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[alsa-devel] [PATCH] ice1712/aureon.c: fix coding style
by Alexander Beregalov 07 Sep '08

07 Sep '08

before: total: 304 errors, 137 warnings, 2259 lines checked after: total: 0 errors, 121 warnings, 2284 lines checked Compile tested, size is different because of include was changed, but without that change md5sum is different because of cmp cx,dx/cmp dx,cx swap and __LINE__ was changed in printk function. Signed-off-by: Alexander Beregalov <a.beregalov(a)gmail.com> --- sound/pci/ice1712/aureon.c | 677 +++++++++++++++++++++++--------------------- 1 files changed, 351 insertions(+), 326 deletions(-) diff --git a/sound/pci/ice1712/aureon.c b/sound/pci/ice1712/aureon.c index a137fc1..110d16e 100644 --- a/sound/pci/ice1712/aureon.c +++ b/sound/pci/ice1712/aureon.c @@ -44,10 +44,9 @@ * not working: prety much everything else, at least i could verify that * we have no digital output, no capture, pretty bad clicks and poops * on mixer switch and other coll stuff. - * - */ + */ -#include <asm/io.h> +#include <linux/io.h> #include <linux/delay.h> #include <linux/interrupt.h> #include <linux/init.h> @@ -131,7 +130,7 @@ static void aureon_pca9554_write(struct snd_ice1712 *ice, unsigned char reg, snd_ice1712_gpio_write(ice, tmp); udelay(50); - /* + /* * send i2c stop condition and start condition * to obtain sane state */ @@ -152,10 +151,16 @@ static void aureon_pca9554_write(struct snd_ice1712 *ice, unsigned char reg, * skipping ack cycles inbetween */ for (j = 0; j < 3; j++) { - switch(j) { - case 0: val = dev; break; - case 1: val = reg; break; - case 2: val = data; break; + switch (j) { + case 0: + val = dev; + break; + case 1: + val = reg; + break; + case 2: + val = data; + break; } for (i = 7; i >= 0; i--) { tmp &= ~AUREON_SPI_CLK; @@ -171,7 +176,7 @@ static void aureon_pca9554_write(struct snd_ice1712 *ice, unsigned char reg, snd_ice1712_gpio_write(ice, tmp); udelay(40); } - tmp &= ~AUREON_SPI_CLK; + tmp &= ~AUREON_SPI_CLK; snd_ice1712_gpio_write(ice, tmp); udelay(40); tmp |= AUREON_SPI_CLK; @@ -203,7 +208,7 @@ static int aureon_universe_inmux_info(struct snd_kcontrol *kcontrol, uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = 3; - if(uinfo->value.enumerated.item >= uinfo->value.enumerated.items) + if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); return 0; @@ -231,12 +236,12 @@ static int aureon_universe_inmux_put(struct snd_kcontrol *kcontrol, return -EINVAL; snd_ice1712_save_gpio_status(ice); oval = spec->pca9554_out; - if ((change = (oval != nval))) { + change = (oval != nval); + if (change) { aureon_pca9554_write(ice, PCA9554_OUT, nval); spec->pca9554_out = nval; } snd_ice1712_restore_gpio_status(ice); - return change; } @@ -256,7 +261,7 @@ static void aureon_ac97_write(struct snd_ice1712 *ice, unsigned short reg, udelay(10); tmp &= ~AUREON_AC97_ADDR; snd_ice1712_gpio_write(ice, tmp); - udelay(10); + udelay(10); /* Send low-order byte to XILINX chip */ tmp &= ~AUREON_AC97_DATA_MASK; @@ -269,7 +274,7 @@ static void aureon_ac97_write(struct snd_ice1712 *ice, unsigned short reg, tmp &= ~AUREON_AC97_DATA_LOW; snd_ice1712_gpio_write(ice, tmp); udelay(10); - + /* Send high-order byte to XILINX chip */ tmp &= ~AUREON_AC97_DATA_MASK; tmp |= (val >> 8) & AUREON_AC97_DATA_MASK; @@ -282,7 +287,7 @@ static void aureon_ac97_write(struct snd_ice1712 *ice, unsigned short reg, tmp &= ~AUREON_AC97_DATA_HIGH; snd_ice1712_gpio_write(ice, tmp); udelay(10); - + /* Instruct XILINX chip to parse the data to the STAC9744 chip */ tmp |= AUREON_AC97_COMMIT; snd_ice1712_gpio_write(ice, tmp); @@ -290,7 +295,7 @@ static void aureon_ac97_write(struct snd_ice1712 *ice, unsigned short reg, tmp &= ~AUREON_AC97_COMMIT; snd_ice1712_gpio_write(ice, tmp); udelay(10); - + /* Store the data in out private buffer */ spec->stac9744[(reg & 0x7F) >> 1] = val; } @@ -304,7 +309,7 @@ static unsigned short aureon_ac97_read(struct snd_ice1712 *ice, unsigned short r /* * Initialize STAC9744 chip */ -static int aureon_ac97_init (struct snd_ice1712 *ice) +static int aureon_ac97_init(struct snd_ice1712 *ice) { struct aureon_spec *spec = ice->spec; int i; @@ -335,20 +340,21 @@ static int aureon_ac97_init (struct snd_ice1712 *ice) tmp = (snd_ice1712_gpio_read(ice) | AUREON_AC97_RESET) & ~AUREON_AC97_DATA_MASK; snd_ice1712_gpio_write(ice, tmp); udelay(3); - + tmp &= ~AUREON_AC97_RESET; snd_ice1712_gpio_write(ice, tmp); udelay(3); - + tmp |= AUREON_AC97_RESET; snd_ice1712_gpio_write(ice, tmp); udelay(3); - + memset(&spec->stac9744, 0, sizeof(spec->stac9744)); - for (i=0; ac97_defaults[i] != (unsigned short)-1; i+=2) + for (i = 0; ac97_defaults[i] != (unsigned short)-1; i += 2) spec->stac9744[(ac97_defaults[i]) >> 1] = ac97_defaults[i+1]; - - aureon_ac97_write(ice, AC97_MASTER, 0x0000); // Unmute AC'97 master volume permanently - muting is done by WM8770 + + /* Unmute AC'97 master volume permanently - muting is done by WM8770 */ + aureon_ac97_write(ice, AC97_MASTER, 0x0000); return 0; } @@ -388,7 +394,7 @@ static int aureon_ac97_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ele struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); unsigned short ovol, nvol; int change; - + snd_ice1712_save_gpio_status(ice); ovol = aureon_ac97_read(ice, kcontrol->private_value & 0x7F); @@ -396,13 +402,14 @@ static int aureon_ac97_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ele if (kcontrol->private_value & AUREON_AC97_STEREO) nvol |= ((0x1F - ucontrol->value.integer.value[1]) << 8) & 0x1F00; nvol |= ovol & ~0x1F1F; - - if ((change = (ovol != nvol))) + + change = (ovol != nvol); + if (change) aureon_ac97_write(ice, kcontrol->private_value & 0x7F, nvol); snd_ice1712_restore_gpio_status(ice); - return change; + return change; } /* @@ -416,7 +423,8 @@ static int aureon_ac97_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_el mutex_lock(&ice->gpio_mutex); - ucontrol->value.integer.value[0] = aureon_ac97_read(ice, kcontrol->private_value & 0x7F) & 0x8000 ? 0 : 1; + ucontrol->value.integer.value[0] = aureon_ac97_read(ice, + kcontrol->private_value & 0x7F) & 0x8000 ? 0 : 1; mutex_unlock(&ice->gpio_mutex); return 0; @@ -429,13 +437,14 @@ static int aureon_ac97_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_el int change; snd_ice1712_save_gpio_status(ice); - + ovol = aureon_ac97_read(ice, kcontrol->private_value & 0x7F); - nvol = (ucontrol->value.integer.value[0] ? 0x0000 : 0x8000) | (ovol & ~ 0x8000); - - if ((change = (ovol != nvol))) + nvol = (ucontrol->value.integer.value[0] ? 0x0000 : 0x8000) | (ovol & ~0x8000); + + change = (ovol != nvol); + if (change) aureon_ac97_write(ice, kcontrol->private_value & 0x7F, nvol); - + snd_ice1712_restore_gpio_status(ice); return change; @@ -465,13 +474,14 @@ static int aureon_ac97_micboost_put(struct snd_kcontrol *kcontrol, struct snd_ct int change; snd_ice1712_save_gpio_status(ice); - + ovol = aureon_ac97_read(ice, AC97_MIC); nvol = (ucontrol->value.integer.value[0] ? 0x0000 : 0x0020) | (ovol & ~0x0020); - - if ((change = (ovol != nvol))) + + change = (ovol != nvol); + if (change) aureon_ac97_write(ice, AC97_MIC, nvol); - + snd_ice1712_restore_gpio_status(ice); return change; @@ -493,16 +503,15 @@ static void aureon_spi_write(struct snd_ice1712 *ice, unsigned int cs, unsigned snd_ice1712_gpio_set_mask(ice, ~(PRODIGY_SPI_MOSI|PRODIGY_SPI_CLK|PRODIGY_WM_CS)); mosi = PRODIGY_SPI_MOSI; clk = PRODIGY_SPI_CLK; - } - else { + } else { snd_ice1712_gpio_set_mask(ice, ~(AUREON_WM_RW|AUREON_SPI_MOSI|AUREON_SPI_CLK| AUREON_WM_CS|AUREON_CS8415_CS)); mosi = AUREON_SPI_MOSI; clk = AUREON_SPI_CLK; - + tmp |= AUREON_WM_RW; } - + tmp &= ~cs; snd_ice1712_gpio_write(ice, tmp); udelay(1); @@ -534,7 +543,9 @@ static void aureon_spi_write(struct snd_ice1712 *ice, unsigned int cs, unsigned /* * Read data in SPI mode */ -static void aureon_spi_read(struct snd_ice1712 *ice, unsigned int cs, unsigned int data, int bits, unsigned char *buffer, int size) { +static void aureon_spi_read(struct snd_ice1712 *ice, unsigned int cs, + unsigned int data, int bits, unsigned char *buffer, int size) +{ int i, j; unsigned int tmp; @@ -544,7 +555,7 @@ static void aureon_spi_read(struct snd_ice1712 *ice, unsigned int cs, unsigned i snd_ice1712_gpio_write(ice, tmp); udelay(1); - for (i=bits-1; i>=0; i--) { + for (i = bits-1; i >= 0; i--) { if (data & (1 << i)) tmp |= AUREON_SPI_MOSI; else @@ -561,9 +572,9 @@ static void aureon_spi_read(struct snd_ice1712 *ice, unsigned int cs, unsigned i udelay(1); } - for (j=0; j<size; j++) { + for (j = 0; j < size; j++) { unsigned char outdata = 0; - for (i=7; i>=0; i--) { + for (i = 7; i >= 0; i--) { tmp = snd_ice1712_gpio_read(ice); outdata <<= 1; outdata |= (tmp & AUREON_SPI_MISO) ? 