diff options
54 files changed, 1568 insertions, 453 deletions
diff --git a/Documentation/sound/alsa/soc/dapm.txt b/Documentation/sound/alsa/soc/dapm.txt index 46f9684d0b2..9e6763264a2 100644 --- a/Documentation/sound/alsa/soc/dapm.txt +++ b/Documentation/sound/alsa/soc/dapm.txt @@ -116,6 +116,9 @@ SOC_DAPM_SINGLE("HiFi Playback Switch", WM8731_APANA, 4, 1, 0), SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1, wm8731_output_mixer_controls, ARRAY_SIZE(wm8731_output_mixer_controls)), +If you dont want the mixer elements prefixed with the name of the mixer widget, +you can use SND_SOC_DAPM_MIXER_NAMED_CTL instead. the parameters are the same +as for SND_SOC_DAPM_MIXER. 2.3 Platform/Machine domain Widgets ----------------------------------- diff --git a/arch/arm/mach-pxa/e740.c b/arch/arm/mach-pxa/e740.c index 6d48e00f4f0..a6fff782e7a 100644 --- a/arch/arm/mach-pxa/e740.c +++ b/arch/arm/mach-pxa/e740.c @@ -135,6 +135,11 @@ static unsigned long e740_pin_config[] __initdata = { /* IrDA */ GPIO38_GPIO | MFP_LPM_DRIVE_HIGH, + /* Audio power control */ + GPIO16_GPIO, /* AC97 codec AVDD2 supply (analogue power) */ + GPIO40_GPIO, /* Mic amp power */ + GPIO41_GPIO, /* Headphone amp power */ + /* PC Card */ GPIO8_GPIO, /* CD0 */ GPIO44_GPIO, /* CD1 */ diff --git a/arch/arm/mach-pxa/e750.c b/arch/arm/mach-pxa/e750.c index be1ab8edb97..665066fd280 100644 --- a/arch/arm/mach-pxa/e750.c +++ b/arch/arm/mach-pxa/e750.c @@ -133,6 +133,11 @@ static unsigned long e750_pin_config[] __initdata = { /* IrDA */ GPIO38_GPIO | MFP_LPM_DRIVE_HIGH, + /* Audio power control */ + GPIO4_GPIO, /* Headphone amp power */ + GPIO7_GPIO, /* Speaker amp power */ + GPIO37_GPIO, /* Headphone detect */ + /* PC Card */ GPIO8_GPIO, /* CD0 */ GPIO44_GPIO, /* CD1 */ diff --git a/arch/arm/mach-pxa/include/mach/eseries-gpio.h b/arch/arm/mach-pxa/include/mach/eseries-gpio.h index efbd2aa9ece..f3e5509820d 100644 --- a/arch/arm/mach-pxa/include/mach/eseries-gpio.h +++ b/arch/arm/mach-pxa/include/mach/eseries-gpio.h @@ -45,6 +45,21 @@ /* e7xx IrDA power control */ #define GPIO_E7XX_IR_OFF 38 +/* e740 audio control GPIOs */ +#define GPIO_E740_WM9705_nAVDD2 16 +#define GPIO_E740_MIC_ON 40 +#define GPIO_E740_AMP_ON 41 + +/* e750 audio control GPIOs */ +#define GPIO_E750_HP_AMP_OFF 4 +#define GPIO_E750_SPK_AMP_OFF 7 +#define GPIO_E750_HP_DETECT 37 + +/* e800 audio control GPIOs */ +#define GPIO_E800_HP_DETECT 81 +#define GPIO_E800_HP_AMP_OFF 82 +#define GPIO_E800_SPK_AMP_ON 83 + /* ASIC related GPIOs */ #define GPIO_ESERIES_TMIO_IRQ 5 #define GPIO_ESERIES_TMIO_PCLR 19 diff --git a/include/linux/mfd/wm8350/audio.h b/include/linux/mfd/wm8350/audio.h index af95a1d2f3a..d899dc0223b 100644 --- a/include/linux/mfd/wm8350/audio.h +++ b/include/linux/mfd/wm8350/audio.h @@ -490,6 +490,7 @@ /* * R231 (0xE7) - Jack Status */ +#define WM8350_JACK_L_LVL 0x0800 #define WM8350_JACK_R_LVL 0x0400 /* diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 93a4edb148b..0accdba211f 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -76,6 +76,11 @@ wcontrols, wncontrols)\ { .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \ .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = wncontrols} +#define SND_SOC_DAPM_MIXER_NAMED_CTL(wname, wreg, wshift, winvert, \ + wcontrols, wncontrols)\ +{ .id = snd_soc_dapm_mixer_named_ctl, .name = wname, .reg = wreg, \ + .shift = wshift, .invert = winvert, .kcontrols = wcontrols, \ + .num_kcontrols = wncontrols} #define SND_SOC_DAPM_MICBIAS(wname, wreg, wshift, winvert) \ { .id = snd_soc_dapm_micbias, .name = wname, .reg = wreg, .shift = wshift, \ .invert = winvert, .kcontrols = NULL, .num_kcontrols = 0} @@ -101,6 +106,11 @@ { .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \ .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = wncontrols, \ .event = wevent, .event_flags = wflags} +#define SND_SOC_DAPM_MIXER_NAMED_CTL_E(wname, wreg, wshift, winvert, \ + wcontrols, wncontrols, wevent, wflags) \ +{ .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \ + .invert = winvert, .kcontrols = wcontrols, \ + .num_kcontrols = wncontrols, .event = wevent, .event_flags = wflags} #define SND_SOC_DAPM_MICBIAS_E(wname, wreg, wshift, winvert, wevent, wflags) \ { .id = snd_soc_dapm_micbias, .name = wname, .reg = wreg, .shift = wshift, \ .invert = winvert, .kcontrols = NULL, .num_kcontrols = 0, \ @@ -250,10 +260,10 @@ int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev, int snd_soc_dapm_sys_add(struct device *dev); /* dapm audio pin control and status */ -int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, char *pin); -int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, char *pin); -int snd_soc_dapm_nc_pin(struct snd_soc_codec *codec, char *pin); -int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, char *pin); +int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, const char *pin); +int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, const char *pin); +int snd_soc_dapm_nc_pin(struct snd_soc_codec *codec, const char *pin); +int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, const char *pin); int snd_soc_dapm_sync(struct snd_soc_codec *codec); /* dapm widget types */ @@ -263,6 +273,7 @@ enum snd_soc_dapm_type { snd_soc_dapm_mux, /* selects 1 analog signal from many inputs */ snd_soc_dapm_value_mux, /* selects 1 analog signal from many inputs */ snd_soc_dapm_mixer, /* mixes several analog signals together */ + snd_soc_dapm_mixer_named_ctl, /* mixer with named controls */ snd_soc_dapm_pga, /* programmable gain/attenuation (volume) */ snd_soc_dapm_adc, /* analog to digital converter */ snd_soc_dapm_dac, /* digital to analog converter */ diff --git a/include/sound/soc.h b/include/sound/soc.h index 24593ac3ea1..7039343e8a7 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -154,6 +154,8 @@ enum snd_soc_bias_level { SND_SOC_BIAS_OFF, }; +struct snd_jack; +struct snd_soc_card; struct snd_soc_device; struct snd_soc_pcm_stream; struct snd_soc_ops; @@ -164,6 +166,8 @@ struct snd_soc_platform; struct snd_soc_codec; struct soc_enum; struct snd_soc_ac97_ops; +struct snd_soc_jack; +struct snd_soc_jack_pin; typedef int (*hw_write_t)(void *,const char* ,int); typedef int (*hw_read_t)(void *,char* ,int); @@ -184,6 +188,13 @@ int snd_soc_init_card(struct snd_soc_device *socdev); int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream, const struct snd_pcm_hardware *hw); +/* Jack reporting */ +int snd_soc_jack_new(struct snd_soc_card *card, const char *id, int type, + struct snd_soc_jack *jack); +void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask); +int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count, + struct snd_soc_jack_pin *pins); + /* codec IO */ #define snd_soc_read(codec, reg) codec->read(codec, reg) #define snd_soc_write(codec, reg, value) codec->write(codec, reg, value) @@ -203,6 +214,8 @@ void snd_soc_free_ac97_codec(struct snd_soc_codec *codec); */ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template, void *data, char *long_name); +int snd_soc_add_controls(struct snd_soc_codec *codec, + const struct snd_kcontrol_new *controls, int num_controls); int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); int snd_soc_info_enum_ext(struct snd_kcontrol *kcontrol, @@ -237,6 +250,27 @@ int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol, int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +/** + * struct snd_soc_jack_pin - Describes a pin to update based on jack detection + * + * @pin: name of the pin to update + * @mask: bits to check for in reported jack status + * @invert: if non-zero then pin is enabled when status is not reported + */ +struct snd_soc_jack_pin { + struct list_head list; + const char *pin; + int mask; + bool invert; +}; + +struct snd_soc_jack { + struct snd_jack *jack; + struct snd_soc_card *card; + struct list_head pins; + int status; +}; + /* SoC PCM stream information */ struct snd_soc_pcm_stream { char *stream_name; diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index ef025c66cc6..3d2bb6fc6dc 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -6,6 +6,7 @@ menuconfig SND_SOC tristate "ALSA for SoC audio support" select SND_PCM select AC97_BUS if SND_SOC_AC97_BUS + select SND_JACK if INPUT=y || INPUT=SND ---help--- If you want ASoC support, you should say Y here and also to the diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 86a9b1f5b0f..0237879fd41 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,4 +1,4 @@ -snd-soc-core-objs := soc-core.o soc-dapm.o +snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o obj-$(CONFIG_SND_SOC) += codecs/ diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c index 3dcdc4e3cfa..9ef6b96373f 100644 --- a/sound/soc/atmel/atmel-pcm.c +++ b/sound/soc/atmel/atmel-pcm.c @@ -347,7 +347,7 @@ static int atmel_pcm_mmap(struct snd_pcm_substream *substream, vma->vm_end - vma->vm_start, vma->vm_page_prot); } -struct snd_pcm_ops atmel_pcm_ops = { +static struct snd_pcm_ops atmel_pcm_ops = { .open = atmel_pcm_open, .close = atmel_pcm_close, .ioctl = snd_pcm_lib_ioctl, diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index bc8d654576c..30490a25914 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -305,7 +305,7 @@ static int au1xpsc_pcm_close(struct snd_pcm_substream *substream) return 0; } -struct snd_pcm_ops au1xpsc_pcm_ops = { +static struct snd_pcm_ops au1xpsc_pcm_ops = { .open = au1xpsc_pcm_open, .close = au1xpsc_pcm_close, .ioctl = snd_pcm_lib_ioctl, diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c index 8067cfafa3a..8cfed1a5dcb 100644 --- a/sound/soc/blackfin/bf5xx-ac97-pcm.c +++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c @@ -297,7 +297,7 @@ static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, } #endif -struct snd_pcm_ops bf5xx_pcm_ac97_ops = { +static struct snd_pcm_ops bf5xx_pcm_ac97_ops = { .open = bf5xx_pcm_open, .ioctl = snd_pcm_lib_ioctl, .hw_params = bf5xx_pcm_hw_params, diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c index 53d290b3ea4..1318c4f627b 100644 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c @@ -184,7 +184,7 @@ static int bf5xx_pcm_mmap(struct snd_pcm_substream *substream, return 0 ; } -struct snd_pcm_ops bf5xx_pcm_i2s_ops = { +static struct snd_pcm_ops bf5xx_pcm_i2s_ops = { .open = bf5xx_pcm_open, .ioctl = snd_pcm_lib_ioctl, .hw_params = bf5xx_pcm_hw_params, diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index d0e0d691ae5..cb5fcd605ac 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -34,6 +34,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8903 if I2C select SND_SOC_WM8971 if I2C select SND_SOC_WM8990 if I2C + select SND_SOC_WM9705 if SND_SOC_AC97_BUS select SND_SOC_WM9712 if SND_SOC_AC97_BUS select SND_SOC_WM9713 if SND_SOC_AC97_BUS help @@ -144,6 +145,9 @@ config