From 7e1f7c8a6e517900cd84da1b8ae020f08f286c3b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 12 Apr 2012 17:29:36 +0100 Subject: ASoC: dapm: Ensure power gets managed for line widgets Line widgets had not been included in either the power up or power down sequences so if a widget had an event associated with it that event would never be run. Fix this minimally by adding them to the sequences, we should probably be doing away with the specific widget types as they all have the same priority anyway. Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/soc-dapm.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 6241490fff3..dc7dbfe61cd 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -67,6 +67,7 @@ static int dapm_up_seq[] = { [snd_soc_dapm_out_drv] = 10, [snd_soc_dapm_hp] = 10, [snd_soc_dapm_spk] = 10, + [snd_soc_dapm_line] = 10, [snd_soc_dapm_post] = 11, }; @@ -75,6 +76,7 @@ static int dapm_down_seq[] = { [snd_soc_dapm_adc] = 1, [snd_soc_dapm_hp] = 2, [snd_soc_dapm_spk] = 2, + [snd_soc_dapm_line] = 2, [snd_soc_dapm_out_drv] = 2, [snd_soc_dapm_pga] = 4, [snd_soc_dapm_mixer_named_ctl] = 5, -- cgit v1.2.3-70-g09d2 From 86fc49982369f6918dd9c6eeb70b38ab2303ed0a Mon Sep 17 00:00:00 2001 From: Jesper Juhl Date: Thu, 12 Apr 2012 21:54:34 +0200 Subject: ASoC: cs42l73: don't use negative array index If cs42l73_get_mclkx_coeff() returns < 0 (which it can) in sound/soc/codecs/cs42l73.c::cs42l73_set_mclk(), then we'll be using the (negative) return value as array index on the very next line of code - that's bad. Catch the negative return value and propagate it to the caller (which checks for it) and things are a bit more sane :-) Signed-off-by: Jesper Juhl Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l73.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 78979b3e0e9..07c44b71f09 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -929,6 +929,8 @@ static int cs42l73_set_mclk(struct snd_soc_dai *dai, unsigned int freq) /* MCLKX -> MCLK */ mclkx_coeff = cs42l73_get_mclkx_coeff(freq); + if (mclkx_coeff < 0) + return mclkx_coeff; mclk = cs42l73_mclkx_coeffs[mclkx_coeff].mclkx / cs42l73_mclkx_coeffs[mclkx_coeff].ratio; -- cgit v1.2.3-70-g09d2 From 8eaeb9393397be8eb700ab38a69c450975463b77 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 3 Apr 2012 11:56:51 +0300 Subject: mfd: Convert twl6040 to i2c driver, and separate it from twl core Complete the separation of the twl6040 from the twl core since it is a separate chip, not part of the twl6030 PMIC. Make the needed Kconfig changes for the depending drivers at the same time to avoid breaking the kernel build (vibra, ASoC components). Signed-off-by: Peter Ujfalusi Reviewed-by: Mark Brown Acked-by: Tony Lindgren Acked-by: Dmitry Torokhov Signed-off-by: Samuel Ortiz --- arch/arm/mach-omap2/board-4430sdp.c | 12 ++-- arch/arm/mach-omap2/board-generic.c | 2 +- arch/arm/mach-omap2/board-omap4panda.c | 13 ++-- arch/arm/mach-omap2/twl-common.c | 37 +++++++++-- arch/arm/mach-omap2/twl-common.h | 10 ++- drivers/input/misc/Kconfig | 3 +- drivers/input/misc/twl6040-vibra.c | 4 +- drivers/mfd/Kconfig | 11 +++- drivers/mfd/twl6040-core.c | 114 ++++++++++++++++++++------------- include/linux/i2c/twl.h | 12 ---- include/linux/mfd/twl6040.h | 27 ++++++++ sound/soc/codecs/Kconfig | 3 +- sound/soc/codecs/twl6040.c | 3 +- sound/soc/omap/Kconfig | 2 +- 14 files changed, 159 insertions(+), 94 deletions(-) (limited to 'sound') diff --git a/arch/arm/mach-omap2/board-4430sdp.c b/arch/arm/mach-omap2/board-4430sdp.c index a39fc4bbd2b..130ab00c09a 100644 --- a/arch/arm/mach-omap2/board-4430sdp.c +++ b/arch/arm/mach-omap2/board-4430sdp.c @@ -20,6 +20,7 @@ #include #include #include +#include #include #include #include @@ -560,7 +561,7 @@ static struct regulator_init_data sdp4430_vusim = { }, }; -static struct twl4030_codec_data twl6040_codec = { +static struct twl6040_codec_data twl6040_codec = { /* single-step ramp for headset and handsfree */ .hs_left_step = 0x0f, .hs_right_step = 0x0f, @@ -568,7 +569,7 @@ static struct twl4030_codec_data twl6040_codec = { .hf_right_step = 0x1d, }; -static struct twl4030_vibra_data twl6040_vibra = { +static struct twl6040_vibra_data twl6040_vibra = { .vibldrv_res = 8, .vibrdrv_res = 3, .viblmotor_res = 10, @@ -577,16 +578,14 @@ static struct twl4030_vibra_data twl6040_vibra = { .vddvibr_uV = 0, /* fixed volt supply - VBAT */ }; -static struct twl4030_audio_data twl6040_audio = { +static struct twl6040_platform_data twl6040_data = { .codec = &twl6040_codec, .vibra = &twl6040_vibra, .audpwron_gpio = 127, - .naudint_irq = OMAP44XX_IRQ_SYS_2N, .irq_base = TWL6040_CODEC_IRQ_BASE, }; static struct twl4030_platform_data sdp4430_twldata = { - .audio = &twl6040_audio, /* Regulators */ .vusim = &sdp4430_vusim, .vaux1 = &sdp4430_vaux1, @@ -617,7 +616,8 @@ static int __init omap4_i2c_init(void) TWL_COMMON_REGULATOR_VCXIO | TWL_COMMON_REGULATOR_VUSB | TWL_COMMON_REGULATOR_CLK32KG); - omap4_pmic_init("twl6030", &sdp4430_twldata); + omap4_pmic_init("twl6030", &sdp4430_twldata, + &twl6040_data, OMAP44XX_IRQ_SYS_2N); omap_register_i2c_bus(2, 400, NULL, 0); omap_register_i2c_bus(3, 400, sdp4430_i2c_3_boardinfo, ARRAY_SIZE(sdp4430_i2c_3_boardinfo)); diff --git a/arch/arm/mach-omap2/board-generic.c b/arch/arm/mach-omap2/board-generic.c index 74e1687b517..098d183a008 100644 --- a/arch/arm/mach-omap2/board-generic.c +++ b/arch/arm/mach-omap2/board-generic.c @@ -137,7 +137,7 @@ static struct twl4030_platform_data sdp4430_twldata = { static void __init omap4_i2c_init(void) { - omap4_pmic_init("twl6030", &sdp4430_twldata); + omap4_pmic_init("twl6030", &sdp4430_twldata, NULL, 0); } static void __init omap4_init(void) diff --git a/arch/arm/mach-omap2/board-omap4panda.c b/arch/arm/mach-omap2/board-omap4panda.c index d8c0e89f012..1b782ba5343 100644 --- a/arch/arm/mach-omap2/board-omap4panda.c +++ b/arch/arm/mach-omap2/board-omap4panda.c @@ -25,6 +25,7 @@ #include #include #include +#include #include #include #include @@ -284,7 +285,7 @@ static int __init omap4_twl6030_hsmmc_init(struct omap2_hsmmc_info *controllers) return 0; } -static struct twl4030_codec_data twl6040_codec = { +static struct twl6040_codec_data twl6040_codec = { /* single-step ramp for headset and handsfree */ .hs_left_step = 0x0f, .hs_right_step = 0x0f, @@ -292,17 +293,14 @@ static struct twl4030_codec_data twl6040_codec = { .hf_right_step = 0x1d, }; -static struct twl4030_audio_data twl6040_audio = { +static struct twl6040_platform_data twl6040_data = { .codec = &twl6040_codec, .audpwron_gpio = 127, - .naudint_irq = OMAP44XX_IRQ_SYS_2N, .irq_base = TWL6040_CODEC_IRQ_BASE, }; /* Panda board uses the common PMIC configuration */ -static struct twl4030_platform_data omap4_panda_twldata = { - .audio = &twl6040_audio, -}; +static struct twl4030_platform_data omap4_panda_twldata; /* * Display monitor features are burnt in their EEPROM as EDID data. The EEPROM @@ -326,7 +324,8 @@ static int __init omap4_panda_i2c_init(void) TWL_COMMON_REGULATOR_VCXIO | TWL_COMMON_REGULATOR_VUSB | TWL_COMMON_REGULATOR_CLK32KG); - omap4_pmic_init("twl6030", &omap4_panda_twldata); + omap4_pmic_init("twl6030", &omap4_panda_twldata, + &twl6040_data, OMAP44XX_IRQ_SYS_2N); omap_register_i2c_bus(2, 400, NULL, 0); /* * Bus 3 is attached to the DVI port where devices like the pico DLP diff --git a/arch/arm/mach-omap2/twl-common.c b/arch/arm/mach-omap2/twl-common.c index 4b57757bf9d..7a7b89304c4 100644 --- a/arch/arm/mach-omap2/twl-common.c +++ b/arch/arm/mach-omap2/twl-common.c @@ -37,6 +37,16 @@ static struct i2c_board_info __initdata pmic_i2c_board_info = { .flags = I2C_CLIENT_WAKE, }; +static struct i2c_board_info __initdata omap4_i2c1_board_info[] = { + { + .addr = 0x48, + .flags = I2C_CLIENT_WAKE, + }, + { + I2C_BOARD_INFO("twl6040", 0x4b), + }, +}; + void __init omap_pmic_init(int bus, u32 clkrate, const char *pmic_type, int pmic_irq, struct twl4030_platform_data *pmic_data) @@ -49,14 +59,31 @@ void __init omap_pmic_init(int bus, u32 clkrate, omap_register_i2c_bus(bus, clkrate, &pmic_i2c_board_info, 1); } +void __init omap4_pmic_init(const char *pmic_type, + struct twl4030_platform_data *pmic_data, + struct twl6040_platform_data *twl6040_data, int twl6040_irq) +{ + /* PMIC part*/ + strncpy(omap4_i2c1_board_info[0].type, pmic_type, + sizeof(omap4_i2c1_board_info[0].type)); + omap4_i2c1_board_info[0].irq = OMAP44XX_IRQ_SYS_1N; + omap4_i2c1_board_info[0].platform_data = pmic_data; + + /* TWL6040 audio IC part */ + omap4_i2c1_board_info[1].irq = twl6040_irq; + omap4_i2c1_board_info[1].platform_data = twl6040_data; + + omap_register_i2c_bus(1, 400, omap4_i2c1_board_info, 2); + +} + void __init omap_pmic_late_init(void) { /* Init the OMAP TWL parameters (if PMIC has been registerd) */ - if (!pmic_i2c_board_info.irq) - return; - - omap3_twl_init(); - omap4_twl_init(); + if (pmic_i2c_board_info.irq) + omap3_twl_init(); + if (omap4_i2c1_board_info[0].irq) + omap4_twl_init(); } #if defined(CONFIG_ARCH_OMAP3) diff --git a/arch/arm/mach-omap2/twl-common.h b/arch/arm/mach-omap2/twl-common.h index 275dde8cb27..09627483a57 100644 --- a/arch/arm/mach-omap2/twl-common.h +++ b/arch/arm/mach-omap2/twl-common.h @@ -29,6 +29,7 @@ struct twl4030_platform_data; +struct twl6040_platform_data; void omap_pmic_init(int bus, u32 clkrate, const char *pmic_type, int pmic_irq, struct twl4030_platform_data *pmic_data); @@ -46,12 +47,9 @@ static inline void omap3_pmic_init(const char *pmic_type, omap_pmic_init(1, 2600, pmic_type, INT_34XX_SYS_NIRQ, pmic_data); } -static inline void omap4_pmic_init(const char *pmic_type, - struct twl4030_platform_data *pmic_data) -{ - /* Phoenix Audio IC needs I2C1 to start with 400 KHz or less */ - omap_pmic_init(1, 400, pmic_type, OMAP44XX_IRQ_SYS_1N, pmic_data); -} +void omap4_pmic_init(const char *pmic_type, + struct twl4030_platform_data *pmic_data, + struct twl6040_platform_data *audio_data, int twl6040_irq); void omap3_pmic_get_config(struct twl4030_platform_data *pmic_data, u32 pdata_flags, u32 regulators_flags); diff --git a/drivers/input/misc/Kconfig b/drivers/input/misc/Kconfig index 2d787796bf5..7faf4a7fcaa 100644 --- a/drivers/input/misc/Kconfig +++ b/drivers/input/misc/Kconfig @@ -380,8 +380,7 @@ config INPUT_TWL4030_VIBRA config INPUT_TWL6040_VIBRA tristate "Support for TWL6040 Vibrator" - depends on TWL4030_CORE - select TWL6040_CORE + depends on TWL6040_CORE select INPUT_FF_MEMLESS help This option enables support for TWL6040 Vibrator Driver. diff --git a/drivers/input/misc/twl6040-vibra.c b/drivers/input/misc/twl6040-vibra.c index 45874fed523..14e94f56cb7 100644 --- a/drivers/input/misc/twl6040-vibra.c +++ b/drivers/input/misc/twl6040-vibra.c @@ -28,7 +28,7 @@ #include #include #include -#include +#include #include #include #include @@ -257,7 +257,7 @@ static SIMPLE_DEV_PM_OPS(twl6040_vibra_pm_ops, twl6040_vibra_suspend, NULL); static int __devinit twl6040_vibra_probe(struct platform_device *pdev) { - struct twl4030_vibra_data *pdata = pdev->dev.platform_data; + struct twl6040_vibra_data *pdata = pdev->dev.platform_data; struct vibra_info *info; int ret; diff --git a/drivers/mfd/Kconfig b/drivers/mfd/Kconfig index 29f463cc09c..11e44386fa9 100644 --- a/drivers/mfd/Kconfig +++ b/drivers/mfd/Kconfig @@ -268,10 +268,17 @@ config TWL6030_PWM This is used to control charging LED brightness. config TWL6040_CORE - bool - depends on TWL4030_CORE && GENERIC_HARDIRQS + bool "Support for TWL6040 audio codec" + depends on I2C=y && GENERIC_HARDIRQS select MFD_CORE + select REGMAP_I2C default n + help + Say yes here if you want support for Texas Instruments TWL6040 audio + codec. + This driver provides common support for accessing the device, + additional drivers must be enabled in order to use the + functionality of the device (audio, vibra). config MFD_STMPE bool "Support STMicroelectronics STMPE" diff --git a/drivers/mfd/twl6040-core.c b/drivers/mfd/twl6040-core.c index b2d8e512d3c..2d6bedadca0 100644 --- a/drivers/mfd/twl6040-core.c +++ b/drivers/mfd/twl6040-core.c @@ -30,7 +30,9 @@ #include #include #include -#include +#include +#include +#include #include #include @@ -39,7 +41,7 @@ int twl6040_reg_read(struct twl6040 *twl6040, unsigned int reg) { int ret; - u8 val = 0; + unsigned int val; mutex_lock(&twl6040->io_mutex); /* Vibra control registers from cache */ @@ -47,7 +49,7 @@ int twl6040_reg_read(struct twl6040 *twl6040, unsigned int reg) reg == TWL6040_REG_VIBCTLR)) { val = twl6040->vibra_ctrl_cache[VIBRACTRL_MEMBER(reg)]; } else { - ret = twl_i2c_read_u8(TWL_MODULE_AUDIO_VOICE, &val, reg); + ret = regmap_read(twl6040->regmap, reg, &val); if (ret < 0) { mutex_unlock(&twl6040->io_mutex); return ret; @@ -64,7 +66,7 @@ int twl6040_reg_write(struct twl6040 *twl6040, unsigned int reg, u8 val) int ret; mutex_lock(&twl6040->io_mutex); - ret = twl_i2c_write_u8(TWL_MODULE_AUDIO_VOICE, val, reg); + ret = regmap_write(twl6040->regmap, reg, val); /* Cache the vibra control registers */ if (reg == TWL6040_REG_VIBCTLL || reg == TWL6040_REG_VIBCTLR) twl6040->vibra_ctrl_cache[VIBRACTRL_MEMBER(reg)] = val; @@ -77,16 +79,9 @@ EXPORT_SYMBOL(twl6040_reg_write); int twl6040_set_bits(struct twl6040 *twl6040, unsigned int reg, u8 mask) { int ret; - u8 val; mutex_lock(&twl6040->io_mutex); - ret = twl_i2c_read_u8(TWL_MODULE_AUDIO_VOICE, &val, reg); - if (ret) - goto out; - - val |= mask; - ret = twl_i2c_write_u8(TWL_MODULE_AUDIO_VOICE, val, reg); -out: + ret = regmap_update_bits(twl6040->regmap, reg, mask, mask); mutex_unlock(&twl6040->io_mutex); return ret; } @@ -95,16 +90,9 @@ EXPORT_SYMBOL(twl6040_set_bits); int twl6040_clear_bits(struct twl6040 *twl6040, unsigned int reg, u8 mask) { int ret; - u8 val; mutex_lock(&twl6040->io_mutex); - ret = twl_i2c_read_u8(TWL_MODULE_AUDIO_VOICE, &val, reg); - if (ret) - goto out; - - val &= ~mask; - ret = twl_i2c_write_u8(TWL_MODULE_AUDIO_VOICE, val, reg); -out: + ret = regmap_update_bits(twl6040->regmap, reg, mask, 0); mutex_unlock(&twl6040->io_mutex); return ret; } @@ -494,32 +482,58 @@ static struct resource twl6040_codec_rsrc[] = { }, }; -static int __devinit twl6040_probe(struct platform_device *pdev) +static bool twl6040_readable_reg(struct device *dev, unsigned int reg) { - struct twl4030_audio_data *pdata = pdev->dev.platform_data; + /* Register 0 is not readable */ + if (!reg) + return false; + return true; +} + +static struct regmap_config twl6040_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + .max_register = TWL6040_REG_STATUS, /* 0x2e */ + + .readable_reg = twl6040_readable_reg, +}; + +static int __devinit twl6040_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct twl6040_platform_data *pdata = client->dev.platform_data; struct twl6040 *twl6040; struct mfd_cell *cell = NULL; int ret, children = 0; if (!pdata) { - dev_err(&pdev->dev, "Platform data is missing\n"); + dev_err(&client->dev, "Platform data is missing\n"); return -EINVAL; } /* In order to operate correctly we need valid interrupt config */ - if (!pdata->naudint_irq || !pdata->irq_base) { - dev_err(&pdev->dev, "Invalid IRQ configuration\n"); + if (!client->irq || !pdata->irq_base) { + dev_err(&client->dev, "Invalid IRQ configuration\n"); return -EINVAL; } - twl6040 = kzalloc(sizeof(struct twl6040), GFP_KERNEL); - if (!twl6040) - return -ENOMEM; + twl6040 = devm_kzalloc(&client->dev, sizeof(struct twl6040), + GFP_KERNEL); + if (!twl6040) { + ret = -ENOMEM; + goto err; + } + + twl6040->regmap = regmap_init_i2c(client, &twl6040_regmap_config); + if (IS_ERR(twl6040->regmap)) { + ret = PTR_ERR(twl6040->regmap); + goto err; + } - platform_set_drvdata(pdev, twl6040); + i2c_set_clientdata(client, twl6040); - twl6040->dev = &pdev->dev; - twl6040->irq = pdata->naudint_irq; + twl6040->dev = &client->dev; + twl6040->irq = client->irq; twl6040->irq_base = pdata->irq_base; mutex_init(&twl6040->mutex); @@ -588,12 +602,12 @@ static int __devinit twl6040_probe(struct platform_device *pdev) } if (children) { - ret = mfd_add_devices(&pdev->dev, pdev->id, twl6040->cells, + ret = mfd_add_devices(&client->dev, -1, twl6040->cells, children, NULL, 0); if (ret) goto mfd_err; } else { - dev_err(&pdev->dev, "No platform data found for children\n"); + dev_err(&client->dev, "No platform data found for children\n"); ret = -ENODEV; goto mfd_err; } @@ -608,14 +622,15 @@ gpio2_err: if (gpio_is_valid(twl6040->audpwron)) gpio_free(twl6040->audpwron); gpio1_err: - platform_set_drvdata(pdev, NULL); - kfree(twl6040); + i2c_set_clientdata(client, NULL); + regmap_exit(twl6040->regmap); +err: return ret; } -static int __devexit twl6040_remove(struct platform_device *pdev) +static int __devexit twl6040_remove(struct i2c_client *client) { - struct twl6040 *twl6040 = platform_get_drvdata(pdev); + struct twl6040 *twl6040 = i2c_get_clientdata(client); if (twl6040->power_count) twl6040_power(twl6040, 0); @@ -626,23 +641,30 @@ static int __devexit twl6040_remove(struct platform_device *pdev) free_irq(twl6040->irq_base + TWL6040_IRQ_READY, twl6040); twl6040_irq_exit(twl6040); - mfd_remove_devices(&pdev->dev); - platform_set_drvdata(pdev, NULL); - kfree(twl6040); + mfd_remove_devices(&client->dev); + i2c_set_clientdata(client, NULL); + regmap_exit(twl6040->regmap); return 0; } -static struct platform_driver twl6040_driver = { +static const struct i2c_device_id twl6040_i2c_id[] = { + { "twl6040", 0, }, + { }, +}; +MODULE_DEVICE_TABLE(i2c, twl6040_i2c_id); + +static struct i2c_driver twl6040_driver = { + .driver = { + .name = "twl6040", + .owner = THIS_MODULE, + }, .probe = twl6040_probe, .remove = __devexit_p(twl6040_remove), - .driver = { - .owner = THIS_MODULE, - .name = "twl6040", - }, + .id_table = twl6040_i2c_id, }; -module_platform_driver(twl6040_driver); +module_i2c_driver(twl6040_driver); MODULE_DESCRIPTION("TWL6040 MFD"); MODULE_AUTHOR("Misael Lopez Cruz "); diff --git a/include/linux/i2c/twl.h b/include/linux/i2c/twl.h index 2463b610033..1f90de0cfdb 100644 --- a/include/linux/i2c/twl.h +++ b/include/linux/i2c/twl.h @@ -666,23 +666,11 @@ struct twl4030_codec_data { unsigned int check_defaults:1; unsigned int reset_registers:1; unsigned int hs_extmute:1; - u16 hs_left_step; - u16 hs_right_step; - u16 hf_left_step; - u16 hf_right_step; void (*set_hs_extmute)(int mute); }; struct twl4030_vibra_data { unsigned int coexist; - - /* twl6040 */ - unsigned int vibldrv_res; /* left driver resistance */ - unsigned int vibrdrv_res; /* right driver resistance */ - unsigned int viblmotor_res; /* left motor resistance */ - unsigned int vibrmotor_res; /* right motor resistance */ - int vddvibl_uV; /* VDDVIBL volt, set 0 for fixed reg */ - int vddvibr_uV; /* VDDVIBR volt, set 0 for fixed reg */ }; struct twl4030_audio_data { diff --git a/include/linux/mfd/twl6040.h b/include/linux/mfd/twl6040.h index 9bc9ac651da..b15b5f03f5c 100644 --- a/include/linux/mfd/twl6040.h +++ b/include/linux/mfd/twl6040.h @@ -174,8 +174,35 @@ #define TWL6040_SYSCLK_SEL_LPPLL 0 #define TWL6040_SYSCLK_SEL_HPPLL 1 +struct twl6040_codec_data { + u16 hs_left_step; + u16 hs_right_step; + u16 hf_left_step; + u16 hf_right_step; +}; + +struct twl6040_vibra_data { + unsigned int vibldrv_res; /* left driver resistance */ + unsigned int vibrdrv_res; /* right driver resistance */ + unsigned int viblmotor_res; /* left motor resistance */ + unsigned int vibrmotor_res; /* right motor resistance */ + int vddvibl_uV; /* VDDVIBL volt, set 0 for fixed reg */ + int vddvibr_uV; /* VDDVIBR volt, set 0 for fixed reg */ +}; + +struct twl6040_platform_data { + int audpwron_gpio; /* audio power-on gpio */ + unsigned int irq_base; + + struct twl6040_codec_data *codec; + struct twl6040_vibra_data *vibra; +}; + +struct regmap; + struct twl6040 { struct device *dev; + struct regmap *regmap; struct mutex mutex; struct mutex io_mutex; struct mutex irq_mutex; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 6508e8b790b..59d8efaa17e 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -57,7 +57,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TPA6130A2 if I2C select SND_SOC_TLV320DAC33 if I2C select SND_SOC_TWL4030 if TWL4030_CORE - select SND_SOC_TWL6040 if TWL4030_CORE + select SND_SOC_TWL6040 if TWL6040_CORE select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C select SND_SOC_WL1273 if MFD_WL1273_CORE @@ -276,7 +276,6 @@ config SND_SOC_TWL4030 tristate config SND_SOC_TWL6040 - select TWL6040_CORE tristate config SND_SOC_UDA134X diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 2d8c6b825e5..dc7509b9d53 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -26,7 +26,6 @@ #include #include #include -#include #include #include @@ -1528,7 +1527,7 @@ static int twl6040_resume(struct snd_soc_codec *codec) static int twl6040_probe(struct snd_soc_codec *codec) { struct twl6040_data *priv; - struct twl4030_codec_data *pdata = dev_get_platdata(codec->dev); + struct twl6040_codec_data *pdata = dev_get_platdata(codec->dev); struct platform_device *pdev = container_of(codec->dev, struct platform_device, dev); int ret = 0; diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index e00dd0b1139..deafbfaacdb 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -97,7 +97,7 @@ config SND_OMAP_SOC_SDP3430 config SND_OMAP_SOC_OMAP_ABE_TWL6040 tristate "SoC Audio support for OMAP boards using ABE and twl6040 codec" - depends on TWL4030_CORE && SND_OMAP_SOC && ARCH_OMAP4 + depends on TWL6040_CORE && SND_OMAP_SOC && ARCH_OMAP4 select SND_OMAP_SOC_DMIC select SND_OMAP_SOC_MCPDM select SND_SOC_TWL6040 -- cgit v1.2.3-70-g09d2 From a7dbb603423d499acacefb5fad65d2b406f16370 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Tue, 17 Apr 2012 18:00:11 +0100 Subject: ASoC: core: Fix card RTD count for deferred probe. Currently we increment the number of RTD's per card during the DAI link bind. This can cause an incorrect RTD count when we cannot find a component and defer the probe (and hence perform the DAI link bind for the card again). Fix the count so that it is cleared before every card registration and bind attempt. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 8d2ebf502df..3a4e93e52b6 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3119,6 +3119,7 @@ int snd_soc_register_card(struct snd_soc_card *card) GFP_KERNEL); if (card->rtd == NULL) return -ENOMEM; + card->num_rtd = 0; card->rtd_aux = &card->rtd[card->num_links]; for (i = 0; i < card->num_links; i++) -- cgit v1.2.3-70-g09d2 From f2ec52d4c3698c995c89c579c34d818eab589d8b Mon Sep 17 00:00:00 2001 From: Randy Dunlap Date: Tue, 17 Apr 2012 17:03:42 -0700 Subject: ALSA: fix core/vmaster.c kernel-doc warning Fix kernel-doc warning in sound/core/vmaster.c: Warning(sound/core/vmaster.c:429): No description found for parameter 'private_data' Signed-off-by: Randy Dunlap Signed-off-by: Takashi Iwai --- sound/core/vmaster.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c index 14a286a7bf2..857586135d1 100644 --- a/sound/core/vmaster.c +++ b/sound/core/vmaster.c @@ -419,6 +419,7 @@ EXPORT_SYMBOL(snd_ctl_make_virtual_master); * snd_ctl_add_vmaster_hook - Add a hook to a vmaster control * @kcontrol: vmaster kctl element * @hook: the hook function + * @private_data: the private_data pointer to be saved * * Adds the given hook to the vmaster control element so that it's called * at each time when the value is changed. -- cgit v1.2.3-70-g09d2 From cdf27f373781d8740b874b0b5c18142df32ebb52 Mon Sep 17 00:00:00 2001 From: Paul Mundt Date: Tue, 17 Apr 2012 19:13:04 -0700 Subject: ASoC: fsi: update for dmaengine prep_slave_sg fallout. Leading up to the ->device_prep_slave_sg change in 185ecb5f4fd43911c35956d4cc7d94a1da30417f 'dmaengine: add context parameter to prep_slave_sg and prep_dma_cyclic' a generic wrapper was added in place to guard against the API change, though the fsi driver wasn't updated in the process (presumably its dmaengine support hadn't been merged yet at the time). This trivially switches over to the new wrapper and gets it building again. Signed-off-by: Paul Mundt Acked-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 378cc5b056d..74ed2dffbff 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1001,11 +1001,10 @@ static void fsi_dma_do_tasklet(unsigned long data) sg_dma_address(&sg) = buf; sg_dma_len(&sg) = len; - desc = chan->device->device_prep_slave_sg(chan, &sg, 1, dir, - DMA_PREP_INTERRUPT | - DMA_CTRL_ACK); + desc = dmaengine_prep_slave_sg(chan, &sg, 1, dir, + DMA_PREP_INTERRUPT | DMA_CTRL_ACK); if (!desc) { - dev_err(dai->dev, "device_prep_slave_sg() fail\n"); + dev_err(dai->dev, "dmaengine_prep_slave_sg() fail\n"); return; } -- cgit v1.2.3-70-g09d2 From 118cb4a408e1c4021ac85d6c05da66bb6f57e556 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Apr 2012 07:33:27 +0200 Subject: ALSA: hda/realtek - Fix regression on Quanta/Gericom KN1 Through the transition to the auto-parser, the support for Quanta/Gericom KN1 got broken. There are two problems behind it: - This machine doesn't like the default COEF setup for ALC260 we take now as default - BIOS doesn't set the pins correctly at all; especially the machine uses only