1 : 0; @@ -584,19 +595,24 @@ static void aureon_spi_read(struct snd_ice1712 *ice, unsigned int cs, unsigned i snd_ice1712_gpio_write(ice, tmp); } -static unsigned char aureon_cs8415_get(struct snd_ice1712 *ice, int reg) { +static unsigned char aureon_cs8415_get(struct snd_ice1712 *ice, int reg) +{ unsigned char val; aureon_spi_write(ice, AUREON_CS8415_CS, 0x2000 | reg, 16); aureon_spi_read(ice, AUREON_CS8415_CS, 0x21, 8, &val, 1); return val; } -static void aureon_cs8415_read(struct snd_ice1712 *ice, int reg, unsigned char *buffer, int size) { +static void aureon_cs8415_read(struct snd_ice1712 *ice, int reg, + unsigned char *buffer, int size) +{ aureon_spi_write(ice, AUREON_CS8415_CS, 0x2000 | reg, 16); aureon_spi_read(ice, AUREON_CS8415_CS, 0x21, 8, buffer, size); } -static void aureon_cs8415_put(struct snd_ice1712 *ice, int reg, unsigned char val) { +static void aureon_cs8415_put(struct snd_ice1712 *ice, int reg, + unsigned char val) +{ aureon_spi_write(ice, AUREON_CS8415_CS, 0x200000 | (reg << 8) | val, 24); } @@ -654,18 +670,20 @@ static int aureon_ac97_mmute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_e return 0; } -static int aureon_ac97_mmute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { +static int aureon_ac97_mmute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); unsigned short ovol, nvol; int change; - + snd_ice1712_save_gpio_status(ice); - + ovol = wm_get(ice, WM_OUT_MUX1); nvol = (ovol & ~0x02) | (ucontrol->value.integer.value[0] ? 0x02 : 0x00); - if ((change = (ovol != nvol))) + change = (ovol != nvol); + if (change) wm_put(ice, WM_OUT_MUX1, nvol); - + snd_ice1712_restore_gpio_status(ice); return change; @@ -702,12 +720,12 @@ static const unsigned char wm_vol[256] = { static void wm_set_vol(struct snd_ice1712 *ice, unsigned int index, unsigned short vol, unsigned short master) { unsigned char nvol; - + if ((master & WM_VOL_MUTE) || (vol & WM_VOL_MUTE)) nvol = 0; else nvol = 127 - wm_vol[(((vol & ~WM_VOL_MUTE) * (master & ~WM_VOL_MUTE)) / 127) & WM_VOL_MAX]; - + wm_put(ice, index, nvol); wm_put_nocache(ice, index, 0x180 | nvol); } @@ -736,7 +754,8 @@ static int wm_pcm_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_va snd_ice1712_save_gpio_status(ice); oval = wm_get(ice, WM_MUTE); nval = (oval & ~0x10) | (ucontrol->value.integer.value[0] ? 0 : 0x10); - if ((change = (nval != oval))) + change = (oval != nval); + if (change) wm_put(ice, WM_MUTE, nval); snd_ice1712_restore_gpio_status(ice); @@ -760,7 +779,7 @@ static int wm_master_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); struct aureon_spec *spec = ice->spec; int i; - for (i=0; i<2; i++) + for (i = 0; i < 2; i++) ucontrol->value.integer.value[i] = spec->master[i] & ~WM_VOL_MUTE; return 0; @@ -849,7 +868,8 @@ static int wm_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value * /* * WM8770 mute control */ -static int wm_mute_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { +static int wm_mute_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; uinfo->count = kcontrol->private_value >> 8; uinfo->value.integer.min = 0; @@ -862,7 +882,7 @@ static int wm_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); struct aureon_spec *spec = ice->spec; int voices, ofs, i; - + voices = kcontrol->private_value >> 8; ofs = kcontrol->private_value & 0xFF; @@ -907,7 +927,7 @@ static int wm_master_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); struct aureon_spec *spec = ice->spec; - + ucontrol->value.integer.value[0] = (spec->master[0] & WM_VOL_MUTE) ? 0 : 1; ucontrol->value.integer.value[1] = @@ -1083,21 +1103,21 @@ static int wm_adc_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_val static int wm_adc_mux_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static const char * const texts[] = { - "CD", //AIN1 - "Aux", //AIN2 - "Line", //AIN3 - "Mic", //AIN4 - "AC97" //AIN5 + "CD", /* AIN1 */ + "Aux", /* AIN2 */ + "Line", /* AIN3 */ + "Mic", /* AIN4 */ + "AC97" /* AIN5 */ }; static const char * const universe_texts[] = { - "Aux1", //AIN1 - "CD", //AIN2 - "Phono", //AIN3 - "Line", //AIN4 - "Aux2", //AIN5 - "Mic", //AIN6 - "Aux3", //AIN7 - "AC97" //AIN8 + "Aux1", /* AIN1 */ + "CD", /* AIN2 */ + "Phono", /* AIN3 */ + "Line", /* AIN4 */ + "Aux2", /* AIN5 */ + "Mic", /* AIN6 */ + "Aux3", /* AIN7 */ + "AC97" /* AIN8 */ }; struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); @@ -1108,8 +1128,7 @@ static int wm_adc_mux_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_in if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; strcpy(uinfo->value.enumerated.name, universe_texts[uinfo->value.enumerated.item]); - } - else { + } else { uinfo->value.enumerated.items = 5; if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; @@ -1156,8 +1175,8 @@ static int aureon_cs8415_mux_info(struct snd_kcontrol *kcontrol, struct snd_ctl_ { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); static const char * const aureon_texts[] = { - "CD", //RXP0 - "Optical" //RXP1 + "CD", /* RXP0 */ + "Optical" /* RXP1 */ }; static const char * const prodigy_texts[] = { "CD", @@ -1180,10 +1199,10 @@ static int aureon_cs8415_mux_get(struct snd_kcontrol *kcontrol, struct snd_ctl_e struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); struct aureon_spec *spec = ice->spec; - //snd_ice1712_save_gpio_status(ice); - //val = aureon_cs8415_get(ice, CS8415_CTRL2); + /* snd_ice1712_save_gpio_status(ice); */ + /* val = aureon_cs8415_get(ice, CS8415_CTRL2); */ ucontrol->value.enumerated.item[0] = spec->cs8415_mux; - //snd_ice1712_restore_gpio_status(ice); + /* snd_ice1712_restore_gpio_status(ice); */ return 0; } @@ -1206,7 +1225,7 @@ static int aureon_cs8415_mux_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e return change; } -static int aureon_cs8415_rate_info (struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +static int aureon_cs8415_rate_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 1; @@ -1215,7 +1234,7 @@ static int aureon_cs8415_rate_info (struct snd_kcontrol *kcontrol, struct snd_ct return 0; } -static int aureon_cs8415_rate_get (struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int aureon_cs8415_rate_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); unsigned char ratio; @@ -1229,7 +1248,7 @@ static int aureon_cs8415_rate_get (struct snd_kcontrol *kcontrol, struct snd_ctl */ #define aureon_cs8415_mute_info snd_ctl_boolean_mono_info -static int aureon_cs8415_mute_get (struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int aureon_cs8415_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); snd_ice1712_save_gpio_status(ice); @@ -1238,7 +1257,7 @@ static int aureon_cs8415_mute_get (struct snd_kcontrol *kcontrol, struct snd_ctl return 0; } -static int aureon_cs8415_mute_put (struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int aureon_cs8415_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); unsigned char oval, nval; @@ -1249,7 +1268,8 @@ static int aureon_cs8415_mute_put (struct snd_kcontrol *kcontrol, struct snd_ctl nval = oval & ~0x20; else nval = oval | 0x20; - if ((change = (oval != nval))) + change = (oval != nval); + if (change) aureon_cs8415_put(ice, CS8415_CTRL1, nval); snd_ice1712_restore_gpio_status(ice); return change; @@ -1258,15 +1278,17 @@ static int aureon_cs8415_mute_put (struct snd_kcontrol *kcontrol, struct snd_ctl /* * CS8415A Q-Sub info */ -static int aureon_cs8415_qsub_info (struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { +static int aureon_cs8415_qsub_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; uinfo->count = 10; return 0; } -static int aureon_cs8415_qsub_get (struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { +static int aureon_cs8415_qsub_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - + snd_ice1712_save_gpio_status(ice); aureon_cs8415_read(ice, CS8415_QSUB, ucontrol->value.bytes.data, 10); snd_ice1712_restore_gpio_status(ice); @@ -1274,18 +1296,21 @@ static int aureon_cs8415_qsub_get (struct snd_kcontrol *kcontrol, struct snd_ctl return 0; } -static int aureon_cs8415_spdif_info (struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { +static int aureon_cs8415_spdif_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958; uinfo->count = 1; return 0; } -static int aureon_cs8415_mask_get (struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { +static int aureon_cs8415_mask_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ memset(ucontrol->value.iec958.status, 0xFF, 24); return 0; } -static int aureon_cs8415_spdif_get (struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { +static int aureon_cs8415_spdif_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); snd_ice1712_save_gpio_status(ice); @@ -1311,9 +1336,9 @@ static int aureon_set_headphone_amp(struct snd_ice1712 *ice, int enable) else if (ice->eeprom.subvendor != VT1724_SUBDEVICE_PRODIGY71LT && ice->eeprom.subvendor != VT1724_SUBDEVICE_PRODIGY71XT) - tmp &= ~ AUREON_HP_SEL; + tmp &= ~AUREON_HP_SEL; else - tmp &= ~ PRODIGY_HP_SEL; + tmp &= ~PRODIGY_HP_SEL; if (tmp != tmp2) { snd_ice1712_gpio_write(ice, tmp); return 1; @@ -1325,7 +1350,7 @@ static int aureon_get_headphone_amp(struct snd_ice1712 *ice) { unsigned int tmp = snd_ice1712_gpio_read(ice); - return ( tmp & AUREON_HP_SEL )!