SND_SOC_WM8971 config SND_SOC_WM8990 tristate +config SND_SOC_WM9705 + tristate + config SND_SOC_WM9712 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index c4ddc9aa2bb..3664cdc300b 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -23,6 +23,7 @@ snd-soc-wm8900-objs := wm8900.o snd-soc-wm8903-objs := wm8903.o snd-soc-wm8971-objs := wm8971.o snd-soc-wm8990-objs := wm8990.o +snd-soc-wm9705-objs := wm9705.o snd-soc-wm9712-objs := wm9712.o snd-soc-wm9713-objs := wm9713.o @@ -51,5 +52,7 @@ obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o obj-$(CONFIG_SND_SOC_WM8971) += snd-soc-wm8971.o obj-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o +obj-$(CONFIG_SND_SOC_WM8991) += snd-soc-wm8991.o +obj-$(CONFIG_SND_SOC_WM9705) += snd-soc-wm9705.o obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index fb53e6511af..89d41277616 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -123,7 +123,6 @@ bus_err: snd_soc_free_pcms(socdev); err: - kfree(socdev->codec->reg_cache); kfree(socdev->codec); socdev->codec = NULL; return ret; @@ -138,7 +137,6 @@ static int ac97_soc_remove(struct platform_device *pdev) return 0; snd_soc_free_pcms(socdev); - kfree(socdev->codec->reg_cache); kfree(socdev->codec); return 0; diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 73fdbb4d4a3..faf358758e1 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -93,20 +93,6 @@ SOC_ENUM("Capture Source", ad1980_cap_src), SOC_SINGLE("Mic Boost Switch", AC97_MIC, 6, 1, 0), }; -/* add non dapm controls */ -static int ad1980_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(ad1980_snd_ac97_controls); i++) { - err = snd_ctl_add(codec->card, snd_soc_cnew( - &ad1980_snd_ac97_controls[i], codec, NULL)); - if (err < 0) - return err; - } - return 0; -} - static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg) { @@ -123,7 +109,7 @@ static unsigned int ac97_read(struct snd_soc_codec *codec, default: reg = reg >> 1; - if (reg >= (ARRAY_SIZE(ad1980_reg))) + if (reg >= ARRAY_SIZE(ad1980_reg)) return -EINVAL; return cache[reg]; @@ -137,7 +123,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, soc_ac97_ops.write(codec->ac97, reg, val); reg = reg >> 1; - if (reg < (ARRAY_SIZE(ad1980_reg))) + if (reg < ARRAY_SIZE(ad1980_reg)) cache[reg] = val; return 0; @@ -269,7 +255,8 @@ static int ad1980_soc_probe(struct platform_device *pdev) ext_status = ac97_read(codec, AC97_EXTENDED_STATUS); ac97_write(codec, AC97_EXTENDED_STATUS, ext_status&~0x3800); - ad1980_add_controls(codec); + snd_soc_add_controls(codec, ad1980_snd_ac97_controls, + ARRAY_SIZE(ad1980_snd_ac97_controls)); ret = snd_soc_init_card(socdev); if (ret < 0) { printk(KERN_ERR "ad1980: failed to register card\n"); diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 81300d8d42c..f17c363cb1d 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -155,21 +155,6 @@ static const struct snd_kcontrol_new ak4535_snd_controls[] = { SOC_SINGLE("Mic Sidetone Volume", AK4535_VOL, 4, 7, 0), }; -/* add non dapm controls */ -static int ak4535_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(ak4535_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&ak4535_snd_controls[i], codec, NULL)); - if (err < 0) - return err; - } - - return 0; -} - /* Mono 1 Mixer */ static const struct snd_kcontrol_new ak4535_mono1_mixer_controls[] = { SOC_DAPM_SINGLE("Mic Sidetone Switch", AK4535_SIG1, 4, 1, 0), @@ -510,7 +495,8 @@ static int ak4535_init(struct snd_soc_device *socdev) /* power on device */ ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - ak4535_add_controls(codec); + snd_soc_add_controls(codec, ak4535_snd_controls, + ARRAY_SIZE(ak4535_snd_controls)); ak4535_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index cac37361676..ec7fe3b7b0c 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -151,21 +151,6 @@ SOC_ENUM("Capture Source", ssm2602_enum[0]), SOC_ENUM("Playback De-emphasis", ssm2602_enum[1]), }; -/* add non dapm controls */ -static int ssm2602_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(ssm2602_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&ssm2602_snd_controls[i], codec, NULL)); - if (err < 0) - return err; - } - - return 0; -} - /* Output Mixer */ static const struct snd_kcontrol_new ssm2602_output_mixer_controls[] = { SOC_DAPM_SINGLE("Line Bypass Switch", SSM2602_APANA, 3, 1, 0), @@ -622,7 +607,8 @@ static int ssm2602_init(struct snd_soc_device *socdev) APANA_ENABLE_MIC_BOOST); ssm2602_write(codec, SSM2602_PWR, 0); - ssm2602_add_controls(codec); + snd_soc_add_controls(codec, ssm2602_snd_controls, + ARRAY_SIZE(ssm2602_snd_controls)); ssm2602_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index cfdea007c4c..a0e47c1dcd6 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -183,24 +183,6 @@ static const struct snd_kcontrol_new tlv320aic23_snd_controls[] = { SOC_ENUM("Playback De-emphasis", tlv320aic23_deemph), }; -/* add non dapm controls */ -static int tlv320aic23_add_controls(struct snd_soc_codec *codec) -{ - - int err, i; - - for (i = 0; i < ARRAY_SIZE(tlv320aic23_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&tlv320aic23_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - - return 0; - -} - /* PGA Mixer controls for Line and Mic switch */ static const struct snd_kcontrol_new tlv320aic23_output_mixer_controls[] = { SOC_DAPM_SINGLE("Line Bypass Switch", TLV320AIC23_ANLG, 3, 1, 0), @@ -718,7 +700,8 @@ static int tlv320aic23_init(struct snd_soc_device *socdev) tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x1); - tlv320aic23_add_controls(codec); + snd_soc_add_controls(codec, tlv320aic23_snd_controls, + ARRAY_SIZE(tlv320aic23_snd_controls)); tlv320aic23_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index b47a749c5ea..36ab0198ca3 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -311,22 +311,6 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]), }; -/* add non dapm controls */ -static int aic3x_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(aic3x_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&aic3x_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - - return 0; -} - /* Left DAC Mux */ static const struct snd_kcontrol_new aic3x_left_dac_mux_controls = SOC_DAPM_ENUM("Route", aic3x_enum[LDAC_ENUM]); @@ -1224,7 +1208,8 @@ static int aic3x_init(struct snd_soc_device *socdev) aic3x_write(codec, AIC3X_GPIO1_REG, (setup->gpio_func[0] & 0xf) << 4); aic3x_write(codec, AIC3X_GPIO2_REG, (setup->gpio_func[1] & 0xf) << 4); - aic3x_add_controls(codec); + snd_soc_add_controls(codec, aic3x_snd_controls, + ARRAY_SIZE(aic3x_snd_controls)); aic3x_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index ea370a4f86d..f530c1e6d9e 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -125,6 +125,9 @@ static inline unsigned int twl4030_read_reg_cache(struct snd_soc_codec *codec, { u8 *cache = codec->reg_cache; + if (reg >= TWL4030_CACHEREGNUM) + return -EIO; + return cache[reg]; } @@ -670,22 +673,6 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = { 0, 3, 5, 0, input_gain_tlv), }; -/* add non dapm controls */ -static int twl4030_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(twl4030_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&twl4030_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - - return 0; -} - static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { /* Left channel inputs */ SND_SOC_DAPM_INPUT("MAINMIC"), @@ -1233,7 +1220,8 @@ static int twl4030_init(struct snd_soc_device *socdev) /* power on device */ twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - twl4030_add_controls(codec); + snd_soc_add_controls(codec, twl4030_snd_controls, + ARRAY_SIZE(twl4030_snd_controls)); twl4030_add_widgets(codec); ret = snd_soc_init_card(socdev); diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index a2c5064a774..277825d155a 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -431,39 +431,6 @@ SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]), SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0), }; -static int uda134x_add_controls(struct snd_soc_codec *codec) -{ - int err, i, n; - const struct snd_kcontrol_new *ctrls; - struct uda134x_platform_data *pd = codec->control_data; - - switch (pd->model) { - case UDA134X_UDA1340: - case UDA134X_UDA1344: - n = ARRAY_SIZE(uda1340_snd_controls); - ctrls = uda1340_snd_controls; - break; - case UDA134X_UDA1341: - n = ARRAY_SIZE(uda1341_snd_controls); - ctrls = uda1341_snd_controls; - break; - default: - printk(KERN_ERR "%s unkown codec type: %d", - __func__, pd->model); - return -EINVAL; - } - - for (i = 0; i < n; i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&ctrls[i], - codec, NULL)); - if (err < 0) - return err; - } - - return 0; -} - struct snd_soc_dai uda134x_dai = { .name = "UDA134X", /* playback capabilities */ @@ -572,7 +539,22 @@ static int uda134x_soc_probe(struct platform_device *pdev) goto pcm_err; } - ret = uda134x_add_controls(codec); + switch (pd->model) { + case UDA134X_UDA1340: + case UDA134X_UDA1344: + ret = snd_soc_add_controls(codec, uda1340_snd_controls, + ARRAY_SIZE(uda1340_snd_controls)); + break; + case UDA134X_UDA1341: + ret = snd_soc_add_controls(codec, uda1341_snd_controls, + ARRAY_SIZE(uda1341_snd_controls)); + break; + default: + printk(KERN_ERR "%s unkown codec type: %d", + __func__, pd->model); + return -EINVAL; + } + if (ret < 0) { printk(KERN_ERR "UDA134X: failed to register controls\n"); goto pcm_err; diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index e6bf0844fbf..a957b4365b9 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -271,21 +271,6 @@ static const struct snd_kcontrol_new uda1380_snd_controls[] = { SOC_SINGLE("AGC Switch", UDA1380_AGC, 0, 1, 0), }; -/* add non dapm controls */ -static int uda1380_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(uda1380_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&uda1380_snd_controls[i], codec, NULL)); - if (err < 0) - return err; - } - - return 0; -} - /* Input mux */ static const struct snd_kcontrol_new uda1380_input_mux_control = SOC_DAPM_ENUM("Route", uda1380_input_sel_enum); @@ -675,7 +660,8 @@ static int uda1380_init(struct snd_soc_device *socdev, int dac_clk) } /* uda1380 init */ - uda1380_add_controls(codec); + snd_soc_add_controls(codec, uda1380_snd_controls, + ARRAY_SIZE(uda1380_snd_controls)); uda1380_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index e3989d406f5..2e0db29b499 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -51,10 +51,17 @@ struct wm8350_output { u16 mute; }; +struct wm8350_jack_data { + struct snd_soc_jack *jack; + int report; +}; + struct wm8350_data { struct snd_soc_codec codec; struct wm8350_output out1; struct wm8350_output out2; + struct wm8350_jack_data hpl; + struct wm8350_jack_data hpr; struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)]; }; @@ -775,21 +782,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Beep", NULL, "IN3R PGA"}, }; -static int wm8350_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm8350_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8350_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - - return 0; -} - static int wm8350_add_widgets(struct snd_soc_codec *codec) { int ret; @@ -1328,6 +1320,95 @@ static int wm8350_resume(struct platform_device *pdev) return 0; } +static void wm8350_hp_jack_handler(struct wm8350 *wm8350, int irq, void *data) +{ + struct wm8350_data *priv = data; + u16 reg; + int report; + int mask; + struct wm8350_jack_data *jack = NULL; + + switch (irq) { + case WM8350_IRQ_CODEC_JCK_DET_L: + jack = &priv->hpl; + mask = WM8350_JACK_L_LVL; + break; + + case WM8350_IRQ_CODEC_JCK_DET_R: + jack = &priv->hpr; + mask = WM8350_JACK_R_LVL; + break; + + default: + BUG(); + } + + if (!jack->jack) { + dev_warn(wm8350->dev, "Jack interrupt called with no jack\n"); + return; + } + + /* Debounce */ + msleep(200); + + reg = wm8350_reg_read(wm8350, WM8350_JACK_PIN_STATUS); + if (reg & mask) + report = jack->report; + else + report = 0; + + snd_soc_jack_report(jack->jack, report, jack->report); +} + +/** + * wm8350_hp_jack_detect - Enable headphone jack detection. + * + * @codec: WM8350 codec + * @which: left or right jack detect signal + * @jack: jack to report detection events on + * @report: value to report + * + * Enables the headphone jack detection of the WM8350. + */ +int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which, + struct snd_soc_jack *jack, int report) +{ + struct wm8350_data *priv = codec->private_data; + struct wm8350 *wm8350 = codec->control_data; + int irq; + int ena; + + switch (which) { + case WM8350_JDL: + priv->hpl.jack = jack; + priv->hpl.report = report; + irq = WM8350_IRQ_CODEC_JCK_DET_L; + ena = WM8350_JDL_ENA; + break; + + case WM8350_JDR: + priv->hpr.jack = jack; + priv->hpr.report = report; + irq = WM8350_IRQ_CODEC_JCK_DET_R; + ena = WM8350_JDR_ENA; + break; + + default: + return -EINVAL; + } + + wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_TOCLK_ENA); + wm8350_set_bits(wm8350, WM8350_JACK_DETECT, ena); + + /* Sync status */ + wm8350_hp_jack_handler(wm8350, irq, priv); + + wm8350_unmask_irq(wm8350, irq); + + return 0; +} +EXPORT_SYMBOL_GPL(wm8350_hp_jack_detect); + static struct snd_soc_codec *wm8350_codec; static int wm8350_probe(struct platform_device *pdev) @@ -1381,13 +1462,21 @@ static int wm8350_probe(struct platform_device *pdev) wm8350_set_bits(wm8350, WM8350_ROUT2_VOLUME, WM8350_OUT2_VU | WM8350_OUT2R_MUTE); + wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L); + wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R); + wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L, + wm8350_hp_jack_handler, priv); + wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R, + wm8350_hp_jack_handler, priv); + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) { dev_err(&pdev->dev, "failed to create pcms\n"); return ret; } - wm8350_add_controls(codec); + snd_soc_add_controls(codec, wm8350_snd_controls, + ARRAY_SIZE(wm8350_snd_controls)); wm8350_add_widgets(codec); wm8350_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -1411,8 +1500,21 @@ static int wm8350_remove(struct platform_device *pdev) struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->codec; struct wm8350 *wm8350 = codec->control_data; + struct wm8350_data *priv = codec->private_data; int ret; + wm8350_clear_bits(wm8350, WM8350_JACK_DETECT, + WM8350_JDL_ENA | WM8350_JDR_ENA); + wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_TOCLK_ENA); + + wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L); + wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R); + wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L); + wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R); + + priv->hpl.jack = NULL; + priv->hpr.jack = NULL; + /* cancel any work waiting to be queued. */ ret = cancel_delayed_work(&codec->delayed_work); diff --git a/sound/soc/codecs/wm8350.h b/sound/soc/codecs/wm8350.h index cc2887aa6c3..d11bd9288cf 100644 --- a/sound/soc/codecs/wm8350.h +++ b/sound/soc/codecs/wm8350.h @@ -17,4 +17,12 @@ extern struct snd_soc_dai wm8350_dai; extern struct snd_soc_codec_device soc_codec_dev_wm8350; +enum wm8350_jack { + WM8350_JDL = 1, + WM8350_JDR = 2, +}; + +int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which, + struct snd_soc_jack *jack, int report); + #endif diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 40f8238df71..abe7cce8771 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -171,22 +171,6 @@ SOC_SINGLE("Capture Boost(+20dB)", WM8510_ADCBOOST, 8, 1, 0), SOC_SINGLE("Mono Playback Switch", WM8510_MONOMIX, 6, 1, 1), }; -/* add non dapm controls */ -static int wm8510_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm8510_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8510_snd_controls[i], codec, - NULL)); - if (err < 0) - return err; - } - - return 0; -} - /* Speaker Output Mixer */ static const struct snd_kcontrol_new wm8510_speaker_mixer_controls[] = { SOC_DAPM_SINGLE("Line Bypass Switch", WM8510_SPKMIX, 1, 1, 0), @@ -656,7 +640,8 @@ static int wm8510_init(struct snd_soc_device *socdev) /* power on device */ codec->bias_level = SND_SOC_BIAS_OFF; wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - wm8510_add_controls(codec); + snd_soc_add_controls(codec, wm8510_snd_controls, + ARRAY_SIZE(wm8510_snd_controls)); wm8510_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index d004e584529..3faf0e70ce1 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -200,7 +200,7 @@ static inline unsigned int wm8580_read_reg_cache(struct snd_soc_codec *codec, unsigned int reg) { u16 *cache = codec->reg_cache; - BUG_ON(reg > ARRAY_SIZE(wm8580_reg)); + BUG_ON(reg >= ARRAY_SIZE(wm8580_reg)); return cache[reg]; } @@ -223,7 +223,7 @@ static int wm8580_write(struct snd_soc_codec *codec, unsigned int reg, { u8 data[2]; - BUG_ON(reg > ARRAY_SIZE(wm8580_reg)); + BUG_ON(reg >= ARRAY_SIZE(wm8580_reg)); /* Registers are 9 bits wide */ value &= 0x1ff; @@ -330,20 +330,6 @@ SOC_DOUBLE("ADC Mute Switch", WM8580_ADC_CONTROL1, 0, 1, 1, 0), SOC_SINGLE("ADC High-Pass Filter Switch", WM8580_ADC_CONTROL1, 4, 1, 0), }; -/* Add non-DAPM controls */ -static int wm8580_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm8580_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8580_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - return 0; -} static const struct snd_soc_dapm_widget wm8580_dapm_widgets[] = { SND_SOC_DAPM_DAC("DAC1", "Playback", WM8580_PWRDN1, 2, 1), SND_SOC_DAPM_DAC("DAC2", "Playback", WM8580_PWRDN1, 3, 1), @@ -866,7 +852,8 @@ static int wm8580_init(struct snd_soc_device *socdev) goto pcm_err; } - wm8580_add_controls(codec); + snd_soc_add_controls(codec, wm8580_snd_controls, + ARRAY_SIZE(wm8580_snd_controls)); wm8580_add_widgets(codec); ret = snd_soc_init_card(socdev); diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index 80b11983e13..f90dc52e975 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -47,7 +47,7 @@ static inline unsigned int wm8728_read_reg_cache(struct snd_soc_codec *codec, unsigned int reg) { u16 *cache = codec->reg_cache; - BUG_ON(reg > ARRAY_SIZE(wm8728_reg_defaults)); + BUG_ON(reg >= ARRAY_SIZE(wm8728_reg_defaults)); return cache[reg]; } @@ -55,7 +55,7 @@ static inline void wm8728_write_reg_cache(struct snd_soc_codec *codec, u16 reg, unsigned int value) { u16 *cache = codec->reg_cache; - BUG_ON(reg > ARRAY_SIZE(wm8728_reg_defaults)); + BUG_ON(reg >= ARRAY_SIZE(wm8728_reg_defaults)); cache[reg] = value; } @@ -92,21 +92,6 @@ SOC_DOUBLE_R_TLV("Digital Playback Volume", WM8728_DACLVOL, WM8728_DACRVOL, SOC_SINGLE("Deemphasis", WM8728_DACCTL, 1, 1, 0), }; -static int wm8728_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm8728_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8728_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - - return 0; -} - /* * DAPM controls. */ @@ -330,7 +315,8 @@ static int wm8728_init(struct snd_soc_device *socdev) /* power on device */ wm8728_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - wm8728_add_controls(codec); + snd_soc_add_controls(codec, wm8728_snd_controls, + ARRAY_SIZE(wm8728_snd_controls)); wm8728_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index c444b9f2701..96d6e1aeaf4 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -129,22 +129,6 @@ SOC_SINGLE("Store DC Offset Switch", WM8731_APDIGI, 4, 1, 0), SOC_ENUM("Playback De-emphasis", wm8731_enum[1]), }; -/* add non dapm controls */ -static int wm8731_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm8731_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8731_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - - return 0; -} - /* Output Mixer */ static const struct snd_kcontrol_new wm8731_output_mixer_controls[] = { SOC_DAPM_SINGLE("Line Bypass Switch", WM8731_APANA, 3, 1, 0), @@ -543,7 +527,8 @@ static int wm8731_init(struct snd_soc_device *socdev) reg = wm8731_read_reg_cache(codec, WM8731_RINVOL); wm8731_write(codec, WM8731_RINVOL, reg & ~0x0100); - wm8731_add_controls(codec); + snd_soc_add_controls(codec, wm8731_snd_controls, + ARRAY_SIZE(wm8731_snd_controls)); wm8731_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 5997fa68e0d..1578569793a 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -231,21 +231,6 @@ SOC_SINGLE("Mono Playback Volume", WM8750_MOUTV, 0, 127, 0), }; -/* add non dapm controls */ -static int wm8750_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm8750_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8750_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - return 0; -} - /* * DAPM Controls */ @@ -816,7 +801,8 @@ static int wm8750_init(struct snd_soc_device *socdev) reg = wm8750_read_reg_cache(codec, WM8750_RINVOL); wm8750_write(codec, WM8750_RINVOL, reg | 0x0100); - wm8750_add_controls(codec); + snd_soc_add_controls(codec, wm8750_snd_controls, + ARRAY_SIZE(wm8750_snd_controls)); wm8750_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 6c21b50c937..5a1c1fca120 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -97,7 +97,7 @@ static inline unsigned int wm8753_read_reg_cache(struct snd_soc_codec *codec, unsigned int reg) { u16 *cache = codec->reg_cache; - if (reg < 1 || reg > (ARRAY_SIZE(wm8753_reg) + 1)) + if (reg < 1 || reg >= (ARRAY_SIZE(wm8753_reg) + 1)) return -1; return cache[reg - 1]; } @@ -109,7 +109,7 @@ static inline void wm8753_write_reg_cache(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { u16 *cache = codec->reg_cache; - if (reg < 1 || reg > 0x3f) + if (reg < 1 || reg >= (ARRAY_SIZE(wm8753_reg) + 1)) return; cache[reg - 1] = value; } @@ -339,21 +339,6 @@ SOC_ENUM("ADC Data