October 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/30] ALSA: HDA VIA: Remove unused IS_VT17xx_VENDORID macro
by Logan Li 10 Oct '09

10 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Remove unused IS_VT17xx_VENDORID macro IS_VT17*_VENDORID macros are used nowhere, so clean them up. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 8 -------- 1 file changed, 8 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -76,14 +76,6 @@ #define VT1702_HP_NID 0x17 #define VT1702_DIGOUT_NID 0x11 -#define IS_VT1708_VENDORID(x) ((x) >= 0x11061708 && (x) <= 0x1106170b) -#define IS_VT1709_10CH_VENDORID(x) ((x) >= 0x1106e710 && (x) <= 0x1106e713) -#define IS_VT1709_6CH_VENDORID(x) ((x) >= 0x1106e714 && (x) <= 0x1106e717) -#define IS_VT1708B_8CH_VENDORID(x) ((x) >= 0x1106e720 && (x) <= 0x1106e723) -#define IS_VT1708B_4CH_VENDORID(x) ((x) >= 0x1106e724 && (x) <= 0x1106e727) -#define IS_VT1708S_VENDORID(x) ((x) >= 0x11060397 && (x) <= 0x11067397) -#define IS_VT1702_VENDORID(x) ((x) >= 0x11060398 && (x) <= 0x11067398) - enum VIA_HDA_CODEC { UNKNOWN = -1, VT1708,

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[alsa-devel] [PATCH] sscape: coding style fixes
by Krzysztof Helt 10 Oct '09