= 0; + return (tmp & AUREON_HP_SEL) != 0; } #define aureon_hpamp_info snd_ctl_boolean_mono_info @@ -1343,7 +1368,7 @@ static int aureon_hpamp_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - return aureon_set_headphone_amp(ice,ucontrol->value.integer.value[0]); + return aureon_set_headphone_amp(ice, ucontrol->value.integer.value[0]); } /* @@ -1390,7 +1415,7 @@ static int aureon_oversampling_info(struct snd_kcontrol *k, struct snd_ctl_elem_ uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return 0; } static int aureon_oversampling_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1434,7 +1459,7 @@ static struct snd_kcontrol_new aureon_dac_controls[] __devinitdata = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ), + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Master Playback Volume", .info = wm_master_vol_info, .get = wm_master_vol_get, @@ -1452,7 +1477,7 @@ static struct snd_kcontrol_new aureon_dac_controls[] __devinitdata = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ), + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Front Playback Volume", .info = wm_vol_info, .get = wm_vol_get, @@ -1471,7 +1496,7 @@ static struct snd_kcontrol_new aureon_dac_controls[] __devinitdata = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ), + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Rear Playback Volume", .info = wm_vol_info, .get = wm_vol_get, @@ -1490,7 +1515,7 @@ static struct snd_kcontrol_new aureon_dac_controls[] __devinitdata = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ), + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Center Playback Volume", .info = wm_vol_info, .get = wm_vol_get, @@ -1509,7 +1534,7 @@ static struct snd_kcontrol_new aureon_dac_controls[] __devinitdata = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ), + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "LFE Playback Volume", .info = wm_vol_info, .get = wm_vol_get, @@ -1528,7 +1553,7 @@ static struct snd_kcontrol_new aureon_dac_controls[] __devinitdata = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ), + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Side Playback Volume", .info = wm_vol_info, .get = wm_vol_get, @@ -1539,23 +1564,23 @@ static struct snd_kcontrol_new aureon_dac_controls[] __devinitdata = { }; static struct snd_kcontrol_new wm_controls[] __devinitdata = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "PCM Playback Switch", .info = wm_pcm_mute_info, .get = wm_pcm_mute_get, .put = wm_pcm_mute_put - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ), + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "PCM Playback Volume", .info = wm_pcm_vol_info, .get = wm_pcm_vol_get, .put = wm_pcm_vol_put, .tlv = { .p = db_scale_wm_pcm } - }, + }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Capture Switch", @@ -1566,7 +1591,7 @@ static struct snd_kcontrol_new wm_controls[] __devinitdata = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ), + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Capture Volume", .info = wm_adc_vol_info, .get = wm_adc_vol_get, @@ -1605,232 +1630,232 @@ static struct snd_kcontrol_new wm_controls[] __devinitdata = { }; static struct snd_kcontrol_new ac97_controls[] __devinitdata = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "AC97 Playback Switch", .info = aureon_ac97_mmute_info, .get = aureon_ac97_mmute_get, .put = aureon_ac97_mmute_put, .private_value = AC97_MASTER - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ), - .name = "AC97 Playback Volume", - .info = aureon_ac97_vol_info, - .get = aureon_ac97_vol_get, - .put = aureon_ac97_vol_put, - .private_value = AC97_MASTER|AUREON_AC97_STEREO, + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "AC97 Playback Volume", + .info = aureon_ac97_vol_info, + .get = aureon_ac97_vol_get, + .put = aureon_ac97_vol_put, + .private_value = AC97_MASTER|AUREON_AC97_STEREO, .tlv = { .p = db_scale_ac97_master } - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "CD Playback Switch", - .info = aureon_ac97_mute_info, - .get = aureon_ac97_mute_get, - .put = aureon_ac97_mute_put, - .private_value = AC97_CD - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "CD Playback Switch", + .info = aureon_ac97_mute_info, + .get = aureon_ac97_mute_get, + .put = aureon_ac97_mute_put, + .private_value = AC97_CD + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ), - .name = "CD Playback Volume", - .info = aureon_ac97_vol_info, - .get = aureon_ac97_vol_get, - .put = aureon_ac97_vol_put, - .private_value = AC97_CD|AUREON_AC97_STEREO, + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "CD Playback Volume", + .info = aureon_ac97_vol_info, + .get = aureon_ac97_vol_get, + .put = aureon_ac97_vol_put, + .private_value = AC97_CD|AUREON_AC97_STEREO, .tlv = { .p = db_scale_ac97_gain } - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Aux Playback Switch", - .info = aureon_ac97_mute_info, - .get = aureon_ac97_mute_get, - .put = aureon_ac97_mute_put, - .private_value = AC97_AUX, - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Aux Playback Switch", + .info = aureon_ac97_mute_info, + .get = aureon_ac97_mute_get, + .put = aureon_ac97_mute_put, + .private_value = AC97_AUX, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ), - .name = "Aux Playback Volume", - .info = aureon_ac97_vol_info, - .get = aureon_ac97_vol_get, - .put = aureon_ac97_vol_put, - .private_value = AC97_AUX|AUREON_AC97_STEREO, + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "Aux Playback Volume", + .info = aureon_ac97_vol_info, + .get = aureon_ac97_vol_get, + .put = aureon_ac97_vol_put, + .private_value = AC97_AUX|AUREON_AC97_STEREO, .tlv = { .p = db_scale_ac97_gain } - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Line Playback Switch", - .info = aureon_ac97_mute_info, - .get = aureon_ac97_mute_get, - .put = aureon_ac97_mute_put, - .private_value = AC97_LINE - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Line Playback Switch", + .info = aureon_ac97_mute_info, + .get = aureon_ac97_mute_get, + .put = aureon_ac97_mute_put, + .private_value = AC97_LINE + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ), - .name = "Line Playback Volume", - .info = aureon_ac97_vol_info, - .get = aureon_ac97_vol_get, - .put = aureon_ac97_vol_put, - .private_value = AC97_LINE|AUREON_AC97_STEREO, + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "Line Playback Volume", + .info = aureon_ac97_vol_info, + .get = aureon_ac97_vol_get, + .put = aureon_ac97_vol_put, + .private_value = AC97_LINE|AUREON_AC97_STEREO, .tlv = { .p = db_scale_ac97_gain } - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Mic Playback Switch", - .info = aureon_ac97_mute_info, - .get = aureon_ac97_mute_get, - .put = aureon_ac97_mute_put, - .private_value = AC97_MIC - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mic Playback Switch", + .info = aureon_ac97_mute_info, + .get = aureon_ac97_mute_get, + .put = aureon_ac97_mute_put, + .private_value = AC97_MIC + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ), - .name = "Mic Playback Volume", - .info = aureon_ac97_vol_info, - .get = aureon_ac97_vol_get, - .put = aureon_ac97_vol_put, - .private_value = AC97_MIC, + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "Mic Playback Volume", + .info = aureon_ac97_vol_info, + .get = aureon_ac97_vol_get, + .put = aureon_ac97_vol_put, + .private_value = AC97_MIC, .tlv = { .p = db_scale_ac97_gain } - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Mic Boost (+20dB)", - .info = aureon_ac97_micboost_info, - .get = aureon_ac97_micboost_get, - .put = aureon_ac97_micboost_put - } + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mic Boost (+20dB)", + .info = aureon_ac97_micboost_info, + .get = aureon_ac97_micboost_get, + .put = aureon_ac97_micboost_put + } }; static struct snd_kcontrol_new universe_ac97_controls[] __devinitdata = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "AC97 Playback Switch", .info = aureon_ac97_mmute_info, .get = aureon_ac97_mmute_get, .put = aureon_ac97_mmute_put, .private_value = AC97_MASTER - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ), - .name = "AC97 Playback Volume", - .info = aureon_ac97_vol_info, - .get = aureon_ac97_vol_get, - .put = aureon_ac97_vol_put, - .private_value = AC97_MASTER|AUREON_AC97_STEREO, + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "AC97 Playback Volume", + .info = aureon_ac97_vol_info, + .get = aureon_ac97_vol_get, + .put = aureon_ac97_vol_put, + .private_value = AC97_MASTER|AUREON_AC97_STEREO, .tlv = { .p = db_scale_ac97_master } - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "CD Playback Switch", - .info = aureon_ac97_mute_info, - .get = aureon_ac97_mute_get, - .put = aureon_ac97_mute_put, - .private_value = AC97_AUX - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "CD Playback Switch", + .info = aureon_ac97_mute_info, + .get = aureon_ac97_mute_get, + .put = aureon_ac97_mute_put, + .private_value = AC97_AUX + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ), - .name = "CD Playback Volume", - .info = aureon_ac97_vol_info, - .get = aureon_ac97_vol_get, - .put = aureon_ac97_vol_put, - .private_value = AC97_AUX|AUREON_AC97_STEREO, + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "CD Playback Volume", + .info = aureon_ac97_vol_info, + .get = aureon_ac97_vol_get, + .put = aureon_ac97_vol_put, + .private_value = AC97_AUX|AUREON_AC97_STEREO, .tlv = { .p = db_scale_ac97_gain } - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Phono Playback Switch", - .info = aureon_ac97_mute_info, - .get = aureon_ac97_mute_get, - .put = aureon_ac97_mute_put, - .private_value = AC97_CD - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Phono Playback Switch", + .info = aureon_ac97_mute_info, + .get = aureon_ac97_mute_get, + .put = aureon_ac97_mute_put, + .private_value = AC97_CD + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ), - .name = "Phono Playback Volume", - .info = aureon_ac97_vol_info, - .get = aureon_ac97_vol_get, - .put = aureon_ac97_vol_put, - .private_value = AC97_CD|AUREON_AC97_STEREO, + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "Phono Playback Volume", + .info = aureon_ac97_vol_info, + .get = aureon_ac97_vol_get, + .put = aureon_ac97_vol_put, + .private_value = AC97_CD|AUREON_AC97_STEREO, .tlv = { .p = db_scale_ac97_gain } - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Line Playback Switch", - .info = aureon_ac97_mute_info, - .get = aureon_ac97_mute_get, - .put = aureon_ac97_mute_put, - .private_value = AC97_LINE - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Line Playback Switch", + .info = aureon_ac97_mute_info, + .get = aureon_ac97_mute_get, + .put = aureon_ac97_mute_put, + .private_value = AC97_LINE + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ), - .name = "Line Playback Volume", - .info = aureon_ac97_vol_info, - .get = aureon_ac97_vol_get, - .put = aureon_ac97_vol_put, - .private_value = AC97_LINE|AUREON_AC97_STEREO, + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "Line Playback Volume", + .info = aureon_ac97_vol_info, + .get = aureon_ac97_vol_get, + .put = aureon_ac97_vol_put, + .private_value = AC97_LINE|AUREON_AC97_STEREO, .tlv = { .p = db_scale_ac97_gain } - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Mic Playback Switch", - .info = aureon_ac97_mute_info, - .get = aureon_ac97_mute_get, - .put = aureon_ac97_mute_put, - .private_value = AC97_MIC - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mic Playback Switch", + .info = aureon_ac97_mute_info, + .get = aureon_ac97_mute_get, + .put = aureon_ac97_mute_put, + .private_value = AC97_MIC + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ), - .name = "Mic Playback Volume", - .info = aureon_ac97_vol_info, - .get = aureon_ac97_vol_get, - .put = aureon_ac97_vol_put, - .private_value = AC97_MIC, + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "Mic Playback Volume", + .info = aureon_ac97_vol_info, + .get = aureon_ac97_vol_get, + .put = aureon_ac97_vol_put, + .private_value = AC97_MIC, .tlv = { .p = db_scale_ac97_gain } - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Mic Boost (+20dB)", - .info = aureon_ac97_micboost_info, - .get = aureon_ac97_micboost_get, - .put = aureon_ac97_micboost_put - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Aux Playback Switch", - .info = aureon_ac97_mute_info, - .get = aureon_ac97_mute_get, - .put = aureon_ac97_mute_put, - .private_value = AC97_VIDEO, - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mic Boost (+20dB)", + .info = aureon_ac97_micboost_info, + .get = aureon_ac97_micboost_get, + .put = aureon_ac97_micboost_put + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Aux Playback Switch", + .info = aureon_ac97_mute_info, + .get = aureon_ac97_mute_get, + .put = aureon_ac97_mute_put, + .private_value = AC97_VIDEO, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ), - .name = "Aux Playback Volume", - .info = aureon_ac97_vol_info, - .get = aureon_ac97_vol_get, - .put = aureon_ac97_vol_put, - .private_value = AC97_VIDEO|AUREON_AC97_STEREO, + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "Aux Playback Volume", + .info = aureon_ac97_vol_info, + .get = aureon_ac97_vol_get, + .put = aureon_ac97_vol_put, + .private_value = AC97_VIDEO|AUREON_AC97_STEREO, .tlv = { .p = db_scale_ac97_gain } - }, + }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Aux Source", @@ -1844,43 +1869,43 @@ static struct snd_kcontrol_new universe_ac97_controls[] __devinitdata = { static struct snd_kcontrol_new cs8415_controls[] __devinitdata = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",CAPTURE,SWITCH), + .name = SNDRV_CTL_NAME_IEC958("", CAPTURE, SWITCH), .info = aureon_cs8415_mute_info, .get = aureon_cs8415_mute_get, .put = aureon_cs8415_mute_put }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",CAPTURE,NONE) "Source", + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SNDRV_CTL_NAME_IEC958("", CAPTURE, NONE) "Source", .info = aureon_cs8415_mux_info, .get = aureon_cs8415_mux_get, .put = aureon_cs8415_mux_put, }, { .iface = SNDRV_CTL_ELEM_IFACE_PCM, - .name = SNDRV_CTL_NAME_IEC958("Q-subcode ",CAPTURE,DEFAULT), + .name = SNDRV_CTL_NAME_IEC958("Q-subcode ", CAPTURE, DEFAULT), .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, .info = aureon_cs8415_qsub_info, .get = aureon_cs8415_qsub_get, }, { .iface = SNDRV_CTL_ELEM_IFACE_PCM, - .name = SNDRV_CTL_NAME_IEC958("",CAPTURE,MASK), + .name = SNDRV_CTL_NAME_IEC958("", CAPTURE, MASK), .access = SNDRV_CTL_ELEM_ACCESS_READ, .info = aureon_cs8415_spdif_info, .get = aureon_cs8415_mask_get }, { .iface = SNDRV_CTL_ELEM_IFACE_PCM, - .name = SNDRV_CTL_NAME_IEC958("",CAPTURE,DEFAULT), + .name = SNDRV_CTL_NAME_IEC958("", CAPTURE, DEFAULT), .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, .info = aureon_cs8415_spdif_info, .get = aureon_cs8415_spdif_get }, { .iface = SNDRV_CTL_ELEM_IFACE_PCM, - .name = SNDRV_CTL_NAME_IEC958("",CAPTURE,NONE) "Rate", - .access =SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .name = SNDRV_CTL_NAME_IEC958("", CAPTURE, NONE) "Rate", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, .info = aureon_cs8415_rate_info, .get = aureon_cs8415_rate_get } @@ -1905,15 +1930,14 @@ static int __devinit aureon_add_controls(struct snd_ice1712 *ice) if (err < 0) return err; } - + if (ice->eeprom.subvendor == VT1724_SUBDEVICE_AUREON71_UNIVERSE) { for (i = 0; i < ARRAY_SIZE(universe_ac97_controls); i++) { err = snd_ctl_add(ice->card, snd_ctl_new1(&universe_ac97_controls[i], ice)); if (err < 0) return err; } - } - else if (ice->eeprom.subvendor != VT1724_SUBDEVICE_PRODIGY71LT && + } else if (ice->eeprom.subvendor != VT1724_SUBDEVICE_PRODIGY71LT && ice->eeprom.subvendor != VT1724_SUBDEVICE_PRODIGY71XT) { for (i = 0; i < ARRAY_SIZE(ac97_controls); i++) { err = snd_ctl_add(ice->card, snd_ctl_new1(&ac97_controls[i], ice)); @@ -1932,7 +1956,7 @@ static int __devinit aureon_add_controls(struct snd_ice1712 *ice) else if ((id & 0x0F) != 0x01) snd_printk(KERN_INFO "Detected unsupported CS8415 rev. (%c)\n", (char)((id & 0x0F) + 'A' - 1)); else { - for (i = 0; i< ARRAY_SIZE(cs8415_controls); i++) { + for (i = 0; i < ARRAY_SIZE(cs8415_controls); i++) { struct snd_kcontrol *kctl; err = snd_ctl_add(ice->card, (kctl = snd_ctl_new1(&cs8415_controls[i], ice))); if (err < 0) @@ -1943,7 +1967,7 @@ static int __devinit aureon_add_controls(struct snd_ice1712 *ice) } snd_ice1712_restore_gpio_status(ice); } - + return 0; } @@ -2059,11 +2083,12 @@ static int __devinit aureon_init(struct snd_ice1712 *ice) /* to remeber the register values of CS8415 */ ice->akm = kzalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL); - if (! ice->akm) + if (!ice->akm) return -ENOMEM; ice->akm_codecs = 1; - - if ((err = aureon_ac97_init(ice)) != 0) + + err = aureon_ac97_init(ice); + if (err != 0) return err; snd_ice1712_gpio_set_dir(ice, 0x5fffff); /* fix this for the time being */ @@ -2086,7 +2111,7 @@ static int __devinit aureon_init(struct snd_ice1712 *ice) /* initialize WM8770 codec */ if (ice->eeprom.subvendor == VT1724_SUBDEVICE_PRODIGY71 || ice->eeprom.subvendor == VT1724_SUBDEVICE_PRODIGY71LT || - ice->eeprom.subvendor == VT1724_SUBDEVICE_PRODIGY71XT) + ice->eeprom.subvendor == VT1724_SUBDEVICE_PRODIGY71XT) p = wm_inits_prodigy; else p = wm_inits_aureon; @@ -2105,10 +2130,10 @@ static int __devinit aureon_init(struct snd_ice1712 *ice) snd_ice1712_restore_gpio_status(ice); - /* initialize PCA9554 pin directions & set default input*/ + /* initialize PCA9554 pin directions & set default input */ aureon_pca9554_write(ice, PCA9554_DIR, 0x00); aureon_pca9554_write(ice, PCA9554_OUT, 0x00); /* internal AUX */ - + spec->master[0] = WM_VOL_MUTE; spec->master[1] = WM_VOL_MUTE; for (i = 0; i < ice->num_total_dacs; i++) { @@ -2215,12 +2240,12 @@ struct snd_ice1712_card_info snd_vt1724_aureon_cards[] __devinitdata = { .eeprom_data = aureon71_eeprom, .driver = "Aureon71", }, - { - .subvendor = VT1724_SUBDEVICE_AUREON71_UNIVERSE, - .name = "Terratec Aureon 7.1-Universe", + { + .subvendor = VT1724_SUBDEVICE_AUREON71_UNIVERSE, + .name = "Terratec Aureon 7.1-Universe", .model = "universe", - .chip_init = aureon_init, - .build_controls = aureon_add_controls, + .chip_init = aureon_init, + .build_controls = aureon_add_controls, .eeprom_size = sizeof(aureon71_universe_eeprom), .eeprom_data = aureon71_universe_eeprom, .driver = "Aureon71Univ", /* keep in 15 letters */