Select", wm8753_enum[27]), SOC_ENUM("ROUT2 Phase", wm8753_enum[28]), }; -/* add non dapm controls */ -static int wm8753_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm8753_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8753_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - return 0; -} - /* * _DAPM_ Controls */ @@ -1603,7 +1588,8 @@ static int wm8753_init(struct snd_soc_device *socdev) reg = wm8753_read_reg_cache(codec, WM8753_RINVOL); wm8753_write(codec, WM8753_RINVOL, reg | 0x0100); - wm8753_add_controls(codec); + snd_soc_add_controls(codec, wm8753_snd_controls, + ARRAY_SIZE(wm8753_snd_controls)); wm8753_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 6767de10ded..1e08d4f065f 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -517,22 +517,6 @@ SOC_SINGLE("LINEOUT2 LP -12dB", WM8900_REG_LOUTMIXCTL1, }; -/* add non dapm controls */ -static int wm8900_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm8900_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8900_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - - return 0; -} - static const struct snd_kcontrol_new wm8900_dapm_loutput2_control = SOC_DAPM_SINGLE("LINEOUT2L Switch", WM8900_REG_POWER3, 6, 1, 0); @@ -1439,7 +1423,8 @@ static int wm8900_probe(struct platform_device *pdev) goto pcm_err; } - wm8900_add_controls(codec); + snd_soc_add_controls(codec, wm8900_snd_controls, + ARRAY_SIZE(wm8900_snd_controls)); wm8900_add_widgets(codec); ret = snd_soc_init_card(socdev); diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index bde74546db4..6ff34b957dc 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -744,21 +744,6 @@ SOC_DOUBLE_R_TLV("Speaker Volume", 0, 63, 0, out_tlv), }; -static int wm8903_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm8903_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8903_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - - return 0; -} - static const struct snd_kcontrol_new linput_mode_mux = SOC_DAPM_ENUM("Left Input Mode Mux", linput_mode_enum); @@ -1737,7 +1722,8 @@ static int wm8903_probe(struct platform_device *pdev) goto err; } - wm8903_add_controls(socdev->codec); + snd_soc_add_controls(socdev->codec, wm8903_snd_controls, + ARRAY_SIZE(wm8903_snd_controls)); wm8903_add_widgets(socdev->codec); ret = snd_soc_init_card(socdev); diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 88ead7f8dd9..c8bd9b06f33 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -195,21 +195,6 @@ static const struct snd_kcontrol_new wm8971_snd_controls[] = { SOC_DOUBLE_R("Mic Boost", WM8971_LADCIN, WM8971_RADCIN, 4, 3, 0), }; -/* add non-DAPM controls */ -static int wm8971_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm8971_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8971_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - return 0; -} - /* * DAPM Controls */ @@ -745,7 +730,8 @@ static int wm8971_init(struct snd_soc_device *socdev) reg = wm8971_read_reg_cache(codec, WM8971_RINVOL); wm8971_write(codec, WM8971_RINVOL, reg | 0x0100); - wm8971_add_controls(codec); + snd_soc_add_controls(codec, wm8971_snd_controls, + ARRAY_SIZE(wm8971_snd_controls)); wm8971_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 5b5afc14447..f93c0955ed9 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -116,7 +116,7 @@ static inline unsigned int wm8990_read_reg_cache(struct snd_soc_codec *codec, unsigned int reg) { u16 *cache = codec->reg_cache; - BUG_ON(reg > (ARRAY_SIZE(wm8990_reg)) - 1); + BUG_ON(reg >= ARRAY_SIZE(wm8990_reg)); return cache[reg]; } @@ -129,7 +129,7 @@ static inline void wm8990_write_reg_cache(struct snd_soc_codec *codec, u16 *cache = codec->reg_cache; /* Reset register and reserved registers are uncached */ - if (reg == 0 || reg > ARRAY_SIZE(wm8990_reg) - 1) + if (reg == 0 || reg >= ARRAY_SIZE(wm8990_reg)) return; cache[reg] = value; @@ -417,21 +417,6 @@ SOC_SINGLE("RIN34 Mute Switch", WM8990_RIGHT_LINE_INPUT_3_4_VOLUME, }; -/* add non dapm controls */ -static int wm8990_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm8990_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8990_snd_controls[i], codec, - NULL)); - if (err < 0) - return err; - } - return 0; -} - /* * _DAPM_ Controls */ @@ -1460,7 +1445,8 @@ static int wm8990_init(struct snd_soc_device *socdev) wm8990_write(codec, WM8990_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8)); wm8990_write(codec, WM8990_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8)); - wm8990_add_controls(codec); + snd_soc_add_controls(codec, wm8990_snd_controls, + ARRAY_SIZE(wm8990_snd_controls)); wm8990_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c new file mode 100644 index 00000000000..5e1937ac0b5 --- /dev/null +++ b/sound/soc/codecs/wm9705.c @@ -0,0 +1,410 @@ +/* + * wm9705.c -- ALSA Soc WM9705 codec support + * + * Copyright 2008 Ian Molton <spyro@f2s.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; Version 2 of the License only. + * + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/ac97_codec.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +/* + * WM9705 register cache + */ +static const u16 wm9705_reg[] = { + 0x6150, 0x8000, 0x8000, 0x8000, /* 0x0 */ + 0x0000, 0x8000, 0x8008, 0x8008, /* 0x8 */ + 0x8808, 0x8808, 0x8808, 0x8808, /* 0x10 */ + 0x8808, 0x0000, 0x8000, 0x0000, /* 0x18 */ + 0x0000, 0x0000, 0x0000, 0x000f, /* 0x20 */ + 0x0605, 0x0000, 0xbb80, 0x0000, /* 0x28 */ + 0x0000, 0xbb80, 0x0000, 0x0000, /* 0x30 */ + 0x0000, 0x2000, 0x0000, 0x0000, /* 0x38 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 0x40 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 0x48 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 0x50 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 0x58 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 0x60 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 0x68 */ + 0x0000, 0x0808, 0x0000, 0x0006, /* 0x70 */ + 0x0000, 0x0000, 0x574d, 0x4c05, /* 0x78 */ +}; + +static const struct snd_kcontrol_new wm9705_snd_ac97_controls[] = { + SOC_DOUBLE("Master Playback Volume", AC97_MASTER, 8, 0, 31, 1), + SOC_SINGLE("Master Playback Switch", AC97_MASTER, 15, 1, 1), + SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1), + SOC_SINGLE("Headphone Playback Switch", AC97_HEADPHONE, 15, 1, 1), + SOC_DOUBLE("PCM Playback Volume", AC97_PCM, 8, 0, 31, 1), + SOC_SINGLE("PCM Playback Switch", AC97_PCM, 15, 1, 1), + SOC_SINGLE("Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 1), + SOC_SINGLE("Mono Playback Switch", AC97_MASTER_MONO, 15, 1, 1), + SOC_SINGLE("PCBeep Playback Volume", AC97_PC_BEEP, 1, 15, 1), + SOC_SINGLE("Phone Playback Volume", AC97_PHONE, 0, 31, 1), + SOC_DOUBLE("Line Playback Volume", AC97_LINE, 8, 0, 31, 1), + SOC_DOUBLE("CD Playback Volume", AC97_CD, 8, 0, 31, 1), + SOC_SINGLE("Mic Playback Volume", AC97_MIC, 0, 31, 1), + SOC_SINGLE("Mic 20dB Boost Switch", AC97_MIC, 6, 1, 0), + SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 15, 0), + SOC_SINGLE("Capture Switch", AC97_REC_GAIN, 15, 1, 1), +}; + +static const char *wm9705_mic[] = {"Mic 1", "Mic 2"}; +static const char *wm9705_rec_sel[] = {"Mic", "CD", "NC", "NC", + "Line", "Stereo Mix", "Mono Mix", "Phone"}; + +static const struct soc_enum wm9705_enum_mic = + SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, wm9705_mic); +static const struct soc_enum wm9705_enum_rec_l = + SOC_ENUM_SINGLE(AC97_REC_SEL, 8, 8, wm9705_rec_sel); +static const struct soc_enum wm9705_enum_rec_r = + SOC_ENUM_SINGLE(AC97_REC_SEL, 0, 8, wm9705_rec_sel); + +/* Headphone Mixer */ +static const struct snd_kcontrol_new wm9705_hp_mixer_controls[] = { + SOC_DAPM_SINGLE("PCBeep Playback Switch", AC97_PC_BEEP, 15, 1, 1), + SOC_DAPM_SINGLE("CD Playback Switch", AC97_CD, 15, 1, 1), + SOC_DAPM_SINGLE("Mic Playback Switch", AC97_MIC, 15, 1, 1), + SOC_DAPM_SINGLE("Phone Playback Switch", AC97_PHONE, 15, 1, 1), + SOC_DAPM_SINGLE("Line Playback Switch", AC97_LINE, 15, 1, 1), +}; + +/* Mic source */ +static const struct snd_kcontrol_new wm9705_mic_src_controls = + SOC_DAPM_ENUM("Route", wm9705_enum_mic); + +/* Capture source */ +static const struct snd_kcontrol_new wm9705_capture_selectl_controls = + SOC_DAPM_ENUM("Route", wm9705_enum_rec_l); +static const struct snd_kcontrol_new wm9705_capture_selectr_controls = + SOC_DAPM_ENUM("Route", wm9705_enum_rec_r); + +/* DAPM widgets */ +static const struct snd_soc_dapm_widget wm9705_dapm_widgets[] = { + SND_SOC_DAPM_MUX("Mic Source", SND_SOC_NOPM, 0, 0, + &wm9705_mic_src_controls), + SND_SOC_DAPM_MUX("Left Capture Source", SND_SOC_NOPM, 0, 0, + &wm9705_capture_selectl_controls), + SND_SOC_DAPM_MUX("Right Capture Source", SND_SOC_NOPM, 0, 0, + &wm9705_capture_selectr_controls), + SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_MIXER_NAMED_CTL("HP Mixer", SND_SOC_NOPM, 0, 0, + &wm9705_hp_mixer_controls[0], + ARRAY_SIZE(wm9705_hp_mixer_controls)), + SND_SOC_DAPM_MIXER("Mono Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_PGA("Headphone PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Speaker PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Line PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Line out PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mono PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Phone PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mic PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PCBEEP PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("CD PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("ADC PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("HPOUTL"), + SND_SOC_DAPM_OUTPUT("HPOUTR"), + SND_SOC_DAPM_OUTPUT("LOUT"), + SND_SOC_DAPM_OUTPUT("ROUT"), + SND_SOC_DAPM_OUTPUT("MONOOUT"), + SND_SOC_DAPM_INPUT("PHONE"), + SND_SOC_DAPM_INPUT("LINEINL"), + SND_SOC_DAPM_INPUT("LINEINR"), + SND_SOC_DAPM_INPUT("CDINL"), + SND_SOC_DAPM_INPUT("CDINR"), + SND_SOC_DAPM_INPUT("PCBEEP"), + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), +}; + +/* Audio map + * WM9705 has no switches to disable the route from the inputs to the HP mixer + * so in order to prevent active inputs from forcing the audio outputs to be + * constantly enabled, we use the mutes on those inputs to simulate such + * controls. + */ +static const struct snd_soc_dapm_route audio_map[] = { + /* HP mixer */ + {"HP Mixer", "PCBeep Playback Switch", "PCBEEP PGA"}, + {"HP Mixer", "CD Playback Switch", "CD PGA"}, + {"HP Mixer", "Mic Playback Switch", "Mic PGA"}, + {"HP Mixer", "Phone Playback Switch", "Phone PGA"}, + {"HP Mixer", "Line Playback Switch", "Line PGA"}, + {"HP Mixer", NULL, "Left DAC"}, + {"HP Mixer", NULL, "Right DAC"}, + + /* mono mixer */ + {"Mono Mixer", NULL, "HP Mixer"}, + + /* outputs */ + {"Headphone PGA", NULL, "HP Mixer"}, + {"HPOUTL", NULL, "Headphone PGA"}, + {"HPOUTR", NULL, "Headphone