10 Oct '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Fix coding style errors in the driver. Also, add missing argument for CMD_XXX_MIDI_VOL command. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- The third trial. I hope I got it right this time. sound/isa/sscape.c | 169 +++++++++++++++++++++++++-------------------------- 1 files changed, 83 insertions(+), 86 deletions(-) diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 279be50..0a706c4 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -109,14 +109,14 @@ MODULE_DEVICE_TABLE(pnp_card, sscape_pnpids); #define RX_READY 0x01 #define TX_READY 0x02 -#define CMD_ACK 0x80 -#define CMD_SET_MIDI_VOL 0x84 -#define CMD_GET_MIDI_VOL 0x85 -#define CMD_XXX_MIDI_VOL 0x86 -#define CMD_SET_EXTMIDI 0x8a -#define CMD_GET_EXTMIDI 0x8b -#define CMD_SET_MT32 0x8c -#define CMD_GET_MT32 0x8d +#define CMD_ACK 0x80 +#define CMD_SET_MIDI_VOL 0x84 +#define CMD_GET_MIDI_VOL 0x85 +#define CMD_XXX_MIDI_VOL 0x86 +#define CMD_SET_EXTMIDI 0x8a +#define CMD_GET_EXTMIDI 0x8b +#define CMD_SET_MT32 0x8c +#define CMD_GET_MT32 0x8d enum GA_REG { GA_INTSTAT_REG = 0, @@ -166,10 +166,12 @@ static inline struct soundscape *get_card_soundscape(struct snd_card *c) * I think this means that the memory has to map to * contiguous pages of physical memory. */ -static struct snd_dma_buffer *get_dmabuf(struct snd_dma_buffer *buf, unsigned long size) +static struct snd_dma_buffer *get_dmabuf(struct snd_dma_buffer *buf, + unsigned long size) { if (buf) { - if (snd_dma_alloc_pages_fallback(SNDRV_DMA_TYPE_DEV, snd_dma_isa_data(), + if (snd_dma_alloc_pages_fallback(SNDRV_DMA_TYPE_DEV, + snd_dma_isa_data(), size, buf) < 0) { snd_printk(KERN_ERR "sscape: Failed to allocate " "%lu bytes for DMA\n", @@ -190,13 +192,13 @@ static void free_dmabuf(struct snd_dma_buffer *buf) snd_dma_free_pages(buf); } - /* * This function writes to the SoundScape's control registers, * but doesn't do any locking. It's up to the caller to do that. * This is why this function is "unsafe" ... */ -static inline void sscape_write_unsafe(unsigned io_base, enum GA_REG reg, unsigned char val) +static inline void sscape_write_unsafe(unsigned io_base, enum GA_REG reg, + unsigned char val) { outb(reg, ODIE_ADDR_IO(io_base)); outb(val, ODIE_DATA_IO(io_base)); @@ -206,7 +208,8 @@ static inline void sscape_write_unsafe(unsigned io_base, enum GA_REG reg, unsign * Write to the SoundScape's control registers, and do the * necessary locking ... */ -static void sscape_write(struct soundscape *s, enum GA_REG reg, unsigned char val) +static void sscape_write(struct soundscape *s, enum GA_REG reg, + unsigned char val) { unsigned long flags; @@ -219,7 +222,8 @@ static void sscape_write(struct soundscape *s, enum GA_REG reg, unsigned char va * Read from the SoundScape's control registers, but leave any * locking to the caller. This is why the function is "unsafe" ... */ -static inline unsigned char sscape_read_unsafe(unsigned io_base, enum GA_REG reg) +static inline unsigned char sscape_read_unsafe(unsigned io_base, + enum GA_REG reg) { outb(reg, ODIE_ADDR_IO(io_base)); return inb(ODIE_DATA_IO(io_base)); @@ -248,9 +252,8 @@ static inline void set_midi_mode_unsafe(unsigned io_base) static inline int host_read_unsafe(unsigned io_base) { int data = -1; - if ((inb(HOST_CTRL_IO(io_base)) & RX_READY) != 0) { + if ((inb(HOST_CTRL_IO(io_base)) & RX_READY) != 0) data = inb(HOST_DATA_IO(io_base)); - } return data; } @@ -292,7 +295,7 @@ static inline int host_write_unsafe(unsigned io_base, unsigned char data) * Also leaves all locking-issues to the caller ... */ static int host_write_ctrl_unsafe(unsigned io_base, unsigned char data, - unsigned timeout) + unsigned timeout) { int err; @@ -311,7 +314,7 @@ static int host_write_ctrl_unsafe(unsigned io_base, unsigned char data, * * NOTE: This check is based upon observation, not documentation. */ -static inline int verify_mpu401(const struct snd_mpu401 * mpu) +static inline int verify_mpu401(const struct snd_mpu401 *mpu) { return ((inb(MPU401C(mpu)) & 0xc0) == 0x80); } @@ -319,7 +322,7 @@ static inline int verify_mpu401(const struct snd_mpu401 * mpu) /* * This is apparently the standard way to initailise an MPU-401 */ -static inline void initialise_mpu401(const struct snd_mpu401 * mpu) +static inline void initialise_mpu401(const struct snd_mpu401 *mpu) { outb(0, MPU401D(mpu)); } @@ -329,9 +332,10 @@ static inline void initialise_mpu401(const struct snd_mpu401 * mpu) * The AD1845 detection fails if we *don't* do this, so I * think that this is a good idea ... */ -static inline void activate_ad1845_unsafe(unsigned io_base) +static void activate_ad1845_unsafe(unsigned io_base) { - sscape_write_unsafe(io_base, GA_HMCTL_REG, (sscape_read_unsafe(io_base, GA_HMCTL_REG) & 0xcf) | 0x10); + unsigned char val = sscape_read_unsafe(io_base, GA_HMCTL_REG); + sscape_write_unsafe(io_base, GA_HMCTL_REG, (val & 0xcf) | 0x10); sscape_write_unsafe(io_base, GA_CDCFG_REG, 0x80); } @@ -350,24 +354,27 @@ static void soundscape_free(struct snd_card *c) * Tell the SoundScape to begin a DMA tranfer using the given channel. * All locking issues are left to the caller. */ -static inline void sscape_start_dma_unsafe(unsigned io_base, enum GA_REG reg) +static void sscape_start_dma_unsafe(unsigned io_base, enum GA_REG reg) { - sscape_write_unsafe(io_base, reg, sscape_read_unsafe(io_base, reg) | 0x01); - sscape_write_unsafe(io_base, reg, sscape_read_unsafe(io_base, reg) & 0xfe); + sscape_write_unsafe(io_base, reg, + sscape_read_unsafe(io_base, reg) | 0x01); + sscape_write_unsafe(io_base, reg, + sscape_read_unsafe(io_base, reg) & 0xfe); } /* * Wait for a DMA transfer to complete. This is a "limited busy-wait", * and all locking issues are left to the caller. */ -static int sscape_wait_dma_unsafe(unsigned io_base, enum GA_REG reg, unsigned timeout) +static int sscape_wait_dma_unsafe(unsigned io_base, enum GA_REG reg, + unsigned timeout) { while (!