3 2

[alsa-devel] [PATCH] hda: multiple SPDIF outputs support
by Matthew Ranostay 07 Sep '08

07 Sep '08

Added support for outputting a stream to multiple SPDIF outs on supporting codecs. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff --git a/pci/hda/hda_codec.c b/pci/hda/hda_codec.c index 4f32911..696d77e 100644 --- a/pci/hda/hda_codec.c +++ b/pci/hda/hda_codec.c @@ -1454,12 +1454,22 @@ static int snd_hda_spdif_default_put(struct snd_kcontrol *kcontrol, codec->spdif_ctls = val; if (change) { + hda_nid_t *d; snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, val & 0xff); snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_2, val >> 8); + + for (d = codec->slave_dig_outs; *d; d++) { + snd_hda_codec_write_cache(codec, *d, 0, + AC_VERB_SET_DIGI_CONVERT_1, + val & 0xff); + snd_hda_codec_write_cache(codec, *d, 0, + AC_VERB_SET_DIGI_CONVERT_2, + val >> 8); + } } mutex_unlock(&codec->spdif_mutex); @@ -1491,10 +1501,16 @@ static int snd_hda_spdif_out_switch_put(struct snd_kcontrol *kcontrol, val |= AC_DIG1_ENABLE; change = codec->spdif_ctls != val; if (change) { + hda_nid_t *d; codec->spdif_ctls = val; snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, val & 0xff); + + for (d = codec->slave_dig_outs; *d; d++) + snd_hda_codec_write_cache(codec, *d, 0, + AC_VERB_SET_DIGI_CONVERT_1, + val & 0xff); /* unmute amp switch (if any) */ if ((get_wcaps(codec, nid) & AC_WCAP_OUT_AMP) && (val & AC_DIG1_ENABLE)) @@ -1643,9 +1659,14 @@ static int snd_hda_spdif_in_switch_put(struct snd_kcontrol *kcontrol, mutex_lock(&codec->spdif_mutex); change = codec->spdif_in_enable != val; if (change) { + hda_nid_t *d; codec->spdif_in_enable = val; snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, val); + + for (d = codec->slave_dig_outs; *d; d++) + snd_hda_codec_write_cache(codec, *d, 0, + AC_VERB_SET_DIGI_CONVERT_1, val); } mutex_unlock(&codec->spdif_mutex); return change; @@ -2589,15 +2610,30 @@ int snd_hda_input_mux_put(struct hda_codec *codec, static void setup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid, unsigned int stream_tag, unsigned int format) { + hda_nid_t *d; + /* turn off SPDIF once; otherwise the IEC958 bits won't be updated */ - if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE)) + if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE)) { snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, + codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff); + + for (d = codec->slave_dig_outs; *d; d++) + snd_hda_codec_write(codec, *d, 0, + AC_VERB_SET_DIGI_CONVERT_1, codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff); + } snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format); /* turn on again (if needed) */ - if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE)) + if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE)) { snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, codec->spdif_ctls & 0xff); + + for (d = codec->slave_dig_outs; *d; d++) + snd_hda_codec_write(codec, *d, 0, + AC_VERB_SET_DIGI_CONVERT_1, + codec->spdif_ctls & 0xff); + } + } /* @@ -2621,8 +2657,12 @@ int snd_hda_multi_out_dig_prepare(struct hda_codec *codec, unsigned int format, struct snd_pcm_substream *substream) { + hda_nid_t *nid; mutex_lock(&codec->spdif_mutex); setup_dig_out_stream(codec, mout->dig_out_nid, stream_tag, format); + if (codec->slave_dig_outs) + for (nid = codec->slave_dig_outs; *nid; nid++) + setup_dig_out_stream(codec, *nid, stream_tag, format); mutex_unlock(&codec->spdif_mutex); return 0; } @@ -2689,6 +2729,7 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, struct snd_pcm_substream *substream) { hda_nid_t *nids = mout->dac_nids; + hda_nid_t *d; int chs = substream->runtime->channels; int i; @@ -2702,9 +2743,16 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, mout->dig_out_used = HDA_DIG_ANALOG_DUP; setup_dig_out_stream(codec, mout->dig_out_nid, stream_tag, format); + if (codec->slave_dig_outs) + for (d = codec->slave_dig_outs; *d; d++) + setup_dig_out_stream(codec, *d, + stream_tag, format); } else { mout->dig_out_used = 0; snd_hda_codec_cleanup_stream(codec, mout->dig_out_nid); + if (codec->slave_dig_outs) + for (d = codec->slave_dig_outs; *d; d++) + snd_hda_codec_cleanup_stream(codec, *d); } } mutex_unlock(&codec->spdif_mutex); diff --git a/pci/hda/hda_codec.h b/pci/hda/hda_codec.h index 780e2ff..60468f5 100644 --- a/pci/hda/hda_codec.h +++ b/pci/hda/hda_codec.h @@ -725,6 +725,7 @@ struct hda_codec { unsigned int spdif_status; /* IEC958 status bits */ unsigned short spdif_ctls; /* SPDIF control bits */ unsigned int spdif_in_enable; /* SPDIF input enable? */ + hda_nid_t *slave_dig_outs; /* optional digital out slave widgets */ struct snd_hwdep *hwdep; /* assigned hwdep device */ diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index 9968ee4..52b3579 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -229,6 +229,10 @@ static hda_nid_t stac92hd73xx_pwr_nids[8] = { 0x0f, 0x10, 0x11 }; +static hda_nid_t stac92hd73xx_slave_dig_outs[2] = { + 0x26, 0, +}; + static hda_nid_t stac92hd73xx_adc_nids[2] = { 0x1a, 0x1b }; @@ -274,6 +278,10 @@ static hda_nid_t stac92hd83xxx_pwr_nids[4] = { 0xa, 0xb, 0xd, 0xe, }; +static hda_nid_t stac92hd83xxx_slave_dig_outs[2] = { + 0x1e, 0, +}; + static unsigned int stac92hd83xxx_pwr_mapping[4] = { 0x03, 0x0c, 0x10, 0x40, }; @@ -303,6 +311,10 @@ static hda_nid_t stac92hd71bxx_dmic_nids[STAC92HD71BXX_NUM_DMICS + 1] = { 0x18, 0x19, 0 }; +static hda_nid_t stac92hd71bxx_slave_dig_outs[2] = { + 0x22, 0 +}; + static hda_nid_t stac925x_adc_nids[1] = { 0x03, }; @@ -396,9 +408,10 @@ static hda_nid_t stac92hd83xxx_pin_nids[14] = { 0x0f, 0x10, 0x11, 0x12, 0x13, 0x1d, 0x1e, 0x1f, 0x20 }; -static hda_nid_t stac92hd71bxx_pin_nids[10] = { +static hda_nid_t stac92hd71bxx_pin_nids[11] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, 0x14, 0x18, 0x19, 0x1e, + 0x1f, }; static hda_nid_t stac927x_pin_nids[14] = { @@ -1424,22 +1437,22 @@ static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { "DFI LanParty", STAC_92HD71BXX_REF), }; -static unsigned int ref92hd71bxx_pin_configs[10] = { +static unsigned int ref92hd71bxx_pin_configs[11] = { 0x02214030, 0x02a19040, 0x01a19020, 0x01014010, 0x0181302e, 0x01114010, 0x01019020, 0x90a000f0, - 0x90a000f0, 0x01452050, + 0x90a000f0, 0x01452050, 0x01452050, }; -static unsigned int dell_m4_1_pin_configs[10] = { +static unsigned int dell_m4_1_pin_configs[11] = { 0x0421101f, 0x04a11221, 0x40f000f0, 0x90170110, 0x23a1902e, 0x23014250, 0x40f000f0, 0x90a000f0, - 0x40f000f0, 0x4f0000f0, + 0x40f000f0, 0x4f0000f0, 0x4f0000f0, }; -static unsigned int dell_m4_2_pin_configs[10] = { +static unsigned int dell_m4_2_pin_configs[11] = { 0x0421101f, 0x04a11221, 0x90a70330, 0x90170110, 0x23a1902e, 0x23014250, 0x40f000f0, 0x40f000f0, - 0x40f000f0, 0x044413b0, + 0x40f000f0, 0x044413b0, 0x044413b0, }; static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = { @@ -3841,8 +3854,9 @@ again: spec->num_pwrs = ARRAY_SIZE(stac92hd73xx_pwr_nids); spec->pwr_nids = stac92hd73xx_pwr_nids; + codec->slave_dig_outs = stac92hd73xx_slave_dig_outs; - err = stac92xx_parse_auto_config(codec, 0x22, 0x24); + err = stac92xx_parse_auto_config(codec, 0x22, 0x25); if (!err) { if (spec->board_config < 0) { @@ -3883,6 +3897,7 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + codec->slave_dig_outs = stac92hd83xxx_slave_dig_outs; spec->mono_nid = 0x19; spec->digbeep_nid = 0x21; spec->dmic_nids = stac92hd83xxx_dmic_nids; @@ -4033,6 +4048,7 @@ again: case 0x111d76b5: spec->mixer = stac92hd71bxx_mixer; spec->init = stac92hd71bxx_core_init; + codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; break; case 0x111d7608: /* 5 Port with Analog Mixer */ if ((codec->revision_id & 0xf) == 0 || @@ -4065,6 +4081,7 @@ again: default: spec->mixer = stac92hd71bxx_analog_mixer; spec->init = stac92hd71bxx_analog_core_init; + codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; } spec->aloopback_mask = 0x20;