PGA"}, + {"Line out PGA", NULL, "HP Mixer"}, + {"LOUT", NULL, "Line out PGA"}, + {"ROUT", NULL, "Line out PGA"}, + {"Mono PGA", NULL, "Mono Mixer"}, + {"MONOOUT", NULL, "Mono PGA"}, + + /* inputs */ + {"CD PGA", NULL, "CDINL"}, + {"CD PGA", NULL, "CDINR"}, + {"Line PGA", NULL, "LINEINL"}, + {"Line PGA", NULL, "LINEINR"}, + {"Phone PGA", NULL, "PHONE"}, + {"Mic Source", "Mic 1", "MIC1"}, + {"Mic Source", "Mic 2", "MIC2"}, + {"Mic PGA", NULL, "Mic Source"}, + {"PCBEEP PGA", NULL, "PCBEEP"}, + + /* Left capture selector */ + {"Left Capture Source", "Mic", "Mic Source"}, + {"Left Capture Source", "CD", "CDINL"}, + {"Left Capture Source", "Line", "LINEINL"}, + {"Left Capture Source", "Stereo Mix", "HP Mixer"}, + {"Left Capture Source", "Mono Mix", "HP Mixer"}, + {"Left Capture Source", "Phone", "PHONE"}, + + /* Right capture source */ + {"Right Capture Source", "Mic", "Mic Source"}, + {"Right Capture Source", "CD", "CDINR"}, + {"Right Capture Source", "Line", "LINEINR"}, + {"Right Capture Source", "Stereo Mix", "HP Mixer"}, + {"Right Capture Source", "Mono Mix", "HP Mixer"}, + {"Right Capture Source", "Phone", "PHONE"}, + + {"ADC PGA", NULL, "Left Capture Source"}, + {"ADC PGA", NULL, "Right Capture Source"}, + + /* ADC's */ + {"Left ADC", NULL, "ADC PGA"}, + {"Right ADC", NULL, "ADC PGA"}, +}; + +static int wm9705_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, wm9705_dapm_widgets, + ARRAY_SIZE(wm9705_dapm_widgets)); + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_new_widgets(codec); + + return 0; +} + +/* We use a register cache to enhance read performance. */ +static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg) +{ + u16 *cache = codec->reg_cache; + + switch (reg) { + case AC97_RESET: + case AC97_VENDOR_ID1: + case AC97_VENDOR_ID2: + return soc_ac97_ops.read(codec->ac97, reg); + default: + reg = reg >> 1; + + if (reg >= (ARRAY_SIZE(wm9705_reg))) + return -EIO; + + return cache[reg]; + } +} + +static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int val) +{ + u16 *cache = codec->reg_cache; + + soc_ac97_ops.write(codec->ac97, reg, val); + reg = reg >> 1; + if (reg < (ARRAY_SIZE(wm9705_reg))) + cache[reg] = val; + + return 0; +} + +static int ac97_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + int reg; + u16 vra; + + vra = ac97_read(codec, AC97_EXTENDED_STATUS); + ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + reg = AC97_PCM_FRONT_DAC_RATE; + else + reg = AC97_PCM_LR_ADC_RATE; + + return ac97_write(codec, reg, runtime->rate); +} + +#define WM9705_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | \ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000) + +struct snd_soc_dai wm9705_dai[] = { + { + .name = "AC97 HiFi", + .ac97_control = 1, + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM9705_AC97_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM9705_AC97_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = { + .prepare = ac97_prepare, + }, + }, + { + .name = "AC97 Aux", + .playback = { + .stream_name = "Aux Playback", + .channels_min = 1, + .channels_max = 1, + .rates = WM9705_AC97_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + } +}; +EXPORT_SYMBOL_GPL(wm9705_dai); + +static int wm9705_reset(struct snd_soc_codec *codec) +{ + if (soc_ac97_ops.reset) { + soc_ac97_ops.reset(codec->ac97); + if (ac97_read(codec, 0) == wm9705_reg[0]) + return 0; /* Success */ + } + + return -EIO; +} + +static int wm9705_soc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + printk(KERN_INFO "WM9705 SoC Audio Codec\n"); + + socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (socdev->codec == NULL) + return -ENOMEM; + codec = socdev->codec; + mutex_init(&codec->mutex); + + codec->reg_cache = kmemdup(wm9705_reg, sizeof(wm9705_reg), GFP_KERNEL); + if (codec->reg_cache == NULL) { + ret = -ENOMEM; + goto cache_err; + } + codec->reg_cache_size = sizeof(wm9705_reg); + codec->reg_cache_step = 2; + + codec->name = "WM9705"; + codec->owner = THIS_MODULE; + codec->dai = wm9705_dai; + codec->num_dai = ARRAY_SIZE(wm9705_dai); + codec->write = ac97_write; + codec->read = ac97_read; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); + if (ret < 0) { + printk(KERN_ERR "wm9705: failed to register AC97 codec\n"); + goto codec_err; + } + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) + goto pcm_err; + + ret = wm9705_reset(codec); + if (ret) + goto reset_err; + + snd_soc_add_controls(codec, wm9705_snd_ac97_controls, + ARRAY_SIZE(wm9705_snd_ac97_controls)); + wm9705_add_widgets(codec); + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + printk(KERN_ERR "wm9705: failed to register card\n"); + goto pcm_err; + } + + return 0; + +reset_err: + snd_soc_free_pcms(socdev); +pcm_err: + snd_soc_free_ac97_codec(codec); +codec_err: + kfree(codec->reg_cache); +cache_err: + kfree(socdev->codec); + socdev->codec = NULL; + return ret; +} + +static int wm9705_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec == NULL) + return 0; + + snd_soc_dapm_free(socdev); + snd_soc_free_pcms(socdev); + snd_soc_free_ac97_codec(codec); + kfree(codec->reg_cache); + kfree(codec); + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm9705 = { + .probe = wm9705_soc_probe, + .remove = wm9705_soc_remove, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm9705); + +MODULE_DESCRIPTION("ASoC WM9705 driver"); +MODULE_AUTHOR("Ian Molton"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/wm9705.h b/sound/soc/codecs/wm9705.h new file mode 100644 index 00000000000..d380f110f9e --- /dev/null +++ b/sound/soc/codecs/wm9705.h @@ -0,0 +1,14 @@ +/* + * wm9705.h -- WM9705 Soc Audio driver + */ + +#ifndef _WM9705_H +#define _WM9705_H + +#define WM9705_DAI_AC97_HIFI 0 +#define WM9705_DAI_AC97_AUX 1 + +extern struct snd_soc_dai wm9705_dai[2]; +extern struct snd_soc_codec_device soc_codec_dev_wm9705; + +#endif diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index af83d629078..4dc90d67530 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -154,21 +154,6 @@ SOC_SINGLE("Mic 2 Volume", AC97_MIC, 0, 31, 1), SOC_SINGLE("Mic 20dB Boost Switch", AC97_MIC, 7, 1, 0), }; -/* add non dapm controls */ -static int wm9712_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm9712_snd_ac97_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm9712_snd_ac97_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - return 0; -} - /* We have to create a fake left and right HP mixers because * the codec only has a single control that is shared by both channels. * This makes it impossible to determine the audio path. @@ -467,7 +452,7 @@ static unsigned int ac97_read(struct snd_soc_codec *codec, else { reg = reg >> 1; - if (reg > (ARRAY_SIZE(wm9712_reg))) + if (reg >= (ARRAY_SIZE(wm9712_reg))) return -EIO; return cache[reg]; @@ -481,7 +466,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, soc_ac97_ops.write(codec->ac97, reg, val); reg = reg >> 1; - if (reg <= (ARRAY_SIZE(wm9712_reg))) + if (reg < (ARRAY_SIZE(wm9712_reg))) cache[reg] = val; return 0; @@ -698,7 +683,8 @@ static int wm9712_soc_probe(struct platform_device *pdev) ac97_write(codec, AC97_VIDEO, ac97_read(codec, AC97_VIDEO) | 0x3000); wm9712_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - wm9712_add_controls(codec); + snd_soc_add_controls(codec, wm9712_snd_ac97_controls, + ARRAY_SIZE(wm9712_snd_ac97_controls)); wm9712_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index f3ca8aaf013..0e60e16973d 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -32,7 +32,6 @@ struct wm9713_priv { u32 pll_in; /* PLL input frequency */ - u32 pll_out; /* PLL output frequency */ }; static unsigned int ac97_read(struct snd_soc_codec *codec, @@ -190,21 +189,6 @@ SOC_SINGLE("3D Lower Cut-off Switch", AC97_REC_GAIN_MIC, 4, 1, 0), SOC_SINGLE("3D Depth", AC97_REC_GAIN_MIC, 0, 15, 1), }; -/* add non dapm controls */ -static int wm9713_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm9713_snd_ac97_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm9713_snd_ac97_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - return 0; -} - /* We have to create a fake left and right HP mixers because * the codec only has a single control that is shared by both channels. * This makes it impossible to determine the audio path using the current @@ -636,7 +620,7 @@ static unsigned int ac97_read(struct snd_soc_codec *codec, else { reg = reg >> 1; - if (reg > (ARRAY_SIZE(wm9713_reg))) + if (reg >= (ARRAY_SIZE(wm9713_reg))) return -EIO; return cache[reg]; @@ -650,7 +634,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, if (reg < 0x7c) soc_ac97_ops.write(codec->ac97, reg, val); reg = reg >> 1; - if (reg <= (ARRAY_SIZE(wm9713_reg))) + if (reg < (ARRAY_SIZE(wm9713_reg))) cache[reg] = val; return 0; @@ -738,13 +722,13 @@ static int wm9713_set_pll(struct snd_soc_codec *codec, struct _pll_div pll_div; /* turn PLL off ? */ - if (freq_in == 0 || freq_out == 0) { + if (freq_in == 0) { /* disable PLL power and select ext source */ reg = ac97_read(codec, AC97_HANDSET_RATE); ac97_write(codec, AC97_HANDSET_RATE, reg | 0x0080); reg = ac97_read(codec, AC97_EXTENDED_MID); ac97_write(codec, AC97_EXTENDED_MID, reg | 0x0200); - wm9713->pll_out = 0; + wm9713->pll_in = 0; return 0; } @@ -788,7 +772,6 @@ static int wm9713_set_pll(struct snd_soc_codec *codec, ac97_write(codec, AC97_EXTENDED_MID, reg & 0xfdff); reg = ac97_read(codec, AC97_HANDSET_RATE); ac97_write(codec, AC97_HANDSET_RATE, reg & 0xff7f); - wm9713->pll_out = freq_out; wm9713->pll_in = freq_in; /* wait 10ms AC97 link frames for the link to stabilise */ @@ -1164,8 +1147,8 @@ static int wm9713_soc_resume(struct platform_device *pdev) wm9713_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* do we need to re-start the PLL ? */ - if (wm9713->pll_out) - wm9713_set_pll(codec, 0, wm9713->pll_in, wm9713->pll_out); + if (wm9713->pll_in) + wm9713_set_pll(codec, 0, wm9713->pll_in, 0); /* only synchronise the codec if warm reset failed */ if (ret == 0) { @@ -1245,7 +1228,8 @@ static int wm9713_soc_probe(struct platform_device *pdev) reg = ac97_read(codec, AC97_CD) & 0x7fff; ac97_write(codec, AC97_CD, reg); - wm9713_add_controls(codec); + snd_soc_add_controls(codec, wm9713_snd_ac97_controls, + ARRAY_SIZE(wm9713_snd_ac97_controls)); wm9713_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 366049d8578..7af3b5b3a53 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -286,7 +286,7 @@ static int davinci_pcm_mmap(struct snd_pcm_substream *substream, runtime->dma_bytes); } -struct snd_pcm_ops davinci_pcm_ops = { +static struct snd_pcm_ops davinci_pcm_ops = { .open = davinci_pcm_open, .close = davinci_pcm_close, .ioctl = snd_pcm_lib_ioctl, diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c index 4935d1bcbd8..50baef1fe5b 100644 --- a/sound/soc/davinci/davinci-sffsdr.c +++ b/sound/soc/davinci/davinci-sffsdr.c @@ -25,7 +25,9 @@ #include <asm/dma.h> #include <asm/mach-types.h> +#ifdef CONFIG_SFFSDR_FPGA #include <asm/plat-sffsdr/sffsdr-fpga.h> +#endif #include <mach/mcbsp.h> #include <mach/edma.h> @@ -43,6 +45,17 @@ static int sffsdr_hw_params(struct snd_pcm_substream *substream, int fs; int ret = 0; + /* Fsref can be 32000, 44100 or 48000. */ + fs = params_rate(params); + +#ifndef CONFIG_SFFSDR_FPGA + /* Without the FPGA module, the Fs is fixed at 44100 Hz */ + if (fs != 44100) { + pr_debug("warning: only 44.1 kHz is supported without SFFSDR FPGA module\n"); + return -EINVAL; + } +#endif + /* Set cpu DAI configuration: * CLKX and CLKR are the inputs for the Sample Rate Generator. * FSX and FSR are outputs, driven by the sample Rate Generator. */ @@ -53,12 +66,13 @@ static int sffsdr_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - /* Fsref can be 32000, 44100 or 48000. */ - fs = params_rate(params); - pr_debug("sffsdr_hw_params: rate = %d Hz\n", fs); +#ifndef CONFIG_SFFSDR_FPGA + return 0; +#else return sffsdr_fpga_set_codec_fs(fs); +#endif } static struct snd_soc_ops sffsdr_ops = { diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 95c12b26fe3..c7c78c39cfe 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -1,17 +1,17 @@ config SND_SOC_OF_SIMPLE tristate +# ASoC platform support for the Freescale MPC8610 SOC. This compiles drivers +# for the SSI and the Elo DMA controller. You will still need to select +# a platform driver and a codec driver. config SND_SOC_MPC8610 - bool "ALSA SoC support for the MPC8610 SOC" - depends on MPC8610_HPCD - default y if MPC8610 - help - Say Y if you want to add support for codecs attached to the SSI - device on an MPC8610. + tristate + depends on MPC8610 config SND_SOC_MPC8610_HPCD - bool "ALSA SoC support for the Freescale MPC8610 HPCD board" - depends on SND_SOC_MPC8610 + tristate "ALSA SoC support for the Freescale MPC8610 HPCD board" + depends on MPC8610_HPCD + select SND_SOC_MPC8610 select SND_SOC_CS4270 select SND_SOC_CS4270_VD33_ERRATA default y if MPC8610_HPCD diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 035da4afec3..f85134c8638 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -2,10 +2,13 @@ obj-$(CONFIG_SND_SOC_OF_SIMPLE) += soc-of-simple.o # MPC8610 HPCD Machine Support -obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += mpc8610_hpcd.o +snd-soc-mpc8610-hpcd-objs := mpc8610_hpcd.o +obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += snd-soc-mpc8610-hpcd.o # MPC8610 Platform Support -obj-$(CONFIG_SND_SOC_MPC8610) += fsl_ssi.o fsl_dma.o +snd-soc-fsl-ssi-objs := fsl_ssi.o +snd-soc-fsl-dma-objs := fsl_dma.o +obj-$(CONFIG_SND_SOC_MPC8610) += snd-soc-fsl-ssi.o snd-soc-fsl-dma.o obj-$(CONFIG_SND_SOC_MPC5200_I2S) += mpc5200_psc_i2s.o diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index b0362dfd5b7..607a38c7ae4 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -264,7 +264,7 @@ static int omap_pcm_mmap(struct snd_pcm_substream *substream, runtime->dma_bytes); } -struct snd_pcm_ops omap_pcm_ops = { +static struct snd_pcm_ops omap_pcm_ops = { .open = omap_pcm_open, .close = omap_pcm_close, .ioctl = snd_pcm_lib_ioctl, diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index f82e1069947..958ac3fe15d 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -61,6 +61,24 @@ config SND_PXA2XX_SOC_TOSA Say Y if you want to add support for SoC audio on Sharp Zaurus SL-C6000x models (Tosa). +config SND_PXA2XX_SOC_E740 + tristate "SoC AC97 Audio support for e740" + depends on SND_PXA2XX_SOC && MACH_E740 + select SND_SOC_WM9705 + select SND_PXA2XX_SOC_AC97 + help + Say Y if you want to add support for SoC audio on the + toshiba e740 PDA + +config SND_PXA2XX_SOC_E750 + tristate "SoC AC97 Audio support for e750" + depends on SND_PXA2XX_SOC && MACH_E750 + select SND_SOC_WM9705 + select SND_PXA2XX_SOC_AC97 + help + Say Y if you want to add support for SoC audio on the + toshiba e750 PDA + config SND_PXA2XX_SOC_E800 tristate "SoC AC97 Audio support for e800" depends on SND_PXA2XX_SOC && MACH_E800 diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 08a9f279772..97a51a8c936 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -13,6 +13,8 @@ obj-$(CONFIG_SND_PXA_SOC_SSP) += snd-soc-pxa-ssp.o snd-soc-corgi-objs := corgi.o snd-soc-poodle-objs := poodle.o snd-soc-tosa-objs := tosa.o +snd-soc-e740-objs := e740_wm9705.o +snd-soc-e750-objs := e750_wm9705.o snd-soc-e800-objs := e800_wm9712.o snd-soc-spitz-objs := spitz.o snd-soc-em-x270-objs := em-x270.o @@ -22,6 +24,8 @@ snd-soc-zylonite-objs := zylonite.o obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o obj-$(CONFIG_SND_PXA2XX_SOC_TOSA) += snd-soc-tosa.o +obj-$(CONFIG_SND_PXA2XX_SOC_E740) += snd-soc-e740.o +obj-$(CONFIG_SND_PXA2XX_SOC_E750) += snd-soc-e750.o obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c new file mode 100644 index 00000000000..ac361765173 --- /dev/null +++ b/sound/soc/pxa/e740_wm9705.c @@ -0,0 +1,213 @@ +/* + * e740-wm9705.c -- SoC audio for e740 + * + * Copyright 2007 (c) Ian Molton <spyro@f2s.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; version 2 ONLY. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/gpio.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <mach/pxa-regs.h> +#include <mach/hardware.h> +#include <mach/audio.h> +#include <mach/eseries-gpio.h> + +#include <asm/mach-types.h> + +#include "../codecs/wm9705.h" +#include "pxa2xx-pcm.h" +#include "pxa2xx-ac97.h" + + +#define E740_AUDIO_OUT 1 +#define E740_AUDIO_IN 2 + +static int e740_audio_power; + +static void e740_sync_audio_power(int status) +{ + gpio_set_value(GPIO_E740_WM9705_nAVDD2, !status); + gpio_set_value(GPIO_E740_AMP_ON, (status & E740_AUDIO_OUT) ? 1 : 0); + gpio_set_value(GPIO_E740_MIC_ON, (status & E740_AUDIO_IN) ? 1 : 0); +} + +static int e740_mic_amp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + if (event & SND_SOC_DAPM_PRE_PMU) + e740_audio_power |= E740_AUDIO_IN; + else if (event & SND_SOC_DAPM_POST_PMD) + e740_audio_power &= ~E740_AUDIO_IN; + + e740_sync_audio_power(e740_audio_power); + + return 0; +} + +static int e740_output_amp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + if (event & SND_SOC_DAPM_PRE_PMU) + e740_audio_power |= E740_AUDIO_OUT; + else if (event & SND_SOC_DAPM_POST_PMD) + e740_audio_power &= ~E740_AUDIO_OUT; + + e740_sync_audio_power(e740_audio_power); + + return 0; +} + +static const struct snd_soc_dapm_widget e740_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_MIC("Mic (Internal)", NULL), + SND_SOC_DAPM_PGA_E("Output Amp", SND_SOC_NOPM, 0, 0, NULL, 0, + e740_output_amp_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA_E("Mic Amp", SND_SOC_NOPM, 0, 0, NULL, 0, + e740_mic_amp_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + {"Output Amp", NULL, "LOUT"}, + {"Output Amp", NULL, "ROUT"}, + {"Output Amp", NULL, "MONOOUT"}, + + {"Speaker", NULL, "Output Amp"}, + {"Headphone Jack", NULL, "Output Amp"}, + + {"MIC1", NULL, "Mic Amp"}, + {"Mic Amp", NULL, "Mic (Internal)"}, +}; + +static int e740_ac97_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_nc_pin(codec, "HPOUTL"); + snd_soc_dapm_nc_pin(codec, "HPOUTR"); + snd_soc_dapm_nc_pin(codec, "PHONE"); + snd_soc_dapm_nc_pin(codec, "LINEINL"); + snd_soc_dapm_nc_pin(codec, "LINEINR"); + snd_soc_dapm_nc_pin(codec, "CDINL"); + snd_soc_dapm_nc_pin(codec, "CDINR"); + snd_soc_dapm_nc_pin(codec, "PCBEEP"); + snd_soc_dapm_nc_pin(codec, "MIC2"); + + snd_soc_dapm_new_controls(codec, e740_dapm_widgets, + ARRAY_SIZE(e740_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link e740_dai[] = { + { + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI], + .codec_dai = &wm9705_dai[WM9705_DAI_AC97_HIFI], + .init = e740_ac97_init, + }, + { + .name = "AC97 Aux", + .stream_name = "AC97 Aux", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX], + .codec_dai = &wm9705_dai[WM9705_DAI_AC97_AUX], + }, +}; + +static struct snd_soc_card e740 = { + .name = "Toshiba e740", + .platform = &pxa2xx_soc_platform, + .dai_link = e740_dai, + .num_links = ARRAY_SIZE(e740_dai), +}; + +static struct snd_soc_device e740_snd_devdata = { + .card = &e740, + .codec_dev = &soc_codec_dev_wm9705, +}; + +static struct platform_device *e740_snd_device; + +static int __init e740_init(void) +{ + int ret; + + if (!machine_is_e740()) + return -ENODEV; + + ret = gpio_request(GPIO_E740_MIC_ON, "Mic amp"); + if (ret) + return ret; + + ret = gpio_request(GPIO_E740_AMP_ON, "Output amp"); + if (ret) + goto free_mic_amp_gpio; + + ret = gpio_request(GPIO_E740_WM9705_nAVDD2, "Audio power"); + if (ret) + goto free_op_amp_gpio; + + /* Disable audio */ + ret = gpio_direction_output(GPIO_E740_MIC_ON, 0); + if (ret) + goto free_apwr_gpio; + ret = gpio_direction_output(GPIO_E740_AMP_ON, 0); + if (ret) + goto free_apwr_gpio; + ret = gpio_direction_output(GPIO_E740_WM9705_nAVDD2, 1); + if (ret) + goto free_apwr_gpio; + + e740_snd_device = platform_device_alloc("soc-audio", -1); + if (!e740_snd_device) { + ret = -ENOMEM; + goto free_apwr_gpio; + } + + platform_set_drvdata(e740_snd_device, &e740_snd_devdata); + e740_snd_devdata.dev = &e740_snd_device->dev; + ret = platform_device_add(e740_snd_device); + + if (!ret) + return 0; + +/* Fail gracefully */ + platform_device_put(e740_snd_device); +free_apwr_gpio: + gpio_free(GPIO_E740_WM9705_nAVDD2); +free_op_amp_gpio: + gpio_free(GPIO_E740_AMP_ON); +free_mic_amp_gpio: + gpio_free(GPIO_E740_MIC_ON); + + return ret; +} + +static void __exit e740_exit(void) +{ + platform_device_unregister(e740_snd_device); +} + +module_init(e740_init); +module_exit(e740_exit); + +/* Module information */ +MODULE_AUTHOR("Ian Molton <spyro@f2s.com>"); +MODULE_DESCRIPTION("ALSA SoC driver for e740"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c new file mode 100644 index 00000000000..20fbdcfa9f7 --- /dev/null +++ b/sound/soc/pxa/e750_wm9705.c @@ -0,0 +1,189 @@ +/* + * e750-wm9705.c -- SoC audio for e750 + * + * Copyright 2007 (c) Ian Molton <spyro@f2s.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; version 2 ONLY. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/gpio.