the pin 0x0f for both headphone and speaker This patch adds the fixup as a workaround for these issues. Reported-and-tested-by: Uros Vampl Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 49 +++++++++++++++++++++++++++++++++++++++---- 1 file changed, 45 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2508f8109f1..e65e3543305 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1445,6 +1445,13 @@ enum { ALC_FIXUP_ACT_BUILD, }; +static void alc_apply_pincfgs(struct hda_codec *codec, + const struct alc_pincfg *cfg) +{ + for (; cfg->nid; cfg++) + snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val); +} + static void alc_apply_fixup(struct hda_codec *codec, int action) { struct alc_spec *spec = codec->spec; @@ -1478,9 +1485,7 @@ static void alc_apply_fixup(struct hda_codec *codec, int action) snd_printdd(KERN_INFO "hda_codec: %s: " "Apply pincfg for %s\n", codec->chip_name, modelname); - for (; cfg->nid; cfg++) - snd_hda_codec_set_pincfg(codec, cfg->nid, - cfg->val); + alc_apply_pincfgs(codec, cfg); break; case ALC_FIXUP_VERBS: if (action != ALC_FIXUP_ACT_PROBE || !fix->v.verbs) @@ -4861,6 +4866,7 @@ enum { ALC260_FIXUP_GPIO1_TOGGLE, ALC260_FIXUP_REPLACER, ALC260_FIXUP_HP_B1900, + ALC260_FIXUP_KN1, }; static void alc260_gpio1_automute(struct hda_codec *codec) @@ -4888,6 +4894,36 @@ static void alc260_fixup_gpio1_toggle(struct hda_codec *codec, } } +static void alc260_fixup_kn1(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + static const struct alc_pincfg pincfgs[] = { + { 0x0f, 0x02214000 }, /* HP/speaker */ + { 0x12, 0x90a60160 }, /* int mic */ + { 0x13, 0x02a19000 }, /* ext mic */ + { 0x18, 0x01446000 }, /* SPDIF out */ + /* disable bogus I/O pins */ + { 0x10, 0x411111f0 }, + { 0x11, 0x411111f0 }, + { 0x14, 0x411111f0 }, + { 0x15, 0x411111f0 }, + { 0x16, 0x411111f0 }, + { 0x17, 0x411111f0 }, + { 0x19, 0x411111f0 }, + { } + }; + + switch (action) { + case ALC_FIXUP_ACT_PRE_PROBE: + alc_apply_pincfgs(codec, pincfgs); + break; + case ALC_FIXUP_ACT_PROBE: + spec->init_amp = ALC_INIT_NONE; + break; + } +} + static const struct alc_fixup alc260_fixups[] = { [ALC260_FIXUP_HP_DC5750] = { .type = ALC_FIXUP_PINS, @@ -4938,7 +4974,11 @@ static const struct alc_fixup alc260_fixups[] = { .v.func = alc260_fixup_gpio1_toggle, .chained = true, .chain_id = ALC260_FIXUP_COEF, - } + }, + [ALC260_FIXUP_KN1] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc260_fixup_kn1, + }, }; static const struct snd_pci_quirk alc260_fixup_tbl[] = { @@ -4948,6 +4988,7 @@ static const struct snd_pci_quirk alc260_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x280a, "HP dc5750", ALC260_FIXUP_HP_DC5750), SND_PCI_QUIRK(0x103c, 0x30ba, "HP Presario B1900", ALC260_FIXUP_HP_B1900), SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FIXUP_GPIO1), + SND_PCI_QUIRK(0x152d, 0x0729, "Quanta KN1", ALC260_FIXUP_KN1), SND_PCI_QUIRK(0x161f, 0x2057, "Replacer 672V", ALC260_FIXUP_REPLACER), SND_PCI_QUIRK(0x1631, 0xc017, "PB V7900", ALC260_FIXUP_COEF), {} -- cgit v1.2.3-70-g09d2 From 3e843196c697ee2c319d96e861980fb4c3e04e24 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Apr 2012 12:04:03 +0200 Subject: ALSA: hda/sigmatel - Fix inverted mute LED While refactoring the mute-LED handling for HP laptops, I messed up the polarity check in a wrong way. The red (or the mute-LED if any) should appear in the muted state, corresponding to GPIO on. Reported-by: Mikko Vinni Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 33a9946b492..4742cac26aa 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5063,12 +5063,11 @@ static void stac92xx_update_led_status(struct hda_codec *codec, int enabled) if (spec->gpio_led_polarity) muted = !muted; - /*polarity defines *not* muted state level*/ if (!spec->vref_mute_led_nid) { if (muted) - spec->gpio_data &= ~spec->gpio_led; /* orange */ + spec->gpio_data |= spec->gpio_led; else - spec->gpio_data |= spec->gpio_led; /* white */ + spec->gpio_data &= ~spec->gpio_led; stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, spec->gpio_data); } else { -- cgit v1.2.3-70-g09d2 From 590b4775d6b628c7ad215fd0335a0a787032e2dd Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 19 Apr 2012 00:00:27 -0700 Subject: ALSA: workaround: change the timing of alsa_sound_last_init() Current alsa_sound_last_init() was called as __initcall(). So, on current ALSA, only devices that had been properly registered at this point were shown. So, it will show "No soundcards found" if driver requests probe deferment. it's often misleading. This patch delays the timing of alsa_sound_last_init() as workaround. Signed-off-by: Kuninori Morimoto Reviwed-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/last.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/last.c b/sound/last.c index bdd0857b887..7ffc182e084 100644 --- a/sound/last.c +++ b/sound/last.c @@ -38,4 +38,4 @@ static int __init alsa_sound_last_init(void) return 0; } -__initcall(alsa_sound_last_init); +late_initcall_sync(alsa_sound_last_init); -- cgit v1.2.3-70-g09d2 From ca3649de026ff95c6f2847e8d096cf2f411c02b3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Apr 2012 15:15:25 +0200 Subject: ALSA: hda/conexant - Don't set HP pin-control bit unconditionally Some output pins on Conexant chips have no HP control bit, but the auto-parser initializes these pins unconditionally with PIN_HP. Check the pin-capability and avoid the HP bit if not supported. Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index d29d6d37790..f52c9ef3cc8 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3951,9 +3951,14 @@ static void cx_auto_init_output(struct hda_codec *codec) int i; mute_outputs(codec, spec->multiout.num_dacs, spec->multiout.dac_nids); - for (i = 0; i < cfg->hp_outs; i++) + for (i = 0; i < cfg->hp_outs; i++) { + unsigned int val = PIN_OUT; + if (snd_hda_query_pin_caps(codec, cfg->hp_pins[i]) & + AC_PINCAP_HP_DRV) + val |= AC_PINCTL_HP_EN; snd_hda_codec_write(codec, cfg->hp_pins[i], 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP); + AC_VERB_SET_PIN_WIDGET_CONTROL, val); + } mute_outputs(codec, cfg->hp_outs, cfg->hp_pins); mute_outputs(codec, cfg->line_outs, cfg->line_out_pins); mute_outputs(codec, cfg->speaker_outs, cfg->speaker_pins); -- cgit v1.2.3-70-g09d2 From d70f363222ef373c2037412f09a600357cfa1c7a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Apr 2012 15:18:08 +0200 Subject: ALSA: hda/conexant - Set up the missing docking-station pins ThinkPad 410,420,510,520 and X201 with cx50585 & co chips have the docking-station ports, but BIOS doesn't initialize for these pins. Thus, like the former X200, we need to set up the pins manually in the driver. The odd part is that the same PCI SSID is used for X200 and T400, thus we need to prepare individual fixup tables for cx5051 and others. Bugzilla entries: https://bugzilla.redhat.com/show_bug.cgi?id=808559 https://bugzilla.redhat.com/show_bug.cgi?id=806217 https://bugzilla.redhat.com/show_bug.cgi?id=810697 Reported-by: Josh Boyer Reported-by: Jens Taprogge Tested-by: Jens Taprogge Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 28 +++++++++++++++++++++++++--- 1 file changed, 25 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index f52c9ef3cc8..58b5de4a6ee 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -4367,8 +4367,10 @@ static void apply_pin_fixup(struct hda_codec *codec, enum { CXT_PINCFG_LENOVO_X200, + CXT_PINCFG_LENOVO_TP410, }; +/* ThinkPad X200 & co with cxt5051 */ static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = { { 0x16, 0x042140ff }, /* HP (seq# overridden) */ { 0x17, 0x21a11000 }, /* dock-mic */ @@ -4376,15 +4378,33 @@ static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = { {} }; +/* ThinkPad 410/420/510/520, X201 & co with cxt5066 */ +static const struct cxt_pincfg cxt_pincfg_lenovo_tp410[] = { + { 0x19, 0x042110ff }, /* HP (seq# overridden) */ + { 0x1a, 0x21a190f0 }, /* dock-mic */ + { 0x1c, 0x212140ff }, /* dock-HP */ + {} +}; + static const struct cxt_pincfg *cxt_pincfg_tbl[] = { [CXT_PINCFG_LENOVO_X200] = cxt_pincfg_lenovo_x200, + [CXT_PINCFG_LENOVO_TP410] = cxt_pincfg_lenovo_tp410, }; -static const struct snd_pci_quirk cxt_fixups[] = { +static const struct snd_pci_quirk cxt5051_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT_PINCFG_LENOVO_X200), {} }; +static const struct snd_pci_quirk