2 1

[alsa-devel] HP tx2000 and snd-hda-intel
by Kory Prince 07 Sep '08

07 Sep '08

Note: The other message did not go through because it was too big. I can email the schematics I mentioned if anyone is interested ************************************************************** Hello, I have an HP Pavillion TX2000z. This laptop has uses the snd-hda-intel and has the ALC861VD realtek codec. The existing configuration for the codec does not support the tx2000 series laptops fully. There are several configurations that have various levels of support. If you specify for the model to use the existing hp model (made for the tx1000 series) then: 1.Headphones Work 2.Speakers Work 3.Front Mic Works (the jack mic by the headphone jack) 4.The External Mics do not work (There are two mics at the top of the laptop... I am guessing left and right for a stereo Mic) 5.It seems like there is something mapped to an input that does not belong there. The simplest way to explain this is when I open audacity and set it to monitor input that there is significant white noise coming in. The Ext. Mics are not working so it can't come from there (as far as I know) and I have nothing plugged into the front Mic. I cannot find any way to mute this noise. When using the 3-stack model (with the option position_fix): 1. Headphones do not work (in either ports) 2. Speakers Work 3. Front Mic Works 4. The external mics work 5. Same problem as expressed above All other models do not work in any significant way. Also one of the Headphone ports (there are two on the front) has Digital Out (optical) that I have no way of confirming if it works. Now, that being said, I have tried working on the driver. I have virtually no programing experience. I was able to have the driver autodetect the soundcard and apply a certain model (before it had to be applied with a module config file.) And was able to interpret some of how the driver worked. But I am so lost. I am not sure how hard it would be for a programmer to modify the driver to support the tx2000. So I want to put out a call to anyone who could help. I am fully available for testing and some debugging if someone could try and help. I am uploading the codec file and the codec schematic ( I don't think this is actually accurate, but It may help...) Also here is the lspci -nn for the soundcard and other info I found 00:10.1 Audio device [0403]: nVidia Corporation MCP51 High Definition Audio [10de:026c] (rev a2) lshw *-multimedia description: Audio device product: MCP51 High Definition Audio vendor: nVidia Corporation physical id: 10.1 bus info: pci@0000:00:10.1 version: a2 width: 32 bits clock: 66MHz capabilities: pm msi ht bus_master cap_list configuration: driver=HDA Intel latency=0 maxlatency=5 mingnt=2 module=snd_hda_intel If anyone could help or use additional information I would appreciate it. Thanks, Kory p.s. if this is the wrong place to post, please redirect me.

1 0

[alsa-devel] snd_pcm_writei error 128 Key has been revoked
by Paulos Siahu 07 Sep '08

07 Sep '08

Hello I'm a newbie to ALSA, in fact to sound programming at all. I tried a simple program for a beginner. When I got to this line: err = snd_pcm_writei (playback_handle, buf, bufsize); printf("Playing Error: %d %s\n", err, snd_strerror(err)); the first time it played the sound that I wanted then I Ctrl-C the app. From then on, every time I run the program it gives me: Playing Error: 128 Key has been revoked I tried to google around but no luck. Hope someone can help. Thanks Paulos Siahu

1 0

[alsa-devel] Clearification needed about structure _snd_pcm
by William Estrada 06 Sep '08

06 Sep '08

Hi group, I have found the documentation about ALSA very confusing. Where can I find the definitions about the fields in the structure_snd_pcm? Specifically Buffer, Rate, Period, Buffer Size, Period size, etc. Basically all of these: > Device: 8c297b8 > PCM Device name = 'default' PCM state = RUNNING > access type = MMAP_NONINTERLEAV format = 'U8' (Unsigned > 8 bit) > subformat = 'STD' (Standard) channels = 1 > rate = 8000 bps period time = 10000 us > period size = 80 frame buffer time = 0 us > buffer size = 65536 frames periods/buffer = 0 frames > exact rate = 8000/1 bps significant bits = 8000 > is batch = 0 is block transfer = 1 > is double = 0 is half duplex = 0 > is joint duplex = 0 can overrange = 0 > can mmap = 1 can pause = 0 > can resume = 1 can sync start = 1 My program is not honoring the 'packet size'. -- William Estrada MrUmunhum(a)popdial.com Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) Ymessenger: MrUmunhum

1 0

[alsa-devel] snd_pcm_hw_param_any returns bad data
by William Estrada 06 Sep '08

06 Sep '08

Hi group, I'm using snd_pcm_hw_param_any to retrieve the parameters from an opened device and I'm getting garbage back in the structures. My code is here: http://64.124.13.3/_ALSA_/ This is a simple test to show the problem. The output is shown in show.txt. as you can see the before ( within the open function ) is much different than the after. I am assuming that it is my code that has the problem, any ideas??? Thanks for your time. -- William Estrada MrUmunhum(a)popdial.com Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) Ymessenger: MrUmunhum

1 0

[alsa-devel] [PATCH] ALSA: ASoC V2: optimize init sequence of Freescale MPC8610 sound drivers
by Timur Tabi 06 Sep '08

