From 8afd0ef2639ea2b7bc1b3a0f927ab14e0df034df Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 7 Dec 2012 17:10:05 +0900 Subject: ASoC: wm8994: Fix variable double use Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 82411009d40..f2e63acb267 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3721,7 +3721,7 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) { struct wm8994_priv *wm8994 = data; struct snd_soc_codec *codec = wm8994->hubs.codec; - int reg, count; + int reg, count, ret; /* * Jack detection may have detected a removal simulataneously @@ -3767,11 +3767,11 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) /* Avoid a transient report when the accessory is being removed */ if (wm8994->jackdet) { - reg = snd_soc_read(codec, WM1811_JACKDET_CTRL); - if (reg < 0) { + ret = snd_soc_read(codec, WM1811_JACKDET_CTRL); + if (ret < 0) { dev_err(codec->dev, "Failed to read jack status: %d\n", - reg); - } else if (!(reg & WM1811_JACKDET_LVL)) { + ret); + } else if (!(ret & WM1811_JACKDET_LVL)) { dev_dbg(codec->dev, "Ignoring removed jack\n"); return IRQ_HANDLED; } -- cgit v1.2.3 From ff541f4b2a7546ffa8edf123f4b3b49bb24574e2 Mon Sep 17 00:00:00 2001 From: Chuansheng Liu Date: Fri, 21 Dec 2012 18:17:12 +0800 Subject: ASoC: core: giving WARN when device starting from non-off bias with idle_bias_off Just found some cases that some codec drivers set the bias to _STANDBY and set idle_bias_off to 1 during probing. It will cause unpaired runtime_get_sync/put() issue. Also as Mark suggested, there is no reason to start from _STANDBY bias with idle_bias_off == 1. So here giving one warning when detected (dapm.idle_bias_off == 1) and (dapm.bias_level != SND_SOC_BIAS_OFF) just after driver->probe(). Signed-off-by: liu chuansheng Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 91d592ff67b..e0e8ce0c031 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1107,6 +1107,10 @@ static int soc_probe_codec(struct snd_soc_card *card, "ASoC: failed to probe CODEC %d\n", ret); goto err_probe; } + WARN(codec->dapm.idle_bias_off && + codec->dapm.bias_level != SND_SOC_BIAS_OFF, + "codec %s can not start from non-off bias" + " with idle_bias_off==1\n", codec->name); } /* If the driver didn't set I/O up try regmap */ -- cgit v1.2.3 From 5ce568329e4fcf9e9050bff878f8157ca43bc882 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 12 Dec 2012 23:28:04 -0200 Subject: ASoC: wm8962: Add device tree support Add device tree support. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index bd4b0db4cda..705d0a0e913 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3758,10 +3758,17 @@ static const struct i2c_device_id wm8962_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, wm8962_i2c_id); +static const struct of_device_id wm8962_of_match[] = { + { .compatible = "wlf,wm8962", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8962_of_match); + static struct i2c_driver wm8962_i2c_driver = { .driver = { .name = "wm8962", .owner = THIS_MODULE, + .of_match_table = wm8962_of_match, .pm = &wm8962_pm, }, .probe = wm8962_i2c_probe, -- cgit v1.2.3 From a2ce64750e07d74c51f02c5652edfc1aa2d8e894 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 20 Dec 2012 13:09:59 +0000 Subject: ASoC: wm8962: Convert to devm_input_allocate_device() Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 705d0a0e913..0b3680315df 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3189,7 +3189,7 @@ static void wm8962_init_beep(struct snd_soc_codec *codec) struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); int ret; - wm8962->beep = input_allocate_device(); + wm8962->beep = devm_input_allocate_device(codec->dev); if (!wm8962->beep) { dev_err(codec->dev, "Failed to allocate beep device\n"); return; @@ -3210,7 +3210,6 @@ static void wm8962_init_beep(struct snd_soc_codec *codec) ret = input_register_device(wm8962->beep); if (ret != 0) { - input_free_device(wm8962->beep); wm8962->beep = NULL; dev_err(codec->dev, "Failed to register beep device\n"); } @@ -3227,7 +3226,6 @@ static void