From 4f9c16ccfa26691dbb9a5d9e7d5098eb934ccdbe Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 30 Apr 2008 16:20:19 +0200 Subject: [ALSA] soc - tlv320aic3x - revisit clock setup This patch cleans up the clocking setup for aic3x codecs. It drops the dividers table and determines the PLL control values programatically. Under certain conditions, the PLL is disabled entirely which could save some power. Signed-off-by: Daniel Mack Acked-by: Jarkko Nikula Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/codecs/tlv320aic3x.c | 225 +++++++++++++++++------------------------ sound/soc/codecs/tlv320aic3x.h | 4 + 2 files changed, 95 insertions(+), 134 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 09b1661b8a3..738b3b634d7 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -49,7 +49,7 @@ #include "tlv320aic3x.h" #define AUDIO_NAME "aic3x" -#define AIC3X_VERSION "0.1" +#define AIC3X_VERSION "0.2" /* codec private data */ struct aic3x_priv { @@ -648,81 +648,6 @@ static int aic3x_add_widgets(struct snd_soc_codec *codec) return 0; } -struct aic3x_rate_divs { - u32 mclk; - u32 rate; - u32 fsref_reg; - u8 sr_reg:4; - u8 pllj_reg; - u16 plld_reg; -}; - -/* AIC3X codec mclk clock divider coefficients */ -static const struct aic3x_rate_divs aic3x_divs[] = { - /* 8k */ - {12000000, 8000, 48000, 0xa, 16, 3840}, - {19200000, 8000, 48000, 0xa, 10, 2400}, - {22579200, 8000, 48000, 0xa, 8, 7075}, - {33868800, 8000, 48000, 0xa, 5, 8049}, - /* 11.025k */ - {12000000, 11025, 44100, 0x6, 15, 528}, - {19200000, 11025, 44100, 0x6, 9, 4080}, - {22579200, 11025, 44100, 0x6, 8, 0}, - {33868800, 11025, 44100, 0x6, 5, 3333}, - /* 16k */ - {12000000, 16000, 48000, 0x4, 16, 3840}, - {19200000, 16000, 48000, 0x4, 10, 2400}, - {22579200, 16000, 48000, 0x4, 8, 7075}, - {33868800, 16000, 48000, 0x4, 5, 8049}, - /* 22.05k */ - {12000000, 22050, 44100, 0x2, 15, 528}, - {19200000, 22050, 44100, 0x2, 9, 4080}, - {22579200, 22050, 44100, 0x2, 8, 0}, - {33868800, 22050, 44100, 0x2, 5, 3333}, - /* 32k */ - {12000000, 32000, 48000, 0x1, 16, 3840}, - {19200000, 32000, 48000, 0x1, 10, 2400}, - {22579200, 32000, 48000, 0x1, 8, 7075}, - {33868800, 32000, 48000, 0x1, 5, 8049}, - /* 44.1k */ - {12000000, 44100, 44100, 0x0, 15, 528}, - {19200000, 44100, 44100, 0x0, 9, 4080}, - {22579200, 44100, 44100, 0x0, 8, 0}, - {33868800, 44100, 44100, 0x0, 5, 3333}, - /* 48k */ - {12000000, 48000, 48000, 0x0, 16, 3840}, - {19200000, 48000, 48000, 0x0, 10, 2400}, - {22579200, 48000, 48000, 0x0, 8, 7075}, - {33868800, 48000, 48000, 0x0, 5, 8049}, - /* 64k */ - {12000000, 64000, 96000, 0x1, 16, 3840}, - {19200000, 64000, 96000, 0x1, 10, 2400}, - {22579200, 64000, 96000, 0x1, 8, 7075}, - {33868800, 64000, 96000, 0x1, 5, 8049}, - /* 88.2k */ - {12000000, 88200, 88200, 0x0, 15, 528}, - {19200000, 88200, 88200, 0x0, 9, 4080}, - {22579200, 88200, 88200, 0x0, 8, 0}, - {33868800, 88200, 88200, 0x0, 5, 3333}, - /* 96k */ - {12000000, 96000, 96000, 0x0, 16, 3840}, - {19200000, 96000, 96000, 0x0, 10, 2400}, - {22579200, 96000, 96000, 0x0, 8, 7075}, - {33868800, 96000, 96000, 0x0, 5, 8049}, -}; - -static inline int aic3x_get_divs(int mclk, int rate) -{ - int i; - - for (i = 0; i < ARRAY_SIZE(aic3x_divs); i++) { - if (aic3x_divs[i].rate == rate && aic3x_divs[i].mclk == mclk) - return i; - } - - return 0; -} - static int aic3x_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -730,49 +655,107 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->codec; struct aic3x_priv *aic3x = codec->private_data; - int i; - u8 data, pll_p, pll_r, pll_j; - u16 pll_d; - - i = aic3x_get_divs(aic3x->sysclk, params_rate(params)); + int codec_clk = 0, bypass_pll = 0, fsref, last_clk = 0; + u8 data, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1; + u16 pll_d = 1; - /* Route Left DAC to left channel input and - * right DAC to right channel input */ - data = (LDAC2LCH | RDAC2RCH); - switch (aic3x_divs[i].fsref_reg) { - case 44100: - data |= FSREF_44100; + /* select data word length */ + data = + aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLB) & (~(0x3 << 4)); + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: break; - case 48000: - data |= FSREF_48000; + case SNDRV_PCM_FORMAT_S20_3LE: + data |= (0x01 << 4); break; - case 88200: - data |= FSREF_44100 | DUAL_RATE_MODE; + case SNDRV_PCM_FORMAT_S24_LE: + data |= (0x02 << 4); break; - case 96000: - data |= FSREF_48000 | DUAL_RATE_MODE; + case SNDRV_PCM_FORMAT_S32_LE: + data |= (0x03 << 4); break; } + aic3x_write(codec, AIC3X_ASD_INTF_CTRLB, data); + + /* Fsref can be 44100 or 48000 */ + fsref = (params_rate(params) % 11025 == 0) ? 44100 : 48000; + + /* Try to find a value for Q which allows us to bypass the PLL and + * generate CODEC_CLK directly. */ + for (pll_q = 2; pll_q < 18; pll_q++) + if (aic3x->sysclk / (128 * pll_q) == fsref) { + bypass_pll = 1; + break; + } + + if (bypass_pll) { + pll_q &= 0xf; + aic3x_write(codec, AIC3X_PLL_PROGA_REG, pll_q << PLLQ_SHIFT); + aic3x_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_CLKDIV); + } else + aic3x_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_PLLDIV); + + /* Route Left DAC to left channel input and + * right DAC to right channel input */ + data = (LDAC2LCH | RDAC2RCH); + data |= (fsref == 44100) ? FSREF_44100 : FSREF_48000; + if (params_rate(params) >= 64000) + data |= DUAL_RATE_MODE; aic3x_write(codec, AIC3X_CODEC_DATAPATH_REG, data); /* codec sample rate select */ - data = aic3x_divs[i].sr_reg; + data = (fsref * 20) / params_rate(params); + if (params_rate(params) < 64000) + data /= 2; + data /= 5; + data -= 2; data |= (data << 4); aic3x_write(codec, AIC3X_SAMPLE_RATE_SEL_REG, data); - /* Use PLL for generation Fsref by equation: - * Fsref = (MCLK * K * R)/(2048 * P); - * Fix P = 2 and R = 1 and calculate K, if - * K = J.D, i.e. J - an interger portion of K and D is the fractional - * one with 4 digits of precision; - * Example: - * For MCLK = 22.5792 MHz and Fsref = 48kHz: - * Select P = 2, R= 1, K = 8.7074, which results in J = 8, D = 7074 + if (bypass_pll) + return 0; + + /* Use PLL + * find an apropriate setup for j, d, r and p by iterating over + * p and r - j and d are calculated for each fraction. + * Up to 128 values are probed, the closest one wins the game. + * The sysclk is divided by 1000 to prevent integer overflows. */ - pll_p = 2; - pll_r = 1; - pll_j = aic3x_divs[i].pllj_reg; - pll_d = aic3x_divs[i].plld_reg; + codec_clk = (2048 * fsref) / (aic3x->sysclk / 1000); + + for (r = 1; r <= 16; r++) + for (p = 1; p <= 8; p++) { + int clk, tmp = (codec_clk * pll_r * 10) / pll_p; + u8 j = tmp / 10000; + u16 d = tmp % 10000; + + if (j > 63) + continue; + + if (d != 0 && aic3x->sysclk < 10000000) + continue; + + /* This is actually 1000 * ((j + (d/10000)) * r) / p + * The term had to be converted to get rid of the + * division by 10000 */ + clk = ((10000 * j * r) + (d * r)) / (10 * p); + + /* check whether this values get closer than the best + * ones we had before */ + if (abs(codec_clk - clk) < abs(codec_clk - last_clk)) { + pll_j = j; pll_d = d; pll_r = r; pll_p = p; + last_clk = clk; + } + + /* Early exit for exact matches */ + if (clk == codec_clk) + break; + } + + if (last_clk == 0) { + printk(KERN_ERR "%s(): unable to setup PLL\n", __func__); + return -EINVAL; + } data = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG); aic3x_write(codec, AIC3X_PLL_PROGA_REG, data | (pll_p << PLLP_SHIFT)); @@ -782,24 +765,6 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, aic3x_write(codec, AIC3X_PLL_PROGD_REG, (pll_d & 0x3F) << PLLD_LSB_SHIFT); - /* select data word length */ - data = - aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLB) & (~(0x3 << 4)); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: - break; - case SNDRV_PCM_FORMAT_S20_3LE: - data |= (0x01 << 4); - break; - case SNDRV_PCM_FORMAT_S24_LE: - data |= (0x02 << 4); - break; - case SNDRV_PCM_FORMAT_S32_LE: - data |= (0x03 << 4); - break; - } - aic3x_write(codec, AIC3X_ASD_INTF_CTRLB, data); - return 0; } @@ -826,16 +791,8 @@ static int aic3x_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, struct snd_soc_codec *codec = codec_dai->codec; struct aic3x_priv *aic3x = codec->private_data; - switch (freq) { - case 12000000: - case 19200000: - case 22579200: - case 33868800: - aic3x->sysclk = freq; - return 0; - } - - return -EINVAL; + aic3x->sysclk = freq; + return 0; } static int aic3x_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index d0cdeeb629d..d49d001e6e4 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -109,6 +109,7 @@ #define LLOPM_CTRL 86 #define RLOPM_CTRL 93 /* Clock generation control register */ +#define AIC3X_GPIOB_REG 101 #define AIC3X_CLKGEN_CTRL_REG 102 /* Page select register bits */ @@ -128,12 +129,15 @@ /* PLL registers bitfields */ #define PLLP_SHIFT 0 +#define PLLQ_SHIFT 3 #define PLLR_SHIFT 0 #define PLLJ_SHIFT 2 #define PLLD_MSB_SHIFT 0 #define PLLD_LSB_SHIFT 2 /* Clock generation register bits */ +#define CODEC_CLKIN_PLLDIV 0 +#define CODEC_CLKIN_CLKDIV 1 #define PLL_CLKIN_SHIFT 4 #define MCLK_SOURCE 0x0 #define PLL_CLKDIV_SHIFT 0 -- cgit From 54e7e6167d29a4a98207884b2fbd28b0b3fe91f6 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 30 Apr 2008 16:20:52 +0200 Subject: [ALSA] soc - tlv320aic3x - add GPIO support This patch adds support for AIC3x GPIO lines. They can be configured for many possible functions as well as be driven manually. I also introduced i2c read functionality since the GPIO state register has to be read from hardware every time and can not be served from cache. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/codecs/tlv320aic3x.c | 53 ++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic3x.h | 49 +++++++++++++++++++++++++++++++++++++- 2 files changed, 101 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 738b3b634d7..957996e0eba 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -138,6 +138,20 @@ static int aic3x_write(struct snd_soc_codec *codec, unsigned int reg, return -EIO; } +/* + * read from the aic3x register space + */ +static int aic3x_read(struct snd_soc_codec *codec, unsigned int reg, + u8 *value) +{ + *value = reg & 0xff; + if (codec->hw_read(codec->control_data, value, 1) != 1) + return -EIO; + + aic3x_write_reg_cache(codec, reg, *value); + return 0; +} + #define SOC_DAPM_SINGLE_AIC3X(xname, reg, shift, mask, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_volsw, \ @@ -911,6 +925,33 @@ static int aic3x_dapm_event(struct snd_soc_codec *codec, int event) return 0; } +void aic3x_set_gpio(struct snd_soc_codec *codec, int gpio, int state) +{ + u8 reg = gpio ? AIC3X_GPIO2_REG : AIC3X_GPIO1_REG; + u8 bit = gpio ? 3: 0; + u8 val = aic3x_read_reg_cache(codec, reg) & ~(1 << bit); + aic3x_write(codec, reg, val | (!!state << bit)); +} +EXPORT_SYMBOL_GPL(aic3x_set_gpio); + +int aic3x_get_gpio(struct snd_soc_codec *codec, int gpio) +{ + u8 reg = gpio ? AIC3X_GPIO2_REG : AIC3X_GPIO1_REG; + u8 val, bit = gpio ? 2: 1; + + aic3x_read(codec, reg, &val); + return (val >> bit) & 1; +} +EXPORT_SYMBOL_GPL(aic3x_get_gpio); + +int aic3x_headset_detected(struct snd_soc_codec *codec) +{ + u8 val; + aic3x_read(codec, AIC3X_RT_IRQ_FLAGS_REG, &val); + return (val >> 2) & 1; +} +EXPORT_SYMBOL_GPL(aic3x_headset_detected); + #define AIC3X_RATES SNDRV_PCM_RATE_8000_96000 #define AIC3X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) @@ -977,6 +1018,7 @@ static int aic3x_resume(struct platform_device *pdev) static int aic3x_init(struct snd_soc_device *socdev) { struct snd_soc_codec *codec = socdev->codec; + struct aic3x_setup_data *setup = socdev->codec_data; int reg, ret = 0; codec->name = "aic3x"; @@ -1067,6 +1109,10 @@ static int aic3x_init(struct snd_soc_device *socdev) /* off, with power on */ aic3x_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + /* setup GPIO functions */ + 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); aic3x_add_widgets(codec); ret = snd_soc_register_card(socdev); @@ -1174,6 +1220,12 @@ static struct i2c_client client_template = { .name = "AIC3X", .driver = &aic3x_i2c_driver, }; + +static int aic3x_i2c_read(struct i2c_client *client, u8 *value, int len) +{ + value[0] = i2c_smbus_read_byte_data(client, value[0]); + return (len == 1); +} #endif static int aic3x_probe(struct platform_device *pdev) @@ -1208,6 +1260,7 @@ static int aic3x_probe(struct platform_device *pdev) if (setup->i2c_address) { normal_i2c[0] = setup->i2c_address; codec->hw_write = (hw_write_t) i2c_master_send; + codec->hw_read = (hw_read_t) aic3x_i2c_read; ret = i2c_add_driver(&aic3x_i2c_driver); if (ret != 0) printk(KERN_ERR "can't add i2c driver"); diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index d49d001e6e4..c1dd1ac0cea 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -108,8 +108,14 @@ #define DACR1_2_RLOPM_VOL 92 #define LLOPM_CTRL 86 #define RLOPM_CTRL 93 -/* Clock generation control register */ +/* GPIO/IRQ registers */ +#define AIC3X_STICKY_IRQ_FLAGS_REG 96 +#define AIC3X_RT_IRQ_FLAGS_REG 97 +#define AIC3X_GPIO1_REG 98 +#define AIC3X_GPIO2_REG 99 +#define AIC3X_GPIOA_REG 100 #define AIC3X_GPIOB_REG 101 +/* Clock generation control register */ #define AIC3X_CLKGEN_CTRL_REG 102 /* Page select register bits */ @@ -175,8 +181,49 @@ /* Default input volume */ #define DEFAULT_GAIN 0x20 +/* GPIO API */ +enum { + AIC3X_GPIO1_FUNC_DISABLED = 0, + AIC3X_GPIO1_FUNC_AUDIO_WORDCLK_ADC = 1, + AIC3X_GPIO1_FUNC_CLOCK_MUX = 2, + AIC3X_GPIO1_FUNC_CLOCK_MUX_DIV2 = 3, + AIC3X_GPIO1_FUNC_CLOCK_MUX_DIV4 = 4, + AIC3X_GPIO1_FUNC_CLOCK_MUX_DIV8 = 5, + AIC3X_GPIO1_FUNC_SHORT_CIRCUIT_IRQ = 6, + AIC3X_GPIO1_FUNC_AGC_NOISE_IRQ = 7, + AIC3X_GPIO1_FUNC_INPUT = 8, + AIC3X_GPIO1_FUNC_OUTPUT = 9, + AIC3X_GPIO1_FUNC_DIGITAL_MIC_MODCLK = 10, + AIC3X_GPIO1_FUNC_AUDIO_WORDCLK = 11, + AIC3X_GPIO1_FUNC_BUTTON_IRQ = 12, + AIC3X_GPIO1_FUNC_HEADSET_DETECT_IRQ = 13, + AIC3X_GPIO1_FUNC_HEADSET_DETECT_OR_BUTTON_IRQ = 14, + AIC3X_GPIO1_FUNC_ALL_IRQ = 16 +}; + +enum { + AIC3X_GPIO2_FUNC_DISABLED = 0, + AIC3X_GPIO2_FUNC_HEADSET_DETECT_IRQ = 2, + AIC3X_GPIO2_FUNC_INPUT = 3, + AIC3X_GPIO2_FUNC_OUTPUT = 4, + AIC3X_GPIO2_FUNC_DIGITAL_MIC_INPUT = 5, + AIC3X_GPIO2_FUNC_AUDIO_BITCLK = 8, + AIC3X_GPIO2_FUNC_HEADSET_DETECT_OR_BUTTON_IRQ = 9, + AIC3X_GPIO2_FUNC_ALL_IRQ = 10, + AIC3X_GPIO2_FUNC_SHORT_CIRCUIT_OR_AGC_IRQ = 11, + AIC3X_GPIO2_FUNC_HEADSET_OR_BUTTON_PRESS_OR_SHORT_CIRCUIT_IRQ = 12, + AIC3X_GPIO2_FUNC_SHORT_CIRCUIT_IRQ = 13, + AIC3X_GPIO2_FUNC_AGC_NOISE_IRQ = 14, + AIC3X_GPIO2_FUNC_BUTTON_PRESS_IRQ = 15 +}; + +void aic3x_set_gpio(struct snd_soc_codec *codec, int gpio, int state); +int aic3x_get_gpio(struct snd_soc_codec *codec, int gpio); +int aic3x_headset_detected(struct snd_soc_codec *codec); + struct aic3x_setup_data { unsigned short i2c_address; + unsigned int gpio_func[2]; }; extern struct snd_soc_codec_dai aic3x_dai; -- cgit From eeec12bf7b7d80d1c9cae5aae0dff7e2f928c64b Mon Sep 17 00:00:00 2001 From: Graeme Gregory Date: Wed, 30 Apr 2008 19:27:40 +0200 Subject: [ALSA] soc - DAPM - add hook to read state of DAPM widget This adds a hook to read the power state of a DAPM widget, I use this in the gta02 driver to expose certain DAPM widgets in the mixer for ease of audio routing. Signed-off-by: Graeme Gregory Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/soc-dapm.c | 23 +++++++++++++++++++++++ 1 file changed, 23 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index af3326c6350..9fd5ee818e8 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1342,6 +1342,29 @@ int snd_soc_dapm_set_endpoint(struct snd_soc_codec *codec, } EXPORT_SYMBOL_GPL(snd_soc_dapm_set_endpoint); +/** + * snd_soc_dapm_get_endpoint_status - get audio endpoint status + * @codec: audio codec + * @endpoint: audio signal endpoint (or start point) + * + * Get audio endpoint status - connected or disconnected. + * + * Returns status + */ +int snd_soc_dapm_get_endpoint_status(struct snd_soc_codec *codec, + char *endpoint) +{ + struct snd_soc_dapm_widget *w; + + list_for_each_entry(w, &codec->dapm_widgets, list) { + if (!strcmp(w->name, endpoint)) + return w->connected; + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_get_endpoint); + /** * snd_soc_dapm_free - free dapm resources * @socdev: SoC device -- cgit From 650f6b13318bce6a8d59bfa48fe15b5832623cbc Mon Sep 17 00:00:00 2001 From: Stephen Rothwell Date: Thu, 1 May 2008 10:49:18 +0200 Subject: [ALSA] sound: fix export symbol typo Signed-off-by: Stephen Rothwell Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 9fd5ee818e8..c60200ccde6 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1363,7 +1363,7 @@ int snd_soc_dapm_get_endpoint_status(struct snd_soc_codec *codec, return 0; } -EXPORT_SYMBOL_GPL(snd_soc_dapm_get_endpoint); +EXPORT_SYMBOL_GPL(snd_soc_dapm_get_endpoint_status); /** * snd_soc_dapm_free - free dapm resources -- cgit From 1894c59fdb63692f5ba2576875cc558b856935ca Mon Sep 17 00:00:00 2001 From: Tim Niemeyer Date: Mon, 5 May 2008 14:16:12 +0200 Subject: [ALSA] soc - Patch to add debug messages to the neo1973_wm8753 (GTA01) sound driver Signed-off-by: Tim Niemeyer Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/s3c24xx/neo1973_wm8753.c | 46 ++++++++++++++++++++++++++++++++++++++ 1 file changed, 46 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 0e9d1c5f248..e469186a407 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -43,6 +43,14 @@ #include "s3c24xx-pcm.h" #include "s3c24xx-i2s.h" +/* Debugging stuff */ +#define S3C24XX_SOC_NEO1973_WM8753_DEBUG 0 +#if S3C24XX_SOC_NEO1973_WM8753_DEBUG +#define DBG(x...) printk(KERN_DEBUG "s3c24xx-soc-neo1973-wm8753: " x) +#else +#define DBG(x...) +#endif + /* define the scenarios */ #define NEO_AUDIO_OFF 0 #define NEO_GSM_CALL_AUDIO_HANDSET 1 @@ -67,6 +75,8 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream, int ret = 0; unsigned long iis_clkrate; + DBG("Entered %s\n", __func__); + iis_clkrate = s3c24xx_i2s_get_clockrate(); switch (params_rate(params)) { @@ -151,6 +161,8 @@ static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; + DBG("Entered %s\n", __func__); + /* disable the PLL */ return codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL1, 0, 0); } @@ -172,6 +184,8 @@ static int neo1973_voice_hw_params(struct snd_pcm_substream *substream, int ret = 0; unsigned long iis_clkrate; + DBG("Entered %s\n", __func__); + iis_clkrate = s3c24xx_i2s_get_clockrate(); if (params_rate(params) != 8000) @@ -213,6 +227,8 @@ static int neo1973_voice_hw_free(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; + DBG("Entered %s\n", __func__); + /* disable the PLL */ return codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL2, 0, 0); } @@ -233,6 +249,8 @@ static int neo1973_get_scenario(struct snd_kcontrol *kcontrol, static int set_scenario_endpoints(struct snd_soc_codec *codec, int scenario) { + DBG("Entered %s\n", __func__); + switch (neo1973_scenario) { case NEO_AUDIO_OFF: snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); @@ -315,6 +333,8 @@ static int neo1973_set_scenario(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + DBG("Entered %s\n", __func__); + if (neo1973_scenario == ucontrol->value.integer.value[0]) return 0; @@ -327,6 +347,8 @@ static u8 lm4857_regs[4] = {0x00, 0x40, 0x80, 0xC0}; static void lm4857_write_regs(void) { + DBG("Entered %s\n", __func__); + if (i2c_master_send(i2c, lm4857_regs, 4) != 4) printk(KERN_ERR "lm4857: i2c write failed\n"); } @@ -338,6 +360,8 @@ static int lm4857_get_reg(struct snd_kcontrol *kcontrol, int shift = (kcontrol->private_value >> 8) & 0x0F; int mask = (kcontrol->private_value >> 16) & 0xFF; + DBG("Entered %s\n", __func__); + ucontrol->value.integer.value[0] = (lm4857_regs[reg] >> shift) & mask; return 0; } @@ -364,6 +388,8 @@ static int lm4857_get_mode(struct snd_kcontrol *kcontrol, { u8 value = lm4857_regs[LM4857_CTRL] & 0x0F; + DBG("Entered %s\n", __func__); + if (value) value -= 5; @@ -376,6 +402,8 @@ static int lm4857_set_mode(struct snd_kcontrol *kcontrol, { u8 value = ucontrol->value.integer.value[0]; + DBG("Entered %s\n", __func__); + if (value) value += 5; @@ -483,6 +511,8 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec) { int i, err; + DBG("Entered %s\n", __func__); + /* set up NC codec pins */ snd_soc_dapm_set_endpoint(codec, "LOUT2", 0); snd_soc_dapm_set_endpoint(codec, "ROUT2", 0); @@ -583,6 +613,8 @@ static int lm4857_amp_probe(struct i2c_adapter *adap, int addr, int kind) { int ret; + DBG("Entered %s\n", __func__); + client_template.adapter = adap; client_template.addr = addr; @@ -606,6 +638,8 @@ exit_err: static int lm4857_i2c_detach(struct i2c_client *client) { + DBG("Entered %s\n", __func__); + i2c_detach_client(client); kfree(client); return 0; @@ -613,6 +647,8 @@ static int lm4857_i2c_detach(struct i2c_client *client) static int lm4857_i2c_attach(struct i2c_adapter *adap) { + DBG("Entered %s\n", __func__); + return i2c_probe(adap, &addr_data, lm4857_amp_probe); } @@ -620,6 +656,8 @@ static u8 lm4857_state; static int lm4857_suspend(struct i2c_client *dev, pm_message_t state) { + DBG("Entered %s\n", __func__); + dev_dbg(&dev->dev, "lm4857_suspend\n"); lm4857_state = lm4857_regs[LM4857_CTRL] & 0xf; if (lm4857_state) { @@ -631,6 +669,8 @@ static int lm4857_suspend(struct i2c_client *dev, pm_message_t state) static int lm4857_resume(struct i2c_client *dev) { + DBG("Entered %s\n", __func__); + if (lm4857_state) { lm4857_regs[LM4857_CTRL] |= (lm4857_state & 0x0f); lm4857_write_regs(); @@ -640,6 +680,8 @@ static int lm4857_resume(struct i2c_client *dev) static void lm4857_shutdown(struct i2c_client *dev) { + DBG("Entered %s\n", __func__); + dev_dbg(&dev->dev, "lm4857_shutdown\n"); lm4857_regs[LM4857_CTRL] &= 0xf0; lm4857_write_regs(); @@ -671,6 +713,8 @@ static int __init neo1973_init(void) { int ret; + DBG("Entered %s\n", __func__); + neo1973_snd_device = platform_device_alloc("soc-audio", -1); if (!neo1973_snd_device) return -ENOMEM; @@ -691,6 +735,8 @@ static int __init neo1973_init(void) static void __exit neo1973_exit(void) { + DBG("Entered %s\n", __func__); + i2c_del_driver(&lm4857_i2c_driver); platform_device_unregister(neo1973_snd_device); } -- cgit From 4ba1327ab8ce179c40862f3dedb4ebaaa491d737 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 13 May 2008 14:51:19 +0200 Subject: [ALSA] soc - DAPM - Add bulk control registration Most SoC drivers cut'n'paste a loop iterating over an array to register their DAPM controls. Provide a function they can call instead. Signed-off-by: Mark Brown Cc: Graeme Gregory Cc: Frank Mandarino Cc: Jarkko Nikula Cc: Richard Purdie Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/soc-dapm.c | 27 +++++++++++++++++++++++++++ 1 file changed, 27 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c60200ccde6..811d6527101 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1233,6 +1233,33 @@ int snd_soc_dapm_new_control(struct snd_soc_codec *codec, } EXPORT_SYMBOL_GPL(snd_soc_dapm_new_control); +/** + * snd_soc_dapm_new_controls - create new dapm controls + * @codec: audio codec + * @widget: widget array + * @num: number of widgets + * + * Creates new DAPM controls based upon the templates. + * + * Returns 0 for success else error. + */ +int snd_soc_dapm_new_controls(struct snd_soc_codec *codec, + const struct snd_soc_dapm_widget *widget, + int num) +{ + int i, ret; + + for (i = 0; i < num; i++) { + ret = snd_soc_dapm_new_control(codec, widget); + if (ret < 0) + return ret; + widget++; + } + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_new_controls); + + /** * snd_soc_dapm_stream_event - send a stream event to the dapm core * @codec: audio codec -- cgit From 105f1c28442301237d20b05a3d52d9987614016f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 13 May 2008 14:52:19 +0200 Subject: [ALSA] soc - DAPM - Bulk route registration ASoC codecs and machine drivers that use DAPM routes all cut'n'paste a loop iterating over a null terminated array of routes. Factor out this into a bulk registration function, improving the error reporting for most users, and deprecate the old API to help out of tree users pick up the changes. Signed-off-by: Mark Brown Cc: Graeme Gregory Cc: Frank Mandarino Cc: Jarkko Nikula Cc: Richard Purdie Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/soc-dapm.c | 72 +++++++++++++++++++++++++++++++++++++++++----------- 1 file changed, 57 insertions(+), 15 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 811d6527101..1ef6d94b835 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -841,21 +841,8 @@ int snd_soc_dapm_sync_endpoints(struct snd_soc_codec *codec) } EXPORT_SYMBOL_GPL(snd_soc_dapm_sync_endpoints); -/** - * snd_soc_dapm_connect_input - connect dapm widgets - * @codec: audio codec - * @sink: name of target widget - * @control: mixer control name - * @source: name of source name - * - * Connects 2 dapm widgets together via a named audio path. The sink is - * the widget receiving the audio signal, whilst the source is the sender - * of the audio signal. - * - * Returns 0 for success else error. - */ -int snd_soc_dapm_connect_input(struct snd_soc_codec *codec, const char *sink, - const char * control, const char *source) +static int snd_soc_dapm_add_route(struct snd_soc_codec *codec, + const char *sink, const char *control, const char *source) { struct snd_soc_dapm_path *path; struct snd_soc_dapm_widget *wsource = NULL, *wsink = NULL, *w; @@ -957,8 +944,63 @@ err: kfree(path); return ret; } + +/** + * snd_soc_dapm_connect_input - connect dapm widgets + * @codec: audio codec + * @sink: name of target widget + * @control: mixer control name + * @source: name of source name + * + * Connects 2 dapm widgets together via a named audio path. The sink is + * the widget receiving the audio signal, whilst the source is the sender + * of the audio signal. + * + * This function has been deprecated in favour of snd_soc_dapm_add_routes(). + * + * Returns 0 for success else error. + */ +int snd_soc_dapm_connect_input(struct snd_soc_codec *codec, const char *sink, + const char *control, const char *source) +{ + return snd_soc_dapm_add_route(codec, sink, control, source); +} EXPORT_SYMBOL_GPL(snd_soc_dapm_connect_input); +/** + * snd_soc_dapm_add_routes - Add routes between DAPM widgets + * @codec: codec + * @route: audio routes + * @num: number of routes + * + * Connects 2 dapm widgets together via a named audio path. The sink is + * the widget receiving the audio signal, whilst the source is the sender + * of the audio signal. + * + * Returns 0 for success else error. On error all resources can be freed + * with a call to snd_soc_card_free(). + */ +int snd_soc_dapm_add_routes(struct snd_soc_codec *codec, + const struct snd_soc_dapm_route *route, int num) +{ + int i, ret; + + for (i = 0; i < num; i++) { + ret = snd_soc_dapm_add_route(codec, route->sink, + route->control, route->source); + if (ret < 0) { + printk(KERN_ERR "Failed to add route %s->%s\n", + route->source, + route->sink); + return ret; + } + route++; + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_add_routes); + /** * snd_soc_dapm_new_widgets - add new dapm widgets * @codec: audio codec -- cgit From d0cc0d3a95cc3c022ee118072d243d3670ec1663 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 13 May 2008 14:55:22 +0200 Subject: [ALSA] soc - tlv320aic3x - Convert to use bulk registration APIs Signed-off-by: Mark Brown Cc: Jarkko Nikula Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/codecs/tlv320aic3x.c | 15 ++++----------- 1 file changed, 4 insertions(+), 11 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 957996e0eba..cb8365ac0c0 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -497,7 +497,7 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { SND_SOC_DAPM_INPUT("LINE2R"), }; -static const char *intercon[][3] = { +static const struct snd_soc_dapm_route intercon[] = { /* Left Output */ {"Left DAC Mux", "DAC_L1", "Left DAC"}, {"Left DAC Mux", "DAC_L2", "Left DAC"}, @@ -641,22 +641,15 @@ static const char *intercon[][3] = { {"Right Line Out", NULL, "Right Line2 Bypass Mixer"}, {"Mono Out", NULL, "Right Line2 Bypass Mixer"}, {"Right HP Out", NULL, "Right Line2 Bypass Mixer"}, - - /* terminator */ - {NULL, NULL, NULL}, }; static int aic3x_add_widgets(struct snd_soc_codec *codec) { - int i; - - for (i = 0; i < ARRAY_SIZE(aic3x_dapm_widgets); i++) - snd_soc_dapm_new_control(codec, &aic3x_dapm_widgets[i]); + snd_soc_dapm_new_controls(codec, aic3x_dapm_widgets, + ARRAY_SIZE(aic3x_dapm_widgets)); /* set up audio path interconnects */ - for (i = 0; intercon[i][0] != NULL; i++) - snd_soc_dapm_connect_input(codec, intercon[i][0], - intercon[i][1], intercon[i][2]); + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); snd_soc_dapm_new_widgets(codec); return 0; -- cgit From 25191c45aec54cd01b53391bb0b0e1e60377a5fc Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 13 May 2008 14:55:48 +0200 Subject: [ALSA] soc - Zaurus - Convert to bulk DAPM registration APIs Signed-off-by: Mark Brown Cc: Richard Purdie Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/pxa/corgi.c | 12 ++++-------- sound/soc/pxa/poodle.c | 14 +++++--------- sound/soc/pxa/spitz.c | 14 +++++--------- sound/soc/pxa/tosa.c | 14 ++++---------- 4 files changed, 18 insertions(+), 36 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 7f32a116757..852f16d963a 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -247,7 +247,7 @@ SND_SOC_DAPM_HP("Headset Jack", NULL), }; /* Corgi machine audio map (connections to the codec pins) */ -static const char *audio_map[][3] = { +static const struct snd_soc_dapm_route audio_map[] = { /* headset Jack - in = micin, out = LHPOUT*/ {"Headset Jack", NULL, "LHPOUT"}, @@ -265,8 +265,6 @@ static const char *audio_map[][3] = { /* Same as the above but no mic bias for line signals */ {"MICIN", NULL, "Line Jack"}, - - {NULL, NULL, NULL}, }; static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset", @@ -303,13 +301,11 @@ static int corgi_wm8731_init(struct snd_soc_codec *codec) } /* Add corgi specific widgets */ - for (i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++) - snd_soc_dapm_new_control(codec, &wm8731_dapm_widgets[i]); + snd_soc_dapm_new_controls(codec, wm8731_dapm_widgets, + ARRAY_SIZE(wm8731_dapm_widgets)); /* Set up corgi specific audio path audio_map */ - for (i = 0; audio_map[i][0] != NULL; i++) - snd_soc_dapm_connect_input(codec, audio_map[i][0], - audio_map[i][1], audio_map[i][2]); + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); snd_soc_dapm_sync_endpoints(codec); return 0; diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 7e830b21894..810f1fe158a 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -215,8 +215,8 @@ SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_SPK("Ext Spk", poodle_amp_event), }; -/* Corgi machine audio_mapnections to the codec pins */ -static const char *audio_map[][3] = { +/* Corgi machine connections to the codec pins */ +static const struct snd_soc_dapm_route audio_map[] = { /* headphone connected to LHPOUT1, RHPOUT1 */ {"Headphone Jack", NULL, "LHPOUT"}, @@ -225,8 +225,6 @@ static const char *audio_map[][3] = { /* speaker connected to LOUT, ROUT */ {"Ext Spk", NULL, "ROUT"}, {"Ext Spk", NULL, "LOUT"}, - - {NULL, NULL, NULL}, }; static const char *jack_function[] = {"Off", "Headphone"}; @@ -263,13 +261,11 @@ static int poodle_wm8731_init(struct snd_soc_codec *codec) } /* Add poodle specific widgets */ - for (i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++) - snd_soc_dapm_new_control(codec, &wm8731_dapm_widgets[i]); + snd_soc_dapm_new_controls(codec, wm8731_dapm_widgets, + ARRAY_SIZE(wm8731_dapm_widgets)); /* Set up poodle specific audio path audio_map */ - for (i = 0; audio_map[i][0] != NULL; i++) - snd_soc_dapm_connect_input(codec, audio_map[i][0], - audio_map[i][1], audio_map[i][2]); + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); snd_soc_dapm_sync_endpoints(codec); return 0; diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index d8b8372db00..73a6df8709e 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -250,7 +250,7 @@ static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = { }; /* Spitz machine audio_map */ -static const char *audio_map[][3] = { +static const struct snd_soc_dapm_route audio_map[] = { /* headphone connected to LOUT1, ROUT1 */ {"Headphone Jack", NULL, "LOUT1"}, @@ -269,8 +269,6 @@ static const char *audio_map[][3] = { /* line is connected to input 1 - no bias */ {"LINPUT1", NULL, "Line Jack"}, - - {NULL, NULL, NULL}, }; static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset", @@ -313,13 +311,11 @@ static int spitz_wm8750_init(struct snd_soc_codec *codec) } /* Add spitz specific widgets */ - for (i = 0; i < ARRAY_SIZE(wm8750_dapm_widgets); i++) - snd_soc_dapm_new_control(codec, &wm8750_dapm_widgets[i]); + snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets, + ARRAY_SIZE(wm8750_dapm_widgets)); - /* Set up spitz specific audio path audio_map */ - for (i = 0; audio_map[i][0] != NULL; i++) - snd_soc_dapm_connect_input(codec, audio_map[i][0], - audio_map[i][1], audio_map[i][2]); + /* Set up spitz specific audio paths */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); snd_soc_dapm_sync_endpoints(codec); return 0; diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index 7346d7e5d06..fda2aa056d5 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -154,7 +154,7 @@ SND_SOC_DAPM_SPK("Speaker", NULL), }; /* tosa audio map */ -static const char *audio_map[][3] = { +static const snd_soc_dapm_route audio_map[] = { /* headphone connected to HPOUTL, HPOUTR */ {"Headphone Jack", NULL, "HPOUTL"}, @@ -173,8 +173,6 @@ static const char *audio_map[][3] = { {"Headset Jack", NULL, "HPOUTR"}, {"LINEINR", NULL, "Mic Bias"}, {"Mic Bias", NULL, "Headset Jack"}, - - {NULL, NULL, NULL}, }; static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset", @@ -208,15 +206,11 @@ static int tosa_ac97_init(struct snd_soc_codec *codec) } /* add tosa specific widgets */ - for (i = 0; i < ARRAY_SIZE(tosa_dapm_widgets); i++) { - snd_soc_dapm_new_control(codec, &tosa_dapm_widgets[i]); - } + snd_soc_dapm_new_controls(codec, &tosa_dapm_widgets, + ARRAY_SIZE(tosa_dapm_widgets)); /* set up tosa specific audio path audio_map */ - for (i = 0; audio_map[i][0] != NULL; i++) { - snd_soc_dapm_connect_input(codec, audio_map[i][0], - audio_map[i][1], audio_map[i][2]); - } + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); snd_soc_dapm_sync_endpoints(codec); return 0; -- cgit From 51e6a8411a9440f0fdba6cdd7d779e74f89debc4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 13 May 2008 14:57:37 +0200 Subject: [ALSA] soc - eti_b1_wm8731 - Convert to use bulk DAPM control registration Signed-off-by: Mark Brown Cc: Frank Mandarino Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/at91/eti_b1_wm8731.c | 17 ++++------------- 1 file changed, 4 insertions(+), 13 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c index 1347dcf3f80..4a383a4a0ff 100644 --- a/sound/soc/at91/eti_b1_wm8731.c +++ b/sound/soc/at91/eti_b1_wm8731.c @@ -191,7 +191,7 @@ static const struct snd_soc_dapm_widget eti_b1_dapm_widgets[] = { SND_SOC_DAPM_SPK("Ext Spk", NULL), }; -static const char *intercon[][3] = { +static const struct snd_soc_dapm_route intercon[] = { /* speaker connected to LHPOUT */ {"Ext Spk", NULL, "LHPOUT"}, @@ -199,9 +199,6 @@ static const char *intercon[][3] = { /* mic is connected to Mic Jack, with WM8731 Mic Bias */ {"MICIN", NULL, "Mic Bias"}, {"Mic Bias", NULL, "Int Mic"}, - - /* terminator */ - {NULL, NULL, NULL}, }; /* @@ -209,20 +206,14 @@ static const char *intercon[][3] = { */ static int eti_b1_wm8731_init(struct snd_soc_codec *codec) { - int i; - DBG("eti_b1_wm8731_init() called\n"); /* Add specific widgets */ - for(i = 0; i < ARRAY_SIZE(eti_b1_dapm_widgets); i++) { - snd_soc_dapm_new_control(codec, &eti_b1_dapm_widgets[i]); - } + snd_soc_dapm_new_controls(codec, eti_b1_dapm_widgets, + ARRAY_SIZE(eti_b1_dapm_widgets)); /* Set up specific audio path interconnects */ - for(i = 0; intercon[i][0] != NULL; i++) { - snd_soc_dapm_connect_input(codec, intercon[i][0], - intercon[i][1], intercon[i][2]); - } + snd_soc_dapm_add_route(codec, intercon, ARRAY_SIZE(intercon)); /* not connected */ snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0); -- cgit From 8f3112d7a847c2933a42ce29f17899f585d09106 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 13 May 2008 14:58:03 +0200 Subject: [ALSA] soc - neo1973_wm8753 - Convert to bulk DAPM registration APIs Signed-off-by: Mark Brown Cc: Graeme Gregory Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/s3c24xx/neo1973_wm8753.c | 17 ++++++----------- 1 file changed, 6 insertions(+), 11 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index e469186a407..79c5027273c 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -425,8 +425,7 @@ static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = { }; -/* example machine audio_mapnections */ -static const char *audio_map[][3] = { +static const struct snd_soc_dapm_route dapm_routes[] = { /* Connections to the lm4857 amp */ {"Audio Out", NULL, "LOUT1"}, @@ -449,8 +448,6 @@ static const char *audio_map[][3] = { /* Connect the ALC pins */ {"ACIN", NULL, "ACOP"}, - - {NULL, NULL, NULL}, }; static const char *lm4857_mode[] = { @@ -526,8 +523,8 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec) set_scenario_endpoints(codec, NEO_AUDIO_OFF); /* Add neo1973 specific widgets */ - for (i = 0; i < ARRAY_SIZE(wm8753_dapm_widgets); i++) - snd_soc_dapm_new_control(codec, &wm8753_dapm_widgets[i]); + snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets, + ARRAY_SIZE(wm8753_dapm_widgets)); /* add neo1973 specific controls */ for (i = 0; i < ARRAY_SIZE(wm8753_neo1973_controls); i++) { @@ -538,11 +535,9 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec) return err; } - /* set up neo1973 specific audio path audio_mapnects */ - for (i = 0; audio_map[i][0] != NULL; i++) { - snd_soc_dapm_connect_input(codec, audio_map[i][0], - audio_map[i][1], audio_map[i][2]); - } + /* set up neo1973 specific audio routes */ + err = snd_soc_dapm_add_routes(codec, dapm_routes, + ARRAY_SIZE(dapm_routes)); snd_soc_dapm_sync_endpoints(codec); return 0; -- cgit From acf497f996aa08f03c62a3150abd7939ae23de4c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 13 May 2008 14:58:30 +0200 Subject: [ALSA] soc - davinci-evm - Update for bulk DAPM registration APIs Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/davinci/davinci-evm.c | 14 ++++---------- 1 file changed, 4 insertions(+), 10 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index fcd16524033..4c70a0ed339 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -71,7 +71,7 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { }; /* davinci-evm machine audio_mapnections to the codec pins */ -static const char *audio_map[][3] = { +static const struct snd_soc_dapm_route audio_map[] = { /* Headphone connected to HPLOUT, HPROUT */ {"Headphone Jack", NULL, "HPLOUT"}, {"Headphone Jack", NULL, "HPROUT"}, @@ -90,23 +90,17 @@ static const char *audio_map[][3] = { {"LINE2L", NULL, "Line In"}, {"LINE1R", NULL, "Line In"}, {"LINE2R", NULL, "Line In"}, - - {NULL, NULL, NULL}, }; /* Logic for a aic3x as connected on a davinci-evm */ static int evm_aic3x_init(struct snd_soc_codec *codec) { - int i; - /* Add davinci-evm specific widgets */ - for (i = 0; i < ARRAY_SIZE(aic3x_dapm_widgets); i++) - snd_soc_dapm_new_control(codec, &aic3x_dapm_widgets[i]); + snd_soc_dapm_new_controls(codec, aic3x_dapm_widgets, + ARRAY_SIZE(aic3x_dapm_widgets)); /* Set up davinci-evm specific audio path audio_map */ - for (i = 0; audio_map[i][0] != NULL; i++) - snd_soc_dapm_connect_input(codec, audio_map[i][0], - audio_map[i][1], audio_map[i][2]); + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); /* not connected */ snd_soc_dapm_set_endpoint(codec, "MONO_LOUT", 0); -- cgit From 1a2505988ea650b61bd07722e99080a40ff27653 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 13 May 2008 14:58:57 +0200 Subject: [ALSA] soc - n810 - Update for bulk DAPM registration APIs Signed-off-by: Mark Brown Cc: Jarkko Nikula Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/omap/n810.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 6533563a601..c2afffbe8da 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -177,7 +177,7 @@ static const struct snd_soc_dapm_widget aic33_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", n810_jack_event), }; -static const char *audio_map[][3] = { +static const struct snd_soc_dapm_route audio_map[] = { {"Headphone Jack", NULL, "HPLOUT"}, {"Headphone Jack", NULL, "HPROUT"}, @@ -217,13 +217,11 @@ static int n810_aic33_init(struct snd_soc_codec *codec) } /* Add N810 specific widgets */ - for (i = 0; i < ARRAY_SIZE(aic33_dapm_widgets); i++) - snd_soc_dapm_new_control(codec, &aic33_dapm_widgets[i]); + snd_soc_dapm_new_controls(codec, aic33_dapm_widgets, + ARRAY_SIZE(aic33_dapm_widgets)); /* Set up N810 specific audio path audio_map */ - for (i = 0; i < ARRAY_SIZE(audio_map); i++) - snd_soc_dapm_connect_input(codec, audio_map[i][0], - audio_map[i][1], audio_map[i][2]); + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); snd_soc_dapm_sync_endpoints(codec); -- cgit From f99a633a151686a599413bef758253dfd04887d1 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Thu, 15 May 2008 11:01:36 +0200 Subject: [ALSA] ASoC: Convert N810 machine driver to use gpiolib Use gpiolib since it is now available for OMAPs. Change also references to HW version RX44 to product name N810. Signed-off-by: Jarkko Nikula Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/omap/n810.c | 25 ++++++++++++++----------- 1 file changed, 14 insertions(+), 11 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index c2afffbe8da..c32487b6c7b 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -30,15 +30,15 @@ #include #include -#include +#include #include #include "omap-mcbsp.h" #include "omap-pcm.h" #include "../codecs/tlv320aic3x.h" -#define RX44_HEADSET_AMP_GPIO 10 -#define RX44_SPEAKER_AMP_GPIO 101 +#define N810_HEADSET_AMP_GPIO 10 +#define N810_SPEAKER_AMP_GPIO 101 static struct clk *sys_clkout2; static struct clk *sys_clkout2_src; @@ -154,9 +154,9 @@ static int n810_spk_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { if (SND_SOC_DAPM_EVENT_ON(event)) - omap_set_gpio_dataout(RX44_SPEAKER_AMP_GPIO, 1); + gpio_set_value(N810_SPEAKER_AMP_GPIO, 1); else - omap_set_gpio_dataout(RX44_SPEAKER_AMP_GPIO, 0); + gpio_set_value(N810_SPEAKER_AMP_GPIO, 0); return 0; } @@ -165,9 +165,9 @@ static int n810_jack_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { if (SND_SOC_DAPM_EVENT_ON(event)) - omap_set_gpio_dataout(RX44_HEADSET_AMP_GPIO, 1); + gpio_set_value(N810_HEADSET_AMP_GPIO, 1); else - omap_set_gpio_dataout(RX44_HEADSET_AMP_GPIO, 0); + gpio_set_value(N810_HEADSET_AMP_GPIO, 0); return 0; } @@ -303,12 +303,12 @@ static int __init n810_soc_init(void) clk_set_parent(sys_clkout2_src, func96m_clk); clk_set_rate(sys_clkout2, 12000000); - if (omap_request_gpio(RX44_HEADSET_AMP_GPIO) < 0) + if (gpio_request(N810_HEADSET_AMP_GPIO, "hs_amp") < 0) BUG(); - if (omap_request_gpio(RX44_SPEAKER_AMP_GPIO) < 0) + if (gpio_request(N810_SPEAKER_AMP_GPIO, "spk_amp") < 0) BUG(); - omap_set_gpio_direction(RX44_HEADSET_AMP_GPIO, 0); - omap_set_gpio_direction(RX44_SPEAKER_AMP_GPIO, 0); + gpio_direction_output(N810_HEADSET_AMP_GPIO, 0); + gpio_direction_output(N810_SPEAKER_AMP_GPIO, 0); return 0; err2: @@ -323,6 +323,9 @@ err1: static void __exit n810_soc_exit(void) { + gpio_free(N810_SPEAKER_AMP_GPIO); + gpio_free(N810_HEADSET_AMP_GPIO); + platform_device_unregister(n810_snd_device); } -- cgit From b2efbbfba24efc8456d68d5af42d082ab1c2febc Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 19 May 2008 12:30:58 +0200 Subject: [ALSA] ASoC: Remove in-code changelogs The overwhelming majority just say 'initial version' anyway. Signed-off-by: Mark Brown Acked-by: Ben Dooks Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/codecs/ac97.c | 3 --- sound/soc/codecs/wm9712.c | 3 --- sound/soc/codecs/wm9713.c | 3 --- sound/soc/pxa/corgi.c | 4 ---- sound/soc/pxa/pxa2xx-i2s.c | 3 --- sound/soc/pxa/spitz.c | 3 --- sound/soc/pxa/tosa.c | 3 --- sound/soc/s3c24xx/neo1973_wm8753.c | 4 ---- sound/soc/s3c24xx/s3c2443-ac97.c | 3 --- sound/soc/s3c24xx/s3c24xx-i2s.c | 5 ----- sound/soc/s3c24xx/s3c24xx-pcm.c | 4 ---- sound/soc/s3c24xx/smdk2443_wm9710.c | 3 --- sound/soc/soc-core.c | 4 ---- sound/soc/soc-dapm.c | 5 ----- 14 files changed, 50 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 2a1ffe39690..7bf2081b46f 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -10,9 +10,6 @@ * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. * - * Revision history - * 17th Oct 2005 Initial version. - * * Generic AC97 support. */ diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 76c1e2d33e7..89efe40c7c3 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -9,9 +9,6 @@ * 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. - * - * Revision history - * 4th Feb 2006 Initial version. */ #include diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 1f241161445..9e6b2fd7262 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -10,9 +10,6 @@ * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. * - * Revision history - * 4th Feb 2006 Initial version. - * * Features:- * * o Support for AC97 Codec, Voice DAC and Aux DAC diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 852f16d963a..edeea63e80e 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -11,10 +11,6 @@ * 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. - * - * Revision history - * 30th Nov 2005 Initial version. - * */ #include diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 42507103097..35090c2870f 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -9,9 +9,6 @@ * 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. - * - * Revision history - * 12th Aug 2005 Initial version. */ #include diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 73a6df8709e..092b5c776b4 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -12,9 +12,6 @@ * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. * - * Revision history - * 30th Nov 2005 Initial version. - * */ #include diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index fda2aa056d5..465ff0f458e 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -12,9 +12,6 @@ * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. * - * Revision history - * 30th Nov 2005 Initial version. - * * GPIO's * 1 - Jack Insertion * 5 - Hookswitch (headset answer/hang up switch) diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 79c5027273c..c1a0161bc72 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -10,10 +10,6 @@ * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. * - * Revision history - * 20th Jan 2007 Initial version. - * 05th Feb 2007 Rename all to Neo1973 - * */ #include diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c index e81d9a6c83d..0eed140dcd9 100644 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ b/sound/soc/s3c24xx/s3c2443-ac97.c @@ -10,9 +10,6 @@ * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. - * - * Revision history - * 21st Mar 2007 Initial Version */ #include diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 1ed6afd4545..ddf87246c77 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -12,11 +12,6 @@ * 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. - * - * - * Revision history - * 11th Dec 2006 Merged with Simtec driver - * 10th Nov 2006 Initial version. */ #include diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c index 7806ae61461..ef599745159 100644 --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c @@ -12,10 +12,6 @@ * 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. - * - * Revision history - * 11th Dec 2006 Merged with Simtec driver - * 10th Nov 2006 Initial version. */ #include diff --git a/sound/soc/s3c24xx/smdk2443_wm9710.c b/sound/soc/s3c24xx/smdk2443_wm9710.c index b4a56302b9a..8515d6ff03f 100644 --- a/sound/soc/s3c24xx/smdk2443_wm9710.c +++ b/sound/soc/s3c24xx/smdk2443_wm9710.c @@ -10,9 +10,6 @@ * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. * - * Revision history - * 8th Mar 2007 Initial version. - * */ #include diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e148db940cf..0318d8abe3e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -14,10 +14,6 @@ * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. * - * Revision history - * 12th Aug 2005 Initial version. - * 25th Oct 2005 Working Codec, Interface and Platform registration. - * * TODO: * o Add hw rules to enforce rates, etc. * o More testing with other codecs/machines. diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1ef6d94b835..8a3192bcee7 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -10,11 +10,6 @@ * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. * - * Revision history - * 12th Aug 2005 Initial version. - * 25th Oct 2005 Implemented path power domain. - * 18th Dec 2005 Implemented machine and stream level power domain. - * * Features: * o Changes power status of internal codec blocks depending on the * dynamic configuration of codec internal audio paths and active -- cgit From 0be9898adb6f58fee44f0fec0bbc0eac997ea9eb Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 19 May 2008 12:31:28 +0200 Subject: [ALSA] ASoC: Clarify API for bias configuration Currently the ASoC core configures the bias levels in the system using a callback on codecs and machines called 'dapm_event', passing it PCI style power levels as SNDRV_CTL_POWER_ constants. This is more obscure than it needs to be and has caused confusion to driver authors, especially given that DAPM is also performing power management. Address this by renaming the callback function to 'set_bias_level' and using constants explicitly representing the off, standby, pre-on and on states which DAPM transitions through. Also unexport the API for setting bias level: there are currently no in-tree users of this API other than the core itself and it is likely that the core would need to be extended to cater for any users. Signed-off-by: Mark Brown Cc: Jarkko Nikula Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/codecs/tlv320aic3x.c | 26 ++++++++++++------------- sound/soc/codecs/wm8731.c | 28 +++++++++++++-------------- sound/soc/codecs/wm8750.c | 36 +++++++++++++++++----------------- sound/soc/codecs/wm8753.c | 36 +++++++++++++++++----------------- sound/soc/codecs/wm9712.c | 28 +++++++++++++-------------- sound/soc/codecs/wm9713.c | 26 ++++++++++++------------- sound/soc/soc-core.c | 25 ++++++++++++------------ sound/soc/soc-dapm.c | 44 ++++++++++++++++++++---------------------- 8 files changed, 124 insertions(+), 125 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index cb8365ac0c0..dc8a38d9e53 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -847,13 +847,14 @@ static int aic3x_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, return 0; } -static int aic3x_dapm_event(struct snd_soc_codec *codec, int event) +static int aic3x_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) { struct aic3x_priv *aic3x = codec->private_data; u8 reg; - switch (event) { - case SNDRV_CTL_POWER_D0: + switch (level) { + case SND_SOC_BIAS_ON: /* all power is driven by DAPM system */ if (aic3x->master) { /* enable pll */ @@ -862,10 +863,9 @@ static int aic3x_dapm_event(struct snd_soc_codec *codec, int event) reg | PLL_ENABLE); } break; - case SNDRV_CTL_POWER_D1: - case SNDRV_CTL_POWER_D2: + case SND_SOC_BIAS_PREPARE: break; - case SNDRV_CTL_POWER_D3hot: + case SND_SOC_BIAS_STANDBY: /* * all power is driven by DAPM system, * so output power is safe if bypass was set @@ -877,7 +877,7 @@ static int aic3x_dapm_event(struct snd_soc_codec *codec, int event) reg & ~PLL_ENABLE); } break; - case SNDRV_CTL_POWER_D3cold: + case SND_SOC_BIAS_OFF: /* force all power off */ reg = aic3x_read_reg_cache(codec, LINE1L_2_LADC_CTRL); aic3x_write(codec, LINE1L_2_LADC_CTRL, reg & ~LADC_PWR_ON); @@ -913,7 +913,7 @@ static int aic3x_dapm_event(struct snd_soc_codec *codec, int event) } break; } - codec->dapm_state = event; + codec->bias_level = level; return 0; } @@ -979,7 +979,7 @@ static int aic3x_suspend(struct platform_device *pdev, pm_message_t state) struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->codec; - aic3x_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + aic3x_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } @@ -999,7 +999,7 @@ static int aic3x_resume(struct platform_device *pdev) codec->hw_write(codec->control_data, data, 2); } - aic3x_dapm_event(codec, codec->suspend_dapm_state); + aic3x_set_bias_level(codec, codec->suspend_bias_level); return 0; } @@ -1018,7 +1018,7 @@ static int aic3x_init(struct snd_soc_device *socdev) codec->owner = THIS_MODULE; codec->read = aic3x_read_reg_cache; codec->write = aic3x_write; - codec->dapm_event = aic3x_dapm_event; + codec->set_bias_level = aic3x_set_bias_level; codec->dai = &aic3x_dai; codec->num_dai = 1; codec->reg_cache_size = sizeof(aic3x_reg); @@ -1100,7 +1100,7 @@ static int aic3x_init(struct snd_soc_device *socdev) aic3x_write(codec, LINE2R_2_MONOLOPM_VOL, DEFAULT_VOL); /* off, with power on */ - aic3x_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + aic3x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* setup GPIO functions */ aic3x_write(codec, AIC3X_GPIO1_REG, (setup->gpio_func[0] & 0xf) << 4); @@ -1271,7 +1271,7 @@ static int aic3x_remove(struct platform_device *pdev) /* power down chip */ if (codec->control_data) - aic3x_dapm_event(codec, SNDRV_CTL_POWER_D3); + aic3x_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 0cf9265fca8..0f28aa4bccc 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -435,29 +435,29 @@ static int wm8731_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, return 0; } -static int wm8731_dapm_event(struct snd_soc_codec *codec, int event) +static int wm8731_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) { u16 reg = wm8731_read_reg_cache(codec, WM8731_PWR) & 0xff7f; - switch (event) { - case SNDRV_CTL_POWER_D0: /* full On */ + switch (level) { + case SND_SOC_BIAS_ON: /* vref/mid, osc on, dac unmute */ wm8731_write(codec, WM8731_PWR, reg); break; - case SNDRV_CTL_POWER_D1: /* partial On */ - case SNDRV_CTL_POWER_D2: /* partial On */ + case SND_SOC_BIAS_PREPARE: break; - case SNDRV_CTL_POWER_D3hot: /* Off, with power */ + case SND_SOC_BIAS_STANDBY: /* everything off except vref/vmid, */ wm8731_write(codec, WM8731_PWR, reg | 0x0040); break; - case SNDRV_CTL_POWER_D3cold: /* Off, without power */ + case SND_SOC_BIAS_OFF: /* everything off, dac mute, inactive */ wm8731_write(codec, WM8731_ACTIVE, 0x0); wm8731_write(codec, WM8731_PWR, 0xffff); break; } - codec->dapm_state = event; + codec->bias_level = level; return 0; } @@ -503,7 +503,7 @@ static int wm8731_suspend(struct platform_device *pdev, pm_message_t state) struct snd_soc_codec *codec = socdev->codec; wm8731_write(codec, WM8731_ACTIVE, 0x0); - wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + wm8731_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } @@ -521,8 +521,8 @@ static int wm8731_resume(struct platform_device *pdev) data[1] = cache[i] & 0x00ff; codec->hw_write(codec->control_data, data, 2); } - wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3hot); - wm8731_dapm_event(codec, codec->suspend_dapm_state); + wm8731_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + wm8731_set_bias_level(codec, codec->suspend_bias_level); return 0; } @@ -539,7 +539,7 @@ static int wm8731_init(struct snd_soc_device *socdev) codec->owner = THIS_MODULE; codec->read = wm8731_read_reg_cache; codec->write = wm8731_write; - codec->dapm_event = wm8731_dapm_event; + codec->set_bias_level = wm8731_set_bias_level; codec->dai = &wm8731_dai; codec->num_dai = 1; codec->reg_cache_size = sizeof(wm8731_reg); @@ -557,7 +557,7 @@ static int wm8731_init(struct snd_soc_device *socdev) } /* power on device */ - wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + wm8731_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* set the update bits */ reg = wm8731_read_reg_cache(codec, WM8731_LOUT1V); @@ -730,7 +730,7 @@ static int wm8731_remove(struct platform_device *pdev) struct snd_soc_codec *codec = socdev->codec; if (codec->control_data) - wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + wm8731_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 16cd5d4d5ad..62423f4493b 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -686,29 +686,29 @@ static int wm8750_mute(struct snd_soc_codec_dai *dai, int mute) return 0; } -static int wm8750_dapm_event(struct snd_soc_codec *codec, int event) +static int wm8750_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) { u16 pwr_reg = wm8750_read_reg_cache(codec, WM8750_PWR1) & 0xfe3e; - switch (event) { - case SNDRV_CTL_POWER_D0: /* full On */ + switch (level) { + case SND_SOC_BIAS_ON: /* set vmid to 50k and unmute dac */ wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x00c0); break; - case SNDRV_CTL_POWER_D1: /* partial On */ - case SNDRV_CTL_POWER_D2: /* partial On */ + case SND_SOC_BIAS_PREPARE: /* set vmid to 5k for quick power up */ wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x01c1); break; - case SNDRV_CTL_POWER_D3hot: /* Off, with power */ + case SND_SOC_BIAS_STANDBY: /* mute dac and set vmid to 500k, enable VREF */ wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x0141); break; - case SNDRV_CTL_POWER_D3cold: /* Off, without power */ + case SND_SOC_BIAS_OFF: wm8750_write(codec, WM8750_PWR1, 0x0001); break; } - codec->dapm_state = event; + codec->bias_level = level; return 0; } @@ -748,7 +748,7 @@ static void wm8750_work(struct work_struct *work) { struct snd_soc_codec *codec = container_of(work, struct snd_soc_codec, delayed_work.work); - wm8750_dapm_event(codec, codec->dapm_state); + wm8750_set_bias_level(codec, codec->bias_level); } static int wm8750_suspend(struct platform_device *pdev, pm_message_t state) @@ -756,7 +756,7 @@ static int wm8750_suspend(struct platform_device *pdev, pm_message_t state) struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->codec; - wm8750_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + wm8750_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } @@ -777,12 +777,12 @@ static int wm8750_resume(struct platform_device *pdev) codec->hw_write(codec->control_data, data, 2); } - wm8750_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + wm8750_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* charge wm8750 caps */ - if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) { - wm8750_dapm_event(codec, SNDRV_CTL_POWER_D2); - codec->dapm_state = SNDRV_CTL_POWER_D0; + if (codec->suspend_bias_level == SND_SOC_BIAS_ON) { + wm8750_set_bias_level(codec, SND_SOC_BIAS_PREPARE); + codec->bias_level = SND_SOC_BIAS_ON; schedule_delayed_work(&codec->delayed_work, msecs_to_jiffies(1000)); } @@ -803,7 +803,7 @@ static int wm8750_init(struct snd_soc_device *socdev) codec->owner = THIS_MODULE; codec->read = wm8750_read_reg_cache; codec->write = wm8750_write; - codec->dapm_event = wm8750_dapm_event; + codec->set_bias_level = wm8750_set_bias_level; codec->dai = &wm8750_dai; codec->num_dai = 1; codec->reg_cache_size = sizeof(wm8750_reg); @@ -821,8 +821,8 @@ static int wm8750_init(struct snd_soc_device *socdev) } /* charge output caps */ - wm8750_dapm_event(codec, SNDRV_CTL_POWER_D2); - codec->dapm_state = SNDRV_CTL_POWER_D3hot; + wm8750_set_bias_level(codec, SND_SOC_BIAS_PREPARE); + codec->bias_level = SND_SOC_BIAS_STANDBY; schedule_delayed_work(&codec->delayed_work, msecs_to_jiffies(1000)); /* set the update bits */ @@ -1021,7 +1021,7 @@ static int wm8750_remove(struct platform_device *pdev) struct snd_soc_codec *codec = socdev->codec; if (codec->control_data) - wm8750_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + wm8750_set_bias_level(codec, SND_SOC_BIAS_OFF); run_delayed_work(&codec->delayed_work); snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index fb41826c4c4..9032b0c07c8 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1274,29 +1274,29 @@ static int wm8753_mute(struct snd_soc_codec_dai *dai, int mute) return 0; } -static int wm8753_dapm_event(struct snd_soc_codec *codec, int event) +static int wm8753_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) { u16 pwr_reg = wm8753_read_reg_cache(codec, WM8753_PWR1) & 0xfe3e; - switch (event) { - case SNDRV_CTL_POWER_D0: /* full On */ + switch (level) { + case SND_SOC_BIAS_ON: /* set vmid to 50k and unmute dac */ wm8753_write(codec, WM8753_PWR1, pwr_reg | 0x00c0); break; - case SNDRV_CTL_POWER_D1: /* partial On */ - case SNDRV_CTL_POWER_D2: /* partial On */ + case SND_SOC_BIAS_PREPARE: /* set vmid to 5k for quick power up */ wm8753_write(codec, WM8753_PWR1, pwr_reg | 0x01c1); break; - case SNDRV_CTL_POWER_D3hot: /* Off, with power */ + case SND_SOC_BIAS_STANDBY: /* mute dac and set vmid to 500k, enable VREF */ wm8753_write(codec, WM8753_PWR1, pwr_reg | 0x0141); break; - case SNDRV_CTL_POWER_D3cold: /* Off, without power */ + case SND_SOC_BIAS_OFF: wm8753_write(codec, WM8753_PWR1, 0x0001); break; } - codec->dapm_state = event; + codec->bias_level = level; return 0; } @@ -1500,7 +1500,7 @@ static void wm8753_work(struct work_struct *work) { struct snd_soc_codec *codec = container_of(work, struct snd_soc_codec, delayed_work.work); - wm8753_dapm_event(codec, codec->dapm_state); + wm8753_set_bias_level(codec, codec->bias_level); } static int wm8753_suspend(struct platform_device *pdev, pm_message_t state) @@ -1512,7 +1512,7 @@ static int wm8753_suspend(struct platform_device *pdev, pm_message_t state) if (!codec->card) return 0; - wm8753_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } @@ -1537,12 +1537,12 @@ static int wm8753_resume(struct platform_device *pdev) codec->hw_write(codec->control_data, data, 2); } - wm8753_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + wm8753_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* charge wm8753 caps */ - if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) { - wm8753_dapm_event(codec, SNDRV_CTL_POWER_D2); - codec->dapm_state = SNDRV_CTL_POWER_D0; + if (codec->suspend_bias_level == SND_SOC_BIAS_ON) { + wm8753_set_bias_level(codec, SND_SOC_BIAS_PREPARE); + codec->bias_level = SND_SOC_BIAS_ON; schedule_delayed_work(&codec->delayed_work, msecs_to_jiffies(caps_charge)); } @@ -1563,7 +1563,7 @@ static int wm8753_init(struct snd_soc_device *socdev) codec->owner = THIS_MODULE; codec->read = wm8753_read_reg_cache; codec->write = wm8753_write; - codec->dapm_event = wm8753_dapm_event; + codec->set_bias_level = wm8753_set_bias_level; codec->dai = wm8753_dai; codec->num_dai = 2; codec->reg_cache_size = sizeof(wm8753_reg); @@ -1584,8 +1584,8 @@ static int wm8753_init(struct snd_soc_device *socdev) } /* charge output caps */ - wm8753_dapm_event(codec, SNDRV_CTL_POWER_D2); - codec->dapm_state = SNDRV_CTL_POWER_D3hot; + wm8753_set_bias_level(codec, SND_SOC_BIAS_PREPARE); + codec->bias_level = SND_SOC_BIAS_STANDBY; schedule_delayed_work(&codec->delayed_work, msecs_to_jiffies(caps_charge)); @@ -1792,7 +1792,7 @@ static int wm8753_remove(struct platform_device *pdev) struct snd_soc_codec *codec = socdev->codec; if (codec->control_data) - wm8753_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF); run_delayed_work(&codec->delayed_work); snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 89efe40c7c3..e26cfcf0b4f 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -571,23 +571,23 @@ struct snd_soc_codec_dai wm9712_dai[] = { }; EXPORT_SYMBOL_GPL(wm9712_dai); -static int wm9712_dapm_event(struct snd_soc_codec *codec, int event) +static int wm9712_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) { - switch (event) { - case SNDRV_CTL_POWER_D0: /* full On */ - case SNDRV_CTL_POWER_D1: /* partial On */ - case SNDRV_CTL_POWER_D2: /* partial On */ + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: break; - case SNDRV_CTL_POWER_D3hot: /* Off, with power */ + case SND_SOC_BIAS_STANDBY: ac97_write(codec, AC97_POWERDOWN, 0x0000); break; - case SNDRV_CTL_POWER_D3cold: /* Off, without power */ + case SND_SOC_BIAS_OFF: /* disable everything including AC link */ ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff); ac97_write(codec, AC97_POWERDOWN, 0xffff); break; } - codec->dapm_state = event; + codec->bias_level = level; return 0; } @@ -615,7 +615,7 @@ static int wm9712_soc_suspend(struct