06 Sep '08

In the Freescale MPC8610 sound drivers, relocate all code from the _prepare functions into the corresponding _hw_params functions. These drivers assumed that the sample size is known in the _prepare function and not in the _hw_params function, but this is not true. Move the code in fsl_dma_prepare() into fsl_dma_hw_param(). Create fsl_ssi_hw_params() and move the code from fsl_ssi_prepare() into it. Turn off snooping for DMA operations to/from I/O registers, since that's not necessary. Some comment blocks were not near the code they reference, so they were moved. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- This patch is for ASoC V2 only. sound/soc/fsl/fsl_dma.c | 283 ++++++++++++++++++++++------------------------- sound/soc/fsl/fsl_ssi.c | 20 ++-- 2 files changed, 141 insertions(+), 162 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 5da0069..8821289 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -319,9 +319,66 @@ error: } /** - * fsl_dma_open: open a new substream. + * fsl_dma_open: open a new substream and allocate DMA resources for it * * Each substream has its own DMA buffer. + * + * ALSA divides the DMA buffer into N periods. We create NUM_DMA_LINKS link + * descriptors that cycle from one period to the next. For example, if + * there are six periods and two link descriptors, this is how they look + * before playback starts: + * + * The last link descriptor + * ____________ points back to the first + * | | + * V | + * ___ ___ | + * | |->| |->| + * |___| |___| + * | | + * | | + * V V + * _________________________________________ + * | | | | | | | The DMA buffer is + * | | | | | | | divided into 6 parts + * |______|______|______|______|______|______| + * + * and here's how they look after the first period is finished playing: + * + * ____________ + * | | + * V | + * ___ ___ | + * | |->| |->| + * |___| |___| + * | | + * |______________ + * | | + * V V + * _________________________________________ + * | | | | | | | + * | | | | | | | + * |______|______|______|______|______|______| + * + * The first link descriptor now points to the third period. The DMA + * controller is currently playing the second period. When it finishes, it + * will jump back to the first descriptor and play the third period. + * + * There are four reasons we do this: + * + * 1. The only way to get the DMA controller to automatically restart the + * transfer when it gets to the end of the buffer is to use chaining + * mode. Basic direct mode doesn't offer that feature. + * 2. We need to receive an interrupt at the end of every period. The DMA + * controller can generate an interrupt at the end of every link transfer + * (aka segment). Making each period into a DMA segment will give us the + * interrupts we need. + * 3. By creating only two link descriptors, regardless of the number of + * periods, we do not need to reallocate the link descriptors if the + * number of periods changes. + * 4. All of the audio data is still stored in a single, contiguous DMA + * buffer, which is what ALSA expects. We're just dividing it into + * contiguous parts, and creating a link descriptor for each one. */ static int fsl_dma_open(struct snd_pcm_substream *substream) { @@ -370,8 +427,8 @@ static int fsl_dma_open(struct snd_pcm_substream *substream) dma_private->ld_buf_phys = ld_buf_phys; dma_private->dma_buf_phys = substream->dma_buffer.addr; - ret = request_irq(dma_private->dma_info->irq, - fsl_dma_isr, 0, "DMA", dma_private); + ret = request_irq(dma_private->dma_info->irq, fsl_dma_isr, 0, + "fsl-audio-dma", dma_private); if (ret) { dev_err(substream->pcm->card->dev, "can't register ISR for IRQ %u (ret=%i)\n", @@ -393,17 +450,14 @@ static int fsl_dma_open(struct snd_pcm_substream *substream) temp_link = dma_private->ld_buf_phys + sizeof(struct fsl_dma_link_descriptor); - for (i = 0; i < NUM_DMA_LINKS; i++) { - struct fsl_dma_link_descriptor *link = &dma_private->link[i]; - - link->source_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); - link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); - link->next = cpu_to_be64(temp_link); + for (i = 0; i < NUM_DMA_LINKS - 1; i++) { + dma_private->link[i].next = cpu_to_be64(temp_link); temp_link += sizeof(struct fsl_dma_link_descriptor); } /* The last link descriptor points to the first */ - dma_private->link[i - 1].next = cpu_to_be64(dma_private->ld_buf_phys); + dma_private->link[NUM_DMA_LINKS - 1].next = + cpu_to_be64(dma_private->ld_buf_phys); /* Tell the DMA controller where the first link descriptor is */ out_be32(&dma_channel->clndar, @@ -417,11 +471,7 @@ static int fsl_dma_open(struct snd_pcm_substream *substream) /* * Program the mode register for interrupts, external master control, * and source/destination hold. Also clear the Channel Abort bit. - */ - mr = in_be32(&dma_channel->mr) & - ~(CCSR_DMA_MR_CA | CCSR_DMA_MR_DAHE | CCSR_DMA_MR_SAHE); - - /* + * * We want External Master Start and External Master Pause enabled, * because the SSI is controlling the DMA controller. We want the DMA * controller to be set up in advance, and then we signal only the SSI @@ -436,7 +486,7 @@ static int fsl_dma_open(struct snd_pcm_substream *substream) * We want Error Interrupt enabled, so that we can get an error if * the DMA controller is mis-programmed somehow. */ - mr |= CCSR_DMA_MR_EOSIE | CCSR_DMA_MR_EIE | CCSR_DMA_MR_EMP_EN | + mr = CCSR_DMA_MR_EOSIE | CCSR_DMA_MR_EIE | CCSR_DMA_MR_EMP_EN | CCSR_DMA_MR_EMS_EN; /* For playback, we want the destination address to be held. For @@ -450,78 +500,47 @@ static int fsl_dma_open(struct snd_pcm_substream *substream) } /** - * fsl_dma_hw_params: allocate the DMA buffer and the DMA link descriptors. - * - * ALSA divides the DMA buffer into N periods. We create NUM_DMA_LINKS link - * descriptors that ping-pong from one period to the next. For example, if - * there are six periods and two link descriptors, this is how they look - * before playback starts: - * - * The last link descriptor - * ____________ points back to the first - * | | - * V | - * ___ ___ | - * | |->| |->| - * |___| |___| - * | | - * | | - * V V - * _________________________________________ - * | | | | | | | The DMA buffer is - * | | | | | | | divided into 6 parts - * |______|______|______|______|______|______| + * fsl_dma_hw_params: program the audio paramters into the DMA controller * - * and here's how they look after the first period is finished playing: + * This function takes the audio parameters (period size, sample size, etc) + * and finishes programming the DMA controller accordingly. * - * ____________ - * | | - * V | - * ___ ___ | - * | |->| |->| - * |___| |___| - * | | - * |______________ - * | | - * V V - * _________________________________________ - * | | | | | | | - * | | | | | | | - * |______|______|______|______|______|______| + * The actual address in STX0 (destination for playback, source for capture) + * is based on the sample size. * - * The first link descriptor now points to the third period. The DMA - * controller is currently playing the second period. When it finishes, it - * will jump back to the first descriptor and play the third period. + * One of the drawbacks with big-endian is that when copying integers of + * different sizes to a fixed-sized register, the address to which the + * integer must be copied is dependent on the size of the integer. * - * There are four reasons we do this: + * For example, if P is the address of a 32-bit register, and X is a 32-bit + * integer, then X should be copied to address P. However, if X is a 16-bit + * integer, then it should be copied to P+2. If X is an 8-bit register, + * then it should be copied to P+3. * - * 1. The only way to get the DMA controller to automatically restart the - * transfer when it gets to the end of the buffer is to use chaining - * mode. Basic direct mode doesn't offer that feature. - * 2. We need to receive an interrupt at the end of every period. The DMA - * controller can generate an interrupt at the end of every link transfer - * (aka segment). Making each period into a DMA segment will give us the - * interrupts we need. - * 3. By creating only two link descriptors, regardless of the number of - * periods, we do not need to reallocate the link descriptors if the - * number of periods changes. - * 4. All of the audio data is still stored in a single, contiguous DMA - * buffer, which is what ALSA expects. We're just dividing it into - * contiguous parts, and creating a link descriptor for each one. + * So for playback of 8-bit samples, the DMA controller must transfer single + * bytes from the DMA buffer to the last byte of the STX0 register, i.e. + * offset by 3 bytes. For 16-bit samples, the offset is two bytes. * - * Note that due to a quirk of the SSI's STX register, the target address - * for the DMA operations depends on the sample size. So we don't program - * the dest_addr (for playback -- source_addr for capture) fields in the - * link descriptors here. We do that in fsl_dma_prepare() + * For 24-bit samples, the offset is 1 byte. However, the DMA controller + * does not support 3-byte copies (the DAHTS register supports only 1, 2, 4, + * and 8 bytes at a time). So we do not support packed 24-bit samples. + * 24-bit data must be padded to 32 bits. */ static int fsl_dma_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { struct snd_pcm_runtime *runtime = substream->runtime; struct fsl_dma_private *dma_private = runtime->private_data; + struct ccsr_dma_channel __iomem *dma_channel = + dma_private->dma_info->channel; + + dma_addr_t ssi_sxx_phys; /* Bus address of SSI STX register */ dma_addr_t temp_addr; /* Pointer to next period */ + uint32_t mr; + unsigned int sample_size; /* Number of bits per sample */ + unsigned int frame_size; /* Number of bytes per frame */ unsigned int i; /* Get all the parameters we need */ @@ -537,83 +556,16 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream, if (dma_private->dma_buf_next >= dma_private->dma_buf_end) dma_private->dma_buf_next = dma_private->dma_buf_phys; - /* - * The actual address in STX0 (destination for playback, source for - * capture) is based on the sample size, but we don't know the sample - * size in this function, so we'll have to adjust that later. See - * comments in fsl_dma_prepare(). - * - * The DMA controller does not have a cache, so the CPU does not - * need to tell it to flush its cache. However, the DMA - * controller does need to tell the CPU to flush its cache. - * That's what the SNOOP bit does. - * - * Also, even though the DMA controller supports 36-bit addressing, for - * simplicity we currently support only 32-bit addresses for the audio - * buffer itself. - */ - temp_addr = substream->dma_buffer.addr; - - for (i = 0; i < NUM_DMA_LINKS; i++) { - struct fsl_dma_link_descriptor *link = &dma_private->link[i]; - - link->count = cpu_to_be32(period_size); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - link->source_addr = cpu_to_be32(temp_addr); - else - link->dest_addr = cpu_to_be32(temp_addr); - - temp_addr += period_size; - } - - return 0; -} - -/** - * fsl_dma_prepare - prepare the DMA registers for playback. - * - * This function is called after the specifics of the audio data are known, - * i.e. snd_pcm_runtime is initialized. - * - * In this function, we finish programming the registers of the DMA - * controller that are dependent on the sample size. - * - * One of the drawbacks with big-endian is that when copying integers of - * different sizes to a fixed-sized register, the address to which the - * integer must be copied is dependent on the size of the integer. - * - * For example, if P is the address of a 32-bit register, and X is a 32-bit - * integer, then X should be copied to address P. However, if X is a 16-bit - * integer, then it should be copied to P+2. If X is an 8-bit register, - * then it should be copied to P+3. - * - * So for playback of 8-bit samples, the DMA controller must transfer single - * bytes from the DMA buffer to the last byte of the STX0 register, i.e. - * offset by 3 bytes. For 16-bit samples, the offset is two bytes. - * - * For 24-bit samples, the offset is 1 byte. However, the DMA controller - * does not support 3-byte copies (the DAHTS register supports only 1, 2, 4, - * and 8 bytes at a time). So we do not support packed 24-bit samples. - * 24-bit data must be padded to 32 bits. - */ -static int fsl_dma_prepare(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct fsl_dma_private *dma_private = runtime->private_data; - struct ccsr_dma_channel __iomem *dma_channel = - dma_private->dma_info->channel; - uint32_t mr; - unsigned int i; - dma_addr_t ssi_sxx_phys; /* Bus address of SSI STX register */ - unsigned int frame_size; /* Number of bytes per frame */ + /* Program the sample size and I/O address into the DMA controller */ ssi_sxx_phys = dma_private->dma_info->ssi_sxx_phys; mr = in_be32(&dma_channel->mr) & ~(CCSR_DMA_MR_BWC_MASK | CCSR_DMA_MR_SAHTS_MASK | CCSR_DMA_MR_DAHTS_MASK); - switch (runtime->sample_bits) { + sample_size = snd_pcm_format_physical_width(params_format(hw_params)); + + switch (sample_size) { case 8: mr |= CCSR_DMA_MR_DAHTS_1 | CCSR_DMA_MR_SAHTS_1; ssi_sxx_phys += 3; @@ -631,7 +583,8 @@ static int fsl_dma_prepare(struct snd_pcm_substream *substream) return -EINVAL; } - frame_size = runtime->frame_bits / 8; + frame_size = 2 * (sample_size / 8); + /* * BWC should always be a multiple of the frame size. BWC determines * how many bytes are sent/received before the DMA controller checks the @@ -648,16 +601,45 @@ static int fsl_dma_prepare(struct snd_pcm_substream *substream) out_be32(&dma_channel->mr, mr); - /* - * Program the address of the DMA transfer to/from the SSI. - */ + temp_addr = substream->dma_buffer.addr; + for (i = 0; i < NUM_DMA_LINKS; i++) { struct fsl_dma_link_descriptor *link = &dma_private->link[i]; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + link->count = cpu_to_be32(period_size); + + /* Even though the DMA controller supports 36-bit addressing, + * for simplicity we allow only 32-bit addresses for the audio + * buffer itself. This was enforced in fsl_dma_new() with the + * DMA mask. + * + * The snoop bit tells the DMA controller whether it should tell + * the ECM to snoop during a read or write to an address. For + * audio, we use DMA to transfer data between memory and an I/O + * device (the SSI's STX0 or SRX0 register). Snooping is only + * needed if there is a cache, so we need to snoop memory + * addresses only. For playback, that means we snoop the source + * but not the destination. For capture, we snoop the + * destination but not the source. + * + * Note that this is unlikely to cause cache incoherency unless + * the DMA buffer is smaller than the size of L1 cache. + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + link->source_addr = cpu_to_be32(temp_addr); + link->source_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); + link->dest_addr = cpu_to_be32(ssi_sxx_phys); - else + link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_NOSNOOP); + } else { link->source_addr = cpu_to_be32(ssi_sxx_phys); + link->source_attr = cpu_to_be32(CCSR_DMA_ATR_NOSNOOP); + + link->dest_addr = cpu_to_be32(temp_addr); + link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); + } + + temp_addr += period_size; } return 0; @@ -800,7 +782,6 @@ static struct snd_pcm_ops fsl_dma_ops = { .ioctl = snd_pcm_lib_ioctl, .hw_params = fsl_dma_hw_params, .hw_free = fsl_dma_hw_free, - .prepare = fsl_dma_prepare, .pointer = fsl_dma_pointer, }; diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 319cc97..fc52d30 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -344,30 +344,28 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, } /** - * fsl_ssi_prepare: prepare the SSI. + * fsl_ssi_hw_params: program the sample size * * Most of the SSI registers have been programmed in the startup function, - * but the word length must be programmed here. Unfortunately, programming - * the SxCCR.WL bits requires the SSI to be temporarily disabled. This can - * cause a problem with supporting simultaneous playback and capture. If - * the SSI is already playing a stream, then that stream may be temporarily - * stopped when you start capture. + * but the word length must be programmed here. Since we support + * synchronous mode only, only the master stream can program STCRR[WL]. * * Note: The SxCCR.DC and SxCCR.PM bits are only used if the SSI is the * clock master. */ -static int fsl_ssi_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *cpu_dai) +static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params, struct snd_soc_dai *cpu_dai) { struct fsl_ssi_info *ssi_info = cpu_dai->private_data; if (substream == ssi_info->master_stream) { - struct snd_pcm_runtime *runtime = substream->runtime; struct ccsr_ssi __iomem *ssi = ssi_info->ssi; + unsigned int sample_size = + snd_pcm_format_width(params_format(hw_params)); u32 wl; /* The SSI should always be disabled at this points (SSIEN=0) */ - wl = CCSR_SSI_SxCCR_WL(snd_pcm_format_width(runtime->format)); + wl = CCSR_SSI_SxCCR_WL(sample_size); /* In synchronous mode, the SSI uses STCCR for capture */ clrsetbits_be32(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl); @@ -588,7 +586,7 @@ static struct snd_soc_dai_caps capture = { static struct snd_soc_dai_ops ops = { .startup = fsl_ssi_startup, - .prepare = fsl_ssi_prepare, + .hw_params = fsl_ssi_hw_params, .shutdown = fsl_ssi_shutdown, .trigger = fsl_ssi_trigger, -- 1.5.5