(sscape_read_unsafe(io_base, reg) & 0x01) && (timeout != 0)) { udelay(100); --timeout; } /* while */ - return (sscape_read_unsafe(io_base, reg) & 0x01); + return sscape_read_unsafe(io_base, reg) & 0x01; } /* @@ -427,13 +434,13 @@ static int host_startup_ack(struct soundscape *s, unsigned timeout) /* * Upload a byte-stream into the SoundScape using DMA channel A. */ -static int upload_dma_data(struct soundscape *s, - const unsigned char *data, - size_t size) +static int upload_dma_data(struct soundscape *s, const unsigned char *data, + size_t size) { unsigned long flags; struct snd_dma_buffer dma; int ret; + unsigned char val; if (!get_dmabuf(&dma, PAGE_ALIGN(32 * 1024))) return -ENOMEM; @@ -443,18 +450,21 @@ static int upload_dma_data(struct soundscape *s, /* * Reset the board ... */ - sscape_write_unsafe(s->io_base, GA_HMCTL_REG, sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f); + val = sscape_read_unsafe(s->io_base, GA_HMCTL_REG); + sscape_write_unsafe(s->io_base, GA_HMCTL_REG, val & 0x3f); /* * Enable the DMA channels and configure them ... */ - sscape_write_unsafe(s->io_base, GA_DMAA_REG, (s->chip->dma1 << 4) | DMA_8BIT); + val = (s->chip->dma1 << 4) | DMA_8BIT; + sscape_write_unsafe(s->io_base, GA_DMAA_REG, val); sscape_write_unsafe(s->io_base, GA_DMAB_REG, 0x20); /* * Take the board out of reset ... */ - sscape_write_unsafe(s->io_base, GA_HMCTL_REG, sscape_read_unsafe(s->io_base, GA_HMCTL_REG) | 0x80); + val = sscape_read_unsafe(s->io_base, GA_HMCTL_REG); + sscape_write_unsafe(s->io_base, GA_HMCTL_REG, val | 0x80); /* * Upload the firmware to the SoundScape @@ -472,7 +482,7 @@ static int upload_dma_data(struct soundscape *s, sscape_start_dma_unsafe(s->io_base, GA_DMAA_REG); if (!sscape_wait_dma_unsafe(s->io_base, GA_DMAA_REG, 5000)) { /* - * Don't forget to release this spinlock we're holding ... + * Don't forget to release this spinlock we're holding */ spin_unlock_irqrestore(&s->lock, flags); @@ -489,7 +499,8 @@ static int upload_dma_data(struct soundscape *s, /* * Boot the board ... (I think) */ - sscape_write_unsafe(s->io_base, GA_HMCTL_REG, sscape_read_unsafe(s->io_base, GA_HMCTL_REG) | 0x40); + val = sscape_read_unsafe(s->io_base, GA_HMCTL_REG); + sscape_write_unsafe(s->io_base, GA_HMCTL_REG, val | 0x40); spin_unlock_irqrestore(&s->lock, flags); /* @@ -591,7 +602,7 @@ static int sscape_upload_microcode(struct snd_card *card, int version) * Mixer control for the SoundScape's MIDI device. */ static int sscape_midi_info(struct snd_kcontrol *ctl, - struct snd_ctl_elem_info *uinfo) + struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 1; @@ -601,7 +612,7 @@ static int sscape_midi_info(struct snd_kcontrol *ctl, } static int sscape_midi_get(struct snd_kcontrol *kctl, - struct snd_ctl_elem_value *uctl) + struct snd_ctl_elem_value *uctl) { struct snd_wss *chip = snd_kcontrol_chip(kctl); struct snd_card *card = chip->card; @@ -615,16 +626,18 @@ static int sscape_midi_get(struct snd_kcontrol *kctl, } static int sscape_midi_put(struct snd_kcontrol *kctl, - struct snd_ctl_elem_value *uctl) + struct snd_ctl_elem_value *uctl) { struct snd_wss *chip = snd_kcontrol_chip(kctl); struct snd_card *card = chip->card; - register struct soundscape *s = get_card_soundscape(card); + struct soundscape *s = get_card_soundscape(card); unsigned long flags; int change; + unsigned char new_val; spin_lock_irqsave(&s->lock, flags); + new_val = uctl->value.integer.value[0] & 127; /* * We need to put the board into HOST mode before we * can send any volume-changing HOST commands ... @@ -637,15 +650,16 @@ static int sscape_midi_put(struct snd_kcontrol *kctl, * and then perform another volume-related command. Perhaps the * first command is an "open" and the second command is a "close"? */ - if (s->midi_vol == ((unsigned char) uctl->value.integer. value[0] & 127)) { + if (s->midi_vol == new_val) { change = 0; goto __skip_change; } - change = (host_write_ctrl_unsafe(s->io_base, CMD_SET_MIDI_VOL, 100) - && host_write_ctrl_unsafe(s->io_base, ((unsigned char) uctl->value.integer. value[0]) & 127, 100) - && host_write_ctrl_unsafe(s->io_base, CMD_XXX_MIDI_VOL, 100)); - s->midi_vol = (unsigned char) uctl->value.integer.value[0] & 127; - __skip_change: + change = host_write_ctrl_unsafe(s->io_base, CMD_SET_MIDI_VOL, 100) + && host_write_ctrl_unsafe(s->io_base, new_val, 100) + && host_write_ctrl_unsafe(s->io_base, CMD_XXX_MIDI_VOL, 100) + && host_write_ctrl_unsafe(s->io_base, new_val, 100); + s->midi_vol = new_val; +__skip_change: /* * Take the board out of HOST mode and back into MIDI mode ... @@ -738,7 +752,7 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) if (s->type == SSCAPE_VIVO) wss_io += 