platform_device *pdev, struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->codec; - wm9712_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + wm9712_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } @@ -632,7 +632,7 @@ static int wm9712_soc_resume(struct platform_device *pdev) return ret; } - wm9712_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + wm9712_set_bias_level(codec, SND_SOC_BIAS_STANDBY); if (ret == 0) { /* Sync reg_cache with the hardware after cold reset */ @@ -644,8 +644,8 @@ static int wm9712_soc_resume(struct platform_device *pdev) } } - if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) - wm9712_dapm_event(codec, SNDRV_CTL_POWER_D0); + if (codec->suspend_bias_level == SND_SOC_BIAS_ON) + wm9712_set_bias_level(codec, SND_SOC_BIAS_ON); return ret; } @@ -679,7 +679,7 @@ static int wm9712_soc_probe(struct platform_device *pdev) codec->num_dai = ARRAY_SIZE(wm9712_dai); codec->write = ac97_write; codec->read = ac97_read; - codec->dapm_event = wm9712_dapm_event; + codec->set_bias_level = wm9712_set_bias_level; INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); @@ -703,7 +703,7 @@ static int wm9712_soc_probe(struct platform_device *pdev) /* set alc mux to none */ ac97_write(codec, AC97_VIDEO, ac97_read(codec, AC97_VIDEO) | 0x3000); - wm9712_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + wm9712_set_bias_level(codec, SND_SOC_BIAS_STANDBY); wm9712_add_controls(codec); wm9712_add_widgets(codec); ret = snd_soc_register_card(socdev); diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 9e6b2fd7262..4863636e9d5 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1094,33 +1094,33 @@ int wm9713_reset(struct snd_soc_codec *codec, int try_warm) } EXPORT_SYMBOL_GPL(wm9713_reset); -static int wm9713_dapm_event(struct snd_soc_codec *codec, int event) +static int wm9713_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) { u16 reg; - switch (event) { - case SNDRV_CTL_POWER_D0: /* full On */ + switch (level) { + case SND_SOC_BIAS_ON: /* enable thermal shutdown */ reg = ac97_read(codec, AC97_EXTENDED_MID) & 0x1bff; ac97_write(codec, AC97_EXTENDED_MID, reg); break; - case SNDRV_CTL_POWER_D1: /* partial On */ - case SNDRV_CTL_POWER_D2: /* partial On */ + case SND_SOC_BIAS_PREPARE: break; - case SNDRV_CTL_POWER_D3hot: /* Off, with power */ + case SND_SOC_BIAS_STANDBY: /* enable master bias and vmid */ reg = ac97_read(codec, AC97_EXTENDED_MID) & 0x3bff; ac97_write(codec, AC97_EXTENDED_MID, reg); ac97_write(codec, AC97_POWERDOWN, 0x0000); break; - case SNDRV_CTL_POWER_D3cold: /* Off, without power */ + case SND_SOC_BIAS_OFF: /* disable everything including AC link */ ac97_write(codec, AC97_EXTENDED_MID, 0xffff); ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff); ac97_write(codec, AC97_POWERDOWN, 0xffff); break; } - codec->dapm_state = event; + codec->bias_level = level; return 0; } @@ -1157,7 +1157,7 @@ static int wm9713_soc_resume(struct platform_device *pdev) return ret; } - wm9713_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + wm9713_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* do we need to re-start the PLL ? */ if (wm9713->pll_out) @@ -1173,8 +1173,8 @@ static int wm9713_soc_resume(struct platform_device *pdev) } } - if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) - wm9713_dapm_event(codec, SNDRV_CTL_POWER_D0); + if (codec->suspend_bias_level == SND_SOC_BIAS_ON) + wm9713_set_bias_level(codec, SND_SOC_BIAS_ON); return ret; } @@ -1213,7 +1213,7 @@ static int wm9713_soc_probe(struct platform_device *pdev) codec->num_dai = ARRAY_SIZE(wm9713_dai); codec->write = ac97_write; codec->read = ac97_read; - codec->dapm_event = wm9713_dapm_event; + codec->set_bias_level = wm9713_set_bias_level; INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); @@ -1235,7 +1235,7 @@ static int wm9713_soc_probe(struct platform_device *pdev) goto reset_err; } - wm9713_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + wm9713_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* unmute the adc - move to kcontrol */ reg = ac97_read(codec, AC97_CD) & 0x7fff; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0318d8abe3e..a05b3450aee 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -283,12 +283,12 @@ static void close_delayed_work(struct work_struct *work) /* are we waiting on this codec DAI stream */ if (codec_dai->pop_wait == 1) { - /* power down the codec to D1 if no longer active */ + /* Reduce power if no longer active */ if (codec->active == 0) { dbg("pop wq D1 %s %s\n", codec->name, codec_dai->playback.stream_name); - snd_soc_dapm_device_event(socdev, - SNDRV_CTL_POWER_D1); + snd_soc_dapm_set_bias_level(socdev, + SND_SOC_BIAS_PREPARE); } codec_dai->pop_wait = 0; @@ -296,12 +296,12 @@ static void close_delayed_work(struct work_struct *work) codec_dai->playback.stream_name, SND_SOC_DAPM_STREAM_STOP); - /* power down the codec power domain if no longer active */ + /* Fall into standby if no longer active */ if (codec->active == 0) { dbg("pop wq D3 %s %s\n", codec->name, codec_dai->playback.stream_name); - snd_soc_dapm_device_event(socdev, - SNDRV_CTL_POWER_D3hot); + snd_soc_dapm_set_bias_level(socdev, + SND_SOC_BIAS_STANDBY); } } } @@ -361,8 +361,8 @@ static int soc_codec_close(struct snd_pcm_substream *substream) SND_SOC_DAPM_STREAM_STOP); if (codec->active == 0 && codec_dai->pop_wait == 0) - snd_soc_dapm_device_event(socdev, - SNDRV_CTL_POWER_D3hot); + snd_soc_dapm_set_bias_level(socdev, + SND_SOC_BIAS_STANDBY); } mutex_unlock(&pcm_mutex); @@ -435,9 +435,10 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) } } else { /* no delayed work - do we need to power up codec */ - if (codec->dapm_state != SNDRV_CTL_POWER_D0) { + if (codec->bias_level != SND_SOC_BIAS_ON) { - snd_soc_dapm_device_event(socdev, SNDRV_CTL_POWER_D1); + snd_soc_dapm_set_bias_level(socdev, + SND_SOC_BIAS_PREPARE); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) snd_soc_dapm_stream_event(codec, @@ -448,7 +449,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) codec_dai->capture.stream_name, SND_SOC_DAPM_STREAM_START); - snd_soc_dapm_device_event(socdev, SNDRV_CTL_POWER_D0); + snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_ON); if (codec_dai->dai_ops.digital_mute) codec_dai->dai_ops.digital_mute(codec_dai, 0); @@ -658,7 +659,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) /* close any waiting streams and save state */ run_delayed_work(&socdev->delayed_work); - codec->suspend_dapm_state = codec->dapm_state; + codec->suspend_bias_level = codec->bias_level; for(i = 0; i < codec->num_dai; i++) { char *stream = codec->dai[i].playback.stream_name; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8a3192bcee7..728f3ac2f30 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -763,21 +763,18 @@ static ssize_t dapm_widget_show(struct device *dev, } } - switch(codec->dapm_state){ - case SNDRV_CTL_POWER_D0: - state = "D0"; + switch (codec->bias_level) { + case SND_SOC_BIAS_ON: + state = "On"; break; - case SNDRV_CTL_POWER_D1: - state = "D1"; + case SND_SOC_BIAS_PREPARE: + state = "Prepare"; break; - case SNDRV_CTL_POWER_D2: - state = "D2"; + case SND_SOC_BIAS_STANDBY: + state = "Standby"; break; - case SNDRV_CTL_POWER_D3hot: - state = "D3hot"; - break; - case SNDRV_CTL_POWER_D3cold: - state = "D3cold"; + case SND_SOC_BIAS_OFF: + state = "Off"; break; } count += sprintf(buf + count, "PM State: %s\n", state); @@ -1358,27 +1355,28 @@ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, EXPORT_SYMBOL_GPL(snd_soc_dapm_stream_event); /** - * snd_soc_dapm_device_event - send a device event to the dapm core + * snd_soc_dapm_set_bias_level - set the bias level for the system * @socdev: audio device - * @event: device event + * @level: level to configure * - * Sends a device event to the dapm core. The core then makes any - * necessary machine or codec power changes.. + * Configure the bias (power) levels for the SoC audio device. * * Returns 0 for success else error. */ -int snd_soc_dapm_device_event(struct snd_soc_device *socdev, int event) +int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev, + enum snd_soc_bias_level level) { struct snd_soc_codec *codec = socdev->codec; struct snd_soc_machine *machine = socdev->machine; + int ret = 0; - if (machine->dapm_event) - machine->dapm_event(machine, event); - if (codec->dapm_event) - codec->dapm_event(codec, event); - return 0; + if (machine->set_bias_level) + ret = machine->set_bias_level(machine, level); + if (ret == 0 && codec->set_bias_level) + ret = codec->set_bias_level(codec, level); + + return ret; } -EXPORT_SYMBOL_GPL(snd_soc_dapm_device_event); /** * snd_soc_dapm_set_endpoint - set audio endpoint status -- cgit From 3ff3f64ba04b3e5a86dce5239b10268798f68ad7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 19 May 2008 12:32:25 +0200 Subject: [ALSA] ASoC: core checkpatch cleanups Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/soc-core.c | 100 +++++++++++++++++++++++++++------------------------ 1 file changed, 54 insertions(+), 46 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a05b3450aee..a3f091e0843 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -108,9 +108,9 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec) } #endif -static inline const char* get_dai_name(int type) +static inline const char *get_dai_name(int type) { - switch(type) { + switch (type) { case SND_SOC_DAI_AC97_BUS: case SND_SOC_DAI_AC97: return "AC97"; @@ -178,9 +178,11 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) /* Check that the codec and cpu DAI's are compatible */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { runtime->hw.rate_min = - max(codec_dai->playback.rate_min, cpu_dai->playback.rate_min); + max(codec_dai->playback.rate_min, + cpu_dai->playback.rate_min); runtime->hw.rate_max = - min(codec_dai->playback.rate_max, cpu_dai->playback.rate_max); + min(codec_dai->playback.rate_max, + cpu_dai->playback.rate_max); runtime->hw.channels_min = max(codec_dai->playback.channels_min, cpu_dai->playback.channels_min); @@ -193,9 +195,11 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) codec_dai->playback.rates & cpu_dai->playback.rates; } else { runtime->hw.rate_min = - max(codec_dai->capture.rate_min, cpu_dai->capture.rate_min); + max(codec_dai->capture.rate_min, + cpu_dai->capture.rate_min); runtime->hw.rate_max = - min(codec_dai->capture.rate_max, cpu_dai->capture.rate_max); + min(codec_dai->capture.rate_max, + cpu_dai->capture.rate_max); runtime->hw.channels_min = max(codec_dai->capture.channels_min, cpu_dai->capture.channels_min); @@ -225,7 +229,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) goto machine_err; } - dbg("asoc: %s <-> %s info:\n",codec_dai->name, cpu_dai->name); + dbg("asoc: %s <-> %s info:\n", codec_dai->name, cpu_dai->name); dbg("asoc: rate mask 0x%x\n", runtime->hw.rates); dbg("asoc: min ch %d max ch %d\n", runtime->hw.channels_min, runtime->hw.channels_max); @@ -272,7 +276,7 @@ static void close_delayed_work(struct work_struct *work) int i; mutex_lock(&pcm_mutex); - for(i = 0; i < codec->num_dai; i++) { + for (i = 0; i < codec->num_dai; i++) { codec_dai = &codec->dai[i]; dbg("pop wq checking: %s status: %s waiting: %s\n", @@ -511,7 +515,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, if (cpu_dai->ops.hw_params) { ret = cpu_dai->ops.hw_params(substream, params); if (ret < 0) { - printk(KERN_ERR "asoc: can't set interface %s hw params\n", + printk(KERN_ERR "asoc: interface %s hw params failed\n", cpu_dai->name); goto interface_err; } @@ -520,7 +524,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, if (platform->pcm_ops->hw_params) { ret = platform->pcm_ops->hw_params(substream, params); if (ret < 0) { - printk(KERN_ERR "asoc: can't set platform %s hw params\n", + printk(KERN_ERR "asoc: platform %s hw params failed\n", platform->name); goto platform_err; } @@ -539,7 +543,7 @@ interface_err: codec_dai->ops.hw_free(substream); codec_err: - if(machine->ops && machine->ops->hw_free) + if (machine->ops && machine->ops->hw_free) machine->ops->hw_free(substream); mutex_unlock(&pcm_mutex); @@ -628,15 +632,15 @@ static struct snd_pcm_ops soc_pcm_ops = { /* powers down audio subsystem for suspend */ static int soc_suspend(struct platform_device *pdev, pm_message_t state) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_machine *machine = socdev->machine; - struct snd_soc_platform *platform = socdev->platform; - struct snd_soc_codec_device *codec_dev = socdev->codec_dev; + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_machine *machine = socdev->machine; + struct snd_soc_platform *platform = socdev->platform; + struct snd_soc_codec_device *codec_dev = socdev->codec_dev; struct snd_soc_codec *codec = socdev->codec; int i; /* mute any active DAC's */ - for(i = 0; i < machine->num_links; i++) { + for (i = 0; i < machine->num_links; i++) { struct snd_soc_codec_dai *dai = machine->dai_link[i].codec_dai; if (dai->dai_ops.digital_mute && dai->playback.active) dai->dai_ops.digital_mute(dai, 1); @@ -649,7 +653,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) if (machine->suspend_pre) machine->suspend_pre(pdev, state); - for(i = 0; i < machine->num_links; i++) { + for (i = 0; i < machine->num_links; i++) { struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->suspend && cpu_dai->type != SND_SOC_DAI_AC97) cpu_dai->suspend(pdev, cpu_dai); @@ -661,7 +665,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) run_delayed_work(&socdev->delayed_work); codec->suspend_bias_level = codec->bias_level; - for(i = 0; i < codec->num_dai; i++) { + for (i = 0; i < codec->num_dai; i++) { char *stream = codec->dai[i].playback.stream_name; if (stream != NULL) snd_soc_dapm_stream_event(codec, stream, @@ -675,7 +679,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) if (codec_dev->suspend) codec_dev->suspend(pdev, state); - for(i = 0; i < machine->num_links; i++) { + for (i = 0; i < machine->num_links; i++) { struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->suspend && cpu_dai->type == SND_SOC_DAI_AC97) cpu_dai->suspend(pdev, cpu_dai); @@ -690,17 +694,17 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) /* powers up audio subsystem after a suspend */ static int soc_resume(struct platform_device *pdev) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_machine *machine = socdev->machine; - struct snd_soc_platform *platform = socdev->platform; - struct snd_soc_codec_device *codec_dev = socdev->codec_dev; + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_machine *machine = socdev->machine; + struct snd_soc_platform *platform = socdev->platform; + struct snd_soc_codec_device *codec_dev = socdev->codec_dev; struct snd_soc_codec *codec = socdev->codec; int i; if (machine->resume_pre) machine->resume_pre(pdev); - for(i = 0; i < machine->num_links; i++) { + for (i = 0; i < machine->num_links; i++) { struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->resume && cpu_dai->type == SND_SOC_DAI_AC97) cpu_dai->resume(pdev, cpu_dai); @@ -709,8 +713,8 @@ static int soc_resume(struct platform_device *pdev) if (codec_dev->resume) codec_dev->resume(pdev); - for(i = 0; i < codec->num_dai; i++) { - char* stream = codec->dai[i].playback.stream_name; + for (i = 0; i < codec->num_dai; i++) { + char *stream = codec->dai[i].playback.stream_name; if (stream != NULL) snd_soc_dapm_stream_event(codec, stream, SND_SOC_DAPM_STREAM_RESUME); @@ -720,14 +724,14 @@ static int soc_resume(struct platform_device *pdev) SND_SOC_DAPM_STREAM_RESUME); } - /* unmute any active DAC's */ - for(i = 0; i < machine->num_links; i++) { + /* unmute any active DACs */ + for (i = 0; i < machine->num_links; i++) { struct snd_soc_codec_dai *dai = machine->dai_link[i].codec_dai; if (dai->dai_ops.digital_mute && dai->playback.active) dai->dai_ops.digital_mute(dai, 0); } - for(i = 0; i < machine->num_links; i++) { + for (i = 0; i < machine->num_links; i++) { struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->resume && cpu_dai->type != SND_SOC_DAI_AC97) cpu_dai->resume(pdev, cpu_dai); @@ -757,7 +761,7 @@ static int soc_probe(struct platform_device *pdev) if (machine->probe) { ret = machine->probe(pdev); - if(ret < 0) + if (ret < 0) return ret; } @@ -765,20 +769,20 @@ static int soc_probe(struct platform_device *pdev) struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->probe) { ret = cpu_dai->probe(pdev); - if(ret < 0) + if (ret < 0) goto cpu_dai_err; } } if (codec_dev->probe) { ret = codec_dev->probe(pdev); - if(ret < 0) + if (ret < 0) goto cpu_dai_err; } if (platform->probe) { ret = platform->probe(pdev); - if(ret < 0) + if (ret < 0) goto platform_err; } @@ -865,7 +869,7 @@ static int soc_new_pcm(struct snd_soc_device *socdev, codec_dai->codec = socdev->codec; /* check client and interface hw capabilities */ - sprintf(new_name, "%s %s-%s-%d",dai_link->stream_name, codec_dai->name, + sprintf(new_name, "%s %s-%s-%d", dai_link->stream_name, codec_dai->name, get_dai_name(cpu_dai->type), num); if (codec_dai->playback.channels_min) @@ -876,7 +880,8 @@ static int soc_new_pcm(struct snd_soc_device *socdev, ret = snd_pcm_new(codec->card, new_name, codec->pcm_devs++, playback, capture, &pcm); if (ret < 0) { - printk(KERN_ERR "asoc: can't create pcm for codec %s\n", codec->name); + printk(KERN_ERR "asoc: can't create pcm for codec %s\n", + codec->name); kfree(rtd); return ret; } @@ -925,8 +930,9 @@ static ssize_t codec_reg_show(struct device *dev, step = codec->reg_cache_step; count += sprintf(buf, "%s registers\n", codec->name); - for(i = 0; i < codec->reg_cache_size; i += step) - count += sprintf(buf + count, "%2x: %4x\n", i, codec->read(codec, i)); + for (i = 0; i < codec->reg_cache_size; i += step) + count += sprintf(buf + count, "%2x: %4x\n", i, + codec->read(codec, i)); return count; } @@ -1069,7 +1075,7 @@ int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid) strncpy(codec->card->driver, codec->name, sizeof(codec->card->driver)); /* create the pcms */ - for(i = 0; i < machine->num_links; i++) { + for (i = 0; i < machine->num_links; i++) { ret = soc_new_pcm(socdev, &machine->dai_link[i], i); if (ret < 0) { printk(KERN_ERR "asoc: can't create pcm %s\n", @@ -1099,7 +1105,7 @@ int snd_soc_register_card(struct snd_soc_device *socdev) struct snd_soc_machine *machine = socdev->machine; int ret = 0, i, ac97 = 0, err = 0; - for(i = 0; i < machine->num_links; i++) { + for (i = 0; i < machine->num_links; i++) { if (socdev->machine->dai_link[i].init) { err = socdev->machine->dai_link[i].init(codec); if (err < 0) { @@ -1108,7 +1114,7 @@ int snd_soc_register_card(struct snd_soc_device *socdev) continue; } } - if (socdev->machine->dai_link[i].codec_dai->type == + if (socdev->machine->dai_link[i].codec_dai->type == SND_SOC_DAI_AC97_BUS) ac97 = 1; } @@ -1119,7 +1125,7 @@ int snd_soc_register_card(struct snd_soc_device *socdev) ret = snd_card_register(codec->card); if (ret < 0) { - printk(KERN_ERR "asoc: failed to register soundcard for codec %s\n", + printk(KERN_ERR "asoc: failed to register soundcard for %s\n", codec->name); goto out; } @@ -1143,7 +1149,7 @@ int snd_soc_register_card(struct snd_soc_device *socdev) err = device_create_file(socdev->dev, &dev_attr_codec_reg); if (err < 0) - printk(KERN_WARNING "asoc: failed to add codec sysfs entries\n"); + printk(KERN_WARNING "asoc: failed to add codec sysfs files\n"); mutex_unlock(&codec->mutex); @@ -1169,7 +1175,7 @@ void snd_soc_free_pcms(struct snd_soc_device *socdev) mutex_lock(&codec->mutex); #ifdef CONFIG_SND_SOC_AC97_BUS - for(i = 0; i < codec->num_dai; i++) { + for (i = 0; i < codec->num_dai; i++) { codec_dai = &codec->dai[i]; if (codec_dai->type == SND_SOC_DAI_AC97_BUS && codec->ac97) { soc_ac97_dev_unregister(codec); @@ -1279,7 +1285,8 @@ int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, for (bitmask = 1; bitmask < e->mask; bitmask <<= 1) ; val = snd_soc_read(codec, e->reg); - ucontrol->value.enumerated.item[0] = (val >> e->shift_l) & (bitmask - 1); + ucontrol->value.enumerated.item[0] + = (val >> e->shift_l) & (bitmask - 1); if (e->shift_l != e->shift_r) ucontrol->value.enumerated.item[1] = (val >> e->shift_r) & (bitmask - 1); @@ -1573,7 +1580,8 @@ int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol, val = val << shift; val2 = val2 << shift; - if ((err = snd_soc_update_bits(codec, reg, val_mask, val)) < 0) + err = snd_soc_update_bits(codec, reg, val_mask, val); + if (err < 0) return err; err = snd_soc_update_bits(codec, reg2, val_mask, val2); @@ -1589,7 +1597,7 @@ static int __devinit snd_soc_init(void) static void snd_soc_exit(void) { - platform_driver_unregister(&soc_driver); + platform_driver_unregister(&soc_driver); } module_init(snd_soc_init); -- cgit From a65f0568f6cc8433877fb71dd7d36b551854b0bc Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 13 May 2008 14:54:43 +0200 Subject: [ALSA] soc - Convert Wolfson codec drivers to use bulk DAPM registration Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/codecs/wm8731.c | 16 ++++------------ sound/soc/codecs/wm8750.c | 16 ++++------------ sound/soc/codecs/wm8753.c | 17 ++++------------- sound/soc/codecs/wm9712.c | 15 ++++----------- sound/soc/codecs/wm9713.c | 15 ++++----------- 5 files changed, 20 insertions(+), 59 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 0f28aa4bccc..5acf43ab104 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -193,7 +193,7 @@ SND_SOC_DAPM_INPUT("RLINEIN"), SND_SOC_DAPM_INPUT("LLINEIN"), }; -static const char *intercon[][3] = { +static const struct snd_soc_dapm_route intercon[] = { /* output mixer */ {"Output Mixer", "Line Bypass Switch", "Line Input"}, {"Output Mixer", "HiFi Playback Switch", "DAC"}, @@ -214,22 +214,14 @@ static const char *intercon[][3] = { {"Line Input", NULL, "LLINEIN"}, {"Line Input", NULL, "RLINEIN"}, {"Mic Bias", NULL, "MICIN"}, - - /* terminator */ - {NULL, NULL, NULL}, }; static int wm8731_add_widgets(struct snd_soc_codec *codec) { - int i; - - for (i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++) - snd_soc_dapm_new_control(codec, &wm8731_dapm_widgets[i]); + snd_soc_dapm_new_controls(codec, wm8731_dapm_widgets, + ARRAY_SIZE(wm8731_dapm_widgets)); - /* set up audio path interconnects */ - for (i = 0; intercon[i][0] != NULL; i++) - snd_soc_dapm_connect_input(codec, intercon[i][0], - intercon[i][1], intercon[i][2]); + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); snd_soc_dapm_new_widgets(codec); return 0; diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 62423f4493b..1f11ad24551 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -378,7 +378,7 @@ static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = { SND_SOC_DAPM_INPUT("RINPUT3"), }; -static const char *audio_map[][3] = { +static const struct snd_soc_dapm_route audio_map[] = { /* left mixer */ {"Left Mixer", "Playback Switch", "Left DAC"}, {"Left Mixer", "Left Bypass Switch", "Left Line Mux"}, @@ -470,22 +470,14 @@ static const char *audio_map[][3] = { /* ADC */ {"Left ADC", NULL, "Left ADC Mux"}, {"Right ADC", NULL, "Right ADC Mux"}, - - /* terminator */ - {NULL, NULL, NULL}, }; static int wm8750_add_widgets(struct snd_soc_codec *codec) { - int i; - - for (i = 0; i < ARRAY_SIZE(wm8750_dapm_widgets); i++) - snd_soc_dapm_new_control(codec, &wm8750_dapm_widgets[i]); + snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets, + ARRAY_SIZE(wm8750_dapm_widgets)); - /* set up audio path audio_mapnects */ - for (i = 0; audio_map[i][0] != NULL; i++) - snd_soc_dapm_connect_input(codec, audio_map[i][0], - audio_map[i][1], audio_map[i][2]); + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); snd_soc_dapm_new_widgets(codec); return 0; diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 9032b0c07c8..c32e6326be6 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -523,7 +523,7 @@ SND_SOC_DAPM_INPUT("MIC2"), SND_SOC_DAPM_VMID("VREF"), }; -static const char *audio_map[][3] = { +static const struct snd_soc_dapm_route audio_map[] = { /* left mixer */ {"Left Mixer", "Left Playback Switch", "Left DAC"}, {"Left Mixer", "Voice Playback Switch", "Voice DAC"}, @@ -674,23 +674,14 @@ static const char *audio_map[][3] = { /* ACOP */ {"ACOP", NULL, "ALC Mixer"}, - - /* terminator */ - {NULL, NULL, NULL}, }; static int wm8753_add_widgets(struct snd_soc_codec *codec) { - int i; + snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets, + ARRAY_SIZE(wm8753_dapm_widgets)); - for (i = 0; i < ARRAY_SIZE(wm8753_dapm_widgets); i++) - snd_soc_dapm_new_control(codec, &wm8753_dapm_widgets[i]); - - /* set up the WM8753 audio map */ - for (i = 0; audio_map[i][0] != NULL; i++) { - snd_soc_dapm_connect_input(codec, audio_map[i][0], - audio_map[i][1], audio_map[i][2]); - } + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); snd_soc_dapm_new_widgets(codec); return 0; diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index e26cfcf0b4f..d9789f1c890 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -348,7 +348,7 @@ SND_SOC_DAPM_INPUT("MIC1"), SND_SOC_DAPM_INPUT("MIC2"), }; -static const char *audio_map[][3] = { +static const struct snd_soc_dapm_route audio_map[] = { /* virtual mixer - mixes left & right channels for spk and mono */ {"AC97 Mixer", NULL, "Left DAC"}, {"AC97 Mixer", NULL, "Right DAC"}, @@ -443,21 +443,14 @@ static const char *audio_map[][3] = { {"Speaker PGA", NULL, "Speaker Mux"}, {"LOUT2", NULL, "Speaker PGA"}, {"ROUT2", NULL, "Speaker PGA"}, - - {NULL, NULL, NULL}, }; static int wm9712_add_widgets(struct snd_soc_codec *codec) { - int i; - - for (i = 0; i < ARRAY_SIZE(wm9712_dapm_widgets); i++) - snd_soc_dapm_new_control(codec, &wm9712_dapm_widgets[i]); + snd_soc_dapm_new_controls(codec, wm9712_dapm_widgets, + ARRAY_SIZE(wm9712_dapm_widgets)); - /* set up audio path connects */ - for (i = 0; audio_map[i][0] != NULL; i++) - snd_soc_dapm_connect_input(codec, audio_map[i][0], - audio_map[i][1], audio_map[i][2]); + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); snd_soc_dapm_new_widgets(codec); return 0; diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 4863636e9d5..4f516a5a561 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -453,7 +453,7 @@ SND_SOC_DAPM_INPUT("MIC2B"), SND_SOC_DAPM_VMID("VMID"), }; -static const char *audio_map[][3] = { +static const struct snd_soc_dapm_route audio_map[] = { /* left HP mixer */ {"Left HP Mixer", "PC Beep Playback Switch", "PCBEEP"}, {"Left HP Mixer", "Voice Playback Switch", "Voice DAC"}, @@ -604,21 +604,14 @@ static const char *audio_map[][3] = { {"Capture Mono Mux", "Stereo", "Capture Mixer"}, {"Capture Mono Mux", "Left", "Left Capture Source"}, {"Capture Mono Mux", "Right", "Right Capture Source"}, - - {NULL, NULL, NULL}, }; static int wm9713_add_widgets(struct snd_soc_codec *codec) { - int i; - - for (i = 0; i < ARRAY_SIZE(wm9713_dapm_widgets); i++) - snd_soc_dapm_new_control(codec, &wm9713_dapm_widgets[i]); + snd_soc_dapm_new_controls(codec, wm9713_dapm_widgets, + ARRAY_SIZE(wm9713_dapm_widgets)); - /* set up audio path audio_mapnects */ - for (i = 0; audio_map[i][0] != NULL; i++) - snd_soc_dapm_connect_input(codec, audio_map[i][0], - audio_map[i][1], audio_map[i][2]); + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); snd_soc_dapm_new_widgets(codec); return 0; -- cgit From 33e5b22285f63ede858c00456f3ffbc2ea79d6cf Mon Sep 17 00:00:00 2001 From: Werner Almesberger Date: Mon, 14 Apr 2008 14:26:44 +0200 Subject: [ALSA] soc - Fix s3c24xx-i2s LR sync while timer ticks are disabled When timer ticks are disabled when calling sound/soc/s3c24xx/s3c24xx-i2s.c:s3c24xx_snd_lrsync and the LR signal never happens, we loop forever. This has been observed in the following call chain: snd_pcm_common_ioctl1 -> snd_pcm_action_lock_irq -> snd_pcm_action_single -> snd_pcm_do_resume -> soc_pcm_trigger -> s3c24xx_i2s_trigger The patch below changes the timeout mechanism to use udelay, which doesn't need timer ticks. Signed-off-by: Werner Almesberger Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/s3c24xx/s3c24xx-i2s.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index ddf87246c77..4c52f7946d9 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -175,7 +175,7 @@ static void s3c24xx_snd_rxctrl(int on) static int s3c24xx_snd_lrsync(void) { u32 iiscon; - unsigned long timeout = jiffies + msecs_to_jiffies(5); + int timeout = 50; /* 5ms */ DBG("Entered %s\n", __func__); @@ -184,8 +184,9 @@ static int s3c24xx_snd_lrsync(void) if (iiscon & S3C2410_IISCON_LRINDEX) break; - if (time_after(jiffies, timeout)) + if (!timeout--) return -ETIMEDOUT; + udelay(100); } return 0; -- cgit From 89fe5117928b2c1272c9376362131ded561c91ad Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 23 May 2008 16:10:37 +0200 Subject: sound: Convert to menuconfig Convert menu in sound Kconfig files to menuconfig and if. Signed-off-by: Takashi Iwai --- sound/soc/Kconfig | 17 ++++++++--------- sound/soc/at91/Kconfig | 2 +- sound/soc/codecs/Kconfig | 10 ++-------- sound/soc/davinci/Kconfig | 2 +- sound/soc/fsl/Kconfig | 2 +- sound/soc/pxa/Kconfig | 2 +- sound/soc/s3c24xx/Kconfig | 4 +--- sound/soc/sh/Kconfig | 5 ++--- 8 files changed, 17 insertions(+), 27 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 18f28ac4bfe..fd7bc4f8907 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -2,15 +2,8 @@ # SoC audio configuration # -menu "System on Chip audio support" - depends on SND!=n - -config SND_SOC_AC97_BUS - bool - -config SND_SOC +menuconfig SND_SOC tristate "ALSA for SoC audio support" - depends on SND select SND_PCM ---help--- @@ -23,6 +16,11 @@ config SND_SOC This ASoC audio support can also be built as a module. If so, the module will be called snd-soc-core. +if SND_SOC + +config SND_SOC_AC97_BUS + bool + # All the supported Soc's source "sound/soc/at91/Kconfig" source "sound/soc/pxa/Kconfig" @@ -35,4 +33,5 @@ source "sound/soc/omap/Kconfig" # Supported codecs source "sound/soc/codecs/Kconfig" -endmenu +endif # SND_SOC + diff --git a/sound/soc/at91/Kconfig b/sound/soc/at91/Kconfig index 5cb93fd3a40..905186502e0 100644 --- a/sound/soc/at91/Kconfig +++ b/sound/soc/at91/Kconfig @@ -1,6 +1,6 @@ config SND_AT91_SOC tristate "SoC Audio for the Atmel AT91 System-on-Chip" - depends on ARCH_AT91 && SND_SOC + depends on ARCH_AT91 help Say Y or M if you want to add support for codecs attached to the AT91 SSC interface. You will also need diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3903ab7dfa4..d4a5fe42f6e 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1,31 +1,25 @@ config SND_SOC_AC97_CODEC tristate - depends on SND_SOC + select SND_AC97_CODEC config SND_SOC_WM8731 tristate - depends on SND_SOC config SND_SOC_WM8750 tristate - depends on SND_SOC config SND_SOC_WM8753 tristate - depends on SND_SOC config SND_SOC_WM9712 tristate - depends on SND_SOC config SND_SOC_WM9713 tristate - depends on SND_SOC # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate - depends on SND_SOC # Cirrus Logic CS4270 Codec Hardware Mute Support # Select if you have external muting circuitry attached to your CS4270. @@ -43,4 +37,4 @@ config SND_SOC_CS4270_VD33_ERRATA config SND_SOC_TLV320AIC3X tristate - depends on SND_SOC && I2C + depends on I2C diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 20680c551aa..8f7e3383490 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -1,6 +1,6 @@ config SND_DAVINCI_SOC tristate "SoC Audio for the TI DAVINCI chip" - depends on ARCH_DAVINCI && SND_SOC + depends on ARCH_DAVINCI help Say Y or M if you want to add support for codecs attached to the DAVINCI AC97 or I2S interface. You will also need diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 257101f44e9..19802e27df4 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -2,7 +2,7 @@ menu "ALSA SoC audio for Freescale SOCs" config SND_SOC_MPC8610 bool "ALSA SoC support for the MPC8610 SOC" - depends on SND_SOC && MPC8610_HPCD + depends on MPC8610_HPCD default y if MPC8610 help Say Y if you want to add support for codecs attached to the SSI diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 484f883459e..18a40dc8ad9 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -1,6 +1,6 @@ config SND_PXA2XX_SOC tristate "SoC Audio for the Intel PXA2xx chip" - depends on ARCH_PXA && SND_SOC + depends on ARCH_PXA help Say Y or M if you want to add support for codecs attached to the PXA2xx AC97, I2S or SSP interface. You will also need diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 1f6dbfc4caa..b9f2353effe 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -1,7 +1,6 @@ config SND_S3C24XX_SOC tristate "SoC Audio for the Samsung S3C24XX chips" - depends on ARCH_S3C2410 && SND_SOC - select SND_PCM + depends on ARCH_S3C2410 help Say Y or M if you want to add support for codecs attached to the S3C24XX AC97, I2S or SSP interface. You will also need @@ -16,7 +15,6 @@ config SND_S3C2412_SOC_I2S config SND_S3C2443_SOC_AC97 tristate select AC97_BUS - select SND_AC97_CODEC select SND_SOC_AC97_BUS config SND_S3C24XX_SOC_NEO1973_WM8753 diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 4c1e013381c..54bd604012a 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -3,7 +3,7 @@ menu "SoC Audio support for SuperH" config SND_SOC_PCM_SH7760 tristate "SoC Audio support for Renesas SH7760" - depends on CPU_SUBTYPE_SH7760 && SND_SOC && SH_DMABRG + depends on CPU_SUBTYPE_SH7760 && SH_DMABRG help Enable this option for SH7760 AC97/I2S audio support. @@ -13,10 +13,9 @@ config SND_SOC_PCM_SH7760 ## config SND_SOC_SH4_HAC + tristate select AC97_BUS select SND_SOC_AC97_BUS - select SND_AC97_CODEC - tristate config SND_SOC_SH4_SSI tristate -- cgit From 142054a389ebf7972b4eee822ad7c55ff852b649 Mon Sep 17 00:00:00 2001 From: Mike Rapoport Date: Mon, 26 May 2008 12:59:16 +0300 Subject: [ALSA] Add EM-X270 ASoC driver This patch adds ASoC support for EM-X270 machine. Signed-off-by: Mike Rapoport Signed-off-by: Takashi Iwai --- sound/soc/pxa/Kconfig | 9 +++++ sound/soc/pxa/Makefile | 3 +- sound/soc/pxa/em-x270.c | 102 ++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 113 insertions(+), 1 deletion(-) create mode 100644 sound/soc/pxa/em-x270.c (limited to 'sound/soc') diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 18a40dc8ad9..12f6ac99b04 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -62,3 +62,12 @@ config SND_PXA2XX_SOC_E800 help Say Y if you want to add support for SoC audio on the Toshiba e800 PDA + +config SND_PXA2XX_SOC_EM_X270 + tristate "SoC Audio support for CompuLab EM-x270" + depends on SND_PXA2XX_SOC && MACH_EM_X270 + select SND_PXA2XX_SOC_AC97 + select SND_SOC_WM9712 + help + Say Y if you want to add support for SoC audio on + CompuLab EM-x270. diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 04e5646f75b..5bc8edf9dca 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -13,10 +13,11 @@ snd-soc-poodle-objs := poodle.o snd-soc-tosa-objs := tosa.o snd-soc-e800-objs := e800_wm9712.o snd-soc-spitz-objs := spitz.o +snd-soc-em-x270-objs := em-x270.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_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/em-x270.c b/sound/soc/pxa/em-x270.c new file mode 100644 index 00000000000..02dcac39cdf --- /dev/null +++ b/sound/soc/pxa/em-x270.c @@ -0,0 +1,102 @@ +/* + * em-x270.c -- SoC audio for EM-X270 + * + * Copyright 2007 CompuLab, Ltd. + * + * Author: Mike Rapoport + * + * Copied from tosa.c: + * Copyright 2005 Wolfson Microelectronics PLC. + * Copyright 2005 Openedhand Ltd. + * + * Authors: Liam Girdwood + * Richard Purdie + * + * 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 +#include +#include + +#include +#include +#include +#include +#include + +#include +#include +#include +#include + +#include "../codecs/wm9712.h" +#include "pxa2xx-pcm.h" +#include "pxa2xx-ac97.h" + +static struct snd_soc_dai_link em_x270_dai[] = { + { + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI], + .codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI], + }, + { + .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_machine em_x270 = { + .name = "EM-X270", + .dai_link = em_x270_dai, + .num_links = ARRAY_SIZE(em_x270_dai), +}; + +static struct snd_soc_device em_x270_snd_devdata = { + .machine = &em_x270, + .platform = &pxa2xx_soc_platform, + .codec_dev = &soc_codec_dev_wm9712, +}; + +static struct platform_device *em_x270_snd_device; + +static int __init em_x270_init(void) +{ + int ret; + + if (!machine_is_em_x270()) + return -ENODEV; + + em_x270_snd_device = platform_device_alloc("soc-audio", -1); + if (!em_x270_snd_device) + return -ENOMEM; + + platform_set_drvdata(em_x270_snd_device, &em_x270_snd_devdata); + em_x270_snd_devdata.dev = &em_x270_snd_device->dev; + ret = platform_device_add(em_x270_snd_device); + + if (ret) + platform_device_put(em_x270_snd_device); + + return ret; +} + +static void __exit em_x270_exit(void) +{ + platform_device_unregister(em_x270_snd_device); +} + +module_init(em_x270_init); +module_exit(em_x270_exit); + +/* Module information */ +MODULE_AUTHOR("Mike Rapoport"); +MODULE_DESCRIPTION("ALSA SoC EM-X270"); +MODULE_LICENSE("GPL"); -- cgit From e13ac2e9b18bde51cf32c69c2209df25791ab3e5 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 28 May 2008 17:58:05 +0100 Subject: [ALSA] ASoC: Add SOC_DOUBLE_S8_TLV control type The SOC_DOUBLE_S8_TLV control type was originally implemented in the UDA1380 driver by Philipp Zabel and was moved into the core by me. Signed-off-by: Philipp Zabel Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/soc-core.c | 72 ++++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 72 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a3f091e0843..f594ab888e1 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1589,6 +1589,78 @@ int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_GPL(snd_soc_put_volsw_2r); +/** + * snd_soc_info_volsw_s8 - signed mixer info callback + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to provide information about a signed mixer control. + * + * Returns 0 for success. + */ +int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int max = (signed char)((kcontrol->private_value >> 16) & 0xff); + int min = (signed char)((kcontrol->private_value >> 24) & 0xff); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = max-min; + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_info_volsw_s8); + +/** + * snd_soc_get_volsw_s8 - signed mixer get callback + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to get the value of a signed mixer control. + * + * Returns 0 for success. + */ +int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int min = (signed char)((kcontrol->private_value >> 24) & 0xff); + int val = snd_soc_read(codec, reg); + + ucontrol->value.integer.value[0] = + ((signed char)(val & 0xff))-min; + ucontrol->value.integer.value[1] = + ((signed char)((val >> 8) & 0xff))-min; + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_get_volsw_s8); + +/** + * snd_soc_put_volsw_sgn - signed mixer put callback + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to set the value of a signed mixer control. + * + * Returns 0 for success. + */ +int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int min = (signed char)((kcontrol->private_value >> 24) & 0xff); + unsigned short val; + + val = (ucontrol->value.integer.value[0]+min) & 0xff; + val |= ((ucontrol->value.integer.value[1]+min) & 0xff) << 8; + + return snd_soc_update_bits(codec, reg, 0xffff, val); +} +EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8); + static int __devinit snd_soc_init(void) { printk(KERN_INFO "ASoC version %s\n", SND_SOC_VERSION); -- cgit From b7482f52789266e2548be5d0f6420c9fc12428d8 Mon Sep 17 00:00:00 2001 From: Philipp Zabel Date: Wed, 28 May 2008 17:58:06 +0100 Subject: [ALSA] ASoC: Add UDA1380 driver The UDA1380 codec is used by the HTC Magician and a number of Samsung reference boards. This driver has had a long out of tree history, having originally been written by Giorgio Padrin and converted to ASoC by Richard Purdie. Since conversion to ASoC most of the work on the driver has been done by Philipp Zabel with some review and updates for new APIs by Liam Girdwood and Mark Brown. Signed-off-by: Richard Purdie Signed-off-by: Philipp Zabel Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/uda1380.c | 852 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/uda1380.h | 89 +++++ 4 files changed, 946 insertions(+) create mode 100644 sound/soc/codecs/uda1380.c create mode 100644 sound/soc/codecs/uda1380.h (limited to 'sound/soc') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index d4a5fe42f6e..7beefccfa82 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -2,6 +2,9 @@ config SND_SOC_AC97_CODEC tristate select SND_AC97_CODEC +config SND_SOC_UDA1380 + tristate + config SND_SOC_WM8731 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 4e1314c9d3e..d5926a11707 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -1,4 +1,5 @@ snd-soc-ac97-objs := ac97.o +snd-soc-uda1380-objs := uda1380.o snd-soc-wm8731-objs := wm8731.o snd-soc-wm8750-objs := wm8750.o snd-soc-wm8753-objs := wm8753.o @@ -8,6 +9,7 @@ snd-soc-cs4270-objs := cs4270.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o +obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c new file mode 100644 index 00000000000..cb50486201f --- /dev/null +++ b/sound/soc/codecs/uda1380.c @@ -0,0 +1,852 @@ +/* + * uda1380.c - Philips UDA1380 ALSA SoC audio driver + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Copyright (c) 2007 Philipp Zabel + * Improved support for DAPM and audio routing/mixing capabilities, + * added TLV support. + * + * Modified by Richard Purdie to fit into SoC + * codec model. + * + * Copyright (c) 2005 Giorgio Padrin + * Copyright 2005 Openedhand Ltd. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "uda1380.h" + +#define UDA1380_VERSION "0.6" +#define AUDIO_NAME "uda1380" + +/* + * uda1380 register cache + */ +static const u16 uda1380_reg[UDA1380_CACHEREGNUM] = { + 0x0502, 0x0000, 0x0000, 0x3f3f, + 0x0202, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0xff00, 0x0000, 0x4800, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x8000, 0x0002, 0x0000, +}; + +/* + * read uda1380 register cache + */ +static inline unsigned int uda1380_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg == UDA1380_RESET) + return 0; + if (reg >= UDA1380_CACHEREGNUM) + return -1; + return cache[reg]; +} + +/* + * write uda1380 register cache + */ +static inline void uda1380_write_reg_cache(struct snd_soc_codec *codec, + u16 reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + if (reg >= UDA1380_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * write to the UDA1380 register space + */ +static int uda1380_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[3]; + + /* data is + * data[0] is register offset + * data[1] is MS byte + * data[2] is LS byte + */ + data[0] = reg; + data[1] = (value & 0xff00) >> 8; + data[2] = value & 0x00ff; + + uda1380_write_reg_cache(codec, reg, value); + + /* the interpolator & decimator regs must only be written when the + * codec DAI is active. + */ + if (!codec->active && (reg >= UDA1380_MVOL)) + return 0; + pr_debug("uda1380: hw write %x val %x\n", reg, value); + if (codec->hw_write(codec->control_data, data, 3) == 3) { + unsigned int val; + i2c_master_send(codec->control_data, data, 1); + i2c_master_recv(codec->control_data, data, 2); + val = (data[0]<<8) | data[1]; + if (val != value) { + pr_debug("uda1380: READ BACK VAL %x\n", + (data[0]<<8) | data[1]); + return -EIO; + } + return 0; + } else + return -EIO; +} + +#define uda1380_reset(c) uda1380_write(c, UDA1380_RESET, 0) + +/* declarations of ALSA reg_elem_REAL controls */ +static const char *uda1380_deemp[] = { + "None", + "32kHz", + "44.1kHz", + "48kHz", + "96kHz", +}; +static const char *uda1380_input_sel[] = { + "Line", + "Mic + Line R", + "Line L", + "Mic", +}; +static const char *uda1380_output_sel[] = { + "DAC", + "Analog Mixer", +}; +static const char *uda1380_spf_mode[] = { + "Flat", + "Minimum1", + "Minimum2", + "Maximum" +}; +static const char *uda1380_capture_sel[] = { + "ADC", + "Digital Mixer" +}; +static const char *uda1380_sel_ns[] = { + "3rd-order", + "5th-order" +}; +static const char *uda1380_mix_control[] = { + "off", + "PCM only", + "before sound processing", + "after sound processing" +}; +static const char *uda1380_sdet_setting[] = { + "3200", + "4800", + "9600", + "19200" +}; +static const char *uda1380_os_setting[] = { + "single-speed", + "double-speed (no mixing)", + "quad-speed (no mixing)" +}; + +static const struct soc_enum uda1380_deemp_enum[] = { + SOC_ENUM_SINGLE(UDA1380_DEEMP, 8, 5, uda1380_deemp), + SOC_ENUM_SINGLE(UDA1380_DEEMP, 0, 5, uda1380_deemp), +}; +static const struct soc_enum uda1380_input_sel_enum = + SOC_ENUM_SINGLE(UDA1380_ADC, 2, 4, uda1380_input_sel); /* SEL_MIC, SEL_LNA */ +static const struct soc_enum uda1380_output_sel_enum = + SOC_ENUM_SINGLE(UDA1380_PM, 7, 2, uda1380_output_sel); /* R02_EN_AVC */ +static const struct soc_enum uda1380_spf_enum = + SOC_ENUM_SINGLE(UDA1380_MODE, 14, 4, uda1380_spf_mode); /* M */ +static const struct soc_enum uda1380_capture_sel_enum = + SOC_ENUM_SINGLE(UDA1380_IFACE, 6, 2, uda1380_capture_sel); /* SEL_SOURCE */ +static const struct soc_enum uda1380_sel_ns_enum = + SOC_ENUM_SINGLE(UDA1380_MIXER, 14, 2, uda1380_sel_ns); /* SEL_NS */ +static const struct soc_enum uda1380_mix_enum = + SOC_ENUM_SINGLE(UDA1380_MIXER, 12, 4, uda1380_mix_control); /* MIX, MIX_POS */ +static const struct soc_enum uda1380_sdet_enum = + SOC_ENUM_SINGLE(UDA1380_MIXER, 4, 4, uda1380_sdet_setting); /* SD_VALUE */ +static const struct soc_enum uda1380_os_enum = + SOC_ENUM_SINGLE(UDA1380_MIXER, 0, 3, uda1380_os_setting); /* OS */ + +/* + * from -48 dB in 1.5 dB steps (mute instead of -49.5 dB) + */ +static DECLARE_TLV_DB_SCALE(amix_tlv, -4950, 150, 1); + +/* + * from -78 dB in 1 dB steps (3 dB steps, really. LSB are ignored), + * from -66 dB in 0.5 dB steps (2 dB steps, really) and + * from -52 dB in 0.25 dB steps + */ +static const unsigned int mvol_tlv[] = { + TLV_DB_RANGE_HEAD(3), + 0, 15, TLV_DB_SCALE_ITEM(-8200, 100, 1), + 16, 43, TLV_DB_SCALE_ITEM(-6600, 50, 0), + 44, 252, TLV_DB_SCALE_ITEM(-5200, 25, 0), +}; + +/* + * from -72 dB in 1.5 dB steps (6 dB steps really), + * from -66 dB in 0.75 dB steps (3 dB steps really), + * from -60 dB in 0.5 dB steps (2 dB steps really) and + * from -46 dB in 0.25 dB steps + */ +static const unsigned int vc_tlv[] = { + TLV_DB_RANGE_HEAD(4), + 0, 7, TLV_DB_SCALE_ITEM(-7800, 150, 1), + 8, 15, TLV_DB_SCALE_ITEM(-6600, 75, 0), + 16, 43, TLV_DB_SCALE_ITEM(-6000, 50, 0), + 44, 228, TLV_DB_SCALE_ITEM(-4600, 25, 0), +}; + +/* from 0 to 6 dB in 2 dB steps if SPF mode != flat */ +static DECLARE_TLV_DB_SCALE(tr_tlv, 0, 200, 0); + +/* from 0 to 24 dB in 2 dB steps, if SPF mode == maximum, otherwise cuts + * off at 18 dB max) */ +static DECLARE_TLV_DB_SCALE(bb_tlv, 0, 200, 0); + +/* from -63 to 24 dB in 0.5 dB steps (-128...48) */ +static DECLARE_TLV_DB_SCALE(dec_tlv, -6400, 50, 1); + +/* from 0 to 24 dB in 3 dB steps */ +static DECLARE_TLV_DB_SCALE(pga_tlv, 0, 300, 0); + +/* from 0 to 30 dB in 2 dB steps */ +static DECLARE_TLV_DB_SCALE(vga_tlv, 0, 200, 0); + +static const struct snd_kcontrol_new uda1380_snd_controls[] = { + SOC_DOUBLE_TLV("Analog Mixer Volume", UDA1380_AMIX, 0, 8, 44, 1, amix_tlv), /* AVCR, AVCL */ + SOC_DOUBLE_TLV("Master Playback Volume", UDA1380_MVOL, 0, 8, 252, 1, mvol_tlv), /* MVCL, MVCR */ + SOC_SINGLE_TLV("ADC Playback Volume", UDA1380_MIXVOL, 8, 228, 1, vc_tlv), /* VC2 */ + SOC_SINGLE_TLV("PCM Playback Volume", UDA1380_MIXVOL, 0, 228, 1, vc_tlv), /* VC1 */ + SOC_ENUM("Sound Processing Filter", uda1380_spf_enum), /* M */ + SOC_DOUBLE_TLV("Tone Control - Treble", UDA1380_MODE, 4, 12, 3, 0, tr_tlv), /* TRL, TRR */ + SOC_DOUBLE_TLV("Tone Control - Bass", UDA1380_MODE, 0, 8, 15, 0, bb_tlv), /* BBL, BBR */ +/**/ SOC_SINGLE("Master Playback Switch", UDA1380_DEEMP, 14, 1, 1), /* MTM */ + SOC_SINGLE("ADC Playback Switch", UDA1380_DEEMP, 11, 1, 1), /* MT2 from decimation filter */ + SOC_ENUM("ADC Playback De-emphasis", uda1380_deemp_enum[0]), /* DE2 */ + SOC_SINGLE("PCM Playback Switch", UDA1380_DEEMP, 3, 1, 1), /* MT1, from digital data input */ + SOC_ENUM("PCM Playback De-emphasis", uda1380_deemp_enum[1]), /* DE1 */ + SOC_SINGLE("DAC Polarity inverting Switch", UDA1380_MIXER, 15, 1, 0), /* DA_POL_INV */ + SOC_ENUM("Noise Shaper", uda1380_sel_ns_enum), /* SEL_NS */ + SOC_ENUM("Digital Mixer Signal Control", uda1380_mix_enum), /* MIX_POS, MIX */ + SOC_SINGLE("Silence Switch", UDA1380_MIXER, 7, 1, 0), /* SILENCE, force DAC output to silence */ + SOC_SINGLE("Silence Detector Switch", UDA1380_MIXER, 6, 1, 0), /* SDET_ON */ + SOC_ENUM("Silence Detector Setting", uda1380_sdet_enum), /* SD_VALUE */ + SOC_ENUM("Oversampling Input", uda1380_os_enum), /* OS */ + SOC_DOUBLE_S8_TLV("ADC Capture Volume", UDA1380_DEC, -128, 48, dec_tlv), /* ML_DEC, MR_DEC */ +/**/ SOC_SINGLE("ADC Capture Switch", UDA1380_PGA, 15, 1, 1), /* MT_ADC */ + SOC_DOUBLE_TLV("Line Capture Volume", UDA1380_PGA, 0, 8, 8, 0, pga_tlv), /* PGA_GAINCTRLL, PGA_GAINCTRLR */ + SOC_SINGLE("ADC Polarity inverting Switch", UDA1380_ADC, 12, 1, 0), /* ADCPOL_INV */ + SOC_SINGLE_TLV("Mic Capture Volume", UDA1380_ADC, 8, 15, 0, vga_tlv), /* VGA_CTRL */ + SOC_SINGLE("DC Filter Bypass Switch", UDA1380_ADC, 1, 1, 0), /* SKIP_DCFIL (before decimator) */ + SOC_SINGLE("DC Filter Enable Switch", UDA1380_ADC, 0, 1, 0), /* EN_DCFIL (at output of decimator) */ + SOC_SINGLE("AGC Timing", UDA1380_AGC, 8, 7, 0), /* TODO: enum, see table 62 */ + SOC_SINGLE("AGC Target level", UDA1380_AGC, 2, 3, 1), /* AGC_LEVEL */ + /* -5.5, -8, -11.5, -14 dBFS */ + 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); + +/* Output mux */ +static const struct snd_kcontrol_new uda1380_output_mux_control = + SOC_DAPM_ENUM("Route", uda1380_output_sel_enum); + +/* Capture mux */ +static const struct snd_kcontrol_new uda1380_capture_mux_control = + SOC_DAPM_ENUM("Route", uda1380_capture_sel_enum); + + +static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = { + SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, + &uda1380_input_mux_control), + SND_SOC_DAPM_MUX("Output Mux", SND_SOC_NOPM, 0, 0, + &uda1380_output_mux_control), + SND_SOC_DAPM_MUX("Capture Mux", SND_SOC_NOPM, 0, 0, + &uda1380_capture_mux_control), + SND_SOC_DAPM_PGA("Left PGA", UDA1380_PM, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right PGA", UDA1380_PM, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mic LNA", UDA1380_PM, 4, 0, NULL, 0), + SND_SOC_DAPM_ADC("Left ADC", "Left Capture", UDA1380_PM, 2, 0), + SND_SOC_DAPM_ADC("Right ADC", "Right Capture", UDA1380_PM, 0, 0), + SND_SOC_DAPM_INPUT("VINM"), + SND_SOC_DAPM_INPUT("VINL"), + SND_SOC_DAPM_INPUT("VINR"), + SND_SOC_DAPM_MIXER("Analog Mixer", UDA1380_PM, 6, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("VOUTLHP"), + SND_SOC_DAPM_OUTPUT("VOUTRHP"), + SND_SOC_DAPM_OUTPUT("VOUTL"), + SND_SOC_DAPM_OUTPUT("VOUTR"), + SND_SOC_DAPM_DAC("DAC", "Playback", UDA1380_PM, 10, 0), + SND_SOC_DAPM_PGA("HeadPhone Driver", UDA1380_PM, 13, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + + /* output mux */ + {"HeadPhone Driver", NULL, "Output Mux"}, + {"VOUTR", NULL, "Output Mux"}, + {"VOUTL", NULL, "Output Mux"}, + + {"Analog Mixer", NULL, "VINR"}, + {"Analog Mixer", NULL, "VINL"}, + {"Analog Mixer", NULL, "DAC"}, + + {"Output Mux", "DAC", "DAC"}, + {"Output Mux", "Analog Mixer", "Analog Mixer"}, + + /* {"DAC", "Digital Mixer", "I2S" } */ + + /* headphone driver */ + {"VOUTLHP", NULL, "HeadPhone Driver"}, + {"VOUTRHP", NULL, "HeadPhone Driver"}, + + /* input mux */ + {"Left ADC", NULL, "Input Mux"}, + {"Input Mux", "Mic", "Mic LNA"}, + {"Input Mux", "Mic + Line R", "Mic LNA"}, + {"Input Mux", "Line L", "Left PGA"}, + {"Input Mux", "Line", "Left PGA"}, + + /* right input */ + {"Right ADC", "Mic + Line R", "Right PGA"}, + {"Right ADC", "Line", "Right PGA"}, + + /* inputs */ + {"Mic LNA", NULL, "VINM"}, + {"Left PGA", NULL, "VINL"}, + {"Right PGA", NULL, "VINR"}, +}; + +static int uda1380_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, uda1380_dapm_widgets, + ARRAY_SIZE(uda1380_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static int uda1380_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + int iface; + + /* set up DAI based upon fmt */ + iface = uda1380_read_reg_cache(codec, UDA1380_IFACE); + iface &= ~(R01_SFORI_MASK | R01_SIM | R01_SFORO_MASK); + + /* FIXME: how to select I2S for DATAO and MSB for DATAI correctly? */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= R01_SFORI_I2S | R01_SFORO_I2S; + break; + case SND_SOC_DAIFMT_LSB: + iface |= R01_SFORI_LSB16 | R01_SFORO_I2S; + break; + case SND_SOC_DAIFMT_MSB: + iface |= R01_SFORI_MSB | R01_SFORO_I2S; + } + + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) == SND_SOC_DAIFMT_CBM_CFM) + iface |= R01_SIM; + + uda1380_write(codec, UDA1380_IFACE, iface); + + return 0; +} + +/* + * Flush reg cache + * We can only write the interpolator and decimator registers + * when the DAI is being clocked by the CPU DAI. It's up to the + * machine and cpu DAI driver to do this before we are called. + */ +static int uda1380_pcm_prepare(struct snd_pcm_substream *substream) +{ + 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, reg_start, reg_end, clk; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + reg_start = UDA1380_MVOL; + reg_end = UDA1380_MIXER; + } else { + reg_start = UDA1380_DEC; + reg_end = UDA1380_AGC; + } + + /* FIXME disable DAC_CLK */ + clk = uda1380_read_reg_cache(codec, UDA1380_CLK); + uda1380_write(codec, UDA1380_CLK, clk & ~R00_DAC_CLK); + + for (reg = reg_start; reg <= reg_end; reg++) { + pr_debug("uda1380: flush reg %x val %x:", reg, + uda1380_read_reg_cache(codec, reg)); + uda1380_write(codec, reg, uda1380_read_reg_cache(codec, reg)); + } + + /* FIXME enable DAC_CLK */ + uda1380_write(codec, UDA1380_CLK, clk | R00_DAC_CLK); + + return 0; +} + +static int uda1380_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK); + + /* set WSPLL power and divider if running from this clock */ + if (clk & R00_DAC_CLK) { + int rate = params_rate(params); + u16 pm = uda1380_read_reg_cache(codec, UDA1380_PM); + clk &= ~0x3; /* clear SEL_LOOP_DIV */ + switch (rate) { + case 6250 ... 12500: + clk |= 0x0; + break; + case 12501 ... 25000: + clk |= 0x1; + break; + case 25001 ... 50000: + clk |= 0x2; + break; + case 50001 ... 100000: + clk |= 0x3; + break; + } + uda1380_write(codec, UDA1380_PM, R02_PON_PLL | pm); + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + clk |= R00_EN_DAC | R00_EN_INT; + else + clk |= R00_EN_ADC | R00_EN_DEC; + + uda1380_write(codec, UDA1380_CLK, clk); + return 0; +} + +static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK); + + /* shut down WSPLL power if running from this clock */ + if (clk & R00_DAC_CLK) { + u16 pm = uda1380_read_reg_cache(codec, UDA1380_PM); + uda1380_write(codec, UDA1380_PM, ~R02_PON_PLL & pm); + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + clk &= ~(R00_EN_DAC | R00_EN_INT); + else + clk &= ~(R00_EN_ADC | R00_EN_DEC); + + uda1380_write(codec, UDA1380_CLK, clk); +} + +static int uda1380_mute(struct snd_soc_codec_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 mute_reg = uda1380_read_reg_cache(codec, UDA1380_DEEMP) & ~R13_MTM; + + /* FIXME: mute(codec,0) is called when the magician clock is already + * set to WSPLL, but for some unknown reason writing to interpolator + * registers works only when clocked by SYSCLK */ + u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK); + uda1380_write(codec, UDA1380_CLK, ~R00_DAC_CLK & clk); + if (mute) + uda1380_write(codec, UDA1380_DEEMP, mute_reg | R13_MTM); + else + uda1380_write(codec, UDA1380_DEEMP, mute_reg); + uda1380_write(codec, UDA1380_CLK, clk); + return 0; +} + +static int uda1380_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + int pm = uda1380_read_reg_cache(codec, UDA1380_PM); + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + uda1380_write(codec, UDA1380_PM, R02_PON_BIAS | pm); + break; + case SND_SOC_BIAS_STANDBY: + uda1380_write(codec, UDA1380_PM, R02_PON_BIAS); + break; + case SND_SOC_BIAS_OFF: + uda1380_write(codec, UDA1380_PM, 0x0); + break; + } + codec->bias_level = level; + return 0; +} + +#define UDA1380_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) + +struct snd_soc_codec_dai uda1380_dai[] = { +{ + .name = "UDA1380", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = UDA1380_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = UDA1380_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = { + .hw_params = uda1380_pcm_hw_params, + .shutdown = uda1380_pcm_shutdown, + .prepare = uda1380_pcm_prepare, + }, + .dai_ops = { + .digital_mute = uda1380_mute, + .set_fmt = uda1380_set_dai_fmt, + }, +}, +{ /* playback only - dual interface */ + .name = "UDA1380", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = UDA1380_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = { + .hw_params = uda1380_pcm_hw_params, + .shutdown = uda1380_pcm_shutdown, + .prepare = uda1380_pcm_prepare, + }, + .dai_ops = { + .digital_mute = uda1380_mute, + .set_fmt = uda1380_set_dai_fmt, + }, +}, +{ /* capture only - dual interface*/ + .name = "UDA1380", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = UDA1380_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = { + .hw_params = uda1380_pcm_hw_params, + .shutdown = uda1380_pcm_shutdown, + .prepare = uda1380_pcm_prepare, + }, + .dai_ops = { + .set_fmt = uda1380_set_dai_fmt, + }, +}, +}; +EXPORT_SYMBOL_GPL(uda1380_dai); + +static int uda1380_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + uda1380_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int uda1380_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + int i; + u8 data[2]; + u16 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(uda1380_reg); i++) { + data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); + data[1] = cache[i] & 0x00ff; + codec->hw_write(codec->control_data, data, 2); + } + uda1380_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + uda1380_set_bias_level(codec, codec->suspend_bias_level); + return 0; +} + +/* + * initialise the UDA1380 driver + * register mixer and dsp interfaces with the kernel + */ +static int uda1380_init(struct snd_soc_device *socdev, int dac_clk) +{ + struct snd_soc_codec *codec = socdev->codec; + int ret = 0; + + codec->name = "UDA1380"; + codec->owner = THIS_MODULE; + codec->read = uda1380_read_reg_cache; + codec->write = uda1380_write; + codec->set_bias_level = uda1380_set_bias_level; + codec->dai = uda1380_dai; + codec->num_dai = ARRAY_SIZE(uda1380_dai); + codec->reg_cache = kmemdup(uda1380_reg, sizeof(uda1380_reg), + GFP_KERNEL); + if (codec->reg_cache == NULL) + return -ENOMEM; + codec->reg_cache_size = sizeof(uda1380_reg); + codec->reg_cache_step = 2; + uda1380_reset(codec); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + pr_err("uda1380: failed to create pcms\n"); + goto pcm_err; + } + + /* power on device */ + uda1380_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + /* set clock input */ + switch (dac_clk) { + case UDA1380_DAC_CLK_SYSCLK: + uda1380_write(codec, UDA1380_CLK, 0); + break; + case UDA1380_DAC_CLK_WSPLL: + uda1380_write(codec, UDA1380_CLK, R00_DAC_CLK); + break; + } + + /* uda1380 init */ + uda1380_add_controls(codec); + uda1380_add_widgets(codec); + ret = snd_soc_register_card(socdev); + if (ret < 0) { + pr_err("uda1380: failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + return ret; +} + +static struct snd_soc_device *uda1380_socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + +#define I2C_DRIVERID_UDA1380 0xfefe /* liam - need a proper id */ + +static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; + +/* Magic definition of all other variables and things */ +I2C_CLIENT_INSMOD; + +static struct i2c_driver uda1380_i2c_driver; +static struct i2c_client client_template; + +/* If the i2c layer weren't so broken, we could pass this kind of data + around */ + +static int uda1380_codec_probe(struct i2c_adapter *adap, int addr, int kind) +{ + struct snd_soc_device *socdev = uda1380_socdev; + struct uda1380_setup_data *setup = socdev->codec_data; + struct snd_soc_codec *codec = socdev->codec; + struct i2c_client *i2c; + int ret; + + if (addr != setup->i2c_address) + return -ENODEV; + + client_template.adapter = adap; + client_template.addr = addr; + + i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); + if (i2c == NULL) { + kfree(codec); + return -ENOMEM; + } + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + ret = i2c_attach_client(i2c); + if (ret < 0) { + pr_err("uda1380: failed to attach codec at addr %x\n", addr); + goto err; + } + + ret = uda1380_init(socdev, setup->dac_clk); + if (ret < 0) { + pr_err("uda1380: failed to initialise UDA1380\n"); + goto err; + } + return ret; + +err: + kfree(codec); + kfree(i2c); + return ret; +} + +static int uda1380_i2c_detach(struct i2c_client *client) +{ + struct snd_soc_codec *codec = i2c_get_clientdata(client); + i2c_detach_client(client); + kfree(codec->reg_cache); + kfree(client); + return 0; +} + +static int uda1380_i2c_attach(struct i2c_adapter *adap) +{ + return i2c_probe(adap, &addr_data, uda1380_codec_probe); +} + +static struct i2c_driver uda1380_i2c_driver = { + .driver = { + .name = "UDA1380 I2C Codec", + .owner = THIS_MODULE, + }, + .id = I2C_DRIVERID_UDA1380, + .attach_adapter = uda1380_i2c_attach, + .detach_client = uda1380_i2c_detach, + .command = NULL, +}; + +static struct i2c_client client_template = { + .name = "UDA1380", + .driver = &uda1380_i2c_driver, +}; +#endif + +static int uda1380_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct uda1380_setup_data *setup; + struct snd_soc_codec *codec; + int ret = 0; + + pr_info("UDA1380 Audio Codec %s", UDA1380_VERSION); + + setup = socdev->codec_data; + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + uda1380_socdev = socdev; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + if (setup->i2c_address) { + normal_i2c[0] = setup->i2c_address; + codec->hw_write = (hw_write_t)i2c_master_send; + ret = i2c_add_driver(&uda1380_i2c_driver); + if (ret != 0) + printk(KERN_ERR "can't add i2c driver"); + } +#else + /* Add other interfaces here */ +#endif + return ret; +} + +/* power down chip */ +static int uda1380_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec->control_data) + uda1380_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&uda1380_i2c_driver); +#endif + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_uda1380 = { + .probe = uda1380_probe, + .remove = uda1380_remove, + .suspend = uda1380_suspend, + .resume = uda1380_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_uda1380); + +MODULE_AUTHOR("Giorgio Padrin"); +MODULE_DESCRIPTION("Audio support for codec Philips UDA1380"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/uda1380.h b/sound/soc/codecs/uda1380.h new file mode 100644 index 00000000000..f9d885c8bf0 --- /dev/null +++ b/sound/soc/codecs/uda1380.h @@ -0,0 +1,89 @@ +/* + * Audio support for Philips UDA1380 + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Copyright (c) 2005 Giorgio Padrin + */ + +#ifndef _UDA1380_H +#define _UDA1380_H + +#define UDA1380_CLK 0x00 +#define UDA1380_IFACE 0x01 +#define UDA1380_PM 0x02 +#define UDA1380_AMIX 0x03 +#define UDA1380_HP 0x04 +#define UDA1380_MVOL 0x10 +#define UDA1380_MIXVOL 0x11 +#define UDA1380_MODE 0x12 +#define UDA1380_DEEMP 0x13 +#define UDA1380_MIXER 0x14 +#define UDA1380_INTSTAT 0x18 +#define UDA1380_DEC 0x20 +#define UDA1380_PGA 0x21 +#define UDA1380_ADC 0x22 +#define UDA1380_AGC 0x23 +#define UDA1380_DECSTAT 0x28 +#define UDA1380_RESET 0x7f + +#define UDA1380_CACHEREGNUM 0x24 + +/* Register flags */ +#define R00_EN_ADC 0x0800 +#define R00_EN_DEC 0x0400 +#define R00_EN_DAC 0x0200 +#define R00_EN_INT 0x0100 +#define R00_DAC_CLK 0x0010 +#define R01_SFORI_I2S 0x0000 +#define R01_SFORI_LSB16 0x0100 +#define R01_SFORI_LSB18 0x0200 +#define R01_SFORI_LSB20 0x0300 +#define R01_SFORI_MSB 0x0500 +#define R01_SFORI_MASK 0x0700 +#define R01_SFORO_I2S 0x0000 +#define R01_SFORO_LSB16 0x0001 +#define R01_SFORO_LSB18 0x0002 +#define R01_SFORO_LSB20 0x0003 +#define R01_SFORO_LSB24 0x0004 +#define R01_SFORO_MSB 0x0005 +#define R01_SFORO_MASK 0x0007 +#define R01_SEL_SOURCE 0x0040 +#define R01_SIM 0x0010 +#define R02_PON_PLL 0x8000 +#define R02_PON_HP 0x2000 +#define R02_PON_DAC 0x0400 +#define R02_PON_BIAS 0x0100 +#define R02_EN_AVC 0x0080 +#define R02_PON_AVC 0x0040 +#define R02_PON_LNA 0x0010 +#define R02_PON_PGAL 0x0008 +#define R02_PON_ADCL 0x0004 +#define R02_PON_PGAR 0x0002 +#define R02_PON_ADCR 0x0001 +#define R13_MTM 0x4000 +#define R14_SILENCE 0x0080 +#define R14_SDET_ON 0x0040 +#define R21_MT_ADC 0x8000 +#define R22_SEL_LNA 0x0008 +#define R22_SEL_MIC 0x0004 +#define R22_SKIP_DCFIL 0x0002 +#define R23_AGC_EN 0x0001 + +struct uda1380_setup_data { + unsigned short i2c_address; + int dac_clk; +#define UDA1380_DAC_CLK_SYSCLK 0 +#define UDA1380_DAC_CLK_WSPLL 1 +}; + +#define UDA1380_DAI_DUPLEX 0 /* playback and capture on single DAI */ +#define UDA1380_DAI_PLAYBACK 1 /* playback DAI */ +#define UDA1380_DAI_CAPTURE 2 /* capture DAI */ + +extern struct snd_soc_codec_dai uda1380_dai[3]; +extern struct snd_soc_codec_device soc_codec_dev_uda1380; + +#endif /* _UDA1380_H */ -- cgit From 5d421516670e8009436e299bd25cff1a6d3a4707 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 5 Jun 2008 13:49:32 +0100 Subject: [ALSA] ASoC: Add WM8510 driver The WM8510 is a mono CODEC with speaker driver optimised for telephony applications, featuring: - 16/20/24/32 bit audio at data rates between 8kHz and 48kHz - On-chip PLL - Dual microphone inputs This driver was originally written by Liam Girdwood with updates from Brett Saunders, Geoffrey Wossum and myself. Signed-off-by: Liam Girdwood Signed-off-by: Brett Saunders Signed-off-by: Geoffrey Wossum Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8510.c | 836 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8510.h | 103 ++++++ 4 files changed, 944 insertions(+) create mode 100644 sound/soc/codecs/wm8510.c create mode 100644 sound/soc/codecs/wm8510.h (limited to 'sound/soc') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7beefccfa82..ef400b36344 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -5,6 +5,9 @@ config SND_SOC_AC97_CODEC config SND_SOC_UDA1380 tristate +config SND_SOC_WM8510 + tristate + config SND_SOC_WM8731 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index d5926a11707..b92c6654166 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -1,5 +1,6 @@ snd-soc-ac97-objs := ac97.o snd-soc-uda1380-objs := uda1380.o +snd-soc-wm8510-objs := wm8510.o snd-soc-wm8731-objs := wm8731.o snd-soc-wm8750-objs := wm8750.o snd-soc-wm8753-objs := wm8753.o @@ -10,6 +11,7 @@ snd-soc-tlv320aic3x-objs := tlv320aic3x.