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[alsa-devel] HP tx2000 and snd-hda-intel
by Kory Prince 06 Sep '08

06 Sep '08

Hello, I have an HP Pavillion TX2000z. This laptop has uses the snd-hda-intel and has the ALC861VD realtek codec. The existing configuration for the codec does not support the tx2000 series laptops fully. There are several configurations that have various levels of support. If you specify for the model to use the existing hp model (made for the tx1000 series) then: 1.Headphones Work 2.Speakers Work 3.Front Mic Works (the jack mic by the headphone jack) 4.The External Mics do not work (There are two mics at the top of the laptop... I am guessing left and right for a stereo Mic) 5.It seems like there is something mapped to an input that does not belong there. The simplest way to explain this is when I open audacity and set it to monitor input that there is significant white noise coming in. The Ext. Mics are not working so it can't come from there (as far as I know) and I have nothing plugged into the front Mic. I cannot find any way to mute this noise. When using the 3-stack model (with the option position_fix): 1. Headphones do not work (in either ports) 2. Speakers Work 3. Front Mic Works 4. The external mics work 5. Same problem as expressed above All other models do not work in any significant way. Also one of the Headphone ports (there are two on the front) has Digital Out (optical) that I have no way of confirming if it works. Now, that being said, I have tried working on the driver. I have virtually no programing experience. I was able to have the driver autodetect the soundcard and apply a certain model (before it had to be applied with a module config file.) And was able to interpret some of how the driver worked. But I am so lost. I am not sure how hard it would be for a programmer to modify the driver to support the tx2000. So I want to put out a call to anyone who could help. I am fully available for testing and some debugging if someone could try and help. I am uploading the codec file and the codec schematic ( I don't think this is actually accurate, but It may help...) Also here is the lspci -nn for the soundcard and other info I found 00:10.1 Audio device [0403]: nVidia Corporation MCP51 High Definition Audio [10de:026c] (rev a2) lshw *-multimedia description: Audio device product: MCP51 High Definition Audio vendor: nVidia Corporation physical id: 10.1 bus info: pci@0000:00:10.1 version: a2 width: 32 bits clock: 66MHz capabilities: pm msi ht bus_master cap_list configuration: driver=HDA Intel latency=0 maxlatency=5 mingnt=2 module=snd_hda_intel If anyone could help or use additional information I would appreciate it. Thanks, Kory p.s. if this is the wrong place to post, please redirect me.

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[alsa-devel] ALSA SOC VI API's for 2 PCM DEVICE ON SAME CARD
by dinesh 05 Sep '08

05 Sep '08

Hi, Actually i am stuck with a problem i have two register 2 channels of my SOC device as 2 different pcm devices on same card. But my driver uses ALSA V1 api's and i am not able to find out any example code for this using v1 api's. (SIMILAR TO MPC5200_PSC.C) If anyone can help me please. Thanks, Dinesh

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[alsa-devel] [PATCH 0/14] ASoC: Blackfin and I2C updats
by Mark Brown 05 Sep '08

05 Sep '08

The following changes since commit 44e2c3045f77c69d18ba4afda888a4cdec4a33fd: Cliff Cai (1): ALSA: ASoC codec: fix compiling error in ad1980 driver after ASoC API changed are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai Cliff Cai (10): ASoC codec: SSM2602 audio codec driver ASoC: Blackfin: SPORT peripheral interface driver ASoC: Blackfin: DMA Driver for AC97 sound chip ASoC: Blackfin: AC97 Blackfin CPU DAI driver ASoC: Blackfin: DMA Driver for I2S sound chip ASoC: Blackfin: I2S CPU DAI driver ASoC: Blackfin: board driver for AD1980/1 audio codec ASoC: Blackfin: board driver for SSM2602 sound chip ASoC: Blackfin: add Blackfin arch ASoC Kconfig and Makefile ASoC: Blackfin: Include Blackfin architecture support in build Jean Delvare (4): ASoC: Fix an error path in neo1973_wm8753 ASoC: Convert wm8753 to a new-style i2c driver ASoC: Convert neo1973/lm4857 to a new-style i2c driver ASoC: Convert wm8510 to a new-style i2c driver sound/soc/Kconfig | 1 + sound/soc/Makefile | 2 +- sound/soc/at32/playpaq_wm8510.c | 1 + sound/soc/blackfin/Kconfig | 85 +++ sound/soc/blackfin/Makefile | 20 + sound/soc/blackfin/bf5xx-ac97-pcm.c | 429 +++++++++++++++ sound/soc/blackfin/bf5xx-ac97-pcm.h | 29 + sound/soc/blackfin/bf5xx-ac97.c | 407 ++++++++++++++ sound/soc/blackfin/bf5xx-ac97.h | 36 ++ sound/soc/blackfin/bf5xx-ad1980.c | 113 ++++ sound/soc/blackfin/bf5xx-i2s-pcm.c | 288 ++++++++++ sound/soc/blackfin/bf5xx-i2s-pcm.h | 29 + sound/soc/blackfin/bf5xx-i2s.c | 292 ++++++++++ sound/soc/blackfin/bf5xx-i2s.h | 14 + sound/soc/blackfin/bf5xx-sport.c | 1032 +++++++++++++++++++++++++++++++++++ sound/soc/blackfin/bf5xx-sport.h | 192 +++++++ sound/soc/blackfin/bf5xx-ssm2602.c | 186 +++++++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ssm2602.c | 776 ++++++++++++++++++++++++++ sound/soc/codecs/ssm2602.h | 130 +++++ sound/soc/codecs/wm8510.c | 110 ++-- sound/soc/codecs/wm8510.h | 1 + sound/soc/codecs/wm8753.c | 106 ++-- sound/soc/codecs/wm8753.h | 1 + sound/soc/s3c24xx/neo1973_wm8753.c | 113 +++-- 26 files changed, 4237 insertions(+), 162 deletions(-) create mode 100644 sound/soc/blackfin/Kconfig create mode 100644 sound/soc/blackfin/Makefile create mode 100644 sound/soc/blackfin/bf5xx-ac97-pcm.c create mode 100644 sound/soc/blackfin/bf5xx-ac97-pcm.h create mode 100644 sound/soc/blackfin/bf5xx-ac97.c create mode 100644 sound/soc/blackfin/bf5xx-ac97.h create mode 100644 sound/soc/blackfin/bf5xx-ad1980.c create mode 100644 sound/soc/blackfin/bf5xx-i2s-pcm.c create mode 100644 sound/soc/blackfin/bf5xx-i2s-pcm.h create mode 100644 sound/soc/blackfin/bf5xx-i2s.c create mode 100644 sound/soc/blackfin/bf5xx-i2s.h create mode 100644 sound/soc/blackfin/bf5xx-sport.c create mode 100644 sound/soc/blackfin/bf5xx-sport.h create mode 100644 sound/soc/blackfin/bf5xx-ssm2602.c create mode 100644 sound/soc/codecs/ssm2602.c create mode 100644 sound/soc/codecs/ssm2602.h

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