4; - d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f; + d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG); sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d | 0xc0); /* wait for WSS codec */ @@ -762,7 +776,7 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) if ((inb(wss_io) & 0x80) != 0) s->type = MEDIA_FX; - d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f; + d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG); sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d | 0xc0); /* wait for WSS codec */ for (d = 0; d < 500; d++) { @@ -778,7 +792,7 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) */ retval = 1; - _done: +_done: spin_unlock_irqrestore(&s->lock, flags); return retval; } @@ -789,7 +803,7 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) * to crash the machine. Also check that someone isn't using the hardware * IOCTL device. */ -static int mpu401_open(struct snd_mpu401 * mpu) +static int mpu401_open(struct snd_mpu401 *mpu) { if (!verify_mpu401(mpu)) { snd_printk(KERN_ERR "sscape: MIDI disabled, " @@ -803,18 +817,18 @@ static int mpu401_open(struct snd_mpu401 * mpu) /* * Initialse an MPU-401 subdevice for MIDI support on the SoundScape. */ -static int __devinit create_mpu401(struct snd_card *card, int devnum, unsigned long port, int irq) +static int __devinit create_mpu401(struct snd_card *card, int devnum, + unsigned long port, int irq) { struct soundscape *sscape = get_card_soundscape(card); struct snd_rawmidi *rawmidi; int err; - if ((err = snd_mpu401_uart_new(card, devnum, - MPU401_HW_MPU401, - port, MPU401_INFO_INTEGRATED, - irq, IRQF_DISABLED, - &rawmidi)) == 0) { - struct snd_mpu401 *mpu = (struct snd_mpu401 *) rawmidi->private_data; + err = snd_mpu401_uart_new(card, devnum, MPU401_HW_MPU401, port, + MPU401_INFO_INTEGRATED, irq, IRQF_DISABLED, + &rawmidi); + if (err == 0) { + struct snd_mpu401 *mpu = rawmidi->private_data; mpu->open_input = mpu401_open; mpu->open_output = mpu401_open; mpu->private_data = sscape; @@ -866,19 +880,6 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, unsigned long flags; struct snd_pcm *pcm; -/* - * It turns out that the PLAYBACK_ENABLE bit is set - * by the lowlevel driver ... - * -#define AD1845_IFACE_CONFIG \ - (CS4231_AUTOCALIB | CS4231_RECORD_ENABLE | CS4231_PLAYBACK_ENABLE) - snd_wss_mce_up(chip); - spin_lock_irqsave(&chip->reg_lock, flags); - snd_wss_out(chip, CS4231_IFACE_CTRL, AD1845_IFACE_CONFIG); - spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_wss_mce_down(chip); - */ - if (sscape->type != SSCAPE_VIVO) { /* * The input clock frequency on the SoundScape must @@ -928,7 +929,7 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, sscape->chip = chip; } - _error: +_error: return err; } @@ -1034,7 +1035,6 @@ static int __devinit create_sscape(int dev, struct snd_card *card) */ spin_lock_irqsave(&sscape->lock, flags); - sscape_write_unsafe(sscape->io_base, GA_INTENA_REG, 0x00); /* disable */ sscape_write_unsafe(sscape->io_base, GA_SMCFGA_REG, 0x2e); sscape_write_unsafe(sscape->io_base, GA_SMCFGB_REG, 0x00); @@ -1055,6 +1055,10 @@ static int __devinit create_sscape(int dev, struct snd_card *card) sscape_write_unsafe(sscape->io_base, GA_CDCFG_REG, 0x09 | DMA_8BIT | (dma[dev] << 4) | (irq_cfg << 1)); + /* + * Enable the master IRQ ... + */ + sscape_write_unsafe(sscape->io_base, GA_INTENA_REG, 0x80); spin_unlock_irqrestore(&sscape->lock, flags); @@ -1094,11 +1098,6 @@ static int __devinit create_sscape(int dev, struct snd_card *card) } /* - * Enable the master IRQ ... - */ - sscape_write(sscape, GA_INTENA_REG, 0x80); - - /* * Initialize mixer */ spin_lock_irqsave(&sscape->lock, flags); @@ -1155,7 +1154,8 @@ static int __devinit snd_sscape_match(struct device *pdev, unsigned int i) mpu_irq[i] == SNDRV_AUTO_IRQ || dma[i] == SNDRV_AUTO_DMA) { printk(KERN_INFO - "sscape: insufficient parameters, need IO, IRQ, MPU-IRQ and DMA\n"); + "sscape: insufficient parameters, " + "need IO, IRQ, MPU-IRQ and DMA\n"); return 0; } @@ -1183,7 +1183,8 @@ static int __devinit snd_sscape_probe(struct device *pdev, unsigned int dev) if (ret < 0) goto _release_card; - if ((ret = snd_card_register(card)) < 0) { + ret = snd_card_register(card); + if (ret < 0) { snd_printk(KERN_ERR "sscape: Failed to register sound card\n"); goto _release_card; } @@ -1236,20 +1237,15 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, * Allow this function to fail *quietly* if all the ISA PnP * devices were configured using module parameters instead. */ - if ((idx = get_next_autoindex(idx)) >= SNDRV_CARDS) + idx = get_next_autoindex(idx); + if (idx >= SNDRV_CARDS) return -ENOSPC; /* - * We have found a candidate ISA PnP card. Now we - * have to check that it has the devices that we - * expect it to have. - */ - - /* * Check that we still have room for another sound card ... */ dev = pnp_request_card_device(pcard, pid->devs[0].id, NULL); - if (! dev) + if (!dev) return -ENODEV; if (!pnp_is_active(dev)) { @@ -1298,7 +1294,8 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, if (ret < 0) goto _release_card; - if ((ret = snd_card_register(card)) < 0) { + ret = snd_card_register(card); + if (ret < 0) { snd_printk(KERN_ERR "sscape: Failed to register sound card\n"); goto _release_card; } -- 1.6.0.3 --------------------------------------------------------------- Przygotuj si� na przysz�o��! Kliknij >>> http://link.interia.pl/f238a