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o +obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c new file mode 100644 index 00000000000..6f54cb0f50f --- /dev/null +++ b/sound/soc/codecs/wm8510.c @@ -0,0 +1,836 @@ +/* + * wm8510.c -- WM8510 ALSA Soc Audio driver + * + * Copyright 2006 Wolfson Microelectronics PLC. + * + * Author: Liam Girdwood + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "wm8510.h" + +#define AUDIO_NAME "wm8510" +#define WM8510_VERSION "0.6" + +/* + * Debug + */ + +#define WM8510_DEBUG 0 + +#ifdef WM8510_DEBUG +#define dbg(format, arg...) \ + printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg) +#else +#define dbg(format, arg...) do {} while (0) +#endif +#define err(format, arg...) \ + printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg) +#define info(format, arg...) \ + printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg) +#define warn(format, arg...) \ + printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg) + +struct snd_soc_codec_device soc_codec_dev_wm8510; + +/* + * wm8510 register cache + * We can't read the WM8510 register space when we are + * using 2 wire for device control, so we cache them instead. + */ +static const u16 wm8510_reg[WM8510_CACHEREGNUM] = { + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0050, 0x0000, 0x0140, 0x0000, + 0x0000, 0x0000, 0x0000, 0x00ff, + 0x0000, 0x0000, 0x0100, 0x00ff, + 0x0000, 0x0000, 0x012c, 0x002c, + 0x002c, 0x002c, 0x002c, 0x0000, + 0x0032, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0038, 0x000b, 0x0032, 0x0000, + 0x0008, 0x000c, 0x0093, 0x00e9, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0003, 0x0010, 0x0000, 0x0000, + 0x0000, 0x0002, 0x0001, 0x0000, + 0x0000, 0x0000, 0x0039, 0x0000, + 0x0000, +}; + +/* + * read wm8510 register cache + */ +static inline unsigned int wm8510_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg == WM8510_RESET) + return 0; + if (reg >= WM8510_CACHEREGNUM) + return -1; + return cache[reg]; +} + +/* + * write wm8510 register cache + */ +static inline void wm8510_write_reg_cache(struct snd_soc_codec *codec, + u16 reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + if (reg >= WM8510_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * write to the WM8510 register space + */ +static int wm8510_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + /* data is + * D15..D9 WM8510 register offset + * D8...D0 register data + */ + data[0] = (reg << 1) | ((value >> 8) & 0x0001); + data[1] = value & 0x00ff; + + wm8510_write_reg_cache(codec, reg, value); + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + else + return -EIO; +} + +#define wm8510_reset(c) wm8510_write(c, WM8510_RESET, 0) + +static const char *wm8510_companding[] = { "Off", "NC", "u-law", "A-law" }; +static const char *wm8510_deemp[] = { "None", "32kHz", "44.1kHz", "48kHz" }; +static const char *wm8510_alc[] = { "ALC", "Limiter" }; + +static const struct soc_enum wm8510_enum[] = { + SOC_ENUM_SINGLE(WM8510_COMP, 1, 4, wm8510_companding), /* adc */ + SOC_ENUM_SINGLE(WM8510_COMP, 3, 4, wm8510_companding), /* dac */ + SOC_ENUM_SINGLE(WM8510_DAC, 4, 4, wm8510_deemp), + SOC_ENUM_SINGLE(WM8510_ALC3, 8, 2, wm8510_alc), +}; + +static const struct snd_kcontrol_new wm8510_snd_controls[] = { + +SOC_SINGLE("Digital Loopback Switch", WM8510_COMP, 0, 1, 0), + +SOC_ENUM("DAC Companding", wm8510_enum[1]), +SOC_ENUM("ADC Companding", wm8510_enum[0]), + +SOC_ENUM("Playback De-emphasis", wm8510_enum[2]), +SOC_SINGLE("DAC Inversion Switch", WM8510_DAC, 0, 1, 0), + +SOC_SINGLE("Master Playback Volume", WM8510_DACVOL, 0, 127, 0), + +SOC_SINGLE("High Pass Filter Switch", WM8510_ADC, 8, 1, 0), +SOC_SINGLE("High Pass Cut Off", WM8510_ADC, 4, 7, 0), +SOC_SINGLE("ADC Inversion Switch", WM8510_COMP, 0, 1, 0), + +SOC_SINGLE("Capture Volume", WM8510_ADCVOL, 0, 127, 0), + +SOC_SINGLE("DAC Playback Limiter Switch", WM8510_DACLIM1, 8, 1, 0), +SOC_SINGLE("DAC Playback Limiter Decay", WM8510_DACLIM1, 4, 15, 0), +SOC_SINGLE("DAC Playback Limiter Attack", WM8510_DACLIM1, 0, 15, 0), + +SOC_SINGLE("DAC Playback Limiter Threshold", WM8510_DACLIM2, 4, 7, 0), +SOC_SINGLE("DAC Playback Limiter Boost", WM8510_DACLIM2, 0, 15, 0), + +SOC_SINGLE("ALC Enable Switch", WM8510_ALC1, 8, 1, 0), +SOC_SINGLE("ALC Capture Max Gain", WM8510_ALC1, 3, 7, 0), +SOC_SINGLE("ALC Capture Min Gain", WM8510_ALC1, 0, 7, 0), + +SOC_SINGLE("ALC Capture ZC Switch", WM8510_ALC2, 8, 1, 0), +SOC_SINGLE("ALC Capture Hold", WM8510_ALC2, 4, 7, 0), +SOC_SINGLE("ALC Capture Target", WM8510_ALC2, 0, 15, 0), + +SOC_ENUM("ALC Capture Mode", wm8510_enum[3]), +SOC_SINGLE("ALC Capture Decay", WM8510_ALC3, 4, 15, 0), +SOC_SINGLE("ALC Capture Attack", WM8510_ALC3, 0, 15, 0), + +SOC_SINGLE("ALC Capture Noise Gate Switch", WM8510_NGATE, 3, 1, 0), +SOC_SINGLE("ALC Capture Noise Gate Threshold", WM8510_NGATE, 0, 7, 0), + +SOC_SINGLE("Capture PGA ZC Switch", WM8510_INPPGA, 7, 1, 0), +SOC_SINGLE("Capture PGA Volume", WM8510_INPPGA, 0, 63, 0), + +SOC_SINGLE("Speaker Playback ZC Switch", WM8510_SPKVOL, 7, 1, 0), +SOC_SINGLE("Speaker Playback Switch", WM8510_SPKVOL, 6, 1, 1), +SOC_SINGLE("Speaker Playback Volume", WM8510_SPKVOL, 0, 63, 0), +SOC_SINGLE("Speaker Boost", WM8510_OUTPUT, 2, 1, 0), + +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), +SOC_DAPM_SINGLE("Aux Playback Switch", WM8510_SPKMIX, 5, 1, 0), +SOC_DAPM_SINGLE("PCM Playback Switch", WM8510_SPKMIX, 0, 1, 0), +}; + +/* Mono Output Mixer */ +static const struct snd_kcontrol_new wm8510_mono_mixer_controls[] = { +SOC_DAPM_SINGLE("Line Bypass Switch", WM8510_MONOMIX, 1, 1, 0), +SOC_DAPM_SINGLE("Aux Playback Switch", WM8510_MONOMIX, 2, 1, 0), +SOC_DAPM_SINGLE("PCM Playback Switch", WM8510_MONOMIX, 0, 1, 0), +}; + +static const struct snd_kcontrol_new wm8510_boost_controls[] = { +SOC_DAPM_SINGLE("Mic PGA Switch", WM8510_INPPGA, 6, 1, 0), +SOC_DAPM_SINGLE("Aux Volume", WM8510_ADCBOOST, 0, 7, 0), +SOC_DAPM_SINGLE("Mic Volume", WM8510_ADCBOOST, 4, 7, 0), +}; + +static const struct snd_kcontrol_new wm8510_micpga_controls[] = { +SOC_DAPM_SINGLE("MICP Switch", WM8510_INPUT, 0, 1, 0), +SOC_DAPM_SINGLE("MICN Switch", WM8510_INPUT, 1, 1, 0), +SOC_DAPM_SINGLE("AUX Switch", WM8510_INPUT, 2, 1, 0), +}; + +static const struct snd_soc_dapm_widget wm8510_dapm_widgets[] = { +SND_SOC_DAPM_MIXER("Speaker Mixer", WM8510_POWER3, 2, 0, + &wm8510_speaker_mixer_controls[0], + ARRAY_SIZE(wm8510_speaker_mixer_controls)), +SND_SOC_DAPM_MIXER("Mono Mixer", WM8510_POWER3, 3, 0, + &wm8510_mono_mixer_controls[0], + ARRAY_SIZE(wm8510_mono_mixer_controls)), +SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8510_POWER3, 0, 0), +SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8510_POWER2, 0, 0), +SND_SOC_DAPM_PGA("Aux Input", WM8510_POWER1, 6, 0, NULL, 0), +SND_SOC_DAPM_PGA("SpkN Out", WM8510_POWER3, 5, 0, NULL, 0), +SND_SOC_DAPM_PGA("SpkP Out", WM8510_POWER3, 6, 0, NULL, 0), +SND_SOC_DAPM_PGA("Mono Out", WM8510_POWER3, 7, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Mic PGA", WM8510_POWER2, 2, 0, + &wm8510_micpga_controls[0], + ARRAY_SIZE(wm8510_micpga_controls)), +SND_SOC_DAPM_MIXER("Boost Mixer", WM8510_POWER2, 4, 0, + &wm8510_boost_controls[0], + ARRAY_SIZE(wm8510_boost_controls)), + +SND_SOC_DAPM_MICBIAS("Mic Bias", WM8510_POWER1, 4, 0), + +SND_SOC_DAPM_INPUT("MICN"), +SND_SOC_DAPM_INPUT("MICP"), +SND_SOC_DAPM_INPUT("AUX"), +SND_SOC_DAPM_OUTPUT("MONOOUT"), +SND_SOC_DAPM_OUTPUT("SPKOUTP"), +SND_SOC_DAPM_OUTPUT("SPKOUTN"), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* Mono output mixer */ + {"Mono Mixer", "PCM Playback Switch", "DAC"}, + {"Mono Mixer", "Aux Playback Switch", "Aux Input"}, + {"Mono Mixer", "Line Bypass Switch", "Boost Mixer"}, + + /* Speaker output mixer */ + {"Speaker Mixer", "PCM Playback Switch", "DAC"}, + {"Speaker Mixer", "Aux Playback Switch", "Aux Input"}, + {"Speaker Mixer", "Line Bypass Switch", "Boost Mixer"}, + + /* Outputs */ + {"Mono Out", NULL, "Mono Mixer"}, + {"MONOOUT", NULL, "Mono Out"}, + {"SpkN Out", NULL, "Speaker Mixer"}, + {"SpkP Out", NULL, "Speaker Mixer"}, + {"SPKOUTN", NULL, "SpkN Out"}, + {"SPKOUTP", NULL, "SpkP Out"}, + + /* Microphone PGA */ + {"Mic PGA", "MICN Switch", "MICN"}, + {"Mic PGA", "MICP Switch", "MICP"}, + { "Mic PGA", "AUX Switch", "Aux Input" }, + + /* Boost Mixer */ + {"Boost Mixer", "Mic PGA Switch", "Mic PGA"}, + {"Boost Mixer", "Mic Volume", "MICP"}, + {"Boost Mixer", "Aux Volume", "Aux Input"}, + + {"ADC", NULL, "Boost Mixer"}, +}; + +static int wm8510_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, wm8510_dapm_widgets, + ARRAY_SIZE(wm8510_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +struct pll_ { + unsigned int pre_div:4; /* prescale - 1 */ + unsigned int n:4; + unsigned int k; +}; + +static struct pll_ pll_div; + +/* The size in bits of the pll divide multiplied by 10 + * to allow rounding later */ +#define FIXED_PLL_SIZE ((1 << 24) * 10) + +static void pll_factors(unsigned int target, unsigned int source) +{ + unsigned long long Kpart; + unsigned int K, Ndiv, Nmod; + + Ndiv = target / source; + if (Ndiv < 6) { + source >>= 1; + pll_div.pre_div = 1; + Ndiv = target / source; + } else + pll_div.pre_div = 0; + + if ((Ndiv < 6) || (Ndiv > 12)) + printk(KERN_WARNING + "WM8510 N value %d outwith recommended range!d\n", + Ndiv); + + pll_div.n = Ndiv; + Nmod = target % source; + Kpart = FIXED_PLL_SIZE * (long long)Nmod; + + do_div(Kpart, source); + + K = Kpart & 0xFFFFFFFF; + + /* Check if we need to round */ + if ((K % 10) >= 5) + K += 5; + + /* Move down to proper range now rounding is done */ + K /= 10; + + pll_div.k = K; +} + +static int wm8510_set_dai_pll(struct snd_soc_codec_dai *codec_dai, + int pll_id, unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 reg; + + if (freq_in == 0 || freq_out == 0) { + /* Clock CODEC directly from MCLK */ + reg = wm8510_read_reg_cache(codec, WM8510_CLOCK); + wm8510_write(codec, WM8510_CLOCK, reg & 0x0ff); + + /* Turn off PLL */ + reg = wm8510_read_reg_cache(codec, WM8510_POWER1); + wm8510_write(codec, WM8510_POWER1, reg & 0x1df); + return 0; + } + + pll_factors(freq_out*8, freq_in); + + wm8510_write(codec, WM8510_PLLN, (pll_div.pre_div << 4) | pll_div.n); + wm8510_write(codec, WM8510_PLLK1, pll_div.k >> 18); + wm8510_write(codec, WM8510_PLLK2, (pll_div.k >> 9) & 0x1ff); + wm8510_write(codec, WM8510_PLLK3, pll_div.k & 0x1ff); + reg = wm8510_read_reg_cache(codec, WM8510_POWER1); + wm8510_write(codec, WM8510_POWER1, reg | 0x020); + + /* Run CODEC from PLL instead of MCLK */ + reg = wm8510_read_reg_cache(codec, WM8510_CLOCK); + wm8510_write(codec, WM8510_CLOCK, reg | 0x100); + + return 0; +} + +/* + * Configure WM8510 clock dividers. + */ +static int wm8510_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai, + int div_id, int div) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 reg; + + switch (div_id) { + case WM8510_OPCLKDIV: + reg = wm8510_read_reg_cache(codec, WM8510_GPIO) & 0x1cf; + wm8510_write(codec, WM8510_GPIO, reg | div); + break; + case WM8510_MCLKDIV: + reg = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1f; + wm8510_write(codec, WM8510_CLOCK, reg | div); + break; + case WM8510_ADCCLK: + reg = wm8510_read_reg_cache(codec, WM8510_ADC) & 0x1f7; + wm8510_write(codec, WM8510_ADC, reg | div); + break; + case WM8510_DACCLK: + reg = wm8510_read_reg_cache(codec, WM8510_DAC) & 0x1f7; + wm8510_write(codec, WM8510_DAC, reg | div); + break; + case WM8510_BCLKDIV: + reg = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1e3; + wm8510_write(codec, WM8510_CLOCK, reg | div); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int wm8510_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface = 0; + u16 clk = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1fe; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + clk |= 0x0001; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= 0x0010; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= 0x0008; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= 0x00018; + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= 0x0180; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= 0x0100; + break; + case SND_SOC_DAIFMT_NB_IF: + iface |= 0x0080; + break; + default: + return -EINVAL; + } + + wm8510_write(codec, WM8510_IFACE, iface); + wm8510_write(codec, WM8510_CLOCK, clk); + return 0; +} + +static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u16 iface = wm8510_read_reg_cache(codec, WM8510_IFACE) & 0x19f; + u16 adn = wm8510_read_reg_cache(codec, WM8510_ADD) & 0x1f1; + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface |= 0x0020; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface |= 0x0040; + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface |= 0x0060; + break; + } + + /* filter coefficient */ + switch (params_rate(params)) { + case SNDRV_PCM_RATE_8000: + adn |= 0x5 << 1; + break; + case SNDRV_PCM_RATE_11025: + adn |= 0x4 << 1; + break; + case SNDRV_PCM_RATE_16000: + adn |= 0x3 << 1; + break; + case SNDRV_PCM_RATE_22050: + adn |= 0x2 << 1; + break; + case SNDRV_PCM_RATE_32000: + adn |= 0x1 << 1; + break; + case SNDRV_PCM_RATE_44100: + case SNDRV_PCM_RATE_48000: + break; + } + + wm8510_write(codec, WM8510_IFACE, iface); + wm8510_write(codec, WM8510_ADD, adn); + return 0; +} + +static int wm8510_mute(struct snd_soc_codec_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 mute_reg = wm8510_read_reg_cache(codec, WM8510_DAC) & 0xffbf; + + if (mute) + wm8510_write(codec, WM8510_DAC, mute_reg | 0x40); + else + wm8510_write(codec, WM8510_DAC, mute_reg); + return 0; +} + +/* liam need to make this lower power with dapm */ +static int wm8510_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + + switch (level) { + case SND_SOC_BIAS_ON: + wm8510_write(codec, WM8510_POWER1, 0x1ff); + wm8510_write(codec, WM8510_POWER2, 0x1ff); + wm8510_write(codec, WM8510_POWER3, 0x1ff); + break; + case SND_SOC_BIAS_PREPARE: + case SND_SOC_BIAS_STANDBY: + break; + case SND_SOC_BIAS_OFF: + /* everything off, dac mute, inactive */ + wm8510_write(codec, WM8510_POWER1, 0x0); + wm8510_write(codec, WM8510_POWER2, 0x0); + wm8510_write(codec, WM8510_POWER3, 0x0); + break; + } + codec->bias_level = level; + return 0; +} + +#define WM8510_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) + +#define WM8510_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_codec_dai wm8510_dai = { + .name = "WM8510 HiFi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = WM8510_RATES, + .formats = WM8510_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = WM8510_RATES, + .formats = WM8510_FORMATS,}, + .ops = { + .hw_params = wm8510_pcm_hw_params, + }, + .dai_ops = { + .digital_mute = wm8510_mute, + .set_fmt = wm8510_set_dai_fmt, + .set_clkdiv = wm8510_set_dai_clkdiv, + .set_pll = wm8510_set_dai_pll, + }, +}; +EXPORT_SYMBOL_GPL(wm8510_dai); + +static int wm8510_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + wm8510_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int wm8510_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + int i; + u8 data[2]; + u16 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(wm8510_reg); i++) { + data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); + data[1] = cache[i] & 0x00ff; + codec->hw_write(codec->control_data, data, 2); + } + wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + wm8510_set_bias_level(codec, codec->suspend_bias_level); + return 0; +} + +/* + * initialise the WM8510 driver + * register the mixer and dsp interfaces with the kernel + */ +static int wm8510_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->codec; + int ret = 0; + + codec->name = "WM8510"; + codec->owner = THIS_MODULE; + codec->read = wm8510_read_reg_cache; + codec->write = wm8510_write; + codec->set_bias_level = wm8510_set_bias_level; + codec->dai = &wm8510_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(wm8510_reg); + codec->reg_cache = kmemdup(wm8510_reg, sizeof(wm8510_reg), GFP_KERNEL); + + if (codec->reg_cache == NULL) + return -ENOMEM; + + wm8510_reset(codec); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "wm8510: failed to create pcms\n"); + goto pcm_err; + } + + /* power on device */ + wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + wm8510_add_controls(codec); + wm8510_add_widgets(codec); + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "wm8510: failed to register card\n"); + goto card_err; + } + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + return ret; +} + +static struct snd_soc_device *wm8510_socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + +/* + * WM8510 2 wire address is 0x1a + */ +#define I2C_DRIVERID_WM8510 0xfefe /* liam - need a proper id */ + +static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; + +/* Magic definition of all other variables and things */ +I2C_CLIENT_INSMOD; + +static struct i2c_driver wm8510_i2c_driver; +static struct i2c_client client_template; + +/* If the i2c layer weren't so broken, we could pass this kind of data + around */ + +static int wm8510_codec_probe(struct i2c_adapter *adap, int addr, int kind) +{ + struct snd_soc_device *socdev = wm8510_socdev; + struct wm8510_setup_data *setup = socdev->codec_data; + struct snd_soc_codec *codec = socdev->codec; + struct i2c_client *i2c; + int ret; + + if (addr != setup->i2c_address) + return -ENODEV; + + client_template.adapter = adap; + client_template.addr = addr; + + i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); + if (i2c == NULL) { + kfree(codec); + return -ENOMEM; + } + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + ret = i2c_attach_client(i2c); + if (ret < 0) { + err("failed to attach codec at addr %x\n", addr); + goto err; + } + + ret = wm8510_init(socdev); + if (ret < 0) { + err("failed to initialise WM8510\n"); + goto err; + } + return ret; + +err: + kfree(codec); + kfree(i2c); + return ret; +} + +static int wm8510_i2c_detach(struct i2c_client *client) +{ + struct snd_soc_codec *codec = i2c_get_clientdata(client); + i2c_detach_client(client); + kfree(codec->reg_cache); + kfree(client); + return 0; +} + +static int wm8510_i2c_attach(struct i2c_adapter *adap) +{ + return i2c_probe(adap, &addr_data, wm8510_codec_probe); +} + +/* corgi i2c codec control layer */ +static struct i2c_driver wm8510_i2c_driver = { + .driver = { + .name = "WM8510 I2C Codec", + .owner = THIS_MODULE, + }, + .id = I2C_DRIVERID_WM8510, + .attach_adapter = wm8510_i2c_attach, + .detach_client = wm8510_i2c_detach, + .command = NULL, +}; + +static struct i2c_client client_template = { + .name = "WM8510", + .driver = &wm8510_i2c_driver, +}; +#endif + +static int wm8510_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct wm8510_setup_data *setup; + struct snd_soc_codec *codec; + int ret = 0; + + info("WM8510 Audio Codec %s", WM8510_VERSION); + + setup = socdev->codec_data; + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + wm8510_socdev = socdev; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + if (setup->i2c_address) { + normal_i2c[0] = setup->i2c_address; + codec->hw_write = (hw_write_t)i2c_master_send; + ret = i2c_add_driver(&wm8510_i2c_driver); + if (ret != 0) + printk(KERN_ERR "can't add i2c driver"); + } +#else + /* Add other interfaces here */ +#endif + return ret; +} + +/* power down chip */ +static int wm8510_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec->control_data) + wm8510_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&wm8510_i2c_driver); +#endif + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8510 = { + .probe = wm8510_probe, + .remove = wm8510_remove, + .suspend = wm8510_suspend, + .resume = wm8510_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8510); + +MODULE_DESCRIPTION("ASoC WM8510 driver"); +MODULE_AUTHOR("Liam Girdwood"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8510.h b/sound/soc/codecs/wm8510.h new file mode 100644 index 00000000000..c862e7b7d53 --- /dev/null +++ b/sound/soc/codecs/wm8510.h @@ -0,0 +1,103 @@ +/* + * wm8510.h -- WM8510 Soc Audio driver + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _WM8510_H +#define _WM8510_H + +/* WM8510 register space */ + +#define WM8510_RESET 0x0 +#define WM8510_POWER1 0x1 +#define WM8510_POWER2 0x2 +#define WM8510_POWER3 0x3 +#define WM8510_IFACE 0x4 +#define WM8510_COMP 0x5 +#define WM8510_CLOCK 0x6 +#define WM8510_ADD 0x7 +#define WM8510_GPIO 0x8 +#define WM8510_DAC 0xa +#define WM8510_DACVOL 0xb +#define WM8510_ADC 0xe +#define WM8510_ADCVOL 0xf +#define WM8510_EQ1 0x12 +#define WM8510_EQ2 0x13 +#define WM8510_EQ3 0x14 +#define WM8510_EQ4 0x15 +#define WM8510_EQ5 0x16 +#define WM8510_DACLIM1 0x18 +#define WM8510_DACLIM2 0x19 +#define WM8510_NOTCH1 0x1b +#define WM8510_NOTCH2 0x1c +#define WM8510_NOTCH3 0x1d +#define WM8510_NOTCH4 0x1e +#define WM8510_ALC1 0x20 +#define WM8510_ALC2 0x21 +#define WM8510_ALC3 0x22 +#define WM8510_NGATE 0x23 +#define WM8510_PLLN 0x24 +#define WM8510_PLLK1 0x25 +#define WM8510_PLLK2 0x26 +#define WM8510_PLLK3 0x27 +#define WM8510_ATTEN 0x28 +#define WM8510_INPUT 0x2c +#define WM8510_INPPGA 0x2d +#define WM8510_ADCBOOST 0x2f +#define WM8510_OUTPUT 0x31 +#define WM8510_SPKMIX 0x32 +#define WM8510_SPKVOL 0x36 +#define WM8510_MONOMIX 0x38 + +#define WM8510_CACHEREGNUM 57 + +/* Clock divider Id's */ +#define WM8510_OPCLKDIV 0 +#define WM8510_MCLKDIV 1 +#define WM8510_ADCCLK 2 +#define WM8510_DACCLK 3 +#define WM8510_BCLKDIV 4 + +/* DAC clock dividers */ +#define WM8510_DACCLK_F2 (1 << 3) +#define WM8510_DACCLK_F4 (0 << 3) + +/* ADC clock dividers */ +#define WM8510_ADCCLK_F2 (1 << 3) +#define WM8510_ADCCLK_F4 (0 << 3) + +/* PLL Out dividers */ +#define WM8510_OPCLKDIV_1 (0 << 4) +#define WM8510_OPCLKDIV_2 (1 << 4) +#define WM8510_OPCLKDIV_3 (2 << 4) +#define WM8510_OPCLKDIV_4 (3 << 4) + +/* BCLK clock dividers */ +#define WM8510_BCLKDIV_1 (0 << 2) +#define WM8510_BCLKDIV_2 (1 << 2) +#define WM8510_BCLKDIV_4 (2 << 2) +#define WM8510_BCLKDIV_8 (3 << 2) +#define WM8510_BCLKDIV_16 (4 << 2) +#define WM8510_BCLKDIV_32 (5 << 2) + +/* MCLK clock dividers */ +#define WM8510_MCLKDIV_1 (0 << 5) +#define WM8510_MCLKDIV_1_5 (1 << 5) +#define WM8510_MCLKDIV_2 (2 << 5) +#define WM8510_MCLKDIV_3 (3 << 5) +#define WM8510_MCLKDIV_4 (4 << 5) +#define WM8510_MCLKDIV_6 (5 << 5) +#define WM8510_MCLKDIV_8 (6 << 5) +#define WM8510_MCLKDIV_12 (7 << 5) + +struct wm8510_setup_data { + unsigned short i2c_address; +}; + +extern struct snd_soc_codec_dai wm8510_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8510; + +#endif -- cgit From f10485e79882bf09cf442a673511154987b43e3b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 5 Jun 2008 13:49:33 +0100 Subject: [ALSA] ASoC: Add WM8990 driver The WM8990 is a highly integrated ultra-low power hi-fi codec designed for handsets rich in multimedia features such as mobile TV, digital audio playback and gaming. The bulk of this driver was written by Liam Girdwood with some additional development and updates for new ASoC APIs by me. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8990.c | 1645 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8990.h | 832 +++++++++++++++++++++++ 4 files changed, 2482 insertions(+) create mode 100644 sound/soc/codecs/wm8990.c create mode 100644 sound/soc/codecs/wm8990.h (limited to 'sound/soc') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index ef400b36344..f439138db84 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -17,6 +17,9 @@ config SND_SOC_WM8750 config SND_SOC_WM8753 tristate +config SND_SOC_WM8990 + tristate + config SND_SOC_WM9712 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index b92c6654166..d9826f35d8c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -4,6 +4,7 @@ snd-soc-wm8510-objs := wm8510.o snd-soc-wm8731-objs := wm8731.o snd-soc-wm8750-objs := wm8750.o snd-soc-wm8753-objs := wm8753.o +snd-soc-wm8990-objs := wm8990.o snd-soc-wm9712-objs := wm9712.o snd-soc-wm9713-objs := wm9713.o snd-soc-cs4270-objs := cs4270.o @@ -15,6 +16,7 @@ obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o +obj-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c new file mode 100644 index 00000000000..a7d25e2f252 --- /dev/null +++ b/sound/soc/codecs/wm8990.c @@ -0,0 +1,1645 @@ +/* + * wm8990.c -- WM8990 ALSA Soc Audio driver + * + * Copyright 2008 Wolfson Microelectronics PLC. + * Author: Liam Girdwood + * lg@opensource.wolfsonmicro.com or linux@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 +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "wm8990.h" + +#define AUDIO_NAME "wm8990" +#define WM8990_VERSION "0.2" + +/* + * Debug + */ + +#define WM8990_DEBUG 0 + +#ifdef WM8990_DEBUG +#define dbg(format, arg...) \ + printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg) +#else +#define dbg(format, arg...) do {} while (0) +#endif +#define err(format, arg...) \ + printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg) +#define info(format, arg...) \ + printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg) +#define warn(format, arg...) \ + printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg) + +/* codec private data */ +struct wm8990_priv { + unsigned int sysclk; + unsigned int pcmclk; +}; + +/* + * wm8990 register cache. Note that register 0 is not included in the + * cache. + */ +static const u16 wm8990_reg[] = { + 0x8990, /* R0 - Reset */ + 0x0000, /* R1 - Power Management (1) */ + 0x6000, /* R2 - Power Management (2) */ + 0x0000, /* R3 - Power Management (3) */ + 0x4050, /* R4 - Audio Interface (1) */ + 0x4000, /* R5 - Audio Interface (2) */ + 0x01C8, /* R6 - Clocking (1) */ + 0x0000, /* R7 - Clocking (2) */ + 0x0040, /* R8 - Audio Interface (3) */ + 0x0040, /* R9 - Audio Interface (4) */ + 0x0004, /* R10 - DAC CTRL */ + 0x00C0, /* R11 - Left DAC Digital Volume */ + 0x00C0, /* R12 - Right DAC Digital Volume */ + 0x0000, /* R13 - Digital Side Tone */ + 0x0100, /* R14 - ADC CTRL */ + 0x00C0, /* R15 - Left ADC Digital Volume */ + 0x00C0, /* R16 - Right ADC Digital Volume */ + 0x0000, /* R17 */ + 0x0000, /* R18 - GPIO CTRL 1 */ + 0x1000, /* R19 - GPIO1 & GPIO2 */ + 0x1010, /* R20 - GPIO3 & GPIO4 */ + 0x1010, /* R21 - GPIO5 & GPIO6 */ + 0x8000, /* R22 - GPIOCTRL 2 */ + 0x0800, /* R23 - GPIO_POL */ + 0x008B, /* R24 - Left Line Input 1&2 Volume */ + 0x008B, /* R25 - Left Line Input 3&4 Volume */ + 0x008B, /* R26 - Right Line Input 1&2 Volume */ + 0x008B, /* R27 - Right Line Input 3&4 Volume */ + 0x0000, /* R28 - Left Output Volume */ + 0x0000, /* R29 - Right Output Volume */ + 0x0066, /* R30 - Line Outputs Volume */ + 0x0022, /* R31 - Out3/4 Volume */ + 0x0079, /* R32 - Left OPGA Volume */ + 0x0079, /* R33 - Right OPGA Volume */ + 0x0003, /* R34 - Speaker Volume */ + 0x0003, /* R35 - ClassD1 */ + 0x0000, /* R36 */ + 0x0100, /* R37 - ClassD3 */ + 0x0000, /* R38 */ + 0x0000, /* R39 - Input Mixer1 */ + 0x0000, /* R40 - Input Mixer2 */ + 0x0000, /* R41 - Input Mixer3 */ + 0x0000, /* R42 - Input Mixer4 */ + 0x0000, /* R43 - Input Mixer5 */ + 0x0000, /* R44 - Input Mixer6 */ + 0x0000, /* R45 - Output Mixer1 */ + 0x0000, /* R46 - Output Mixer2 */ + 0x0000, /* R47 - Output Mixer3 */ + 0x0000, /* R48 - Output Mixer4 */ + 0x0000, /* R49 - Output Mixer5 */ + 0x0000, /* R50 - Output Mixer6 */ + 0x0180, /* R51 - Out3/4 Mixer */ + 0x0000, /* R52 - Line Mixer1 */ + 0x0000, /* R53 - Line Mixer2 */ + 0x0000, /* R54 - Speaker Mixer */ + 0x0000, /* R55 - Additional Control */ + 0x0000, /* R56 - AntiPOP1 */ + 0x0000, /* R57 - AntiPOP2 */ + 0x0000, /* R58 - MICBIAS */ + 0x0000, /* R59 */ + 0x0008, /* R60 - PLL1 */ + 0x0031, /* R61 - PLL2 */ + 0x0026, /* R62 - PLL3 */ +}; + +/* + * read wm8990 register cache + */ +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); + return cache[reg]; +} + +/* + * write wm8990 register cache + */ +static inline void wm8990_write_reg_cache(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + BUG_ON(reg > (ARRAY_SIZE(wm8990_reg)) - 1); + + /* Reset register is uncached */ + if (reg == 0) + return; + + cache[reg] = value; +} + +/* + * write to the wm8990 register space + */ +static int wm8990_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[3]; + + data[0] = reg & 0xFF; + data[1] = (value >> 8) & 0xFF; + data[2] = value & 0xFF; + + wm8990_write_reg_cache(codec, reg, value); + + if (codec->hw_write(codec->control_data, data, 3) == 2) + return 0; + else + return -EIO; +} + +#define wm8990_reset(c) wm8990_write(c, WM8990_RESET, 0) + +static const DECLARE_TLV_DB_LINEAR(rec_mix_tlv, -1500, 600); + +static const DECLARE_TLV_DB_LINEAR(in_pga_tlv, -1650, 3000); + +static const DECLARE_TLV_DB_LINEAR(out_mix_tlv, 0, -2100); + +static const DECLARE_TLV_DB_LINEAR(out_pga_tlv, -7300, 600); + +static const DECLARE_TLV_DB_LINEAR(out_omix_tlv, -600, 0); + +static const DECLARE_TLV_DB_LINEAR(out_dac_tlv, -7163, 0); + +static const DECLARE_TLV_DB_LINEAR(in_adc_tlv, -7163, 1763); + +static const DECLARE_TLV_DB_LINEAR(out_sidetone_tlv, -3600, 0); + +static int wm899x_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int ret; + u16 val; + + ret = snd_soc_put_volsw(kcontrol, ucontrol); + if (ret < 0) + return ret; + + /* now hit the volume update bits (always bit 8) */ + val = wm8990_read_reg_cache(codec, reg); + return wm8990_write(codec, reg, val | 0x0100); +} + +#define SOC_WM899X_OUTPGA_SINGLE_R_TLV(xname, reg, shift, max, invert,\ + tlv_array) {\ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ + SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw, \ + .get = snd_soc_get_volsw, .put = wm899x_outpga_put_volsw_vu, \ + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } + + +static const char *wm8990_digital_sidetone[] = + {"None", "Left ADC", "Right ADC", "Reserved"}; + +static const struct soc_enum wm8990_left_digital_sidetone_enum = +SOC_ENUM_SINGLE(WM8990_DIGITAL_SIDE_TONE, + WM8990_ADC_TO_DACL_SHIFT, + WM8990_ADC_TO_DACL_MASK, + wm8990_digital_sidetone); + +static const struct soc_enum wm8990_right_digital_sidetone_enum = +SOC_ENUM_SINGLE(WM8990_DIGITAL_SIDE_TONE, + WM8990_ADC_TO_DACR_SHIFT, + WM8990_ADC_TO_DACR_MASK, + wm8990_digital_sidetone); + +static const char *wm8990_adcmode[] = + {"Hi-fi mode", "Voice mode 1", "Voice mode 2", "Voice mode 3"}; + +static const struct soc_enum wm8990_right_adcmode_enum = +SOC_ENUM_SINGLE(WM8990_ADC_CTRL, + WM8990_ADC_HPF_CUT_SHIFT, + WM8990_ADC_HPF_CUT_MASK, + wm8990_adcmode); + +static const struct snd_kcontrol_new wm8990_snd_controls[] = { +/* INMIXL */ +SOC_SINGLE("LIN12 PGA Boost", WM8990_INPUT_MIXER3, WM8990_L12MNBST_BIT, 1, 0), +SOC_SINGLE("LIN34 PGA Boost", WM8990_INPUT_MIXER3, WM8990_L34MNBST_BIT, 1, 0), +/* INMIXR */ +SOC_SINGLE("RIN12 PGA Boost", WM8990_INPUT_MIXER3, WM8990_R12MNBST_BIT, 1, 0), +SOC_SINGLE("RIN34 PGA Boost", WM8990_INPUT_MIXER3, WM8990_R34MNBST_BIT, 1, 0), + +/* LOMIX */ +SOC_SINGLE_TLV("LOMIX LIN3 Bypass Volume", WM8990_OUTPUT_MIXER3, + WM8990_LLI3LOVOL_SHIFT, WM8990_LLI3LOVOL_MASK, 1, out_mix_tlv), +SOC_SINGLE_TLV("LOMIX RIN12 PGA Bypass Volume", WM8990_OUTPUT_MIXER3, + WM8990_LR12LOVOL_SHIFT, WM8990_LR12LOVOL_MASK, 1, out_mix_tlv), +SOC_SINGLE_TLV("LOMIX LIN12 PGA Bypass Volume", WM8990_OUTPUT_MIXER3, + WM8990_LL12LOVOL_SHIFT, WM8990_LL12LOVOL_MASK, 1, out_mix_tlv), +SOC_SINGLE_TLV("LOMIX RIN3 Bypass Volume", WM8990_OUTPUT_MIXER5, + WM8990_LRI3LOVOL_SHIFT, WM8990_LRI3LOVOL_MASK, 1, out_mix_tlv), +SOC_SINGLE_TLV("LOMIX AINRMUX Bypass Volume", WM8990_OUTPUT_MIXER5, + WM8990_LRBLOVOL_SHIFT, WM8990_LRBLOVOL_MASK, 1, out_mix_tlv), +SOC_SINGLE_TLV("LOMIX AINLMUX Bypass Volume", WM8990_OUTPUT_MIXER5, + WM8990_LRBLOVOL_SHIFT, WM8990_LRBLOVOL_MASK, 1, out_mix_tlv), + +/* ROMIX */ +SOC_SINGLE_TLV("ROMIX RIN3 Bypass Volume", WM8990_OUTPUT_MIXER4, + WM8990_RRI3ROVOL_SHIFT, WM8990_RRI3ROVOL_MASK, 1, out_mix_tlv), +SOC_SINGLE_TLV("ROMIX LIN12 PGA Bypass Volume", WM8990_OUTPUT_MIXER4, + WM8990_RL12ROVOL_SHIFT, WM8990_RL12ROVOL_MASK, 1, out_mix_tlv), +SOC_SINGLE_TLV("ROMIX RIN12 PGA Bypass Volume", WM8990_OUTPUT_MIXER4, + WM8990_RR12ROVOL_SHIFT, WM8990_RR12ROVOL_MASK, 1, out_mix_tlv), +SOC_SINGLE_TLV("ROMIX LIN3 Bypass Volume", WM8990_OUTPUT_MIXER6, + WM8990_RLI3ROVOL_SHIFT, WM8990_RLI3ROVOL_MASK, 1, out_mix_tlv), +SOC_SINGLE_TLV("ROMIX AINLMUX Bypass Volume", WM8990_OUTPUT_MIXER6, + WM8990_RLBROVOL_SHIFT, WM8990_RLBROVOL_MASK, 1, out_mix_tlv), +SOC_SINGLE_TLV("ROMIX AINRMUX Bypass Volume", WM8990_OUTPUT_MIXER6, + WM8990_RRBROVOL_SHIFT, WM8990_RRBROVOL_MASK, 1, out_mix_tlv), + +/* LOUT */ +SOC_WM899X_OUTPGA_SINGLE_R_TLV("LOUT Volume", WM8990_LEFT_OUTPUT_VOLUME, + WM8990_LOUTVOL_SHIFT, WM8990_LOUTVOL_MASK, 0, out_pga_tlv), +SOC_SINGLE("LOUT ZC", WM8990_LEFT_OUTPUT_VOLUME, WM8990_LOZC_BIT, 1, 0), + +/* ROUT */ +SOC_WM899X_OUTPGA_SINGLE_R_TLV("ROUT Volume", WM8990_RIGHT_OUTPUT_VOLUME, + WM8990_ROUTVOL_SHIFT, WM8990_ROUTVOL_MASK, 0, out_pga_tlv), +SOC_SINGLE("ROUT ZC", WM8990_RIGHT_OUTPUT_VOLUME, WM8990_ROZC_BIT, 1, 0), + +/* LOPGA */ +SOC_WM899X_OUTPGA_SINGLE_R_TLV("LOPGA Volume", WM8990_LEFT_OPGA_VOLUME, + WM8990_LOPGAVOL_SHIFT, WM8990_LOPGAVOL_MASK, 0, out_pga_tlv), +SOC_SINGLE("LOPGA ZC Switch", WM8990_LEFT_OPGA_VOLUME, + WM8990_LOPGAZC_BIT, 1, 0), + +/* ROPGA */ +SOC_WM899X_OUTPGA_SINGLE_R_TLV("ROPGA Volume", WM8990_RIGHT_OPGA_VOLUME, + WM8990_ROPGAVOL_SHIFT, WM8990_ROPGAVOL_MASK, 0, out_pga_tlv), +SOC_SINGLE("ROPGA ZC Switch", WM8990_RIGHT_OPGA_VOLUME, + WM8990_ROPGAZC_BIT, 1, 0), + +SOC_SINGLE("LON Mute Switch", WM8990_LINE_OUTPUTS_VOLUME, + WM8990_LONMUTE_BIT, 1, 0), +SOC_SINGLE("LOP Mute Switch", WM8990_LINE_OUTPUTS_VOLUME, + WM8990_LOPMUTE_BIT, 1, 0), +SOC_SINGLE("LOP Attenuation Switch", WM8990_LINE_OUTPUTS_VOLUME, + WM8990_LOATTN_BIT, 1, 0), +SOC_SINGLE("RON Mute Switch", WM8990_LINE_OUTPUTS_VOLUME, + WM8990_RONMUTE_BIT, 1, 0), +SOC_SINGLE("ROP Mute Switch", WM8990_LINE_OUTPUTS_VOLUME, + WM8990_ROPMUTE_BIT, 1, 0), +SOC_SINGLE("ROP Attenuation Switch", WM8990_LINE_OUTPUTS_VOLUME, + WM8990_ROATTN_BIT, 1, 0), + +SOC_SINGLE("OUT3 Mute Switch", WM8990_OUT3_4_VOLUME, + WM8990_OUT3MUTE_BIT, 1, 0), +SOC_SINGLE("OUT3 Attenuation Switch", WM8990_OUT3_4_VOLUME, + WM8990_OUT3ATTN_BIT, 1, 0), + +SOC_SINGLE("OUT4 Mute Switch", WM8990_OUT3_4_VOLUME, + WM8990_OUT4MUTE_BIT, 1, 0), +SOC_SINGLE("OUT4 Attenuation Switch", WM8990_OUT3_4_VOLUME, + WM8990_OUT4ATTN_BIT, 1, 0), + +SOC_SINGLE("Speaker Mode Switch", WM8990_CLASSD1, + WM8990_CDMODE_BIT, 1, 0), + +SOC_SINGLE("Speaker Output Attenuation Volume", WM8990_SPEAKER_VOLUME, + WM8990_SPKVOL_SHIFT, WM8990_SPKVOL_MASK, 0), +SOC_SINGLE("Speaker DC Boost Volume", WM8990_CLASSD3, + WM8990_DCGAIN_SHIFT, WM8990_DCGAIN_MASK, 0), +SOC_SINGLE("Speaker AC Boost Volume", WM8990_CLASSD3, + WM8990_ACGAIN_SHIFT, WM8990_ACGAIN_MASK, 0), + +SOC_WM899X_OUTPGA_SINGLE_R_TLV("Left DAC Digital Volume", + WM8990_LEFT_DAC_DIGITAL_VOLUME, + WM8990_DACL_VOL_SHIFT, + WM8990_DACL_VOL_MASK, + 0, + out_dac_tlv), + +SOC_WM899X_OUTPGA_SINGLE_R_TLV("Right DAC Digital Volume", + WM8990_RIGHT_DAC_DIGITAL_VOLUME, + WM8990_DACR_VOL_SHIFT, + WM8990_DACR_VOL_MASK, + 0, + out_dac_tlv), + +SOC_ENUM("Left Digital Sidetone", wm8990_left_digital_sidetone_enum), +SOC_ENUM("Right Digital Sidetone", wm8990_right_digital_sidetone_enum), + +SOC_SINGLE_TLV("Left Digital Sidetone Volume", WM8990_DIGITAL_SIDE_TONE, + WM8990_ADCL_DAC_SVOL_SHIFT, WM8990_ADCL_DAC_SVOL_MASK, 0, + out_sidetone_tlv), +SOC_SINGLE_TLV("Right Digital Sidetone Volume", WM8990_DIGITAL_SIDE_TONE, + WM8990_ADCR_DAC_SVOL_SHIFT, WM8990_ADCR_DAC_SVOL_MASK, 0, + out_sidetone_tlv), + +SOC_SINGLE("ADC Digital High Pass Filter Switch", WM8990_ADC_CTRL, + WM8990_ADC_HPF_ENA_BIT, 1, 0), + +SOC_ENUM("ADC HPF Mode", wm8990_right_adcmode_enum), + +SOC_WM899X_OUTPGA_SINGLE_R_TLV("Left ADC Digital Volume", + WM8990_LEFT_ADC_DIGITAL_VOLUME, + WM8990_ADCL_VOL_SHIFT, + WM8990_ADCL_VOL_MASK, + 0, + in_adc_tlv), + +SOC_WM899X_OUTPGA_SINGLE_R_TLV("Right ADC Digital Volume", + WM8990_RIGHT_ADC_DIGITAL_VOLUME, + WM8990_ADCR_VOL_SHIFT, + WM8990_ADCR_VOL_MASK, + 0, + in_adc_tlv), + +SOC_WM899X_OUTPGA_SINGLE_R_TLV("LIN12 Volume", + WM8990_LEFT_LINE_INPUT_1_2_VOLUME, + WM8990_LIN12VOL_SHIFT, + WM8990_LIN12VOL_MASK, + 0, + in_pga_tlv), + +SOC_SINGLE("LIN12 ZC Switch", WM8990_LEFT_LINE_INPUT_1_2_VOLUME, + WM8990_LI12ZC_BIT, 1, 0), + +SOC_SINGLE("LIN12 Mute Switch", WM8990_LEFT_LINE_INPUT_1_2_VOLUME, + WM8990_LI12MUTE_BIT, 1, 0), + +SOC_WM899X_OUTPGA_SINGLE_R_TLV("LIN34 Volume", + WM8990_LEFT_LINE_INPUT_3_4_VOLUME, + WM8990_LIN34VOL_SHIFT, + WM8990_LIN34VOL_MASK, + 0, + in_pga_tlv), + +SOC_SINGLE("LIN34 ZC Switch", WM8990_LEFT_LINE_INPUT_3_4_VOLUME, + WM8990_LI34ZC_BIT, 1, 0), + +SOC_SINGLE("LIN34 Mute Switch", WM8990_LEFT_LINE_INPUT_3_4_VOLUME, + WM8990_LI34MUTE_BIT, 1, 0), + +SOC_WM899X_OUTPGA_SINGLE_R_TLV("RIN12 Volume", + WM8990_RIGHT_LINE_INPUT_1_2_VOLUME, + WM8990_RIN12VOL_SHIFT, + WM8990_RIN12VOL_MASK, + 0, + in_pga_tlv), + +SOC_SINGLE("RIN12 ZC Switch", WM8990_RIGHT_LINE_INPUT_1_2_VOLUME, + WM8990_RI12ZC_BIT, 1, 0), + +SOC_SINGLE("RIN12 Mute Switch", WM8990_RIGHT_LINE_INPUT_1_2_VOLUME, + WM8990_RI12MUTE_BIT, 1, 0), + +SOC_WM899X_OUTPGA_SINGLE_R_TLV("RIN34 Volume", + WM8990_RIGHT_LINE_INPUT_3_4_VOLUME, + WM8990_RIN34VOL_SHIFT, + WM8990_RIN34VOL_MASK, + 0, + in_pga_tlv), + +SOC_SINGLE("RIN34 ZC Switch", WM8990_RIGHT_LINE_INPUT_3_4_VOLUME, + WM8990_RI34ZC_BIT, 1, 0), + +SOC_SINGLE("RIN34 Mute Switch", WM8990_RIGHT_LINE_INPUT_3_4_VOLUME, + WM8990_RI34MUTE_BIT, 1, 0), + +}; + +/* 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 + */ + +static int inmixer_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + u16 reg, fakepower; + + reg = wm8990_read_reg_cache(w->codec, WM8990_POWER_MANAGEMENT_2); + fakepower = wm8990_read_reg_cache(w->codec, WM8990_INTDRIVBITS); + + if (fakepower & ((1 << WM8990_INMIXL_PWR_BIT) | + (1 << WM8990_AINLMUX_PWR_BIT))) { + reg |= WM8990_AINL_ENA; + } else { + reg &= ~WM8990_AINL_ENA; + } + + if (fakepower & ((1 << WM8990_INMIXR_PWR_BIT) | + (1 << WM8990_AINRMUX_PWR_BIT))) { + reg |= WM8990_AINR_ENA; + } else { + reg &= ~WM8990_AINL_ENA; + } + wm8990_write(w->codec, WM8990_POWER_MANAGEMENT_2, reg); + + return 0; +} + +static int outmixer_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + u32 reg_shift = kcontrol->private_value & 0xfff; + int ret = 0; + u16 reg; + + switch (reg_shift) { + case WM8990_SPEAKER_MIXER | (WM8990_LDSPK_BIT << 8) : + reg = wm8990_read_reg_cache(w->codec, WM8990_OUTPUT_MIXER1); + if (reg & WM8990_LDLO) { + printk(KERN_WARNING + "Cannot set as Output Mixer 1 LDLO Set\n"); + ret = -1; + } + break; + case WM8990_SPEAKER_MIXER | (WM8990_RDSPK_BIT << 8): + reg = wm8990_read_reg_cache(w->codec, WM8990_OUTPUT_MIXER2); + if (reg & WM8990_RDRO) { + printk(KERN_WARNING + "Cannot set as Output Mixer 2 RDRO Set\n"); + ret = -1; + } + break; + case WM8990_OUTPUT_MIXER1 | (WM8990_LDLO_BIT << 8): + reg = wm8990_read_reg_cache(w->codec, WM8990_SPEAKER_MIXER); + if (reg & WM8990_LDSPK) { + printk(KERN_WARNING + "Cannot set as Speaker Mixer LDSPK Set\n"); + ret = -1; + } + break; + case WM8990_OUTPUT_MIXER2 | (WM8990_RDRO_BIT << 8): + reg = wm8990_read_reg_cache(w->codec, WM8990_SPEAKER_MIXER); + if (reg & WM8990_RDSPK) { + printk(KERN_WARNING + "Cannot set as Speaker Mixer RDSPK Set\n"); + ret = -1; + } + break; + } + + return ret; +} + +/* INMIX dB values */ +static const unsigned int in_mix_tlv[] = { + TLV_DB_RANGE_HEAD(1), + 0, 7, TLV_DB_LINEAR_ITEM(-1200, 600), +}; + +/* Left In PGA Connections */ +static const struct snd_kcontrol_new wm8990_dapm_lin12_pga_controls[] = { +SOC_DAPM_SINGLE("LIN1 Switch", WM8990_INPUT_MIXER2, WM8990_LMN1_BIT, 1, 0), +SOC_DAPM_SINGLE("LIN2 Switch", WM8990_INPUT_MIXER2, WM8990_LMP2_BIT, 1, 0), +}; + +static const struct snd_kcontrol_new wm8990_dapm_lin34_pga_controls[] = { +SOC_DAPM_SINGLE("LIN3 Switch", WM8990_INPUT_MIXER2, WM8990_LMN3_BIT, 1, 0), +SOC_DAPM_SINGLE("LIN4 Switch", WM8990_INPUT_MIXER2, WM8990_LMP4_BIT, 1, 0), +}; + +/* Right In PGA Connections */ +static const struct snd_kcontrol_new wm8990_dapm_rin12_pga_controls[] = { +SOC_DAPM_SINGLE("RIN1 Switch", WM8990_INPUT_MIXER2, WM8990_RMN1_BIT, 1, 0), +SOC_DAPM_SINGLE("RIN2 Switch", WM8990_INPUT_MIXER2, WM8990_RMP2_BIT, 1, 0), +}; + +static const struct snd_kcontrol_new wm8990_dapm_rin34_pga_controls[] = { +SOC_DAPM_SINGLE("RIN3 Switch", WM8990_INPUT_MIXER2, WM8990_RMN3_BIT, 1, 0), +SOC_DAPM_SINGLE("RIN4 Switch", WM8990_INPUT_MIXER2, WM8990_RMP4_BIT, 1, 0), +}; + +/* INMIXL */ +static const struct snd_kcontrol_new wm8990_dapm_inmixl_controls[] = { +SOC_DAPM_SINGLE_TLV("Record Left Volume", WM8990_INPUT_MIXER3, + WM8990_LDBVOL_SHIFT, WM8990_LDBVOL_MASK, 0, in_mix_tlv), +SOC_DAPM_SINGLE_TLV("LIN2 Volume", WM8990_INPUT_MIXER5, WM8990_LI2BVOL_SHIFT, + 7, 0, in_mix_tlv), +SOC_DAPM_SINGLE("LINPGA12 Switch", WM8990_INPUT_MIXER3, WM8990_L12MNB_BIT, + 1, 0), +SOC_DAPM_SINGLE("LINPGA34 Switch", WM8990_INPUT_MIXER3, WM8990_L34MNB_BIT, + 1, 0), +}; + +/* INMIXR */ +static const struct snd_kcontrol_new wm8990_dapm_inmixr_controls[] = { +SOC_DAPM_SINGLE_TLV("Record Right Volume", WM8990_INPUT_MIXER4, + WM8990_RDBVOL_SHIFT, WM8990_RDBVOL_MASK, 0, in_mix_tlv), +SOC_DAPM_SINGLE_TLV("RIN2 Volume", WM8990_INPUT_MIXER6, WM8990_RI2BVOL_SHIFT, + 7, 0, in_mix_tlv), +SOC_DAPM_SINGLE("RINPGA12 Switch", WM8990_INPUT_MIXER3, WM8990_L12MNB_BIT, + 1, 0), +SOC_DAPM_SINGLE("RINPGA34 Switch", WM8990_INPUT_MIXER3, WM8990_L34MNB_BIT, + 1, 0), +}; + +/* AINLMUX */ +static const char *wm8990_ainlmux[] = + {"INMIXL Mix", "RXVOICE Mix", "DIFFINL Mix"}; + +static const struct soc_enum wm8990_ainlmux_enum = +SOC_ENUM_SINGLE(WM8990_INPUT_MIXER1, WM8990_AINLMODE_SHIFT, + ARRAY_SIZE(wm8990_ainlmux), wm8990_ainlmux); + +static const struct snd_kcontrol_new wm8990_dapm_ainlmux_controls = +SOC_DAPM_ENUM("Route", wm8990_ainlmux_enum); + +/* DIFFINL */ + +/* AINRMUX */ +static const char *wm8990_ainrmux[] = + {"INMIXR Mix", "RXVOICE Mix", "DIFFINR Mix"}; + +static const struct soc_enum wm8990_ainrmux_enum = +SOC_ENUM_SINGLE(WM8990_INPUT_MIXER1, WM8990_AINRMODE_SHIFT, + ARRAY_SIZE(wm8990_ainrmux), wm8990_ainrmux); + +static const struct snd_kcontrol_new wm8990_dapm_ainrmux_controls = +SOC_DAPM_ENUM("Route", wm8990_ainrmux_enum); + +/* RXVOICE */ +static const struct snd_kcontrol_new wm8990_dapm_rxvoice_controls[] = { +SOC_DAPM_SINGLE_TLV("LIN4/RXN", WM8990_INPUT_MIXER5, WM8990_LR4BVOL_SHIFT, + WM8990_LR4BVOL_MASK, 0, in_mix_tlv), +SOC_DAPM_SINGLE_TLV("RIN4/RXP", WM8990_INPUT_MIXER6, WM8990_RL4BVOL_SHIFT, + WM8990_RL4BVOL_MASK, 0, in_mix_tlv), +}; + +/* LOMIX */ +static const struct snd_kcontrol_new wm8990_dapm_lomix_controls[] = { +SOC_DAPM_SINGLE("LOMIX Right ADC Bypass Switch", WM8990_OUTPUT_MIXER1, + WM8990_LRBLO_BIT, 1, 0), +SOC_DAPM_SINGLE("LOMIX Left ADC Bypass Switch", WM8990_OUTPUT_MIXER1, + WM8990_LLBLO_BIT, 1, 0), +SOC_DAPM_SINGLE("LOMIX RIN3 Bypass Switch", WM8990_OUTPUT_MIXER1, + WM8990_LRI3LO_BIT, 1, 0), +SOC_DAPM_SINGLE("LOMIX LIN3 Bypass Switch", WM8990_OUTPUT_MIXER1, + WM8990_LLI3LO_BIT, 1, 0), +SOC_DAPM_SINGLE("LOMIX RIN12 PGA Bypass Switch", WM8990_OUTPUT_MIXER1, + WM8990_LR12LO_BIT, 1, 0), +SOC_DAPM_SINGLE("LOMIX LIN12 PGA Bypass Switch", WM8990_OUTPUT_MIXER1, + WM8990_LL12LO_BIT, 1, 0), +SOC_DAPM_SINGLE("LOMIX Left DAC Switch", WM8990_OUTPUT_MIXER1, + WM8990_LDLO_BIT, 1, 0), +}; + +/* ROMIX */ +static const struct snd_kcontrol_new wm8990_dapm_romix_controls[] = { +SOC_DAPM_SINGLE("ROMIX Left ADC Bypass Switch", WM8990_OUTPUT_MIXER2, + WM8990_RLBRO_BIT, 1, 0), +SOC_DAPM_SINGLE("ROMIX Right ADC Bypass Switch", WM8990_OUTPUT_MIXER2, + WM8990_RRBRO_BIT, 1, 0), +SOC_DAPM_SINGLE("ROMIX LIN3 Bypass Switch", WM8990_OUTPUT_MIXER2, + WM8990_RLI3RO_BIT, 1, 0), +SOC_DAPM_SINGLE("ROMIX RIN3 Bypass Switch", WM8990_OUTPUT_MIXER2, + WM8990_RRI3RO_BIT, 1, 0), +SOC_DAPM_SINGLE("ROMIX LIN12 PGA Bypass Switch", WM8990_OUTPUT_MIXER2, + WM8990_RL12RO_BIT, 1, 0), +SOC_DAPM_SINGLE("ROMIX RIN12 PGA Bypass Switch", WM8990_OUTPUT_MIXER2, + WM8990_RR12RO_BIT, 1, 0), +SOC_DAPM_SINGLE("ROMIX Right DAC Switch", WM8990_OUTPUT_MIXER2, + WM8990_RDRO_BIT, 1, 0), +}; + +/* LONMIX */ +static const struct snd_kcontrol_new wm8990_dapm_lonmix_controls[] = { +SOC_DAPM_SINGLE("LONMIX Left Mixer PGA Switch", WM8990_LINE_MIXER1, + WM8990_LLOPGALON_BIT, 1, 0), +SOC_DAPM_SINGLE("LONMIX Right Mixer PGA Switch", WM8990_LINE_MIXER1, + WM8990_LROPGALON_BIT, 1, 0), +SOC_DAPM_SINGLE("LONMIX Inverted LOP Switch", WM8990_LINE_MIXER1, + WM8990_LOPLON_BIT, 1, 0), +}; + +/* LOPMIX */ +static const struct snd_kcontrol_new wm8990_dapm_lopmix_controls[] = { +SOC_DAPM_SINGLE("LOPMIX Right Mic Bypass Switch", WM8990_LINE_MIXER1, + WM8990_LR12LOP_BIT, 1, 0), +SOC_DAPM_SINGLE("LOPMIX Left Mic Bypass Switch", WM8990_LINE_MIXER1, + WM8990_LL12LOP_BIT, 1, 0), +SOC_DAPM_SINGLE("LOPMIX Left Mixer PGA Switch", WM8990_LINE_MIXER1, + WM8990_LLOPGALOP_BIT, 1, 0), +}; + +/* RONMIX */ +static const struct snd_kcontrol_new wm8990_dapm_ronmix_controls[] = { +SOC_DAPM_SINGLE("RONMIX Right Mixer PGA Switch", WM8990_LINE_MIXER2, + WM8990_RROPGARON_BIT, 1, 0), +SOC_DAPM_SINGLE("RONMIX Left Mixer PGA Switch", WM8990_LINE_MIXER2, + WM8990_RLOPGARON_BIT, 1, 0), +SOC_DAPM_SINGLE("RONMIX Inverted ROP Switch", WM8990_LINE_MIXER2, + WM8990_ROPRON_BIT, 1, 0), +}; + +/* ROPMIX */ +static const struct snd_kcontrol_new wm8990_dapm_ropmix_controls[] = { +SOC_DAPM_SINGLE("ROPMIX Left Mic Bypass Switch", WM8990_LINE_MIXER2, + WM8990_RL12ROP_BIT, 1, 0), +SOC_DAPM_SINGLE("ROPMIX Right Mic Bypass Switch", WM8990_LINE_MIXER2, + WM8990_RR12ROP_BIT, 1, 0), +SOC_DAPM_SINGLE("ROPMIX Right Mixer PGA Switch", WM8990_LINE_MIXER2, + WM8990_RROPGAROP_BIT, 1, 0), +}; + +/* OUT3MIX */ +static const struct snd_kcontrol_new wm8990_dapm_out3mix_controls[] = { +SOC_DAPM_SINGLE("OUT3MIX LIN4/RXP Bypass Switch", WM8990_OUT3_4_MIXER, + WM8990_LI4O3_BIT, 1, 0), +SOC_DAPM_SINGLE("OUT3MIX Left Out PGA Switch", WM8990_OUT3_4_MIXER, + WM8990_LPGAO3_BIT, 1, 0), +}; + +/* OUT4MIX */ +static const struct snd_kcontrol_new wm8990_dapm_out4mix_controls[] = { +SOC_DAPM_SINGLE("OUT4MIX Right Out PGA Switch", WM8990_OUT3_4_MIXER, + WM8990_RPGAO4_BIT, 1, 0), +SOC_DAPM_SINGLE("OUT4MIX RIN4/RXP Bypass Switch", WM8990_OUT3_4_MIXER, + WM8990_RI4O4_BIT, 1, 0), +}; + +/* SPKMIX */ +static const struct snd_kcontrol_new wm8990_dapm_spkmix_controls[] = { +SOC_DAPM_SINGLE("SPKMIX LIN2 Bypass Switch", WM8990_SPEAKER_MIXER, + WM8990_LI2SPK_BIT, 1, 0), +SOC_DAPM_SINGLE("SPKMIX LADC Bypass Switch", WM8990_SPEAKER_MIXER, + WM8990_LB2SPK_BIT, 1, 0), +SOC_DAPM_SINGLE("SPKMIX Left Mixer PGA Switch", WM8990_SPEAKER_MIXER, + WM8990_LOPGASPK_BIT, 1, 0), +SOC_DAPM_SINGLE("SPKMIX Left DAC Switch", WM8990_SPEAKER_MIXER, + WM8990_LDSPK_BIT, 1, 0), +SOC_DAPM_SINGLE("SPKMIX Right DAC Switch", WM8990_SPEAKER_MIXER, + WM8990_RDSPK_BIT, 1, 0), +SOC_DAPM_SINGLE("SPKMIX Right Mixer PGA Switch", WM8990_SPEAKER_MIXER, + WM8990_ROPGASPK_BIT, 1, 0), +SOC_DAPM_SINGLE("SPKMIX RADC Bypass Switch", WM8990_SPEAKER_MIXER, + WM8990_RL12ROP_BIT, 1, 0), +SOC_DAPM_SINGLE("SPKMIX RIN2 Bypass Switch", WM8990_SPEAKER_MIXER, + WM8990_RI2SPK_BIT, 1, 0), +}; + +static const struct snd_soc_dapm_widget wm8990_dapm_widgets[] = { +/* Input Side */ +/* Input Lines */ +SND_SOC_DAPM_INPUT("LIN1"), +SND_SOC_DAPM_INPUT("LIN2"), +SND_SOC_DAPM_INPUT("LIN3"), +SND_SOC_DAPM_INPUT("LIN4/RXN"), +SND_SOC_DAPM_INPUT("RIN3"), +SND_SOC_DAPM_INPUT("RIN4/RXP"), +SND_SOC_DAPM_INPUT("RIN1"), +SND_SOC_DAPM_INPUT("RIN2"), +SND_SOC_DAPM_INPUT("Internal ADC Source"), + +/* DACs */ +SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8990_POWER_MANAGEMENT_2, + WM8990_ADCL_ENA_BIT, 0), +SND_SOC_DAPM_ADC("Right ADC", "Right Capture", WM8990_POWER_MANAGEMENT_2, + WM8990_ADCR_ENA_BIT, 0), + +/* Input PGAs */ +SND_SOC_DAPM_MIXER("LIN12 PGA", WM8990_POWER_MANAGEMENT_2, WM8990_LIN12_ENA_BIT, + 0, &wm8990_dapm_lin12_pga_controls[0], + ARRAY_SIZE(wm8990_dapm_lin12_pga_controls)), +SND_SOC_DAPM_MIXER("LIN34 PGA", WM8990_POWER_MANAGEMENT_2, WM8990_LIN34_ENA_BIT, + 0, &wm8990_dapm_lin34_pga_controls[0], + ARRAY_SIZE(wm8990_dapm_lin34_pga_controls)), +SND_SOC_DAPM_MIXER("RIN12 PGA", WM8990_POWER_MANAGEMENT_2, WM8990_RIN12_ENA_BIT, + 0, &wm8990_dapm_rin12_pga_controls[0], + ARRAY_SIZE(wm8990_dapm_rin12_pga_controls)), +SND_SOC_DAPM_MIXER("RIN34 PGA", WM8990_POWER_MANAGEMENT_2, WM8990_RIN34_ENA_BIT, + 0, &wm8990_dapm_rin34_pga_controls[0], + ARRAY_SIZE(wm8990_dapm_rin34_pga_controls)), + +/* INMIXL */ +SND_SOC_DAPM_MIXER_E("INMIXL", WM8990_INTDRIVBITS, WM8990_INMIXL_PWR_BIT, 0, + &wm8990_dapm_inmixl_controls[0], + ARRAY_SIZE(wm8990_dapm_inmixl_controls), + inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + +/* AINLMUX */ +SND_SOC_DAPM_MUX_E("AILNMUX", WM8990_INTDRIVBITS, WM8990_AINLMUX_PWR_BIT, 0, + &wm8990_dapm_ainlmux_controls, inmixer_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + +/* INMIXR */ +SND_SOC_DAPM_MIXER_E("INMIXR", WM8990_INTDRIVBITS, WM8990_INMIXR_PWR_BIT, 0, + &wm8990_dapm_inmixr_controls[0], + ARRAY_SIZE(wm8990_dapm_inmixr_controls), + inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + +/* AINRMUX */ +SND_SOC_DAPM_MUX_E("AIRNMUX", WM8990_INTDRIVBITS, WM8990_AINRMUX_PWR_BIT, 0, + &wm8990_dapm_ainrmux_controls, inmixer_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + +/* Output Side */ +/* DACs */ +SND_SOC_DAPM_DAC("Left DAC", "Left Playback", WM8990_POWER_MANAGEMENT_3, + WM8990_DACL_ENA_BIT, 0), +SND_SOC_DAPM_DAC("Right DAC", "Right Playback", WM8990_POWER_MANAGEMENT_3, + WM8990_DACR_ENA_BIT, 0), + +/* LOMIX */ +SND_SOC_DAPM_MIXER_E("LOMIX", WM8990_POWER_MANAGEMENT_3, WM8990_LOMIX_ENA_BIT, + 0, &wm8990_dapm_lomix_controls[0], + ARRAY_SIZE(wm8990_dapm_lomix_controls), + outmixer_event, SND_SOC_DAPM_PRE_REG), + +/* LONMIX */ +SND_SOC_DAPM_MIXER("LONMIX", WM8990_POWER_MANAGEMENT_3, WM8990_LON_ENA_BIT, 0, + &wm8990_dapm_lonmix_controls[0], + ARRAY_SIZE(wm8990_dapm_lonmix_controls)), + +/* LOPMIX */ +SND_SOC_DAPM_MIXER("LOPMIX", WM8990_POWER_MANAGEMENT_3, WM8990_LOP_ENA_BIT, 0, + &wm8990_dapm_lopmix_controls[0], + ARRAY_SIZE(wm8990_dapm_lopmix_controls)), + +/* OUT3MIX */ +SND_SOC_DAPM_MIXER("OUT3MIX", WM8990_POWER_MANAGEMENT_1, WM8990_OUT3_ENA_BIT, 0, + &wm8990_dapm_out3mix_controls[0], + ARRAY_SIZE(wm8990_dapm_out3mix_controls)), + +/* SPKMIX */ +SND_SOC_DAPM_MIXER_E("SPKMIX", WM8990_POWER_MANAGEMENT_1, WM8990_SPK_ENA_BIT, 0, + &wm8990_dapm_spkmix_controls[0], + ARRAY_SIZE(wm8990_dapm_spkmix_controls), outmixer_event, + SND_SOC_DAPM_PRE_REG), + +/* OUT4MIX */ +SND_SOC_DAPM_MIXER("OUT4MIX", WM8990_POWER_MANAGEMENT_1, WM8990_OUT4_ENA_BIT, 0, + &wm8990_dapm_out4mix_controls[0], + ARRAY_SIZE(wm8990_dapm_out4mix_controls)), + +/* ROPMIX */ +SND_SOC_DAPM_MIXER("ROPMIX", WM8990_POWER_MANAGEMENT_3, WM8990_ROP_ENA_BIT, 0, + &wm8990_dapm_ropmix_controls[0], + ARRAY_SIZE(wm8990_dapm_ropmix_controls)), + +/* RONMIX */ +SND_SOC_DAPM_MIXER("RONMIX", WM8990_POWER_MANAGEMENT_3, WM8990_RON_ENA_BIT, 0, + &wm8990_dapm_ronmix_controls[0], + ARRAY_SIZE(wm8990_dapm_ronmix_controls)), + +/* ROMIX */ +SND_SOC_DAPM_MIXER_E("ROMIX", WM8990_POWER_MANAGEMENT_3, WM8990_ROMIX_ENA_BIT, + 0, &wm8990_dapm_romix_controls[0], + ARRAY_SIZE(wm8990_dapm_romix_controls), + outmixer_event, SND_SOC_DAPM_PRE_REG), + +/* LOUT PGA */ +SND_SOC_DAPM_PGA("LOUT PGA", WM8990_POWER_MANAGEMENT_1, WM8990_LOUT_ENA_BIT, 0, + NULL, 0), + +/* ROUT PGA */ +SND_SOC_DAPM_PGA("ROUT PGA", WM8990_POWER_MANAGEMENT_1, WM8990_ROUT_ENA_BIT, 0, + NULL, 0), + +/* LOPGA */ +SND_SOC_DAPM_PGA("LOPGA", WM8990_POWER_MANAGEMENT_3, WM8990_LOPGA_ENA_BIT, 0, + NULL, 0), + +/* ROPGA */ +SND_SOC_DAPM_PGA("ROPGA", WM8990_POWER_MANAGEMENT_3, WM8990_ROPGA_ENA_BIT, 0, + NULL, 0), + +/* MICBIAS */ +SND_SOC_DAPM_MICBIAS("MICBIAS", WM8990_POWER_MANAGEMENT_1, + WM8990_MICBIAS_ENA_BIT, 0), + +SND_SOC_DAPM_OUTPUT("LON"), +SND_SOC_DAPM_OUTPUT("LOP"), +SND_SOC_DAPM_OUTPUT("OUT3"), +SND_SOC_DAPM_OUTPUT("LOUT"), +SND_SOC_DAPM_OUTPUT("SPKN"), +SND_SOC_DAPM_OUTPUT("SPKP"), +SND_SOC_DAPM_OUTPUT("ROUT"), +SND_SOC_DAPM_OUTPUT("OUT4"), +SND_SOC_DAPM_OUTPUT("ROP"), +SND_SOC_DAPM_OUTPUT("RON"), + +SND_SOC_DAPM_OUTPUT("Internal DAC Sink"), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* Make DACs turn on when playing even if not mixed into any outputs */ + {"Internal DAC Sink", NULL, "Left DAC"}, + {"Internal DAC Sink", NULL, "Right DAC"}, + + /* Make ADCs turn on when recording even if not mixed from any inputs */ + {"Left ADC", NULL, "Internal ADC Source"}, + {"Right ADC", NULL, "Internal ADC Source"}, + + /* Input Side */ + /* LIN12 PGA */ + {"LIN12 PGA", "LIN1 Switch", "LIN1"}, + {"LIN12 PGA", "LIN2 Switch", "LIN2"}, + /* LIN34 PGA */ + {"LIN34 PGA", "LIN3 Switch", "LIN3"}, + {"LIN34 PGA", "LIN4 Switch", "LIN4"}, + /* INMIXL */ + {"INMIXL", "Record Left Volume", "LOMIX"}, + {"INMIXL", "LIN2 Volume", "LIN2"}, + {"INMIXL", "LINPGA12 Switch", "LIN12 PGA"}, + {"INMIXL", "LINPGA34 Switch", "LIN34 PGA"}, + /* AILNMUX */ + {"AILNMUX", "INMIXL Mix", "INMIXL"}, + {"AILNMUX", "DIFFINL Mix", "LIN12PGA"}, + {"AILNMUX", "DIFFINL Mix", "LIN34PGA"}, + {"AILNMUX", "RXVOICE Mix", "LIN4/RXN"}, + {"AILNMUX", "RXVOICE Mix", "RIN4/RXP"}, + /* ADC */ + {"Left ADC", NULL, "AILNMUX"}, + + /* RIN12 PGA */ + {"RIN12 PGA", "RIN1 Switch", "RIN1"}, + {"RIN12 PGA", "RIN2 Switch", "RIN2"}, + /* RIN34 PGA */ + {"RIN34 PGA", "RIN3 Switch", "RIN3"}, + {"RIN34 PGA", "RIN4 Switch", "RIN4"}, + /* INMIXL */ + {"INMIXR", "Record Right Volume", "ROMIX"}, + {"INMIXR", "RIN2 Volume", "RIN2"}, + {"INMIXR", "RINPGA12 Switch", "RIN12 PGA"}, + {"INMIXR", "RINPGA34 Switch", "RIN34 PGA"}, + /* AIRNMUX */ + {"AIRNMUX", "INMIXR Mix", "INMIXR"}, + {"AIRNMUX", "DIFFINR Mix", "RIN12PGA"}, + {"AIRNMUX", "DIFFINR Mix", "RIN34PGA"}, + {"AIRNMUX", "RXVOICE Mix", "RIN4/RXN"}, + {"AIRNMUX", "RXVOICE Mix", "RIN4/RXP"}, + /* ADC */ + {"Right ADC", NULL, "AIRNMUX"}, + + /* LOMIX */ + {"LOMIX", "LOMIX RIN3 Bypass Switch", "RIN3"}, + {"LOMIX", "LOMIX LIN3 Bypass Switch", "LIN3"}, + {"LOMIX", "LOMIX LIN12 PGA Bypass Switch", "LIN12 PGA"}, + {"LOMIX", "LOMIX RIN12 PGA Bypass Switch", "RIN12 PGA"}, + {"LOMIX", "LOMIX Right ADC Bypass Switch", "AINRMUX"}, + {"LOMIX", "LOMIX Left ADC Bypass Switch", "AINLMUX"}, + {"LOMIX", "LOMIX Left DAC Switch", "Left DAC"}, + + /* ROMIX */ + {"ROMIX", "ROMIX RIN3 Bypass Switch", "RIN3"}, + {"ROMIX", "ROMIX LIN3 Bypass Switch", "LIN3"}, + {"ROMIX", "ROMIX LIN12 PGA Bypass Switch", "LIN12 PGA"}, + {"ROMIX", "ROMIX RIN12 PGA Bypass Switch", "RIN12 PGA"}, + {"ROMIX", "ROMIX Right ADC Bypass Switch", "AINRMUX"}, + {"ROMIX", "ROMIX Left ADC Bypass Switch", "AINLMUX"}, + {"ROMIX", "ROMIX Right DAC Switch", "Right DAC"}, + + /* SPKMIX */ + {"SPKMIX", "SPKMIX LIN2 Bypass Switch", "LIN2"}, + {"SPKMIX", "SPKMIX RIN2 Bypass Switch", "RIN2"}, + {"SPKMIX", "SPKMIX LADC Bypass Switch", "AINLMUX"}, + {"SPKMIX", "SPKMIX RADC Bypass Switch", "AINRMUX"}, + {"SPKMIX", "SPKMIX Left Mixer PGA Switch", "LOPGA"}, + {"SPKMIX", "SPKMIX Right Mixer PGA Switch", "ROPGA"}, + {"SPKMIX", "SPKMIX Right DAC Switch", "Right DAC"}, + {"SPKMIX", "SPKMIX Left DAC Switch", "Right DAC"}, + + /* LONMIX */ + {"LONMIX", "LONMIX Left Mixer PGA Switch", "LOPGA"}, + {"LONMIX", "LONMIX Right Mixer PGA Switch", "ROPGA"}, + {"LONMIX", "LONMIX Inverted LOP Switch", "LOPMIX"}, + + /* LOPMIX */ + {"LOPMIX", "LOPMIX Right Mic Bypass Switch", "RIN12 PGA"}, + {"LOPMIX", "LOPMIX Left Mic Bypass Switch", "LIN12 PGA"}, + {"LOPMIX", "LOPMIX Left Mixer PGA Switch", "LOPGA"}, + + /* OUT3MIX */ + {"OUT3MIX", "OUT3MIX LIN4/RXP Bypass Switch", "LIN4/RXP"}, + {"OUT3MIX", "OUT3MIX Left Out PGA Switch", "LOPGA"}, + + /* OUT4MIX */ + {"OUT4MIX", "OUT4MIX Right Out PGA Switch", "ROPGA"}, + {"OUT4MIX", "OUT4MIX RIN4/RXP Bypass Switch", "RIN4/RXP"}, + + /* RONMIX */ + {"RONMIX", "RONMIX Right Mixer PGA Switch", "ROPGA"}, + {"RONMIX", "RONMIX Left Mixer PGA Switch", "LOPGA"}, + {"RONMIX", "RONMIX Inverted ROP Switch", "ROPMIX"}, + + /* ROPMIX */ + {"ROPMIX", "ROPMIX Left Mic Bypass Switch", "LIN12 PGA"}, + {"ROPMIX", "ROPMIX Right Mic Bypass Switch", "RIN12 PGA"}, + {"ROPMIX", "ROPMIX Right Mixer PGA Switch", "ROPGA"}, + + /* Out Mixer PGAs */ + {"LOPGA", NULL, "LOMIX"}, + {"ROPGA", NULL, "ROMIX"}, + + {"LOUT PGA", NULL, "LOMIX"}, + {"ROUT PGA", NULL, "ROMIX"}, + + /* Output Pins */ + {"LON", NULL, "LONMIX"}, + {"LOP", NULL, "LOPMIX"}, + {"OUT", NULL, "OUT3MIX"}, + {"LOUT", NULL, "LOUT PGA"}, + {"SPKN", NULL, "SPKMIX"}, + {"ROUT", NULL, "ROUT PGA"}, + {"OUT4", NULL, "OUT4MIX"}, + {"ROP", NULL, "ROPMIX"}, + {"RON", NULL, "RONMIX"}, +}; + +static int wm8990_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, wm8990_dapm_widgets, + ARRAY_SIZE(wm8990_dapm_widgets)); + + /* set up the WM8990 audio map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +/* PLL divisors */ +struct _pll_div { + u32 div2; + u32 n; + u32 k; +}; + +/* The size in bits of the pll divide multiplied by 10 + * to allow rounding later */ +#define FIXED_PLL_SIZE ((1 << 16) * 10) + +static void pll_factors(struct _pll_div *pll_div, unsigned int target, + unsigned int source) +{ + u64 Kpart; + unsigned int K, Ndiv, Nmod; + + + Ndiv = target / source; + if (Ndiv < 6) { + source >>= 1; + pll_div->div2 = 1; + Ndiv = target / source; + } else + pll_div->div2 = 0; + + if ((Ndiv < 6) || (Ndiv > 12)) + printk(KERN_WARNING + "WM8990 N value outwith recommended range! N = %d\n", Ndiv); + + pll_div->n = Ndiv; + Nmod = target % source; + Kpart = FIXED_PLL_SIZE * (long long)Nmod; + + do_div(Kpart, source); + + K = Kpart & 0xFFFFFFFF; + + /* Check if we need to round */ + if ((K % 10) >= 5) + K += 5; + + /* Move down to proper range now rounding is done */ + K /= 10; + + pll_div->k = K; +} + +static int wm8990_set_dai_pll(struct snd_soc_codec_dai *codec_dai, + int pll_id, unsigned int freq_in, unsigned int freq_out) +{ + u16 reg; + struct snd_soc_codec *codec = codec_dai->codec; + struct _pll_div pll_div; + + if (freq_in && freq_out) { + pll_factors(&pll_div, freq_out * 4, freq_in); + + /* Turn on PLL */ + reg = wm8990_read_reg_cache(codec, WM8990_POWER_MANAGEMENT_2); + reg |= WM8990_PLL_ENA; + wm8990_write(codec, WM8990_POWER_MANAGEMENT_2, reg); + + /* sysclk comes from PLL */ + reg = wm8990_read_reg_cache(codec, WM8990_CLOCKING_2); + wm8990_write(codec, WM8990_CLOCKING_2, reg | WM8990_SYSCLK_SRC); + + /* set up N , fractional mode and pre-divisor if neccessary */ + wm8990_write(codec, WM8990_PLL1, pll_div.n | WM8990_SDM | + (pll_div.div2?WM8990_PRESCALE:0)); + wm8990_write(codec, WM8990_PLL2, (u8)(pll_div.k>>8)); + wm8990_write(codec, WM8990_PLL3, (u8)(pll_div.k & 0xFF)); + } else { + /* Turn on PLL */ + reg = wm8990_read_reg_cache(codec, WM8990_POWER_MANAGEMENT_2); + reg &= ~WM8990_PLL_ENA; + wm8990_write(codec, WM8990_POWER_MANAGEMENT_2, reg); + } + return 0; +} + +/* + * Clock after PLL and dividers + */ +static int wm8990_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8990_priv *wm8990 = codec->private_data; + + wm8990->sysclk = freq; + return 0; +} + +/* + * Set's ADC and Voice DAC format. + */ +static int wm8990_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 audio1, audio3; + + audio1 = wm8990_read_reg_cache(codec, WM8990_AUDIO_INTERFACE_1); + audio3 = wm8990_read_reg_cache(codec, WM8990_AUDIO_INTERFACE_3); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + audio3 &= ~WM8990_AIF_MSTR1; + break; + case SND_SOC_DAIFMT_CBM_CFM: + audio3 |= WM8990_AIF_MSTR1; + break; + default: + return -EINVAL; + } + + audio1 &= ~WM8990_AIF_FMT_MASK; + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + audio1 |= WM8990_AIF_TMF_I2S; + audio1 &= ~WM8990_AIF_LRCLK_INV; + break; + case SND_SOC_DAIFMT_RIGHT_J: + audio1 |= WM8990_AIF_TMF_RIGHTJ; + audio1 &= ~WM8990_AIF_LRCLK_INV; + break; + case SND_SOC_DAIFMT_LEFT_J: + audio1 |= WM8990_AIF_TMF_LEFTJ; + audio1 &= ~WM8990_AIF_LRCLK_INV; + break; + case SND_SOC_DAIFMT_DSP_A: + audio1 |= WM8990_AIF_TMF_DSP; + audio1 &= ~WM8990_AIF_LRCLK_INV; + break; + case SND_SOC_DAIFMT_DSP_B: + audio1 |= WM8990_AIF_TMF_DSP | WM8990_AIF_LRCLK_INV; + break; + default: + return -EINVAL; + } + + wm8990_write(codec, WM8990_AUDIO_INTERFACE_1, audio1); + wm8990_write(codec, WM8990_AUDIO_INTERFACE_3, audio3); + return 0; +} + +static int wm8990_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai, + int div_id, int div) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 reg; + + switch (div_id) { + case WM8990_MCLK_DIV: + reg = wm8990_read_reg_cache(codec, WM8990_CLOCKING_2) & + ~WM8990_MCLK_DIV_MASK; + wm8990_write(codec, WM8990_CLOCKING_2, reg | div); + break; + case WM8990_DACCLK_DIV: + reg = wm8990_read_reg_cache(codec, WM8990_CLOCKING_2) & + ~WM8990_DAC_CLKDIV_MASK; + wm8990_write(codec, WM8990_CLOCKING_2, reg | div); + break; + case WM8990_ADCCLK_DIV: + reg = wm8990_read_reg_cache(codec, WM8990_CLOCKING_2) & + ~WM8990_ADC_CLKDIV_MASK; + wm8990_write(codec, WM8990_CLOCKING_2, reg | div); + break; + case WM8990_BCLK_DIV: + reg = wm8990_read_reg_cache(codec, WM8990_CLOCKING_1) & + ~WM8990_BCLK_DIV_MASK; + wm8990_write(codec, WM8990_CLOCKING_1, reg | div); + break; + default: + return -EINVAL; + } + + return 0; +} + +/* + * Set PCM DAI bit size and sample rate. + */ +static int wm8990_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u16 audio1 = wm8990_read_reg_cache(codec, WM8990_AUDIO_INTERFACE_1); + + audio1 &= ~WM8990_AIF_WL_MASK; + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + audio1 |= WM8990_AIF_WL_20BITS; + break; + case SNDRV_PCM_FORMAT_S24_LE: + audio1 |= WM8990_AIF_WL_24BITS; + break; + case SNDRV_PCM_FORMAT_S32_LE: + audio1 |= WM8990_AIF_WL_32BITS; + break; + } + + wm8990_write(codec, WM8990_AUDIO_INTERFACE_1, audio1); + return 0; +} + +static int wm8990_mute(struct snd_soc_codec_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 val; + + val = wm8990_read_reg_cache(codec, WM8990_DAC_CTRL) & ~WM8990_DAC_MUTE; + + if (mute) + wm8990_write(codec, WM8990_DAC_CTRL, val | WM8990_DAC_MUTE); + else + wm8990_write(codec, WM8990_DAC_CTRL, val); + + return 0; +} + +static int wm8990_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 val; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (codec->bias_level == SND_SOC_BIAS_OFF) { + /* Enable all output discharge bits */ + wm8990_write(codec, WM8990_ANTIPOP1, WM8990_DIS_LLINE | + WM8990_DIS_RLINE | WM8990_DIS_OUT3 | + WM8990_DIS_OUT4 | WM8990_DIS_LOUT | + WM8990_DIS_ROUT); + + /* Enable POBCTRL, SOFT_ST, VMIDTOG and BUFDCOPEN */ + wm8990_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST | + WM8990_BUFDCOPEN | WM8990_POBCTRL | + WM8990_VMIDTOG); + + /* Delay to allow output caps to discharge */ + msleep(msecs_to_jiffies(300)); + + /* Disable VMIDTOG */ + wm8990_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST | + WM8990_BUFDCOPEN | WM8990_POBCTRL); + + /* disable all output discharge bits */ + wm8990_write(codec, WM8990_ANTIPOP1, 0); + + /* Enable outputs */ + wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1b00); + + msleep(msecs_to_jiffies(50)); + + /* Enable VMID at 2x50k */ + wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f02); + + msleep(msecs_to_jiffies(100)); + + /* Enable VREF */ + wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f03); + + msleep(msecs_to_jiffies(600)); + + /* Enable BUFIOEN */ + wm8990_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST | + WM8990_BUFDCOPEN | WM8990_POBCTRL | + WM8990_BUFIOEN); + + /* Disable outputs */ + wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x3); + + /* disable POBCTRL, SOFT_ST and BUFDCOPEN */ + wm8990_write(codec, WM8990_ANTIPOP2, WM8990_BUFIOEN); + } else { + /* ON -> standby */ + + } + break; + + case SND_SOC_BIAS_OFF: + /* Enable POBCTRL and SOFT_ST */ + wm8990_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST | + WM8990_POBCTRL | WM8990_BUFIOEN); + + /* Enable POBCTRL, SOFT_ST and BUFDCOPEN */ + wm8990_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST | + WM8990_BUFDCOPEN | WM8990_POBCTRL | + WM8990_BUFIOEN); + + /* mute DAC */ + val = wm8990_read_reg_cache(codec, WM8990_DAC_CTRL); + wm8990_write(codec, WM8990_DAC_CTRL, val | WM8990_DAC_MUTE); + + /* Enable any disabled outputs */ + wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f03); + + /* Disable VMID */ + wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f01); + + msleep(msecs_to_jiffies(300)); + + /* Enable all output discharge bits */ + wm8990_write(codec, WM8990_ANTIPOP1, WM8990_DIS_LLINE | + WM8990_DIS_RLINE | WM8990_DIS_OUT3 | + WM8990_DIS_OUT4 | WM8990_DIS_LOUT | + WM8990_DIS_ROUT); + + /* Disable VREF */ + wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x0); + + /* disable POBCTRL, SOFT_ST and BUFDCOPEN */ + wm8990_write(codec, WM8990_ANTIPOP2, 0x0); + break; + } + + codec->bias_level = level; + return 0; +} + +#define WM8990_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000) + +#define WM8990_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +/* + * The WM8990 supports 2 different and mutually exclusive DAI + * configurations. + * + * 1. ADC/DAC on Primary Interface + * 2. ADC on Primary Interface/DAC on secondary + */ +struct snd_soc_codec_dai wm8990_dai = { +/* ADC/DAC on primary */ + .name = "WM8990 ADC/DAC Primary", + .id = 1, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8990_RATES, + .formats = WM8990_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM8990_RATES, + .formats = WM8990_FORMATS,}, + .ops = { + .hw_params = wm8990_hw_params,}, + .dai_ops = { + .digital_mute = wm8990_mute, + .set_fmt = wm8990_set_dai_fmt, + .set_clkdiv = wm8990_set_dai_clkdiv, + .set_pll = wm8990_set_dai_pll, + .set_sysclk = wm8990_set_dai_sysclk, + }, +}; +EXPORT_SYMBOL_GPL(wm8990_dai); + +static int wm8990_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + /* we only need to suspend if we are a valid card */ + if (!codec->card) + return 0; + + wm8990_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int wm8990_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + int i; + u8 data[2]; + u16 *cache = codec->reg_cache; + + /* we only need to resume if we are a valid card */ + if (!codec->card) + return 0; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(wm8990_reg); i++) { + if (i + 1 == WM8990_RESET) + continue; + data[0] = ((i + 1) << 1) | ((cache[i] >> 8) & 0x0001); + data[1] = cache[i] & 0x00ff; + codec->hw_write(codec->control_data, data, 2); + } + + wm8990_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} + +/* + * initialise the WM8990 driver + * register the mixer and dsp interfaces with the kernel + */ +static int wm8990_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->codec; + u16 reg; + int ret = 0; + + codec->name = "WM8990"; + codec->owner = THIS_MODULE; + codec->read = wm8990_read_reg_cache; + codec->write = wm8990_write; + codec->set_bias_level = wm8990_set_bias_level; + codec->dai = &wm8990_dai; + codec->num_dai = 2; + codec->reg_cache_size = ARRAY_SIZE(wm8990_reg); + codec->reg_cache = kmemdup(wm8990_reg, sizeof(wm8990_reg), GFP_KERNEL); + + if (codec->reg_cache == NULL) + return -ENOMEM; + + wm8990_reset(codec); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "wm8990: failed to create pcms\n"); + goto pcm_err; + } + + /* charge output caps */ + codec->bias_level = SND_SOC_BIAS_OFF; + wm8990_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + reg = wm8990_read_reg_cache(codec, WM8990_AUDIO_INTERFACE_4); + wm8990_write(codec, WM8990_AUDIO_INTERFACE_4, reg | WM8990_ALRCGPIO1); + + reg = wm8990_read_reg_cache(codec, WM8990_GPIO1_GPIO2) & + ~WM8990_GPIO1_SEL_MASK; + wm8990_write(codec, WM8990_GPIO1_GPIO2, reg | 1); + + reg = wm8990_read_reg_cache(codec, WM8990_POWER_MANAGEMENT_2); + wm8990_write(codec, WM8990_POWER_MANAGEMENT_2, reg | WM8990_OPCLK_ENA); + + wm8990_write(codec, WM8990_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8)); + wm8990_write(codec, WM8990_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8)); + + wm8990_add_controls(codec); + wm8990_add_widgets(codec); + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "wm8990: failed to register card\n"); + goto card_err; + } + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + return ret; +} + +/* If the i2c layer weren't so broken, we could pass this kind of data + around */ +static struct snd_soc_device *wm8990_socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + +/* + * WM891 2 wire address is determined by GPIO5 + * state during powerup. + * low = 0x34 + * high = 0x36 + */ +static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; + +/* Magic definition of all other variables and things */ +I2C_CLIENT_INSMOD; + +static struct i2c_driver wm8990_i2c_driver; +static struct i2c_client client_template; + +static int wm8990_codec_probe(struct i2c_adapter *adap, int addr, int kind) +{ + struct snd_soc_device *socdev = wm8990_socdev; + struct wm8990_setup_data *setup = socdev->codec_data; + struct snd_soc_codec *codec = socdev->codec; + struct i2c_client *i2c; + int ret; + + if (addr != setup->i2c_address) + return -ENODEV; + + client_template.adapter = adap; + client_template.addr = addr; + + i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); + if (i2c == NULL) { + kfree(codec); + return -ENOMEM; + } + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + ret = i2c_attach_client(i2c); + if (ret < 0) { + err("failed to attach codec at addr %x\n", addr); + goto err; + } + + ret = wm8990_init(socdev); + if (ret < 0) { + err("failed to initialise WM8990\n"); + goto err; + } + return ret; + +err: + kfree(codec); + kfree(i2c); + return ret; +} + +static int wm8990_i2c_detach(struct i2c_client *client) +{ + struct snd_soc_codec *codec = i2c_get_clientdata(client); + i2c_detach_client(client); + kfree(codec->reg_cache); + kfree(client); + return 0; +} + +static int wm8990_i2c_attach(struct i2c_adapter *adap) +{ + return i2c_probe(adap, &addr_data, wm8990_codec_probe); +} + +static struct i2c_driver wm8990_i2c_driver = { + .driver = { + .name = "WM8990 I2C Codec", + .owner = THIS_MODULE, + }, + .attach_adapter = wm8990_i2c_attach, + .detach_client = wm8990_i2c_detach, + .command = NULL, +}; + +static struct i2c_client client_template = { + .name = "WM8990", + .driver = &wm8990_i2c_driver, +}; +#endif + +static int wm8990_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct wm8990_setup_data *setup; + struct snd_soc_codec *codec; + struct wm8990_priv *wm8990; + int ret = 0; + + info("WM8990 Audio Codec %s\n", WM8990_VERSION); + + setup = socdev->codec_data; + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + wm8990 = kzalloc(sizeof(struct wm8990_priv), GFP_KERNEL); + if (wm8990 == NULL) { + kfree(codec); + return -ENOMEM; + } + + codec->private_data = wm8990; + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + wm8990_socdev = socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + if (setup->i2c_address) { + normal_i2c[0] = setup->i2c_address; + codec->hw_write = (hw_write_t)i2c_master_send; + ret = i2c_add_driver(&wm8990_i2c_driver); + if (ret != 0) + printk(KERN_ERR "can't add i2c driver"); + } +#else + /* Add other interfaces here */ +#endif + return ret; +} + +/* power down chip */ +static int wm8990_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec->control_data) + wm8990_set_bias_level(codec, SND_SOC_BIAS_OFF); + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&wm8990_i2c_driver); +#endif + kfree(codec->private_data); + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8990 = { + .probe = wm8990_probe, + .remove = wm8990_remove, + .suspend = wm8990_suspend, + .resume = wm8990_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8990); + +MODULE_DESCRIPTION("ASoC WM8990 driver"); +MODULE_AUTHOR("Liam Girdwood"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8990.h b/sound/soc/codecs/wm8990.h new file mode 100644 index 00000000000..bf9f8823dfc --- /dev/null +++ b/sound/soc/codecs/wm8990.h @@ -0,0 +1,832 @@ +/* + * wm8990.h -- audio driver for WM8990 + * + * Copyright 2007 Wolfson Microelectronics PLC. + * Author: Graeme Gregory + * graeme.gregory@wolfsonmicro.com or linux@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. + * + */ + +#ifndef __WM8990REGISTERDEFS_H__ +#define __WM8990REGISTERDEFS_H__ + +/* + * Register values. + */ +#define WM8990_RESET 0x00 +#define WM8990_POWER_MANAGEMENT_1 0x01 +#define WM8990_POWER_MANAGEMENT_2 0x02 +#define WM8990_POWER_MANAGEMENT_3 0x03 +#define WM8990_AUDIO_INTERFACE_1 0x04 +#define WM8990_AUDIO_INTERFACE_2 0x05 +#define WM8990_CLOCKING_1 0x06 +#define WM8990_CLOCKING_2 0x07 +#define WM8990_AUDIO_INTERFACE_3 0x08 +#define WM8990_AUDIO_INTERFACE_4 0x09 +#define WM8990_DAC_CTRL 0x0A +#define WM8990_LEFT_DAC_DIGITAL_VOLUME 0x0B +#define WM8990_RIGHT_DAC_DIGITAL_VOLUME 0x0C +#define WM8990_DIGITAL_SIDE_TONE 0x0D +#define WM8990_ADC_CTRL 0x0E +#define WM8990_LEFT_ADC_DIGITAL_VOLUME 0x0F +#define WM8990_RIGHT_ADC_DIGITAL_VOLUME 0x10 +#define WM8990_GPIO_CTRL_1 0x12 +#define WM8990_GPIO1_GPIO2 0x13 +#define WM8990_GPIO3_GPIO4 0x14 +#define WM8990_GPIO5_GPIO6 0x15 +#define WM8990_GPIOCTRL_2 0x16 +#define WM8990_GPIO_POL 0x17 +#define WM8990_LEFT_LINE_INPUT_1_2_VOLUME 0x18 +#define WM8990_LEFT_LINE_INPUT_3_4_VOLUME 0x19 +#define WM8990_RIGHT_LINE_INPUT_1_2_VOLUME 0x1A +#define WM8990_RIGHT_LINE_INPUT_3_4_VOLUME 0x1B +#define WM8990_LEFT_OUTPUT_VOLUME 0x1C +#define WM8990_RIGHT_OUTPUT_VOLUME 0x1D +#define WM8990_LINE_OUTPUTS_VOLUME 0x1E +#define WM8990_OUT3_4_VOLUME 0x1F +#define WM8990_LEFT_OPGA_VOLUME 0x20 +#define WM8990_RIGHT_OPGA_VOLUME 0x21 +#define WM8990_SPEAKER_VOLUME 0x22 +#define WM8990_CLASSD1 0x23 +#define WM8990_CLASSD3 0x25 +#define WM8990_INPUT_MIXER1 0x27 +#define WM8990_INPUT_MIXER2 0x28 +#define WM8990_INPUT_MIXER3 0x29 +#define WM8990_INPUT_MIXER4 0x2A +#define WM8990_INPUT_MIXER5 0x2B +#define WM8990_INPUT_MIXER6 0x2C +#define WM8990_OUTPUT_MIXER1 0x2D +#define WM8990_OUTPUT_MIXER2 0x2E +#define WM8990_OUTPUT_MIXER3 0x2F +#define WM8990_OUTPUT_MIXER4 0x30 +#define WM8990_OUTPUT_MIXER5 0x31 +#define WM8990_OUTPUT_MIXER6 0x32 +#define WM8990_OUT3_4_MIXER 0x33 +#define WM8990_LINE_MIXER1 0x34 +#define WM8990_LINE_MIXER2 0x35 +#define WM8990_SPEAKER_MIXER 0x36 +#define WM8990_ADDITIONAL_CONTROL 0x37 +#define WM8990_ANTIPOP1 0x38 +#define WM8990_ANTIPOP2 0x39 +#define WM8990_MICBIAS 0x3A +#define WM8990_PLL1 0x3C +#define WM8990_PLL2 0x3D +#define WM8990_PLL3 0x3E +#define WM8990_INTDRIVBITS 0x3F + +#define WM8990_REGISTER_COUNT 60 +#define WM8990_MAX_REGISTER 0x3F + +/* + * Field Definitions. + */ + +/* + * R0 (0x00) - Reset + */ +#define WM8990_SW_RESET_CHIP_ID_MASK 0xFFFF /* SW_RESET_CHIP_ID */ + +/* + * R1 (0x01) - Power Management (1) + */ +#define WM8990_SPK_ENA 0x1000 /* SPK_ENA */ +#define WM8990_SPK_ENA_BIT 12 +#define WM8990_OUT3_ENA 0x0800 /* OUT3_ENA */ +#define WM8990_OUT3_ENA_BIT 11 +#define WM8990_OUT4_ENA 0x0400 /* OUT4_ENA */ +#define WM8990_OUT4_ENA_BIT 10 +#define WM8990_LOUT_ENA 0x0200 /* LOUT_ENA */ +#define WM8990_LOUT_ENA_BIT 9 +#define WM8990_ROUT_ENA 0x0100 /* ROUT_ENA */ +#define WM8990_ROUT_ENA_BIT 8 +#define WM8990_MICBIAS_ENA 0x0010 /* MICBIAS_ENA */ +#define WM8990_MICBIAS_ENA_BIT 4 +#define WM8990_VMID_MODE_MASK 0x0006 /* VMID_MODE - [2:1] */ +#define WM8990_VREF_ENA 0x0001 /* VREF_ENA */ +#define WM8990_VREF_ENA_BIT 0 + +/* + * R2 (0x02) - Power Management (2) + */ +#define WM8990_PLL_ENA 0x8000 /* PLL_ENA */ +#define WM8990_PLL_ENA_BIT 15 +#define WM8990_TSHUT_ENA 0x4000 /* TSHUT_ENA */ +#define WM8990_TSHUT_ENA_BIT 14 +#define WM8990_TSHUT_OPDIS 0x2000 /* TSHUT_OPDIS */ +#define WM8990_TSHUT_OPDIS_BIT 13 +#define WM8990_OPCLK_ENA 0x0800 /* OPCLK_ENA */ +#define WM8990_OPCLK_ENA_BIT 11 +#define WM8990_AINL_ENA 0x0200 /* AINL_ENA */ +#define WM8990_AINL_ENA_BIT 9 +#define WM8990_AINR_ENA 0x0100 /* AINR_ENA */ +#define WM8990_AINR_ENA_BIT 8 +#define WM8990_LIN34_ENA 0x0080 /* LIN34_ENA */ +#define WM8990_LIN34_ENA_BIT 7 +#define WM8990_LIN12_ENA 0x0040 /* LIN12_ENA */ +#define WM8990_LIN12_ENA_BIT 6 +#define WM8990_RIN34_ENA 0x0020 /* RIN34_ENA */ +#define WM8990_RIN34_ENA_BIT 5 +#define WM8990_RIN12_ENA 0x0010 /* RIN12_ENA */ +#define WM8990_RIN12_ENA_BIT 4 +#define WM8990_ADCL_ENA 0x0002 /* ADCL_ENA */ +#define WM8990_ADCL_ENA_BIT 1 +#define WM8990_ADCR_ENA 0x0001 /* ADCR_ENA */ +#define WM8990_ADCR_ENA_BIT 0 + +/* + * R3 (0x03) - Power Management (3) + */ +#define WM8990_LON_ENA 0x2000 /* LON_ENA */ +#define WM8990_LON_ENA_BIT 13 +#define WM8990_LOP_ENA 0x1000 /* LOP_ENA */ +#define WM8990_LOP_ENA_BIT 12 +#define WM8990_RON_ENA 0x0800 /* RON_ENA */ +#define WM8990_RON_ENA_BIT 11 +#define WM8990_ROP_ENA 0x0400 /* ROP_ENA */ +#define WM8990_ROP_ENA_BIT 10 +#define WM8990_LOPGA_ENA 0x0080 /* LOPGA_ENA */ +#define WM8990_LOPGA_ENA_BIT 7 +#define WM8990_ROPGA_ENA 0x0040 /* ROPGA_ENA */ +#define WM8990_ROPGA_ENA_BIT 6 +#define WM8990_LOMIX_ENA 0x0020 /* LOMIX_ENA */ +#define WM8990_LOMIX_ENA_BIT 5 +#define WM8990_ROMIX_ENA 0x0010 /* ROMIX_ENA */ +#define WM8990_ROMIX_ENA_BIT 4 +#define WM8990_DACL_ENA 0x0002 /* DACL_ENA */ +#define WM8990_DACL_ENA_BIT 1 +#define WM8990_DACR_ENA 0x0001 /* DACR_ENA */ +#define WM8990_DACR_ENA_BIT 0 + +/* + * R4 (0x04) - Audio Interface (1) + */ +#define WM8990_AIFADCL_SRC 0x8000 /* AIFADCL_SRC */ +#define WM8990_AIFADCR_SRC 0x4000 /* AIFADCR_SRC */ +#define WM8990_AIFADC_TDM 0x2000 /* AIFADC_TDM */ +#define WM8990_AIFADC_TDM_CHAN 0x1000 /* AIFADC_TDM_CHAN */ +#define WM8990_AIF_BCLK_INV 0x0100 /* AIF_BCLK_INV */ +#define WM8990_AIF_LRCLK_INV 0x0080 /* AIF_LRCLK_INV */ +#define WM8990_AIF_WL_MASK 0x0060 /* AIF_WL - [6:5] */ +#define WM8990_AIF_WL_16BITS (0 << 5) +#define WM8990_AIF_WL_20BITS (1 << 5) +#define WM8990_AIF_WL_24BITS (2 << 5) +#define WM8990_AIF_WL_32BITS (3 << 5) +#define WM8990_AIF_FMT_MASK 0x0018 /* AIF_FMT - [4:3] */ +#define WM8990_AIF_TMF_RIGHTJ (0 << 3) +#define WM8990_AIF_TMF_LEFTJ (1 << 3) +#define WM8990_AIF_TMF_I2S (2 << 3) +#define WM8990_AIF_TMF_DSP (3 << 3) + +/* + * R5 (0x05) - Audio Interface (2) + */ +#define WM8990_DACL_SRC 0x8000 /* DACL_SRC */ +#define WM8990_DACR_SRC 0x4000 /* DACR_SRC */ +#define WM8990_AIFDAC_TDM 0x2000 /* AIFDAC_TDM */ +#define WM8990_AIFDAC_TDM_CHAN 0x1000 /* AIFDAC_TDM_CHAN */ +#define WM8990_DAC_BOOST_MASK 0x0C00 /* DAC_BOOST */ +#define WM8990_DAC_COMP 0x0010 /* DAC_COMP */ +#define WM8990_DAC_COMPMODE 0x0008 /* DAC_COMPMODE */ +#define WM8990_ADC_COMP 0x0004 /* ADC_COMP */ +#define WM8990_ADC_COMPMODE 0x0002 /* ADC_COMPMODE */ +#define WM8990_LOOPBACK 0x0001 /* LOOPBACK */ + +/* + * R6 (0x06) - Clocking (1) + */ +#define WM8990_TOCLK_RATE 0x8000 /* TOCLK_RATE */ +#define WM8990_TOCLK_ENA 0x4000 /* TOCLK_ENA */ +#define WM8990_OPCLKDIV_MASK 0x1E00 /* OPCLKDIV - [12:9] */ +#define WM8990_DCLKDIV_MASK 0x01C0 /* DCLKDIV - [8:6] */ +#define WM8990_BCLK_DIV_MASK 0x001E /* BCLK_DIV - [4:1] */ +#define WM8990_BCLK_DIV_1 (0x0 << 1) +#define WM8990_BCLK_DIV_1_5 (0x1 << 1) +#define WM8990_BCLK_DIV_2 (0x2 << 1) +#define WM8990_BCLK_DIV_3 (0x3 << 1) +#define WM8990_BCLK_DIV_4 (0x4 << 1) +#define WM8990_BCLK_DIV_5_5 (0x5 << 1) +#define WM8990_BCLK_DIV_6 (0x6 << 1) +#define WM8990_BCLK_DIV_8 (0x7 << 1) +#define WM8990_BCLK_DIV_11 (0x8 << 1) +#define WM8990_BCLK_DIV_12 (0x9 << 1) +#define WM8990_BCLK_DIV_16 (0xA << 1) +#define WM8990_BCLK_DIV_22 (0xB << 1) +#define WM8990_BCLK_DIV_24 (0xC << 1) +#define WM8990_BCLK_DIV_32 (0xD << 1) +#define WM8990_BCLK_DIV_44 (0xE << 1) +#define WM8990_BCLK_DIV_48 (0xF << 1) + +/* + * R7 (0x07) - Clocking (2) + */ +#define WM8990_MCLK_SRC 0x8000 /* MCLK_SRC */ +#define WM8990_SYSCLK_SRC 0x4000 /* SYSCLK_SRC */ +#define WM8990_CLK_FORCE 0x2000 /* CLK_FORCE */ +#define WM8990_MCLK_DIV_MASK 0x1800 /* MCLK_DIV - [12:11] */ +#define WM8990_MCLK_DIV_1 (0 << 11) +#define WM8990_MCLK_DIV_2 (2 << 11) +#define WM8990_MCLK_INV 0x0400 /* MCLK_INV */ +#define WM8990_ADC_CLKDIV_MASK 0x00E0 /* ADC_CLKDIV */ +#define WM8990_ADC_CLKDIV_1 (0 << 5) +#define WM8990_ADC_CLKDIV_1_5 (1 << 5) +#define WM8990_ADC_CLKDIV_2 (2 << 5) +#define WM8990_ADC_CLKDIV_3 (3 << 5) +#define WM8990_ADC_CLKDIV_4 (4 << 5) +#define WM8990_ADC_CLKDIV_5_5 (5 << 5) +#define WM8990_ADC_CLKDIV_6 (6 << 5) +#define WM8990_DAC_CLKDIV_MASK 0x001C /* DAC_CLKDIV - [4:2] */ +#define WM8990_DAC_CLKDIV_1 (0 << 2) +#define WM8990_DAC_CLKDIV_1_5 (1 << 2) +#define WM8990_DAC_CLKDIV_2 (2 << 2) +#define WM8990_DAC_CLKDIV_3 (3 << 2) +#define WM8990_DAC_CLKDIV_4 (4 << 2) +#define WM8990_DAC_CLKDIV_5_5 (5 << 2) +#define WM8990_DAC_CLKDIV_6 (6 << 2) + +/* + * R8 (0x08) - Audio Interface (3) + */ +#define WM8990_AIF_MSTR1 0x8000 /* AIF_MSTR1 */ +#define WM8990_AIF_MSTR2 0x4000 /* AIF_MSTR2 */ +#define WM8990_AIF_SEL 0x2000 /* AIF_SEL */ +#define WM8990_ADCLRC_DIR 0x0800 /* ADCLRC_DIR */ +#define WM8990_ADCLRC_RATE_MASK 0x07FF /* ADCLRC_RATE */ + +/* + * R9 (0x09) - Audio Interface (4) + */ +#define WM8990_ALRCGPIO1 0x8000 /* ALRCGPIO1 */ +#define WM8990_ALRCBGPIO6 0x4000 /* ALRCBGPIO6 */ +#define WM8990_AIF_TRIS 0x2000 /* AIF_TRIS */ +#define WM8990_DACLRC_DIR 0x0800 /* DACLRC_DIR */ +#define WM8990_DACLRC_RATE_MASK 0x07FF /* DACLRC_RATE */ + +/* + * R10 (0x0A) - DAC CTRL + */ +#define WM8990_AIF_LRCLKRATE 0x0400 /* AIF_LRCLKRATE */ +#define WM8990_DAC_MONO 0x0200 /* DAC_MONO */ +#define WM8990_DAC_SB_FILT 0x0100 /* DAC_SB_FILT */ +#define WM8990_DAC_MUTERATE 0x0080 /* DAC_MUTERATE */ +#define WM8990_DAC_MUTEMODE 0x0040 /* DAC_MUTEMODE */ +#define WM8990_DEEMP_MASK 0x0030 /* DEEMP - [5:4] */ +#define WM8990_DAC_MUTE 0x0004 /* DAC_MUTE */ +#define WM8990_DACL_DATINV 0x0002 /* DACL_DATINV */ +#define WM8990_DACR_DATINV 0x0001 /* DACR_DATINV */ + +/* + * R11 (0x0B) - Left DAC Digital Volume + */ +#define WM8990_DAC_VU 0x0100 /* DAC_VU */ +#define WM8990_DACL_VOL_MASK 0x00FF /* DACL_VOL - [7:0] */ +#define WM8990_DACL_VOL_SHIFT 0 +/* + * R12 (0x0C) - Right DAC Digital Volume + */ +#define WM8990_DAC_VU 0x0100 /* DAC_VU */ +#define WM8990_DACR_VOL_MASK 0x00FF /* DACR_VOL - [7:0] */ +#define WM8990_DACR_VOL_SHIFT 0 +/* + * R13 (0x0D) - Digital Side Tone + */ +#define WM8990_ADCL_DAC_SVOL_MASK 0x0F /* ADCL_DAC_SVOL */ +#define WM8990_ADCL_DAC_SVOL_SHIFT 9 +#define WM8990_ADCR_DAC_SVOL_MASK 0x0F /* ADCR_DAC_SVOL */ +#define WM8990_ADCR_DAC_SVOL_SHIFT 5 +#define WM8990_ADC_TO_DACL_MASK 0x03 /* ADC_TO_DACL - [3:2] */ +#define WM8990_ADC_TO_DACL_SHIFT 2 +#define WM8990_ADC_TO_DACR_MASK 0x03 /* ADC_TO_DACR - [1:0] */ +#define WM8990_ADC_TO_DACR_SHIFT 0 + +/* + * R14 (0x0E) - ADC CTRL + */ +#define WM8990_ADC_HPF_ENA 0x0100 /* ADC_HPF_ENA */ +#define WM8990_ADC_HPF_ENA_BIT 8 +#define WM8990_ADC_HPF_CUT_MASK 0x03 /* ADC_HPF_CUT - [6:5] */ +#define WM8990_ADC_HPF_CUT_SHIFT 5 +#define WM8990_ADCL_DATINV 0x0002 /* ADCL_DATINV */ +#define WM8990_ADCL_DATINV_BIT 1 +#define WM8990_ADCR_DATINV 0x0001 /* ADCR_DATINV */ +#define WM8990_ADCR_DATINV_BIT 0 + +/* + * R15 (0x0F) - Left ADC Digital Volume + */ +#define WM8990_ADC_VU 0x0100 /* ADC_VU */ +#define WM8990_ADCL_VOL_MASK 0x00FF /* ADCL_VOL - [7:0] */ +#define WM8990_ADCL_VOL_SHIFT 0 + +/* + * R16 (0x10) - Right ADC Digital Volume + */ +#define WM8990_ADC_VU 0x0100 /* ADC_VU */ +#define WM8990_ADCR_VOL_MASK 0x00FF /* ADCR_VOL - [7:0] */ +#define WM8990_ADCR_VOL_SHIFT 0 + +/* + * R18 (0x12) - GPIO CTRL 1 + */ +#define WM8990_IRQ 0x1000 /* IRQ */ +#define WM8990_TEMPOK 0x0800 /* TEMPOK */ +#define WM8990_MICSHRT 0x0400 /* MICSHRT */ +#define WM8990_MICDET 0x0200 /* MICDET */ +#define WM8990_PLL_LCK 0x0100 /* PLL_LCK */ +#define WM8990_GPI8_STATUS 0x0080 /* GPI8_STATUS */ +#define WM8990_GPI7_STATUS 0x0040 /* GPI7_STATUS */ +#define WM8990_GPIO6_STATUS 0x0020 /* GPIO6_STATUS */ +#define WM8990_GPIO5_STATUS 0x0010 /* GPIO5_STATUS */ +#define WM8990_GPIO4_STATUS 0x0008 /* GPIO4_STATUS */ +#define WM8990_GPIO3_STATUS 0x0004 /* GPIO3_STATUS */ +#define WM8990_GPIO2_STATUS 0x0002 /* GPIO2_STATUS */ +#define WM8990_GPIO1_STATUS 0x0001 /* GPIO1_STATUS */ + +/* + * R19 (0x13) - GPIO1 & GPIO2 + */ +#define WM8990_GPIO2_DEB_ENA 0x8000 /* GPIO2_DEB_ENA */ +#define WM8990_GPIO2_IRQ_ENA 0x4000 /* GPIO2_IRQ_ENA */ +#define WM8990_GPIO2_PU 0x2000 /* GPIO2_PU */ +#define WM8990_GPIO2_PD 0x1000 /* GPIO2_PD */ +#define WM8990_GPIO2_SEL_MASK 0x0F00 /* GPIO2_SEL - [11:8] */ +#define WM8990_GPIO1_DEB_ENA 0x0080 /* GPIO1_DEB_ENA */ +#define WM8990_GPIO1_IRQ_ENA 0x0040 /* GPIO1_IRQ_ENA */ +#define WM8990_GPIO1_PU 0x0020 /* GPIO1_PU */ +#define WM8990_GPIO1_PD 0x0010 /* GPIO1_PD */ +#define WM8990_GPIO1_SEL_MASK 0x000F /* GPIO1_SEL - [3:0] */ + +/* + * R20 (0x14) - GPIO3 & GPIO4 + */ +#define WM8990_GPIO4_DEB_ENA 0x8000 /* GPIO4_DEB_ENA */ +#define WM8990_GPIO4_IRQ_ENA 0x4000 /* GPIO4_IRQ_ENA */ +#define WM8990_GPIO4_PU 0x2000 /* GPIO4_PU */ +#define WM8990_GPIO4_PD 0x1000 /* GPIO4_PD */ +#define WM8990_GPIO4_SEL_MASK 0x0F00 /* GPIO4_SEL - [11:8] */ +#define WM8990_GPIO3_DEB_ENA 0x0080 /* GPIO3_DEB_ENA */ +#define WM8990_GPIO3_IRQ_ENA 0x0040 /* GPIO3_IRQ_ENA */ +#define WM8990_GPIO3_PU 0x0020 /* GPIO3_PU */ +#define WM8990_GPIO3_PD 0x0010 /* GPIO3_PD */ +#define WM8990_GPIO3_SEL_MASK 0x000F /* GPIO3_SEL - [3:0] */ + +/* + * R21 (0x15) - GPIO5 & GPIO6 + */ +#define WM8990_GPIO6_DEB_ENA 0x8000 /* GPIO6_DEB_ENA */ +#define WM8990_GPIO6_IRQ_ENA 0x4000 /* GPIO6_IRQ_ENA */ +#define WM8990_GPIO6_PU 0x2000 /* GPIO6_PU */ +#define WM8990_GPIO6_PD 0x1000 /* GPIO6_PD */ +#define WM8990_GPIO6_SEL_MASK 0x0F00 /* GPIO6_SEL - [11:8] */ +#define WM8990_GPIO5_DEB_ENA 0x0080 /* GPIO5_DEB_ENA */ +#define WM8990_GPIO5_IRQ_ENA 0x0040 /* GPIO5_IRQ_ENA */ +#define WM8990_GPIO5_PU 0x0020 /* GPIO5_PU */ +#define WM8990_GPIO5_PD 0x0010 /* GPIO5_PD */ +#define WM8990_GPIO5_SEL_MASK 0x000F /* GPIO5_SEL - [3:0] */ + +/* + * R22 (0x16) - GPIOCTRL 2 + */ +#define WM8990_RD_3W_ENA 0x8000 /* RD_3W_ENA */ +#define WM8990_MODE_3W4W 0x4000 /* MODE_3W4W */ +#define WM8990_TEMPOK_IRQ_ENA 0x0800 /* TEMPOK_IRQ_ENA */ +#define WM8990_MICSHRT_IRQ_ENA 0x0400 /* MICSHRT_IRQ_ENA */ +#define WM8990_MICDET_IRQ_ENA 0x0200 /* MICDET_IRQ_ENA */ +#define WM8990_PLL_LCK_IRQ_ENA 0x0100 /* PLL_LCK_IRQ_ENA */ +#define WM8990_GPI8_DEB_ENA 0x0080 /* GPI8_DEB_ENA */ +#define WM8990_GPI8_IRQ_ENA 0x0040 /* GPI8_IRQ_ENA */ +#define WM8990_GPI8_ENA 0x0010 /* GPI8_ENA */ +#define WM8990_GPI7_DEB_ENA 0x0008 /* GPI7_DEB_ENA */ +#define WM8990_GPI7_IRQ_ENA 0x0004 /* GPI7_IRQ_ENA */ +#define WM8990_GPI7_ENA 0x0001 /* GPI7_ENA */ + +/* + * R23 (0x17) - GPIO_POL + */ +#define WM8990_IRQ_INV 0x1000 /* IRQ_INV */ +#define WM8990_TEMPOK_POL 0x0800 /* TEMPOK_POL */ +#define WM8990_MICSHRT_POL 0x0400 /* MICSHRT_POL */ +#define WM8990_MICDET_POL 0x0200 /* MICDET_POL */ +#define WM8990_PLL_LCK_POL 0x0100 /* PLL_LCK_POL */ +#define WM8990_GPI8_POL 0x0080 /* GPI8_POL */ +#define WM8990_GPI7_POL 0x0040 /* GPI7_POL */ +#define WM8990_GPIO6_POL 0x0020 /* GPIO6_POL */ +#define WM8990_GPIO5_POL 0x0010 /* GPIO5_POL */ +#define WM8990_GPIO4_POL 0x0008 /* GPIO4_POL */ +#define WM8990_GPIO3_POL 0x0004 /* GPIO3_POL */ +#define WM8990_GPIO2_POL 0x0002 /* GPIO2_POL */ +#define WM8990_GPIO1_POL 0x0001 /* GPIO1_POL */ + +/* + * R24 (0x18) - Left Line Input 1&2 Volume + */ +#define WM8990_IPVU 0x0100 /* IPVU */ +#define WM8990_LI12MUTE 0x0080 /* LI12MUTE */ +#define WM8990_LI12MUTE_BIT 7 +#define WM8990_LI12ZC 0x0040 /* LI12ZC */ +#define WM8990_LI12ZC_BIT 6 +#define WM8990_LIN12VOL_MASK 0x001F /* LIN12VOL - [4:0] */ +#define WM8990_LIN12VOL_SHIFT 0 +/* + * R25 (0x19) - Left Line Input 3&4 Volume + */ +#define WM8990_IPVU 0x0100 /* IPVU */ +#define WM8990_LI34MUTE 0x0080 /* LI34MUTE */ +#define WM8990_LI34MUTE_BIT 7 +#define WM8990_LI34ZC 0x0040 /* LI34ZC */ +#define WM8990_LI34ZC_BIT 6 +#define WM8990_LIN34VOL_MASK 0x001F /* LIN34VOL - [4:0] */ +#define WM8990_LIN34VOL_SHIFT 0 + +/* + * R26 (0x1A) - Right Line Input 1&2 Volume + */ +#define WM8990_IPVU 0x0100 /* IPVU */ +#define WM8990_RI12MUTE 0x0080 /* RI12MUTE */ +#define WM8990_RI12MUTE_BIT 7 +#define WM8990_RI12ZC 0x0040 /* RI12ZC */ +#define WM8990_RI12ZC_BIT 6 +#define WM8990_RIN12VOL_MASK 0x001F /* RIN12VOL - [4:0] */ +#define WM8990_RIN12VOL_SHIFT 0 + +/* + * R27 (0x1B) - Right Line Input 3&4 Volume + */ +#define WM8990_IPVU 0x0100 /* IPVU */ +#define WM8990_RI34MUTE 0x0080 /* RI34MUTE */ +#define WM8990_RI34MUTE_BIT 7 +#define WM8990_RI34ZC 0x0040 /* RI34ZC */ +#define WM8990_RI34ZC_BIT 6 +#define WM8990_RIN34VOL_MASK 0x001F /* RIN34VOL - [4:0] */ +#define WM8990_RIN34VOL_SHIFT 0 + +/* + * R28 (0x1C) - Left Output Volume + */ +#define WM8990_OPVU 0x0100 /* OPVU */ +#define WM8990_LOZC 0x0080 /* LOZC */ +#define WM8990_LOZC_BIT 7 +#define WM8990_LOUTVOL_MASK 0x007F /* LOUTVOL - [6:0] */ +#define WM8990_LOUTVOL_SHIFT 0 +/* + * R29 (0x1D) - Right Output Volume + */ +#define WM8990_OPVU 0x0100 /* OPVU */ +#define WM8990_ROZC 0x0080 /* ROZC */ +#define WM8990_ROZC_BIT 7 +#define WM8990_ROUTVOL_MASK 0x007F /* ROUTVOL - [6:0] */ +#define WM8990_ROUTVOL_SHIFT 0 +/* + * R30 (0x1E) - Line Outputs Volume + */ +#define WM8990_LONMUTE 0x0040 /* LONMUTE */ +#define WM8990_LONMUTE_BIT 6 +#define WM8990_LOPMUTE 0x0020 /* LOPMUTE */ +#define WM8990_LOPMUTE_BIT 5 +#define WM8990_LOATTN 0x0010 /* LOATTN */ +#define WM8990_LOATTN_BIT 4 +#define WM8990_RONMUTE 0x0004 /* RONMUTE */ +#define WM8990_RONMUTE_BIT 2 +#define WM8990_ROPMUTE 0x0002 /* ROPMUTE */ +#define WM8990_ROPMUTE_BIT 1 +#define WM8990_ROATTN 0x0001 /* ROATTN */ +#define WM8990_ROATTN_BIT 0 + +/* + * R31 (0x1F) - Out3/4 Volume + */ +#define WM8990_OUT3MUTE 0x0020 /* OUT3MUTE */ +#define WM8990_OUT3MUTE_BIT 5 +#define WM8990_OUT3ATTN 0x0010 /* OUT3ATTN */ +#define WM8990_OUT3ATTN_BIT 4 +#define WM8990_OUT4MUTE 0x0002 /* OUT4MUTE */ +#define WM8990_OUT4MUTE_BIT 1 +#define WM8990_OUT4ATTN 0x0001 /* OUT4ATTN */ +#define WM8990_OUT4ATTN_BIT 0 + +/* + * R32 (0x20) - Left OPGA Volume + */ +#define WM8990_OPVU 0x0100 /* OPVU */ +#define WM8990_LOPGAZC 0x0080 /* LOPGAZC */ +#define WM8990_LOPGAZC_BIT 7 +#define WM8990_LOPGAVOL_MASK 0x007F /* LOPGAVOL - [6:0] */ +#define WM8990_LOPGAVOL_SHIFT 0 + +/* + * R33 (0x21) - Right OPGA Volume + */ +#define WM8990_OPVU 0x0100 /* OPVU */ +#define WM8990_ROPGAZC 0x0080 /* ROPGAZC */ +#define WM8990_ROPGAZC_BIT 7 +#define WM8990_ROPGAVOL_MASK 0x007F /* ROPGAVOL - [6:0] */ +#define WM8990_ROPGAVOL_SHIFT 0 +/* + * R34 (0x22) - Speaker Volume + */ +#define WM8990_SPKVOL_MASK 0x0003 /* SPKVOL - [1:0] */ +#define WM8990_SPKVOL_SHIFT 0 + +/* + * R35 (0x23) - ClassD1 + */ +#define WM8990_CDMODE 0x0100 /* CDMODE */ +#define WM8990_CDMODE_BIT 8 + +/* + * R37 (0x25) - ClassD3 + */ +#define WM8990_DCGAIN_MASK 0x0007 /* DCGAIN - [5:3] */ +#define WM8990_DCGAIN_SHIFT 3 +#define WM8990_ACGAIN_MASK 0x0007 /* ACGAIN - [2:0] */ +#define WM8990_ACGAIN_SHIFT 0 +/* + * R39 (0x27) - Input Mixer1 + */ +#define WM8990_AINLMODE_MASK 0x000C /* AINLMODE - [3:2] */ +#define WM8990_AINLMODE_SHIFT 2 +#define WM8990_AINRMODE_MASK 0x0003 /* AINRMODE - [1:0] */ +#define WM8990_AINRMODE_SHIFT 0 + +/* + * R40 (0x28) - Input Mixer2 + */ +#define WM8990_LMP4 0x0080 /* LMP4 */ +#define WM8990_LMP4_BIT 7 /* LMP4 */ +#define WM8990_LMN3 0x0040 /* LMN3 */ +#define WM8990_LMN3_BIT 6 /* LMN3 */ +#define WM8990_LMP2 0x0020 /* LMP2 */ +#define WM8990_LMP2_BIT 5 /* LMP2 */ +#define WM8990_LMN1 0x0010 /* LMN1 */ +#define WM8990_LMN1_BIT 4 /* LMN1 */ +#define WM8990_RMP4 0x0008 /* RMP4 */ +#define WM8990_RMP4_BIT 3 /* RMP4 */ +#define WM8990_RMN3 0x0004 /* RMN3 */ +#define WM8990_RMN3_BIT 2 /* RMN3 */ +#define WM8990_RMP2 0x0002 /* RMP2 */ +#define WM8990_RMP2_BIT 1 /* RMP2 */ +#define WM8990_RMN1 0x0001 /* RMN1 */ +#define WM8990_RMN1_BIT 0 /* RMN1 */ + +/* + * R41 (0x29) - Input Mixer3 + */ +#define WM8990_L34MNB 0x0100 /* L34MNB */ +#define WM8990_L34MNB_BIT 8 +#define WM8990_L34MNBST 0x0080 /* L34MNBST */ +#define WM8990_L34MNBST_BIT 7 +#define WM8990_L12MNB 0x0020 /* L12MNB */ +#define WM8990_L12MNB_BIT 5 +#define WM8990_L12MNBST 0x0010 /* L12MNBST */ +#define WM8990_L12MNBST_BIT 4 +#define WM8990_LDBVOL_MASK 0x0007 /* LDBVOL - [2:0] */ +#define WM8990_LDBVOL_SHIFT 0 + +/* + * R42 (0x2A) - Input Mixer4 + */ +#define WM8990_R34MNB 0x0100 /* R34MNB */ +#define WM8990_R34MNB_BIT 8 +#define WM8990_R34MNBST 0x0080 /* R34MNBST */ +#define WM8990_R34MNBST_BIT 7 +#define WM8990_R12MNB 0x0020 /* R12MNB */ +#define WM8990_R12MNB_BIT 5 +#define WM8990_R12MNBST 0x0010 /* R12MNBST */ +#define WM8990_R12MNBST_BIT 4 +#define WM8990_RDBVOL_MASK 0x0007 /* RDBVOL - [2:0] */ +#define WM8990_RDBVOL_SHIFT 0 + +/* + * R43 (0x2B) - Input Mixer5 + */ +#define WM8990_LI2BVOL_MASK 0x07 /* LI2BVOL - [8:6] */ +#define WM8990_LI2BVOL_SHIFT 6 +#define WM8990_LR4BVOL_MASK 0x07 /* LR4BVOL - [5:3] */ +#define WM8990_LR4BVOL_SHIFT 3 +#define WM8990_LL4BVOL_MASK 0x07 /* LL4BVOL - [2:0] */ +#define WM8990_LL4BVOL_SHIFT 0 + +/* + * R44 (0x2C) - Input Mixer6 + */ +#define WM8990_RI2BVOL_MASK 0x07 /* RI2BVOL - [8:6] */ +#define WM8990_RI2BVOL_SHIFT 6 +#define WM8990_RL4BVOL_MASK 0x07 /* RL4BVOL - [5:3] */ +#define WM8990_RL4BVOL_SHIFT 3 +#define WM8990_RR4BVOL_MASK 0x07 /* RR4BVOL - [2:0] */ +#define WM8990_RR4BVOL_SHIFT 0 + +/* + * R45 (0x2D) - Output Mixer1 + */ +#define WM8990_LRBLO 0x0080 /* LRBLO */ +#define WM8990_LRBLO_BIT 7 +#define WM8990_LLBLO 0x0040 /* LLBLO */ +#define WM8990_LLBLO_BIT 6 +#define WM8990_LRI3LO 0x0020 /* LRI3LO */ +#define WM8990_LRI3LO_BIT 5 +#define WM8990_LLI3LO 0x0010 /* LLI3LO */ +#define WM8990_LLI3LO_BIT 4 +#define WM8990_LR12LO 0x0008 /* LR12LO */ +#define WM8990_LR12LO_BIT 3 +#define WM8990_LL12LO 0x0004 /* LL12LO */ +#define WM8990_LL12LO_BIT 2 +#define WM8990_LDLO 0x0001 /* LDLO */ +#define WM8990_LDLO_BIT 0 + +/* + * R46 (0x2E) - Output Mixer2 + */ +#define WM8990_RLBRO 0x0080 /* RLBRO */ +#define WM8990_RLBRO_BIT 7 +#define WM8990_RRBRO 0x0040 /* RRBRO */ +#define WM8990_RRBRO_BIT 6 +#define WM8990_RLI3RO 0x0020 /* RLI3RO */ +#define WM8990_RLI3RO_BIT 5 +#define WM8990_RRI3RO 0x0010 /* RRI3RO */ +#define WM8990_RRI3RO_BIT 4 +#define WM8990_RL12RO 0x0008 /* RL12RO */ +#define WM8990_RL12RO_BIT 3 +#define WM8990_RR12RO 0x0004 /* RR12RO */ +#define WM8990_RR12RO_BIT 2 +#define WM8990_RDRO 0x0001 /* RDRO */ +#define WM8990_RDRO_BIT 0 + +/* + * R47 (0x2F) - Output Mixer3 + */ +#define WM8990_LLI3LOVOL_MASK 0x07 /* LLI3LOVOL - [8:6] */ +#define WM8990_LLI3LOVOL_SHIFT 6 +#define WM8990_LR12LOVOL_MASK 0x07 /* LR12LOVOL - [5:3] */ +#define WM8990_LR12LOVOL_SHIFT 3 +#define WM8990_LL12LOVOL_MASK 0x07 /* LL12LOVOL - [2:0] */ +#define WM8990_LL12LOVOL_SHIFT 0 + +/* + * R48 (0x30) - Output Mixer4 + */ +#define WM8990_RRI3ROVOL_MASK 0x07 /* RRI3ROVOL - [8:6] */ +#define WM8990_RRI3ROVOL_SHIFT 6 +#define WM8990_RL12ROVOL_MASK 0x07 /* RL12ROVOL - [5:3] */ +#define WM8990_RL12ROVOL_SHIFT 3 +#define WM8990_RR12ROVOL_MASK 0x07 /* RR12ROVOL - [2:0] */ +#define WM8990_RR12ROVOL_SHIFT 0 + +/* + * R49 (0x31) - Output Mixer5 + */ +#define WM8990_LRI3LOVOL_MASK 0x07 /* LRI3LOVOL - [8:6] */ +#define WM8990_LRI3LOVOL_SHIFT 6 +#define WM8990_LRBLOVOL_MASK 0x07 /* LRBLOVOL - [5:3] */ +#define WM8990_LRBLOVOL_SHIFT 3 +#define WM8990_LLBLOVOL_MASK 0x07 /* LLBLOVOL - [2:0] */ +#define WM8990_LLBLOVOL_SHIFT 0 + +/* + * R50 (0x32) - Output Mixer6 + */ +#define WM8990_RLI3ROVOL_MASK 0x07 /* RLI3ROVOL - [8:6] */ +#define WM8990_RLI3ROVOL_SHIFT 6 +#define WM8990_RLBROVOL_MASK 0x07 /* RLBROVOL - [5:3] */ +#define WM8990_RLBROVOL_SHIFT 3 +#define WM8990_RRBROVOL_MASK 0x07 /* RRBROVOL - [2:0] */ +#define WM8990_RRBROVOL_SHIFT 0 + +/* + * R51 (0x33) - Out3/4 Mixer + */ +#define WM8990_VSEL_MASK 0x0180 /* VSEL - [8:7] */ +#define WM8990_LI4O3 0x0020 /* LI4O3 */ +#define WM8990_LI4O3_BIT 5 +#define WM8990_LPGAO3 0x0010 /* LPGAO3 */ +#define WM8990_LPGAO3_BIT 4 +#define WM8990_RI4O4 0x0002 /* RI4O4 */ +#define WM8990_RI4O4_BIT 1 +#define WM8990_RPGAO4 0x0001 /* RPGAO4 */ +#define WM8990_RPGAO4_BIT 0 +/* + * R52 (0x34) - Line Mixer1 + */ +#define WM8990_LLOPGALON 0x0040 /* LLOPGALON */ +#define WM8990_LLOPGALON_BIT 6 +#define WM8990_LROPGALON 0x0020 /* LROPGALON */ +#define WM8990_LROPGALON_BIT 5 +#define WM8990_LOPLON 0x0010 /* LOPLON */ +#define WM8990_LOPLON_BIT 4 +#define WM8990_LR12LOP 0x0004 /* LR12LOP */ +#define WM8990_LR12LOP_BIT 2 +#define WM8990_LL12LOP 0x0002 /* LL12LOP */ +#define WM8990_LL12LOP_BIT 1 +#define WM8990_LLOPGALOP 0x0001 /* LLOPGALOP */ +#define WM8990_LLOPGALOP_BIT 0 +/* + * R53 (0x35) - Line Mixer2 + */ +#define WM8990_RROPGARON 0x0040 /* RROPGARON */ +#define WM8990_RROPGARON_BIT 6 +#define WM8990_RLOPGARON 0x0020 /* RLOPGARON */ +#define WM8990_RLOPGARON_BIT 5 +#define WM8990_ROPRON 0x0010 /* ROPRON */ +#define WM8990_ROPRON_BIT 4 +#define WM8990_RL12ROP 0x0004 /* RL12ROP */ +#define WM8990_RL12ROP_BIT 2 +#define WM8990_RR12ROP 0x0002 /* RR12ROP */ +#define WM8990_RR12ROP_BIT 1 +#define WM8990_RROPGAROP 0x0001 /* RROPGAROP */ +#define WM8990_RROPGAROP_BIT 0 + +/* + * R54 (0x36) - Speaker Mixer + */ +#define WM8990_LB2SPK 0x0080 /* LB2SPK */ +#define WM8990_LB2SPK_BIT 7 +#define WM8990_RB2SPK 0x0040 /* RB2SPK */ +#define WM8990_RB2SPK_BIT 6 +#define WM8990_LI2SPK 0x0020 /* LI2SPK */ +#define WM8990_LI2SPK_BIT 5 +#define WM8990_RI2SPK 0x0010 /* RI2SPK */ +#define WM8990_RI2SPK_BIT 4 +#define WM8990_LOPGASPK 0x0008 /* LOPGASPK */ +#define WM8990_LOPGASPK_BIT 3 +#define WM8990_ROPGASPK 0x0004 /* ROPGASPK */ +#define WM8990_ROPGASPK_BIT 2 +#define WM8990_LDSPK 0x0002 /* LDSPK */ +#define WM8990_LDSPK_BIT 1 +#define WM8990_RDSPK 0x0001 /* RDSPK */ +#define WM8990_RDSPK_BIT 0 + +/* + * R55 (0x37) - Additional Control + */ +#define WM8990_VROI 0x0001 /* VROI */ + +/* + * R56 (0x38) - AntiPOP1 + */ +#define WM8990_DIS_LLINE 0x0020 /* DIS_LLINE */ +#define WM8990_DIS_RLINE 0x0010 /* DIS_RLINE */ +#define WM8990_DIS_OUT3 0x0008 /* DIS_OUT3 */ +#define WM8990_DIS_OUT4 0x0004 /* DIS_OUT4 */ +#define WM8990_DIS_LOUT 0x0002 /* DIS_LOUT */ +#define WM8990_DIS_ROUT 0x0001 /* DIS_ROUT */ + +/* + * R57 (0x39) - AntiPOP2 + */ +#define WM8990_SOFTST 0x0040 /* SOFTST */ +#define WM8990_BUFIOEN 0x0008 /* BUFIOEN */ +#define WM8990_BUFDCOPEN 0x0004 /* BUFDCOPEN */ +#define WM8990_POBCTRL 0x0002 /* POBCTRL */ +#define WM8990_VMIDTOG 0x0001 /* VMIDTOG */ + +/* + * R58 (0x3A) - MICBIAS + */ +#define WM8990_MCDSCTH_MASK 0x00C0 /* MCDSCTH - [7:6] */ +#define WM8990_MCDTHR_MASK 0x0038 /* MCDTHR - [5:3] */ +#define WM8990_MCD 0x0004 /* MCD */ +#define WM8990_MBSEL 0x0001 /* MBSEL */ + +/* + * R60 (0x3C) - PLL1 + */ +#define WM8990_SDM 0x0080 /* SDM */ +#define WM8990_PRESCALE 0x0040 /* PRESCALE */ +#define WM8990_PLLN_MASK 0x000F /* PLLN - [3:0] */ + +/* + * R61 (0x3D) - PLL2 + */ +#define WM8990_PLLK1_MASK 0x00FF /* PLLK1 - [7:0] */ + +/* + * R62 (0x3E) - PLL3 + */ +#define WM8990_PLLK2_MASK 0x00FF /* PLLK2 - [7:0] */ + +/* + * R63 (0x3F) - Internal Driver Bits + */ +#define WM8990_INMIXL_PWR_BIT 0 +#define WM8990_AINLMUX_PWR_BIT 1 +#define WM8990_INMIXR_PWR_BIT 2 +#define WM8990_AINRMUX_PWR_BIT 3 + +struct wm8990_setup_data { + unsigned short i2c_address; +}; + +#define WM8990_MCLK_DIV 0 +#define WM8990_DACCLK_DIV 1 +#define WM8990_ADCCLK_DIV 2 +#define WM8990_BCLK_DIV 3 + +extern struct snd_soc_codec_dai wm8990_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8990; + +#endif /* __WM8990REGISTERDEFS_H__ */ +/*------------------------------ END OF FILE ---------------------------------*/ -- cgit From 9aaca9683b014c4d718f32cbddb42e5b36ea3c0f Mon Sep 17 00:00:00 2001 From: Geoffrey Wossum Date: Thu, 5 Jun 2008 13:49:34 +0100 Subject: [ALSA] Revised AT32 ASoC Patch Attached is a revised version of my patch to add AT32 to ASoC. This cleans most of the style issues associated with the previous patch. Also fixes an issue with the playpaq_wm8510.c code depending on a non-released patch to th AT32 portmux support. Patch is against 2.6.24.3.atmel.3 kernel, the latest AVR32 kernel Atmel has released, with the linux-2.6-asoc patches from when v2.6.24 was tagged also applied. [Fixed up minor checkpatch issues and updated for current kernels -- broonie] Signed-off-by: Geoffrey Wossum Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/Kconfig | 1 + sound/soc/Makefile | 3 +- sound/soc/at32/Kconfig | 34 ++ sound/soc/at32/Makefile | 11 + sound/soc/at32/at32-pcm.c | 491 +++++++++++++++++++++++ sound/soc/at32/at32-pcm.h | 79 ++++ sound/soc/at32/at32-ssc.c | 849 ++++++++++++++++++++++++++++++++++++++++ sound/soc/at32/at32-ssc.h | 59 +++ sound/soc/at32/playpaq_wm8510.c | 524 +++++++++++++++++++++++++ 9 files changed, 2050 insertions(+), 1 deletion(-) create mode 100644 sound/soc/at32/Kconfig create mode 100644 sound/soc/at32/Makefile create mode 100644 sound/soc/at32/at32-pcm.c create mode 100644 sound/soc/at32/at32-pcm.h create mode 100644 sound/soc/at32/at32-ssc.c create mode 100644 sound/soc/at32/at32-ssc.h create mode 100644 sound/soc/at32/playpaq_wm8510.c (limited to 'sound/soc') diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index fd7bc4f8907..b939e22db7b 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -22,6 +22,7 @@ config SND_SOC_AC97_BUS bool # All the supported Soc's +source "sound/soc/at32/Kconfig" source "sound/soc/at91/Kconfig" source "sound/soc/pxa/Kconfig" source "sound/soc/s3c24xx/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 782db212710..3645f959c26 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,4 +1,5 @@ snd-soc-core-objs := soc-core.o soc-dapm.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o -obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ omap/ +obj-$(CONFIG_SND_SOC) += codecs/ at32/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ +obj-$(CONFIG_SND_SOC) += omap/ diff --git a/sound/soc/at32/Kconfig b/sound/soc/at32/Kconfig new file mode 100644 index 00000000000..b0765e86c08 --- /dev/null +++ b/sound/soc/at32/Kconfig @@ -0,0 +1,34 @@ +config SND_AT32_SOC + tristate "SoC Audio for the Atmel AT32 System-on-a-Chip" + depends on AVR32 && SND_SOC + help + Say Y or M if you want to add support for codecs attached to + the AT32 SSC interface. You will also need to + to select the audio interfaces to support below. + + +config SND_AT32_SOC_SSC + tristate + + + +config SND_AT32_SOC_PLAYPAQ + tristate "SoC Audio support for PlayPaq with WM8510" + depends on SND_AT32_SOC && BOARD_PLAYPAQ + select SND_AT32_SOC_SSC + select SND_SOC_WM8510 + help + Say Y or M here if you want to add support for SoC audio + on the LRS PlayPaq. + + + +config SND_AT32_SOC_PLAYPAQ_SLAVE + bool "Run CODEC on PlayPaq in slave mode" + depends on SND_AT32_SOC_PLAYPAQ + default n + help + Say Y if you want to run with the AT32 SSC generating the BCLK + and FRAME signals on the PlayPaq. Unless you want to play + with the AT32 as the SSC master, you probably want to say N here, + as this will give you better sound quality. diff --git a/sound/soc/at32/Makefile b/sound/soc/at32/Makefile new file mode 100644 index 00000000000..c03e55ecece --- /dev/null +++ b/sound/soc/at32/Makefile @@ -0,0 +1,11 @@ +# AT32 Platform Support +snd-soc-at32-objs := at32-pcm.o +snd-soc-at32-ssc-objs := at32-ssc.o + +obj-$(CONFIG_SND_AT32_SOC) += snd-soc-at32.o +obj-$(CONFIG_SND_AT32_SOC_SSC) += snd-soc-at32-ssc.o + +# AT32 Machine Support +snd-soc-playpaq-objs := playpaq_wm8510.o + +obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o diff --git a/sound/soc/at32/at32-pcm.c b/sound/soc/at32/at32-pcm.c new file mode 100644 index 00000000000..cf76e89fae6 --- /dev/null +++ b/sound/soc/at32/at32-pcm.c @@ -0,0 +1,491 @@ +/* sound/soc/at32/at32-pcm.c + * ASoC PCM interface for Atmel AT32 SoC + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Note that this is basically a port of the sound/soc/at91-pcm.c to + * the AVR32 kernel. Thanks to Frank Mandarino for that code. + */ + +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include + +#include "at32-pcm.h" + + + +/*--------------------------------------------------------------------------*\ + * Hardware definition +\*--------------------------------------------------------------------------*/ +/* TODO: These values were taken from the AT91 platform driver, check + * them against real values for AT32 + */ +static const struct snd_pcm_hardware at32_pcm_hardware = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_PAUSE), + + .formats = SNDRV_PCM_FMTBIT_S16, + .period_bytes_min = 32, + .period_bytes_max = 8192, /* 512 frames * 16 bytes / frame */ + .periods_min = 2, + .periods_max = 1024, + .buffer_bytes_max = 32 * 1024, +}; + + + +/*--------------------------------------------------------------------------*\ + * Data types +\*--------------------------------------------------------------------------*/ +struct at32_runtime_data { + struct at32_pcm_dma_params *params; + dma_addr_t dma_buffer; /* physical address of DMA buffer */ + dma_addr_t dma_buffer_end; /* first address beyond DMA buffer */ + size_t period_size; + + dma_addr_t period_ptr; /* physical address of next period */ + int periods; /* period index of period_ptr */ + + /* Save PDC registers (for power management) */ + u32 pdc_xpr_save; + u32 pdc_xcr_save; + u32 pdc_xnpr_save; + u32 pdc_xncr_save; +}; + + + +/*--------------------------------------------------------------------------*\ + * Helper functions +\*--------------------------------------------------------------------------*/ +static int at32_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *dmabuf = &substream->dma_buffer; + size_t size = at32_pcm_hardware.buffer_bytes_max; + + dmabuf->dev.type = SNDRV_DMA_TYPE_DEV; + dmabuf->dev.dev = pcm->card->dev; + dmabuf->private_data = NULL; + dmabuf->area = dma_alloc_coherent(pcm->card->dev, size, + &dmabuf->addr, GFP_KERNEL); + pr_debug("at32_pcm: preallocate_dma_buffer: " + "area=%p, addr=%p, size=%ld\n", + (void *)dmabuf->area, (void *)dmabuf->addr, size); + + if (!dmabuf->area) + return -ENOMEM; + + dmabuf->bytes = size; + return 0; +} + + + +/*--------------------------------------------------------------------------*\ + * ISR +\*--------------------------------------------------------------------------*/ +static void at32_pcm_dma_irq(u32 ssc_sr, struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *rtd = substream->runtime; + struct at32_runtime_data *prtd = rtd->private_data; + struct at32_pcm_dma_params *params = prtd->params; + static int count; + + count++; + if (ssc_sr & params->mask->ssc_endbuf) { + pr_warning("at32-pcm: buffer %s on %s (SSC_SR=%#x, count=%d)\n", + substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + "underrun" : "overrun", params->name, ssc_sr, count); + + /* re-start the PDC */ + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + prtd->period_ptr += prtd->period_size; + if (prtd->period_ptr >= prtd->dma_buffer_end) + prtd->period_ptr = prtd->dma_buffer; + + + ssc_writex(params->ssc->regs, params->pdc->xpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xcr, + prtd->period_size / params->pdc_xfer_size); + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_enable); + } + + + if (ssc_sr & params->mask->ssc_endx) { + /* Load the PDC next pointer and counter registers */ + prtd->period_ptr += prtd->period_size; + if (prtd->period_ptr >= prtd->dma_buffer_end) + prtd->period_ptr = prtd->dma_buffer; + ssc_writex(params->ssc->regs, params->pdc->xnpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xncr, + prtd->period_size / params->pdc_xfer_size); + } + + + snd_pcm_period_elapsed(substream); +} + + + +/*--------------------------------------------------------------------------*\ + * PCM operations +\*--------------------------------------------------------------------------*/ +static int at32_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct at32_runtime_data *prtd = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + /* this may get called several times by oss emulation + * with different params + */ + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = params_buffer_bytes(params); + + prtd->params = rtd->dai->cpu_dai->dma_data; + prtd->params->dma_intr_handler = at32_pcm_dma_irq; + + prtd->dma_buffer = runtime->dma_addr; + prtd->dma_buffer_end = runtime->dma_addr + runtime->dma_bytes; + prtd->period_size = params_period_bytes(params); + + pr_debug("hw_params: DMA for %s initialized " + "(dma_bytes=%ld, period_size=%ld)\n", + prtd->params->name, runtime->dma_bytes, prtd->period_size); + + return 0; +} + + + +static int at32_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct at32_runtime_data *prtd = substream->runtime->private_data; + struct at32_pcm_dma_params *params = prtd->params; + + if (params != NULL) { + ssc_writex(params->ssc->regs, SSC_PDC_PTCR, + params->mask->pdc_disable); + prtd->params->dma_intr_handler = NULL; + } + + return 0; +} + + + +static int at32_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct at32_runtime_data *prtd = substream->runtime->private_data; + struct at32_pcm_dma_params *params = prtd->params; + + ssc_writex(params->ssc->regs, SSC_IDR, + params->mask->ssc_endx | params->mask->ssc_endbuf); + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + + return 0; +} + + +static int at32_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_pcm_runtime *rtd = substream->runtime; + struct at32_runtime_data *prtd = rtd->private_data; + struct at32_pcm_dma_params *params = prtd->params; + int ret = 0; + + pr_debug("at32_pcm_trigger: buffer_size = %ld, " + "dma_area = %p, dma_bytes = %ld\n", + rtd->buffer_size, rtd->dma_area, rtd->dma_bytes); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + prtd->period_ptr = prtd->dma_buffer; + + ssc_writex(params->ssc->regs, params->pdc->xpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xcr, + prtd->period_size / params->pdc_xfer_size); + + prtd->period_ptr += prtd->period_size; + ssc_writex(params->ssc->regs, params->pdc->xnpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xncr, + prtd->period_size / params->pdc_xfer_size); + + pr_debug("trigger: period_ptr=%lx, xpr=%x, " + "xcr=%d, xnpr=%x, xncr=%d\n", + (unsigned long)prtd->period_ptr, + ssc_readx(params->ssc->regs, params->pdc->xpr), + ssc_readx(params->ssc->regs, params->pdc->xcr), + ssc_readx(params->ssc->regs, params->pdc->xnpr), + ssc_readx(params->ssc->regs, params->pdc->xncr)); + + ssc_writex(params->ssc->regs, SSC_IER, + params->mask->ssc_endx | params->mask->ssc_endbuf); + ssc_writex(params->ssc->regs, SSC_PDC_PTCR, + params->mask->pdc_enable); + + pr_debug("sr=%x, imr=%x\n", + ssc_readx(params->ssc->regs, SSC_SR), + ssc_readx(params->ssc->regs, SSC_IER)); + break; /* SNDRV_PCM_TRIGGER_START */ + + + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + break; + + + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_enable); + break; + + default: + ret = -EINVAL; + } + + return ret; +} + + + +static snd_pcm_uframes_t at32_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct at32_runtime_data *prtd = runtime->private_data; + struct at32_pcm_dma_params *params = prtd->params; + dma_addr_t ptr; + snd_pcm_uframes_t x; + + ptr = (dma_addr_t) ssc_readx(params->ssc->regs, params->pdc->xpr); + x = bytes_to_frames(runtime, ptr - prtd->dma_buffer); + + if (x == runtime->buffer_size) + x = 0; + + return x; +} + + + +static int at32_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct at32_runtime_data *prtd; + int ret = 0; + + snd_soc_set_runtime_hwparams(substream, &at32_pcm_hardware); + + /* ensure that buffer size is a multiple of period size */ + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + goto out; + + prtd = kzalloc(sizeof(*prtd), GFP_KERNEL); + if (prtd == NULL) { + ret = -ENOMEM; + goto out; + } + runtime->private_data = prtd; + + +out: + return ret; +} + + + +static int at32_pcm_close(struct snd_pcm_substream *substream) +{ + struct at32_runtime_data *prtd = substream->runtime->private_data; + + kfree(prtd); + return 0; +} + + +static int at32_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + return remap_pfn_range(vma, vma->vm_start, + substream->dma_buffer.addr >> PAGE_SHIFT, + vma->vm_end - vma->vm_start, vma->vm_page_prot); +} + + + +static struct snd_pcm_ops at32_pcm_ops = { + .open = at32_pcm_open, + .close = at32_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = at32_pcm_hw_params, + .hw_free = at32_pcm_hw_free, + .prepare = at32_pcm_prepare, + .trigger = at32_pcm_trigger, + .pointer = at32_pcm_pointer, + .mmap = at32_pcm_mmap, +}; + + + +/*--------------------------------------------------------------------------*\ + * ASoC platform driver +\*--------------------------------------------------------------------------*/ +static u64 at32_pcm_dmamask = 0xffffffff; + +static int at32_pcm_new(struct snd_card *card, + struct snd_soc_codec_dai *dai, + struct snd_pcm *pcm) +{ + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &at32_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = 0xffffffff; + + if (dai->playback.channels_min) { + ret = at32_pcm_preallocate_dma_buffer( + pcm, SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + } + + if (dai->capture.channels_min) { + pr_debug("at32-pcm: Allocating PCM capture DMA buffer\n"); + ret = at32_pcm_preallocate_dma_buffer( + pcm, SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + } + + +out: + return ret; +} + + + +static void at32_pcm_free_dma_buffers(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (substream == NULL) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + dma_free_coherent(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + + + +#ifdef CONFIG_PM +static int at32_pcm_suspend(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) +{ + struct snd_pcm_runtime *runtime = dai->runtime; + struct at32_runtime_data *prtd; + struct at32_pcm_dma_params *params; + + if (runtime == NULL) + return 0; + prtd = runtime->private_data; + params = prtd->params; + + /* Disable the PDC and save the PDC registers */ + ssc_writex(params->ssc->regs, PDC_PTCR, params->mask->pdc_disable); + + prtd->pdc_xpr_save = ssc_readx(params->ssc->regs, params->pdc->xpr); + prtd->pdc_xcr_save = ssc_readx(params->ssc->regs, params->pdc->xcr); + prtd->pdc_xnpr_save = ssc_readx(params->ssc->regs, params->pdc->xnpr); + prtd->pdc_xncr_save = ssc_readx(params->ssc->regs, params->pdc->xncr); + + return 0; +} + + + +static int at32_pcm_resume(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) +{ + struct snd_pcm_runtime *runtime = dai->runtime; + struct at32_runtime_data *prtd; + struct at32_pcm_dma_params *params; + + if (runtime == NULL) + return 0; + prtd = runtime->private_data; + params = prtd->params; + + /* Restore the PDC registers and enable the PDC */ + ssc_writex(params->ssc->regs, params->pdc->xpr, prtd->pdc_xpr_save); + ssc_writex(params->ssc->regs, params->pdc->xcr, prtd->pdc_xcr_save); + ssc_writex(params->ssc->regs, params->pdc->xnpr, prtd->pdc_xnpr_save); + ssc_writex(params->ssc->regs, params->pdc->xncr, prtd->pdc_xncr_save); + + ssc_writex(params->ssc->regs, PDC_PTCR, params->mask->pdc_enable); + return 0; +} +#else /* CONFIG_PM */ +# define at32_pcm_suspend NULL +# define at32_pcm_resume NULL +#endif /* CONFIG_PM */ + + + +struct snd_soc_platform at32_soc_platform = { + .name = "at32-audio", + .pcm_ops = &at32_pcm_ops, + .pcm_new = at32_pcm_new, + .pcm_free = at32_pcm_free_dma_buffers, + .suspend = at32_pcm_suspend, + .resume = at32_pcm_resume, +}; +EXPORT_SYMBOL_GPL(at32_soc_platform); + + + +MODULE_AUTHOR("Geoffrey Wossum "); +MODULE_DESCRIPTION("Atmel AT32 PCM module"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/at32/at32-pcm.h b/sound/soc/at32/at32-pcm.h new file mode 100644 index 00000000000..2a52430417d --- /dev/null +++ b/sound/soc/at32/at32-pcm.h @@ -0,0 +1,79 @@ +/* sound/soc/at32/at32-pcm.h + * ASoC PCM interface for Atmel AT32 SoC + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __SOUND_SOC_AT32_AT32_PCM_H +#define __SOUND_SOC_AT32_AT32_PCM_H __FILE__ + +#include + + +/* + * Registers and status bits that are required by the PCM driver + * TODO: Is ptcr really used? + */ +struct at32_pdc_regs { + u32 xpr; /* PDC RX/TX pointer */ + u32 xcr; /* PDC RX/TX counter */ + u32 xnpr; /* PDC next RX/TX pointer */ + u32 xncr; /* PDC next RX/TX counter */ + u32 ptcr; /* PDC transfer control */ +}; + + + +/* + * SSC mask info + */ +struct at32_ssc_mask { + u32 ssc_enable; /* SSC RX/TX enable */ + u32 ssc_disable; /* SSC RX/TX disable */ + u32 ssc_endx; /* SSC ENDTX or ENDRX */ + u32 ssc_endbuf; /* SSC TXBUFF or RXBUFF */ + u32 pdc_enable; /* PDC RX/TX enable */ + u32 pdc_disable; /* PDC RX/TX disable */ +}; + + + +/* + * This structure, shared between the PCM driver and the interface, + * contains all information required by the PCM driver to perform the + * PDC DMA operation. All fields except dma_intr_handler() are initialized + * by the interface. The dms_intr_handler() pointer is set by the PCM + * driver and called by the interface SSC interrupt handler if it is + * non-NULL. + */ +struct at32_pcm_dma_params { + char *name; /* stream identifier */ + int pdc_xfer_size; /* PDC counter increment in bytes */ + struct ssc_device *ssc; /* SSC device for stream */ + struct at32_pdc_regs *pdc; /* PDC register info */ + struct at32_ssc_mask *mask; /* SSC mask info */ + struct snd_pcm_substream *substream; + void (*dma_intr_handler) (u32, struct snd_pcm_substream *); +}; + + + +/* + * The AT32 ASoC platform driver + */ +extern struct snd_soc_platform at32_soc_platform; + + + +/* + * SSC register access (since ssc_writel() / ssc_readl() require literal name) + */ +#define ssc_readx(base, reg) (__raw_readl((base) + (reg))) +#define ssc_writex(base, reg, value) __raw_writel((value), (base) + (reg)) + +#endif /* __SOUND_SOC_AT32_AT32_PCM_H */ diff --git a/sound/soc/at32/at32-ssc.c b/sound/soc/at32/at32-ssc.c new file mode 100644 index 00000000000..0ca44107f18 --- /dev/null +++ b/sound/soc/at32/at32-ssc.c @@ -0,0 +1,849 @@ +/* sound/soc/at32/at32-ssc.c + * ASoC platform driver for AT32 using SSC as DAI + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Note that this is basically a port of the sound/soc/at91-ssc.c to + * the AVR32 kernel. Thanks to Frank Mandarino for that code. + */ + +/* #define DEBUG */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +#include "at32-pcm.h" +#include "at32-ssc.h" + + + +/*-------------------------------------------------------------------------*\ + * Constants +\*-------------------------------------------------------------------------*/ +#define NUM_SSC_DEVICES 3 + +/* + * SSC direction masks + */ +#define SSC_DIR_MASK_UNUSED 0 +#define SSC_DIR_MASK_PLAYBACK 1 +#define SSC_DIR_MASK_CAPTURE 2 + +/* + * SSC register values that Atmel left out of . These + * are expected to be used with SSC_BF + */ +/* START bit field values */ +#define SSC_START_CONTINUOUS 0 +#define SSC_START_TX_RX 1 +#define SSC_START_LOW_RF 2 +#define SSC_START_HIGH_RF 3 +#define SSC_START_FALLING_RF 4 +#define SSC_START_RISING_RF 5 +#define SSC_START_LEVEL_RF 6 +#define SSC_START_EDGE_RF 7 +#define SSS_START_COMPARE_0 8 + +/* CKI bit field values */ +#define SSC_CKI_FALLING 0 +#define SSC_CKI_RISING 1 + +/* CKO bit field values */ +#define SSC_CKO_NONE 0 +#define SSC_CKO_CONTINUOUS 1 +#define SSC_CKO_TRANSFER 2 + +/* CKS bit field values */ +#define SSC_CKS_DIV 0 +#define SSC_CKS_CLOCK 1 +#define SSC_CKS_PIN 2 + +/* FSEDGE bit field values */ +#define SSC_FSEDGE_POSITIVE 0 +#define SSC_FSEDGE_NEGATIVE 1 + +/* FSOS bit field values */ +#define SSC_FSOS_NONE 0 +#define SSC_FSOS_NEGATIVE 1 +#define SSC_FSOS_POSITIVE 2 +#define SSC_FSOS_LOW 3 +#define SSC_FSOS_HIGH 4 +#define SSC_FSOS_TOGGLE 5 + +#define START_DELAY 1 + + + +/*-------------------------------------------------------------------------*\ + * Module data +\*-------------------------------------------------------------------------*/ +/* + * SSC PDC registered required by the PCM DMA engine + */ +static struct at32_pdc_regs pdc_tx_reg = { + .xpr = SSC_PDC_TPR, + .xcr = SSC_PDC_TCR, + .xnpr = SSC_PDC_TNPR, + .xncr = SSC_PDC_TNCR, +}; + + + +static struct at32_pdc_regs pdc_rx_reg = { + .xpr = SSC_PDC_RPR, + .xcr = SSC_PDC_RCR, + .xnpr = SSC_PDC_RNPR, + .xncr = SSC_PDC_RNCR, +}; + + + +/* + * SSC and PDC status bits for transmit and receive + */ +static struct at32_ssc_mask ssc_tx_mask = { + .ssc_enable = SSC_BIT(CR_TXEN), + .ssc_disable = SSC_BIT(CR_TXDIS), + .ssc_endx = SSC_BIT(SR_ENDTX), + .ssc_endbuf = SSC_BIT(SR_TXBUFE), + .pdc_enable = SSC_BIT(PDC_PTCR_TXTEN), + .pdc_disable = SSC_BIT(PDC_PTCR_TXTDIS), +}; + + + +static struct at32_ssc_mask ssc_rx_mask = { + .ssc_enable = SSC_BIT(CR_RXEN), + .ssc_disable = SSC_BIT(CR_RXDIS), + .ssc_endx = SSC_BIT(SR_ENDRX), + .ssc_endbuf = SSC_BIT(SR_RXBUFF), + .pdc_enable = SSC_BIT(PDC_PTCR_RXTEN), + .pdc_disable = SSC_BIT(PDC_PTCR_RXTDIS), +}; + + + +/* + * DMA parameters for each SSC + */ +static struct at32_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = { + { + { + .name = "SSC0 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC0 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + }, + }, + { + { + .name = "SSC1 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC1 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + }, + }, + { + { + .name = "SSC2 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC2 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + }, + }, +}; + + + +static struct at32_ssc_info ssc_info[NUM_SSC_DEVICES] = { + { + .name = "ssc0", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, + { + .name = "ssc1", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, + { + .name = "ssc2", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, +}; + + + + +/*-------------------------------------------------------------------------*\ + * ISR +\*-------------------------------------------------------------------------*/ +/* + * SSC interrupt handler. Passes PDC interrupts to the DMA interrupt + * handler in the PCM driver. + */ +static irqreturn_t at32_ssc_interrupt(int irq, void *dev_id) +{ + struct at32_ssc_info *ssc_p = dev_id; + struct at32_pcm_dma_params *dma_params; + u32 ssc_sr; + u32 ssc_substream_mask; + int i; + + ssc_sr = (ssc_readl(ssc_p->ssc->regs, SR) & + ssc_readl(ssc_p->ssc->regs, IMR)); + + /* + * Loop through substreams attached to this SSC. If a DMA-related + * interrupt occured on that substream, call the DMA interrupt + * handler function, if one has been registered in the dma_param + * structure by the PCM driver. + */ + for (i = 0; i < ARRAY_SIZE(ssc_p->dma_params); i++) { + dma_params = ssc_p->dma_params[i]; + + if ((dma_params != NULL) && + (dma_params->dma_intr_handler != NULL)) { + ssc_substream_mask = (dma_params->mask->ssc_endx | + dma_params->mask->ssc_endbuf); + if (ssc_sr & ssc_substream_mask) { + dma_params->dma_intr_handler(ssc_sr, + dma_params-> + substream); + } + } + } + + + return IRQ_HANDLED; +} + +/*-------------------------------------------------------------------------*\ + * DAI functions +\*-------------------------------------------------------------------------*/ +/* + * Startup. Only that one substream allowed in each direction. + */ +static int at32_ssc_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + int dir_mask; + + dir_mask = ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + SSC_DIR_MASK_PLAYBACK : SSC_DIR_MASK_CAPTURE); + + spin_lock_irq(&ssc_p->lock); + if (ssc_p->dir_mask & dir_mask) { + spin_unlock_irq(&ssc_p->lock); + return -EBUSY; + } + ssc_p->dir_mask |= dir_mask; + spin_unlock_irq(&ssc_p->lock); + + return 0; +} + + + +/* + * Shutdown. Clear DMA parameters and shutdown the SSC if there + * are no other substreams open. + */ +static void at32_ssc_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + struct at32_pcm_dma_params *dma_params; + int dir_mask; + + dma_params = ssc_p->dma_params[substream->stream]; + + if (dma_params != NULL) { + ssc_writel(dma_params->ssc->regs, CR, + dma_params->mask->ssc_disable); + pr_debug("%s disabled SSC_SR=0x%08x\n", + (substream->stream ? "receiver" : "transmit"), + ssc_readl(ssc_p->ssc->regs, SR)); + + dma_params->ssc = NULL; + dma_params->substream = NULL; + ssc_p->dma_params[substream->stream] = NULL; + } + + + dir_mask = 1 << substream->stream; + spin_lock_irq(&ssc_p->lock); + ssc_p->dir_mask &= ~dir_mask; + if (!ssc_p->dir_mask) { + /* Shutdown the SSC clock */ + pr_debug("at32-ssc: Stopping user %d clock\n", + ssc_p->ssc->user); + clk_disable(ssc_p->ssc->clk); + + if (ssc_p->initialized) { + free_irq(ssc_p->ssc->irq, ssc_p); + ssc_p->initialized = 0; + } + + /* Reset the SSC */ + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); + + /* clear the SSC dividers */ + ssc_p->cmr_div = 0; + ssc_p->tcmr_period = 0; + ssc_p->rcmr_period = 0; + } + spin_unlock_irq(&ssc_p->lock); +} + + + +/* + * Set the SSC system clock rate + */ +static int at32_ssc_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + /* TODO: What the heck do I do here? */ + return 0; +} + + + +/* + * Record DAI format for use by hw_params() + */ +static int at32_ssc_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, + unsigned int fmt) +{ + struct at32_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; + + ssc_p->daifmt = fmt; + return 0; +} + + + +/* + * Record SSC clock dividers for use in hw_params() + */ +static int at32_ssc_set_dai_clkdiv(struct snd_soc_cpu_dai *cpu_dai, + int div_id, int div) +{ + struct at32_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; + + switch (div_id) { + case AT32_SSC_CMR_DIV: + /* + * The same master clock divider is used for both + * transmit and receive, so if a value has already + * been set, it must match this value + */ + if (ssc_p->cmr_div == 0) + ssc_p->cmr_div = div; + else if (div != ssc_p->cmr_div) + return -EBUSY; + break; + + case AT32_SSC_TCMR_PERIOD: + ssc_p->tcmr_period = div; + break; + + case AT32_SSC_RCMR_PERIOD: + ssc_p->rcmr_period = div; + break; + + default: + return -EINVAL; + } + + return 0; +} + + + +/* + * Configure the SSC + */ +static int at32_ssc_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int id = rtd->dai->cpu_dai->id; + struct at32_ssc_info *ssc_p = &ssc_info[id]; + struct at32_pcm_dma_params *dma_params; + int channels, bits; + u32 tfmr, rfmr, tcmr, rcmr; + int start_event; + int ret; + + + /* + * Currently, there is only one set of dma_params for each direction. + * If more are added, this code will have to be changed to select + * the proper set + */ + dma_params = &ssc_dma_params[id][substream->stream]; + dma_params->ssc = ssc_p->ssc; + dma_params->substream = substream; + + ssc_p->dma_params[substream->stream] = dma_params; + + + /* + * The cpu_dai->dma_data field is only used to communicate the + * appropriate DMA parameters to the PCM driver's hw_params() + * function. It should not be used for other purposes as it + * is common to all substreams. + */ + rtd->dai->cpu_dai->dma_data = dma_params; + + channels = params_channels(params); + + + /* + * Determine sample size in bits and the PDC increment + */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + bits = 8; + dma_params->pdc_xfer_size = 1; + break; + + case