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <mach/pxa-regs.h> +#include <mach/hardware.h> +#include <mach/audio.h> +#include <mach/eseries-gpio.h> + +#include <asm/mach-types.h> + +#include "../codecs/wm9705.h" +#include "pxa2xx-pcm.h" +#include "pxa2xx-ac97.h" + +static int e750_spk_amp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + if (event & SND_SOC_DAPM_PRE_PMU) + gpio_set_value(GPIO_E750_SPK_AMP_OFF, 0); + else if (event & SND_SOC_DAPM_POST_PMD) + gpio_set_value(GPIO_E750_SPK_AMP_OFF, 1); + + return 0; +} + +static int e750_hp_amp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + if (event & SND_SOC_DAPM_PRE_PMU) + gpio_set_value(GPIO_E750_HP_AMP_OFF, 0); + else if (event & SND_SOC_DAPM_POST_PMD) + gpio_set_value(GPIO_E750_HP_AMP_OFF, 1); + + return 0; +} + +static const struct snd_soc_dapm_widget e750_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_MIC("Mic (Internal)", NULL), + SND_SOC_DAPM_PGA_E("Headphone Amp", SND_SOC_NOPM, 0, 0, NULL, 0, + e750_hp_amp_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA_E("Speaker Amp", SND_SOC_NOPM, 0, 0, NULL, 0, + e750_spk_amp_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + {"Headphone Amp", NULL, "HPOUTL"}, + {"Headphone Amp", NULL, "HPOUTR"}, + {"Headphone Jack", NULL, "Headphone Amp"}, + + {"Speaker Amp", NULL, "MONOOUT"}, + {"Speaker", NULL, "Speaker Amp"}, + + {"MIC1", NULL, "Mic (Internal)"}, +}; + +static int e750_ac97_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_nc_pin(codec, "LOUT"); + snd_soc_dapm_nc_pin(codec, "ROUT"); + snd_soc_dapm_nc_pin(codec, "PHONE"); + snd_soc_dapm_nc_pin(codec, "LINEINL"); + snd_soc_dapm_nc_pin(codec, "LINEINR"); + snd_soc_dapm_nc_pin(codec, "CDINL"); + snd_soc_dapm_nc_pin(codec, "CDINR"); + snd_soc_dapm_nc_pin(codec, "PCBEEP"); + snd_soc_dapm_nc_pin(codec, "MIC2"); + + snd_soc_dapm_new_controls(codec, e750_dapm_widgets, + ARRAY_SIZE(e750_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link e750_dai[] = { + { + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI], + .codec_dai = &wm9705_dai[WM9705_DAI_AC97_HIFI], + .init = e750_ac97_init, + /* use ops to check startup state */ + }, + { + .name = "AC97 Aux", + .stream_name = "AC97 Aux", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX], + .codec_dai = &wm9705_dai[WM9705_DAI_AC97_AUX], + }, +}; + +static struct snd_soc_card e750 = { + .name = "Toshiba e750", + .platform = &pxa2xx_soc_platform, + .dai_link = e750_dai, + .num_links = ARRAY_SIZE(e750_dai), +}; + +static struct snd_soc_device e750_snd_devdata = { + .card = &e750, + .codec_dev = &soc_codec_dev_wm9705, +}; + +static struct platform_device *e750_snd_device; + +static int __init e750_init(void) +{ + int ret; + + if (!machine_is_e750()) + return -ENODEV; + + ret = gpio_request(GPIO_E750_HP_AMP_OFF, "Headphone amp"); + if (ret) + return ret; + + ret = gpio_request(GPIO_E750_SPK_AMP_OFF, "Speaker amp"); + if (ret) + goto free_hp_amp_gpio; + + ret = gpio_direction_output(GPIO_E750_HP_AMP_OFF, 1); + if (ret) + goto free_spk_amp_gpio; + + ret = gpio_direction_output(GPIO_E750_SPK_AMP_OFF, 1); + if (ret) + goto free_spk_amp_gpio; + + e750_snd_device = platform_device_alloc("soc-audio", -1); + if (!e750_snd_device) { + ret = -ENOMEM; + goto free_spk_amp_gpio; + } + + platform_set_drvdata(e750_snd_device, &e750_snd_devdata); + e750_snd_devdata.dev = &e750_snd_device->dev; + ret = platform_device_add(e750_snd_device); + + if (!ret) + return 0; + +/* Fail gracefully */ + platform_device_put(e750_snd_device); +free_spk_amp_gpio: + gpio_free(GPIO_E750_SPK_AMP_OFF); +free_hp_amp_gpio: + gpio_free(GPIO_E750_HP_AMP_OFF); + + return ret; +} + +static void __exit e750_exit(void) +{ + platform_device_unregister(e750_snd_device); + gpio_free(GPIO_E750_SPK_AMP_OFF); + gpio_free(GPIO_E750_HP_AMP_OFF); +} + +module_init(e750_init); +module_exit(e750_exit); + +/* Module information */ +MODULE_AUTHOR("Ian Molton <spyro@f2s.com>"); +MODULE_DESCRIPTION("ALSA SoC driver for e750"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c index 2e3386dfa0f..78a1770b986 100644 --- a/sound/soc/pxa/e800_wm9712.c +++ b/sound/soc/pxa/e800_wm9712.c @@ -1,8 +1,6 @@ /* * e800-wm9712.c -- SoC audio for e800 * - * Based on tosa.c - * * Copyright 2007 (c) Ian Molton <spyro@f2s.com> * * This program is free software; you can redistribute it and/or modify it @@ -13,31 +11,96 @@ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/device.h> +#include <linux/gpio.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> #include <sound/soc-dapm.h> -#include <asm/mach-types.h> #include <mach/pxa-regs.h> #include <mach/hardware.h> #include <mach/audio.h> +#include <mach/eseries-gpio.h> + +#include <asm/mach-types.h> #include "../codecs/wm9712.h" #include "pxa2xx-pcm.h" #include "pxa2xx-ac97.h" -static struct snd_soc_card e800; +static int e800_spk_amp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + if (event & SND_SOC_DAPM_PRE_PMU) + gpio_set_value(GPIO_E800_SPK_AMP_ON, 1); + else if (event & SND_SOC_DAPM_POST_PMD) + gpio_set_value(GPIO_E800_SPK_AMP_ON, 0); -static struct snd_soc_dai_link e800_dai[] = { + return 0; +} + +static int e800_hp_amp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) { - .name = "AC97 Aux", - .stream_name = "AC97 Aux", - .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX], - .codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX], -}, + if (event & SND_SOC_DAPM_PRE_PMU) + gpio_set_value(GPIO_E800_HP_AMP_OFF, 0); + else if (event & SND_SOC_DAPM_POST_PMD) + gpio_set_value(GPIO_E800_HP_AMP_OFF, 1); + + return 0; +} + +static const struct snd_soc_dapm_widget e800_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Mic (Internal1)", NULL), + SND_SOC_DAPM_MIC("Mic (Internal2)", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_PGA_E("Headphone Amp", SND_SOC_NOPM, 0, 0, NULL, 0, + e800_hp_amp_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA_E("Speaker Amp", SND_SOC_NOPM, 0, 0, NULL, 0, + e800_spk_amp_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + {"Headphone Jack", NULL, "HPOUTL"}, + {"Headphone Jack", NULL, "HPOUTR"}, + {"Headphone Jack", NULL, "Headphone Amp"}, + + {"Speaker Amp", NULL, "MONOOUT"}, + {"Speaker", NULL, "Speaker Amp"}, + + {"MIC1", NULL, "Mic (Internal1)"}, + {"MIC2", NULL, "Mic (Internal2)"}, +}; + +static int e800_ac97_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, e800_dapm_widgets, + ARRAY_SIZE(e800_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link e800_dai[] = { + { + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI], + .codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI], + .init = e800_ac97_init, + }, + { + .name = "AC97 Aux", + .stream_name = "AC97 Aux", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX], + .codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX], + }, }; static struct snd_soc_card e800 = { @@ -61,6 +124,22 @@ static int __init e800_init(void) if (!machine_is_e800()) return -ENODEV; + ret = gpio_request(GPIO_E800_HP_AMP_OFF, "Headphone amp"); + if (ret) + return ret; + + ret = gpio_request(GPIO_E800_SPK_AMP_ON, "Speaker amp"); + if (ret) + goto free_hp_amp_gpio; + + ret = gpio_direction_output(GPIO_E800_HP_AMP_OFF, 1); + if (ret) + goto free_spk_amp_gpio; + + ret = gpio_direction_output(GPIO_E800_SPK_AMP_ON, 1); + if (ret) + goto free_spk_amp_gpio; + e800_snd_device = platform_device_alloc("soc-audio", -1); if (!e800_snd_device) return -ENOMEM; @@ -69,8 +148,15 @@ static int __init e800_init(void) e800_snd_devdata.dev = &e800_snd_device->dev; ret = platform_device_add(e800_snd_device); - if (ret) - platform_device_put(e800_snd_device); + if (!ret) + return 0; + +/* Fail gracefully */ + platform_device_put(e800_snd_device); +free_spk_amp_gpio: + gpio_free(GPIO_E800_SPK_AMP_ON); +free_hp_amp_gpio: + gpio_free(GPIO_E800_HP_AMP_OFF); return ret; } @@ -78,6 +164,8 @@ static int __init e800_init(void) static void __exit e800_exit(void) { platform_device_unregister(e800_snd_device); + gpio_free(GPIO_E800_SPK_AMP_ON); + gpio_free(GPIO_E800_HP_AMP_OFF); } module_init(e800_init); @@ -86,4 +174,4 @@ module_exit(e800_exit); /* Module information */ MODULE_AUTHOR("Ian Molton <spyro@f2s.com>"); MODULE_DESCRIPTION("ALSA SoC driver for e800"); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index f8e9ecd589d..8541b679f6e 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -14,6 +14,7 @@ #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/device.h> +#include <linux/clk.h> #include <linux/i2c.h> #include <sound/core.h> #include <sound/pcm.h> @@ -26,6 +27,17 @@ #include "pxa2xx-ac97.h" #include "pxa-ssp.h" +/* + * There is a physical switch SW15 on the board which changes the MCLK + * for the WM9713 between the standard AC97 master clock and the + * output of the CLK_POUT signal from the PXA. + */ +static int clk_pout; +module_param(clk_pout, int, 0); +MODULE_PARM_DESC(clk_pout, "Use CLK_POUT as WM9713 MCLK (SW15 on board)."); + +static struct clk *pout; + static struct snd_soc_card zylonite; static const struct snd_soc_dapm_widget zylonite_dapm_widgets[] = { @@ -61,10 +73,8 @@ static const struct snd_soc_dapm_route audio_map[] = { static int zylonite_wm9713_init(struct snd_soc_codec *codec) { - /* Currently we only support use of the AC97 clock here. If - * CLK_POUT is selected by SW15 then the clock API will need - * to be used to request and enable it here. - */ + if (clk_pout) + snd_soc_dai_set_pll(&codec->dai[0], 0, clk_get_rate(pout), 0); snd_soc_dapm_new_controls(codec, zylonite_dapm_widgets, ARRAY_SIZE(zylonite_dapm_widgets)); @@ -85,7 +95,6 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - unsigned int pll_out = 0; unsigned int acds = 0; unsigned int wm9713_div = 0; int ret = 0; @@ -93,16 +102,13 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, switch (params_rate(params)) { case 8000: wm9713_div = 12; - pll_out = 2048000; break; case 16000: wm9713_div = 6; - pll_out = 4096000; break; case 48000: default: wm9713_div = 2; - pll_out = 12288000; acds = 1; break; } @@ -123,10 +129,6 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, pll_out); - if (ret < 0) - return ret; - ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_AUDIO_DIV_ACDS, acds); if (ret < 0) return ret; @@ -135,11 +137,12 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - /* Note that if the PLL is in use the WM9713_PCMCLK_PLL_DIV needs - * to be set instead. - */ - ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_DIV, - WM9713_PCMDIV(wm9713_div)); + if (clk_pout) + ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_PLL_DIV, + WM9713_PCMDIV(wm9713_div)); + else + ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_DIV, + WM9713_PCMDIV(wm9713_div)); if (ret < 0) return ret; @@ -173,8 +176,72 @@ static struct snd_soc_dai_link zylonite_dai[] = { }, }; +static int zylonite_probe(struct platform_device *pdev) +{ + int ret; + + if (clk_pout) { + pout = clk_get(NULL, "CLK_POUT"); + if (IS_ERR(pout)) { + dev_err(&pdev->dev, "Unable to obtain CLK_POUT: %ld\n", + PTR_ERR(pout)); + return PTR_ERR(pout); + } + + ret = clk_enable(pout); + if (ret != 0) { + dev_err(&pdev->dev, "Unable to enable CLK_POUT: %d\n", + ret); + clk_put(pout); + return ret; + } + + dev_dbg(&pdev->dev, "MCLK enabled at %luHz\n", + clk_get_rate(pout)); + } + + return 0; +} + +static int zylonite_remove(struct platform_device *pdev) +{ + if (clk_pout) { + clk_disable(pout); + clk_put(pout); + } + + return 0; +} + +static int