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[alsa-devel] [PATCH] wss: convert CS4231 mixer to dB scale
by Krzysztof Helt 10 Oct '09

10 Oct '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Convert CS4231 mixer to dB scale after AD1848 mixer. Also, add missing microphone boost control for the AD1848 and correct wrong bits for loopback volume on the AD1848. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- I wonder if it is a good idea to reorder the snd_wss_controls table so the first part is identical to the snd_ad1848_controls. The snd_ad1848_controls can be removed completely and the first part of the snd_wss_controls can be used for the AD1848 chips. sound/isa/wss/wss_lib.c | 43 ++++++++++++++++++++++++++----------------- 1 files changed, 26 insertions(+), 17 deletions(-) diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index 5d2ba1b..754a208 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -2198,6 +2198,7 @@ EXPORT_SYMBOL(snd_wss_put_double); static const DECLARE_TLV_DB_SCALE(db_scale_6bit, -9450, 150, 0); static const DECLARE_TLV_DB_SCALE(db_scale_5bit_12db_max, -3450, 150, 0); static const DECLARE_TLV_DB_SCALE(db_scale_rec_gain, 0, 150, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_4bit, -4500, 300, 0); static struct snd_kcontrol_new snd_ad1848_controls[] = { WSS_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, @@ -2224,38 +2225,45 @@ WSS_DOUBLE_TLV("Capture Volume", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, .get = snd_wss_get_mux, .put = snd_wss_put_mux, }, +WSS_DOUBLE("Mic Boost", 0, + CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 5, 5, 1, 0), WSS_SINGLE("Loopback Capture Switch", 0, CS4231_LOOPBACK, 0, 1, 0), -WSS_SINGLE_TLV("Loopback Capture Volume", 0, CS4231_LOOPBACK, 1, 63, 0, +WSS_SINGLE_TLV("Loopback Capture Volume", 0, CS4231_LOOPBACK, 2, 63, 1, db_scale_6bit), }; static struct snd_kcontrol_new snd_wss_controls[] = { WSS_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), -WSS_DOUBLE("PCM Playback Volume", 0, - CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1), +WSS_DOUBLE_TLV("PCM Playback Volume", 0, + CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1, + db_scale_6bit), WSS_DOUBLE("Line Playback Switch", 0, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), -WSS_DOUBLE("Line Playback Volume", 0, - CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1), +WSS_DOUBLE_TLV("Line Playback Volume", 0, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1, + db_scale_5bit_12db_max), WSS_DOUBLE("Aux Playback Switch", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE("Aux Playback Volume", 0, - CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1), +WSS_DOUBLE_TLV("Aux Playback Volume", 0, + CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1, + db_scale_5bit_12db_max), WSS_DOUBLE("Aux Playback Switch", 1, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE("Aux Playback Volume", 1, - CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1), +WSS_DOUBLE_TLV("Aux Playback Volume", 1, + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1, + db_scale_5bit_12db_max), WSS_SINGLE("Mono Playback Switch", 0, CS4231_MONO_CTRL, 7, 1, 1), -WSS_SINGLE("Mono Playback Volume", 0, - CS4231_MONO_CTRL, 0, 15, 1), +WSS_SINGLE_TLV("Mono Playback Volume", 0, + CS4231_MONO_CTRL, 0, 15, 1, + db_scale_4bit), WSS_SINGLE("Mono Output Playback Switch", 0, CS4231_MONO_CTRL, 6, 1, 1), WSS_SINGLE("Mono Output Playback Bypass", 0, CS4231_MONO_CTRL, 5, 1, 0), -WSS_DOUBLE("Capture Volume", 0, - CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0, 15, 0), +WSS_DOUBLE_TLV("Capture Volume", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, + 0, 0, 15, 0, db_scale_rec_gain), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Capture Source", @@ -2267,15 +2275,16 @@ WSS_DOUBLE("Mic Boost", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 5, 5, 1, 0), WSS_SINGLE("Loopback Capture Switch", 0, CS4231_LOOPBACK, 0, 1, 0), -WSS_SINGLE("Loopback Capture Volume", 0, - CS4231_LOOPBACK, 2, 63, 1) +WSS_SINGLE_TLV("Loopback Capture Volume", 0, CS4231_LOOPBACK, 2, 63, 1, + db_scale_6bit), }; static struct snd_kcontrol_new snd_opti93x_controls[] = { WSS_DOUBLE("Master Playback Switch", 0, OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 7, 7, 1, 1), -WSS_DOUBLE("Master Playback Volume", 0, - OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 1, 1, 31, 1), +WSS_DOUBLE_TLV("Master Playback Volume", 0, + OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 1, 1, 31, 1, + db_scale_6bit), WSS_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), WSS_DOUBLE("PCM Playback Volume", 0, -- 1.6.0.3 ---------------------------------------------------------------------- Szukasz pracy? Chcesz lepiej zarabia�? Sprawd� oferty na http://link.interia.pl/f23ba