SNDRV_PCM_FORMAT_S16: + bits = 16; + dma_params->pdc_xfer_size = 2; + break; + + case SNDRV_PCM_FORMAT_S24: + bits = 24; + dma_params->pdc_xfer_size = 4; + break; + + case SNDRV_PCM_FORMAT_S32: + bits = 32; + dma_params->pdc_xfer_size = 4; + break; + + default: + pr_warning("at32-ssc: Unsupported PCM format %d", + params_format(params)); + return -EINVAL; + } + pr_debug("at32-ssc: bits = %d, pdc_xfer_size = %d, channels = %d\n", + bits, dma_params->pdc_xfer_size, channels); + + + /* + * The SSC only supports up to 16-bit samples in I2S format, due + * to the size of the Frame Mode Register FSLEN field. + */ + if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S) + if (bits > 16) { + pr_warning("at32-ssc: " + "sample size %d is too large for I2S\n", + bits); + return -EINVAL; + } + + + /* + * Compute the SSC register settings + */ + switch (ssc_p->daifmt & (SND_SOC_DAIFMT_FORMAT_MASK | + SND_SOC_DAIFMT_MASTER_MASK)) { + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS: + /* + * I2S format, SSC provides BCLK and LRS clocks. + * + * The SSC transmit and receive clocks are generated from the + * MCK divider, and the BCLK signal is output on the SSC TK line + */ + pr_debug("at32-ssc: SSC mode is I2S BCLK / FRAME master\n"); + rcmr = (SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) | + SSC_BF(RCMR_STTDLY, START_DELAY) | + SSC_BF(RCMR_START, SSC_START_FALLING_RF) | + SSC_BF(RCMR_CKI, SSC_CKI_RISING) | + SSC_BF(RCMR_CKO, SSC_CKO_NONE) | + SSC_BF(RCMR_CKS, SSC_CKS_DIV)); + + rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(RFMR_FSOS, SSC_FSOS_NEGATIVE) | + SSC_BF(RFMR_FSLEN, bits - 1) | + SSC_BF(RFMR_DATNB, channels - 1) | + SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1)); + + tcmr = (SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) | + SSC_BF(TCMR_STTDLY, START_DELAY) | + SSC_BF(TCMR_START, SSC_START_FALLING_RF) | + SSC_BF(TCMR_CKI, SSC_CKI_FALLING) | + SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) | + SSC_BF(TCMR_CKS, SSC_CKS_DIV)); + + tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(TFMR_FSOS, SSC_FSOS_NEGATIVE) | + SSC_BF(TFMR_FSLEN, bits - 1) | + SSC_BF(TFMR_DATNB, channels - 1) | SSC_BIT(TFMR_MSBF) | + SSC_BF(TFMR_DATLEN, bits - 1)); + break; + + + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: + /* + * I2S format, CODEC supplies BCLK and LRC clock. + * + * The SSC transmit clock is obtained from the BCLK signal + * on the TK line, and the SSC receive clock is generated from + * the transmit clock. + * + * For single channel data, one sample is transferred on the + * falling edge of the LRC clock. For two channel data, one + * sample is transferred on both edges of the LRC clock. + */ + pr_debug("at32-ssc: SSC mode is I2S BCLK / FRAME slave\n"); + start_event = ((channels == 1) ? + SSC_START_FALLING_RF : SSC_START_EDGE_RF); + + rcmr = (SSC_BF(RCMR_STTDLY, START_DELAY) | + SSC_BF(RCMR_START, start_event) | + SSC_BF(RCMR_CKI, SSC_CKI_RISING) | + SSC_BF(RCMR_CKO, SSC_CKO_NONE) | + SSC_BF(RCMR_CKS, SSC_CKS_CLOCK)); + + rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(RFMR_FSOS, SSC_FSOS_NONE) | + SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1)); + + tcmr = (SSC_BF(TCMR_STTDLY, START_DELAY) | + SSC_BF(TCMR_START, start_event) | + SSC_BF(TCMR_CKI, SSC_CKI_FALLING) | + SSC_BF(TCMR_CKO, SSC_CKO_NONE) | + SSC_BF(TCMR_CKS, SSC_CKS_PIN)); + + tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(TFMR_FSOS, SSC_FSOS_NONE) | + SSC_BIT(TFMR_MSBF) | SSC_BF(TFMR_DATLEN, bits - 1)); + break; + + + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS: + /* + * DSP/PCM Mode A format, SSC provides BCLK and LRC clocks. + * + * The SSC transmit and receive clocks are generated from the + * MCK divider, and the BCLK signal is output on the SSC TK line + */ + pr_debug("at32-ssc: SSC mode is DSP A BCLK / FRAME master\n"); + rcmr = (SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) | + SSC_BF(RCMR_STTDLY, 1) | + SSC_BF(RCMR_START, SSC_START_RISING_RF) | + SSC_BF(RCMR_CKI, SSC_CKI_RISING) | + SSC_BF(RCMR_CKO, SSC_CKO_NONE) | + SSC_BF(RCMR_CKS, SSC_CKS_DIV)); + + rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(RFMR_FSOS, SSC_FSOS_POSITIVE) | + SSC_BF(RFMR_DATNB, channels - 1) | + SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1)); + + tcmr = (SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) | + SSC_BF(TCMR_STTDLY, 1) | + SSC_BF(TCMR_START, SSC_START_RISING_RF) | + SSC_BF(TCMR_CKI, SSC_CKI_RISING) | + SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) | + SSC_BF(TCMR_CKS, SSC_CKS_DIV)); + + tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(TFMR_FSOS, SSC_FSOS_POSITIVE) | + SSC_BF(TFMR_DATNB, channels - 1) | + SSC_BIT(TFMR_MSBF) | SSC_BF(TFMR_DATLEN, bits - 1)); + break; + + + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM: + default: + pr_warning("at32-ssc: unsupported DAI format 0x%x\n", + ssc_p->daifmt); + return -EINVAL; + break; + } + pr_debug("at32-ssc: RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n", + rcmr, rfmr, tcmr, tfmr); + + + if (!ssc_p->initialized) { + /* enable peripheral clock */ + pr_debug("at32-ssc: Starting clock\n"); + clk_enable(ssc_p->ssc->clk); + + /* Reset the SSC and its PDC registers */ + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); + + ssc_writel(ssc_p->ssc->regs, PDC_RPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RCR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RNPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RNCR, 0); + + ssc_writel(ssc_p->ssc->regs, PDC_TPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TCR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TNPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TNCR, 0); + + ret = request_irq(ssc_p->ssc->irq, at32_ssc_interrupt, 0, + ssc_p->name, ssc_p); + if (ret < 0) { + pr_warning("at32-ssc: request irq failed (%d)\n", ret); + pr_debug("at32-ssc: Stopping clock\n"); + clk_disable(ssc_p->ssc->clk); + return ret; + } + + ssc_p->initialized = 1; + } + + /* Set SSC clock mode register */ + ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->cmr_div); + + /* set receive clock mode and format */ + ssc_writel(ssc_p->ssc->regs, RCMR, rcmr); + ssc_writel(ssc_p->ssc->regs, RFMR, rfmr); + + /* set transmit clock mode and format */ + ssc_writel(ssc_p->ssc->regs, TCMR, tcmr); + ssc_writel(ssc_p->ssc->regs, TFMR, tfmr); + + pr_debug("at32-ssc: SSC initialized\n"); + return 0; +} + + + +static int at32_ssc_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + struct at32_pcm_dma_params *dma_params; + + dma_params = ssc_p->dma_params[substream->stream]; + + ssc_writel(dma_params->ssc->regs, CR, dma_params->mask->ssc_enable); + + return 0; +} + + + +#ifdef CONFIG_PM +static int at32_ssc_suspend(struct platform_device *pdev, + struct snd_soc_cpu_dai *cpu_dai) +{ + struct at32_ssc_info *ssc_p; + + if (!cpu_dai->active) + return 0; + + ssc_p = &ssc_info[cpu_dai->id]; + + /* Save the status register before disabling transmit and receive */ + ssc_p->ssc_state.ssc_sr = ssc_readl(ssc_p->ssc->regs, SR); + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_TXDIS) | SSC_BIT(CR_RXDIS)); + + /* Save the current interrupt mask, then disable unmasked interrupts */ + ssc_p->ssc_state.ssc_imr = ssc_readl(ssc_p->ssc->regs, IMR); + ssc_writel(ssc_p->ssc->regs, IDR, ssc_p->ssc_state.ssc_imr); + + ssc_p->ssc_state.ssc_cmr = ssc_readl(ssc_p->ssc->regs, CMR); + ssc_p->ssc_state.ssc_rcmr = ssc_readl(ssc_p->ssc->regs, RCMR); + ssc_p->ssc_state.ssc_rfmr = ssc_readl(ssc_p->ssc->regs, RFMR); + ssc_p->ssc_state.ssc_tcmr = ssc_readl(ssc_p->ssc->regs, TCMR); + ssc_p->ssc_state.ssc_tfmr = ssc_readl(ssc_p->ssc->regs, TFMR); + + return 0; +} + + + +static int at32_ssc_resume(struct platform_device *pdev, + struct snd_soc_cpu_dai *cpu_dai) +{ + struct at32_ssc_info *ssc_p; + u32 cr; + + if (!cpu_dai->active) + return 0; + + ssc_p = &ssc_info[cpu_dai->id]; + + /* restore SSC register settings */ + ssc_writel(ssc_p->ssc->regs, TFMR, ssc_p->ssc_state.ssc_tfmr); + ssc_writel(ssc_p->ssc->regs, TCMR, ssc_p->ssc_state.ssc_tcmr); + ssc_writel(ssc_p->ssc->regs, RFMR, ssc_p->ssc_state.ssc_rfmr); + ssc_writel(ssc_p->ssc->regs, RCMR, ssc_p->ssc_state.ssc_rcmr); + ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->ssc_state.ssc_cmr); + + /* re-enable interrupts */ + ssc_writel(ssc_p->ssc->regs, IER, ssc_p->ssc_state.ssc_imr); + + /* Re-enable recieve and transmit as appropriate */ + cr = 0; + cr |= + (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_RXEN)) ? SSC_BIT(CR_RXEN) : 0; + cr |= + (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_TXEN)) ? SSC_BIT(CR_TXEN) : 0; + ssc_writel(ssc_p->ssc->regs, CR, cr); + + return 0; +} +#else /* CONFIG_PM */ +# define at32_ssc_suspend NULL +# define at32_ssc_resume NULL +#endif /* CONFIG_PM */ + + +#define AT32_SSC_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 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) + + +#define AT32_SSC_FORMATS \ + (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16 | \ + SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_S32) + + +struct snd_soc_cpu_dai at32_ssc_dai[NUM_SSC_DEVICES] = { + { + .name = "at32-ssc0", + .id = 0, + .type = SND_SOC_DAI_PCM, + .suspend = at32_ssc_suspend, + .resume = at32_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .ops = { + .startup = at32_ssc_startup, + .shutdown = at32_ssc_shutdown, + .prepare = at32_ssc_prepare, + .hw_params = at32_ssc_hw_params, + }, + .dai_ops = { + .set_sysclk = at32_ssc_set_dai_sysclk, + .set_fmt = at32_ssc_set_dai_fmt, + .set_clkdiv = at32_ssc_set_dai_clkdiv, + }, + .private_data = &ssc_info[0], + }, + { + .name = "at32-ssc1", + .id = 1, + .type = SND_SOC_DAI_PCM, + .suspend = at32_ssc_suspend, + .resume = at32_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .ops = { + .startup = at32_ssc_startup, + .shutdown = at32_ssc_shutdown, + .prepare = at32_ssc_prepare, + .hw_params = at32_ssc_hw_params, + }, + .dai_ops = { + .set_sysclk = at32_ssc_set_dai_sysclk, + .set_fmt = at32_ssc_set_dai_fmt, + .set_clkdiv = at32_ssc_set_dai_clkdiv, + }, + .private_data = &ssc_info[1], + }, + { + .name = "at32-ssc2", + .id = 2, + .type = SND_SOC_DAI_PCM, + .suspend = at32_ssc_suspend, + .resume = at32_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .ops = { + .startup = at32_ssc_startup, + .shutdown = at32_ssc_shutdown, + .prepare = at32_ssc_prepare, + .hw_params = at32_ssc_hw_params, + }, + .dai_ops = { + .set_sysclk = at32_ssc_set_dai_sysclk, + .set_fmt = at32_ssc_set_dai_fmt, + .set_clkdiv = at32_ssc_set_dai_clkdiv, + }, + .private_data = &ssc_info[2], + }, +}; +EXPORT_SYMBOL_GPL(at32_ssc_dai); + + +MODULE_AUTHOR("Geoffrey Wossum "); +MODULE_DESCRIPTION("AT32 SSC ASoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/at32/at32-ssc.h b/sound/soc/at32/at32-ssc.h new file mode 100644 index 00000000000..3c6901a6c03 --- /dev/null +++ b/sound/soc/at32/at32-ssc.h @@ -0,0 +1,59 @@ +/* sound/soc/at32/at32-ssc.h + * ASoC SSC interface for Atmel AT32 SoC + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __SOUND_SOC_AT32_AT32_SSC_H +#define __SOUND_SOC_AT32_AT32_SSC_H __FILE__ + +#include +#include + +#include "at32-pcm.h" + + + +struct at32_ssc_state { + u32 ssc_cmr; + u32 ssc_rcmr; + u32 ssc_rfmr; + u32 ssc_tcmr; + u32 ssc_tfmr; + u32 ssc_sr; + u32 ssc_imr; +}; + + + +struct at32_ssc_info { + char *name; + struct ssc_device *ssc; + spinlock_t lock; /* lock for dir_mask */ + unsigned short dir_mask; /* 0=unused, 1=playback, 2=capture */ + unsigned short initialized; /* true if SSC has been initialized */ + unsigned short daifmt; + unsigned short cmr_div; + unsigned short tcmr_period; + unsigned short rcmr_period; + struct at32_pcm_dma_params *dma_params[2]; + struct at32_ssc_state ssc_state; +}; + + +/* SSC divider ids */ +#define AT32_SSC_CMR_DIV 0 /* MCK divider for BCLK */ +#define AT32_SSC_TCMR_PERIOD 1 /* BCLK divider for transmit FS */ +#define AT32_SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */ + + +extern struct snd_soc_cpu_dai at32_ssc_dai[]; + + + +#endif /* __SOUND_SOC_AT32_AT32_SSC_H */ diff --git a/sound/soc/at32/playpaq_wm8510.c b/sound/soc/at32/playpaq_wm8510.c new file mode 100644 index 00000000000..d6b9fd5e612 --- /dev/null +++ b/sound/soc/at32/playpaq_wm8510.c @@ -0,0 +1,524 @@ +/* sound/soc/at32/playpaq_wm8510.c + * ASoC machine driver for PlayPaq using WM8510 codec + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This code is largely inspired by sound/soc/at91/eti_b1_wm8731.c + * + * NOTE: If you don't have the AT32 enhanced portmux configured (which + * isn't currently in the mainline or Atmel patched kernel), you will + * need to set the MCLK pin (PA30) to peripheral A in your board initialization + * code. Something like: + * at32_select_periph(GPIO_PIN_PA(30), GPIO_PERIPH_A, 0); + * + */ + +/* #define DEBUG */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +#include +#include + +#include "../codecs/wm8510.h" +#include "at32-pcm.h" +#include "at32-ssc.h" + + +/*-------------------------------------------------------------------------*\ + * constants +\*-------------------------------------------------------------------------*/ +#define MCLK_PIN GPIO_PIN_PA(30) +#define MCLK_PERIPH GPIO_PERIPH_A + + +/*-------------------------------------------------------------------------*\ + * data types +\*-------------------------------------------------------------------------*/ +/* SSC clocking data */ +struct ssc_clock_data { + /* CMR div */ + unsigned int cmr_div; + + /* Frame period (as needed by xCMR.PERIOD) */ + unsigned int period; + + /* The SSC clock rate these settings where calculated for */ + unsigned long ssc_rate; +}; + + +/*-------------------------------------------------------------------------*\ + * module data +\*-------------------------------------------------------------------------*/ +static struct clk *_gclk0; +static struct clk *_pll0; + +#define CODEC_CLK (_gclk0) + + +/*-------------------------------------------------------------------------*\ + * Sound SOC operations +\*-------------------------------------------------------------------------*/ +#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE +static struct ssc_clock_data playpaq_wm8510_calc_ssc_clock( + struct snd_pcm_hw_params *params, + struct snd_soc_cpu_dai *cpu_dai) +{ + struct at32_ssc_info *ssc_p = cpu_dai->private_data; + struct ssc_device *ssc = ssc_p->ssc; + struct ssc_clock_data cd; + unsigned int rate, width_bits, channels; + unsigned int bitrate, ssc_div; + unsigned actual_rate; + + + /* + * Figure out required bitrate + */ + rate = params_rate(params); + channels = params_channels(params); + width_bits = snd_pcm_format_physical_width(params_format(params)); + bitrate = rate * width_bits * channels; + + + /* + * Figure out required SSC divider and period for required bitrate + */ + cd.ssc_rate = clk_get_rate(ssc->clk); + ssc_div = cd.ssc_rate / bitrate; + cd.cmr_div = ssc_div / 2; + if (ssc_div & 1) { + /* round cmr_div up */ + cd.cmr_div++; + } + cd.period = width_bits - 1; + + + /* + * Find actual rate, compare to requested rate + */ + actual_rate = (cd.ssc_rate / (cd.cmr_div * 2)) / (2 * (cd.period + 1)); + pr_debug("playpaq_wm8510: Request rate = %d, actual rate = %d\n", + rate, actual_rate); + + + return cd; +} +#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ + + + +static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct at32_ssc_info *ssc_p = cpu_dai->private_data; + struct ssc_device *ssc = ssc_p->ssc; + unsigned int pll_out = 0, bclk = 0, mclk_div = 0; + int ret; + + + /* Due to difficulties with getting the correct clocks from the AT32's + * PLL0, we're going to let the CODEC be in charge of all the clocks + */ +#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE + const unsigned int fmt = (SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); +#else + struct ssc_clock_data cd; + const unsigned int fmt = (SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); +#endif + + if (ssc == NULL) { + pr_warning("playpaq_wm8510_hw_params: ssc is NULL!\n"); + return -EINVAL; + } + + + /* + * Figure out PLL and BCLK dividers for WM8510 + */ + switch (params_rate(params)) { + case 48000: + pll_out = 12288000; + mclk_div = WM8510_MCLKDIV_1; + bclk = WM8510_BCLKDIV_8; + break; + + case 44100: + pll_out = 11289600; + mclk_div = WM8510_MCLKDIV_1; + bclk = WM8510_BCLKDIV_8; + break; + + case 22050: + pll_out = 11289600; + mclk_div = WM8510_MCLKDIV_2; + bclk = WM8510_BCLKDIV_8; + break; + + case 16000: + pll_out = 12288000; + mclk_div = WM8510_MCLKDIV_3; + bclk = WM8510_BCLKDIV_8; + break; + + case 11025: + pll_out = 11289600; + mclk_div = WM8510_MCLKDIV_4; + bclk = WM8510_BCLKDIV_8; + break; + + case 8000: + pll_out = 12288000; + mclk_div = WM8510_MCLKDIV_6; + bclk = WM8510_BCLKDIV_8; + break; + + default: + pr_warning("playpaq_wm8510: Unsupported sample rate %d\n", + params_rate(params)); + return -EINVAL; + } + + + /* + * set CPU and CODEC DAI configuration + */ + ret = codec_dai->dai_ops.set_fmt(codec_dai, fmt); + if (ret < 0) { + pr_warning("playpaq_wm8510: " + "Failed to set CODEC DAI format (%d)\n", + ret); + return ret; + } + ret = cpu_dai->dai_ops.set_fmt(cpu_dai, fmt); + if (ret < 0) { + pr_warning("playpaq_wm8510: " + "Failed to set CPU DAI format (%d)\n", + ret); + return ret; + } + + + /* + * Set CPU clock configuration + */ +#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE + cd = playpaq_wm8510_calc_ssc_clock(params, cpu_dai); + pr_debug("playpaq_wm8510: cmr_div = %d, period = %d\n", + cd.cmr_div, cd.period); + ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, + AT32_SSC_CMR_DIV, cd.cmr_div); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CPU CMR_DIV (%d)\n", + ret); + return ret; + } + ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, AT32_SSC_TCMR_PERIOD, + cd.period); + if (ret < 0) { + pr_warning("playpaq_wm8510: " + "Failed to set CPU transmit period (%d)\n", + ret); + return ret; + } +#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ + + + /* + * Set CODEC clock configuration + */ + pr_debug("playpaq_wm8510: " + "pll_in = %ld, pll_out = %u, bclk = %x, mclk = %x\n", + clk_get_rate(CODEC_CLK), pll_out, bclk, mclk_div); + + +#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE + ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8510_BCLKDIV, bclk); + if (ret < 0) { + pr_warning + ("playpaq_wm8510: Failed to set CODEC DAI BCLKDIV (%d)\n", + ret); + return ret; + } +#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ + + + ret = codec_dai->dai_ops.set_pll(codec_dai, 0, + clk_get_rate(CODEC_CLK), pll_out); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CODEC DAI PLL (%d)\n", + ret); + return ret; + } + + + ret = codec_dai->dai_ops.set_clkdiv(codec_dai, + WM8510_MCLKDIV, mclk_div); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CODEC MCLKDIV (%d)\n", + ret); + return ret; + } + + + return 0; +} + + + +static struct snd_soc_ops playpaq_wm8510_ops = { + .hw_params = playpaq_wm8510_hw_params, +}; + + + +static const struct snd_soc_dapm_widget playpaq_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + + + +static const char *intercon[][3] = { + /* speaker connected to SPKOUT */ + {"Ext Spk", NULL, "SPKOUTP"}, + {"Ext Spk", NULL, "SPKOUTN"}, + + {"Mic Bias", NULL, "Int Mic"}, + {"MICN", NULL, "Mic Bias"}, + {"MICP", NULL, "Mic Bias"}, + + /* Terminator */ + {NULL, NULL, NULL}, +}; + + + +static int playpaq_wm8510_init(struct snd_soc_codec *codec) +{ + int i; + + /* + * Add DAPM widgets + */ + for (i = 0; i < ARRAY_SIZE(playpaq_dapm_widgets); i++) + snd_soc_dapm_new_control(codec, &playpaq_dapm_widgets[i]); + + + + /* + * Setup audio path interconnects + */ + for (i = 0; intercon[i][0] != NULL; i++) { + snd_soc_dapm_connect_input(codec, + intercon[i][0], + intercon[i][1], intercon[i][2]); + } + + + /* always connected endpoints */ + snd_soc_dapm_set_endpoint(codec, "Int Mic", 1); + snd_soc_dapm_set_endpoint(codec, "Ext Spk", 1); + snd_soc_dapm_sync_endpoints(codec); + + + + /* Make CSB show PLL rate */ + codec->dai->dai_ops.set_clkdiv(codec->dai, WM8510_OPCLKDIV, + WM8510_OPCLKDIV_1 | 4); + + return 0; +} + + + +static struct snd_soc_dai_link playpaq_wm8510_dai = { + .name = "WM8510", + .stream_name = "WM8510 PCM", + .cpu_dai = &at32_ssc_dai[0], + .codec_dai = &wm8510_dai, + .init = playpaq_wm8510_init, + .ops = &playpaq_wm8510_ops, +}; + + + +static struct snd_soc_machine snd_soc_machine_playpaq = { + .name = "LRS_PlayPaq_WM8510", + .dai_link = &playpaq_wm8510_dai, + .num_links = 1, +}; + + + +static struct wm8510_setup_data playpaq_wm8510_setup = { + .i2c_address = 0x1a, +}; + + + +static struct snd_soc_device playpaq_wm8510_snd_devdata = { + .machine = &snd_soc_machine_playpaq, + .platform = &at32_soc_platform, + .codec_dev = &soc_codec_dev_wm8510, + .codec_data = &playpaq_wm8510_setup, +}; + +static struct platform_device *playpaq_snd_device; + + +static int __init playpaq_asoc_init(void) +{ + int ret = 0; + struct at32_ssc_info *ssc_p = playpaq_wm8510_dai.cpu_dai->private_data; + struct ssc_device *ssc = NULL; + + + /* + * Request SSC device + */ + ssc = ssc_request(0); + if (IS_ERR(ssc)) { + ret = PTR_ERR(ssc); + ssc = NULL; + goto err_ssc; + } + ssc_p->ssc = ssc; + + + /* + * Configure MCLK for WM8510 + */ + _gclk0 = clk_get(NULL, "gclk0"); + if (IS_ERR(_gclk0)) { + _gclk0 = NULL; + goto err_gclk0; + } + _pll0 = clk_get(NULL, "pll0"); + if (IS_ERR(_pll0)) { + _pll0 = NULL; + goto err_pll0; + } + if (clk_set_parent(_gclk0, _pll0)) { + pr_warning("snd-soc-playpaq: " + "Failed to set PLL0 as parent for DAC clock\n"); + goto err_set_clk; + } + clk_set_rate(CODEC_CLK, 12000000); + clk_enable(CODEC_CLK); + +#if defined CONFIG_AT32_ENHANCED_PORTMUX + at32_select_periph(MCLK_PIN, MCLK_PERIPH, 0); +#endif + + + /* + * Create and register platform device + */ + playpaq_snd_device = platform_device_alloc("soc-audio", 0); + if (playpaq_snd_device == NULL) { + ret = -ENOMEM; + goto err_device_alloc; + } + + platform_set_drvdata(playpaq_snd_device, &playpaq_wm8510_snd_devdata); + playpaq_wm8510_snd_devdata.dev = &playpaq_snd_device->dev; + + ret = platform_device_add(playpaq_snd_device); + if (ret) { + pr_warning("playpaq_wm8510: platform_device_add failed (%d)\n", + ret); + goto err_device_add; + } + + return 0; + + +err_device_add: + if (playpaq_snd_device != NULL) { + platform_device_put(playpaq_snd_device); + playpaq_snd_device = NULL; + } +err_device_alloc: +err_set_clk: + if (_pll0 != NULL) { + clk_put(_pll0); + _pll0 = NULL; + } +err_pll0: + if (_gclk0 != NULL) { + clk_put(_gclk0); + _gclk0 = NULL; + } +err_gclk0: + if (ssc != NULL) { + ssc_free(ssc); + ssc = NULL; + } +err_ssc: + return ret; +} + + +static void __exit playpaq_asoc_exit(void) +{ + struct at32_ssc_info *ssc_p = playpaq_wm8510_dai.cpu_dai->private_data; + struct ssc_device *ssc; + + if (ssc_p != NULL) { + ssc = ssc_p->ssc; + if (ssc != NULL) + ssc_free(ssc); + ssc_p->ssc = NULL; + } + + if (_gclk0 != NULL) { + clk_put(_gclk0); + _gclk0 = NULL; + } + if (_pll0 != NULL) { + clk_put(_pll0); + _pll0 = NULL; + } + +#if defined CONFIG_AT32_ENHANCED_PORTMUX + at32_free_pin(MCLK_PIN); +#endif + + platform_device_unregister(playpaq_snd_device); + playpaq_snd_device = NULL; +} + +module_init(playpaq_asoc_init); +module_exit(playpaq_asoc_exit); + +MODULE_AUTHOR("Geoffrey Wossum "); +MODULE_DESCRIPTION("ASoC machine driver for LRS PlayPaq"); +MODULE_LICENSE("GPL"); -- cgit From 05a076de83b507238cd186e93daeaf15820d570e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 6 Jun 2008 17:13:53 +0100 Subject: [ALSA] ASoC: Fix default mono mixer configuration for WM8510 Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/codecs/wm8510.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 6f54cb0f50f..152e6f21154 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -71,7 +71,7 @@ static const u16 wm8510_reg[WM8510_CACHEREGNUM] = { 0x0003, 0x0010, 0x0000, 0x0000, 0x0000, 0x0002, 0x0001, 0x0000, 0x0000, 0x0000, 0x0039, 0x0000, - 0x0000, + 0x0001, }; /* -- cgit From d751b233bb8568f1de1ccbe3824ca69090326251 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 11 Jun 2008 13:47:06 +0100 Subject: ALSA: ASoC: Fix register cache sizes for Wolfson codecs The register cache size is used by the codec_reg sysfs file which works in terms of the register cache access functions rather than in terms of raw access to the cache so the size specified needs to be in terms of the number of elements. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/codecs/wm8731.c | 2 +- sound/soc/codecs/wm8750.c | 2 +- sound/soc/codecs/wm8753.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 5acf43ab104..77880537a3c 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -534,7 +534,7 @@ static int wm8731_init(struct snd_soc_device *socdev) codec->set_bias_level = wm8731_set_bias_level; codec->dai = &wm8731_dai; codec->num_dai = 1; - codec->reg_cache_size = sizeof(wm8731_reg); + codec->reg_cache_size = ARRAY_SIZE(wm8731_reg); codec->reg_cache = kmemdup(wm8731_reg, sizeof(wm8731_reg), GFP_KERNEL); if (codec->reg_cache == NULL) return -ENOMEM; diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 1f11ad24551..1ae670a98c5 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -798,7 +798,7 @@ static int wm8750_init(struct snd_soc_device *socdev) codec->set_bias_level = wm8750_set_bias_level; codec->dai = &wm8750_dai; codec->num_dai = 1; - codec->reg_cache_size = sizeof(wm8750_reg); + codec->reg_cache_size = ARRAY_SIZE(wm8750_reg); codec->reg_cache = kmemdup(wm8750_reg, sizeof(wm8750_reg), GFP_KERNEL); if (codec->reg_cache == NULL) return -ENOMEM; diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index c32e6326be6..285c5eaefe0 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1557,7 +1557,7 @@ static int wm8753_init(struct snd_soc_device *socdev) codec->set_bias_level = wm8753_set_bias_level; codec->dai = wm8753_dai; codec->num_dai = 2; - codec->reg_cache_size = sizeof(wm8753_reg); + codec->reg_cache_size = ARRAY_SIZE(wm8753_reg); codec->reg_cache = kmemdup(wm8753_reg, sizeof(wm8753_reg), GFP_KERNEL); if (codec->reg_cache == NULL) -- cgit From 8ddd440768e04e0c158c2bfa6178928e1f1eef3e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 11 Jun 2008 13:47:07 +0100 Subject: ALSA: ASoC: Fix register cache size for UDA1380 The register cache size is used by the codec_reg sysfs file which works in terms of the register cache access functions rather than in terms of raw access to the cache so the size specified needs to be in terms of the number of elements. Signed-off-by: Mark Brown Cc: Philipp Zabel Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/codecs/uda1380.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index cb50486201f..6d5335b14d5 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -655,8 +655,8 @@ static int uda1380_init(struct snd_soc_device *socdev, int dac_clk) GFP_KERNEL); if (codec->reg_cache == NULL) return -ENOMEM; - codec->reg_cache_size = sizeof(uda1380_reg); - codec->reg_cache_step = 2; + codec->reg_cache_size = ARRAY_SIZE(uda1380_reg); + codec->reg_cache_step = 1; uda1380_reset(codec); /* register pcms */ -- cgit From ae2ff19180d7dcac438e399f874d4de230b7f326 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 11 Jun 2008 13:47:08 +0100 Subject: ALSA: ASoC: Tweak tlv320aicx reg_cache_size MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit ASoC codec drivers frequently set the register cache size using sizeof() rather than ARRAY_SIZE(). For tlv320aicx either is correct since the registers are 8 bit but update to use ARRAY_SIZE() for clarity. Signed-off-by: Mark Brown Acked-by: Jarkko Nikula  Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/codecs/tlv320aic3x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index dc8a38d9e53..528c26acdbe 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1021,7 +1021,7 @@ static int aic3x_init(struct snd_soc_device *socdev) codec->set_bias_level = aic3x_set_bias_level; codec->dai = &aic3x_dai; codec->num_dai = 1; - codec->reg_cache_size = sizeof(aic3x_reg); + codec->reg_cache_size = ARRAY_SIZE(aic3x_reg); codec->reg_cache = kmemdup(aic3x_reg, sizeof(aic3x_reg), GFP_KERNEL); if (codec->reg_cache == NULL) return -ENOMEM; -- cgit From dbac7cb8a7fcae34ffda04e4bc30673c2ea841ae Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 11 Jun 2008 13:47:09 +0100 Subject: ALSA: ASoC: Add missing includes Ensure that DAIs are prototyped in the codec drivers that define them. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/codecs/ac97.c | 1 + sound/soc/codecs/wm9712.c | 1 + 2 files changed, 2 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 7bf2081b46f..300d007f233 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -21,6 +21,7 @@ #include #include #include +#include "ac97.h" #define AC97_VERSION "0.6" diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index d9789f1c890..28ac66f6b6b 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -22,6 +22,7 @@ #include #include #include +#include "wm9712.h" #define WM9712_VERSION "0.4" -- cgit From bdb92876f0a9d2b431199e385732ede89ff0b97d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 11 Jun 2008 13:47:10 +0100 Subject: ALSA: ASoC: Pass the DAI being configured into CPU DAI probe and remove This allows per-DAI initialisation to be done by the CPU DAI drivers. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/davinci/davinci-i2s.c | 6 ++++-- sound/soc/pxa/pxa2xx-ac97.c | 3 ++- sound/soc/s3c24xx/s3c2412-i2s.c | 3 ++- sound/soc/s3c24xx/s3c2443-ac97.c | 6 ++++-- sound/soc/s3c24xx/s3c24xx-i2s.c | 3 ++- sound/soc/soc-core.c | 6 +++--- 6 files changed, 17 insertions(+), 10 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index c421774b33e..c3b545ccff7 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -295,7 +295,8 @@ static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd) return ret; } -static int davinci_i2s_probe(struct platform_device *pdev) +static int davinci_i2s_probe(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_machine *machine = socdev->machine; @@ -356,7 +357,8 @@ err_release_region: return ret; } -static void davinci_i2s_remove(struct platform_device *pdev) +static void davinci_i2s_remove(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_machine *machine = socdev->machine; diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 97ec2d90547..cb947956ed1 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -310,7 +310,8 @@ static int pxa2xx_ac97_resume(struct platform_device *pdev, #define pxa2xx_ac97_resume NULL #endif -static int pxa2xx_ac97_probe(struct platform_device *pdev) +static int pxa2xx_ac97_probe(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) { int ret; diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index c4a46dd589b..c463a82dec3 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -601,7 +601,8 @@ struct clk *s3c2412_get_iisclk(void) EXPORT_SYMBOL_GPL(s3c2412_get_iisclk); -static int s3c2412_i2s_probe(struct platform_device *pdev) +static int s3c2412_i2s_probe(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) { DBG("Entered %s\n", __func__); diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c index 0eed140dcd9..533565b61b2 100644 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ b/sound/soc/s3c24xx/s3c2443-ac97.c @@ -209,7 +209,8 @@ static struct s3c24xx_pcm_dma_params s3c2443_ac97_mic_mono_in = { .dma_size = 4, }; -static int s3c2443_ac97_probe(struct platform_device *pdev) +static int s3c2443_ac97_probe(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) { int ret; u32 ac_glbctrl; @@ -260,7 +261,8 @@ static int s3c2443_ac97_probe(struct platform_device *pdev) return ret; } -static void s3c2443_ac97_remove(struct platform_device *pdev) +static void s3c2443_ac97_remove(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) { free_irq(IRQ_S3C244x_AC97, NULL); clk_disable(s3c24xx_ac97.ac97_clk); diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 4c52f7946d9..42e96b5ff82 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -377,7 +377,8 @@ u32 s3c24xx_i2s_get_clockrate(void) } EXPORT_SYMBOL_GPL(s3c24xx_i2s_get_clockrate); -static int s3c24xx_i2s_probe(struct platform_device *pdev) +static int s3c24xx_i2s_probe(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) { DBG("Entered %s\n", __func__); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f594ab888e1..c96a6184d66 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -768,7 +768,7 @@ static int soc_probe(struct platform_device *pdev) for (i = 0; i < machine->num_links; i++) { struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->probe) { - ret = cpu_dai->probe(pdev); + ret = cpu_dai->probe(pdev, cpu_dai); if (ret < 0) goto cpu_dai_err; } @@ -798,7 +798,7 @@ cpu_dai_err: for (i--; i >= 0; i--) { struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->remove) - cpu_dai->remove(pdev); + cpu_dai->remove(pdev, cpu_dai); } if (machine->remove) @@ -827,7 +827,7 @@ static int soc_remove(struct platform_device *pdev) for (i = 0; i < machine->num_links; i++) { struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->remove) - cpu_dai->remove(pdev); + cpu_dai->remove(pdev, cpu_dai); } if (machine->remove) -- cgit From d133b0ce6ceaf744184c7cb9f95ba92f333c246a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 11 Jun 2008 13:47:11 +0100 Subject: ALSA: ASoC: Fix WM9713 voice PCM slave mode configuration Reported-by: Rodolfo Giometti Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/codecs/wm9713.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 4f516a5a561..aba33011878 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -876,7 +876,7 @@ static int wm9713_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, gpio |= 0x0018; break; case SND_SOC_DAIFMT_CBS_CFS: - reg |= 0x0200; + reg |= 0x2000; gpio |= 0x001a; break; case SND_SOC_DAIFMT_CBS_CFM: -- cgit From 2cc8c609798957b90adf90b5bfb9859d1643fade Mon Sep 17 00:00:00 2001 From: Mike Montour Date: Wed, 11 Jun 2008 13:47:12 +0100 Subject: ALSA: ASoC: Add TLV information to remaining WM8753 controls Signed-off-by: Mike Montour Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/codecs/wm8753.c | 75 +++++++++++++++++++++++++++++++---------------- 1 file changed, 50 insertions(+), 25 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 285c5eaefe0..00b481183d4 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -260,28 +260,50 @@ static int wm8753_set_dai(struct snd_kcontrol *kcontrol, return 1; } -static const DECLARE_TLV_DB_LINEAR(rec_mix_tlv, -1500, 600); +static const DECLARE_TLV_DB_SCALE(rec_mix_tlv, -1500, 300, 0); +static const DECLARE_TLV_DB_SCALE(mic_preamp_tlv, 1200, 600, 0); +static const DECLARE_TLV_DB_SCALE(adc_tlv, -9750, 50, 1); +static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1); +static const unsigned int out_tlv[] = { + TLV_DB_RANGE_HEAD(2), + /* 0000000 - 0101111 = "Analogue mute" */ + 0, 48, TLV_DB_SCALE_ITEM(-25500, 0, 0), + 48, 127, TLV_DB_SCALE_ITEM(-7300, 100, 0), +}; +static const DECLARE_TLV_DB_SCALE(mix_tlv, -1500, 300, 0); +static const DECLARE_TLV_DB_SCALE(voice_mix_tlv, -1200, 300, 0); +static const DECLARE_TLV_DB_SCALE(pga_tlv, -1725, 75, 0); static const struct snd_kcontrol_new wm8753_snd_controls[] = { -SOC_DOUBLE_R("PCM Volume", WM8753_LDAC, WM8753_RDAC, 0, 255, 0), - -SOC_DOUBLE_R("ADC Capture Volume", WM8753_LADC, WM8753_RADC, 0, 255, 0), - -SOC_DOUBLE_R("Headphone Playback Volume", WM8753_LOUT1V, WM8753_ROUT1V, 0, 127, 0), -SOC_DOUBLE_R("Speaker Playback Volume", WM8753_LOUT2V, WM8753_ROUT2V, 0, 127, 0), - -SOC_SINGLE("Mono Playback Volume", WM8753_MOUTV, 0, 127, 0), - -SOC_DOUBLE_R("Bypass Playback Volume", WM8753_LOUTM1, WM8753_ROUTM1, 4, 7, 1), -SOC_DOUBLE_R("Sidetone Playback Volume", WM8753_LOUTM2, WM8753_ROUTM2, 4, 7, 1), -SOC_DOUBLE_R("Voice Playback Volume", WM8753_LOUTM2, WM8753_ROUTM2, 0, 7, 1), - -SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8753_LOUT1V, WM8753_ROUT1V, 7, 1, 0), -SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8753_LOUT2V, WM8753_ROUT2V, 7, 1, 0), - -SOC_SINGLE("Mono Bypass Playback Volume", WM8753_MOUTM1, 4, 7, 1), -SOC_SINGLE("Mono Sidetone Playback Volume", WM8753_MOUTM2, 4, 7, 1), -SOC_SINGLE("Mono Voice Playback Volume", WM8753_MOUTM2, 0, 7, 1), +SOC_DOUBLE_R_TLV("PCM Volume", WM8753_LDAC, WM8753_RDAC, 0, 255, 0, dac_tlv), + +SOC_DOUBLE_R_TLV("ADC Capture Volume", WM8753_LADC, WM8753_RADC, 0, 255, 0, + adc_tlv), + +SOC_DOUBLE_R_TLV("Headphone Playback Volume", WM8753_LOUT1V, WM8753_ROUT1V, + 0, 127, 0, out_tlv), +SOC_DOUBLE_R_TLV("Speaker Playback Volume", WM8753_LOUT2V, WM8753_ROUT2V, 0, + 127, 0, out_tlv), + +SOC_SINGLE_TLV("Mono Playback Volume", WM8753_MOUTV, 0, 127, 0, out_tlv), + +SOC_DOUBLE_R_TLV("Bypass Playback Volume", WM8753_LOUTM1, WM8753_ROUTM1, 4, 7, + 1, mix_tlv), +SOC_DOUBLE_R_TLV("Sidetone Playback Volume", WM8753_LOUTM2, WM8753_ROUTM2, 4, + 7, 1, mix_tlv), +SOC_DOUBLE_R_TLV("Voice Playback Volume", WM8753_LOUTM2, WM8753_ROUTM2, 0, 7, + 1, voice_mix_tlv), + +SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8753_LOUT1V, WM8753_ROUT1V, 7, + 1, 0), +SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8753_LOUT2V, WM8753_ROUT2V, 7, + 1, 0), + +SOC_SINGLE_TLV("Mono Bypass Playback Volume", WM8753_MOUTM1, 4, 7, 1, mix_tlv), +SOC_SINGLE_TLV("Mono Sidetone Playback Volume", WM8753_MOUTM2, 4, 7, 1, + mix_tlv), +SOC_SINGLE_TLV("Mono Voice Playback Volume", WM8753_MOUTM2, 0, 7, 1, + voice_mix_tlv), SOC_SINGLE("Mono Playback ZC Switch", WM8753_MOUTV, 7, 1, 0), SOC_ENUM("Bass Boost", wm8753_enum[0]), @@ -291,10 +313,13 @@ SOC_SINGLE("Bass Volume", WM8753_BASS, 0, 15, 1), SOC_SINGLE("Treble Volume", WM8753_TREBLE, 0, 15, 1), SOC_ENUM("Treble Cut-off", wm8753_enum[2]), -SOC_DOUBLE_TLV("Sidetone Capture Volume", WM8753_RECMIX1, 0, 4, 7, 1, rec_mix_tlv), -SOC_SINGLE_TLV("Voice Sidetone Capture Volume", WM8753_RECMIX2, 0, 7, 1, rec_mix_tlv), +SOC_DOUBLE_TLV("Sidetone Capture Volume", WM8753_RECMIX1, 0, 4, 7, 1, + rec_mix_tlv), +SOC_SINGLE_TLV("Voice Sidetone Capture Volume", WM8753_RECMIX2, 0, 7, 1, + rec_mix_tlv), -SOC_DOUBLE_R("Capture Volume", WM8753_LINVOL, WM8753_RINVOL, 0, 63, 0), +SOC_DOUBLE_R_TLV("Capture Volume", WM8753_LINVOL, WM8753_RINVOL, 0, 63, 0, + pga_tlv), SOC_DOUBLE_R("Capture ZC Switch", WM8753_LINVOL, WM8753_RINVOL, 6, 1, 0), SOC_DOUBLE_R("Capture Switch", WM8753_LINVOL, WM8753_RINVOL, 7, 1, 1), @@ -326,8 +351,8 @@ SOC_ENUM("De-emphasis", wm8753_enum[8]), SOC_ENUM("Playback Mono Mix", wm8753_enum[9]), SOC_ENUM("Playback Phase", wm8753_enum[10]), -SOC_SINGLE("Mic2 Capture Volume", WM8753_INCTL1, 7, 3, 0), -SOC_SINGLE("Mic1 Capture Volume", WM8753_INCTL1, 5, 3, 0), +SOC_SINGLE_TLV("Mic2 Capture Volume", WM8753_INCTL1, 7, 3, 0, mic_preamp_tlv), +SOC_SINGLE_TLV("Mic1 Capture Volume", WM8753_INCTL1, 5, 3, 0, mic_preamp_tlv), SOC_ENUM_EXT("DAI Mode", wm8753_enum[26], wm8753_get_dai, wm8753_set_dai), -- cgit From a2e31a595ea74120a5e9de91fce56dd835edff94 Mon Sep 17 00:00:00 2001 From: Mike Montour Date: Wed, 11 Jun 2008 13:47:14 +0100 Subject: ALSA: ASoC: Add TLV information to the LM4857 controls on the GTA01 Signed-off-by: Mike Montour Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/s3c24xx/neo1973_wm8753.c | 16 ++++++++++------ 1 file changed, 10 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index c1a0161bc72..34851238dea 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -22,6 +22,7 @@ #include #include #include +#include #include #include @@ -474,13 +475,16 @@ static const struct soc_enum neo_scenario_enum[] = { SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(neo_scenarios), neo_scenarios), }; +static const DECLARE_TLV_DB_SCALE(stereo_tlv, -4050, 150, 0); +static const DECLARE_TLV_DB_SCALE(mono_tlv, -3450, 150, 0); + static const struct snd_kcontrol_new wm8753_neo1973_controls[] = { - SOC_SINGLE_EXT("Amp Left Playback Volume", LM4857_LVOL, 0, 31, 0, - lm4857_get_reg, lm4857_set_reg), - SOC_SINGLE_EXT("Amp Right Playback Volume", LM4857_RVOL, 0, 31, 0, - lm4857_get_reg, lm4857_set_reg), - SOC_SINGLE_EXT("Amp Mono Playback Volume", LM4857_MVOL, 0, 31, 0, - lm4857_get_reg, lm4857_set_reg), + SOC_SINGLE_EXT_TLV("Amp Left Playback Volume", LM4857_LVOL, 0, 31, 0, + lm4857_get_reg, lm4857_set_reg, stereo_tlv), + SOC_SINGLE_EXT_TLV("Amp Right Playback Volume", LM4857_RVOL, 0, 31, 0, + lm4857_get_reg, lm4857_set_reg, stereo_tlv), + SOC_SINGLE_EXT_TLV("Amp Mono Playback Volume", LM4857_MVOL, 0, 31, 0, + lm4857_get_reg, lm4857_set_reg, mono_tlv), SOC_ENUM_EXT("Amp Mode", lm4857_mode_enum[0], lm4857_get_mode, lm4857_set_mode), SOC_ENUM_EXT("Neo Mode", neo_scenario_enum[0], -- cgit From 6e132fa668d48a5a6fae6c24f6104e44b1ec1049 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Wed, 11 Jun 2008 13:47:15 +0100 Subject: ALSA: ASoC: Cover also Nokia N810 WiMAX Edition in N810 machine driver Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/omap/n810.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index c32487b6c7b..c168a64e316 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -265,7 +265,7 @@ static int __init n810_soc_init(void) int err; struct device *dev; - if (!machine_is_nokia_n810()) + if (!(machine_is_nokia_n810() || machine_is_nokia_n810_wimax())) return -ENODEV; n810_snd_device = platform_device_alloc("soc-audio", -1); -- cgit From abb68c26ba15f8e84e580a40c0b1bc349cb534b0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 13 Jun 2008 16:24:04 +0100 Subject: ALSA: ASoC: Check for exact register match in wm97xx_reset() To provide added robustness in case an AC97 controller reads back all zeros in error cases check for an exact match when testing to see if resets have brought the codec back. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/codecs/wm9712.c | 4 ++-- sound/soc/codecs/wm9713.c | 4 ++-- 2 files changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 28ac66f6b6b..47390113bd0 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -589,12 +589,12 @@ static int wm9712_reset(struct snd_soc_codec *codec, int try_warm) { if (try_warm && soc_ac97_ops.warm_reset) { soc_ac97_ops.warm_reset(codec->ac97); - if (!(ac97_read(codec, 0) & 0x8000)) + if (ac97_read(codec, 0) == wm9712_reg[0]) return 1; } soc_ac97_ops.reset(codec->ac97); - if (ac97_read(codec, 0) & 0x8000) + if (ac97_read(codec, 0) != wm9712_reg[0]) goto err; return 0; diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index aba33011878..0db96374736 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1076,12 +1076,12 @@ int wm9713_reset(struct snd_soc_codec *codec, int try_warm) { if (try_warm && soc_ac97_ops.warm_reset) { soc_ac97_ops.warm_reset(codec->ac97); - if (!(ac97_read(codec, 0) & 0x8000)) + if (ac97_read(codec, 0) == wm9713_reg[0]) return 1; } soc_ac97_ops.reset(codec->ac97); - if (ac97_read(codec, 0) & 0x8000) + if (ac97_read(codec, 0) != wm9713_reg[0]) return -EIO; return 0; } -- cgit From 6ed2597883b1b03ca94f62f0cfe908314cba6d6b Mon Sep 17 00:00:00 2001 From: Andy Green Date: Fri, 13 Jun 2008 16:24:05 +0100 Subject: ALSA: ASoC: Don't block system resume On OpenMoko soc-audio resume is taking 700ms of the whole resume time of 1.3s, dominated by writes to the codec over I2C. This patch shunts the resume guts into a workqueue which then is done asynchronously. The "card" is locked using the ALSA power state APIs as suggested by Mark Brown. [Added fix for race with resume to suspend and fixed a couple of nits from checkpatch -- broonie.] Signed-off-by: Andy Green Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/soc-core.c | 46 +++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 43 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c96a6184d66..b931039632c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -639,6 +639,16 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) struct snd_soc_codec *codec = socdev->codec; int i; + /* Due to the resume being scheduled into a workqueue we could + * suspend before that's finished - wait for it to complete. + */ + snd_power_lock(codec->card); + snd_power_wait(codec->card, SNDRV_CTL_POWER_D0); + snd_power_unlock(codec->card); + + /* we're going to block userspace touching us until resume completes */ + snd_power_change_state(codec->card, SNDRV_CTL_POWER_D3hot); + /* mute any active DAC's */ for (i = 0; i < machine->num_links; i++) { struct snd_soc_codec_dai *dai = machine->dai_link[i].codec_dai; @@ -691,16 +701,27 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) return 0; } -/* powers up audio subsystem after a suspend */ -static int soc_resume(struct platform_device *pdev) +/* deferred resume work, so resume can complete before we finished + * setting our codec back up, which can be very slow on I2C + */ +static void soc_resume_deferred(struct work_struct *work) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_device *socdev = container_of(work, + struct snd_soc_device, + deferred_resume_work); struct snd_soc_machine *machine = socdev->machine; struct snd_soc_platform *platform = socdev->platform; struct snd_soc_codec_device *codec_dev = socdev->codec_dev; struct snd_soc_codec *codec = socdev->codec; + struct platform_device *pdev = to_platform_device(socdev->dev); int i; + /* our power state is still SNDRV_CTL_POWER_D3hot from suspend time, + * so userspace apps are blocked from touching us + */ + + dev_info(socdev->dev, "starting resume work\n"); + if (machine->resume_pre) machine->resume_pre(pdev); @@ -742,6 +763,22 @@ static int soc_resume(struct platform_device *pdev) if (machine->resume_post) machine->resume_post(pdev); + dev_info(socdev->dev, "resume work completed\n"); + + /* userspace can access us now we are back as we were before */ + snd_power_change_state(codec->card, SNDRV_CTL_POWER_D0); +} + +/* powers up audio subsystem after a suspend */ +static int soc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + dev_info(socdev->dev, "scheduling resume work\n"); + + if (!schedule_work(&socdev->deferred_resume_work)) + dev_err(socdev->dev, "work item may be lost\n"); + return 0; } @@ -788,6 +825,9 @@ static int soc_probe(struct platform_device *pdev) /* DAPM stream work */ INIT_DELAYED_WORK(&socdev->delayed_work, close_delayed_work); + /* deferred resume work */ + INIT_WORK(&socdev->deferred_resume_work, soc_resume_deferred); + return 0; platform_err: -- cgit From 83544994a6a592073e9f6b937c6da12cf5ed2d1e Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Fri, 13 Jun 2008 14:02:31 -0500 Subject: ALSA: Fix register programming in Freescale MPC8610 HPCD sound driver Fix the Freescale MPC8610 HPCD sound driver so that it programs the DMACR and PMUXCR registers in the global utilities correctly. Signed-off-by: Timur Tabi Acked-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/fsl/mpc8610_hpcd.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index a00aac7a71f..8820c3f8d17 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -58,9 +58,9 @@ static int mpc8610_hpcd_machine_probe(struct platform_device *sound_device) sound_device->dev.platform_data; /* Program the signal routing between the SSI and the DMA */ - guts_set_dmacr(machine_data->guts, machine_data->dma_id + 1, + guts_set_dmacr(machine_data->guts, machine_data->dma_id, machine_data->dma_channel_id[0], CCSR_GUTS_DMACR_DEV_SSI); - guts_set_dmacr(machine_data->guts, machine_data->dma_id + 1, + guts_set_dmacr(machine_data->guts, machine_data->dma_id, machine_data->dma_channel_id[1], CCSR_GUTS_DMACR_DEV_SSI); guts_set_pmuxcr_dma(machine_data->guts, machine_data->dma_id, @@ -170,9 +170,9 @@ int mpc8610_hpcd_machine_remove(struct platform_device *sound_device) /* Restore the signal routing */ - guts_set_dmacr(machine_data->guts, machine_data->dma_id + 1, + guts_set_dmacr(machine_data->guts, machine_data->dma_id, machine_data->dma_channel_id[0], 0); - guts_set_dmacr(machine_data->guts, machine_data->dma_id + 1, + guts_set_dmacr(machine_data->guts, machine_data->dma_id, machine_data->dma_channel_id[1], 0); switch (machine_data->ssi_id) { @@ -182,7 +182,7 @@ int mpc8610_hpcd_machine_remove(struct platform_device *sound_device) break; case 1: clrsetbits_be32(&machine_data->guts->pmuxcr, - CCSR_GUTS_PMUXCR_SSI2_MASK, CCSR_GUTS_PMUXCR_SSI1_LA); + CCSR_GUTS_PMUXCR_SSI2_MASK, CCSR_GUTS_PMUXCR_SSI2_LA); break; } -- cgit From 1301a964cef2759d46b4ce366eeeee1f869731f9 Mon Sep 17 00:00:00 2001 From: Randy Dunlap Date: Tue, 17 Jun 2008 19:19:34 +0100 Subject: ALSA: ASoC: fix PM=n build Fix sound/soc build failure when CONFIG_PM=n: linux-next-20080617/sound/soc/soc-core.c:829: error: 'soc_resume_deferred' undeclared (first use in this function) soc3.out:make[3]: *** [sound/soc/soc-core.o] Error 1 Signed-off-by: Randy Dunlap Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/soc-core.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b931039632c..bdbbc6a980f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -825,8 +825,10 @@ static int soc_probe(struct platform_device *pdev) /* DAPM stream work */ INIT_DELAYED_WORK(&socdev->delayed_work, close_delayed_work); +#ifdef CONFIG_PM /* deferred resume work */ INIT_WORK(&socdev->deferred_resume_work, soc_resume_deferred); +#endif return 0; -- cgit From 796d2ca84859d1fdb11ff06cd9707ffab5642fca Mon Sep 17 00:00:00 2001 From: Richard Purdie Date: Mon, 23 Jun 2008 14:51:28 +0100 Subject: ALSA: ASoC: Add AK4535 driver The AK4535 codec is included in some HP iPAQ systems. This driver was originally written by Richard Purdie and with some bug fixes from Milan Plzik. While out of tree it has also had some mechanical updates for new APIs and current best practices from Liam Girdwood, Graeme Gregory and Mark Brown. Signed-off-by: Richard Purdie Signed-off-by: Milan Plzik Signed-off-by: Liam Girdwood Signed-off-by: Graeme Gregory Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4535.c | 696 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ak4535.h | 46 +++ 4 files changed, 747 insertions(+) create mode 100644 sound/soc/codecs/ak4535.c create mode 100644 sound/soc/codecs/ak4535.h (limited to 'sound/soc') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f439138db84..1db04a28a53 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -2,6 +2,9 @@ config SND_SOC_AC97_CODEC tristate select SND_AC97_CODEC +config SND_SOC_AK4535 + tristate + config SND_SOC_UDA1380 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index d9826f35d8c..d7b97abcf72 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -1,4 +1,5 @@ snd-soc-ac97-objs := ac97.o +snd-soc-ak4535-objs := ak4535.o snd-soc-uda1380-objs := uda1380.o snd-soc-wm8510-objs := wm8510.o snd-soc-wm8731-objs := wm8731.o @@ -11,6 +12,7 @@ snd-soc-cs4270-objs := cs4270.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o +obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c new file mode 100644 index 00000000000..469266e881d --- /dev/null +++ b/sound/soc/codecs/ak4535.c @@ -0,0 +1,696 @@ +/* + * ak4535.c -- AK4535 ALSA Soc Audio driver + * + * Copyright 2005 Openedhand Ltd. + * + * Author: Richard Purdie + * + * Based on wm8753.c by Liam Girdwood + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "ak4535.h" + +#define AUDIO_NAME "ak4535" +#define AK4535_VERSION "0.3" + +struct snd_soc_codec_device soc_codec_dev_ak4535; + +/* codec private data */ +struct ak4535_priv { + unsigned int sysclk; +}; + +/* + * ak4535 register cache + */ +static const u16 ak4535_reg[AK4535_CACHEREGNUM] = { + 0x0000, 0x0080, 0x0000, 0x0003, + 0x0002, 0x0000, 0x0011, 0x0001, + 0x0000, 0x0040, 0x0036, 0x0010, + 0x0000, 0x0000, 0x0057, 0x0000, +}; + +/* + * read ak4535 register cache + */ +static inline unsigned int ak4535_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg >= AK4535_CACHEREGNUM) + return -1; + return cache[reg]; +} + +static inline unsigned int ak4535_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 data; + data = reg; + + if (codec->hw_write(codec->control_data, &data, 1) != 1) + return -EIO; + + if (codec->hw_read(codec->control_data, &data, 1) != 1) + return -EIO; + + return data; +}; + +/* + * write ak4535 register cache + */ +static inline void ak4535_write_reg_cache(struct snd_soc_codec *codec, + u16 reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + if (reg >= AK4535_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * write to the AK4535 register space + */ +static int ak4535_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + /* data is + * D15..D8 AK4535 register offset + * D7...D0 register data + */ + data[0] = reg & 0xff; + data[1] = value & 0xff; + + ak4535_write_reg_cache(codec, reg, value); + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + else + return -EIO; +} + +static int ak4535_sync(struct snd_soc_codec *codec) +{ + u16 *cache = codec->reg_cache; + int i, r = 0; + + for (i = 0; i < AK4535_CACHEREGNUM; i++) + r |= ak4535_write(codec, i, cache[i]); + + return r; +}; + +static const char *ak4535_mono_gain[] = {"+6dB", "-17dB"}; +static const char *ak4535_mono_out[] = {"(L + R)/2", "Hi-Z"}; +static const char *ak4535_hp_out[] = {"Stereo", "Mono"}; +static const char *ak4535_deemp[] = {"44.1kHz", "Off", "48kHz", "32kHz"}; +static const char *ak4535_mic_select[] = {"Internal", "External"}; + +static const struct soc_enum ak4535_enum[] = { + SOC_ENUM_SINGLE(AK4535_SIG1, 7, 2, ak4535_mono_gain), + SOC_ENUM_SINGLE(AK4535_SIG1, 6, 2, ak4535_mono_out), + SOC_ENUM_SINGLE(AK4535_MODE2, 2, 2, ak4535_hp_out), + SOC_ENUM_SINGLE(AK4535_DAC, 0, 4, ak4535_deemp), + SOC_ENUM_SINGLE(AK4535_MIC, 1, 2, ak4535_mic_select), +}; + +static const struct snd_kcontrol_new ak4535_snd_controls[] = { + SOC_SINGLE("ALC2 Switch", AK4535_SIG1, 1, 1, 0), + SOC_ENUM("Mono 1 Output", ak4535_enum[1]), + SOC_ENUM("Mono 1 Gain", ak4535_enum[0]), + SOC_ENUM("Headphone Output", ak4535_enum[2]), + SOC_ENUM("Playback Deemphasis", ak4535_enum[3]), + SOC_SINGLE("Bass Volume", AK4535_DAC, 2, 3, 0), + SOC_SINGLE("Mic Boost (+20dB) Switch", AK4535_MIC, 0, 1, 0), + SOC_ENUM("Mic Select", ak4535_enum[4]), + SOC_SINGLE("ALC Operation Time", AK4535_TIMER, 0, 3, 0), + SOC_SINGLE("ALC Recovery Time", AK4535_TIMER, 2, 3, 0), + SOC_SINGLE("ALC ZC Time", AK4535_TIMER, 4, 3, 0), + SOC_SINGLE("ALC 1 Switch", AK4535_ALC1, 5, 1, 0), + SOC_SINGLE("ALC 2 Switch", AK4535_ALC1, 6, 1, 0), + SOC_SINGLE("ALC Volume", AK4535_ALC2, 0, 127, 0), + SOC_SINGLE("Capture Volume", AK4535_PGA, 0, 127, 0), + SOC_SINGLE("Left Playback Volume", AK4535_LATT, 0, 127, 1), + SOC_SINGLE("Right Playback Volume", AK4535_RATT, 0, 127, 1), + SOC_SINGLE("AUX Bypass Volume", AK4535_VOL, 0, 15, 0), + 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), + SOC_DAPM_SINGLE("Mono Playback Switch", AK4535_SIG1, 5, 1, 0), +}; + +/* Stereo Mixer */ +static const struct snd_kcontrol_new ak4535_stereo_mixer_controls[] = { + SOC_DAPM_SINGLE("Mic Sidetone Switch", AK4535_SIG2, 4, 1, 0), + SOC_DAPM_SINGLE("Playback Switch", AK4535_SIG2, 7, 1, 0), + SOC_DAPM_SINGLE("Aux Bypass Switch", AK4535_SIG2, 5, 1, 0), +}; + +/* Input Mixer */ +static const struct snd_kcontrol_new ak4535_input_mixer_controls[] = { + SOC_DAPM_SINGLE("Mic Capture Switch", AK4535_MIC, 2, 1, 0), + SOC_DAPM_SINGLE("Aux Capture Switch", AK4535_MIC, 5, 1, 0), +}; + +/* Input mux */ +static const struct snd_kcontrol_new ak4535_input_mux_control = + SOC_DAPM_ENUM("Input Select", ak4535_enum[4]); + +/* HP L switch */ +static const struct snd_kcontrol_new ak4535_hpl_control = + SOC_DAPM_SINGLE("Switch", AK4535_SIG2, 1, 1, 1); + +/* HP R switch */ +static const struct snd_kcontrol_new ak4535_hpr_control = + SOC_DAPM_SINGLE("Switch", AK4535_SIG2, 0, 1, 1); + +/* mono 2 switch */ +static const struct snd_kcontrol_new ak4535_mono2_control = + SOC_DAPM_SINGLE("Switch", AK4535_SIG1, 0, 1, 0); + +/* Line out switch */ +static const struct snd_kcontrol_new ak4535_line_control = + SOC_DAPM_SINGLE("Switch", AK4535_SIG2, 6, 1, 0); + +/* ak4535 dapm widgets */ +static const struct snd_soc_dapm_widget ak4535_dapm_widgets[] = { + SND_SOC_DAPM_MIXER("Stereo Mixer", SND_SOC_NOPM, 0, 0, + &ak4535_stereo_mixer_controls[0], + ARRAY_SIZE(ak4535_stereo_mixer_controls)), + SND_SOC_DAPM_MIXER("Mono1 Mixer", SND_SOC_NOPM, 0, 0, + &ak4535_mono1_mixer_controls[0], + ARRAY_SIZE(ak4535_mono1_mixer_controls)), + SND_SOC_DAPM_MIXER("Input Mixer", SND_SOC_NOPM, 0, 0, + &ak4535_input_mixer_controls[0], + ARRAY_SIZE(ak4535_input_mixer_controls)), + SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, + &ak4535_input_mux_control), + SND_SOC_DAPM_DAC("DAC", "Playback", AK4535_PM2, 0, 0), + SND_SOC_DAPM_SWITCH("Mono 2 Enable", SND_SOC_NOPM, 0, 0, + &ak4535_mono2_control), + /* speaker powersave bit */ + SND_SOC_DAPM_PGA("Speaker Enable", AK4535_MODE2, 0, 0, NULL, 0), + SND_SOC_DAPM_SWITCH("Line Out Enable", SND_SOC_NOPM, 0, 0, + &ak4535_line_control), + SND_SOC_DAPM_SWITCH("Left HP Enable", SND_SOC_NOPM, 0, 0, + &ak4535_hpl_control), + SND_SOC_DAPM_SWITCH("Right HP Enable", SND_SOC_NOPM, 0, 0, + &ak4535_hpr_control), + SND_SOC_DAPM_OUTPUT("LOUT"), + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("ROUT"), + SND_SOC_DAPM_OUTPUT("HPR"), + SND_SOC_DAPM_OUTPUT("SPP"), + SND_SOC_DAPM_OUTPUT("SPN"), + SND_SOC_DAPM_OUTPUT("MOUT1"), + SND_SOC_DAPM_OUTPUT("MOUT2"), + SND_SOC_DAPM_OUTPUT("MICOUT"), + SND_SOC_DAPM_ADC("ADC", "Capture", AK4535_PM1, 0, 0), + SND_SOC_DAPM_PGA("Spk Amp", AK4535_PM2, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("HP R Amp", AK4535_PM2, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("HP L Amp", AK4535_PM2, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mic", AK4535_PM1, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("Line Out", AK4535_PM1, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mono Out", AK4535_PM1, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("AUX In", AK4535_PM1, 2, 0, NULL, 0), + + SND_SOC_DAPM_MICBIAS("Mic Int Bias", AK4535_MIC, 3, 0), + SND_SOC_DAPM_MICBIAS("Mic Ext Bias", AK4535_MIC, 4, 0), + SND_SOC_DAPM_INPUT("MICIN"), + SND_SOC_DAPM_INPUT("MICEXT"), + SND_SOC_DAPM_INPUT("AUX"), + SND_SOC_DAPM_INPUT("MIN"), + SND_SOC_DAPM_INPUT("AIN"), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /*stereo mixer */ + {"Stereo Mixer", "Playback Switch", "DAC"}, + {"Stereo Mixer", "Mic Sidetone Switch", "Mic"}, + {"Stereo Mixer", "Aux Bypass Switch", "AUX In"}, + + /* mono1 mixer */ + {"Mono1 Mixer", "Mic Sidetone Switch", "Mic"}, + {"Mono1 Mixer", "Mono Playback Switch", "DAC"}, + + /* Mic */ + {"Mic", NULL, "AIN"}, + {"Input Mux", "Internal", "Mic Int Bias"}, + {"Input Mux", "External", "Mic Ext Bias"}, + {"Mic Int Bias", NULL, "MICIN"}, + {"Mic Ext Bias", NULL, "MICEXT"}, + {"MICOUT", NULL, "Input Mux"}, + + /* line out */ + {"LOUT", NULL, "Line Out Enable"}, + {"ROUT", NULL, "Line Out Enable"}, + {"Line Out Enable", "Switch", "Line Out"}, + {"Line Out", NULL, "Stereo Mixer"}, + + /* mono1 out */ + {"MOUT1", NULL, "Mono Out"}, + {"Mono Out", NULL, "Mono1 Mixer"}, + + /* left HP */ + {"HPL", NULL, "Left HP Enable"}, + {"Left HP Enable", "Switch", "HP L Amp"}, + {"HP L Amp", NULL, "Stereo Mixer"}, + + /* right HP */ + {"HPR", NULL, "Right HP Enable"}, + {"Right HP Enable", "Switch", "HP R Amp"}, + {"HP R Amp", NULL, "Stereo Mixer"}, + + /* speaker */ + {"SPP", NULL, "Speaker Enable"}, + {"SPN", NULL, "Speaker Enable"}, + {"Speaker Enable", "Switch", "Spk Amp"}, + {"Spk Amp", NULL, "MIN"}, + + /* mono 2 */ + {"MOUT2", NULL, "Mono 2 Enable"}, + {"Mono 2 Enable", "Switch", "Stereo Mixer"}, + + /* Aux In */ + {"Aux In", NULL, "AUX"}, + + /* ADC */ + {"ADC", NULL, "Input Mixer"}, + {"Input Mixer", "Mic Capture Switch", "Mic"}, + {"Input Mixer", "Aux Capture Switch", "Aux In"}, +}; + +static int ak4535_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, ak4535_dapm_widgets, + ARRAY_SIZE(ak4535_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static int ak4535_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct ak4535_priv *ak4535 = codec->private_data; + + ak4535->sysclk = freq; + return 0; +} + +static int ak4535_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + struct ak4535_priv *ak4535 = codec->private_data; + u8 mode2 = ak4535_read_reg_cache(codec, AK4535_MODE2) & ~(0x3 << 5); + int rate = params_rate(params), fs = 256; + + if (rate) + fs = ak4535->sysclk / rate; + + /* set fs */ + switch (fs) { + case 1024: + mode2 |= (0x2 << 5); + break; + case 512: + mode2 |= (0x1 << 5); + break; + case 256: + break; + } + + /* set rate */ + ak4535_write(codec, AK4535_MODE2, mode2); + return 0; +} + +static int ak4535_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 mode1 = 0; + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + mode1 = 0x0002; + break; + case SND_SOC_DAIFMT_LEFT_J: + mode1 = 0x0001; + break; + default: + return -EINVAL; + } + + /* use 32 fs for BCLK to save power */ + mode1 |= 0x4; + + ak4535_write(codec, AK4535_MODE1, mode1); + return 0; +} + +static int ak4535_mute(struct snd_soc_codec_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC) & 0xffdf; + if (!mute) + ak4535_write(codec, AK4535_DAC, mute_reg); + else + ak4535_write(codec, AK4535_DAC, mute_reg | 0x20); + return 0; +} + +static int ak4535_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 i; + + switch (level) { + case SND_SOC_BIAS_ON: + ak4535_mute(codec->dai, 0); + break; + case SND_SOC_BIAS_PREPARE: + ak4535_mute(codec->dai, 1); + break; + case SND_SOC_BIAS_STANDBY: + i = ak4535_read_reg_cache(codec, AK4535_PM1); + ak4535_write(codec, AK4535_PM1, i | 0x80); + i = ak4535_read_reg_cache(codec, AK4535_PM2); + ak4535_write(codec, AK4535_PM2, i & (~0x80)); + break; + case SND_SOC_BIAS_OFF: + i = ak4535_read_reg_cache(codec, AK4535_PM1); + ak4535_write(codec, AK4535_PM1, i & (~0x80)); + break; + } + codec->bias_level = level; + return 0; +} + +#define AK4535_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) + +struct snd_soc_codec_dai ak4535_dai = { + .name = "AK4535", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = AK4535_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = AK4535_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = { + .hw_params = ak4535_hw_params, + }, + .dai_ops = { + .set_fmt = ak4535_set_dai_fmt, + .digital_mute = ak4535_mute, + .set_sysclk = ak4535_set_dai_sysclk, + }, +}; +EXPORT_SYMBOL_GPL(ak4535_dai); + +static int ak4535_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + ak4535_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int ak4535_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + ak4535_sync(codec); + ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + ak4535_set_bias_level(codec, codec->suspend_bias_level); + return 0; +} + +/* + * initialise the AK4535 driver + * register the mixer and dsp interfaces with the kernel + */ +static int ak4535_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->codec; + int ret = 0; + + codec->name = "AK4535"; + codec->owner = THIS_MODULE; + codec->read = ak4535_read_reg_cache; + codec->write = ak4535_write; + codec->set_bias_level = ak4535_set_bias_level; + codec->dai = &ak4535_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(ak4535_reg); + codec->reg_cache = kmemdup(ak4535_reg, sizeof(ak4535_reg), GFP_KERNEL); + + if (codec->reg_cache == NULL) + return -ENOMEM; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "ak4535: failed to create pcms\n"); + goto pcm_err; + } + + /* power on device */ + ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + ak4535_add_controls(codec); + ak4535_add_widgets(codec); + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "ak4535: failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + + return ret; +} + +static struct snd_soc_device *ak4535_socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + +#define I2C_DRIVERID_AK4535 0xfefe /* liam - need a proper id */ + +static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; + +/* Magic definition of all other variables and things */ +I2C_CLIENT_INSMOD; + +static struct i2c_driver ak4535_i2c_driver; +static struct i2c_client client_template; + +/* If the i2c layer weren't so broken, we could pass this kind of data + around */ +static int ak4535_codec_probe(struct i2c_adapter *adap, int addr, int kind) +{ + struct snd_soc_device *socdev = ak4535_socdev; + struct ak4535_setup_data *setup = socdev->codec_data; + struct snd_soc_codec *codec = socdev->codec; + struct i2c_client *i2c; + int ret; + + if (addr != setup->i2c_address) + return -ENODEV; + + client_template.adapter = adap; + client_template.addr = addr; + + i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); + if (i2c == NULL) { + kfree(codec); + return -ENOMEM; + } + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + ret = i2c_attach_client(i2c); + if (ret < 0) { + printk(KERN_ERR "failed to attach codec at addr %x\n", addr); + goto err; + } + + ret = ak4535_init(socdev); + if (ret < 0) { + printk(KERN_ERR "failed to initialise AK4535\n"); + goto err; + } + return ret; + +err: + kfree(codec); + kfree(i2c); + return ret; +} + +static int ak4535_i2c_detach(struct i2c_client *client) +{ + struct snd_soc_codec *codec = i2c_get_clientdata(client); + i2c_detach_client(client); + kfree(codec->reg_cache); + kfree(client); + return 0; +} + +static int ak4535_i2c_attach(struct i2c_adapter *adap) +{ + return i2c_probe(adap, &addr_data, ak4535_codec_probe); +} + +/* corgi i2c codec control layer */ +static struct i2c_driver ak4535_i2c_driver = { + .driver = { + .name = "AK4535 I2C Codec", + .owner = THIS_MODULE, + }, + .id = I2C_DRIVERID_AK4535, + .attach_adapter = ak4535_i2c_attach, + .detach_client = ak4535_i2c_detach, + .command = NULL, +}; + +static struct i2c_client client_template = { + .name = "AK4535", + .driver = &ak4535_i2c_driver, +}; +#endif + +static int ak4535_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct ak4535_setup_data *setup; + struct snd_soc_codec *codec; + struct ak4535_priv *ak4535; + int ret = 0; + + printk(KERN_INFO "AK4535 Audio Codec %s", AK4535_VERSION); + + setup = socdev->codec_data; + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + ak4535 = kzalloc(sizeof(struct ak4535_priv), GFP_KERNEL); + if (ak4535 == NULL) { + kfree(codec); + return -ENOMEM; + } + + codec->private_data = ak4535; + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + ak4535_socdev = socdev; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + if (setup->i2c_address) { + normal_i2c[0] = setup->i2c_address; + codec->hw_write = (hw_write_t)i2c_master_send; + codec->hw_read = (hw_read_t)i2c_master_recv; + ret = i2c_add_driver(&ak4535_i2c_driver); + if (ret != 0) + printk(KERN_ERR "can't add i2c driver"); + } +#else + /* Add other interfaces here */ +#endif + return ret; +} + +/* power down chip */ +static int ak4535_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec->control_data) + ak4535_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&ak4535_i2c_driver); +#endif + kfree(codec->private_data); + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_ak4535 = { + .probe = ak4535_probe, + .remove = ak4535_remove, + .suspend = ak4535_suspend, + .resume = ak4535_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_ak4535); + +MODULE_DESCRIPTION("Soc AK4535 driver"); +MODULE_AUTHOR("Richard Purdie"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ak4535.h b/sound/soc/codecs/ak4535.h new file mode 100644 index 00000000000..fc686ddf753 --- /dev/null +++ b/sound/soc/codecs/ak4535.h @@ -0,0 +1,46 @@ +/* + * ak4535.h -- AK4535 Soc Audio driver + * + * Copyright 2005 Openedhand Ltd. + * + * Author: Richard Purdie + * + * Based on wm8753.h + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _AK4535_H +#define _AK4535_H + +/* AK4535 register space */ + +#define AK4535_PM1 0x0 +#define AK4535_PM2 0x1 +#define AK4535_SIG1 0x2 +#define AK4535_SIG2 0x3 +#define AK4535_MODE1 0x4 +#define AK4535_MODE2 0x5 +#define AK4535_DAC 0x6 +#define AK4535_MIC 0x7 +#define AK4535_TIMER 0x8 +#define AK4535_ALC1 0x9 +#define AK4535_ALC2 0xa +#define AK4535_PGA 0xb +#define AK4535_LATT 0xc +#define AK4535_RATT 0xd +#define AK4535_VOL 0xe +#define AK4535_STATUS 0xf + +#define AK4535_CACHEREGNUM 0x10 + +struct ak4535_setup_data { + unsigned short i2c_address; +}; + +extern struct snd_soc_codec_dai ak4535_dai; +extern struct snd_soc_codec_device soc_codec_dev_ak4535; + +#endif -- cgit From a5c95e90c1baa9c1114875264bbd283526eb8377 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 23 Jun 2008 14:51:29 +0100 Subject: ALSA: ASoC: Replace custom debug macros with pr_ equivalents Several ASoC codec drivers use custom macros equivalent to the standard pr_ macros, most of which are not actually used. Replace these custom macros with the standard ones. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/codecs/wm8510.c | 25 +++---------------------- sound/soc/codecs/wm8731.c | 25 +++---------------------- sound/soc/codecs/wm8750.c | 25 +++---------------------- sound/soc/codecs/wm8753.c | 25 +++---------------------- sound/soc/codecs/wm8990.c | 25 +++---------------------- 5 files changed, 15 insertions(+), 110 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 152e6f21154..b549f6753ab 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -30,25 +30,6 @@ #define AUDIO_NAME "wm8510" #define WM8510_VERSION "0.6" -/* - * Debug - */ - -#define WM8510_DEBUG 0 - -#ifdef WM8510_DEBUG -#define dbg(format, arg...) \ - printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg) -#else -#define dbg(format, arg...) do {} while (0) -#endif -#define err(format, arg...) \ - printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg) -#define info(format, arg...) \ - printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg) -#define warn(format, arg...) \ - printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg) - struct snd_soc_codec_device soc_codec_dev_wm8510; /* @@ -721,13 +702,13 @@ static int wm8510_codec_probe(struct i2c_adapter *adap, int addr, int kind) ret = i2c_attach_client(i2c); if (ret < 0) { - err("failed to attach codec at addr %x\n", addr); + pr_err("failed to attach codec at addr %x\n", addr); goto err; } ret = wm8510_init(socdev); if (ret < 0) { - err("failed to initialise WM8510\n"); + pr_err("failed to initialise WM8510\n"); goto err; } return ret; @@ -777,7 +758,7 @@ static int wm8510_probe(struct platform_device *pdev) struct snd_soc_codec *codec; int ret = 0; - info("WM8510 Audio Codec %s", WM8510_VERSION); + pr_info("WM8510 Audio Codec %s", WM8510_VERSION); setup = socdev->codec_data; codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 77880537a3c..3ff42ad65ed 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -31,25 +31,6 @@ #define AUDIO_NAME "wm8731" #define WM8731_VERSION "0.13" -/* - * Debug - */ - -#define WM8731_DEBUG 0 - -#ifdef WM8731_DEBUG -#define dbg(format, arg...) \ - printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg) -#else -#define dbg(format, arg...) do {} while (0) -#endif -#define err(format, arg...) \ - printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg) -#define info(format, arg...) \ - printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg) -#define warn(format, arg...) \ - printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg) - struct snd_soc_codec_device soc_codec_dev_wm8731; /* codec private data */ @@ -624,13 +605,13 @@ static int wm8731_codec_probe(struct i2c_adapter *adap, int addr, int kind) ret = i2c_attach_client(i2c); if (ret < 0) { - err("failed to attach codec at addr %x\n", addr); + pr_err("failed to attach codec at addr %x\n", addr); goto err; } ret = wm8731_init(socdev); if (ret < 0) { - err("failed to initialise WM8731\n"); + pr_err("failed to initialise WM8731\n"); goto err; } return ret; @@ -681,7 +662,7 @@ static int wm8731_probe(struct platform_device *pdev) struct wm8731_priv *wm8731; int ret = 0; - info("WM8731 Audio Codec %s", WM8731_VERSION); + pr_info("WM8731 Audio Codec %s", WM8731_VERSION); setup = socdev->codec_data; codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 1ae670a98c5..eb460c9aa63 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -31,25 +31,6 @@ #define AUDIO_NAME "WM8750" #define WM8750_VERSION "0.12" -/* - * Debug - */ - -#define WM8750_DEBUG 0 - -#ifdef WM8750_DEBUG -#define dbg(format, arg...) \ - printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg) -#else -#define dbg(format, arg...) do {} while (0) -#endif -#define err(format, arg...) \ - printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg) -#define info(format, arg...) \ - printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg) -#define warn(format, arg...) \ - printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg) - /* codec private data */ struct wm8750_priv { unsigned int sysclk; @@ -896,13 +877,13 @@ static int wm8750_codec_probe(struct i2c_adapter *adap, int addr, int kind) ret = i2c_attach_client(i2c); if (ret < 0) { - err("failed to attach codec at addr %x\n", addr); + pr_err("failed to attach codec at addr %x\n", addr); goto err; } ret = wm8750_init(socdev); if (ret < 0) { - err("failed to initialise WM8750\n"); + pr_err("failed to initialise WM8750\n"); goto err; } return ret; @@ -953,7 +934,7 @@ static int wm8750_probe(struct platform_device *pdev) struct wm8750_priv *wm8750; int ret = 0; - info("WM8750 Audio Codec %s", WM8750_VERSION); + pr_info("WM8750 Audio Codec %s", WM8750_VERSION); codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); if (codec == NULL) return -ENOMEM; diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 00b481183d4..be01a738f18 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -55,25 +55,6 @@ #define AUDIO_NAME "wm8753" #define WM8753_VERSION "0.16" -/* - * Debug - */ - -#define WM8753_DEBUG 0 - -#ifdef WM8753_DEBUG -#define dbg(format, arg...) \ - printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg) -#else -#define dbg(format, arg...) do {} while (0) -#endif -#define err(format, arg...) \ - printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg) -#define info(format, arg...) \ - printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg) -#define warn(format, arg...) \ - printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg) - static int caps_charge = 2000; module_param(caps_charge, int, 0); MODULE_PARM_DESC(caps_charge, "WM8753 cap charge time (msecs)"); @@ -1689,13 +1670,13 @@ static int wm8753_codec_probe(struct i2c_adapter *adap, int addr, int kind) ret = i2c_attach_client(i2c); if (ret < 0) { - err("failed to attach codec at addr %x\n", addr); + pr_err("failed to attach codec at addr %x\n", addr); goto err; } ret = wm8753_init(socdev); if (ret < 0) { - err("failed to initialise WM8753\n"); + pr_err("failed to initialise WM8753\n"); goto err; } @@ -1747,7 +1728,7 @@ static int wm8753_probe(struct platform_device *pdev) struct wm8753_priv *wm8753; int ret = 0; - info("WM8753 Audio Codec %s", WM8753_VERSION); + pr_info("WM8753 Audio Codec %s", WM8753_VERSION); setup = socdev->codec_data; codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index a7d25e2f252..a1371b73ba7 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -33,25 +33,6 @@ #define AUDIO_NAME "wm8990" #define WM8990_VERSION "0.2" -/* - * Debug - */ - -#define WM8990_DEBUG 0 - -#ifdef WM8990_DEBUG -#define dbg(format, arg...) \ - printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg) -#else -#define dbg(format, arg...) do {} while (0) -#endif -#define err(format, arg...) \ - printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg) -#define info(format, arg...) \ - printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg) -#define warn(format, arg...) \ - printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg) - /* codec private data */ struct wm8990_priv { unsigned int sysclk; @@ -1524,13 +1505,13 @@ static int wm8990_codec_probe(struct i2c_adapter *adap, int addr, int kind) ret = i2c_attach_client(i2c); if (ret < 0) { - err("failed to attach codec at addr %x\n", addr); + pr_err("failed to attach codec at addr %x\n", addr); goto err; } ret = wm8990_init(socdev); if (ret < 0) { - err("failed to initialise WM8990\n"); + pr_err("failed to initialise WM8990\n"); goto err; } return ret; @@ -1579,7 +1560,7 @@ static int wm8990_probe(struct platform_device *pdev) struct wm8990_priv *wm8990; int ret = 0; - info("WM8990 Audio Codec %s\n", WM8990_VERSION); + pr_info("WM8990 Audio Codec %s\n", WM8990_VERSION); setup = socdev->codec_data; codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); -- cgit From e2be2ccf9416bb4e0eb5f851967e79261f41d7e5 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Wed, 25 Jun 2008 14:42:07 +0300 Subject: ALSA: ASoC: Add support for generic DAPM register modifier widget This generic register modifier widget is for updating multiple codec register bits at once when the widget changes its power state. Signed-off-by: Jarkko Nikula Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/soc-dapm.c | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 728f3ac2f30..25363829e60 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -442,6 +442,25 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) return con; } +/* + * Handler for generic register modifier widget. + */ +int dapm_reg_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + unsigned int val; + + if (SND_SOC_DAPM_EVENT_ON(event)) + val = w->on_val; + else + val = w->off_val; + + snd_soc_update_bits(w->codec, -(w->reg + 1), + w->mask << w->shift, val << w->shift); + + return 0; +} + /* * Scan each dapm widget for complete audio path. * A complete path is a route that has valid endpoints i.e.:- -- cgit From 0bd72a3d49fc9fcac0202e4a2f394c7ee82af914 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Wed, 25 Jun 2008 14:42:08 +0300 Subject: ALSA: ASoC: TLV320AIC3X: Use register modifier widget for mic bias Signed-off-by: Jarkko Nikula Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/codecs/tlv320aic3x.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 528c26acdbe..0c7452f932e 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -456,10 +456,12 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { &aic3x_right_line2_mux_controls), /* Mic Bias */ - SND_SOC_DAPM_MICBIAS("Mic Bias 2V", MICBIAS_CTRL, 6, 0), - SND_SOC_DAPM_MICBIAS("Mic Bias 2.5V", MICBIAS_CTRL, 7, 0), - SND_SOC_DAPM_MICBIAS("Mic Bias AVDD", MICBIAS_CTRL, 6, 0), - SND_SOC_DAPM_MICBIAS("Mic Bias AVDD", MICBIAS_CTRL, 7, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "Mic Bias 2V", + MICBIAS_CTRL, 6, 3, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "Mic Bias 2.5V", + MICBIAS_CTRL, 6, 3, 2, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "Mic Bias AVDD", + MICBIAS_CTRL, 6, 3, 3, 0), /* Left PGA to Left Output bypass */ SND_SOC_DAPM_MIXER("Left PGA Bypass Mixer", SND_SOC_NOPM, 0, 0, -- cgit From 81971a14947b344ee7f6e3781f7ef36d8996f57a Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Wed, 25 Jun 2008 14:58:45 +0300 Subject: ALSA: ASoC: TLV320AIC3X: Modify only interface related bits in aic3x_set_dai_fmt Those two serial data interface control register bits have also other functions and they can be set before aic3x_set_dai_fmt is called. Signed-off-by: Jarkko Nikula Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/codecs/tlv320aic3x.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 0c7452f932e..29dc0ec3dd2 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -809,8 +809,10 @@ static int aic3x_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, { struct snd_soc_codec *codec = codec_dai->codec; struct aic3x_priv *aic3x = codec->private_data; - u8 iface_areg = 0; - u8 iface_breg = 0; + u8 iface_areg, iface_breg; + + iface_areg = aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLA) & 0x3f; + iface_breg = aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLB) & 0x3f; /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { -- cgit From ee15ffdb140b1a4fda5f6a3a8b9994031de5edb4 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Wed, 25 Jun 2008 14:58:46 +0300 Subject: ALSA: ASoC: TLV320AIC3X: Add support for digital microphone input AIC33 and AIC34 codecs in TLV320AIC3x family support digital microphone input. When enabled, the codec ADC takes bitstream input to low-pass filter from GPIO2 instead of its own delta-sigma modulator while providing oversampling clock through GPIO1. Signed-off-by: Jarkko Nikula Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/codecs/tlv320aic3x.c | 31 +++++++++++++++++++++++++++++++ 1 file changed, 31 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 29dc0ec3dd2..4f0bf26f170 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -455,6 +455,27 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { SND_SOC_DAPM_MUX("Right Line2R Mux", SND_SOC_NOPM, 0, 0, &aic3x_right_line2_mux_controls), + /* + * Not a real mic bias widget but similar function. This is for dynamic + * control of GPIO1 digital mic modulator clock output function when + * using digital mic. + */ + SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "GPIO1 dmic modclk", + AIC3X_GPIO1_REG, 4, 0xf, + AIC3X_GPIO1_FUNC_DIGITAL_MIC_MODCLK, + AIC3X_GPIO1_FUNC_DISABLED), + + /* + * Also similar function like mic bias. Selects digital mic with + * configurable oversampling rate instead of ADC converter. + */ + SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "DMic Rate 128", + AIC3X_ASD_INTF_CTRLA, 0, 3, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "DMic Rate 64", + AIC3X_ASD_INTF_CTRLA, 0, 3, 2, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "DMic Rate 32", + AIC3X_ASD_INTF_CTRLA, 0, 3, 3, 0), + /* Mic Bias */ SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "Mic Bias 2V", MICBIAS_CTRL, 6, 3, 1, 0), @@ -570,6 +591,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"Left PGA Mixer", "Mic3L Switch", "MIC3L"}, {"Left ADC", NULL, "Left PGA Mixer"}, + {"Left ADC", NULL, "GPIO1 dmic modclk"}, /* Right Input */ {"Right Line1R Mux", "single-ended", "LINE1R"}, @@ -583,6 +605,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"Right PGA Mixer", "Mic3R Switch", "MIC3R"}, {"Right ADC", NULL, "Right PGA Mixer"}, + {"Right ADC", NULL, "GPIO1 dmic modclk"}, /* Left PGA Bypass */ {"Left PGA Bypass Mixer", "Line Switch", "Left PGA Mixer"}, @@ -643,6 +666,14 @@ static const struct snd_soc_dapm_route intercon[] = { {"Right Line Out", NULL, "Right Line2 Bypass Mixer"}, {"Mono Out", NULL, "Right Line2 Bypass Mixer"}, {"Right HP Out", NULL, "Right Line2 Bypass Mixer"}, + + /* + * Logical path between digital mic enable and GPIO1 modulator clock + * output function + */ + {"GPIO1 dmic modclk", NULL, "DMic Rate 128"}, + {"GPIO1 dmic modclk", NULL, "DMic Rate 64"}, + {"GPIO1 dmic modclk", NULL, "DMic Rate 32"}, }; static int aic3x_add_widgets(struct snd_soc_codec *codec) -- cgit From 90b9e47671d561fdf2802aac4ac749b6c86405e0 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Wed, 25 Jun 2008 14:58:47 +0300 Subject: ALSA: ASoC: Add digital mic configuration to N810 machine driver Signed-off-by: Jarkko Nikula Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/omap/n810.c | 34 ++++++++++++++++++++++++++++++++++ 1 file changed, 34 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index c168a64e316..767b39f339a 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -46,11 +46,13 @@ static struct clk *func96m_clk; static int n810_spk_func; static int n810_jack_func; +static int n810_dmic_func; static void n810_ext_control(struct snd_soc_codec *codec) { snd_soc_dapm_set_endpoint(codec, "Ext Spk", n810_spk_func); snd_soc_dapm_set_endpoint(codec, "Headphone Jack", n810_jack_func); + snd_soc_dapm_set_endpoint(codec, "DMic", n810_dmic_func); snd_soc_dapm_sync_endpoints(codec); } @@ -150,6 +152,28 @@ static int n810_set_jack(struct snd_kcontrol *kcontrol, return 1; } +static int n810_get_input(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = n810_dmic_func; + + return 0; +} + +static int n810_set_input(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (n810_dmic_func == ucontrol->value.integer.value[0]) + return 0; + + n810_dmic_func = ucontrol->value.integer.value[0]; + n810_ext_control(codec); + + return 1; +} + static int n810_spk_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { @@ -175,6 +199,7 @@ static int n810_jack_event(struct snd_soc_dapm_widget *w, static const struct snd_soc_dapm_widget aic33_dapm_widgets[] = { SND_SOC_DAPM_SPK("Ext Spk", n810_spk_event), SND_SOC_DAPM_HP("Headphone Jack", n810_jack_event), + SND_SOC_DAPM_MIC("DMic", NULL), }; static const struct snd_soc_dapm_route audio_map[] = { @@ -183,13 +208,18 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Ext Spk", NULL, "LLOUT"}, {"Ext Spk", NULL, "RLOUT"}, + + {"DMic Rate 64", NULL, "Mic Bias 2V"}, + {"Mic Bias 2V", NULL, "DMic"}, }; static const char *spk_function[] = {"Off", "On"}; static const char *jack_function[] = {"Off", "Headphone"}; +static const char *input_function[] = {"ADC", "Digital Mic"}; static const struct soc_enum n810_enum[] = { SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(spk_function), spk_function), SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(jack_function), jack_function), + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(input_function), input_function), }; static const struct snd_kcontrol_new aic33_n810_controls[] = { @@ -197,6 +227,8 @@ static const struct snd_kcontrol_new aic33_n810_controls[] = { n810_get_spk, n810_set_spk), SOC_ENUM_EXT("Jack Function", n810_enum[1], n810_get_jack, n810_set_jack), + SOC_ENUM_EXT("Input Select", n810_enum[2], + n810_get_input, n810_set_input), }; static int n810_aic33_init(struct snd_soc_codec *codec) @@ -248,6 +280,8 @@ static struct snd_soc_machine snd_soc_machine_n810 = { /* Audio private data */ static struct aic3x_setup_data n810_aic33_setup = { .i2c_address = 0x18, + .gpio_func[0] = AIC3X_GPIO1_FUNC_DISABLED, + .gpio_func[1] = AIC3X_GPIO2_FUNC_DIGITAL_MIC_INPUT, }; /* Audio subsystem */ -- cgit From 5a2cc50f166babc26103279c4fbc9f2bf73b79de Mon Sep 17 00:00:00 2001 From: eric miao Date: Mon, 26 May 2008 03:28:09 +0100 Subject: [ARM] 5063/1: pxa: add clk support for pxa2xx I2S Signed-off-by: Eric Miao Acked-by: Mark Brown Signed-off-by: Russell King --- sound/soc/pxa/pxa2xx-i2s.c | 12 ++++++++++-- 1 file changed, 10 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 42507103097..e130346732b 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include #include @@ -40,6 +41,7 @@ struct pxa_i2s_port { u32 fmt; }; static struct pxa_i2s_port pxa_i2s; +static struct clk *clk_i2s; static struct pxa2xx_pcm_dma_params pxa2xx_i2s_pcm_stereo_out = { .name = "I2S PCM Stereo out", @@ -82,6 +84,10 @@ static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + clk_i2s = clk_get(NULL, "I2SCLK"); + if (IS_ERR(clk_i2s)) + return PTR_ERR(clk_i2s); + if (!cpu_dai->active) { SACR0 |= SACR0_RST; SACR0 = 0; @@ -149,7 +155,7 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, pxa_gpio_mode(gpio_bus[pxa_i2s.master].tx); pxa_gpio_mode(gpio_bus[pxa_i2s.master].frm); pxa_gpio_mode(gpio_bus[pxa_i2s.master].clk); - pxa_set_cken(CKEN_I2S, 1); + clk_enable(clk_i2s); pxa_i2s_wait(); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) @@ -234,8 +240,10 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream) if (SACR1 & (SACR1_DREC | SACR1_DRPL)) { SACR0 &= ~SACR0_ENB; pxa_i2s_wait(); - pxa_set_cken(CKEN_I2S, 0); + clk_disable(clk_i2s); } + + clk_put(clk_i2s); } #ifdef CONFIG_PM -- cgit From 4d20f70a787e03530786152ea27a6ed157205cdc Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Fri, 27 Jun 2008 14:07:57 +0300 Subject: ALSA: ASoC: TLV320AIC3X: Add mixer control for ADC highpass filter Signed-off-by: Jarkko Nikula Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/codecs/tlv320aic3x.c | 6 ++++++ sound/soc/codecs/tlv320aic3x.h | 2 ++ 2 files changed, 8 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 4f0bf26f170..d13830623db 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -223,6 +223,8 @@ static const char *aic3x_right_hpcom_mux[] = { "differential of HPROUT", "constant VCM", "single-ended", "differential of HPLCOM", "external feedback" }; static const char *aic3x_linein_mode_mux[] = { "single-ended", "differential" }; +static const char *aic3x_adc_hpf[] = + { "Disabled", "0.0045xFs", "0.0125xFs", "0.025xFs" }; #define LDAC_ENUM 0 #define RDAC_ENUM 1 @@ -232,6 +234,7 @@ static const char *aic3x_linein_mode_mux[] = { "single-ended", "differential" }; #define LINE1R_ENUM 5 #define LINE2L_ENUM 6 #define LINE2R_ENUM 7 +#define ADC_HPF_ENUM 8 static const struct soc_enum aic3x_enum[] = { SOC_ENUM_SINGLE(DAC_LINE_MUX, 6, 3, aic3x_left_dac_mux), @@ -242,6 +245,7 @@ static const struct soc_enum aic3x_enum[] = { SOC_ENUM_SINGLE(LINE1R_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux), SOC_ENUM_SINGLE(LINE2L_2_LADC_CTRL, 7, 2, aic3x_linein_mode_mux), SOC_ENUM_SINGLE(LINE2R_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux), + SOC_ENUM_DOUBLE(AIC3X_CODEC_DFILT_CTRL, 6, 4, 4, aic3x_adc_hpf), }; static const struct snd_kcontrol_new aic3x_snd_controls[] = { @@ -292,6 +296,8 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { /* Input */ SOC_DOUBLE_R("PGA Capture Volume", LADC_VOL, RADC_VOL, 0, 0x7f, 0), SOC_DOUBLE_R("PGA Capture Switch", LADC_VOL, RADC_VOL, 7, 0x01, 1), + + SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]), }; /* add non dapm controls */ diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index c1dd1ac0cea..e6009461063 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -37,6 +37,8 @@ #define AIC3X_ASD_INTF_CTRLB 9 /* Audio overflow status and PLL R value programming register */ #define AIC3X_OVRF_STATUS_AND_PLLR_REG 11 +/* Audio codec digital filter control register */ +#define AIC3X_CODEC_DFILT_CTRL 12 /* ADC PGA Gain control registers */ #define LADC_VOL 15 -- cgit From 5415552d21150dbfbff012de2b59e685ea232b1e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 2 Jul 2008 11:51:19 +0100 Subject: ALSA: ASoC: Advertise 16000Hz rate for WM9713 PCM interface Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/codecs/wm9713.c | 19 ++++++++++++++----- 1 file changed, 14 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 0db96374736..a4806189044 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1001,9 +1001,18 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream) return ac97_write(codec, AC97_PCM_SURR_DAC_RATE, runtime->rate); } -#define WM9713_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ - SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\ - SNDRV_PCM_RATE_48000) +#define WM9713_RATES (SNDRV_PCM_RATE_8000 | \ + SNDRV_PCM_RATE_11025 | \ + SNDRV_PCM_RATE_22050 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000) + +#define WM9713_PCM_RATES (SNDRV_PCM_RATE_8000 | \ + SNDRV_PCM_RATE_11025 | \ + SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_22050 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000) #define WM9713_PCM_FORMATS \ (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \ @@ -1051,13 +1060,13 @@ struct snd_soc_codec_dai wm9713_dai[] = { .stream_name = "Voice Playback", .channels_min = 1, .channels_max = 1, - .rates = WM9713_RATES, + .rates = WM9713_PCM_RATES, .formats = WM9713_PCM_FORMATS,}, .capture = { .stream_name = "Voice Capture", .channels_min = 1, .channels_max = 2, - .rates = WM9713_RATES, + .rates = WM9713_PCM_RATES, .formats = WM9713_PCM_FORMATS,}, .ops = { .hw_params = wm9713_pcm_hw_params, -- cgit From 15e4c72f520d1db9adc38ba157547a7c1fca45b2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 2 Jul 2008 11:51:20 +0100 Subject: ALSA: ASoC: Make pop/click debug wait times dynamically configurable DAPM supports adding a compile time configurable delay to the widget power sequences, aiding diagnosis of problems with pops and clicks being generated during them. This patch converts this to be configurable at run time via a sysfs file. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/soc-dapm.c | 67 +++++++++++++++++++++++++++++++++++++++++----------- 1 file changed, 53 insertions(+), 14 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 25363829e60..7c2dd4ec8df 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -54,16 +54,6 @@ #define dbg(format, arg...) #endif -#define POP_DEBUG 0 -#if POP_DEBUG -#define POP_TIME 500 /* 500 msecs - change if pop debug is too fast */ -#define pop_wait(time) schedule_timeout_uninterruptible(msecs_to_jiffies(time)) -#define pop_dbg(format, arg...) printk(format, ## arg); pop_wait(POP_TIME) -#else -#define pop_dbg(format, arg...) -#define pop_wait(time) -#endif - /* dapm power sequences - make this per codec in the future */ static int dapm_up_seq[] = { snd_soc_dapm_pre, snd_soc_dapm_micbias, snd_soc_dapm_mic, @@ -80,6 +70,28 @@ static int dapm_status = 1; module_param(dapm_status, int, 0); MODULE_PARM_DESC(dapm_status, "enable DPM sysfs entries"); +static unsigned int pop_time; + +static void pop_wait(void) +{ + if (pop_time) + schedule_timeout_uninterruptible(msecs_to_jiffies(pop_time)); +} + +static void pop_dbg(const char *fmt, ...) +{ + va_list args; + + va_start(args, fmt); + + if (pop_time) { + vprintk(fmt, args); + pop_wait(); + } + + va_end(args); +} + /* create a new dapm widget */ static inline struct snd_soc_dapm_widget *dapm_cnew_widget( const struct snd_soc_dapm_widget *_widget) @@ -217,9 +229,9 @@ static int dapm_update_bits(struct snd_soc_dapm_widget *widget) change = old != new; if (change) { pop_dbg("pop test %s : %s in %d ms\n", widget->name, - widget->power ? "on" : "off", POP_TIME); + widget->power ? "on" : "off", pop_time); snd_soc_write(codec, widget->reg, new); - pop_wait(POP_TIME); + pop_wait(); } dbg("reg %x old %x new %x change %d\n", widget->reg, old, new, change); return change; @@ -803,20 +815,47 @@ static ssize_t dapm_widget_show(struct device *dev, static DEVICE_ATTR(dapm_widget, 0444, dapm_widget_show, NULL); +/* pop/click delay times */ +static ssize_t dapm_pop_time_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + return sprintf(buf, "%d\n", pop_time); +} + +static ssize_t dapm_pop_time_store(struct device *dev, + struct device_attribute *attr, + const char *buf, size_t count) + +{ + if (strict_strtoul(buf, 10, &pop_time) < 0) + printk(KERN_ERR "Unable to parse pop_time setting\n"); + + return count; +} + +static DEVICE_ATTR(dapm_pop_time, 0744, dapm_pop_time_show, + dapm_pop_time_store); + int snd_soc_dapm_sys_add(struct device *dev) { int ret = 0; - if (dapm_status) + if (dapm_status) { ret = device_create_file(dev, &dev_attr_dapm_widget); + if (ret == 0) + ret = device_create_file(dev, &dev_attr_dapm_pop_time); + } + return ret; } static void snd_soc_dapm_sys_remove(struct device *dev) { - if (dapm_status) + if (dapm_status) { + device_remove_file(dev, &dev_attr_dapm_pop_time); device_remove_file(dev, &dev_attr_dapm_widget); + } } /* free all dapm widgets and resources */ -- cgit From 3f775987030cf7ff922765c18ddd0d311b4b46ef Mon Sep 17 00:00:00 2001 From: Manuel Lauss Date: Thu, 3 Jul 2008 09:33:10 +0200 Subject: ALSA: ASoC: AC97 codec PM Simple suspend/resume for AC97 ASoC codec. Signed-off-by: Manuel Lauss Acked-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/codecs/ac97.c | 25 +++++++++++++++++++++++++ 1 file changed, 25 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 300d007f233..e4516f3ce64 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -144,9 +144,34 @@ static int ac97_soc_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_PM +static int ac97_soc_suspend(struct platform_device *pdev, pm_message_t msg) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_ac97_suspend(socdev->codec->ac97); + + return 0; +} + +static int ac97_soc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_ac97_resume(socdev->codec->ac97); + + return 0; +} +#else +#define ac97_soc_suspend NULL +#define ac97_soc_resume NULL +#endif + struct snd_soc_codec_device soc_codec_dev_ac97 = { .probe = ac97_soc_probe, .remove = ac97_soc_remove, + .suspend = ac97_soc_suspend, + .resume = ac97_soc_resume, }; EXPORT_SYMBOL_GPL(soc_codec_dev_ac97); -- cgit From 73ead4855d98e330f2840523e1648575b9f2aae9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 4 Jul 2008 16:01:14 +0100 Subject: ALSA: ASoC: Fix warning from strict_strtoul() Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/soc-dapm.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7c2dd4ec8df..91cbbefefb0 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -827,7 +827,11 @@ static ssize_t dapm_pop_time_store(struct device *dev, const char *buf, size_t count) { - if (strict_strtoul(buf, 10, &pop_time) < 0) + unsigned long val; + + if (strict_strtoul(buf, 10, &val) >= 0) + pop_time = val; + else printk(KERN_ERR "Unable to parse pop_time setting\n"); return count; -- cgit From a5302181e5321664047f75715242aac4e0bbd17c Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 7 Jul 2008 13:35:17 +0100 Subject: ALSA: asoc: core - refactored DAPM pin control API. Refactored snd_soc_dapm_set_endpoint() to snd_soc_dapm_enable_pin() and snd_soc_dapm_disable_pin(). Renamed snd_soc_dapm_sync_endpoints() to snd_soc_dapm_sync(). Renamed snd_soc_dapm_get_endpoint_status() to snd_soc_dapm_get_pin_status(). Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/at91/eti_b1_wm8731.c | 10 ++-- sound/soc/codecs/tlv320aic3x.c | 4 +- sound/soc/davinci/davinci-evm.c | 16 ++--- sound/soc/omap/n810.c | 27 ++++++--- sound/soc/pxa/corgi.c | 42 ++++++++------ sound/soc/pxa/poodle.c | 24 ++++---- sound/soc/pxa/spitz.c | 62 ++++++++++---------- sound/soc/pxa/tosa.c | 30 +++++----- sound/soc/s3c24xx/neo1973_wm8753.c | 116 ++++++++++++++++++------------------- sound/soc/sh/sh7760-ac97.c | 2 +- sound/soc/soc-dapm.c | 81 ++++++++++++++++---------- 11 files changed, 225 insertions(+), 189 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c index 4a383a4a0ff..ad971e7061f 100644 --- a/sound/soc/at91/eti_b1_wm8731.c +++ b/sound/soc/at91/eti_b1_wm8731.c @@ -216,14 +216,14 @@ static int eti_b1_wm8731_init(struct snd_soc_codec *codec) snd_soc_dapm_add_route(codec, intercon, ARRAY_SIZE(intercon)); /* not connected */ - snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0); - snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0); + snd_soc_dapm_disable_pin(codec, "RLINEIN"); + snd_soc_dapm_disable_pin(codec, "LLINEIN"); /* always connected */ - snd_soc_dapm_set_endpoint(codec, "Int Mic", 1); - snd_soc_dapm_set_endpoint(codec, "Ext Spk", 1); + snd_soc_dapm_enable_pin(codec, "Int Mic"); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index d13830623db..954d39b7c04 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -29,7 +29,7 @@ * --------------------------------------- * * Hence the machine layer should disable unsupported inputs/outputs by - * snd_soc_dapm_set_endpoint(codec, "MONO_LOUT", 0), etc. + * snd_soc_dapm_disable_pin(codec, "MONO_LOUT"), etc. */ #include @@ -206,7 +206,7 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, } if (found) - snd_soc_dapm_sync_endpoints(widget->codec); + snd_soc_dapm_sync(widget->codec); } ret = snd_soc_update_bits(widget->codec, reg, val_mask, val); diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 4c70a0ed339..091eae3a963 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -103,17 +103,17 @@ static int evm_aic3x_init(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); /* not connected */ - snd_soc_dapm_set_endpoint(codec, "MONO_LOUT", 0); - snd_soc_dapm_set_endpoint(codec, "HPLCOM", 0); - snd_soc_dapm_set_endpoint(codec, "HPRCOM", 0); + snd_soc_dapm_disable_pin(codec, "MONO_LOUT"); + snd_soc_dapm_disable_pin(codec, "HPLCOM"); + snd_soc_dapm_disable_pin(codec, "HPRCOM"); /* always connected */ - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1); - snd_soc_dapm_set_endpoint(codec, "Line Out", 1); - snd_soc_dapm_set_endpoint(codec, "Mic Jack", 1); - snd_soc_dapm_set_endpoint(codec, "Line In", 1); + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Line Out"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + snd_soc_dapm_enable_pin(codec, "Line In"); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 767b39f339a..74f4599b4d7 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -50,11 +50,22 @@ static int n810_dmic_func; static void n810_ext_control(struct snd_soc_codec *codec) { - snd_soc_dapm_set_endpoint(codec, "Ext Spk", n810_spk_func); - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", n810_jack_func); - snd_soc_dapm_set_endpoint(codec, "DMic", n810_dmic_func); + if (n810_spk_func) + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + else + snd_soc_dapm_disable_pin(codec, "Ext Spk"); + + if (n810_jack_func) + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + else + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + + if (n810_dmic_func) + snd_soc_dapm_enable_pin(codec, "DMic"); + else + snd_soc_dapm_disable_pin(codec, "DMic); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); } static int n810_startup(struct snd_pcm_substream *substream) @@ -236,9 +247,9 @@ static int n810_aic33_init(struct snd_soc_codec *codec) int i, err; /* Not connected */ - snd_soc_dapm_set_endpoint(codec, "MONO_LOUT", 0); - snd_soc_dapm_set_endpoint(codec, "HPLCOM", 0); - snd_soc_dapm_set_endpoint(codec, "HPRCOM", 0); + snd_soc_dapm_disable_pin(codec, "MONO_LOUT"); + snd_soc_dapm_disable_pin(codec, "HPLCOM"); + snd_soc_dapm_disable_pin(codec, "HPRCOM"); /* Add N810 specific controls */ for (i = 0; i < ARRAY_SIZE(aic33_n810_controls); i++) { @@ -255,7 +266,7 @@ static int n810_aic33_init(struct snd_soc_codec *codec) /* Set up N810 specific audio path audio_map */ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index edeea63e80e..db18ef68b69 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -50,47 +50,51 @@ static int corgi_spk_func; static void corgi_ext_control(struct snd_soc_codec *codec) { - int spk = 0, mic = 0, line = 0, hp = 0, hs = 0; - /* set up jack connection */ switch (corgi_jack_func) { case CORGI_HP: - hp = 1; /* set = unmute headphone */ set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L); set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R); + snd_soc_dapm_disable_pin(codec, "Mic Jack"); + snd_soc_dapm_disable_pin(codec, "Line Jack"); + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); break; case CORGI_MIC: - mic = 1; /* reset = mute headphone */ reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L); reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + snd_soc_dapm_disable_pin(codec, "Line Jack"); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); break; case CORGI_LINE: - line = 1; reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L); reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R); + snd_soc_dapm_disable_pin(codec, "Mic Jack"); + snd_soc_dapm_enable_pin(codec, "Line Jack"); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); break; case CORGI_HEADSET: - hs = 1; - mic = 1; reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L); set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + snd_soc_dapm_disable_pin(codec, "Line Jack"); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Headset Jack"); break; } if (corgi_spk_func == CORGI_SPK_ON) - spk = 1; - - /* set the enpoints to their new connetion states */ - snd_soc_dapm_set_endpoint(codec, "Ext Spk", spk); - snd_soc_dapm_set_endpoint(codec, "Mic Jack", mic); - snd_soc_dapm_set_endpoint(codec, "Line Jack", line); - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", hp); - snd_soc_dapm_set_endpoint(codec, "Headset Jack", hs); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + else + snd_soc_dapm_disable_pin(codec, "Ext Spk"); /* signal a DAPM event */ - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); } static int corgi_startup(struct snd_pcm_substream *substream) @@ -285,8 +289,8 @@ static int corgi_wm8731_init(struct snd_soc_codec *codec) { int i, err; - snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0); - snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0); + snd_soc_dapm_disable_pin(codec, "LLINEIN"); + snd_soc_dapm_disable_pin(codec, "RLINEIN"); /* Add corgi specific controls */ for (i = 0; i < ARRAY_SIZE(wm8731_corgi_controls); i++) { @@ -303,7 +307,7 @@ static int corgi_wm8731_init(struct snd_soc_codec *codec) /* Set up corgi specific audio path audio_map */ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 810f1fe158a..36cbf69f5f8 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -48,8 +48,6 @@ static int poodle_spk_func; static void poodle_ext_control(struct snd_soc_codec *codec) { - int spk = 0; - /* set up jack connection */ if (poodle_jack_func == POODLE_HP) { /* set = unmute headphone */ @@ -57,23 +55,23 @@ static void poodle_ext_control(struct snd_soc_codec *codec) POODLE_LOCOMO_GPIO_MUTE_L, 1); locomo_gpio_write(&poodle_locomo_device.dev, POODLE_LOCOMO_GPIO_MUTE_R, 1); - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1); + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); } else { locomo_gpio_write(&poodle_locomo_device.dev, POODLE_LOCOMO_GPIO_MUTE_L, 0); locomo_gpio_write(&poodle_locomo_device.dev, POODLE_LOCOMO_GPIO_MUTE_R, 0); - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); } - if (poodle_spk_func == POODLE_SPK_ON) - spk = 1; - /* set the enpoints to their new connetion states */ - snd_soc_dapm_set_endpoint(codec, "Ext Spk", spk); + if (poodle_spk_func == POODLE_SPK_ON) + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + else + snd_soc_dapm_disable_pin(codec, "Ext Spk"); /* signal a DAPM event */ - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); } static int poodle_startup(struct snd_pcm_substream *substream) @@ -248,9 +246,9 @@ static int poodle_wm8731_init(struct snd_soc_codec *codec) { int i, err; - snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0); - snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0); - snd_soc_dapm_set_endpoint(codec, "MICIN", 1); + snd_soc_dapm_disable_pin(codec, "LLINEIN"); + snd_soc_dapm_disable_pin(codec, "RLINEIN"); + snd_soc_dapm_enable_pin(codec, "MICIN"); /* Add poodle specific controls */ for (i = 0; i < ARRAY_SIZE(wm8731_poodle_controls); i++) { @@ -267,7 +265,7 @@ static int poodle_wm8731_init(struct snd_soc_codec *codec) /* Set up poodle specific audio path audio_map */ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 092b5c776b4..ec18163fddd 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -51,60 +51,60 @@ static int spitz_spk_func; static void spitz_ext_control(struct snd_soc_codec *codec) { if (spitz_spk_func == SPITZ_SPK_ON) - snd_soc_dapm_set_endpoint(codec, "Ext Spk", 1); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); else - snd_soc_dapm_set_endpoint(codec, "Ext Spk", 0); + snd_soc_dapm_disable_pin(codec, "Ext Spk"); /* set up jack connection */ switch (spitz_jack_func) { case SPITZ_HP: /* enable and unmute hp jack, disable mic bias */ - snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Line Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); + snd_soc_dapm_disable_pin(codec, "Mic Jack"); + snd_soc_dapm_disable_pin(codec, "Line Jack"); + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); break; case SPITZ_MIC: /* enable mic jack and bias, mute hp */ - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Line Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Mic Jack", 1); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); + snd_soc_dapm_disable_pin(codec, "Line Jack"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); break; case SPITZ_LINE: /* enable line jack, disable mic bias and mute hp */ - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Line Jack", 1); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); + snd_soc_dapm_disable_pin(codec, "Mic Jack"); + snd_soc_dapm_enable_pin(codec, "Line Jack"); reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); break; case SPITZ_HEADSET: /* enable and unmute headset jack enable mic bias, mute L hp */ - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Mic Jack", 1); - snd_soc_dapm_set_endpoint(codec, "Line Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Jack", 1); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + snd_soc_dapm_disable_pin(codec, "Line Jack"); + snd_soc_dapm_enable_pin(codec, "Headset Jack"); reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); break; case SPITZ_HP_OFF: /* jack removed, everything off */ - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Line Jack", 0); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); + snd_soc_dapm_disable_pin(codec, "Mic Jack"); + snd_soc_dapm_disable_pin(codec, "Line Jack"); reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); break; } - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); } static int spitz_startup(struct snd_pcm_substream *substream) @@ -291,13 +291,13 @@ static int spitz_wm8750_init(struct snd_soc_codec *codec) int i, err; /* NC codec pins */ - snd_soc_dapm_set_endpoint(codec, "RINPUT1", 0); - snd_soc_dapm_set_endpoint(codec, "LINPUT2", 0); - snd_soc_dapm_set_endpoint(codec, "RINPUT2", 0); - snd_soc_dapm_set_endpoint(codec, "LINPUT3", 0); - snd_soc_dapm_set_endpoint(codec, "RINPUT3", 0); - snd_soc_dapm_set_endpoint(codec, "OUT3", 0); - snd_soc_dapm_set_endpoint(codec, "MONO", 0); + snd_soc_dapm_disable_pin(codec, "RINPUT1"); + snd_soc_dapm_disable_pin(codec, "LINPUT2"); + snd_soc_dapm_disable_pin(codec, "RINPUT2"); + snd_soc_dapm_disable_pin(codec, "LINPUT3"); + snd_soc_dapm_disable_pin(codec, "RINPUT3"); + snd_soc_dapm_disable_pin(codec, "OUT3"); + snd_soc_dapm_disable_pin(codec, "MONO"); /* Add spitz specific controls */ for (i = 0; i < ARRAY_SIZE(wm8750_spitz_controls); i++) { @@ -314,7 +314,7 @@ static int spitz_wm8750_init(struct snd_soc_codec *codec) /* Set up spitz specific audio paths */ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index 465ff0f458e..dba7689c508 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -52,29 +52,31 @@ static int tosa_spk_func; static void tosa_ext_control(struct snd_soc_codec *codec) { - int spk = 0, mic_int = 0, hp = 0, hs = 0; - /* set up jack connection */ switch (tosa_jack_func) { case TOSA_HP: - hp = 1; + snd_soc_dapm_disable_pin(codec, "Mic (Internal)"); + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); break; case TOSA_MIC_INT: - mic_int = 1; + snd_soc_dapm_enable_pin(codec, "Mic (Internal)"); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); break; case TOSA_HEADSET: - hs = 1; + snd_soc_dapm_disable_pin(codec, "Mic (Internal)"); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Headset Jack"); break; } if (tosa_spk_func == TOSA_SPK_ON) - spk = 1; + snd_soc_dapm_enable_pin(codec, "Speaker"); + else + snd_soc_dapm_disable_pin(codec, "Speaker"); - snd_soc_dapm_set_endpoint(codec, "Speaker", spk); - snd_soc_dapm_set_endpoint(codec, "Mic (Internal)", mic_int); - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", hp); - snd_soc_dapm_set_endpoint(codec, "Headset Jack", hs); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); } static int tosa_startup(struct snd_pcm_substream *substream) @@ -191,8 +193,8 @@ static int tosa_ac97_init(struct snd_soc_codec *codec) { int i, err; - snd_soc_dapm_set_endpoint(codec, "OUT3", 0); - snd_soc_dapm_set_endpoint(codec, "MONOOUT", 0); + snd_soc_dapm_disable_pin(codec, "OUT3"); + snd_soc_dapm_disable_pin(codec, "MONOOUT"); /* add tosa specific controls */ for (i = 0; i < ARRAY_SIZE(tosa_controls); i++) { @@ -209,7 +211,7 @@ static int tosa_ac97_init(struct snd_soc_codec *codec) /* set up tosa specific audio path audio_map */ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 34851238dea..f053e85ff60 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -250,77 +250,77 @@ static int set_scenario_endpoints(struct snd_soc_codec *codec, int scenario) switch (neo1973_scenario) { case NEO_AUDIO_OFF: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); break; case NEO_GSM_CALL_AUDIO_HANDSET: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 1); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 1); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 1); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 1); + snd_soc_dapm_enable_pin(codec, "Audio Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_enable_pin(codec, "Call Mic"); break; case NEO_GSM_CALL_AUDIO_HEADSET: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 1); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 1); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 1); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 1); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_enable_pin(codec, "Audio Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line In"); + snd_soc_dapm_enable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); break; case NEO_GSM_CALL_AUDIO_BLUETOOTH: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 1); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 1); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); break; case NEO_STEREO_TO_SPEAKERS: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 1); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_enable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); break; case NEO_STEREO_TO_HEADPHONES: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 1); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_enable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); break; case NEO_CAPTURE_HANDSET: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 1); + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_enable_pin(codec, "Call Mic"); break; case NEO_CAPTURE_HEADSET: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 1); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_enable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); break; case NEO_CAPTURE_BLUETOOTH: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); break; default: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); } - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } @@ -511,12 +511,12 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec) DBG("Entered %s\n", __func__); /* set up NC codec pins */ - snd_soc_dapm_set_endpoint(codec, "LOUT2", 0); - snd_soc_dapm_set_endpoint(codec, "ROUT2", 0); - snd_soc_dapm_set_endpoint(codec, "OUT3", 0); - snd_soc_dapm_set_endpoint(codec, "OUT4", 0); - snd_soc_dapm_set_endpoint(codec, "LINE1", 0); - snd_soc_dapm_set_endpoint(codec, "LINE2", 0); + snd_soc_dapm_disable_pin(codec, "LOUT2"); + snd_soc_dapm_disable_pin(codec, "ROUT2"); + snd_soc_dapm_disable_pin(codec, "OUT3"); + snd_soc_dapm_disable_pin(codec, "OUT4"); + snd_soc_dapm_disable_pin(codec, "LINE1"); + snd_soc_dapm_disable_pin(codec, "LINE2"); /* set endpoints to default mode */ @@ -539,7 +539,7 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec) err = snd_soc_dapm_add_routes(codec, dapm_routes, ARRAY_SIZE(dapm_routes)); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } diff --git a/sound/soc/sh/sh7760-ac97.c b/sound/soc/sh/sh7760-ac97.c index 2f91de84c5c..846d1b3a630 100644 --- a/sound/soc/sh/sh7760-ac97.c +++ b/sound/soc/sh/sh7760-ac97.c @@ -25,7 +25,7 @@ extern struct snd_soc_platform sh7760_soc_platform; static int machine_init(struct snd_soc_codec *codec) { - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 91cbbefefb0..94296b5dc58 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -880,8 +880,25 @@ 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) +{ + struct snd_soc_dapm_widget *w; + + list_for_each_entry(w, &codec->dapm_widgets, list) { + if (!strcmp(w->name, pin)) { + dbg("dapm: %s: pin %s\n", codec->name, pin); + w->connected = status; + return 0; + } + } + + dbg("dapm: %s: configuring unknown pin %s\n", codec->name, pin); + return -EINVAL; +} + /** - * snd_soc_dapm_sync_endpoints - scan and power dapm paths + * snd_soc_dapm_sync - scan and power dapm paths * @codec: audio codec * * Walks all dapm audio paths and powers widgets according to their @@ -889,11 +906,11 @@ static void dapm_free_widgets(struct snd_soc_codec *codec) * * Returns 0 for success. */ -int snd_soc_dapm_sync_endpoints(struct snd_soc_codec *codec) +int snd_soc_dapm_sync(struct snd_soc_codec *codec) { return dapm_power_widgets(codec, SND_SOC_DAPM_STREAM_NOP); } -EXPORT_SYMBOL_GPL(snd_soc_dapm_sync_endpoints); +EXPORT_SYMBOL_GPL(snd_soc_dapm_sync); static int snd_soc_dapm_add_route(struct snd_soc_codec *codec, const char *sink, const char *control, const char *source) @@ -1441,53 +1458,57 @@ int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev, } /** - * snd_soc_dapm_set_endpoint - set audio endpoint status - * @codec: audio codec - * @endpoint: audio signal endpoint (or start point) - * @status: point status - * - * Set audio endpoint status - connected or disconnected. + * snd_soc_dapm_enable_pin - enable pin. + * @snd_soc_codec: SoC codec + * @pin: pin name * - * Returns 0 for success else error. + * Enables input/output pin and it's parents or children widgets iff there is + * a valid audio route and active audio stream. + * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to + * do any widget power switching. */ -int snd_soc_dapm_set_endpoint(struct snd_soc_codec *codec, - char *endpoint, int status) +int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, char *pin) { - struct snd_soc_dapm_widget *w; - - list_for_each_entry(w, &codec->dapm_widgets, list) { - if (!strcmp(w->name, endpoint)) { - w->connected = status; - return 0; - } - } + return snd_soc_dapm_set_pin(codec, pin, 1); +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_enable_pin); - return -ENODEV; +/** + * snd_soc_dapm_disable_pin - disable pin. + * @codec: SoC codec + * @pin: pin name + * + * Disables input/output pin and it's parents or children widgets. + * 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) +{ + return snd_soc_dapm_set_pin(codec, pin, 0); } -EXPORT_SYMBOL_GPL(snd_soc_dapm_set_endpoint); +EXPORT_SYMBOL_GPL(snd_soc_dapm_disable_pin); /** - * snd_soc_dapm_get_endpoint_status - get audio endpoint status + * snd_soc_dapm_get_pin_status - get audio pin status * @codec: audio codec - * @endpoint: audio signal endpoint (or start point) + * @pin: audio signal pin endpoint (or start point) * - * Get audio endpoint status - connected or disconnected. + * Get audio pin status - connected or disconnected. * - * Returns status + * Returns 1 for connected otherwise 0. */ -int snd_soc_dapm_get_endpoint_status(struct snd_soc_codec *codec, - char *endpoint) +int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, char *pin) { struct snd_soc_dapm_widget *w; list_for_each_entry(w, &codec->dapm_widgets, list) { - if (!strcmp(w->name, endpoint)) + if (!strcmp(w->name, pin)) return w->connected; } return 0; } -EXPORT_SYMBOL_GPL(snd_soc_dapm_get_endpoint_status); +EXPORT_SYMBOL_GPL(snd_soc_dapm_get_pin_status); /** * snd_soc_dapm_free - free dapm resources -- cgit From 3c4b266fe642bcaebe2b95edb56c3f8802924ff9 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 7 Jul 2008 16:07:17 +0100 Subject: ALSA: asoc: core - merge structs snd_soc_codec_dai and snd_soc_cpu_dai. This patch series merges struct snd_soc_codec_dai and struct snd_soc_cpu_dai into struct snd_soc_dai in preparation for further ASoC v2 patches. This merger removes duplication in both DAI structures and simplifies the API for other users. Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/soc-core.c | 50 +++++++++++++++++++++++++------------------------- 1 file changed, 25 insertions(+), 25 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index bdbbc6a980f..4d626b47b2f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -134,8 +134,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_dai_link *machine = rtd->dai; struct snd_soc_platform *platform = socdev->platform; - struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_codec_dai *codec_dai = machine->codec_dai; + struct snd_soc_dai *cpu_dai = machine->cpu_dai; + struct snd_soc_dai *codec_dai = machine->codec_dai; int ret = 0; mutex_lock(&pcm_mutex); @@ -272,7 +272,7 @@ static void close_delayed_work(struct work_struct *work) struct snd_soc_device *socdev = container_of(work, struct snd_soc_device, delayed_work.work); struct snd_soc_codec *codec = socdev->codec; - struct snd_soc_codec_dai *codec_dai; + struct snd_soc_dai *codec_dai; int i; mutex_lock(&pcm_mutex); @@ -323,8 +323,8 @@ static int soc_codec_close(struct snd_pcm_substream *substream) struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_dai_link *machine = rtd->dai; struct snd_soc_platform *platform = socdev->platform; - struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_codec_dai *codec_dai = machine->codec_dai; + struct snd_soc_dai *cpu_dai = machine->cpu_dai; + struct snd_soc_dai *codec_dai = machine->codec_dai; struct snd_soc_codec *codec = socdev->codec; mutex_lock(&pcm_mutex); @@ -384,8 +384,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_dai_link *machine = rtd->dai; struct snd_soc_platform *platform = socdev->platform; - struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_codec_dai *codec_dai = machine->codec_dai; + struct snd_soc_dai *cpu_dai = machine->cpu_dai; + struct snd_soc_dai *codec_dai = machine->codec_dai; struct snd_soc_codec *codec = socdev->codec; int ret = 0; @@ -489,8 +489,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_dai_link *machine = rtd->dai; struct snd_soc_platform *platform = socdev->platform; - struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_codec_dai *codec_dai = machine->codec_dai; + struct snd_soc_dai *cpu_dai = machine->cpu_dai; + struct snd_soc_dai *codec_dai = machine->codec_dai; int ret = 0; mutex_lock(&pcm_mutex); @@ -559,8 +559,8 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_dai_link *machine = rtd->dai; struct snd_soc_platform *platform = socdev->platform; - struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_codec_dai *codec_dai = machine->codec_dai; + struct snd_soc_dai *cpu_dai = machine->cpu_dai; + struct snd_soc_dai *codec_dai = machine->codec_dai; struct snd_soc_codec *codec = socdev->codec; mutex_lock(&pcm_mutex); @@ -594,8 +594,8 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_dai_link *machine = rtd->dai; struct snd_soc_platform *platform = socdev->platform; - struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_codec_dai *codec_dai = machine->codec_dai; + struct snd_soc_dai *cpu_dai = machine->cpu_dai; + struct snd_soc_dai *codec_dai = machine->codec_dai; int ret; if (codec_dai->ops.trigger) { @@ -651,7 +651,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) /* mute any active DAC's */ for (i = 0; i < machine->num_links; i++) { - struct snd_soc_codec_dai *dai = machine->dai_link[i].codec_dai; + struct snd_soc_dai *dai = machine->dai_link[i].codec_dai; if (dai->dai_ops.digital_mute && dai->playback.active) dai->dai_ops.digital_mute(dai, 1); } @@ -664,7 +664,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) machine->suspend_pre(pdev, state); for (i = 0; i < machine->num_links; i++) { - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->suspend && cpu_dai->type != SND_SOC_DAI_AC97) cpu_dai->suspend(pdev, cpu_dai); if (platform->suspend) @@ -690,7 +690,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) codec_dev->suspend(pdev, state); for (i = 0; i < machine->num_links; i++) { - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->suspend && cpu_dai->type == SND_SOC_DAI_AC97) cpu_dai->suspend(pdev, cpu_dai); } @@ -726,7 +726,7 @@ static void soc_resume_deferred(struct work_struct *work) machine->resume_pre(pdev); for (i = 0; i < machine->num_links; i++) { - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->resume && cpu_dai->type == SND_SOC_DAI_AC97) cpu_dai->resume(pdev, cpu_dai); } @@ -747,13 +747,13 @@ static void soc_resume_deferred(struct work_struct *work) /* unmute any active DACs */ for (i = 0; i < machine->num_links; i++) { - struct snd_soc_codec_dai *dai = machine->dai_link[i].codec_dai; + struct snd_soc_dai *dai = machine->dai_link[i].codec_dai; if (dai->dai_ops.digital_mute && dai->playback.active) dai->dai_ops.digital_mute(dai, 0); } for (i = 0; i < machine->num_links; i++) { - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->resume && cpu_dai->type != SND_SOC_DAI_AC97) cpu_dai->resume(pdev, cpu_dai); if (platform->resume) @@ -803,7 +803,7 @@ static int soc_probe(struct platform_device *pdev) } for (i = 0; i < machine->num_links; i++) { - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->probe) { ret = cpu_dai->probe(pdev, cpu_dai); if (ret < 0) @@ -838,7 +838,7 @@ platform_err: cpu_dai_err: for (i--; i >= 0; i--) { - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->remove) cpu_dai->remove(pdev, cpu_dai); } @@ -867,7 +867,7 @@ static int soc_remove(struct platform_device *pdev) codec_dev->remove(pdev); for (i = 0; i < machine->num_links; i++) { - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->remove) cpu_dai->remove(pdev, cpu_dai); } @@ -895,8 +895,8 @@ static int soc_new_pcm(struct snd_soc_device *socdev, struct snd_soc_dai_link *dai_link, int num) { struct snd_soc_codec *codec = socdev->codec; - struct snd_soc_codec_dai *codec_dai = dai_link->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = dai_link->cpu_dai; + struct snd_soc_dai *codec_dai = dai_link->codec_dai; + struct snd_soc_dai *cpu_dai = dai_link->cpu_dai; struct snd_soc_pcm_runtime *rtd; struct snd_pcm *pcm; char new_name[64]; @@ -1211,7 +1211,7 @@ void snd_soc_free_pcms(struct snd_soc_device *socdev) { struct snd_soc_codec *codec = socdev->codec; #ifdef CONFIG_SND_SOC_AC97_BUS - struct snd_soc_codec_dai *codec_dai; + struct snd_soc_dai *codec_dai; int i; #endif -- cgit From 453ba20b2800bc07a3eaf220010dc0127cc552fd Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 7 Jul 2008 16:07:29 +0100 Subject: ALSA: asoc: at32 - merge structs snd_soc_codec_dai and snd_soc_cpu_dai. This patch series merges struct snd_soc_codec_dai and struct snd_soc_cpu_dai into struct snd_soc_dai for AT32 platform. Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/at32/at32-pcm.c | 6 +++--- sound/soc/at32/at32-ssc.c | 12 ++++++------ sound/soc/at32/at32-ssc.h | 2 +- sound/soc/at32/playpaq_wm8510.c | 6 +++--- 4 files changed, 13 insertions(+), 13 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/at32/at32-pcm.c b/sound/soc/at32/at32-pcm.c index cf76e89fae6..435f1daf177 100644 --- a/sound/soc/at32/at32-pcm.c +++ b/sound/soc/at32/at32-pcm.c @@ -366,7 +366,7 @@ static struct snd_pcm_ops at32_pcm_ops = { static u64 at32_pcm_dmamask = 0xffffffff; static int at32_pcm_new(struct snd_card *card, - struct snd_soc_codec_dai *dai, + struct snd_soc_dai *dai, struct snd_pcm *pcm) { int ret = 0; @@ -422,7 +422,7 @@ static void at32_pcm_free_dma_buffers(struct snd_pcm *pcm) #ifdef CONFIG_PM static int at32_pcm_suspend(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { struct snd_pcm_runtime *runtime = dai->runtime; struct at32_runtime_data *prtd; @@ -447,7 +447,7 @@ static int at32_pcm_suspend(struct platform_device *pdev, static int at32_pcm_resume(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { struct snd_pcm_runtime *runtime = dai->runtime; struct at32_runtime_data *prtd; diff --git a/sound/soc/at32/at32-ssc.c b/sound/soc/at32/at32-ssc.c index 0ca44107f18..4ef6492c902 100644 --- a/sound/soc/at32/at32-ssc.c +++ b/sound/soc/at32/at32-ssc.c @@ -336,7 +336,7 @@ static void at32_ssc_shutdown(struct snd_pcm_substream *substream) /* * Set the SSC system clock rate */ -static int at32_ssc_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, +static int at32_ssc_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { /* TODO: What the heck do I do here? */ @@ -348,7 +348,7 @@ static int at32_ssc_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, /* * Record DAI format for use by hw_params() */ -static int at32_ssc_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, +static int at32_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct at32_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; @@ -362,7 +362,7 @@ static int at32_ssc_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, /* * Record SSC clock dividers for use in hw_params() */ -static int at32_ssc_set_dai_clkdiv(struct snd_soc_cpu_dai *cpu_dai, +static int at32_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, int div_id, int div) { struct at32_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; @@ -670,7 +670,7 @@ static int at32_ssc_prepare(struct snd_pcm_substream *substream) #ifdef CONFIG_PM static int at32_ssc_suspend(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai) + struct snd_soc_dai *cpu_dai) { struct at32_ssc_info *ssc_p; @@ -699,7 +699,7 @@ static int at32_ssc_suspend(struct platform_device *pdev, static int at32_ssc_resume(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai) + struct snd_soc_dai *cpu_dai) { struct at32_ssc_info *ssc_p; u32 cr; @@ -746,7 +746,7 @@ static int at32_ssc_resume(struct platform_device *pdev, SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_S32) -struct snd_soc_cpu_dai at32_ssc_dai[NUM_SSC_DEVICES] = { +struct snd_soc_dai at32_ssc_dai[NUM_SSC_DEVICES] = { { .name = "at32-ssc0", .id = 0, diff --git a/sound/soc/at32/at32-ssc.h b/sound/soc/at32/at32-ssc.h index 3c6901a6c03..3c052dbbe46 100644 --- a/sound/soc/at32/at32-ssc.h +++ b/sound/soc/at32/at32-ssc.h @@ -52,7 +52,7 @@ struct at32_ssc_info { #define AT32_SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */ -extern struct snd_soc_cpu_dai at32_ssc_dai[]; +extern struct snd_soc_dai at32_ssc_dai[]; diff --git a/sound/soc/at32/playpaq_wm8510.c b/sound/soc/at32/playpaq_wm8510.c index d6b9fd5e612..18ac7d7391c 100644 --- a/sound/soc/at32/playpaq_wm8510.c +++ b/sound/soc/at32/playpaq_wm8510.c @@ -82,7 +82,7 @@ static struct clk *_pll0; #if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE static struct ssc_clock_data playpaq_wm8510_calc_ssc_clock( struct snd_pcm_hw_params *params, - struct snd_soc_cpu_dai *cpu_dai) + struct snd_soc_dai *cpu_dai) { struct at32_ssc_info *ssc_p = cpu_dai->private_data; struct ssc_device *ssc = ssc_p->ssc; @@ -132,8 +132,8 @@ static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct at32_ssc_info *ssc_p = cpu_dai->private_data; struct ssc_device *ssc = ssc_p->ssc; unsigned int pll_out = 0, bclk = 0, mclk_div = 0; -- cgit From d37ae539a1d76da8fe5a939ce8b6d818501c8716 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 7 Jul 2008 16:07:37 +0100 Subject: ALSA: asoc: at91 - merge structs snd_soc_codec_dai and snd_soc_cpu_dai. This patch merges struct snd_soc_codec_dai and struct snd_soc_cpu_dai into struct snd_soc_dai for the AT91 platform. Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/at91/at91-pcm.c | 6 +++--- sound/soc/at91/at91-ssc.c | 12 ++++++------ sound/soc/at91/at91-ssc.h | 2 +- sound/soc/at91/eti_b1_wm8731.c | 8 ++++---- 4 files changed, 14 insertions(+), 14 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/at91/at91-pcm.c b/sound/soc/at91/at91-pcm.c index ccac6bd2889..d47492b2b6e 100644 --- a/sound/soc/at91/at91-pcm.c +++ b/sound/soc/at91/at91-pcm.c @@ -318,7 +318,7 @@ static int at91_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, static u64 at91_pcm_dmamask = 0xffffffff; static int at91_pcm_new(struct snd_card *card, - struct snd_soc_codec_dai *dai, struct snd_pcm *pcm) + struct snd_soc_dai *dai, struct snd_pcm *pcm) { int ret = 0; @@ -367,7 +367,7 @@ static void at91_pcm_free_dma_buffers(struct snd_pcm *pcm) #ifdef CONFIG_PM static int at91_pcm_suspend(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { struct snd_pcm_runtime *runtime = dai->runtime; struct at91_runtime_data *prtd; @@ -392,7 +392,7 @@ static int at91_pcm_suspend(struct platform_device *pdev, } static int at91_pcm_resume(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { struct snd_pcm_runtime *runtime = dai->runtime; struct at91_runtime_data *prtd; diff --git a/sound/soc/at91/at91-ssc.c b/sound/soc/at91/at91-ssc.c index bc35d00a38f..c3625b665c5 100644 --- a/sound/soc/at91/at91-ssc.c +++ b/sound/soc/at91/at91-ssc.c @@ -281,7 +281,7 @@ static void at91_ssc_shutdown(struct snd_pcm_substream *substream) /* * Record the SSC system clock rate. */ -static int at91_ssc_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, +static int at91_ssc_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { /* @@ -303,7 +303,7 @@ static int at91_ssc_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, /* * Record the DAI format for use in hw_params(). */ -static int at91_ssc_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, +static int at91_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct at91_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; @@ -315,7 +315,7 @@ static int at91_ssc_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, /* * Record SSC clock dividers for use in hw_params(). */ -static int at91_ssc_set_dai_clkdiv(struct snd_soc_cpu_dai *cpu_dai, +static int at91_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, int div_id, int div) { struct at91_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; @@ -634,7 +634,7 @@ static int at91_ssc_prepare(struct snd_pcm_substream *substream) #ifdef CONFIG_PM static int at91_ssc_suspend(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai) + struct snd_soc_dai *cpu_dai) { struct at91_ssc_info *ssc_p; @@ -662,7 +662,7 @@ static int at91_ssc_suspend(struct platform_device *pdev, } static int at91_ssc_resume(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai) + struct snd_soc_dai *cpu_dai) { struct at91_ssc_info *ssc_p; @@ -700,7 +700,7 @@ static int at91_ssc_resume(struct platform_device *pdev, #define AT91_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -struct snd_soc_cpu_dai at91_ssc_dai[NUM_SSC_DEVICES] = { +struct snd_soc_dai at91_ssc_dai[NUM_SSC_DEVICES] = { { .name = "at91-ssc0", .id = 0, .type = SND_SOC_DAI_PCM, diff --git a/sound/soc/at91/at91-ssc.h b/sound/soc/at91/at91-ssc.h index b188f973df9..6b7bf382d06 100644 --- a/sound/soc/at91/at91-ssc.h +++ b/sound/soc/at91/at91-ssc.h @@ -21,7 +21,7 @@ #define AT91SSC_TCMR_PERIOD 1 /* BCLK divider for transmit FS */ #define AT91SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */ -extern struct snd_soc_cpu_dai at91_ssc_dai[]; +extern struct snd_soc_dai at91_ssc_dai[]; #endif /* _AT91_SSC_H */ diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c index ad971e7061f..9d4213963c2 100644 --- a/sound/soc/at91/eti_b1_wm8731.c +++ b/sound/soc/at91/eti_b1_wm8731.c @@ -53,8 +53,8 @@ static struct clk *pllb_clk; static int eti_b1_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; int ret; /* cpu clock is the AT91 master clock sent to the SSC */ @@ -87,8 +87,8 @@ static int eti_b1_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; int ret; #ifdef CONFIG_SND_AT91_SOC_ETI_SLAVE -- cgit From 9cb132d743cf39b3bbe4288e9035217e7237a0bb Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 7 Jul 2008 16:07:42 +0100 Subject: ALSA: asoc: davinci - merge structs snd_soc_codec_dai and snd_soc_cpu_dai. This patch merges struct snd_soc_codec_dai and struct snd_soc_cpu_dai into struct snd_soc_dai for the DaVinci platform. Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/davinci/davinci-evm.c | 4 ++-- sound/soc/davinci/davinci-i2s.c | 14 +++++++------- sound/soc/davinci/davinci-i2s.h | 2 +- sound/soc/davinci/davinci-pcm.c | 2 +- 4 files changed, 11 insertions(+), 11 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 091eae3a963..4249e6a8574 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -33,8 +33,8 @@ static int evm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; int ret = 0; /* set codec DAI configuration */ diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index c3b545ccff7..5ebf1ff71c4 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -147,7 +147,7 @@ static void davinci_mcbsp_stop(struct snd_pcm_substream *substream) static int davinci_i2s_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data; cpu_dai->dma_data = dev->dma_params[substream->stream]; @@ -155,7 +155,7 @@ static int davinci_i2s_startup(struct snd_pcm_substream *substream) return 0; } -static int davinci_i2s_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, +static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct davinci_mcbsp_dev *dev = cpu_dai->private_data; @@ -296,11 +296,11 @@ static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd) } static int davinci_i2s_probe(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_machine *machine = socdev->machine; - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai; struct davinci_mcbsp_dev *dev; struct resource *mem, *ioarea; struct evm_snd_platform_data *pdata; @@ -358,11 +358,11 @@ err_release_region: } static void davinci_i2s_remove(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_machine *machine = socdev->machine; - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai; struct davinci_mcbsp_dev *dev = cpu_dai->private_data; struct resource *mem; @@ -378,7 +378,7 @@ static void davinci_i2s_remove(struct platform_device *pdev, #define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 -struct snd_soc_cpu_dai davinci_i2s_dai = { +struct snd_soc_dai davinci_i2s_dai = { .name = "davinci-i2s", .id = 0, .type = SND_SOC_DAI_I2S, diff --git a/sound/soc/davinci/davinci-i2s.h b/sound/soc/davinci/davinci-i2s.h index 9592d17db32..c5b091807ee 100644 --- a/sound/soc/davinci/davinci-i2s.h +++ b/sound/soc/davinci/davinci-i2s.h @@ -12,6 +12,6 @@ #ifndef _DAVINCI_I2S_H #define _DAVINCI_I2S_H -extern struct snd_soc_cpu_dai davinci_i2s_dai; +extern struct snd_soc_dai davinci_i2s_dai; #endif diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 6a76927c997..6a5e56a782b 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -350,7 +350,7 @@ static void davinci_pcm_free(struct snd_pcm *pcm) static u64 davinci_pcm_dmamask = 0xffffffff; static int davinci_pcm_new(struct snd_card *card, - struct snd_soc_codec_dai *dai, struct snd_pcm *pcm) + struct snd_soc_dai *dai, struct snd_pcm *pcm) { int ret; -- cgit From e550e17ffeb8cf8db27724eaf2ad05f77388afb9 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 7 Jul 2008 16:07:52 +0100 Subject: ALSA: asoc: codecs - merge structs snd_soc_codec_dai and snd_soc_cpu_dai. This patch merges struct snd_soc_codec_dai and struct snd_soc_cpu_dai into struct snd_soc_dai for the codec drivers. Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/codecs/ac97.c | 2 +- sound/soc/codecs/ac97.h | 2 +- sound/soc/codecs/ak4535.c | 8 ++++---- sound/soc/codecs/ak4535.h | 2 +- sound/soc/codecs/cs4270.c | 8 ++++---- sound/soc/codecs/cs4270.h | 2 +- sound/soc/codecs/tlv320aic3x.c | 8 ++++---- sound/soc/codecs/tlv320aic3x.h | 2 +- sound/soc/codecs/uda1380.c | 6 +++--- sound/soc/codecs/uda1380.h | 2 +- sound/soc/codecs/wm8510.c | 10 +++++----- sound/soc/codecs/wm8510.h | 2 +- sound/soc/codecs/wm8731.c | 8 ++++---- sound/soc/codecs/wm8731.h | 2 +- sound/soc/codecs/wm8750.c | 8 ++++---- sound/soc/codecs/wm8750.h | 2 +- sound/soc/codecs/wm8753.c | 28 ++++++++++++++-------------- sound/soc/codecs/wm8753.h | 2 +- sound/soc/codecs/wm8990.c | 12 ++++++------ sound/soc/codecs/wm8990.h | 2 +- sound/soc/codecs/wm9712.c | 2 +- sound/soc/codecs/wm9712.h | 2 +- sound/soc/codecs/wm9713.c | 10 +++++----- sound/soc/codecs/wm9713.h | 2 +- 24 files changed, 67 insertions(+), 67 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index e4516f3ce64..61fd96ca7bc 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -41,7 +41,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream) SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\ SNDRV_PCM_RATE_48000) -struct snd_soc_codec_dai ac97_dai = { +struct snd_soc_dai ac97_dai = { .name = "AC97 HiFi", .type = SND_SOC_DAI_AC97, .playback = { diff --git a/sound/soc/codecs/ac97.h b/sound/soc/codecs/ac97.h index 2bf6d69fd06..281aa42e2bb 100644 --- a/sound/soc/codecs/ac97.h +++ b/sound/soc/codecs/ac97.h @@ -14,6 +14,6 @@ #define __LINUX_SND_SOC_AC97_H extern struct snd_soc_codec_device soc_codec_dev_ac97; -extern struct snd_soc_codec_dai ac97_dai; +extern struct snd_soc_dai ac97_dai; #endif diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 469266e881d..b26003c4f3e 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -329,7 +329,7 @@ static int ak4535_add_widgets(struct snd_soc_codec *codec) return 0; } -static int ak4535_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, +static int ak4535_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; @@ -369,7 +369,7 @@ static int ak4535_hw_params(struct snd_pcm_substream *substream, return 0; } -static int ak4535_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int ak4535_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -394,7 +394,7 @@ static int ak4535_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, return 0; } -static int ak4535_mute(struct snd_soc_codec_dai *dai, int mute) +static int ak4535_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; u16 mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC) & 0xffdf; @@ -436,7 +436,7 @@ static int ak4535_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) -struct snd_soc_codec_dai ak4535_dai = { +struct snd_soc_dai ak4535_dai = { .name = "AK4535", .playback = { .stream_name = "Playback", diff --git a/sound/soc/codecs/ak4535.h b/sound/soc/codecs/ak4535.h index fc686ddf753..e9fe30e2c05 100644 --- a/sound/soc/codecs/ak4535.h +++ b/sound/soc/codecs/ak4535.h @@ -40,7 +40,7 @@ struct ak4535_setup_data { unsigned short i2c_address; }; -extern struct snd_soc_codec_dai ak4535_dai; +extern struct snd_soc_dai ak4535_dai; extern struct snd_soc_codec_device soc_codec_dev_ak4535; #endif diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index e73fcfd9f5c..9deb8c74fdf 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -201,7 +201,7 @@ static struct { * driver what the input settings can be. This would need to be implemented * for stand-alone mode to work. */ -static int cs4270_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, +static int cs4270_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; @@ -251,7 +251,7 @@ static int cs4270_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, * data for playback only, but ASoC currently does not support different * formats for playback vs. record. */ -static int cs4270_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int format) { struct snd_soc_codec *codec = codec_dai->codec; @@ -471,7 +471,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, * board does not have the MUTEA or MUTEB pins connected to such circuitry, * then this function will do nothing. */ -static int cs4270_mute(struct snd_soc_codec_dai *dai, int mute) +static int cs4270_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; int reg6; @@ -667,7 +667,7 @@ error: #endif /* USE_I2C*/ -struct snd_soc_codec_dai cs4270_dai = { +struct snd_soc_dai cs4270_dai = { .name = "CS4270", .playback = { .stream_name = "Playback", diff --git a/sound/soc/codecs/cs4270.h b/sound/soc/codecs/cs4270.h index 0ced49b7804..adc6cd9667d 100644 --- a/sound/soc/codecs/cs4270.h +++ b/sound/soc/codecs/cs4270.h @@ -16,7 +16,7 @@ * The ASoC codec DAI structure for the CS4270. Assign this structure to * the .codec_dai field of your machine driver's snd_soc_dai_link structure. */ -extern struct snd_soc_codec_dai cs4270_dai; +extern struct snd_soc_dai cs4270_dai; /* * The ASoC codec device structure for the CS4270. Assign this structure diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 954d39b7c04..b1dce5f459d 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -814,7 +814,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, return 0; } -static int aic3x_mute(struct snd_soc_codec_dai *dai, int mute) +static int aic3x_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; u8 ldac_reg = aic3x_read_reg_cache(codec, LDAC_VOL) & ~MUTE_ON; @@ -831,7 +831,7 @@ static int aic3x_mute(struct snd_soc_codec_dai *dai, int mute) return 0; } -static int aic3x_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, +static int aic3x_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; @@ -841,7 +841,7 @@ static int aic3x_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, return 0; } -static int aic3x_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -990,7 +990,7 @@ EXPORT_SYMBOL_GPL(aic3x_headset_detected); #define AIC3X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) -struct snd_soc_codec_dai aic3x_dai = { +struct snd_soc_dai aic3x_dai = { .name = "aic3x", .playback = { .stream_name = "Playback", diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index e6009461063..d76c079b86e 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -228,7 +228,7 @@ struct aic3x_setup_data { unsigned int gpio_func[2]; }; -extern struct snd_soc_codec_dai aic3x_dai; +extern struct snd_soc_dai aic3x_dai; extern struct snd_soc_codec_device soc_codec_dev_aic3x; #endif /* _AIC3X_H */ diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 6d5335b14d5..a52d6d9e007 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -372,7 +372,7 @@ static int uda1380_add_widgets(struct snd_soc_codec *codec) return 0; } -static int uda1380_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int uda1380_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -499,7 +499,7 @@ static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream) uda1380_write(codec, UDA1380_CLK, clk); } -static int uda1380_mute(struct snd_soc_codec_dai *codec_dai, int mute) +static int uda1380_mute(struct snd_soc_dai *codec_dai, int mute) { struct snd_soc_codec *codec = codec_dai->codec; u16 mute_reg = uda1380_read_reg_cache(codec, UDA1380_DEEMP) & ~R13_MTM; @@ -542,7 +542,7 @@ static int uda1380_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) -struct snd_soc_codec_dai uda1380_dai[] = { +struct snd_soc_dai uda1380_dai[] = { { .name = "UDA1380", .playback = { diff --git a/sound/soc/codecs/uda1380.h b/sound/soc/codecs/uda1380.h index f9d885c8bf0..50c603e2c9f 100644 --- a/sound/soc/codecs/uda1380.h +++ b/sound/soc/codecs/uda1380.h @@ -83,7 +83,7 @@ struct uda1380_setup_data { #define UDA1380_DAI_PLAYBACK 1 /* playback DAI */ #define UDA1380_DAI_CAPTURE 2 /* capture DAI */ -extern struct snd_soc_codec_dai uda1380_dai[3]; +extern struct snd_soc_dai uda1380_dai[3]; extern struct snd_soc_codec_device soc_codec_dev_uda1380; #endif /* _UDA1380_H */ diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index b549f6753ab..67325fd9544 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -332,7 +332,7 @@ static void pll_factors(unsigned int target, unsigned int source) pll_div.k = K; } -static int wm8510_set_dai_pll(struct snd_soc_codec_dai *codec_dai, +static int wm8510_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, unsigned int freq_in, unsigned int freq_out) { struct snd_soc_codec *codec = codec_dai->codec; @@ -368,7 +368,7 @@ static int wm8510_set_dai_pll(struct snd_soc_codec_dai *codec_dai, /* * Configure WM8510 clock dividers. */ -static int wm8510_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai, +static int wm8510_set_dai_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div) { struct snd_soc_codec *codec = codec_dai->codec; @@ -402,7 +402,7 @@ static int wm8510_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai, return 0; } -static int wm8510_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8510_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -510,7 +510,7 @@ static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } -static int wm8510_mute(struct snd_soc_codec_dai *dai, int mute) +static int wm8510_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; u16 mute_reg = wm8510_read_reg_cache(codec, WM8510_DAC) & 0xffbf; @@ -554,7 +554,7 @@ static int wm8510_set_bias_level(struct snd_soc_codec *codec, #define WM8510_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -struct snd_soc_codec_dai wm8510_dai = { +struct snd_soc_dai wm8510_dai = { .name = "WM8510 HiFi", .playback = { .stream_name = "Playback", diff --git a/sound/soc/codecs/wm8510.h b/sound/soc/codecs/wm8510.h index c862e7b7d53..f5d2e42eb3f 100644 --- a/sound/soc/codecs/wm8510.h +++ b/sound/soc/codecs/wm8510.h @@ -97,7 +97,7 @@ struct wm8510_setup_data { unsigned short i2c_address; }; -extern struct snd_soc_codec_dai wm8510_dai; +extern struct snd_soc_dai wm8510_dai; extern struct snd_soc_codec_device soc_codec_dev_wm8510; #endif diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 3ff42ad65ed..369d39c3f74 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -318,7 +318,7 @@ static void wm8731_shutdown(struct snd_pcm_substream *substream) } } -static int wm8731_mute(struct snd_soc_codec_dai *dai, int mute) +static int wm8731_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; u16 mute_reg = wm8731_read_reg_cache(codec, WM8731_APDIGI) & 0xfff7; @@ -330,7 +330,7 @@ static int wm8731_mute(struct snd_soc_codec_dai *dai, int mute) return 0; } -static int wm8731_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, +static int wm8731_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; @@ -349,7 +349,7 @@ static int wm8731_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, } -static int wm8731_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8731_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -443,7 +443,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, #define WM8731_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) -struct snd_soc_codec_dai wm8731_dai = { +struct snd_soc_dai wm8731_dai = { .name = "WM8731", .playback = { .stream_name = "Playback", diff --git a/sound/soc/codecs/wm8731.h b/sound/soc/codecs/wm8731.h index 5bcab6a7afb..99f2e3c60e3 100644 --- a/sound/soc/codecs/wm8731.h +++ b/sound/soc/codecs/wm8731.h @@ -38,7 +38,7 @@ struct wm8731_setup_data { unsigned short i2c_address; }; -extern struct snd_soc_codec_dai wm8731_dai; +extern struct snd_soc_dai wm8731_dai; extern struct snd_soc_codec_device soc_codec_dev_wm8731; #endif diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index eb460c9aa63..e23cb09f0d1 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -536,7 +536,7 @@ static inline int get_coeff(int mclk, int rate) return -EINVAL; } -static int wm8750_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, +static int wm8750_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; @@ -554,7 +554,7 @@ static int wm8750_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, return -EINVAL; } -static int wm8750_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8750_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -647,7 +647,7 @@ static int wm8750_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } -static int wm8750_mute(struct snd_soc_codec_dai *dai, int mute) +static int wm8750_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; u16 mute_reg = wm8750_read_reg_cache(codec, WM8750_ADCDAC) & 0xfff7; @@ -692,7 +692,7 @@ static int wm8750_set_bias_level(struct snd_soc_codec *codec, #define WM8750_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) -struct snd_soc_codec_dai wm8750_dai = { +struct snd_soc_dai wm8750_dai = { .name = "WM8750", .playback = { .stream_name = "Playback", diff --git a/sound/soc/codecs/wm8750.h b/sound/soc/codecs/wm8750.h index a97a54a6348..8ef30e628b2 100644 --- a/sound/soc/codecs/wm8750.h +++ b/sound/soc/codecs/wm8750.h @@ -61,7 +61,7 @@ struct wm8750_setup_data { unsigned short i2c_address; }; -extern struct snd_soc_codec_dai wm8750_dai; +extern struct snd_soc_dai wm8750_dai; extern struct snd_soc_codec_device soc_codec_dev_wm8750; #endif diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index be01a738f18..8604809f0c3 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -740,7 +740,7 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int target, pll_div->k = K; } -static int wm8753_set_dai_pll(struct snd_soc_codec_dai *codec_dai, +static int wm8753_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, unsigned int freq_in, unsigned int freq_out) { u16 reg, enable; @@ -863,7 +863,7 @@ static int get_coeff(int mclk, int rate) /* * Clock after PLL and dividers */ -static int wm8753_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, +static int wm8753_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; @@ -890,7 +890,7 @@ static int wm8753_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, /* * Set's ADC and Voice DAC format. */ -static int wm8753_vdac_adc_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8753_vdac_adc_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -960,7 +960,7 @@ static int wm8753_pcm_hw_params(struct snd_pcm_substream *substream, /* * Set's PCM dai fmt and BCLK. */ -static int wm8753_pcm_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8753_pcm_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1026,7 +1026,7 @@ static int wm8753_pcm_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, return 0; } -static int wm8753_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai, +static int wm8753_set_dai_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1054,7 +1054,7 @@ static int wm8753_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai, /* * Set's HiFi DAC format. */ -static int wm8753_hdac_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8753_hdac_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1087,7 +1087,7 @@ static int wm8753_hdac_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, /* * Set's I2S DAI format. */ -static int wm8753_i2s_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8753_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1195,7 +1195,7 @@ static int wm8753_i2s_hw_params(struct snd_pcm_substream *substream, return 0; } -static int wm8753_mode1v_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8753_mode1v_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1210,7 +1210,7 @@ static int wm8753_mode1v_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, return wm8753_pcm_set_dai_fmt(codec_dai, fmt); } -static int wm8753_mode1h_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8753_mode1h_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { if (wm8753_hdac_set_dai_fmt(codec_dai, fmt) < 0) @@ -1218,7 +1218,7 @@ static int wm8753_mode1h_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, return wm8753_i2s_set_dai_fmt(codec_dai, fmt); } -static int wm8753_mode2_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8753_mode2_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1233,7 +1233,7 @@ static int wm8753_mode2_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, return wm8753_i2s_set_dai_fmt(codec_dai, fmt); } -static int wm8753_mode3_4_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8753_mode3_4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1250,7 +1250,7 @@ static int wm8753_mode3_4_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, return wm8753_i2s_set_dai_fmt(codec_dai, fmt); } -static int wm8753_mute(struct snd_soc_codec_dai *dai, int mute) +static int wm8753_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; u16 mute_reg = wm8753_read_reg_cache(codec, WM8753_DAC) & 0xfff7; @@ -1316,7 +1316,7 @@ static int wm8753_set_bias_level(struct snd_soc_codec *codec, * 3. Voice disabled - HIFI over HIFI * 4. Voice disabled - HIFI over HIFI, uses voice DAI LRC for capture */ -static const struct snd_soc_codec_dai wm8753_all_dai[] = { +static const struct snd_soc_dai wm8753_all_dai[] = { /* DAI HiFi mode 1 */ { .name = "WM8753 HiFi", .id = 1, @@ -1456,7 +1456,7 @@ static const struct snd_soc_codec_dai wm8753_all_dai[] = { }, }; -struct snd_soc_codec_dai wm8753_dai[2]; +struct snd_soc_dai wm8753_dai[2]; EXPORT_SYMBOL_GPL(wm8753_dai); static void wm8753_set_dai_mode(struct snd_soc_codec *codec, unsigned int mode) diff --git a/sound/soc/codecs/wm8753.h b/sound/soc/codecs/wm8753.h index 95e2a1f5316..44f5f1ff0cc 100644 --- a/sound/soc/codecs/wm8753.h +++ b/sound/soc/codecs/wm8753.h @@ -120,7 +120,7 @@ struct wm8753_setup_data { #define WM8753_DAI_HIFI 0 #define WM8753_DAI_VOICE 1 -extern struct snd_soc_codec_dai wm8753_dai[2]; +extern struct snd_soc_dai wm8753_dai[2]; extern struct snd_soc_codec_device soc_codec_dev_wm8753; #endif diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index a1371b73ba7..3ecce5168e9 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1029,7 +1029,7 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int target, pll_div->k = K; } -static int wm8990_set_dai_pll(struct snd_soc_codec_dai *codec_dai, +static int wm8990_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, unsigned int freq_in, unsigned int freq_out) { u16 reg; @@ -1065,7 +1065,7 @@ static int wm8990_set_dai_pll(struct snd_soc_codec_dai *codec_dai, /* * Clock after PLL and dividers */ -static int wm8990_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, +static int wm8990_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1078,7 +1078,7 @@ static int wm8990_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, /* * Set's ADC and Voice DAC format. */ -static int wm8990_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8990_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1131,7 +1131,7 @@ static int wm8990_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, return 0; } -static int wm8990_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai, +static int wm8990_set_dai_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1196,7 +1196,7 @@ static int wm8990_hw_params(struct snd_pcm_substream *substream, return 0; } -static int wm8990_mute(struct snd_soc_codec_dai *dai, int mute) +static int wm8990_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; u16 val; @@ -1329,7 +1329,7 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, * 1. ADC/DAC on Primary Interface * 2. ADC on Primary Interface/DAC on secondary */ -struct snd_soc_codec_dai wm8990_dai = { +struct snd_soc_dai wm8990_dai = { /* ADC/DAC on primary */ .name = "WM8990 ADC/DAC Primary", .id = 1, diff --git a/sound/soc/codecs/wm8990.h b/sound/soc/codecs/wm8990.h index bf9f8823dfc..6bea5748528 100644 --- a/sound/soc/codecs/wm8990.h +++ b/sound/soc/codecs/wm8990.h @@ -825,7 +825,7 @@ struct wm8990_setup_data { #define WM8990_ADCCLK_DIV 2 #define WM8990_BCLK_DIV 3 -extern struct snd_soc_codec_dai wm8990_dai; +extern struct snd_soc_dai wm8990_dai; extern struct snd_soc_codec_device soc_codec_dev_wm8990; #endif /* __WM8990REGISTERDEFS_H__ */ diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 47390113bd0..9fc8edd8222 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -532,7 +532,7 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream) SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\ SNDRV_PCM_RATE_48000) -struct snd_soc_codec_dai wm9712_dai[] = { +struct snd_soc_dai wm9712_dai[] = { { .name = "AC97 HiFi", .type = SND_SOC_DAI_AC97_BUS, diff --git a/sound/soc/codecs/wm9712.h b/sound/soc/codecs/wm9712.h index 719105d61e6..d29e8a18ca6 100644 --- a/sound/soc/codecs/wm9712.h +++ b/sound/soc/codecs/wm9712.h @@ -8,7 +8,7 @@ #define WM9712_DAI_AC97_HIFI 0 #define WM9712_DAI_AC97_AUX 1 -extern struct snd_soc_codec_dai wm9712_dai[2]; +extern struct snd_soc_dai wm9712_dai[2]; extern struct snd_soc_codec_device soc_codec_dev_wm9712; #endif diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index a4806189044..38d1fe0971f 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -789,7 +789,7 @@ static int wm9713_set_pll(struct snd_soc_codec *codec, return 0; } -static int wm9713_set_dai_pll(struct snd_soc_codec_dai *codec_dai, +static int wm9713_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, unsigned int freq_in, unsigned int freq_out) { struct snd_soc_codec *codec = codec_dai->codec; @@ -800,7 +800,7 @@ static int wm9713_set_dai_pll(struct snd_soc_codec_dai *codec_dai, * Tristate the PCM DAI lines, tristate can be disabled by calling * wm9713_set_dai_fmt() */ -static int wm9713_set_dai_tristate(struct snd_soc_codec_dai *codec_dai, +static int wm9713_set_dai_tristate(struct snd_soc_dai *codec_dai, int tristate) { struct snd_soc_codec *codec = codec_dai->codec; @@ -816,7 +816,7 @@ static int wm9713_set_dai_tristate(struct snd_soc_codec_dai *codec_dai, * Configure WM9713 clock dividers. * Voice DAC needs 256 FS */ -static int wm9713_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai, +static int wm9713_set_dai_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div) { struct snd_soc_codec *codec = codec_dai->codec; @@ -858,7 +858,7 @@ static int wm9713_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai, return 0; } -static int wm9713_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm9713_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1018,7 +1018,7 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream) (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \ SNDRV_PCM_FORMAT_S24_LE) -struct snd_soc_codec_dai wm9713_dai[] = { +struct snd_soc_dai wm9713_dai[] = { { .name = "AC97 HiFi", .type = SND_SOC_DAI_AC97_BUS, diff --git a/sound/soc/codecs/wm9713.h b/sound/soc/codecs/wm9713.h index d357b6c8134..63b8d81756e 100644 --- a/sound/soc/codecs/wm9713.h +++ b/sound/soc/codecs/wm9713.h @@ -46,7 +46,7 @@ #define WM9713_DAI_PCM_VOICE 2 extern struct snd_soc_codec_device soc_codec_dev_wm9713; -extern struct snd_soc_codec_dai wm9713_dai[3]; +extern struct snd_soc_dai wm9713_dai[3]; int wm9713_reset(struct snd_soc_codec *codec, int try_warm); -- cgit From 8cf7b2b393b26b9ba1ccbeeb271a88e51d58f3e3 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 7 Jul 2008 16:08:00 +0100 Subject: ALSA: asoc: fsl - merge structs snd_soc_codec_dai and snd_soc_cpu_dai. This patch merges struct snd_soc_codec_dai and struct snd_soc_cpu_dai into struct snd_soc_dai for the Freescale PPC platform. Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/fsl/fsl_dma.c | 2 +- sound/soc/fsl/fsl_dma.h | 2 +- sound/soc/fsl/fsl_ssi.c | 24 ++++++++++++------------ sound/soc/fsl/fsl_ssi.h | 4 ++-- sound/soc/fsl/mpc8610_hpcd.c | 4 ++-- 5 files changed, 18 insertions(+), 18 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 78de7168d2b..da2bc590286 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -282,7 +282,7 @@ static irqreturn_t fsl_dma_isr(int irq, void *dev_id) * once for each .dai_link in the machine driver's snd_soc_machine * structure. */ -static int fsl_dma_new(struct snd_card *card, struct snd_soc_codec_dai *dai, +static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) { static u64 fsl_dma_dmamask = DMA_BIT_MASK(32); diff --git a/sound/soc/fsl/fsl_dma.h b/sound/soc/fsl/fsl_dma.h index 430a6ce8b0d..385d4a42603 100644 --- a/sound/soc/fsl/fsl_dma.h +++ b/sound/soc/fsl/fsl_dma.h @@ -126,7 +126,7 @@ struct fsl_dma_link_descriptor { u8 res[4]; /* Reserved */ } __attribute__ ((aligned(32), packed)); -/* DMA information needed to create a snd_soc_cpu_dai object +/* DMA information needed to create a snd_soc_dai object * * ssi_stx_phys: bus address of SSI STX register to use * ssi_srx_phys: bus address of SSI SRX register to use diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index f588545698f..71bff33f552 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -82,7 +82,7 @@ struct fsl_ssi_private { struct device *dev; unsigned int playback; unsigned int capture; - struct snd_soc_cpu_dai cpu_dai; + struct snd_soc_dai cpu_dai; struct device_attribute dev_attr; struct { @@ -479,7 +479,7 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream) * @freq: the frequency of the given clock ID, currently ignored * @dir: SND_SOC_CLOCK_IN (clock slave) or SND_SOC_CLOCK_OUT (clock master) */ -static int fsl_ssi_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, +static int fsl_ssi_set_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { @@ -497,7 +497,7 @@ static int fsl_ssi_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, * * @format: one of SND_SOC_DAIFMT_xxx */ -static int fsl_ssi_set_fmt(struct snd_soc_cpu_dai *cpu_dai, unsigned int format) +static int fsl_ssi_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format) { return (format == SND_SOC_DAIFMT_I2S) ? 0 : -EINVAL; } @@ -505,7 +505,7 @@ static int fsl_ssi_set_fmt(struct snd_soc_cpu_dai *cpu_dai, unsigned int format) /** * fsl_ssi_dai_template: template CPU DAI for the SSI */ -static struct snd_soc_cpu_dai fsl_ssi_dai_template = { +static struct snd_soc_dai fsl_ssi_dai_template = { .playback = { /* The SSI does not support monaural audio. */ .channels_min = 2, @@ -569,15 +569,15 @@ static ssize_t fsl_sysfs_ssi_show(struct device *dev, } /** - * fsl_ssi_create_dai: create a snd_soc_cpu_dai structure + * fsl_ssi_create_dai: create a snd_soc_dai structure * - * This function is called by the machine driver to create a snd_soc_cpu_dai + * This function is called by the machine driver to create a snd_soc_dai * structure. The function creates an ssi_private object, which contains - * the snd_soc_cpu_dai. It also creates the sysfs statistics device. + * the snd_soc_dai. It also creates the sysfs statistics device. */ -struct snd_soc_cpu_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info) +struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info) { - struct snd_soc_cpu_dai *fsl_ssi_dai; + struct snd_soc_dai *fsl_ssi_dai; struct fsl_ssi_private *ssi_private; int ret = 0; struct device_attribute *dev_attr; @@ -588,7 +588,7 @@ struct snd_soc_cpu_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info) return NULL; } memcpy(&ssi_private->cpu_dai, &fsl_ssi_dai_template, - sizeof(struct snd_soc_cpu_dai)); + sizeof(struct snd_soc_dai)); fsl_ssi_dai = &ssi_private->cpu_dai; dev_attr = &ssi_private->dev_attr; @@ -623,11 +623,11 @@ struct snd_soc_cpu_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info) EXPORT_SYMBOL_GPL(fsl_ssi_create_dai); /** - * fsl_ssi_destroy_dai: destroy the snd_soc_cpu_dai object + * fsl_ssi_destroy_dai: destroy the snd_soc_dai object * * This function undoes the operations of fsl_ssi_create_dai() */ -void fsl_ssi_destroy_dai(struct snd_soc_cpu_dai *fsl_ssi_dai) +void fsl_ssi_destroy_dai(struct snd_soc_dai *fsl_ssi_dai) { struct fsl_ssi_private *ssi_private = container_of(fsl_ssi_dai, struct fsl_ssi_private, cpu_dai); diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h index c5ce88e1565..83b44d700e3 100644 --- a/sound/soc/fsl/fsl_ssi.h +++ b/sound/soc/fsl/fsl_ssi.h @@ -217,8 +217,8 @@ struct fsl_ssi_info { struct device *dev; }; -struct snd_soc_cpu_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info); -void fsl_ssi_destroy_dai(struct snd_soc_cpu_dai *fsl_ssi_dai); +struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info); +void fsl_ssi_destroy_dai(struct snd_soc_dai *fsl_ssi_dai); #endif diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 8820c3f8d17..59d7e49bd66 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -96,8 +96,8 @@ static int mpc8610_hpcd_machine_probe(struct platform_device *sound_device) static int mpc8610_hpcd_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct mpc8610_hpcd_data *machine_data = rtd->socdev->dev->platform_data; int ret = 0; -- cgit From 8687eb8bded8c3c5842a85bd0c30e43fc5a3e0e0 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 7 Jul 2008 16:08:07 +0100 Subject: ALSA: asoc: omap - merge structs snd_soc_codec_dai and snd_soc_cpu_dai. This patch merges struct snd_soc_codec_dai and struct snd_soc_cpu_dai into struct snd_soc_dai for the Omap platform. Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/omap/n810.c | 4 ++-- sound/soc/omap/omap-mcbsp.c | 16 ++++++++-------- sound/soc/omap/omap-mcbsp.h | 2 +- sound/soc/omap/omap-pcm.c | 2 +- 4 files changed, 12 insertions(+), 12 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 74f4599b4d7..d1233c01398 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -86,8 +86,8 @@ static int n810_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; int err; /* Set codec DAI configuration */ diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 40d87e6d0de..00b0c9d73cd 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -103,7 +103,7 @@ static const unsigned long omap2420_mcbsp_port[][2] = {}; static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); int err = 0; @@ -116,7 +116,7 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream) static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); if (!cpu_dai->active) { @@ -128,7 +128,7 @@ static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream) static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); int err = 0; @@ -157,7 +157,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; int dma, bus_id = mcbsp_data->bus_id, id = cpu_dai->id; @@ -223,7 +223,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, * This must be called before _set_clkdiv and _set_sysclk since McBSP register * cache is initialized here */ -static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, +static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); @@ -292,7 +292,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, return 0; } -static int omap_mcbsp_dai_set_clkdiv(struct snd_soc_cpu_dai *cpu_dai, +static int omap_mcbsp_dai_set_clkdiv(struct snd_soc_dai *cpu_dai, int div_id, int div) { struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); @@ -347,7 +347,7 @@ static int omap_mcbsp_dai_set_clks_src(struct omap_mcbsp_data *mcbsp_data, return 0; } -static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, +static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { @@ -376,7 +376,7 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, return err; } -struct snd_soc_cpu_dai omap_mcbsp_dai[NUM_LINKS] = { +struct snd_soc_dai omap_mcbsp_dai[NUM_LINKS] = { { .name = "omap-mcbsp-dai", .id = 0, diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h index 9965fd4b042..ed8afb55067 100644 --- a/sound/soc/omap/omap-mcbsp.h +++ b/sound/soc/omap/omap-mcbsp.h @@ -44,6 +44,6 @@ enum omap_mcbsp_div { */ #define NUM_LINKS 1 -extern struct snd_soc_cpu_dai omap_mcbsp_dai[NUM_LINKS]; +extern struct snd_soc_dai omap_mcbsp_dai[NUM_LINKS]; #endif diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 62370202c64..e092f3d836d 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -316,7 +316,7 @@ static void omap_pcm_free_dma_buffers(struct snd_pcm *pcm) } } -int omap_pcm_new(struct snd_card *card, struct snd_soc_codec_dai *dai, +int omap_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) { int ret = 0; -- cgit From 917f93ac80a1c007d4a3ce269a3712f93a75728f Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 7 Jul 2008 16:08:11 +0100 Subject: ALSA: asoc: pxa - merge structs snd_soc_codec_dai and snd_soc_cpu_dai. This patch merges struct snd_soc_codec_dai and struct snd_soc_cpu_dai into struct snd_soc_dai for the PXA platform. Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/pxa/corgi.c | 4 ++-- sound/soc/pxa/poodle.c | 4 ++-- sound/soc/pxa/pxa2xx-ac97.c | 14 +++++++------- sound/soc/pxa/pxa2xx-ac97.h | 2 +- sound/soc/pxa/pxa2xx-i2s.c | 14 +++++++------- sound/soc/pxa/pxa2xx-i2s.h | 2 +- sound/soc/pxa/pxa2xx-pcm.c | 2 +- sound/soc/pxa/spitz.c | 4 ++-- 8 files changed, 23 insertions(+), 23 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index db18ef68b69..782afbf7ada 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -123,8 +123,8 @@ static int corgi_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; unsigned int clk = 0; int ret = 0; diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 36cbf69f5f8..ce25b6bf340 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -102,8 +102,8 @@ static int poodle_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; unsigned int clk = 0; int ret = 0; diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index cb947956ed1..b458667538c 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -283,7 +283,7 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_mic_mono_in = { #ifdef CONFIG_PM static int pxa2xx_ac97_suspend(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { GCR |= GCR_ACLINK_OFF; clk_disable(ac97_clk); @@ -291,7 +291,7 @@ static int pxa2xx_ac97_suspend(struct platform_device *pdev, } static int pxa2xx_ac97_resume(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { pxa_gpio_mode(GPIO31_SYNC_AC97_MD); pxa_gpio_mode(GPIO30_SDATA_OUT_AC97_MD); @@ -311,7 +311,7 @@ static int pxa2xx_ac97_resume(struct platform_device *pdev, #endif static int pxa2xx_ac97_probe(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { int ret; @@ -373,7 +373,7 @@ static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) cpu_dai->dma_data = &pxa2xx_ac97_pcm_stereo_out; @@ -387,7 +387,7 @@ static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) cpu_dai->dma_data = &pxa2xx_ac97_pcm_aux_mono_out; @@ -401,7 +401,7 @@ static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) return -ENODEV; @@ -419,7 +419,7 @@ static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream, * There is only 1 physical AC97 interface for pxa2xx, but it * has extra fifo's that can be used for aux DACs and ADCs. */ -struct snd_soc_cpu_dai pxa_ac97_dai[] = { +struct snd_soc_dai pxa_ac97_dai[] = { { .name = "pxa2xx-ac97", .id = 0, diff --git a/sound/soc/pxa/pxa2xx-ac97.h b/sound/soc/pxa/pxa2xx-ac97.h index b8ccfee095c..e390de8edcd 100644 --- a/sound/soc/pxa/pxa2xx-ac97.h +++ b/sound/soc/pxa/pxa2xx-ac97.h @@ -14,7 +14,7 @@ #define PXA2XX_DAI_AC97_AUX 1 #define PXA2XX_DAI_AC97_MIC 2 -extern struct snd_soc_cpu_dai pxa_ac97_dai[3]; +extern struct snd_soc_dai pxa_ac97_dai[3]; /* platform data */ extern struct snd_ac97_bus_ops pxa2xx_ac97_ops; diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 35090c2870f..9c06553b926 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -77,7 +77,7 @@ static struct pxa2xx_gpio gpio_bus[] = { static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; if (!cpu_dai->active) { SACR0 |= SACR0_RST; @@ -98,7 +98,7 @@ static int pxa_i2s_wait(void) return 0; } -static int pxa2xx_i2s_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, +static int pxa2xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { /* interface format */ @@ -124,7 +124,7 @@ static int pxa2xx_i2s_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, return 0; } -static int pxa2xx_i2s_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, +static int pxa2xx_i2s_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { if (clk_id != PXA2XX_I2S_SYSCLK) @@ -140,7 +140,7 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; pxa_gpio_mode(gpio_bus[pxa_i2s.master].rx); pxa_gpio_mode(gpio_bus[pxa_i2s.master].tx); @@ -237,7 +237,7 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream) #ifdef CONFIG_PM static int pxa2xx_i2s_suspend(struct platform_device *dev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { if (!dai->active) return 0; @@ -255,7 +255,7 @@ static int pxa2xx_i2s_suspend(struct platform_device *dev, } static int pxa2xx_i2s_resume(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { if (!dai->active) return 0; @@ -280,7 +280,7 @@ static int pxa2xx_i2s_resume(struct platform_device *pdev, SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000) -struct snd_soc_cpu_dai pxa_i2s_dai = { +struct snd_soc_dai pxa_i2s_dai = { .name = "pxa2xx-i2s", .id = 0, .type = SND_SOC_DAI_I2S, diff --git a/sound/soc/pxa/pxa2xx-i2s.h b/sound/soc/pxa/pxa2xx-i2s.h index 4435bd9f884..e2def441153 100644 --- a/sound/soc/pxa/pxa2xx-i2s.h +++ b/sound/soc/pxa/pxa2xx-i2s.h @@ -15,6 +15,6 @@ /* I2S clock */ #define PXA2XX_I2S_SYSCLK 0 -extern struct snd_soc_cpu_dai pxa_i2s_dai; +extern struct snd_soc_dai pxa_i2s_dai; #endif diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index 01ad7bf716b..2df03ee5819 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -330,7 +330,7 @@ static void pxa2xx_pcm_free_dma_buffers(struct snd_pcm *pcm) static u64 pxa2xx_pcm_dmamask = DMA_32BIT_MASK; -int pxa2xx_pcm_new(struct snd_card *card, struct snd_soc_codec_dai *dai, +int pxa2xx_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) { int ret = 0; diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index ec18163fddd..fd1abc7b08d 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -121,8 +121,8 @@ static int spitz_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; unsigned int clk = 0; int ret = 0; -- cgit From 1992a6fbd929196aebe95e0e7b04c4da66c3bfec Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 7 Jul 2008 16:08:24 +0100 Subject: ALSA: asoc: s3c24xx - merge structs snd_soc_codec_dai and snd_soc_cpu_dai. This patch merges struct snd_soc_codec_dai and struct snd_soc_cpu_dai into struct snd_soc_dai for the S3C24xx platform. Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/s3c24xx/neo1973_wm8753.c | 12 ++++++------ sound/soc/s3c24xx/s3c2412-i2s.c | 14 +++++++------- sound/soc/s3c24xx/s3c2412-i2s.h | 2 +- sound/soc/s3c24xx/s3c2443-ac97.c | 10 +++++----- sound/soc/s3c24xx/s3c24xx-ac97.h | 2 +- sound/soc/s3c24xx/s3c24xx-i2s.c | 14 +++++++------- sound/soc/s3c24xx/s3c24xx-i2s.h | 2 +- sound/soc/s3c24xx/s3c24xx-pcm.c | 2 +- 8 files changed, 29 insertions(+), 29 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index f053e85ff60..51a4ce3dbd1 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -66,8 +66,8 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + 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, bclk = 0; int ret = 0; unsigned long iis_clkrate; @@ -156,7 +156,7 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream, static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; DBG("Entered %s\n", __func__); @@ -176,7 +176,7 @@ static int neo1973_voice_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; unsigned int pcmdiv = 0; int ret = 0; unsigned long iis_clkrate; @@ -222,7 +222,7 @@ static int neo1973_voice_hw_params(struct snd_pcm_substream *substream, static int neo1973_voice_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; DBG("Entered %s\n", __func__); @@ -546,7 +546,7 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec) /* * BT Codec DAI */ -static struct snd_soc_cpu_dai bt_dai = { +static struct snd_soc_dai bt_dai = { .name = "Bluetooth", .id = 0, .type = SND_SOC_DAI_PCM, diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index c463a82dec3..ee4676ed128 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -295,7 +295,7 @@ static inline int s3c2412_snd_is_clkmaster(void) /* * Set S3C2412 I2S DAI format */ -static int s3c2412_i2s_set_fmt(struct snd_soc_cpu_dai *cpu_dai, +static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { u32 iismod; @@ -500,7 +500,7 @@ EXPORT_SYMBOL_GPL(s3c2412_iis_calc_rate); /* * Set S3C2412 Clock source */ -static int s3c2412_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, +static int s3c2412_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { u32 iismod = readl(s3c2412_i2s.regs + S3C2412_IISMOD); @@ -528,7 +528,7 @@ static int s3c2412_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, /* * Set S3C2412 Clock dividers */ -static int s3c2412_i2s_set_clkdiv(struct snd_soc_cpu_dai *cpu_dai, +static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, int div_id, int div) { struct s3c2412_i2s_info *i2s = &s3c2412_i2s; @@ -602,7 +602,7 @@ EXPORT_SYMBOL_GPL(s3c2412_get_iisclk); static int s3c2412_i2s_probe(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { DBG("Entered %s\n", __func__); @@ -648,7 +648,7 @@ static int s3c2412_i2s_probe(struct platform_device *pdev, #ifdef CONFIG_PM static int s3c2412_i2s_suspend(struct platform_device *dev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { struct s3c2412_i2s_info *i2s = &s3c2412_i2s; u32 iismod; @@ -676,7 +676,7 @@ static int s3c2412_i2s_suspend(struct platform_device *dev, } static int s3c2412_i2s_resume(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { struct s3c2412_i2s_info *i2s = &s3c2412_i2s; @@ -708,7 +708,7 @@ static int s3c2412_i2s_resume(struct platform_device *pdev, SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) -struct snd_soc_cpu_dai s3c2412_i2s_dai = { +struct snd_soc_dai s3c2412_i2s_dai = { .name = "s3c2412-i2s", .id = 0, .type = SND_SOC_DAI_I2S, diff --git a/sound/soc/s3c24xx/s3c2412-i2s.h b/sound/soc/s3c24xx/s3c2412-i2s.h index 27f48e1ffa8..aac08a25e54 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.h +++ b/sound/soc/s3c24xx/s3c2412-i2s.h @@ -24,7 +24,7 @@ extern struct clk *s3c2412_get_iisclk(void); -extern struct snd_soc_cpu_dai s3c2412_i2s_dai; +extern struct snd_soc_dai s3c2412_i2s_dai; struct s3c2412_rate_calc { unsigned int clk_div; /* for prescaler */ diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c index 533565b61b2..783349b7fed 100644 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ b/sound/soc/s3c24xx/s3c2443-ac97.c @@ -210,7 +210,7 @@ static struct s3c24xx_pcm_dma_params s3c2443_ac97_mic_mono_in = { }; static int s3c2443_ac97_probe(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { int ret; u32 ac_glbctrl; @@ -262,7 +262,7 @@ static int s3c2443_ac97_probe(struct platform_device *pdev, } static void s3c2443_ac97_remove(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { free_irq(IRQ_S3C244x_AC97, NULL); clk_disable(s3c24xx_ac97.ac97_clk); @@ -274,7 +274,7 @@ static int s3c2443_ac97_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) cpu_dai->dma_data = &s3c2443_ac97_pcm_stereo_out; @@ -316,7 +316,7 @@ static int s3c2443_ac97_hw_mic_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) return -ENODEV; @@ -352,7 +352,7 @@ static int s3c2443_ac97_mic_trigger(struct snd_pcm_substream *substream, SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) -struct snd_soc_cpu_dai s3c2443_ac97_dai[] = { +struct snd_soc_dai s3c2443_ac97_dai[] = { { .name = "s3c2443-ac97", .id = 0, diff --git a/sound/soc/s3c24xx/s3c24xx-ac97.h b/sound/soc/s3c24xx/s3c24xx-ac97.h index bf03e8ed16c..a96dcadf28b 100644 --- a/sound/soc/s3c24xx/s3c24xx-ac97.h +++ b/sound/soc/s3c24xx/s3c24xx-ac97.h @@ -26,6 +26,6 @@ #define IRQ_S3C244x_AC97 IRQ_S3C2443_AC97 #endif -extern struct snd_soc_cpu_dai s3c2443_ac97_dai[]; +extern struct snd_soc_dai s3c2443_ac97_dai[]; #endif /*S3C24XXAC97_H_*/ diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 42e96b5ff82..397524282b5 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -205,7 +205,7 @@ static inline int s3c24xx_snd_is_clkmaster(void) /* * Set S3C24xx I2S DAI format */ -static int s3c24xx_i2s_set_fmt(struct snd_soc_cpu_dai *cpu_dai, +static int s3c24xx_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { u32 iismod; @@ -313,7 +313,7 @@ exit_err: /* * Set S3C24xx Clock source */ -static int s3c24xx_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, +static int s3c24xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { u32 iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); @@ -339,7 +339,7 @@ static int s3c24xx_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, /* * Set S3C24xx Clock dividers */ -static int s3c24xx_i2s_set_clkdiv(struct snd_soc_cpu_dai *cpu_dai, +static int s3c24xx_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, int div_id, int div) { u32 reg; @@ -378,7 +378,7 @@ u32 s3c24xx_i2s_get_clockrate(void) EXPORT_SYMBOL_GPL(s3c24xx_i2s_get_clockrate); static int s3c24xx_i2s_probe(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { DBG("Entered %s\n", __func__); @@ -411,7 +411,7 @@ static int s3c24xx_i2s_probe(struct platform_device *pdev, #ifdef CONFIG_PM static int s3c24xx_i2s_suspend(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai) + struct snd_soc_dai *cpu_dai) { DBG("Entered %s\n", __func__); @@ -426,7 +426,7 @@ static int s3c24xx_i2s_suspend(struct platform_device *pdev, } static int s3c24xx_i2s_resume(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai) + struct snd_soc_dai *cpu_dai) { DBG("Entered %s\n", __func__); clk_enable(s3c24xx_i2s.iis_clk); @@ -449,7 +449,7 @@ static int s3c24xx_i2s_resume(struct platform_device *pdev, SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) -struct snd_soc_cpu_dai s3c24xx_i2s_dai = { +struct snd_soc_dai s3c24xx_i2s_dai = { .name = "s3c24xx-i2s", .id = 0, .type = SND_SOC_DAI_I2S, diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.h b/sound/soc/s3c24xx/s3c24xx-i2s.h index 537b4ecce8a..726d91cf4e1 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.h +++ b/sound/soc/s3c24xx/s3c24xx-i2s.h @@ -32,6 +32,6 @@ u32 s3c24xx_i2s_get_clockrate(void); -extern struct snd_soc_cpu_dai s3c24xx_i2s_dai; +extern struct snd_soc_dai s3c24xx_i2s_dai; #endif /*S3C24XXI2S_H_*/ diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c index ef599745159..cef79b34dc6 100644 --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c @@ -429,7 +429,7 @@ static void s3c24xx_pcm_free_dma_buffers(struct snd_pcm *pcm) static u64 s3c24xx_pcm_dmamask = DMA_32BIT_MASK; static int s3c24xx_pcm_new(struct snd_card *card, - struct snd_soc_codec_dai *dai, struct snd_pcm *pcm) + struct snd_soc_dai *dai, struct snd_pcm *pcm) { int ret = 0; -- cgit From 53640650ffe07f6201ecae0b0dca078b97327247 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 7 Jul 2008 16:08:29 +0100 Subject: ALSA: asoc: sh - merge structs snd_soc_codec_dai and snd_soc_cpu_dai. This patch merges struct snd_soc_codec_dai and struct snd_soc_cpu_dai into struct snd_soc_dai for the SuperH platform. Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/sh/dma-sh7760.c | 2 +- sound/soc/sh/hac.c | 2 +- sound/soc/sh/sh7760-ac97.c | 2 +- sound/soc/sh/ssi.c | 8 ++++---- 4 files changed, 7 insertions(+), 7 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c index 7a3ce80d672..9faa12622d0 100644 --- a/sound/soc/sh/dma-sh7760.c +++ b/sound/soc/sh/dma-sh7760.c @@ -326,7 +326,7 @@ static void camelot_pcm_free(struct snd_pcm *pcm) } static int camelot_pcm_new(struct snd_card *card, - struct snd_soc_codec_dai *dai, + struct snd_soc_dai *dai, struct snd_pcm *pcm) { /* dont use SNDRV_DMA_TYPE_DEV, since it will oops the SH kernel diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c index b7b676b3d67..df7bc345c32 100644 --- a/sound/soc/sh/hac.c +++ b/sound/soc/sh/hac.c @@ -266,7 +266,7 @@ static int hac_hw_params(struct snd_pcm_substream *substream, #define AC97_FMTS \ SNDRV_PCM_FMTBIT_S16_LE -struct snd_soc_cpu_dai sh4_hac_dai[] = { +struct snd_soc_dai sh4_hac_dai[] = { { .name = "HAC0", .id = 0, diff --git a/sound/soc/sh/sh7760-ac97.c b/sound/soc/sh/sh7760-ac97.c index 846d1b3a630..92bfaf4774a 100644 --- a/sound/soc/sh/sh7760-ac97.c +++ b/sound/soc/sh/sh7760-ac97.c @@ -20,7 +20,7 @@ #define IPSEL 0xFE400034 /* platform specific structs can be declared here */ -extern struct snd_soc_cpu_dai sh4_hac_dai[2]; +extern struct snd_soc_dai sh4_hac_dai[2]; extern struct snd_soc_platform sh7760_soc_platform; static int machine_init(struct snd_soc_codec *codec) diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c index 3388bc3d62d..55c3464163a 100644 --- a/sound/soc/sh/ssi.c +++ b/sound/soc/sh/ssi.c @@ -208,7 +208,7 @@ static int ssi_hw_params(struct snd_pcm_substream *substream, return 0; } -static int ssi_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, int clk_id, +static int ssi_set_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { struct ssi_priv *ssi = &ssi_cpu_data[cpu_dai->id]; @@ -222,7 +222,7 @@ static int ssi_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, int clk_id, * This divider is used to generate the SSI_SCK (I2S bitclock) from the * clock at the HAC_BIT_CLK ("oversampling clock") pin. */ -static int ssi_set_clkdiv(struct snd_soc_cpu_dai *dai, int did, int div) +static int ssi_set_clkdiv(struct snd_soc_dai *dai, int did, int div) { struct ssi_priv *ssi = &ssi_cpu_data[dai->id]; unsigned long ssicr; @@ -245,7 +245,7 @@ static int ssi_set_clkdiv(struct snd_soc_cpu_dai *dai, int did, int div) return 0; } -static int ssi_set_fmt(struct snd_soc_cpu_dai *dai, unsigned int fmt) +static int ssi_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct ssi_priv *ssi = &ssi_cpu_data[dai->id]; unsigned long ssicr = SSIREG(SSICR); @@ -332,7 +332,7 @@ static int ssi_set_fmt(struct snd_soc_cpu_dai *dai, unsigned int fmt) SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3LE | \ SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE) -struct snd_soc_cpu_dai sh4_ssi_dai[] = { +struct snd_soc_dai sh4_ssi_dai[] = { { .name = "SSI0", .id = 0, -- cgit From c1286b8671135006eeb66b0267fa2895561b78e4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 7 Jul 2008 19:26:03 +0100 Subject: ALSA: ASoC: Switch DAPM to use of standard DEBUG macro DAPM contains debug output controlled by a DAPM_DEBUG macro. Change this to be controlled by the standard DEBUG, dropping the custom dbg() macro as we go. Also fix the error printed when configuring an unknown pin to be an unconditionally displayed error rather than debug output. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/soc-dapm.c | 22 ++++++++++------------ 1 file changed, 10 insertions(+), 12 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 94296b5dc58..d18ebc62786 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -45,13 +45,10 @@ #include /* debug */ -#define DAPM_DEBUG 0 -#if DAPM_DEBUG +#ifdef DEBUG #define dump_dapm(codec, action) dbg_dump_dapm(codec, action) -#define dbg(format, arg...) printk(format, ## arg) #else #define dump_dapm(codec, action) -#define dbg(format, arg...) #endif /* dapm power sequences - make this per codec in the future */ @@ -233,7 +230,8 @@ static int dapm_update_bits(struct snd_soc_dapm_widget *widget) snd_soc_write(codec, widget->reg, new); pop_wait(); } - dbg("reg %x old %x new %x change %d\n", widget->reg, old, new, change); + pr_debug("reg %x old %x new %x change %d\n", widget->reg, + old, new, change); return change; } @@ -591,8 +589,8 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) /* call any power change event handlers */ if (power_change) { if (w->event) { - dbg("power %s event for %s flags %x\n", - w->power ? "on" : "off", w->name, w->event_flags); + pr_debug("power %s event for %s flags %x\n", + w->power ? "on" : "off", w->name, w->event_flags); if (power) { /* power up event */ if (w->event_flags & SND_SOC_DAPM_PRE_PMU) { @@ -634,7 +632,7 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) return ret; } -#if DAPM_DEBUG +#ifdef DEBUG static void dbg_dump_dapm(struct snd_soc_codec* codec, const char *action) { struct snd_soc_dapm_widget *w; @@ -887,13 +885,13 @@ static int snd_soc_dapm_set_pin(struct snd_soc_codec *codec, list_for_each_entry(w, &codec->dapm_widgets, list) { if (!strcmp(w->name, pin)) { - dbg("dapm: %s: pin %s\n", codec->name, pin); + pr_debug("dapm: %s: pin %s\n", codec->name, pin); w->connected = status; return 0; } } - dbg("dapm: %s: configuring unknown pin %s\n", codec->name, pin); + pr_err("dapm: %s: configuring unknown pin %s\n", codec->name, pin); return -EINVAL; } @@ -1397,8 +1395,8 @@ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, { if (!w->sname) continue; - dbg("widget %s\n %s stream %s event %d\n", w->name, w->sname, - stream, event); + pr_debug("widget %s\n %s stream %s event %d\n", + w->name, w->sname, stream, event); if (strstr(w->sname, stream)) { switch(event) { case SND_SOC_DAPM_STREAM_START: -- cgit From 3fccd8b1bf716f1314c76689a3554ea37ad282b6 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 7 Jul 2008 19:26:04 +0100 Subject: ALSA: ASoC: Dump DAPM state for non-stream changes Explicit DAPM syncs are likely to cause DAPM state updates, as are mixer and mux configuration changes, so display the DAPM status after them too. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/soc-dapm.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d18ebc62786..2c87061c2a6 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -717,8 +717,10 @@ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, path->connect = 0; /* old connection must be powered down */ } - if (found) + if (found) { dapm_power_widgets(widget->codec, SND_SOC_DAPM_STREAM_NOP); + dump_dapm(widget->codec, "mux power update"); + } return 0; } @@ -754,8 +756,10 @@ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, break; } - if (found) + if (found) { dapm_power_widgets(widget->codec, SND_SOC_DAPM_STREAM_NOP); + dump_dapm(widget->codec, "mixer power update"); + } return 0; } @@ -906,7 +910,9 @@ static int snd_soc_dapm_set_pin(struct snd_soc_codec *codec, */ int snd_soc_dapm_sync(struct snd_soc_codec *codec) { - return dapm_power_widgets(codec, SND_SOC_DAPM_STREAM_NOP); + int ret = dapm_power_widgets(codec, SND_SOC_DAPM_STREAM_NOP); + dump_dapm(codec, "sync"); + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_sync); -- cgit From 8c6529dbf881303920a415c2d14a500218661949 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Tue, 8 Jul 2008 13:19:13 +0100 Subject: ALSA: asoc: core - add Digital Audio Interface (DAI) control functions. This patch adds several functions for DAI control and config and replaces the current method of calling function pointers within the DAI struct. Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/soc-core.c | 140 ++++++++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 132 insertions(+), 8 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4d626b47b2f..83f1190293a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -434,8 +434,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) else { codec_dai->pop_wait = 0; cancel_delayed_work(&socdev->delayed_work); - if (codec_dai->dai_ops.digital_mute) - codec_dai->dai_ops.digital_mute(codec_dai, 0); + snd_soc_dai_digital_mute(codec_dai, 0); } } else { /* no delayed work - do we need to power up codec */ @@ -454,8 +453,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) SND_SOC_DAPM_STREAM_START); snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_ON); - if (codec_dai->dai_ops.digital_mute) - codec_dai->dai_ops.digital_mute(codec_dai, 0); + snd_soc_dai_digital_mute(codec_dai, 0); } else { /* codec already powered - power on widgets */ @@ -467,8 +465,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) snd_soc_dapm_stream_event(codec, codec_dai->capture.stream_name, SND_SOC_DAPM_STREAM_START); - if (codec_dai->dai_ops.digital_mute) - codec_dai->dai_ops.digital_mute(codec_dai, 0); + + snd_soc_dai_digital_mute(codec_dai, 0); } } @@ -566,8 +564,8 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) mutex_lock(&pcm_mutex); /* apply codec digital mute */ - if (!codec->active && codec_dai->dai_ops.digital_mute) - codec_dai->dai_ops.digital_mute(codec_dai, 1); + if (!codec->active) + snd_soc_dai_digital_mute(codec_dai, 1); /* free any machine hw params */ if (machine->ops && machine->ops->hw_free) @@ -1703,6 +1701,132 @@ int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8); +/** + * snd_soc_dai_set_sysclk - configure DAI system or master clock. + * @dai: DAI + * @clk_id: DAI specific clock ID + * @freq: new clock frequency in Hz + * @dir: new clock direction - input/output. + * + * Configures the DAI master (MCLK) or system (SYSCLK) clocking. + */ +int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + if (dai->dai_ops.set_sysclk) + return dai->dai_ops.set_sysclk(dai, clk_id, freq, dir); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk); + +/** + * snd_soc_dai_set_clkdiv - configure DAI clock dividers. + * @dai: DAI + * @clk_id: DAI specific clock divider ID + * @div: new clock divisor. + * + * Configures the clock dividers. This is used to derive the best DAI bit and + * frame clocks from the system or master clock. It's best to set the DAI bit + * and frame clocks as low as possible to save system power. + */ +int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, + int div_id, int div) +{ + if (dai->dai_ops.set_clkdiv) + return dai->dai_ops.set_clkdiv(dai, div_id, div); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_clkdiv); + +/** + * snd_soc_dai_set_pll - configure DAI PLL. + * @dai: DAI + * @pll_id: DAI specific PLL ID + * @freq_in: PLL input clock frequency in Hz + * @freq_out: requested PLL output clock frequency in Hz + * + * Configures and enables PLL to generate output clock based on input clock. + */ +int snd_soc_dai_set_pll(struct snd_soc_dai *dai, + int pll_id, unsigned int freq_in, unsigned int freq_out) +{ + if (dai->dai_ops.set_pll) + return dai->dai_ops.set_pll(dai, pll_id, freq_in, freq_out); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll); + +/** + * snd_soc_dai_set_fmt - configure DAI hardware audio format. + * @dai: DAI + * @clk_id: DAI specific clock ID + * @fmt: SND_SOC_DAIFMT_ format value. + * + * Configures the DAI hardware format and clocking. + */ +int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + if (dai->dai_ops.set_fmt) + return dai->dai_ops.set_fmt(dai, fmt); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt); + +/** + * snd_soc_dai_set_tdm_slot - configure DAI TDM. + * @dai: DAI + * @mask: DAI specific mask representing used slots. + * @slots: Number of slots in use. + * + * Configures a DAI for TDM operation. Both mask and slots are codec and DAI + * specific. + */ +int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int mask, int slots) +{ + if (dai->dai_ops.set_sysclk) + return dai->dai_ops.set_tdm_slot(dai, mask, slots); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot); + +/** + * snd_soc_dai_set_tristate - configure DAI system or master clock. + * @dai: DAI + * @tristate: tristate enable + * + * Tristates the DAI so that others can use it. + */ +int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + if (dai->dai_ops.set_sysclk) + return dai->dai_ops.set_tristate(dai, tristate); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate); + +/** + * snd_soc_dai_digital_mute - configure DAI system or master clock. + * @dai: DAI + * @mute: mute enable + * + * Mutes the DAI DAC. + */ +int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute) +{ + if (dai->dai_ops.digital_mute) + return dai->dai_ops.digital_mute(dai, mute); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); + static int __devinit snd_soc_init(void) { printk(KERN_INFO "ASoC version %s\n", SND_SOC_VERSION); -- cgit From 64105cfd65df74fdf82c1d053b2c9953304a94ea Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Tue, 8 Jul 2008 13:19:18 +0100 Subject: ALSA: asoc: machines - add Digital Audio Interface (DAI) control functions. This patch adds several functions for DAI control and config and replaces the current method of calling function pointers within the DAI struct within the machine drivers. Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/at32/playpaq_wm8510.c | 16 +++++------ sound/soc/at91/eti_b1_wm8731.c | 18 ++++++------ sound/soc/davinci/davinci-evm.c | 6 ++-- sound/soc/fsl/mpc8610_hpcd.c | 58 ++++++++++++++++---------------------- sound/soc/omap/n810.c | 6 ++-- sound/soc/pxa/corgi.c | 8 +++--- sound/soc/pxa/poodle.c | 8 +++--- sound/soc/pxa/spitz.c | 8 +++--- sound/soc/s3c24xx/neo1973_wm8753.c | 26 ++++++++--------- 9 files changed, 71 insertions(+), 83 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/at32/playpaq_wm8510.c b/sound/soc/at32/playpaq_wm8510.c index 18ac7d7391c..fd62f256975 100644 --- a/sound/soc/at32/playpaq_wm8510.c +++ b/sound/soc/at32/playpaq_wm8510.c @@ -210,14 +210,14 @@ static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, /* * set CPU and CODEC DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, fmt); + ret = snd_soc_dai_set_fmt(codec_dai, fmt); if (ret < 0) { pr_warning("playpaq_wm8510: " "Failed to set CODEC DAI format (%d)\n", ret); return ret; } - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, fmt); + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); if (ret < 0) { pr_warning("playpaq_wm8510: " "Failed to set CPU DAI format (%d)\n", @@ -233,14 +233,13 @@ static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, cd = playpaq_wm8510_calc_ssc_clock(params, cpu_dai); pr_debug("playpaq_wm8510: cmr_div = %d, period = %d\n", cd.cmr_div, cd.period); - ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, - AT32_SSC_CMR_DIV, cd.cmr_div); + ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_CMR_DIV, cd.cmr_div); if (ret < 0) { pr_warning("playpaq_wm8510: Failed to set CPU CMR_DIV (%d)\n", ret); return ret; } - ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, AT32_SSC_TCMR_PERIOD, + ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_TCMR_PERIOD, cd.period); if (ret < 0) { pr_warning("playpaq_wm8510: " @@ -260,7 +259,7 @@ static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, #if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE - ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8510_BCLKDIV, bclk); + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_BCLKDIV, bclk); if (ret < 0) { pr_warning ("playpaq_wm8510: Failed to set CODEC DAI BCLKDIV (%d)\n", @@ -270,7 +269,7 @@ static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, #endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ - ret = codec_dai->dai_ops.set_pll(codec_dai, 0, + ret = snd_soc_dai_set_pll(codec_dai, 0, clk_get_rate(CODEC_CLK), pll_out); if (ret < 0) { pr_warning("playpaq_wm8510: Failed to set CODEC DAI PLL (%d)\n", @@ -279,8 +278,7 @@ static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, } - ret = codec_dai->dai_ops.set_clkdiv(codec_dai, - WM8510_MCLKDIV, mclk_div); + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_MCLKDIV, mclk_div); if (ret < 0) { pr_warning("playpaq_wm8510: Failed to set CODEC MCLKDIV (%d)\n", ret); diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c index 9d4213963c2..d532de95424 100644 --- a/sound/soc/at91/eti_b1_wm8731.c +++ b/sound/soc/at91/eti_b1_wm8731.c @@ -58,13 +58,13 @@ static int eti_b1_startup(struct snd_pcm_substream *substream) int ret; /* cpu clock is the AT91 master clock sent to the SSC */ - ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, AT91_SYSCLK_MCK, + ret = snd_soc_dai_set_sysclk(cpu_dai, AT91_SYSCLK_MCK, 60000000, SND_SOC_CLOCK_IN); if (ret < 0) return ret; /* codec system clock is supplied by PCK1, set to 12MHz */ - ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8731_SYSCLK, + ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, 12000000, SND_SOC_CLOCK_IN); if (ret < 0) return ret; @@ -96,13 +96,13 @@ static int eti_b1_hw_params(struct snd_pcm_substream *substream, int cmr_div, period; /* set codec DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; /* set cpu DAI configuration */ - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; @@ -141,17 +141,17 @@ static int eti_b1_hw_params(struct snd_pcm_substream *substream, } /* set the MCK divider for BCLK */ - ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, AT91SSC_CMR_DIV, cmr_div); + ret = snd_soc_dai_set_clkdiv(cpu_dai, AT91SSC_CMR_DIV, cmr_div); if (ret < 0) return ret; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { /* set the BCLK divider for DACLRC */ - ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, + ret = snd_soc_dai_set_clkdiv(cpu_dai, AT91SSC_TCMR_PERIOD, period); } else { /* set the BCLK divider for ADCLRC */ - ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, + ret = snd_soc_dai_set_clkdiv(cpu_dai, AT91SSC_RCMR_PERIOD, period); } if (ret < 0) @@ -163,13 +163,13 @@ static int eti_b1_hw_params(struct snd_pcm_substream *substream, */ /* set codec DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) return ret; /* set cpu DAI configuration */ - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) return ret; diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 4249e6a8574..5e2c306399e 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -38,19 +38,19 @@ static int evm_hw_params(struct snd_pcm_substream *substream, int ret = 0; /* set codec DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) return ret; /* set cpu DAI configuration */ - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_CBM_CFM | + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF); if (ret < 0) return ret; /* set the codec system clock */ - ret = codec_dai->dai_ops.set_sysclk(codec_dai, 0, EVM_CODEC_CLOCK, + ret = snd_soc_dai_set_sysclk(codec_dai, 0, EVM_CODEC_CLOCK, SND_SOC_CLOCK_OUT); if (ret < 0) return ret; diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 59d7e49bd66..4bdc9d8fc90 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -103,55 +103,45 @@ static int mpc8610_hpcd_startup(struct snd_pcm_substream *substream) int ret = 0; /* Tell the CPU driver what the serial protocol is. */ - if (cpu_dai->dai_ops.set_fmt) { - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, - machine_data->dai_format); - if (ret < 0) { - dev_err(substream->pcm->card->dev, - "could not set CPU driver audio format\n"); - return ret; - } + ret = snd_soc_dai_set_fmt(cpu_dai, machine_data->dai_format); + if (ret < 0) { + dev_err(substream->pcm->card->dev, + "could not set CPU driver audio format\n"); + return ret; } /* Tell the codec driver what the serial protocol is. */ - if (codec_dai->dai_ops.set_fmt) { - ret = codec_dai->dai_ops.set_fmt(codec_dai, - machine_data->dai_format); - if (ret < 0) { - dev_err(substream->pcm->card->dev, - "could not set codec driver audio format\n"); - return ret; - } + ret = snd_soc_dai_set_fmt(codec_dai, machine_data->dai_format); + if (ret < 0) { + dev_err(substream->pcm->card->dev, + "could not set codec driver audio format\n"); + return ret; } /* * Tell the CPU driver what the clock frequency is, and whether it's a * slave or master. */ - if (cpu_dai->dai_ops.set_sysclk) { - ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, 0, - machine_data->clk_frequency, - machine_data->cpu_clk_direction); - if (ret < 0) { - dev_err(substream->pcm->card->dev, - "could not set CPU driver clock parameters\n"); - return ret; - } + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, + machine_data->clk_frequency, + machine_data->cpu_clk_direction); + if (ret < 0) { + dev_err(substream->pcm->card->dev, + "could not set CPU driver clock parameters\n"); + return ret; } /* * Tell the codec driver what the MCLK frequency is, and whether it's * a slave or master. */ - if (codec_dai->dai_ops.set_sysclk) { - ret = codec_dai->dai_ops.set_sysclk(codec_dai, 0, - machine_data->clk_frequency, - machine_data->codec_clk_direction); - if (ret < 0) { - dev_err(substream->pcm->card->dev, - "could not set codec driver clock params\n"); - return ret; - } + ret = snd_soc_dai_set_sysclk(codec_dai, 0, + machine_data->clk_frequency, + machine_data->codec_clk_direction); + if (ret < 0) { + dev_err(substream->pcm->card->dev, + "could not set codec driver clock params\n"); + return ret; } return 0; diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index d1233c01398..e53c055412c 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -91,7 +91,7 @@ static int n810_hw_params(struct snd_pcm_substream *substream, int err; /* Set codec DAI configuration */ - err = codec_dai->dai_ops.set_fmt(codec_dai, + err = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); @@ -99,7 +99,7 @@ static int n810_hw_params(struct snd_pcm_substream *substream, return err; /* Set cpu DAI configuration */ - err = cpu_dai->dai_ops.set_fmt(cpu_dai, + err = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); @@ -107,7 +107,7 @@ static int n810_hw_params(struct snd_pcm_substream *substream, return err; /* Set the codec system clock for DAC and ADC */ - err = codec_dai->dai_ops.set_sysclk(codec_dai, 0, 12000000, + err = snd_soc_dai_set_sysclk(codec_dai, 0, 12000000, SND_SOC_CLOCK_IN); return err; diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 782afbf7ada..c0294464a23 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -143,25 +143,25 @@ static int corgi_hw_params(struct snd_pcm_substream *substream, } /* set codec DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; /* set cpu DAI configuration */ - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; /* set the codec system clock for DAC and ADC */ - ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8731_SYSCLK, clk, + ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, clk, SND_SOC_CLOCK_IN); if (ret < 0) return ret; /* set the I2S system clock as input (unused) */ - ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0, + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0, SND_SOC_CLOCK_IN); if (ret < 0) return ret; diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index ce25b6bf340..65a4e9a8c39 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -122,25 +122,25 @@ static int poodle_hw_params(struct snd_pcm_substream *substream, } /* set codec DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; /* set cpu DAI configuration */ - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; /* set the codec system clock for DAC and ADC */ - ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8731_SYSCLK, clk, + ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, clk, SND_SOC_CLOCK_IN); if (ret < 0) return ret; /* set the I2S system clock as input (unused) */ - ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0, + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0, SND_SOC_CLOCK_IN); if (ret < 0) return ret; diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index fd1abc7b08d..64385797da5 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -141,25 +141,25 @@ static int spitz_hw_params(struct snd_pcm_substream *substream, } /* set codec DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; /* set cpu DAI configuration */ - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; /* set the codec system clock for DAC and ADC */ - ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8750_SYSCLK, clk, + ret = snd_soc_dai_set_sysclk(codec_dai, WM8750_SYSCLK, clk, SND_SOC_CLOCK_IN); if (ret < 0) return ret; /* set the I2S system clock as input (unused) */ - ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0, + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0, SND_SOC_CLOCK_IN); if (ret < 0) return ret; diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 51a4ce3dbd1..4d7a9aa15f1 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -108,44 +108,44 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream, } /* set codec DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) return ret; /* set cpu DAI configuration */ - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) return ret; /* set the codec system clock for DAC and ADC */ - ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8753_MCLK, pll_out, + ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_MCLK, pll_out, SND_SOC_CLOCK_IN); if (ret < 0) return ret; /* set MCLK division for sample rate */ - ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, S3C2410_IISMOD_32FS); if (ret < 0) return ret; /* set codec BCLK division for sample rate */ - ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8753_BCLKDIV, bclk); + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_BCLKDIV, bclk); if (ret < 0) return ret; /* set prescaler division for sample rate */ - ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, S3C24XX_PRESCALE(4, 4)); if (ret < 0) return ret; /* codec PLL input is PCLK/4 */ - ret = codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL1, + ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, iis_clkrate / 4, pll_out); if (ret < 0) return ret; @@ -161,7 +161,7 @@ static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream) DBG("Entered %s\n", __func__); /* disable the PLL */ - return codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL1, 0, 0); + return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0); } /* @@ -194,24 +194,24 @@ static int neo1973_voice_hw_params(struct snd_pcm_substream *substream, /* todo: gg check mode (DSP_B) against CSR datasheet */ /* set codec DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B | + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; /* set the codec system clock for DAC and ADC */ - ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8753_PCMCLK, 12288000, + ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_PCMCLK, 12288000, SND_SOC_CLOCK_IN); if (ret < 0) return ret; /* set codec PCM division for sample rate */ - ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8753_PCMDIV, pcmdiv); + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_PCMDIV, pcmdiv); if (ret < 0) return ret; /* configue and enable PLL for 12.288MHz output */ - ret = codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL2, + ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, iis_clkrate / 4, 12288000); if (ret < 0) return ret; @@ -227,7 +227,7 @@ static int neo1973_voice_hw_free(struct snd_pcm_substream *substream) DBG("Entered %s\n", __func__); /* disable the PLL */ - return codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL2, 0, 0); + return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0); } static struct snd_soc_ops neo1973_voice_ops = { -- cgit From ac8615baf279f9bcb90c8ae017321d63a94e1762 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Tue, 8 Jul 2008 13:20:39 +0100 Subject: ALSA: asoc: n810 - fix build error. This patch adds a missing " that was recently introduced (removed) in the DAI struct merge patch series. Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/omap/n810.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index e53c055412c..02cec96859b 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -63,7 +63,7 @@ static void n810_ext_control(struct snd_soc_codec *codec) if (n810_dmic_func) snd_soc_dapm_enable_pin(codec, "DMic"); else - snd_soc_dapm_disable_pin(codec, "DMic); + snd_soc_dapm_disable_pin(codec, "DMic"); snd_soc_dapm_sync(codec); } -- cgit From 6f48413dd0f28d81a20fba2e264dd1b9f205893e Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Tue, 8 Jul 2008 13:21:52 +0100 Subject: ALSA: asoc: at32 - DAI struct merge and enable_pin() change. This adds support for the recent DAI struct merge and new DAPM pin API. Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/at32/playpaq_wm8510.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/at32/playpaq_wm8510.c b/sound/soc/at32/playpaq_wm8510.c index fd62f256975..fee5f8e5895 100644 --- a/sound/soc/at32/playpaq_wm8510.c +++ b/sound/soc/at32/playpaq_wm8510.c @@ -341,15 +341,15 @@ static int playpaq_wm8510_init(struct snd_soc_codec *codec) } - /* always connected endpoints */ - snd_soc_dapm_set_endpoint(codec, "Int Mic", 1); - snd_soc_dapm_set_endpoint(codec, "Ext Spk", 1); - snd_soc_dapm_sync_endpoints(codec); + /* always connected pins */ + snd_soc_dapm_enable_pin(codec, "Int Mic"); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + snd_soc_dapm_sync(codec); /* Make CSB show PLL rate */ - codec->dai->dai_ops.set_clkdiv(codec->dai, WM8510_OPCLKDIV, + snd_soc_dai_set_clkdiv(codec->dai, WM8510_OPCLKDIV, WM8510_OPCLKDIV_1 | 4); return 0; -- cgit From 76d39d0a1947fee4c29976e507fb93188eaa2b38 Mon Sep 17 00:00:00 2001 From: Dmitry Baryshkov Date: Tue, 8 Jul 2008 19:45:20 +0400 Subject: ALSA: tosa: fix compilation with new DAPM API Signed-off-by: Dmitry Baryshkov Acked-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/pxa/tosa.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index dba7689c508..b6edb61a3a3 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -153,7 +153,7 @@ SND_SOC_DAPM_SPK("Speaker", NULL), }; /* tosa audio map */ -static const snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route audio_map[] = { /* headphone connected to HPOUTL, HPOUTR */ {"Headphone Jack", NULL, "HPOUTL"}, @@ -205,7 +205,7 @@ static int tosa_ac97_init(struct snd_soc_codec *codec) } /* add tosa specific widgets */ - snd_soc_dapm_new_controls(codec, &tosa_dapm_widgets, + snd_soc_dapm_new_controls(codec, tosa_dapm_widgets, ARRAY_SIZE(tosa_dapm_widgets)); /* set up tosa specific audio path audio_map */ -- cgit From a84534dc179d3e9f501e5a12598cc5271cd28f5b Mon Sep 17 00:00:00 2001 From: Dmitry Baryshkov Date: Tue, 8 Jul 2008 19:45:42 +0400 Subject: ALSA: ASoC: pxa2xx-ac97: fix warning due to missing argument in fuction declaration Signed-off-by: Dmitry Baryshkov Acked-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/pxa/pxa2xx-ac97.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index b458667538c..059af815ea0 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -356,7 +356,8 @@ static int pxa2xx_ac97_probe(struct platform_device *pdev, return ret; } -static void pxa2xx_ac97_remove(struct platform_device *pdev) +static void pxa2xx_ac97_remove(struct platform_device *pdev, + struct snd_soc_dai *dai) { GCR |= GCR_ACLINK_OFF; free_irq(IRQ_AC97, NULL); -- cgit From 4a161d235b68eb7234f40106560c488a1bdb3851 Mon Sep 17 00:00:00 2001 From: Manuel Lauss Date: Wed, 9 Jul 2008 16:27:56 +0200 Subject: ALSA: ASoC: Au12x0/Au1550 PSC Audio support Audio for Au12x0/Au1550 PSCs in AC97 and I2S mode, for ASoC v1 framework. - DBDMA, AC97 and I2S drivers - sample AC97 machine code (Db1200) Signed-off-by: Manuel Lauss Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/Kconfig | 1 + sound/soc/Makefile | 2 +- sound/soc/au1x/Kconfig | 32 ++++ sound/soc/au1x/Makefile | 13 ++ sound/soc/au1x/dbdma2.c | 421 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/au1x/psc-ac97.c | 387 +++++++++++++++++++++++++++++++++++++++ sound/soc/au1x/psc-i2s.c | 414 ++++++++++++++++++++++++++++++++++++++++++ sound/soc/au1x/psc.h | 53 ++++++ sound/soc/au1x/sample-ac97.c | 144 +++++++++++++++ 9 files changed, 1466 insertions(+), 1 deletion(-) create mode 100644 sound/soc/au1x/Kconfig create mode 100644 sound/soc/au1x/Makefile create mode 100644 sound/soc/au1x/dbdma2.c create mode 100644 sound/soc/au1x/psc-ac97.c create mode 100644 sound/soc/au1x/psc-i2s.c create mode 100644 sound/soc/au1x/psc.h create mode 100644 sound/soc/au1x/sample-ac97.c (limited to 'sound/soc') diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index b939e22db7b..f743530add8 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -24,6 +24,7 @@ config SND_SOC_AC97_BUS # All the supported Soc's source "sound/soc/at32/Kconfig" source "sound/soc/at91/Kconfig" +source "sound/soc/au1x/Kconfig" source "sound/soc/pxa/Kconfig" source "sound/soc/s3c24xx/Kconfig" source "sound/soc/sh/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 3645f959c26..933a66d3080 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -2,4 +2,4 @@ snd-soc-core-objs := soc-core.o soc-dapm.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o obj-$(CONFIG_SND_SOC) += codecs/ at32/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ -obj-$(CONFIG_SND_SOC) += omap/ +obj-$(CONFIG_SND_SOC) += omap/ au1x/ diff --git a/sound/soc/au1x/Kconfig b/sound/soc/au1x/Kconfig new file mode 100644 index 00000000000..410a893aa66 --- /dev/null +++ b/sound/soc/au1x/Kconfig @@ -0,0 +1,32 @@ +## +## Au1200/Au1550 PSC + DBDMA +## +config SND_SOC_AU1XPSC + tristate "SoC Audio for Au1200/Au1250/Au1550" + depends on SOC_AU1200 || SOC_AU1550 + help + This option enables support for the Programmable Serial + Controllers in AC97 and I2S mode, and the Descriptor-Based DMA + Controller (DBDMA) as found on the Au1200/Au1250/Au1550 SoC. + +config SND_SOC_AU1XPSC_I2S + tristate + +config SND_SOC_AU1XPSC_AC97 + tristate + select AC97_BUS + select SND_AC97_CODEC + select SND_SOC_AC97_BUS + + +## +## Boards +## +config SND_SOC_SAMPLE_PSC_AC97 + tristate "Sample Au12x0/Au1550 PSC AC97 sound machine" + depends on SND_SOC_AU1XPSC + select SND_SOC_AU1XPSC_AC97 + select SND_SOC_AC97_CODEC + help + This is a sample AC97 sound machine for use in Au12x0/Au1550 + based systems which have audio on PSC1 (e.g. Db1200 demoboard). diff --git a/sound/soc/au1x/Makefile b/sound/soc/au1x/Makefile new file mode 100644 index 00000000000..6c6950b8003 --- /dev/null +++ b/sound/soc/au1x/Makefile @@ -0,0 +1,13 @@ +# Au1200/Au1550 PSC audio +snd-soc-au1xpsc-dbdma-objs := dbdma2.o +snd-soc-au1xpsc-i2s-objs := psc-i2s.o +snd-soc-au1xpsc-ac97-objs := psc-ac97.o + +obj-$(CONFIG_SND_SOC_AU1XPSC) += snd-soc-au1xpsc-dbdma.o +obj-$(CONFIG_SND_SOC_AU1XPSC_I2S) += snd-soc-au1xpsc-i2s.o +obj-$(CONFIG_SND_SOC_AU1XPSC_AC97) += snd-soc-au1xpsc-ac97.o + +# Boards +snd-soc-sample-ac97-objs := sample-ac97.o + +obj-$(CONFIG_SND_SOC_SAMPLE_PSC_AC97) += snd-soc-sample-ac97.o diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c new file mode 100644 index 00000000000..1466d932880 --- /dev/null +++ b/sound/soc/au1x/dbdma2.c @@ -0,0 +1,421 @@ +/* + * Au12x0/Au1550 PSC ALSA ASoC audio support. + * + * (c) 2007-2008 MSC Vertriebsges.m.b.H., + * Manuel Lauss + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * DMA glue for Au1x-PSC audio. + * + * NOTE: all of these drivers can only work with a SINGLE instance + * of a PSC. Multiple independent audio devices are impossible + * with ASoC v1. + */ + + +#include +#include +#include +#include +#include + +#include +#include +#include +#include + +#include +#include +#include + +#include "psc.h" + +/*#define PCM_DEBUG*/ + +#define MSG(x...) printk(KERN_INFO "au1xpsc_pcm: " x) +#ifdef PCM_DEBUG +#define DBG MSG +#else +#define DBG(x...) do {} while (0) +#endif + +struct au1xpsc_audio_dmadata { + /* DDMA control data */ + unsigned int ddma_id; /* DDMA direction ID for this PSC */ + u32 ddma_chan; /* DDMA context */ + + /* PCM context (for irq handlers) */ + struct snd_pcm_substream *substream; + unsigned long curr_period; /* current segment DDMA is working on */ + unsigned long q_period; /* queue period(s) */ + unsigned long dma_area; /* address of queued DMA area */ + unsigned long dma_area_s; /* start address of DMA area */ + unsigned long pos; /* current byte position being played */ + unsigned long periods; /* number of SG segments in total */ + unsigned long period_bytes; /* size in bytes of one SG segment */ + + /* runtime data */ + int msbits; +}; + +/* instance data. There can be only one, MacLeod!!!! */ +static struct au1xpsc_audio_dmadata *au1xpsc_audio_pcmdma[2]; + +/* + * These settings are somewhat okay, at least on my machine audio plays + * almost skip-free. Especially the 64kB buffer seems to help a LOT. + */ +#define AU1XPSC_PERIOD_MIN_BYTES 1024 +#define AU1XPSC_BUFFER_MIN_BYTES 65536 + +#define AU1XPSC_PCM_FMTS \ + (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | \ + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE | \ + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE | \ + SNDRV_PCM_FMTBIT_U32_LE | SNDRV_PCM_FMTBIT_U32_BE | \ + 0) + +/* PCM hardware DMA capabilities - platform specific */ +static const struct snd_pcm_hardware au1xpsc_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED, + .formats = AU1XPSC_PCM_FMTS, + .period_bytes_min = AU1XPSC_PERIOD_MIN_BYTES, + .period_bytes_max = 4096 * 1024 - 1, + .periods_min = 2, + .periods_max = 4096, /* 2 to as-much-as-you-like */ + .buffer_bytes_max = 4096 * 1024 - 1, + .fifo_size = 16, /* fifo entries of AC97/I2S PSC */ +}; + +static void au1x_pcm_queue_tx(struct au1xpsc_audio_dmadata *cd) +{ + au1xxx_dbdma_put_source_flags(cd->ddma_chan, + (void *)phys_to_virt(cd->dma_area), + cd->period_bytes, DDMA_FLAGS_IE); + + /* update next-to-queue period */ + ++cd->q_period; + cd->dma_area += cd->period_bytes; + if (cd->q_period >= cd->periods) { + cd->q_period = 0; + cd->dma_area = cd->dma_area_s; + } +} + +static void au1x_pcm_queue_rx(struct au1xpsc_audio_dmadata *cd) +{ + au1xxx_dbdma_put_dest_flags(cd->ddma_chan, + (void *)phys_to_virt(cd->dma_area), + cd->period_bytes, DDMA_FLAGS_IE); + + /* update next-to-queue period */ + ++cd->q_period; + cd->dma_area += cd->period_bytes; + if (cd->q_period >= cd->periods) { + cd->q_period = 0; + cd->dma_area = cd->dma_area_s; + } +} + +static void au1x_pcm_dmatx_cb(int irq, void *dev_id) +{ + struct au1xpsc_audio_dmadata *cd = dev_id; + + cd->pos += cd->period_bytes; + if (++cd->curr_period >= cd->periods) { + cd->pos = 0; + cd->curr_period = 0; + } + snd_pcm_period_elapsed(cd->substream); + au1x_pcm_queue_tx(cd); +} + +static void au1x_pcm_dmarx_cb(int irq, void *dev_id) +{ + struct au1xpsc_audio_dmadata *cd = dev_id; + + cd->pos += cd->period_bytes; + if (++cd->curr_period >= cd->periods) { + cd->pos = 0; + cd->curr_period = 0; + } + snd_pcm_period_elapsed(cd->substream); + au1x_pcm_queue_rx(cd); +} + +static void au1x_pcm_dbdma_free(struct au1xpsc_audio_dmadata *pcd) +{ + if (pcd->ddma_chan) { + au1xxx_dbdma_stop(pcd->ddma_chan); + au1xxx_dbdma_reset(pcd->ddma_chan); + au1xxx_dbdma_chan_free(pcd->ddma_chan); + pcd->ddma_chan = 0; + pcd->msbits = 0; + } +} + +/* in case of missing DMA ring or changed TX-source / RX-dest bit widths, + * allocate (or reallocate) a 2-descriptor DMA ring with bit depth according + * to ALSA-supplied sample depth. This is due to limitations in the dbdma api + * (cannot adjust source/dest widths of already allocated descriptor ring). + */ +static int au1x_pcm_dbdma_realloc(struct au1xpsc_audio_dmadata *pcd, + int stype, int msbits) +{ + /* DMA only in 8/16/32 bit widths */ + if (msbits == 24) + msbits = 32; + + /* check current config: correct bits and descriptors allocated? */ + if ((pcd->ddma_chan) && (msbits == pcd->msbits)) + goto out; /* all ok! */ + + au1x_pcm_dbdma_free(pcd); + + if (stype == PCM_RX) + pcd->ddma_chan = au1xxx_dbdma_chan_alloc(pcd->ddma_id, + DSCR_CMD0_ALWAYS, + au1x_pcm_dmarx_cb, (void *)pcd); + else + pcd->ddma_chan = au1xxx_dbdma_chan_alloc(DSCR_CMD0_ALWAYS, + pcd->ddma_id, + au1x_pcm_dmatx_cb, (void *)pcd); + + if (!pcd->ddma_chan) + return -ENOMEM;; + + au1xxx_dbdma_set_devwidth(pcd->ddma_chan, msbits); + au1xxx_dbdma_ring_alloc(pcd->ddma_chan, 2); + + pcd->msbits = msbits; + + au1xxx_dbdma_stop(pcd->ddma_chan); + au1xxx_dbdma_reset(pcd->ddma_chan); + +out: + return 0; +} + +static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct au1xpsc_audio_dmadata *pcd; + int stype, ret; + + ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); + if (ret < 0) + goto out; + + stype = SUBSTREAM_TYPE(substream); + pcd = au1xpsc_audio_pcmdma[stype]; + + DBG("runtime->dma_area = 0x%08lx dma_addr_t = 0x%08lx dma_size = %d " + "runtime->min_align %d\n", + (unsigned long)runtime->dma_area, + (unsigned long)runtime->dma_addr, runtime->dma_bytes, + runtime->min_align); + + DBG("bits %d frags %d frag_bytes %d is_rx %d\n", params->msbits, + params_periods(params), params_period_bytes(params), stype); + + ret = au1x_pcm_dbdma_realloc(pcd, stype, params->msbits); + if (ret) { + MSG("DDMA channel (re)alloc failed!\n"); + goto out; + } + + pcd->substream = substream; + pcd->period_bytes = params_period_bytes(params); + pcd->periods = params_periods(params); + pcd->dma_area_s = pcd->dma_area = (unsigned long)runtime->dma_addr; + pcd->q_period = 0; + pcd->curr_period = 0; + pcd->pos = 0; + + ret = 0; +out: + return ret; +} + +static int au1xpsc_pcm_hw_free(struct snd_pcm_substream *substream) +{ + snd_pcm_lib_free_pages(substream); + return 0; +} + +static int au1xpsc_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct au1xpsc_audio_dmadata *pcd = + au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)]; + + au1xxx_dbdma_reset(pcd->ddma_chan); + + if (SUBSTREAM_TYPE(substream) == PCM_RX) { + au1x_pcm_queue_rx(pcd); + au1x_pcm_queue_rx(pcd); + } else { + au1x_pcm_queue_tx(pcd); + au1x_pcm_queue_tx(pcd); + } + + return 0; +} + +static int au1xpsc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + u32 c = au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)]->ddma_chan; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + au1xxx_dbdma_start(c); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + au1xxx_dbdma_stop(c); + break; + default: + return -EINVAL; + } + return 0; +} + +static snd_pcm_uframes_t +au1xpsc_pcm_pointer(struct snd_pcm_substream *substream) +{ + return bytes_to_frames(substream->runtime, + au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)]->pos); +} + +static int au1xpsc_pcm_open(struct snd_pcm_substream *substream) +{ + snd_soc_set_runtime_hwparams(substream, &au1xpsc_pcm_hardware); + return 0; +} + +static int au1xpsc_pcm_close(struct snd_pcm_substream *substream) +{ + au1x_pcm_dbdma_free(au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)]); + return 0; +} + +struct snd_pcm_ops au1xpsc_pcm_ops = { + .open = au1xpsc_pcm_open, + .close = au1xpsc_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = au1xpsc_pcm_hw_params, + .hw_free = au1xpsc_pcm_hw_free, + .prepare = au1xpsc_pcm_prepare, + .trigger = au1xpsc_pcm_trigger, + .pointer = au1xpsc_pcm_pointer, +}; + +static void au1xpsc_pcm_free_dma_buffers(struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static int au1xpsc_pcm_new(struct snd_card *card, + struct snd_soc_dai *dai, + struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + card->dev, AU1XPSC_BUFFER_MIN_BYTES, (4096 * 1024) - 1); + + return 0; +} + +static int au1xpsc_pcm_probe(struct platform_device *pdev) +{ + struct resource *r; + int ret; + + if (au1xpsc_audio_pcmdma[PCM_TX] || au1xpsc_audio_pcmdma[PCM_RX]) + return -EBUSY; + + /* TX DMA */ + au1xpsc_audio_pcmdma[PCM_TX] + = kzalloc(sizeof(struct au1xpsc_audio_dmadata), GFP_KERNEL); + if (!au1xpsc_audio_pcmdma[PCM_TX]) + return -ENOMEM; + + r = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!r) { + ret = -ENODEV; + goto out1; + } + (au1xpsc_audio_pcmdma[PCM_TX])->ddma_id = r->start; + + /* RX DMA */ + au1xpsc_audio_pcmdma[PCM_RX] + = kzalloc(sizeof(struct au1xpsc_audio_dmadata), GFP_KERNEL); + if (!au1xpsc_audio_pcmdma[PCM_RX]) + return -ENOMEM; + + r = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!r) { + ret = -ENODEV; + goto out2; + } + (au1xpsc_audio_pcmdma[PCM_RX])->ddma_id = r->start; + + return 0; + +out2: + kfree(au1xpsc_audio_pcmdma[PCM_RX]); + au1xpsc_audio_pcmdma[PCM_RX] = NULL; +out1: + kfree(au1xpsc_audio_pcmdma[PCM_TX]); + au1xpsc_audio_pcmdma[PCM_TX] = NULL; + return ret; +} + +static int au1xpsc_pcm_remove(struct platform_device *pdev) +{ + int i; + + for (i = 0; i < 2; i++) { + if (au1xpsc_audio_pcmdma[i]) { + au1x_pcm_dbdma_free(au1xpsc_audio_pcmdma[i]); + kfree(au1xpsc_audio_pcmdma[i]); + au1xpsc_audio_pcmdma[i] = NULL; + } + } + + return 0; +} + +/* au1xpsc audio platform */ +struct snd_soc_platform au1xpsc_soc_platform = { + .name = "au1xpsc-pcm-dbdma", + .probe = au1xpsc_pcm_probe, + .remove = au1xpsc_pcm_remove, + .pcm_ops = &au1xpsc_pcm_ops, + .pcm_new = au1xpsc_pcm_new, + .pcm_free = au1xpsc_pcm_free_dma_buffers, +}; +EXPORT_SYMBOL_GPL(au1xpsc_soc_platform); + +static int __init au1xpsc_audio_dbdma_init(void) +{ + au1xpsc_audio_pcmdma[PCM_TX] = NULL; + au1xpsc_audio_pcmdma[PCM_RX] = NULL; + return 0; +} + +static void __exit au1xpsc_audio_dbdma_exit(void) +{ +} + +module_init(au1xpsc_audio_dbdma_init); +module_exit(au1xpsc_audio_dbdma_exit); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Au12x0/Au1550 PSC Audio DMA driver"); +MODULE_AUTHOR("Manuel Lauss "); diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c new file mode 100644 index 00000000000..57facbad682 --- /dev/null +++ b/sound/soc/au1x/psc-ac97.c @@ -0,0 +1,387 @@ +/* + * Au12x0/Au1550 PSC ALSA ASoC audio support. + * + * (c) 2007-2008 MSC Vertriebsges.m.b.H., + * Manuel Lauss + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Au1xxx-PSC AC97 glue. + * + * NOTE: all of these drivers can only work with a SINGLE instance + * of a PSC. Multiple independent audio devices are impossible + * with ASoC v1. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "psc.h" + +#define AC97_DIR \ + (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) + +#define AC97_RATES \ + SNDRV_PCM_RATE_8000_48000 + +#define AC97_FMTS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3BE) + +#define AC97PCR_START(stype) \ + ((stype) == PCM_TX ? PSC_AC97PCR_TS : PSC_AC97PCR_RS) +#define AC97PCR_STOP(stype) \ + ((stype) == PCM_TX ? PSC_AC97PCR_TP : PSC_AC97PCR_RP) +#define AC97PCR_CLRFIFO(stype) \ + ((stype) == PCM_TX ? PSC_AC97PCR_TC : PSC_AC97PCR_RC) + +/* instance data. There can be only one, MacLeod!!!! */ +static struct au1xpsc_audio_data *au1xpsc_ac97_workdata; + +/* AC97 controller reads codec register */ +static unsigned short au1xpsc_ac97_read(struct snd_ac97 *ac97, + unsigned short reg) +{ + /* FIXME */ + struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; + unsigned short data, tmo; + + au_writel(PSC_AC97CDC_RD | PSC_AC97CDC_INDX(reg), AC97_CDC(pscdata)); + au_sync(); + + tmo = 1000; + while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) && --tmo) + udelay(2); + + if (!tmo) + data = 0xffff; + else + data = au_readl(AC97_CDC(pscdata)) & 0xffff; + + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); + au_sync(); + + return data; +} + +/* AC97 controller writes to codec register */ +static void au1xpsc_ac97_write(struct snd_ac97 *ac97, unsigned short reg, + unsigned short val) +{ + /* FIXME */ + struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; + unsigned int tmo; + + au_writel(PSC_AC97CDC_INDX(reg) | (val & 0xffff), AC97_CDC(pscdata)); + au_sync(); + tmo = 1000; + while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) && --tmo) + au_sync(); + + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); + au_sync(); +} + +/* AC97 controller asserts a warm reset */ +static void au1xpsc_ac97_warm_reset(struct snd_ac97 *ac97) +{ + /* FIXME */ + struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; + + au_writel(PSC_AC97RST_SNC, AC97_RST(pscdata)); + au_sync(); + msleep(10); + au_writel(0, AC97_RST(pscdata)); + au_sync(); +} + +static void au1xpsc_ac97_cold_reset(struct snd_ac97 *ac97) +{ + /* FIXME */ + struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; + int i; + + /* disable PSC during cold reset */ + au_writel(0, AC97_CFG(au1xpsc_ac97_workdata)); + au_sync(); + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(pscdata)); + au_sync(); + + /* issue cold reset */ + au_writel(PSC_AC97RST_RST, AC97_RST(pscdata)); + au_sync(); + msleep(500); + au_writel(0, AC97_RST(pscdata)); + au_sync(); + + /* enable PSC */ + au_writel(PSC_CTRL_ENABLE, PSC_CTRL(pscdata)); + au_sync(); + + /* wait for PSC to indicate it's ready */ + i = 100000; + while (!((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_SR)) && (--i)) + au_sync(); + + if (i == 0) { + printk(KERN_ERR "au1xpsc-ac97: PSC not ready!\n"); + return; + } + + /* enable the ac97 function */ + au_writel(pscdata->cfg | PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata)); + au_sync(); + + /* wait for AC97 core to become ready */ + i = 100000; + while (!((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)) && (--i)) + au_sync(); + if (i == 0) + printk(KERN_ERR "au1xpsc-ac97: AC97 ctrl not ready\n"); +} + +/* AC97 controller operations */ +struct snd_ac97_bus_ops soc_ac97_ops = { + .read = au1xpsc_ac97_read, + .write = au1xpsc_ac97_write, + .reset = au1xpsc_ac97_cold_reset, + .warm_reset = au1xpsc_ac97_warm_reset, +}; +EXPORT_SYMBOL_GPL(soc_ac97_ops); + +static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + /* FIXME */ + struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; + unsigned long r, stat; + int chans, stype = SUBSTREAM_TYPE(substream); + + chans = params_channels(params); + + r = au_readl(AC97_CFG(pscdata)); + stat = au_readl(AC97_STAT(pscdata)); + + /* already active? */ + if (stat & (PSC_AC97STAT_TB | PSC_AC97STAT_RB)) { + /* reject parameters not currently set up */ + if ((PSC_AC97CFG_GET_LEN(r) != params->msbits) || + (pscdata->rate != params_rate(params))) + return -EINVAL; + } else { + /* disable AC97 device controller first */ + au_writel(r & ~PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata)); + au_sync(); + + /* set sample bitdepth: REG[24:21]=(BITS-2)/2 */ + r &= ~PSC_AC97CFG_LEN_MASK; + r |= PSC_AC97CFG_SET_LEN(params->msbits); + + /* channels: enable slots for front L/R channel */ + if (stype == PCM_TX) { + r &= ~PSC_AC97CFG_TXSLOT_MASK; + r |= PSC_AC97CFG_TXSLOT_ENA(3); + r |= PSC_AC97CFG_TXSLOT_ENA(4); + } else { + r &= ~PSC_AC97CFG_RXSLOT_MASK; + r |= PSC_AC97CFG_RXSLOT_ENA(3); + r |= PSC_AC97CFG_RXSLOT_ENA(4); + } + + /* finally enable the AC97 controller again */ + au_writel(r | PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata)); + au_sync(); + + pscdata->cfg = r; + pscdata->rate = params_rate(params); + } + + return 0; +} + +static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + /* FIXME */ + struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; + int ret, stype = SUBSTREAM_TYPE(substream); + + ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + au_writel(AC97PCR_START(stype), AC97_PCR(pscdata)); + au_sync(); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + au_writel(AC97PCR_STOP(stype), AC97_PCR(pscdata)); + au_sync(); + break; + default: + ret = -EINVAL; + } + return ret; +} + +static int au1xpsc_ac97_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + int ret; + struct resource *r; + unsigned long sel; + + if (au1xpsc_ac97_workdata) + return -EBUSY; + + au1xpsc_ac97_workdata = + kzalloc(sizeof(struct au1xpsc_audio_data), GFP_KERNEL); + if (!au1xpsc_ac97_workdata) + return -ENOMEM; + + r = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!r) { + ret = -ENODEV; + goto out0; + } + + ret = -EBUSY; + au1xpsc_ac97_workdata->ioarea = + request_mem_region(r->start, r->end - r->start + 1, + "au1xpsc_ac97"); + if (!au1xpsc_ac97_workdata->ioarea) + goto out0; + + au1xpsc_ac97_workdata->mmio = ioremap(r->start, 0xffff); + if (!au1xpsc_ac97_workdata->mmio) + goto out1; + + /* configuration: max dma trigger threshold, enable ac97 */ + au1xpsc_ac97_workdata->cfg = PSC_AC97CFG_RT_FIFO8 | + PSC_AC97CFG_TT_FIFO8 | + PSC_AC97CFG_DE_ENABLE; + + /* preserve PSC clock source set up by platform (dev.platform_data + * is already occupied by soc layer) + */ + sel = au_readl(PSC_SEL(au1xpsc_ac97_workdata)) & PSC_SEL_CLK_MASK; + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_ac97_workdata)); + au_sync(); + au_writel(0, PSC_SEL(au1xpsc_ac97_workdata)); + au_sync(); + au_writel(PSC_SEL_PS_AC97MODE | sel, PSC_SEL(au1xpsc_ac97_workdata)); + au_sync(); + /* next up: cold reset. Dont check for PSC-ready now since + * there may not be any codec clock yet. + */ + + return 0; + +out1: + release_resource(au1xpsc_ac97_workdata->ioarea); + kfree(au1xpsc_ac97_workdata->ioarea); +out0: + kfree(au1xpsc_ac97_workdata); + au1xpsc_ac97_workdata = NULL; + return ret; +} + +static void au1xpsc_ac97_remove(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + /* disable PSC completely */ + au_writel(0, AC97_CFG(au1xpsc_ac97_workdata)); + au_sync(); + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_ac97_workdata)); + au_sync(); + + iounmap(au1xpsc_ac97_workdata->mmio); + release_resource(au1xpsc_ac97_workdata->ioarea); + kfree(au1xpsc_ac97_workdata->ioarea); + kfree(au1xpsc_ac97_workdata); + au1xpsc_ac97_workdata = NULL; +} + +static int au1xpsc_ac97_suspend(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + /* save interesting registers and disable PSC */ + au1xpsc_ac97_workdata->pm[0] = + au_readl(PSC_SEL(au1xpsc_ac97_workdata)); + + au_writel(0, AC97_CFG(au1xpsc_ac97_workdata)); + au_sync(); + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_ac97_workdata)); + au_sync(); + + return 0; +} + +static int au1xpsc_ac97_resume(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + /* restore PSC clock config */ + au_writel(au1xpsc_ac97_workdata->pm[0] | PSC_SEL_PS_AC97MODE, + PSC_SEL(au1xpsc_ac97_workdata)); + au_sync(); + + /* after this point the ac97 core will cold-reset the codec. + * During cold-reset the PSC is reinitialized and the last + * configuration set up in hw_params() is restored. + */ + return 0; +} + +struct snd_soc_dai au1xpsc_ac97_dai = { + .name = "au1xpsc_ac97", + .type = SND_SOC_DAI_AC97, + .probe = au1xpsc_ac97_probe, + .remove = au1xpsc_ac97_remove, + .suspend = au1xpsc_ac97_suspend, + .resume = au1xpsc_ac97_resume, + .playback = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .capture = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .ops = { + .trigger = au1xpsc_ac97_trigger, + .hw_params = au1xpsc_ac97_hw_params, + }, +}; +EXPORT_SYMBOL_GPL(au1xpsc_ac97_dai); + +static int __init au1xpsc_ac97_init(void) +{ + au1xpsc_ac97_workdata = NULL; + return 0; +} + +static void __exit au1xpsc_ac97_exit(void) +{ +} + +module_init(au1xpsc_ac97_init); +module_exit(au1xpsc_ac97_exit); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Au12x0/Au1550 PSC AC97 ALSA ASoC audio driver"); +MODULE_AUTHOR("Manuel Lauss "); diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c new file mode 100644 index 00000000000..ba4b5c199f2 --- /dev/null +++ b/sound/soc/au1x/psc-i2s.c @@ -0,0 +1,414 @@ +/* + * Au12x0/Au1550 PSC ALSA ASoC audio support. + * + * (c) 2007-2008 MSC Vertriebsges.m.b.H., + * Manuel Lauss + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Au1xxx-PSC I2S glue. + * + * NOTE: all of these drivers can only work with a SINGLE instance + * of a PSC. Multiple independent audio devices are impossible + * with ASoC v1. + * NOTE: so far only PSC slave mode (bit- and frameclock) is supported. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "psc.h" + +/* supported I2S DAI hardware formats */ +#define AU1XPSC_I2S_DAIFMT \ + (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | \ + SND_SOC_DAIFMT_NB_NF) + +/* supported I2S direction */ +#define AU1XPSC_I2S_DIR \ + (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) + +#define AU1XPSC_I2S_RATES \ + SNDRV_PCM_RATE_8000_192000 + +#define AU1XPSC_I2S_FMTS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) + +#define I2SSTAT_BUSY(stype) \ + ((stype) == PCM_TX ? PSC_I2SSTAT_TB : PSC_I2SSTAT_RB) +#define I2SPCR_START(stype) \ + ((stype) == PCM_TX ? PSC_I2SPCR_TS : PSC_I2SPCR_RS) +#define I2SPCR_STOP(stype) \ + ((stype) == PCM_TX ? PSC_I2SPCR_TP : PSC_I2SPCR_RP) +#define I2SPCR_CLRFIFO(stype) \ + ((stype) == PCM_TX ? PSC_I2SPCR_TC : PSC_I2SPCR_RC) + + +/* instance data. There can be only one, MacLeod!!!! */ +static struct au1xpsc_audio_data *au1xpsc_i2s_workdata; + +static int au1xpsc_i2s_set_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata; + unsigned long ct; + int ret; + + ret = -EINVAL; + + ct = pscdata->cfg; + + ct &= ~(PSC_I2SCFG_XM | PSC_I2SCFG_MLJ); /* left-justified */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ct |= PSC_I2SCFG_XM; /* enable I2S mode */ + break; + case SND_SOC_DAIFMT_MSB: + break; + case SND_SOC_DAIFMT_LSB: + ct |= PSC_I2SCFG_MLJ; /* LSB (right-) justified */ + break; + default: + goto out; + } + + ct &= ~(PSC_I2SCFG_BI | PSC_I2SCFG_WI); /* IB-IF */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + ct |= PSC_I2SCFG_BI | PSC_I2SCFG_WI; + break; + case SND_SOC_DAIFMT_NB_IF: + ct |= PSC_I2SCFG_BI; + break; + case SND_SOC_DAIFMT_IB_NF: + ct |= PSC_I2SCFG_WI; + break; + case SND_SOC_DAIFMT_IB_IF: + break; + default: + goto out; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: /* CODEC master */ + ct |= PSC_I2SCFG_MS; /* PSC I2S slave mode */ + break; + case SND_SOC_DAIFMT_CBS_CFS: /* CODEC slave */ + ct &= ~PSC_I2SCFG_MS; /* PSC I2S Master mode */ + break; + default: + goto out; + } + + pscdata->cfg = ct; + ret = 0; +out: + return ret; +} + +static int au1xpsc_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata; + + int cfgbits; + unsigned long stat; + + /* check if the PSC is already streaming data */ + stat = au_readl(I2S_STAT(pscdata)); + if (stat & (PSC_I2SSTAT_TB | PSC_I2SSTAT_RB)) { + /* reject parameters not currently set up in hardware */ + cfgbits = au_readl(I2S_CFG(pscdata)); + if ((PSC_I2SCFG_GET_LEN(cfgbits) != params->msbits) || + (params_rate(params) != pscdata->rate)) + return -EINVAL; + } else { + /* set sample bitdepth */ + pscdata->cfg &= ~(0x1f << 4); + pscdata->cfg |= PSC_I2SCFG_SET_LEN(params->msbits); + /* remember current rate for other stream */ + pscdata->rate = params_rate(params); + } + return 0; +} + +/* Configure PSC late: on my devel systems the codec is I2S master and + * supplies the i2sbitclock __AND__ i2sMclk (!) to the PSC unit. ASoC + * uses aggressive PM and switches the codec off when it is not in use + * which also means the PSC unit doesn't get any clocks and is therefore + * dead. That's why this chunk here gets called from the trigger callback + * because I can be reasonably certain the codec is driving the clocks. + */ +static int au1xpsc_i2s_configure(struct au1xpsc_audio_data *pscdata) +{ + unsigned long tmo; + + /* bring PSC out of sleep, and configure I2S unit */ + au_writel(PSC_CTRL_ENABLE, PSC_CTRL(pscdata)); + au_sync(); + + tmo = 1000000; + while (!(au_readl(I2S_STAT(pscdata)) & PSC_I2SSTAT_SR) && tmo) + tmo--; + + if (!tmo) + goto psc_err; + + au_writel(0, I2S_CFG(pscdata)); + au_sync(); + au_writel(pscdata->cfg | PSC_I2SCFG_DE_ENABLE, I2S_CFG(pscdata)); + au_sync(); + + /* wait for I2S controller to become ready */ + tmo = 1000000; + while (!(au_readl(I2S_STAT(pscdata)) & PSC_I2SSTAT_DR) && tmo) + tmo--; + + if (tmo) + return 0; + +psc_err: + au_writel(0, I2S_CFG(pscdata)); + au_writel(PSC_CTRL_SUSPEND, PSC_CTRL(pscdata)); + au_sync(); + return -ETIMEDOUT; +} + +static int au1xpsc_i2s_start(struct au1xpsc_audio_data *pscdata, int stype) +{ + unsigned long tmo, stat; + int ret; + + ret = 0; + + /* if both TX and RX are idle, configure the PSC */ + stat = au_readl(I2S_STAT(pscdata)); + if (!(stat & (PSC_I2SSTAT_TB | PSC_I2SSTAT_RB))) { + ret = au1xpsc_i2s_configure(pscdata); + if (ret) + goto out; + } + + au_writel(I2SPCR_CLRFIFO(stype), I2S_PCR(pscdata)); + au_sync(); + au_writel(I2SPCR_START(stype), I2S_PCR(pscdata)); + au_sync(); + + /* wait for start confirmation */ + tmo = 1000000; + while (!(au_readl(I2S_STAT(pscdata)) & I2SSTAT_BUSY(stype)) && tmo) + tmo--; + + if (!tmo) { + au_writel(I2SPCR_STOP(stype), I2S_PCR(pscdata)); + au_sync(); + ret = -ETIMEDOUT; + } +out: + return ret; +} + +static int au1xpsc_i2s_stop(struct au1xpsc_audio_data *pscdata, int stype) +{ + unsigned long tmo, stat; + + au_writel(I2SPCR_STOP(stype), I2S_PCR(pscdata)); + au_sync(); + + /* wait for stop confirmation */ + tmo = 1000000; + while ((au_readl(I2S_STAT(pscdata)) & I2SSTAT_BUSY(stype)) && tmo) + tmo--; + + /* if both TX and RX are idle, disable PSC */ + stat = au_readl(I2S_STAT(pscdata)); + if (!(stat & (PSC_I2SSTAT_RB | PSC_I2SSTAT_RB))) { + au_writel(0, I2S_CFG(pscdata)); + au_sync(); + au_writel(PSC_CTRL_SUSPEND, PSC_CTRL(pscdata)); + au_sync(); + } + return 0; +} + +static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata; + int ret, stype = SUBSTREAM_TYPE(substream); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + ret = au1xpsc_i2s_start(pscdata, stype); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + ret = au1xpsc_i2s_stop(pscdata, stype); + break; + default: + ret = -EINVAL; + } + return ret; +} + +static int au1xpsc_i2s_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct resource *r; + unsigned long sel; + int ret; + + if (au1xpsc_i2s_workdata) + return -EBUSY; + + au1xpsc_i2s_workdata = + kzalloc(sizeof(struct au1xpsc_audio_data), GFP_KERNEL); + if (!au1xpsc_i2s_workdata) + return -ENOMEM; + + r = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!r) { + ret = -ENODEV; + goto out0; + } + + ret = -EBUSY; + au1xpsc_i2s_workdata->ioarea = + request_mem_region(r->start, r->end - r->start + 1, + "au1xpsc_i2s"); + if (!au1xpsc_i2s_workdata->ioarea) + goto out0; + + au1xpsc_i2s_workdata->mmio = ioremap(r->start, 0xffff); + if (!au1xpsc_i2s_workdata->mmio) + goto out1; + + /* preserve PSC clock source set up by platform (dev.platform_data + * is already occupied by soc layer) + */ + sel = au_readl(PSC_SEL(au1xpsc_i2s_workdata)) & PSC_SEL_CLK_MASK; + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata)); + au_sync(); + au_writel(PSC_SEL_PS_I2SMODE | sel, PSC_SEL(au1xpsc_i2s_workdata)); + au_writel(0, I2S_CFG(au1xpsc_i2s_workdata)); + au_sync(); + + /* preconfigure: set max rx/tx fifo depths */ + au1xpsc_i2s_workdata->cfg |= + PSC_I2SCFG_RT_FIFO8 | PSC_I2SCFG_TT_FIFO8; + + /* don't wait for I2S core to become ready now; clocks may not + * be running yet; depending on clock input for PSC a wait might + * time out. + */ + + return 0; + +out1: + release_resource(au1xpsc_i2s_workdata->ioarea); + kfree(au1xpsc_i2s_workdata->ioarea); +out0: + kfree(au1xpsc_i2s_workdata); + au1xpsc_i2s_workdata = NULL; + return ret; +} + +static void au1xpsc_i2s_remove(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + au_writel(0, I2S_CFG(au1xpsc_i2s_workdata)); + au_sync(); + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata)); + au_sync(); + + iounmap(au1xpsc_i2s_workdata->mmio); + release_resource(au1xpsc_i2s_workdata->ioarea); + kfree(au1xpsc_i2s_workdata->ioarea); + kfree(au1xpsc_i2s_workdata); + au1xpsc_i2s_workdata = NULL; +} + +static int au1xpsc_i2s_suspend(struct platform_device *pdev, + struct snd_soc_dai *cpu_dai) +{ + /* save interesting register and disable PSC */ + au1xpsc_i2s_workdata->pm[0] = + au_readl(PSC_SEL(au1xpsc_i2s_workdata)); + + au_writel(0, I2S_CFG(au1xpsc_i2s_workdata)); + au_sync(); + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata)); + au_sync(); + + return 0; +} + +static int au1xpsc_i2s_resume(struct platform_device *pdev, + struct snd_soc_dai *cpu_dai) +{ + /* select I2S mode and PSC clock */ + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata)); + au_sync(); + au_writel(0, PSC_SEL(au1xpsc_i2s_workdata)); + au_sync(); + au_writel(au1xpsc_i2s_workdata->pm[0], + PSC_SEL(au1xpsc_i2s_workdata)); + au_sync(); + + return 0; +} + +struct snd_soc_dai au1xpsc_i2s_dai = { + .name = "au1xpsc_i2s", + .type = SND_SOC_DAI_I2S, + .probe = au1xpsc_i2s_probe, + .remove = au1xpsc_i2s_remove, + .suspend = au1xpsc_i2s_suspend, + .resume = au1xpsc_i2s_resume, + .playback = { + .rates = AU1XPSC_I2S_RATES, + .formats = AU1XPSC_I2S_FMTS, + .channels_min = 2, + .channels_max = 8, /* 2 without external help */ + }, + .capture = { + .rates = AU1XPSC_I2S_RATES, + .formats = AU1XPSC_I2S_FMTS, + .channels_min = 2, + .channels_max = 8, /* 2 without external help */ + }, + .ops = { + .trigger = au1xpsc_i2s_trigger, + .hw_params = au1xpsc_i2s_hw_params, + }, + .dai_ops = { + .set_fmt = au1xpsc_i2s_set_fmt, + }, +}; +EXPORT_SYMBOL(au1xpsc_i2s_dai); + +static int __init au1xpsc_i2s_init(void) +{ + au1xpsc_i2s_workdata = NULL; + return 0; +} + +static void __exit au1xpsc_i2s_exit(void) +{ +} + +module_init(au1xpsc_i2s_init); +module_exit(au1xpsc_i2s_exit); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Au12x0/Au1550 PSC I2S ALSA ASoC audio driver"); +MODULE_AUTHOR("Manuel Lauss "); diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h new file mode 100644 index 00000000000..8fdb1a04a07 --- /dev/null +++ b/sound/soc/au1x/psc.h @@ -0,0 +1,53 @@ +/* + * Au12x0/Au1550 PSC ALSA ASoC audio support. + * + * (c) 2007-2008 MSC Vertriebsges.m.b.H., + * Manuel Lauss + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * NOTE: all of these drivers can only work with a SINGLE instance + * of a PSC. Multiple independent audio devices are impossible + * with ASoC v1. + */ + +#ifndef _AU1X_PCM_H +#define _AU1X_PCM_H + +extern struct snd_soc_dai au1xpsc_ac97_dai; +extern struct snd_soc_dai au1xpsc_i2s_dai; +extern struct snd_soc_platform au1xpsc_soc_platform; +extern struct snd_ac97_bus_ops soc_ac97_ops; + +struct au1xpsc_audio_data { + void __iomem *mmio; + + unsigned long cfg; + unsigned long rate; + + unsigned long pm[2]; + struct resource *ioarea; +}; + +#define PCM_TX 0 +#define PCM_RX 1 + +#define SUBSTREAM_TYPE(substream) \ + ((substream)->stream == SNDRV_PCM_STREAM_PLAYBACK ? PCM_TX : PCM_RX) + +/* easy access macros */ +#define PSC_CTRL(x) ((unsigned long)((x)->mmio) + PSC_CTRL_OFFSET) +#define PSC_SEL(x) ((unsigned long)((x)->mmio) + PSC_SEL_OFFSET) +#define I2S_STAT(x) ((unsigned long)((x)->mmio) + PSC_I2SSTAT_OFFSET) +#define I2S_CFG(x) ((unsigned long)((x)->mmio) + PSC_I2SCFG_OFFSET) +#define I2S_PCR(x) ((unsigned long)((x)->mmio) + PSC_I2SPCR_OFFSET) +#define AC97_CFG(x) ((unsigned long)((x)->mmio) + PSC_AC97CFG_OFFSET) +#define AC97_CDC(x) ((unsigned long)((x)->mmio) + PSC_AC97CDC_OFFSET) +#define AC97_EVNT(x) ((unsigned long)((x)->mmio) + PSC_AC97EVNT_OFFSET) +#define AC97_PCR(x) ((unsigned long)((x)->mmio) + PSC_AC97PCR_OFFSET) +#define AC97_RST(x) ((unsigned long)((x)->mmio) + PSC_AC97RST_OFFSET) +#define AC97_STAT(x) ((unsigned long)((x)->mmio) + PSC_AC97STAT_OFFSET) + +#endif diff --git a/sound/soc/au1x/sample-ac97.c b/sound/soc/au1x/sample-ac97.c new file mode 100644 index 00000000000..f75ae7f62c3 --- /dev/null +++ b/sound/soc/au1x/sample-ac97.c @@ -0,0 +1,144 @@ +/* + * Sample Au12x0/Au1550 PSC AC97 sound machine. + * + * Copyright (c) 2007-2008 Manuel Lauss + * + * This program is free software; you can redistribute it and/or modify + * it under the terms outlined in the file COPYING at the root of this + * source archive. + * + * This is a very generic AC97 sound machine driver for boards which + * have (AC97) audio at PSC1 (e.g. DB1200 demoboards). + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "../codecs/ac97.h" +#include "psc.h" + +static int au1xpsc_sample_ac97_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_sync(codec); + return 0; +} + +static struct snd_soc_dai_link au1xpsc_sample_ac97_dai = { + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &au1xpsc_ac97_dai, /* see psc-ac97.c */ + .codec_dai = &ac97_dai, /* see codecs/ac97.c */ + .init = au1xpsc_sample_ac97_init, + .ops = NULL, +}; + +static struct snd_soc_machine au1xpsc_sample_ac97_machine = { + .name = "Au1xxx PSC AC97 Audio", + .dai_link = &au1xpsc_sample_ac97_dai, + .num_links = 1, +}; + +static struct snd_soc_device au1xpsc_sample_ac97_devdata = { + .machine = &au1xpsc_sample_ac97_machine, + .platform = &au1xpsc_soc_platform, /* see dbdma2.c */ + .codec_dev = &soc_codec_dev_ac97, +}; + +static struct resource au1xpsc_psc1_res[] = { + [0] = { + .start = CPHYSADDR(PSC1_BASE_ADDR), + .end = CPHYSADDR(PSC1_BASE_ADDR) + 0x000fffff, + .flags = IORESOURCE_MEM, + }, + [1] = { +#ifdef CONFIG_SOC_AU1200 + .start = AU1200_PSC1_INT, + .end = AU1200_PSC1_INT, +#elif defined(CONFIG_SOC_AU1550) + .start = AU1550_PSC1_INT, + .end = AU1550_PSC1_INT, +#endif + .flags = IORESOURCE_IRQ, + }, + [2] = { + .start = DSCR_CMD0_PSC1_TX, + .end = DSCR_CMD0_PSC1_TX, + .flags = IORESOURCE_DMA, + }, + [3] = { + .start = DSCR_CMD0_PSC1_RX, + .end = DSCR_CMD0_PSC1_RX, + .flags = IORESOURCE_DMA, + }, +}; + +static struct platform_device *au1xpsc_sample_ac97_dev; + +static int __init au1xpsc_sample_ac97_load(void) +{ + int ret; + +#ifdef CONFIG_SOC_AU1200 + unsigned long io; + + /* modify sys_pinfunc for AC97 on PSC1 */ + io = au_readl(SYS_PINFUNC); + io |= SYS_PINFUNC_P1C; + io &= ~(SYS_PINFUNC_P1A | SYS_PINFUNC_P1B); + au_writel(io, SYS_PINFUNC); + au_sync(); +#endif + + ret = -ENOMEM; + + /* setup PSC clock source for AC97 part: external clock provided + * by codec. The psc-ac97.c driver depends on this setting! + */ + au_writel(PSC_SEL_CLK_SERCLK, PSC1_BASE_ADDR + PSC_SEL_OFFSET); + au_sync(); + + au1xpsc_sample_ac97_dev = platform_device_alloc("soc-audio", -1); + if (!au1xpsc_sample_ac97_dev) + goto out; + + au1xpsc_sample_ac97_dev->resource = + kmemdup(au1xpsc_psc1_res, sizeof(struct resource) * + ARRAY_SIZE(au1xpsc_psc1_res), GFP_KERNEL); + au1xpsc_sample_ac97_dev->num_resources = ARRAY_SIZE(au1xpsc_psc1_res); + au1xpsc_sample_ac97_dev->id = 1; + + platform_set_drvdata(au1xpsc_sample_ac97_dev, + &au1xpsc_sample_ac97_devdata); + au1xpsc_sample_ac97_devdata.dev = &au1xpsc_sample_ac97_dev->dev; + ret = platform_device_add(au1xpsc_sample_ac97_dev); + + if (ret) { + platform_device_put(au1xpsc_sample_ac97_dev); + au1xpsc_sample_ac97_dev = NULL; + } + +out: + return ret; +} + +static void __exit au1xpsc_sample_ac97_exit(void) +{ + platform_device_unregister(au1xpsc_sample_ac97_dev); +} + +module_init(au1xpsc_sample_ac97_load); +module_exit(au1xpsc_sample_ac97_exit); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Au1xxx PSC sample AC97 machine"); +MODULE_AUTHOR("Manuel Lauss "); -- cgit From 613526677a74c2b3d1b1696ea7334b2cf35155b3 Mon Sep 17 00:00:00 2001 From: sedji gaouaou Date: Thu, 10 Jul 2008 10:15:35 +0100 Subject: [ARM] 5130/4: Support for the at91sam9g20 Support for the at91sam9g20 : Atmel 400Mhz ARM 926ej-s SOC. AT91sam9g20 is an evolution of the at91sam9260 with a faster clock speed. We created a new board for this device but based the chip support directly on 9260 files with little updates. Here is the chip page on Atmel wabsite: http://atmel.com/dyn/products/product_card.asp?part_id=4337 Signed-off-by: Sedji Gaouaou Signed-off-by: Justin Waters Acked-by: Andrew Victor Signed-off-by: Russell King --- sound/soc/at91/at91-ssc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/at91/at91-ssc.c b/sound/soc/at91/at91-ssc.c index bc35d00a38f..1a426095014 100644 --- a/sound/soc/at91/at91-ssc.c +++ b/sound/soc/at91/at91-ssc.c @@ -41,7 +41,7 @@ #define DBG(x...) #endif -#if defined(CONFIG_ARCH_AT91SAM9260) +#if defined(CONFIG_ARCH_AT91SAM9260) || defined(CONFIG_ARCH_AT91SAM9G20) #define NUM_SSC_DEVICES 1 #else #define NUM_SSC_DEVICES 3 -- cgit From 1e066322c26562621811effb1eb14097bc67a9ee Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 11 Jul 2008 14:05:17 +0100 Subject: ALSA: asoc: kbuild - only show menus for the current ASoC CPU platform. We don't want to see ASoC platform menus for other non selected architectures in our config. Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/fsl/Kconfig | 4 ---- sound/soc/omap/Kconfig | 4 ---- 2 files changed, 8 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 19802e27df4..3368ace6097 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -1,5 +1,3 @@ -menu "ALSA SoC audio for Freescale SOCs" - config SND_SOC_MPC8610 bool "ALSA SoC support for the MPC8610 SOC" depends on MPC8610_HPCD @@ -16,5 +14,3 @@ config SND_SOC_MPC8610_HPCD default y if MPC8610_HPCD help Say Y if you want to enable audio on the Freescale MPC8610 HPCD. - -endmenu diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 0230d83e8e5..aea27e70043 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -1,5 +1,3 @@ -menu "SoC Audio for the Texas Instruments OMAP" - config SND_OMAP_SOC tristate "SoC Audio for the Texas Instruments OMAP chips" depends on ARCH_OMAP && SND_SOC @@ -15,5 +13,3 @@ config SND_OMAP_SOC_N810 select SND_SOC_TLV320AIC3X help Say Y if you want to add support for SoC audio on Nokia N810. - -endmenu -- cgit