cxt5066_fixups[] = { + SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410), + SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T410", CXT_PINCFG_LENOVO_TP410), + SND_PCI_QUIRK(0x17aa, 0x215f, "Lenovo T510", CXT_PINCFG_LENOVO_TP410), + SND_PCI_QUIRK(0x17aa, 0x21ce, "Lenovo T420", CXT_PINCFG_LENOVO_TP410), + SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520", CXT_PINCFG_LENOVO_TP410), + {} +}; + /* add "fake" mute amp-caps to DACs on cx5051 so that mixer mute switches * can be created (bko#42825) */ @@ -4421,11 +4441,13 @@ static int patch_conexant_auto(struct hda_codec *codec) break; case 0x14f15051: add_cx5051_fake_mutes(codec); + apply_pin_fixup(codec, cxt5051_fixups, cxt_pincfg_tbl); + break; + default: + apply_pin_fixup(codec, cxt5066_fixups, cxt_pincfg_tbl); break; } - apply_pin_fixup(codec, cxt_fixups, cxt_pincfg_tbl); - err = cx_auto_search_adcs(codec); if (err < 0) return err; -- cgit v1.2.3-70-g09d2 From 5ac57550f279c3d991ef0b398681bcaca18169f7 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Fri, 20 Apr 2012 10:01:46 +0200 Subject: ALSA: HDA: Add external mic quirk for Asus Zenbook UX31E According to the reporter, external mic starts to work if the laptop-dmic model is used. According to BIOS pin config, all pins are consistent with the alc269vb_laptop_dmic fixup, except for the external mic, which is not present. Cc: stable@kernel.org BugLink: https://bugs.launchpad.net/bugs/950490 Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e65e3543305..818f90bc7d5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6109,6 +6109,7 @@ static const struct alc_fixup alc269_fixups[] = { static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_MIC2_MUTE_LED), + SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), -- cgit v1.2.3-70-g09d2 From 1a38336b8611a04f0a624330c1f815421f4bf5f4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 12 Apr 2012 19:47:11 +0100 Subject: ASoC: wm8994: Improve sequencing of AIF channel enables This ensures a clean startup of the channels, without this change some use cases could result in issues in a small proportion of cases. Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm8994.c | 276 +++++++++++++++++++++++++++++++++++++--------- 1 file changed, 222 insertions(+), 54 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 7c49642af05..6c1fe3afd4b 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1000,61 +1000,170 @@ static void wm8994_update_class_w(struct snd_soc_codec *codec) } } -static int late_enable_ev(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) +static int aif1clk_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; - struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + struct wm8994 *control = codec->control_data; + int mask = WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC1R_ENA; + int dac; + int adc; + int val; + + switch (control->type) { + case WM8994: + case WM8958: + mask |= WM8994_AIF1DAC2L_ENA | WM8994_AIF1DAC2R_ENA; + break; + default: + break; + } switch (event) { case SND_SOC_DAPM_PRE_PMU: - if (wm8994->aif1clk_enable) { - snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, - WM8994_AIF1CLK_ENA_MASK, - WM8994_AIF1CLK_ENA); - wm8994->aif1clk_enable = 0; - } - if (wm8994->aif2clk_enable) { - snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, - WM8994_AIF2CLK_ENA_MASK, - WM8994_AIF2CLK_ENA); - wm8994->aif2clk_enable = 0; - } + val = snd_soc_read(codec, WM8994_AIF1_CONTROL_1); + if ((val & WM8994_AIF1ADCL_SRC) && + (val & WM8994_AIF1ADCR_SRC)) + adc = WM8994_AIF1ADC1R_ENA | WM8994_AIF1ADC2R_ENA; + else if (!(val & WM8994_AIF1ADCL_SRC) && + !(val & WM8994_AIF1ADCR_SRC)) + adc = WM8994_AIF1ADC1L_ENA | WM8994_AIF1ADC2L_ENA; + else + adc = WM8994_AIF1ADC1R_ENA | WM8994_AIF1ADC2R_ENA | + WM8994_AIF1ADC1L_ENA | WM8994_AIF1ADC2L_ENA; + + val = snd_soc_read(codec, WM8994_AIF1_CONTROL_2); + if ((val & WM8994_AIF1DACL_SRC) && + (val & WM8994_AIF1DACR_SRC)) + dac = WM8994_AIF1DAC1R_ENA | WM8994_AIF1DAC2R_ENA; + else if (!(val & WM8994_AIF1DACL_SRC) && + !(val & WM8994_AIF1DACR_SRC)) + dac = WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC2L_ENA; + else + dac = WM8994_AIF1DAC1R_ENA | WM8994_AIF1DAC2R_ENA | + WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC2L_ENA; + + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_4, + mask, adc); + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, + mask, dac); + snd_soc_update_bits(codec, WM8994_CLOCKING_1, + WM8994_AIF1DSPCLK_ENA | + WM8994_SYSDSPCLK_ENA, + WM8994_AIF1DSPCLK_ENA | + WM8994_SYSDSPCLK_ENA); + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_4, mask, + WM8994_AIF1ADC1R_ENA | + WM8994_AIF1ADC1L_ENA | + WM8994_AIF1ADC2R_ENA | + WM8994_AIF1ADC2L_ENA); + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, mask, + WM8994_AIF1DAC1R_ENA | + WM8994_AIF1DAC1L_ENA | + WM8994_AIF1DAC2R_ENA | + WM8994_AIF1DAC2L_ENA); break; - } - /* We may also have postponed startup of DSP, handle that. */ - wm8958_aif_ev(w, kcontrol, event); + case SND_SOC_DAPM_PRE_PMD: + case SND_SOC_DAPM_POST_PMD: + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, + mask, 0); + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_4, + mask, 0); + + val = snd_soc_read(codec, WM8994_CLOCKING_1); + if (val & WM8994_AIF2DSPCLK_ENA) + val = WM8994_SYSDSPCLK_ENA; + else + val = 0; + snd_soc_update_bits(codec, WM8994_CLOCKING_1, + WM8994_SYSDSPCLK_ENA | + WM8994_AIF1DSPCLK_ENA, val); + break; + } return 0; } -static int late_disable_ev(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) +static int aif2clk_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; - struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + int dac; + int adc; + int val; switch (event) { + case SND_SOC_DAPM_PRE_PMU: + val = snd_soc_read(codec, WM8994_AIF2_CONTROL_1); + if ((val & WM8994_AIF2ADCL_SRC) && + (val & WM8994_AIF2ADCR_SRC)) + adc = WM8994_AIF2ADCR_ENA; + else if (!(val & WM8994_AIF2ADCL_SRC) && + !(val & WM8994_AIF2ADCR_SRC)) + adc = WM8994_AIF2ADCL_ENA; + else + adc = WM8994_AIF2ADCL_ENA | WM8994_AIF2ADCR_ENA; + + + val = snd_soc_read(codec, WM8994_AIF2_CONTROL_2); + if ((val & WM8994_AIF2DACL_SRC) && + (val & WM8994_AIF2DACR_SRC)) + dac = WM8994_AIF2DACR_ENA; + else if (!(val & WM8994_AIF2DACL_SRC) && + !(val & WM8994_AIF2DACR_SRC)) + dac = WM8994_AIF2DACL_ENA; + else + dac = WM8994_AIF2DACL_ENA | WM8994_AIF2DACR_ENA; + + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_4, + WM8994_AIF2ADCL_ENA | + WM8994_AIF2ADCR_ENA, adc); + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, + WM8994_AIF2DACL_ENA | + WM8994_AIF2DACR_ENA, dac); + snd_soc_update_bits(codec, WM8994_CLOCKING_1, + WM8994_AIF2DSPCLK_ENA | + WM8994_SYSDSPCLK_ENA, + WM8994_AIF2DSPCLK_ENA | + WM8994_SYSDSPCLK_ENA); + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_4, + WM8994_AIF2ADCL_ENA | + WM8994_AIF2ADCR_ENA, + WM8994_AIF2ADCL_ENA | + WM8994_AIF2ADCR_ENA); + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, + WM8994_AIF2DACL_ENA | + WM8994_AIF2DACR_ENA, + WM8994_AIF2DACL_ENA | + WM8994_AIF2DACR_ENA); + break; + + case SND_SOC_DAPM_PRE_PMD: case SND_SOC_DAPM_POST_PMD: - if (wm8994->aif1clk_disable) { - snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, - WM8994_AIF1CLK_ENA_MASK, 0); - wm8994->aif1clk_disable = 0; - } - if (wm8994->aif2clk_disable) { - snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, - WM8994_AIF2CLK_ENA_MASK, 0); - wm8994->aif2clk_disable = 0; - } + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, + WM8994_AIF2DACL_ENA | + WM8994_AIF2DACR_ENA, 0); + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, + WM8994_AIF2ADCL_ENA | + WM8994_AIF2ADCR_ENA, 0); + + val = snd_soc_read(codec, WM8994_CLOCKING_1); + if (val & WM8994_AIF1DSPCLK_ENA) + val = WM8994_SYSDSPCLK_ENA; + else + val = 0; + snd_soc_update_bits(codec, WM8994_CLOCKING_1, + WM8994_SYSDSPCLK_ENA | + WM8994_AIF2DSPCLK_ENA, val); break; } return 0; } -static int aif1clk_ev(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) +static int aif1clk_late_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); @@ -1071,8 +1180,8 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w, return 0; } -static int aif2clk_ev(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) +static int aif2clk_late_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); @@ -1089,6 +1198,63 @@ static int aif2clk_ev(struct snd_soc_dapm_widget *w, return 0; } +static int late_enable_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (wm8994->aif1clk_enable) { + aif1clk_ev(w, kcontrol, event); + snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, + WM8994_AIF1CLK_ENA_MASK, + WM8994_AIF1CLK_ENA); + wm8994->aif1clk_enable = 0; + } + if (wm8994->aif2clk_enable) { + aif2clk_ev(w, kcontrol, event); + snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, + WM8994_AIF2CLK_ENA_MASK, + WM8994_AIF2CLK_ENA); + wm8994->aif2clk_enable = 0; + } + break; + } + + /* We may also have postponed startup of DSP, handle that. */ + wm8958_aif_ev(w, kcontrol, event); + + return 0; +} + +static int late_disable_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMD: + if (wm8994->aif1clk_disable) { + snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, + WM8994_AIF1CLK_ENA_MASK, 0); + aif1clk_ev(w, kcontrol, event); + wm8994->aif1clk_disable = 0; + } + if (wm8994->aif2clk_disable) { + snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, + WM8994_AIF2CLK_ENA_MASK, 0); + aif2clk_ev(w, kcontrol, event); + wm8994->aif2clk_disable = 0; + } + break; + } + + return 0; +} + static int adc_mux_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -1385,9 +1551,9 @@ static const struct snd_kcontrol_new aif2dacr_src_mux = SOC_DAPM_ENUM("AIF2DACR Mux", aif2dacr_src_enum); static const struct snd_soc_dapm_widget wm8994_lateclk_revd_widgets[] = { -SND_SOC_DAPM_SUPPLY("AIF1CLK", SND_SOC_NOPM, 0, 0, aif1clk_ev, +SND_SOC_DAPM_SUPPLY("AIF1CLK", SND_SOC_NOPM, 0, 0, aif1clk_late_ev, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), -SND_SOC_DAPM_SUPPLY("AIF2CLK", SND_SOC_NOPM, 0, 0, aif2clk_ev, +SND_SOC_DAPM_SUPPLY("AIF2CLK", SND_SOC_NOPM, 0, 0, aif2clk_late_ev, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_PGA_E("Late DAC1L Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0, @@ -1416,8 +1582,10 @@ SND_SOC_DAPM_POST("Late Disable PGA", late_disable_ev) }; static const struct snd_soc_dapm_widget wm8994_lateclk_widgets[] = { -SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, aif1clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, aif2clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_PGA("Direct Voice", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MIXER("SPKL", WM8994_POWER_MANAGEMENT_3, 8, 0, left_speaker_mixer, ARRAY_SIZE(left_speaker_mixer)), @@ -1470,30 +1638,30 @@ SND_SOC_DAPM_SUPPLY("VMID", SND_SOC_NOPM, 0, 0, vmid_event, SND_SOC_DAPM_SUPPLY("CLK_SYS", SND_SOC_NOPM, 0, 0, clk_sys_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), -SND_SOC_DAPM_SUPPLY("DSP1CLK", WM8994_CLOCKING_1, 3, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("DSP2CLK", WM8994_CLOCKING_1, 2, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("DSPINTCLK", WM8994_CLOCKING_1, 1, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("DSP1CLK", SND_SOC_NOPM, 3, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("DSP2CLK", SND_SOC_NOPM, 2, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("DSPINTCLK", SND_SOC_NOPM, 1, 0, NULL, 0), SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", NULL, - 0, WM8994_POWER_MANAGEMENT_4, 9, 0), + 0, SND_SOC_NOPM, 9, 0), SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", NULL, - 0, WM8994_POWER_MANAGEMENT_4, 8, 0), + 0, SND_SOC_NOPM, 8, 0), SND_SOC_DAPM_AIF_IN_E("AIF1DAC1L", NULL, 0, - WM8994_POWER_MANAGEMENT_5, 9, 0, wm8958_aif_ev, + SND_SOC_NOPM, 9, 0, wm8958_aif_ev, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_AIF_IN_E("AIF1DAC1R", NULL, 0, - WM8994_POWER_MANAGEMENT_5, 8, 0, wm8958_aif_ev, + SND_SOC_NOPM, 8, 0, wm8958_aif_ev, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_AIF_OUT("AIF1ADC2L", NULL, - 0, WM8994_POWER_MANAGEMENT_4, 11, 0), + 0, SND_SOC_NOPM, 11, 0), SND_SOC_DAPM_AIF_OUT("AIF1ADC2R", NULL, - 0, WM8994_POWER_MANAGEMENT_4, 10, 0), + 0, SND_SOC_NOPM, 10, 0), SND_SOC_DAPM_AIF_IN_E("AIF1DAC2L", NULL, 0, - WM8994_POWER_MANAGEMENT_5, 11, 0, wm8958_aif_ev, + SND_SOC_NOPM, 11, 0, wm8958_aif_ev, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_AIF_IN_E("AIF1DAC2R", NULL, 0, - WM8994_POWER_MANAGEMENT_5, 10, 0, wm8958_aif_ev, + SND_SOC_NOPM, 10, 0, wm8958_aif_ev, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_MIXER("AIF1ADC1L Mixer", SND_SOC_NOPM, 0, 0, @@ -1520,14 +1688,14 @@ SND_SOC_DAPM_MIXER("DAC1R Mixer", SND_SOC_NOPM, 0, 0, dac1r_mix, ARRAY_SIZE(dac1r_mix)), SND_SOC_DAPM_AIF_OUT("AIF2ADCL", NULL, 0, - WM8994_POWER_MANAGEMENT_4, 13, 0), + SND_SOC_NOPM, 13, 0), SND_SOC_DAPM_AIF_OUT("AIF2ADCR", NULL, 0, - WM8994_POWER_MANAGEMENT_4, 12, 0), + SND_SOC_NOPM, 12, 0), SND_SOC_DAPM_AIF_IN_E("AIF2DACL", NULL, 0, - WM8994_POWER_MANAGEMENT_5, 13, 0, wm8958_aif_ev, + SND_SOC_NOPM, 13, 0, wm8958_aif_ev, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_AIF_IN_E("AIF2DACR", NULL, 0, - WM8994_POWER_MANAGEMENT_5, 12, 0, wm8958_aif_ev, + SND_SOC_NOPM, 12, 0, wm8958_aif_ev, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_AIF_IN("AIF1DACDAT", NULL, 0, SND_SOC_NOPM, 0, 0), -- cgit v1.2.3-70-g09d2 From de050acaa1fdba4852cb195baf2bfed75368e0be Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 17 Apr 2012 20:28:10 +0100 Subject: ASoC: wm_hubs: Make sure we don't disable differential line outputs While we need to clean up unused single ended line outputs we don't want to do this if the outputs are in differential mode. Signed-off-by: Mark Brown --- sound/soc/codecs/wm_hubs.c | 15 +++++++++------ 1 file changed, 9 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index f13f2886339..6c028c47060 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -1035,7 +1035,7 @@ void wm_hubs_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); - int val; + int mask, val; switch (level) { case SND_SOC_BIAS_STANDBY: @@ -1047,6 +1047,13 @@ void wm_hubs_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_ON: /* Turn off any unneded single ended outputs */ val = 0; + mask = 0; + + if (hubs->lineout1_se) + mask |= WM8993_LINEOUT1N_ENA | WM8993_LINEOUT1P_ENA; + + if (hubs->lineout2_se) + mask |= WM8993_LINEOUT2N_ENA | WM8993_LINEOUT2P_ENA; if (hubs->lineout1_se && hubs->lineout1n_ena) val |= WM8993_LINEOUT1N_ENA; @@ -1061,11 +1068,7 @@ void wm_hubs_set_bias_level(struct snd_soc_codec *codec, val |= WM8993_LINEOUT2P_ENA; snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_3, - WM8993_LINEOUT1N_ENA | - WM8993_LINEOUT1P_ENA | - WM8993_LINEOUT2N_ENA | - WM8993_LINEOUT2P_ENA, - val); + mask, val); /* Remove the input clamps */ snd_soc_update_bits(codec, WM8993_INPUTS_CLAMP_REG, -- cgit v1.2.3-70-g09d2 From c34ce320d9fe328e3272def20b152f39ccfa045e Mon Sep 17 00:00:00 2001 From: Richard Zhao Date: Tue, 24 Apr 2012 15:24:43 +0800 Subject: ASoC: core: check of_property_count_strings failure Signed-off-by: Richard Zhao Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/soc-core.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3a4e93e52b6..b390f00b4e9 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3631,10 +3631,10 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, int i, ret; num_routes = of_property_count_strings(np, propname); - if (num_routes & 1) { + if (num_routes < 0 || num_routes & 1) { dev_err(card->dev, - "Property '%s's length is not even\n", - propname); + "Property '%s' does not exist or its length is not even\n", + propname); return -EINVAL; } num_routes /= 2; -- cgit v1.2.3-70-g09d2 From a3a53fe1545a87337cc539f415810128bbdad465 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 25 Apr 2012 11:29:47 +0200 Subject: ASoC: bf5xx-ssm2602: Set DAI format Commit 980b0bc69 ("ASoC: blackfin: Use dai_fmt") converted the blackfin ASoC machine drivers to use the dai_links dai_fmt field to setup their DAI format. For the bf5xx-ssm2602 the commit removed the manual call to snd_soc_dai_set_fmt, but missed to set the dai_links dai_fmt field. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-ssm2602.