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[alsa-devel] [PATCH] ice1724: Fix surround on Chaintech AV-710
by Robert Hancock 10 Oct '09

10 Oct '09

Fix the num_total_dacs setting for Chaintech AV710. The existing comment that only PSDOUT0 is connected is correct, but since the card is using packed AC97 mode to send 6 channels to the codec, num_total_dacs should be set to 6 and not 2. This allows 6-channel surround to work. Also clarify a comment regarding the additional WM8728 codec on this card (it's connected to the SPDIF output and always receives the same data). Signed-off-by: Robert Hancock <hancockrwd(a)gmail.com> diff --git a/sound/pci/ice1712/amp.c b/sound/pci/ice1712/amp.c index 3756430..bfcf989 100644 --- a/sound/pci/ice1712/amp.c +++ b/sound/pci/ice1712/amp.c @@ -52,11 +52,13 @@ static int __devinit snd_vt1724_amp_init(struct snd_ice1712 *ice) /* only use basic functionality for now */ - ice->num_total_dacs = 2; /* only PSDOUT0 is connected */ + /* VT1616 6ch codec connected to PSDOUT0 using packed mode */ + ice->num_total_dacs = 6; ice->num_total_adcs = 2; - /* Chaintech AV-710 has another codecs, which need initialization */ - /* initialize WM8728 codec */ + /* Chaintech AV-710 has another WM8728 codec connected to PSDOUT4 + (shared with the SPDIF output). Mixer control for this codec + is not yet supported. */ if (ice->eeprom.subvendor == VT1724_SUBDEVICE_AV710) { for (i = 0; i < ARRAY_SIZE(wm_inits); i += 2) wm_put(ice, wm_inits[i], wm_inits[i+1]);

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[alsa-devel] ASoC updates for 2.6.33
by Mark Brown 10 Oct '09

10 Oct '09

The following changes since commit 907bc6c7fc7071b00083fc11e510e47dd93df45d: Mark Brown (1): Merge branch 'for-2.6.32' into for-2.6.33 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.33 Mark Brown (5): ASoC: Remove absent SYNC and TDM DAI format options from i.MX SSI ASoC: WM8350 capture PGA mutes are inverted Merge branch 'upstream/wm8350' into for-2.6.32 Merge branch 'for-2.6.32' into for-2.6.33 ASoC: Minor fixups to tpa6130a2 driver Nicolas Ferre (1): ASoC: at91sam9g20ek_2mmc board uses same audio connexion as at91sam9g20ek Peter Ujfalusi (1): ASoC: TPA6130A2 amplifier driver include/sound/tpa6130a2-plat.h | 30 +++ sound/soc/atmel/sam9g20_wm8731.c | 2 +- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tpa6130a2.c | 463 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tpa6130a2.h | 62 +++++ sound/soc/codecs/wm8350.c | 4 +- sound/soc/imx/mxc-ssi.c | 8 - 8 files changed, 564 insertions(+), 11 deletions(-) create mode 100644 include/sound/tpa6130a2-plat.h create mode 100644 sound/soc/codecs/tpa6130a2.c create mode 100644 sound/soc/codecs/tpa6130a2.h

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[alsa-devel] VIA HDA patch problem (was [PATCH 17/31] HDA patch_via.c: Add second S/PDIF out control for VT1708S and VT1702.)
by Robert Hancock 10 Oct '09

10 Oct '09

It looks like there's some problem somewhere in the patch set posted on Oct. 7th that wasn't in the original patch set from the 5th: I can't open the mixer for the HDA device: alsamixer -c 1 cannot load mixer controls: Operation not permitted _From strace: open("/dev/snd/controlC1", O_RDWR) = 3 ioctl(3, USBDEVFS_CONTROL, 0x7fff338e57fc) = 0 fcntl(3, F_GETFL) = 0x8002 (flags O_RDWR|O_LARGEFILE) fcntl(3, F_SETFL, O_RDWR|O_NONBLOCK|O_LARGEFILE) = 0 ioctl(3, USBDEVFS_RELEASEINTERFACE, 0x7fff338e5990) = 0 ioctl(3, USBDEVFS_RELEASEINTERFACE, 0x7fff338e5990) = 0 ioctl(3, USBDEVFS_CONNECTINFO, 0x7fff338e5670) = 0 ioctl(3, USBDEVFS_IOCTL, 0x7fff338e4ea0) = -1 EPERM (Operation not permitted) Presumably something is creating a control that isn't readable, or writable or something, and this is causing the mixer to fail to load? Also, with regard to the SPDIF problem I'm seeing on my Asus P7P55D (VT1828S), I was able to dig up a little bit more info: by using hda_analyzer to set the ENABLE flag on the Digital Converter, I was able to get the optical SPDIF port to light up and my receiver indicates PCM input. Unfortunately there's still no actual audio coming through on the receiver, just silence. However, obviously that ENABLE flag is going to have to get set somehow to have any hope of SPDIF working, and currently it seems like it's not..

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[alsa-devel] ascenario: Last batch of fixes.
by Stefan Schmidt 09 Oct '09

09 Oct '09

Hello. This is the last set of fixes I have in my queue. [PATCH 1/3] ascenario: Add check for set scenario in snd_scenario_get_control_id() Debugged some hours that I should test it after _set() and not only after _list(). Better let the user know that no scenario is set instead of silently using a non existing scenario and fail. [PATCH 2/3] ascenario: Convert last sprintf() to snprintf() Some more sprintf() replaced with the more secure snprintf(). [PATCH 3/3] ascenario: Make sequences unique for every scenario. That one actually got me some grey hairs. I'm pretty sure I tested it earlier and it worked as it should, but I can find now traces of this. Fixed for good now. [PATCH] ascenario: Add dump and configure utils for ascneario. Reference patch against alsa-utils for some small utilities using the new scenario API. Only change from the last time is that _dump() now has two parameters even if the new one is still unused. regards Stefan Schmidt

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[alsa-devel] [PATCH 0/30] ALSA HDA VIA patches
by LoganLi＠viatech.com.cn 09 Oct '09

