| /* |
| * 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"); |