zylonite_suspend_post(struct platform_device *pdev, + pm_message_t state) +{ + if (clk_pout) + clk_disable(pout); + + return 0; +} + +static int zylonite_resume_pre(struct platform_device *pdev) +{ + int ret = 0; + + if (clk_pout) { + ret = clk_enable(pout); + if (ret != 0) + dev_err(&pdev->dev, "Unable to enable CLK_POUT: %d\n", + ret); + } + + return ret; +} + static struct snd_soc_card zylonite = { .name = "Zylonite", + .probe = &zylonite_probe, + .remove = &zylonite_remove, + .suspend_post = &zylonite_suspend_post, + .resume_pre = &zylonite_resume_pre, .platform = &pxa2xx_soc_platform, .dai_link = zylonite_dai, .num_links = ARRAY_SIZE(zylonite_dai), diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 55fdb4abb17..8313d52a6e8 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1495,6 +1495,37 @@ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template, EXPORT_SYMBOL_GPL(snd_soc_cnew); /** + * snd_soc_add_controls - add an array of controls to a codec. + * Convienience function to add a list of controls. Many codecs were + * duplicating this code. + * + * @codec: codec to add controls to + * @controls: array of controls to add + * @num_controls: number of elements in the array + * + * Return 0 for success, else error. + */ +int snd_soc_add_controls(struct snd_soc_codec *codec, + const struct snd_kcontrol_new *controls, int num_controls) +{ + struct snd_card *card = codec->card; + int err, i; + + for (i = 0; i < num_controls; i++) { + const struct snd_kcontrol_new *control = &controls[i]; + err = snd_ctl_add(card, snd_soc_cnew(control, codec, NULL)); + if (err < 0) { + dev_err(codec->dev, "%s: Failed to add %s\n", + codec->name, control->name); + return err; + } + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_add_controls); + +/** * snd_soc_info_enum_double - enumerated double mixer info callback * @kcontrol: mixer control * @uinfo: control element information diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a2f1da8b464..54b4564b82b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -54,14 +54,15 @@ static int dapm_up_seq[] = { snd_soc_dapm_pre, snd_soc_dapm_micbias, snd_soc_dapm_mic, snd_soc_dapm_mux, snd_soc_dapm_value_mux, snd_soc_dapm_dac, - snd_soc_dapm_mixer, snd_soc_dapm_pga, snd_soc_dapm_adc, snd_soc_dapm_hp, - snd_soc_dapm_spk, snd_soc_dapm_post + snd_soc_dapm_mixer, snd_soc_dapm_mixer_named_ctl, snd_soc_dapm_pga, + snd_soc_dapm_adc, snd_soc_dapm_hp, snd_soc_dapm_spk, snd_soc_dapm_post }; + static int dapm_down_seq[] = { snd_soc_dapm_pre, snd_soc_dapm_adc, snd_soc_dapm_hp, snd_soc_dapm_spk, - snd_soc_dapm_pga, snd_soc_dapm_mixer, snd_soc_dapm_dac, snd_soc_dapm_mic, - snd_soc_dapm_micbias, snd_soc_dapm_mux, snd_soc_dapm_value_mux, - snd_soc_dapm_post + snd_soc_dapm_pga, snd_soc_dapm_mixer_named_ctl, snd_soc_dapm_mixer, + snd_soc_dapm_dac, snd_soc_dapm_mic, snd_soc_dapm_micbias, + snd_soc_dapm_mux, snd_soc_dapm_value_mux, snd_soc_dapm_post }; static int dapm_status = 1; @@ -101,7 +102,8 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, { switch (w->id) { case snd_soc_dapm_switch: - case snd_soc_dapm_mixer: { + case snd_soc_dapm_mixer: + case snd_soc_dapm_mixer_named_ctl: { int val; struct soc_mixer_control *mc = (struct soc_mixer_control *) w->kcontrols[i].private_value; @@ -323,15 +325,33 @@ static int dapm_new_mixer(struct snd_soc_codec *codec, if (path->name != (char*)w->kcontrols[i].name) continue; - /* add dapm control with long name */ - name_len = 2 + strlen(w->name) - + strlen(w->kcontrols[i].name); + /* add dapm control with long name. + * for dapm_mixer this is the concatenation of the + * mixer and kcontrol name. + * for dapm_mixer_named_ctl this is simply the + * kcontrol name. + */ + name_len = strlen(w->kcontrols[i].name) + 1; + if (w->id == snd_soc_dapm_mixer) + name_len += 1 + strlen(w->name); + path->long_name = kmalloc(name_len, GFP_KERNEL); + if (path->long_name == NULL) return -ENOMEM; - snprintf(path->long_name, name_len, "%s %s", - w->name, w->kcontrols[i].name); + switch (w->id) { + case snd_soc_dapm_mixer: + default: + snprintf(path->long_name, name_len, "%s %s", + w->name, w->kcontrols[i].name); + break; + case snd_soc_dapm_mixer_named_ctl: + snprintf(path->long_name, name_len, "%s", + w->kcontrols[i].name); + break; + } + path->long_name[name_len - 1] = '\0'; path->kcontrol = snd_soc_cnew(&w->kcontrols[i], w, @@ -687,6 +707,7 @@ static void dbg_dump_dapm(struct snd_soc_codec* codec, const char *action) case snd_soc_dapm_adc: case snd_soc_dapm_pga: case snd_soc_dapm_mixer: + case snd_soc_dapm_mixer_named_ctl: if (w->name) { in = is_connected_input_ep(w); dapm_clear_walk(w->codec); @@ -760,6 +781,7 @@ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, int found = 0; if (widget->id != snd_soc_dapm_mixer && + widget->id != snd_soc_dapm_mixer_named_ctl && widget->id != snd_soc_dapm_switch) return -ENODEV; @@ -813,6 +835,7 @@ static ssize_t dapm_widget_show(struct device *dev, case snd_soc_dapm_adc: case snd_soc_dapm_pga: case snd_soc_dapm_mixer: + case snd_soc_dapm_mixer_named_ctl: if (w->name) count += sprintf(buf + count, "%s: %s\n", w->name, w->power ? "On":"Off"); @@ -876,7 +899,7 @@ static void dapm_free_widgets(struct snd_soc_codec *codec) } static int snd_soc_dapm_set_pin(struct snd_soc_codec *codec, - char *pin, int status) + const char *pin, int status) { struct snd_soc_dapm_widget *w; @@ -991,6 +1014,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_codec *codec, break; case snd_soc_dapm_switch: case snd_soc_dapm_mixer: + case snd_soc_dapm_mixer_named_ctl: ret = dapm_connect_mixer(codec, wsource, wsink, path, control); if (ret != 0) goto err; @@ -1068,6 +1092,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec) switch(w->id) { case snd_soc_dapm_switch: case snd_soc_dapm_mixer: + case snd_soc_dapm_mixer_named_ctl: dapm_new_mixer(codec, w); break; case snd_soc_dapm_mux: @@ -1549,7 +1574,7 @@ int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev, * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to * do any widget power switching. */ -int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, char *pin) +int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, const char *pin) { return snd_soc_dapm_set_pin(codec, pin, 1); } @@ -1564,7 +1589,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_enable_pin); * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to * do any widget power switching. */ -int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, char *pin) +int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, const char *pin) { return snd_soc_dapm_set_pin(codec, pin, 0); } @@ -1584,7 +1609,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_disable_pin); * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to * do any widget power switching. */ -int snd_soc_dapm_nc_pin(struct snd_soc_codec *codec, char *pin) +int snd_soc_dapm_nc_pin(struct snd_soc_codec *codec, const char *pin) { return snd_soc_dapm_set_pin(codec, pin, 0); } @@ -1599,7 +1624,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_nc_pin); * * Returns 1 for connected otherwise 0. */ -int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, char *pin) +int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, const char *pin) { struct snd_soc_dapm_widget *w; diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c new file mode 100644 index 00000000000..8cc00c3cdf3 --- /dev/null +++ b/sound/soc/soc-jack.c @@ -0,0 +1,138 @@ +/* + * soc-jack.c -- ALSA SoC jack handling + * + * Copyright 2008 Wolfson Microelectronics PLC. + * + * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <sound/jack.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +/** + * snd_soc_jack_new - Create a new jack + * @card: ASoC card + * @id: an identifying string for this jack + * @type: a bitmask of enum snd_jack_type values that can be detected by + * this jack + * @jack: structure to use for the jack + * + * Creates a new jack object. + * + * Returns zero if successful, or a negative error code on failure. + * On success jack will be initialised. + */ +int snd_soc_jack_new(struct snd_soc_card *card, const char *id, int type, + struct snd_soc_jack *jack) +{ + jack->card = card; + INIT_LIST_HEAD(&jack->pins); + + return snd_jack_new(card->socdev->codec->card, id, type, &jack->jack); +} +EXPORT_SYMBOL_GPL(snd_soc_jack_new); + +/** + * snd_soc_jack_report - Report the current status for a jack + * + * @jack: the jack + * @status: a bitmask of enum snd_jack_type values that are currently detected. + * @mask: a bitmask of enum snd_jack_type values that being reported. + * + * If configured using snd_soc_jack_add_pins() then the associated + * DAPM pins will be enabled or disabled as appropriate and DAPM + * synchronised. + * + * Note: This function uses mutexes and should be called from a + * context which can sleep (such as a workqueue). + */ +void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) +{ + struct snd_soc_codec *codec = jack->card->socdev->codec; + struct snd_soc_jack_pin *pin; + int enable; + int oldstatus; + + if (!jack) { + WARN_ON_ONCE(!jack); + return; + } + + mutex_lock(&codec->mutex); + + oldstatus = jack->status; + + jack->status &= ~mask; + jack->status |= status; + + /* The DAPM sync is expensive enough to be worth skipping */ + if (jack->status == oldstatus) + goto out; + + list_for_each_entry(pin, &jack->pins, list) { + enable = pin->mask & status; + + if (pin->invert) + enable = !enable; + + if (enable) + snd_soc_dapm_enable_pin(codec, pin->pin); + else + snd_soc_dapm_disable_pin(codec, pin->pin); + } + + snd_soc_dapm_sync(codec); + + snd_jack_report(jack->jack, status); + +out: + mutex_unlock(&codec->mutex); +} +EXPORT_SYMBOL_GPL(snd_soc_jack_report); + +/** + * snd_soc_jack_add_pins - Associate DAPM pins with an ASoC jack + * + * @jack: ASoC jack + * @count: Number of pins + * @pins: Array of pins + * + * After this function has been called the DAPM pins specified in the + * pins array will have their status updated to reflect the current + * state of the jack whenever the jack status is updated. + */ +int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count, + struct snd_soc_jack_pin *pins) +{ + int i; + + for (i = 0; i < count; i++) { + if (!pins[i].pin) { + printk(KERN_ERR "No name for pin %d\n", i); + return -EINVAL; + } + if (!pins[i].mask) { + printk(KERN_ERR "No mask for pin %d (%s)\n", i, + pins[i].pin); + return -EINVAL; + } + + INIT_LIST_HEAD(&pins[i].list); + list_add(&(pins[i].list), &jack->pins); + } + + /* Update to reflect the last reported status; canned jack + * implementations are likely to set their state before the + * card has an opportunity to associate pins. + */ + snd_soc_jack_report(jack, 0, 0); + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_jack_add_pins); |