09 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Add Jack detect feature for VT1708. VT1708 does not support unsolicited response, but we need hp detect to automute speaker. Implemented in workqueue. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- pci/hda/patch_via.c | 230 +++++++++++++++++++++++++++++++++++++++------------- 1 file changed, 173 insertions(+), 57 deletions(-) --- a/pci/hda/patch_via.c +++ b/pci/hda/patch_via.c @@ -89,6 +89,64 @@ CODEC_TYPES, }; +struct via_spec { + /* codec parameterization */ + struct snd_kcontrol_new *mixers[4]; + unsigned int num_mixers; + + struct hda_verb *init_verbs[5]; + unsigned int num_iverbs; + + char *stream_name_analog; + struct hda_pcm_stream *stream_analog_playback; + struct hda_pcm_stream *stream_analog_capture; + + char *stream_name_digital; + struct hda_pcm_stream *stream_digital_playback; + struct hda_pcm_stream *stream_digital_capture; + + /* playback */ + struct hda_multi_out multiout; + hda_nid_t slave_dig_outs[2]; + + /* capture */ + unsigned int num_adc_nids; + hda_nid_t *adc_nids; + hda_nid_t mux_nids[3]; + hda_nid_t dig_in_nid; + hda_nid_t dig_in_pin; + + /* capture source */ + const struct hda_input_mux *input_mux; + unsigned int cur_mux[3]; + + /* PCM information */ + struct hda_pcm pcm_rec[3]; + + /* dynamic controls, init_verbs and input_mux */ + struct auto_pin_cfg autocfg; + struct snd_array kctls; + struct hda_input_mux private_imux[2]; + hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS]; + + /* HP mode source */ + const struct hda_input_mux *hp_mux; + unsigned int hp_independent_mode; + unsigned int hp_independent_mode_index; + unsigned int smart51_enabled; + + enum VIA_HDA_CODEC codec_type; + + /* work to check hp jack state */ + struct hda_codec *codec; + struct delayed_work vt1708_hp_work; + int vt1708_jack_detectect; + int vt1708_hp_present; +#ifdef CONFIG_SND_HDA_POWER_SAVE + struct hda_loopback_check loopback; +#endif +}; + static enum VIA_HDA_CODEC get_codec_type(struct hda_codec *codec) { u32 vendor_id = codec->vendor_id; @@ -181,6 +239,31 @@ static void analog_low_current_mode(struct hda_codec *codec, int stream_idle); static void set_jack_power_state(struct hda_codec *codec); +static int is_aa_path_mute(struct hda_codec *codec); + +static void vt1708_start_hp_work(struct via_spec *spec) +{ + if (spec->codec_type != VT1708 || spec->autocfg.hp_pins[0] == 0) + return; + snd_hda_codec_write(spec->codec, 0x1, 0, 0xf81, + !spec->vt1708_jack_detectect); + if (!delayed_work_pending(&spec->vt1708_hp_work)) + schedule_delayed_work(&spec->vt1708_hp_work, + msecs_to_jiffies(100)); +} + +static void vt1708_stop_hp_work(struct via_spec *spec) +{ + if (spec->codec_type != VT1708 || spec->autocfg.hp_pins[0] == 0) + return; + if (snd_hda_get_bool_hint(spec->codec, "analog_loopback_hp_detect") == 1 + && !is_aa_path_mute(spec->codec)) + return; + snd_hda_codec_write(spec->codec, 0x1, 0, 0xf81, + !spec->vt1708_jack_detectect); + cancel_delayed_work(&spec->vt1708_hp_work); + flush_scheduled_work(); +} static int analog_input_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -190,6 +273,12 @@ set_jack_power_state(codec); analog_low_current_mode(snd_kcontrol_chip(kcontrol), -1); + if (snd_hda_get_bool_hint(codec, "analog_loopback_hp_detect") == 1) { + if (is_aa_path_mute(codec)) + vt1708_start_hp_work(codec->spec); + else + vt1708_stop_hp_work(codec->spec); + } return change; } @@ -210,59 +299,6 @@ }; -struct via_spec { - /* codec parameterization */ - struct snd_kcontrol_new *mixers[4]; - unsigned int num_mixers; - - struct hda_verb *init_verbs[5]; - unsigned int num_iverbs; - - char *stream_name_analog; - struct hda_pcm_stream *stream_analog_playback; - struct hda_pcm_stream *stream_analog_capture; - - char *stream_name_digital; - struct hda_pcm_stream *stream_digital_playback; - struct hda_pcm_stream *stream_digital_capture; - - /* playback */ - struct hda_multi_out multiout; - hda_nid_t slave_dig_outs[2]; - - /* capture */ - unsigned int num_adc_nids; - hda_nid_t *adc_nids; - hda_nid_t mux_nids[3]; - hda_nid_t dig_in_nid; - hda_nid_t dig_in_pin; - - /* capture source */ - const struct hda_input_mux *input_mux; - unsigned int cur_mux[3]; - - /* PCM information */ - struct hda_pcm pcm_rec[3]; - - /* dynamic controls, init_verbs and input_mux */ - struct auto_pin_cfg autocfg; - struct snd_array kctls; - struct hda_input_mux private_imux[2]; - hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS]; - - /* HP mode source */ - const struct hda_input_mux *hp_mux; - unsigned int hp_independent_mode; - unsigned int hp_independent_mode_index; - unsigned int smart51_enabled; - - enum VIA_HDA_CODEC codec_type; - -#ifdef CONFIG_SND_HDA_POWER_SAVE - struct hda_loopback_check loopback; -#endif -}; - static hda_nid_t vt1708_adc_nids[2] = { /* ADC1-2 */ 0x15, 0x27 @@ -981,7 +1017,6 @@ struct via_spec *spec = codec->spec; int idle = substream->pstr->substream_opened == 1 && substream->ref_count == 0; - analog_low_current_mode(codec, idle); return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, hinfo); @@ -994,6 +1029,7 @@ struct snd_pcm_substream *substream) { struct via_spec *spec = codec->spec; + vt1708_start_hp_work(spec); return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, stream_tag, format, substream); } @@ -1003,6 +1039,7 @@ struct snd_pcm_substream *substream) { struct via_spec *spec = codec->spec; + vt1708_stop_hp_work(spec); return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout); } @@ -1094,7 +1131,7 @@ snd_hda_codec_setup_stream(codec, mout->hp_nid, stream_tag, 0, format); } - + vt1708_start_hp_work(spec); return 0; } @@ -1134,7 +1171,7 @@ snd_hda_codec_setup_stream(codec, mout->hp_nid, 0, 0, 0); } - + vt1708_stop_hp_work(spec); return 0; } @@ -1345,6 +1382,7 @@ return; via_free_kctls(codec); + vt1708_stop_hp_work(spec); kfree(codec->spec); } @@ -1464,6 +1502,15 @@ return 0; } +#ifdef SND_HDA_NEEDS_RESUME +static int via_suspend(struct hda_codec *codec, pm_message_t state) +{ + struct via_spec *spec = codec->spec; + vt1708_stop_hp_work(spec); + return 0; +} +#endif + #ifdef CONFIG_SND_HDA_POWER_SAVE static int via_check_power_status(struct hda_codec *codec, hda_nid_t nid) { @@ -1479,6 +1526,9 @@ .build_pcms = via_build_pcms, .init = via_init, .free = via_free, +#ifdef SND_HDA_NEEDS_RESUME + .suspend = via_suspend, +#endif #ifdef CONFIG_SND_HDA_POWER_SAVE .check_power_status = via_check_power_status, #endif @@ -1728,6 +1778,51 @@ return; } +static int vt1708_jack_detectect_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct via_spec *spec = codec->spec; + + if (spec->codec_type != VT1708) + return 0; + spec->vt1708_jack_detectect = + !((snd_hda_codec_read(codec, 0x1, 0, 0xf84, 0) >> 8) & 0x1); + ucontrol->value.integer.value[0] = spec->vt1708_jack_detectect; + return 0; +} + +static int vt1708_jack_detectect_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct via_spec *spec = codec->spec; + int change; + + if (spec->codec_type != VT1708) + return 0; + spec->vt1708_jack_detectect = ucontrol->value.integer.value[0]; + change = (0x1 & (snd_hda_codec_read(codec, 0x1, 0, 0xf84, 0) >> 8)) + == !spec->vt1708_jack_detectect; + if (spec->vt1708_jack_detectect) { + mute_aa_path(codec, 1); + notify_aa_path_ctls(codec); + } + return change; +} + +static struct snd_kcontrol_new vt1708_jack_detectect[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Jack Detect", + .count = 1, + .info = snd_ctl_boolean_mono_info, + .get = vt1708_jack_detectect_get, + .put = vt1708_jack_detectect_put, + }, + {} /* end */ +}; + static int vt1708_parse_auto_config(struct hda_codec *codec) { struct via_spec *spec = codec->spec; @@ -1755,6 +1850,10 @@ err = vt1708_auto_create_analog_input_ctls(spec, &spec->autocfg); if (err < 0) return err; + /* add jack detect on/off control */ + err = snd_hda_add_new_ctls(codec, vt1708_jack_detectect); + if (err < 0) + return err; spec->multiout.max_channels = spec->multiout.num_dacs * 2; @@ -1788,6 +1887,22 @@ return 0; } +static void vt1708_update_hp_jack_state(struct work_struct *work) +{ + struct via_spec *spec = container_of(work, struct via_spec, + vt1708_hp_work.work); + if (spec->codec_type != VT1708) + return; + /* if jack state toggled */ + if (spec->vt1708_hp_present + != (snd_hda_codec_read(spec->codec, spec->autocfg.hp_pins[0], 0, + AC_VERB_GET_PIN_SENSE, 0) >> 31)) { + spec->vt1708_hp_present ^= 1; + via_hp_automute(spec->codec); + } + vt1708_start_hp_work(spec); +} + static int get_mux_nids(struct hda_codec *codec) { struct via_spec *spec = codec->spec; @@ -1863,7 +1978,8 @@ #ifdef CONFIG_SND_HDA_POWER_SAVE spec->loopback.amplist = vt1708_loopbacks; #endif - + spec->codec = codec; + INIT_DELAYED_WORK(&spec->vt1708_hp_work, vt1708_update_hp_jack_state); return 0; }

