.12-rc2v2.6.12-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.c | 1576 |
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"); |