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c index df3ac73f877..b39ad356b92 100644 --- a/sound/soc/blackfin/bf5xx-ssm2602.c +++ b/sound/soc/blackfin/bf5xx-ssm2602.c @@ -99,6 +99,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai[] = { .platform_name = "bfin-i2s-pcm-audio", .codec_name = "ssm2602.0-001b", .ops = &bf5xx_ssm2602_ops, + .dai_fmt = BF5XX_SSM2602_DAIFMT, }, { .name = "ssm2602", @@ -108,6 +109,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai[] = { .platform_name = "bfin-i2s-pcm-audio", .codec_name = "ssm2602.0-001b", .ops = &bf5xx_ssm2602_ops, + .dai_fmt = BF5XX_SSM2602_DAIFMT, }, }; -- cgit v1.2.3-70-g09d2 From e875c1e3e758447ba81ca450d89434b3b0496d37 Mon Sep 17 00:00:00 2001 From: Eric Bénard Date: Sun, 29 Apr 2012 17:37:57 +0200 Subject: ASoC: tlv312aic23: unbreak resume MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit * commit f9dfbf9 "ASoC: tlv320aic23: convert to soc-cache" leads to a bug preventing resumeof the codec as regmap expects a 9 bits data register but 0xFFFF is passed in tlv320aic23_set_bias_level and this values gets cached preventing any write to the TLV320AIC23_PWR register as the final value produced by regmap is (register << 9) | value * this patch solves the problem by only working on the 9 bits the register contains. Signed-off-by: Eric Bénard Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/tlv320aic23.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 16d55f91a65..df1e07ffac3 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -472,7 +472,7 @@ static int tlv320aic23_set_dai_sysclk(struct snd_soc_dai *codec_dai, static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 reg = snd_soc_read(codec, TLV320AIC23_PWR) & 0xff7f; + u16 reg = snd_soc_read(codec, TLV320AIC23_PWR) & 0x17f; switch (level) { case SND_SOC_BIAS_ON: @@ -491,7 +491,7 @@ static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_OFF: /* everything off, dac mute, inactive */ snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0); - snd_soc_write(codec, TLV320AIC23_PWR, 0xffff); + snd_soc_write(codec, TLV320AIC23_PWR, 0x1ff); break; } codec->dapm.bias_level = level; -- cgit v1.2.3-70-g09d2 From 30facd4d51d630b6cba386badd7f42456962089b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 30 Apr 2012 20:11:55 +0100 Subject: ASoC: wm8350: Don't use locally allocated codec struct The core allocates the live copies, we shouldn't try to duplicate it and were buggy trying to do so as we were using uninitialised data for the control data. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8350.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 8c4c9591ec0..aa12c6b6bee 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -60,7 +60,7 @@ struct wm8350_jack_data { }; struct wm8350_data { - struct snd_soc_codec codec; + struct wm8350 *wm8350; struct wm8350_output out1; struct wm8350_output out2; struct wm8350_jack_data hpl; @@ -1309,7 +1309,7 @@ static void wm8350_hp_work(struct wm8350_data *priv, struct wm8350_jack_data *jack, u16 mask) { - struct wm8350 *wm8350 = priv->codec.control_data; + struct wm8350 *wm8350 = priv->wm8350; u16 reg; int report; @@ -1342,7 +1342,7 @@ static void wm8350_hpr_work(struct work_struct *work) static irqreturn_t wm8350_hp_jack_handler(int irq, void *data) { struct wm8350_data *priv = data; - struct wm8350 *wm8350 = priv->codec.control_data; + struct wm8350 *wm8350 = priv->wm8350; struct wm8350_jack_data *jack = NULL; switch (irq - wm8350->irq_base) { @@ -1427,7 +1427,7 @@ EXPORT_SYMBOL_GPL(wm8350_hp_jack_detect); static irqreturn_t wm8350_mic_handler(int irq, void *data) { struct wm8350_data *priv = data; - struct wm8350 *wm8350 = priv->codec.control_data; + struct wm8350 *wm8350 = priv->wm8350; u16 reg; int report = 0; @@ -1536,6 +1536,8 @@ static int wm8350_codec_probe(struct snd_soc_codec *codec) return -ENOMEM; snd_soc_codec_set_drvdata(codec, priv); + priv->wm8350 = wm8350; + for (i = 0; i < ARRAY_SIZE(supply_names); i++) priv->supplies[i].supply = supply_names[i]; @@ -1544,7 +1546,6 @@ static int wm8350_codec_probe(struct snd_soc_codec *codec) if (ret != 0) return ret; - wm8350->codec.codec = codec; codec->control_data = wm8350; /* Put the codec into reset if it wasn't already */ -- cgit v1.2.3-70-g09d2 From 06412088ce98f745405b8f65cfc51ddd6b842bbf Mon Sep 17 00:00:00 2001 From: Heiko Stübner Date: Mon, 30 Apr 2012 13:17:21 +0200 Subject: ASoC: s3c2412-i2s: Fix dai registration As s3c2412-i2s is using the s3c_i2sv2 it should call the more specialised s3c_i2sv2_register_dai instead of simply calling snd_soc_register_dai. Without this call the snd_soc_dai_ops structure isn't initialised correctly. Signed-off-by: Heiko Stuebner Signed-off-by: Mark Brown --- sound/soc/samsung/s3c2412-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 72185078ddf..79fbeea99d4 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -166,7 +166,7 @@ static struct snd_soc_dai_driver s3c2412_i2s_dai = { static __devinit int s3c2412_iis_dev_probe(struct platform_device *pdev) { - return snd_soc_register_dai(&pdev->dev, &s3c2412_i2s_dai); + return s3c_i2sv2_register_dai(&pdev->dev, -1, &s3c2412_i2s_dai); } static __devexit int s3c2412_iis_dev_remove(struct platform_device *pdev) -- cgit v1.2.3-70-g09d2 From fad9365bcc2a69ae16adc092e8ac192354980665 Mon Sep 17 00:00:00 2001 From: Oleg Matcovschi Date: Tue, 24 Apr 2012 19:02:02 -0700 Subject: ASoC: omap-pcm: Free dma buffers in case of error. Signed-off-by: Oleg Matcovschi Acked-by: Peter Ujfalusi Acked-by: Jarkko Nikula Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/omap/omap-pcm.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index a59bd352d34..5a649da9122 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -401,6 +401,10 @@ static int omap_pcm_new(struct snd_soc_pcm_runtime *rtd) } out: + /* free preallocated buffers in case of error */ + if (ret) + omap_pcm_free_dma_buffers(pcm); + return ret; } -- cgit v1.2.3-70-g09d2 From c914f55f7cdfafe9d7d5b248751902c7ab57691e Mon Sep 17 00:00:00 2001 From: Mark Hills Date: Mon, 30 Apr 2012 19:39:22 +0100 Subject: ALSA: echoaudio: Remove incorrect part of assertion This assertion seems to imply that chip->dsp_code_to_load is a pointer. It's actually an integer handle on the actual firmware, and 0 has no special meaning. The assertion prevents initialisation of a Darla20 card, but would also affect other models. It seems it was introduced in commit dd7b254d. ALSA sound/pci/echoaudio/echoaudio.c:2061 Echoaudio driver starting... ALSA sound/pci/echoaudio/echoaudio.c:1969 chip=ebe4e000 ALSA sound/pci/echoaudio/echoaudio.c:2007 pci=ed568000 irq=19 subdev=0010 Init hardware... ALSA sound/pci/echoaudio/darla20_dsp.c:36 init_hw() - Darla20 ------------[ cut here ]------------ WARNING: at sound/pci/echoaudio/echoaudio_dsp.c:478 init_hw+0x1d1/0x86c [snd_darla20]() Hardware name: Dell DM051 BUG? (!chip->dsp_code_to_load || !chip->comm_page) Signed-off-by: Mark Hills Cc: Signed-off-by: Takashi Iwai --- sound/pci/echoaudio/echoaudio_dsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c index 64417a73322..d8c670c9d62 100644 --- a/sound/pci/echoaudio/echoaudio_dsp.c +++ b/sound/pci/echoaudio/echoaudio_dsp.c @@ -475,7 +475,7 @@ static int load_firmware(struct echoaudio *chip) const struct firmware *fw; int box_type, err; - if (snd_BUG_ON(!chip->dsp_code_to_load || !chip->comm_page)) + if (snd_BUG_ON(!chip->comm_page)) return -EPERM; /* See if the ASIC is present and working - only if the DSP is already loaded */ -- cgit v1.2.3-70-g09d2 From f5c53d898cc34079373c63a290528963db31d681 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 7 May 2012 10:07:33 +0200 Subject: ALSA: hda/realtek - Add a fixup for Acer Aspire 5739G Acer Aspire 5739G requires the same fix-up for 4930G to support the surround / bass speakers. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=43180 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 818f90bc7d5..27d0f637864 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5405,6 +5405,8 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0142, "Acer Aspire 7730G", ALC882_FIXUP_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", ALC882_FIXUP_PB_M5210), + SND_PCI_QUIRK(0x1025, 0x021e, "Acer Aspire 5739G", + ALC882_FIXUP_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0x0259, "Acer Aspire 5935", ALC889_FIXUP_DAC_ROUTE), SND_PCI_QUIRK(0x1025, 0x026b, "Acer Aspire 8940G", ALC882_FIXUP_ACER_ASPIRE_8930G), SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", ALC882_FIXUP_ACER_ASPIRE_7736), -- cgit v1.2.3-70-g09d2 From bca40138558f0b39357fd1ca477868e4f52f4b1e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 7 May 2012 11:13:14 +0200 Subject: ALSA: hda/realtek - Add missing CD-input pin for MSI-7350 mobo Reported-by: Philipp Matthias Hahn Cc: [v3.3+] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 27d0f637864..8ea613eb73f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5440,6 +5440,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x106b, 0x4a00, "Macbook 5,2", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC882_FIXUP_EAPD), + SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD), SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3", ALC889_FIXUP_CD), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX), -- cgit v1.2.3-70-g09d2 From 42eb92380f73f28e3a5a51973af1183fdbac82f2 Mon Sep 17 00:00:00 2001 From: Andre Schramm Date: Mon, 7 May 2012 18:52:51 +0200 Subject: ALSA: hdsp - Provide ioctl_compat snd_hdsp uses its own ioctls to acquire config- and status information. Expose the corresponding ioctl handler via ioctl_compat, so that 32bit applications can use it on 64bit kernels. Signed-off-by: Andre Schramm Reviewed-by: Adrian Knoth Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdsp.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index b68cdec03b9..0b2aea2ce17 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -5170,6 +5170,7 @@ static int snd_hdsp_create_hwdep(struct snd_card *card, struct hdsp *hdsp) strcpy(hw->name, "HDSP hwdep interface"); hw->ops.ioctl = snd_hdsp_hwdep_ioctl; + hw->ops.ioctl_compat = snd_hdsp_hwdep_ioctl; return 0; } -- cgit v1.2.3-70-g09d2 From af741c150f66db8d1da6f82ac75e2571f7f1dd38 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 7 May 2012 18:09:48 +0200 Subject: ALSA: hda/realtek - Call alc_auto_parse_customize_define() always after fixup The call for alc_auto_parse_customize_define() must be done after the fixup pre-probe initialization. Otherwise SKU_IGNORE fixup won't work properly (e.g. HP RP5800 with ALC662 codec). Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8ea613eb73f..7810913d07a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5641,13 +5641,13 @@ static int patch_alc262(struct hda_codec *codec) snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PROC_COEF, tmp | 0x80); } #endif - alc_auto_parse_customize_define(codec); - alc_fix_pll_init(codec, 0x20, 0x0a, 10); alc_pick_fixup(codec, NULL, alc262_fixup_tbl, alc262_fixups); alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); + alc_auto_parse_customize_define(codec); + /* automatic parse from the BIOS config */ err = alc262_parse_auto_config(codec); if (err < 0) @@ -6252,8 +6252,6 @@ static int patch_alc269(struct hda_codec *codec) spec->mixer_nid = 0x0b; - alc_auto_parse_customize_define(codec); - err = alc_codec_rename_from_preset(codec); if (err < 0) goto error; @@ -6286,6 +6284,8 @@ static int patch_alc269(struct hda_codec *codec) alc269_fixup_tbl, alc269_fixups); alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); + alc_auto_parse_customize_define(codec); + /* automatic parse from the BIOS config */ err = alc269_parse_auto_config(codec); if (err < 0) @@ -6862,8 +6862,6 @@ static int patch_alc662(struct hda_codec *codec) /* handle multiple HPs as is */ spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP; - alc_auto_parse_customize_define(codec); - alc_fix_pll_init(codec, 0x20, 0x04, 15); err = alc_codec_rename_from_preset(codec); @@ -6880,6 +6878,9 @@ static int patch_alc662(struct hda_codec *codec) alc_pick_fixup(codec, alc662_fixup_models, alc662_fixup_tbl, alc662_fixups); alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); + + alc_auto_parse_customize_define(codec); + /* automatic parse from the BIOS config */ err = alc662_parse_auto_config(codec); if (err < 0) -- cgit v1.2.3-70-g09d2 From 619a341b78f17fb86d92e89c04612676cd05e26f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 8 May 2012 16:30:59 +0200 Subject: Revert "ALSA: hda - Set codec to D3 forcibly even if not used" This reverts commit 785f857d1cb0856b612b46a0545b74aa2596e44a. The commit causes a problem with the wrong D3 state after suspend because the call of hda_set_power_state() involves with the power-up sequence, which changes the power_count, and this confuses the resume sequence that checks the power_count as well. Originally, this go-to-D3 sequence should be a simple task without the power-up sequence. But, it'd need some proper sanity checks in the case of power-saved state, so it's not too easy to write now in the 3.4-rc cycle. In short, the safest option now is to revert this affecting commit. Of course, we need to clean up and robustify the power-saving code better for 3.5 kernel. Reported-by: Konstantin Khlebnikov Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 4 ---- sound/pci/hda/hda_intel.c | 14 +++++++++++++- 2 files changed, 13 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 7a8fcc4c15f..841475cc13b 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5444,10 +5444,6 @@ int snd_hda_suspend(struct hda_bus *bus) list_for_each_entry(codec, &bus->codec_list, list) { if (hda_codec_is_power_on(codec)) hda_call_codec_suspend(codec); - else /* forcibly change the power to D3 even if not used */ - hda_set_power_state(codec, - codec->afg ? codec->afg : codec->mfg, - AC_PWRST_D3); if (codec->patch_ops.post_suspend) codec->patch_ops.post_suspend(codec); } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c19e71a94e1..6e958bf9419 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2351,6 +2351,17 @@ static void azx_power_notify(struct hda_bus *bus) * power management */ +static int snd_hda_codecs_inuse(struct hda_bus *bus) +{ + struct hda_codec *codec; + + list_for_each_entry(codec, &bus->codec_list, list) { + if (snd_hda_codec_needs_resume(codec)) + return 1; + } + return 0; +} + static int azx_suspend(struct pci_dev *pci, pm_message_t state) { struct snd_card *card = pci_get_drvdata(pci); @@ -2397,7 +2408,8 @@ static int azx_resume(struct pci_dev *pci) return -EIO; azx_init_pci(chip); - azx_init_chip(chip, 1); + if (snd_hda_codecs_inuse(chip->bus)) + azx_init_chip(chip, 1); snd_hda_resume(chip->bus); snd_power_change_state(card, SNDRV_CTL_POWER_D0); -- cgit v1.2.3-70-g09d2 From 32cf4023e689ad5b3a81a749d8cc99d7f184cb99 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Fri, 4 May 2012 11:05:55 +0200 Subject: ALSA: HDA: Lessen CPU usage when waiting for chip to respond When an IRQ for some reason gets lost, we wait up to a second using udelay, which is CPU intensive. This patch improves the situation by waiting about 30 ms in the CPU intensive mode, then stepping down to using msleep(2) instead. In essence, we trade some granularity in exchange for less CPU consumption when the waiting time is a bit longer. As a result, PulseAudio should no longer be killed by the kernel for taking up to much RT-prio CPU time. At least not for *this* reason. Signed-off-by: David Henningsson Tested-by: Arun Raghavan Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 6e958bf9419..1f350522bed 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -783,11 +783,13 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus, { struct azx *chip = bus->private_data; unsigned long timeout; + unsigned long loopcounter; int do_poll = 0; again: timeout = jiffies + msecs_to_jiffies(1000); - for (;;) { + + for (loopcounter = 0;; loopcounter++) { if (chip->polling_mode || do_poll) { spin_lock_irq(&chip->reg_lock); azx_update_rirb(chip); @@ -803,7 +805,7 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus, } if (time_after(jiffies, timeout)) break; - if (bus->needs_damn_long_delay) + if (bus->needs_damn_long_delay || loopcounter > 3000) msleep(2); /* temporary workaround */ else { udelay(10); -- cgit v1.2.3-70-g09d2 From c8587193ba511b788a9888e5e701a9747e70c0d8 Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski Date: Wed, 9 May 2012 12:57:05 +0200 Subject: ASoC: sh: fix migor.c compilation Fix a recent compilation breakage, caused by a change in SH clock API. Signed-off-by: Guennadi Liakhovetski Signed-off-by: Mark Brown --- sound/soc/sh/migor.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c index 9d9ad8d61c0..8526e1edaf4 100644 --- a/sound/soc/sh/migor.c +++ b/sound/soc/sh/migor.c @@ -35,7 +35,7 @@ static unsigned long siumckb_recalc(struct clk *clk) return codec_freq; } -static struct clk_ops siumckb_clk_ops = { +static struct sh_clk_ops siumckb_clk_ops = { .recalc = siumckb_recalc, }; -- cgit v1.2.3-70-g09d2