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[alsa-devel] [PATCH] sound: at91sam9g20ek_2mmc board uses same audio connexion as at91sam9g20ek
by Nicolas Ferre 09 Oct '09

09 Oct '09

The modified revision of at91sam9g20 Evaluation Kit rev. C and onwards share with previous ones its audio connexion to Wolfson wm8731. Modify the SoC file to extend the machine ID checking. Signed-off-by: Nicolas Ferre <nicolas.ferre(a)atmel.com> --- sound/soc/atmel/sam9g20_wm8731.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index 885ba01..e028744 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -207,7 +207,7 @@ static int __init at91sam9g20ek_init(void) struct clk *pllb; int ret; - if (!machine_is_at91sam9g20ek()) + if (!(machine_is_at91sam9g20ek() || machine_is_at91sam9g20ek_2mmc())) return -ENODEV; /* -- 1.5.6.5

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[alsa-devel] [PATCH 30/30] ALSA: HDA VIA: Only cosmetic changes
by LoganLi＠viatech.com.cn 09 Oct '09

09 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Only cosmetic changes Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- pci/hda/patch_via.c | 64 ++++++++++++++++++++++++++-------------------------- 1 file changed, 33 insertions(+), 31 deletions(-) --- a/pci/hda/patch_via.c +++ b/pci/hda/patch_via.c @@ -22,26 +22,26 @@ */ /* * * * * * * * * * * * * * Release History * * * * * * * * * * * * * * * * */ -/* */ +/* */ /* 2006-03-03 Lydia Wang Create the basic patch to support VT1708 codec */ -/* 2006-03-14 Lydia Wang Modify hard code for some pin widget nid */ -/* 2006-08-02 Lydia Wang Add support to VT1709 codec */ +/* 2006-03-14 Lydia Wang Modify hard code for some pin widget nid */ +/* 2006-08-02 Lydia Wang Add support to VT1709 codec */ /* 2006-09-08 Lydia Wang Fix internal loopback recording source select bug */ -/* 2007-09-12 Lydia Wang Add EAPD enable during driver initialization */ -/* 2007-09-17 Lydia Wang Add VT1708B codec support */ +/* 2007-09-12 Lydia Wang Add EAPD enable during driver initialization */ +/* 2007-09-17 Lydia Wang Add VT1708B codec support */ /* 2007-11-14 Lydia Wang Add VT1708A codec HP and CD pin connect config */ /* 2008-02-03 Lydia Wang Fix Rear channels and Back channels inverse issue */ -/* 2008-03-06 Lydia Wang Add VT1702 codec and VT1708S codec support */ -/* 2008-04-09 Lydia Wang Add mute front speaker when HP plugin */ -/* 2008-04-09 Lydia Wang Add Independent HP feature */ +/* 2008-03-06 Lydia Wang Add VT1702 codec and VT1708S codec support */ +/* 2008-04-09 Lydia Wang Add mute front speaker when HP plugin */ +/* 2008-04-09 Lydia Wang Add Independent HP feature */ /* 2008-05-28 Lydia Wang Add second S/PDIF Out support for VT1702 */ -/* 2008-09-15 Logan Li Add VT1708S Mic Boost workaround/backdoor */ +/* 2008-09-15 Logan Li Add VT1708S Mic Boost workaround/backdoor */ /* 2009-02-16 Logan Li Add support for VT1718S */ /* 2009-03-13 Logan Li Add support for VT1716S */ /* 2009-04-14 Lydai Wang Add support for VT1828S and VT2020 */ /* 2009-07-08 Lydia Wang Add support for VT2002P */ /* 2009-07-21 Lydia Wang Add support for VT1812 */ -/* */ +/* */ /* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * */ @@ -486,7 +486,7 @@ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); if (snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_EAPD) - snd_hda_codec_write(codec, nid, 0, + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_EAPD_BTLENABLE, 0x02); } @@ -1545,7 +1545,7 @@ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - + /* Setup default input MW0 to PW4 */ {0x20, AC_VERB_SET_CONNECT_SEL, 0}, /* PW9 Output enable */ @@ -1887,8 +1887,10 @@ codec->pcm_info = info; info->name = spec->stream_name_analog; - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *(spec->stream_analog_playback); - info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dac_nids[0]; + info->stream[SNDRV_PCM_STREAM_PLAYBACK] = + *(spec->stream_analog_playback); + info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = + spec->multiout.dac_nids[0]; info->stream[SNDRV_PCM_STREAM_CAPTURE] = *(spec->stream_analog_capture); info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0]; @@ -2133,7 +2135,7 @@ } else /* enable SPDIF-input pin */ snd_hda_codec_write(codec, spec->autocfg.dig_in_pin, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN); - return 0; + return 0; } #ifdef SND_HDA_NEEDS_RESUME @@ -2178,8 +2180,8 @@ spec->multiout.num_dacs = cfg->line_outs; spec->multiout.dac_nids = spec->private_dac_nids; - - for(i = 0; i < 4; i++) { + + for (i = 0; i < 4; i++) { nid = cfg->line_out_pins[i]; if (nid) { /* config dac list */ @@ -2217,7 +2219,7 @@ if (!nid) continue; - + nid_vol = nid_vols[i]; if (i == AUTO_SEQ_CENLFE) { @@ -2246,7 +2248,7 @@ HDA_OUTPUT)); if (err < 0) return err; - } else if (i == AUTO_SEQ_FRONT){ + } else if (i == AUTO_SEQ_FRONT) { /* add control to mixer index 0 */ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Master Front Playback Volume", @@ -2260,7 +2262,7 @@ HDA_INPUT)); if (err < 0) return err; - + /* add control to PW3 */ sprintf(name, "%s Playback Volume", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name, @@ -2360,7 +2362,7 @@ case 0x1d: /* Mic */ idx = 2; break; - + case 0x1e: /* Line In */ idx = 3; break; @@ -2593,7 +2595,7 @@ "from BIOS. Using genenic mode...\n"); } - + spec->stream_name_analog = "VT1708 Analog"; spec->stream_analog_playback = &vt1708_pcm_analog_playback; /* disable 32bit format on VT1708 */ @@ -2605,7 +2607,7 @@ spec->stream_digital_playback = &vt1708_pcm_digital_playback; spec->stream_digital_capture = &vt1708_pcm_digital_capture; - + if (!spec->adc_nids && spec->input_mux) { spec->adc_nids = vt1708_adc_nids; spec->num_adc_nids = ARRAY_SIZE(vt1708_adc_nids); @@ -2791,11 +2793,11 @@ spec->multiout.dac_nids[cfg->line_outs] = 0x28; /* AOW4 */ } else if (cfg->line_outs == 3) { /* 6 channels */ - for(i = 0; i < cfg->line_outs; i++) { + for (i = 0; i < cfg->line_outs; i++) { nid = cfg->line_out_pins[i]; if (nid) { /* config dac list */ - switch(i) { + switch (i) { case AUTO_SEQ_FRONT: /* AOW0 */ spec->multiout.dac_nids[i] = 0x10; @@ -2830,7 +2832,7 @@ for (i = 0; i <= AUTO_SEQ_SIDE; i++) { nid = cfg->line_out_pins[i]; - if (!nid) + if (!nid) continue; nid_vol = nid_vols[i]; @@ -2861,7 +2863,7 @@ HDA_OUTPUT)); if (err < 0) return err; - } else if (i == AUTO_SEQ_FRONT){ + } else if (i == AUTO_SEQ_FRONT) { /* ADD control to mixer index 0 */ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Master Front Playback Volume", @@ -2875,7 +2877,7 @@ HDA_INPUT)); if (err < 0) return err; - + /* add control to PW3 */ sprintf(name, "%s Playback Volume", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name, @@ -2971,7 +2973,7 @@ case 0x1d: /* Mic */ idx = 2; break; - + case 0x1e: /* Line In */ idx = 3; break; @@ -3080,7 +3082,7 @@ spec->stream_digital_playback = &vt1709_pcm_digital_playback; spec->stream_digital_capture = &vt1709_pcm_digital_capture; - + if (!spec->adc_nids && spec->input_mux) { spec->adc_nids = vt1709_adc_nids; spec->num_adc_nids = ARRAY_SIZE(vt1709_adc_nids); @@ -3174,7 +3176,7 @@ spec->stream_digital_playback = &vt1709_pcm_digital_playback; spec->stream_digital_capture = &vt1709_pcm_digital_capture; - + if (!spec->adc_nids && spec->input_mux) { spec->adc_nids = vt1709_adc_nids; spec->num_adc_nids = ARRAY_SIZE(vt1709_adc_nids);

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