wm8962_free_beep(struct snd_soc_codec *codec) struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); device_remove_file(codec->dev, &dev_attr_beep); - input_unregister_device(wm8962->beep); cancel_work_sync(&wm8962->beep_work); wm8962->beep = NULL; -- cgit v1.2.3 From 346f1d40833c75ddb5e9e4b47fb00e20f8152762 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 12 Dec 2012 11:28:01 +0900 Subject: ASoC: wm8962: Unconditionally wait for the FLL to lock If the FLL is being shut down we will exit early so there is no need to check here and in fact we're checking the wrong thing anyway. Reported-by: Graeme Gregory Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 26 ++++++++++++-------------- 1 file changed, 12 insertions(+), 14 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 0b3680315df..e9710280e5e 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2873,22 +2873,20 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source, ret = 0; - if (fll1 & WM8962_FLL_ENA) { - /* This should be a massive overestimate but go even - * higher if we'll error out - */ - if (wm8962->irq) - timeout = msecs_to_jiffies(5); - else - timeout = msecs_to_jiffies(1); + /* This should be a massive overestimate but go even + * higher if we'll error out + */ + if (wm8962->irq) + timeout = msecs_to_jiffies(5); + else + timeout = msecs_to_jiffies(1); - timeout = wait_for_completion_timeout(&wm8962->fll_lock, - timeout); + timeout = wait_for_completion_timeout(&wm8962->fll_lock, + timeout); - if (timeout == 0 && wm8962->irq) { - dev_err(codec->dev, "FLL lock timed out"); - ret = -ETIMEDOUT; - } + if (timeout == 0 && wm8962->irq) { + dev_err(codec->dev, "FLL lock timed out"); + ret = -ETIMEDOUT; } wm8962->fll_fref = Fref; -- cgit v1.2.3 From c63f650c0dd7cee8c3684901b70cdb2d69acb1cd Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 14 Dec 2012 14:46:52 +0900 Subject: ASoC: wm5102: Split input PGA controls Though the controls are named as stereo controls in the part the main use case for the analogue inputs to the WM5102 is mono. Reflect this in the controls exposed to userspace, providing a series of mono controls rather than stereo ones. Signed-off-by: Mark Brown --- sound/soc/codecs/wm5102.c | 64 ++++++++++++++++++++++++++++------------------- 1 file changed, 38 insertions(+), 26 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 688ade08058..a16239f594e 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -608,32 +608,44 @@ SOC_SINGLE("IN2 High Performance Switch", ARIZONA_IN2L_CONTROL, SOC_SINGLE("IN3 High Performance Switch", ARIZONA_IN3L_CONTROL, ARIZONA_IN3_OSR_SHIFT, 1, 0), -SOC_DOUBLE_R_RANGE_TLV("IN1 Volume", ARIZONA_IN1L_CONTROL, - ARIZONA_IN1R_CONTROL, - ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), -SOC_DOUBLE_R_RANGE_TLV("IN2 Volume", ARIZONA_IN2L_CONTROL, - ARIZONA_IN2R_CONTROL, - ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), -SOC_DOUBLE_R_RANGE_TLV("IN3 Volume", ARIZONA_IN3L_CONTROL, - ARIZONA_IN3R_CONTROL, - ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), - -SOC_DOUBLE_R("IN1 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_1L, - ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_MUTE_SHIFT, 1, 1), -SOC_DOUBLE_R("IN2 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_2L, - ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_MUTE_SHIFT, 1, 1), -SOC_DOUBLE_R("IN3 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_3L, - ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_MUTE_SHIFT, 1, 1), - -SOC_DOUBLE_R_TLV("IN1 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L, - ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_DIG_VOL_SHIFT, - 0xbf, 0, digital_tlv), -SOC_DOUBLE_R_TLV("IN2 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2L, - ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_DIG_VOL_SHIFT, - 0xbf, 0, digital_tlv), -SOC_DOUBLE_R_TLV("IN3 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_3L, - ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_DIG_VOL_SHIFT, - 0xbf, 0, digital_tlv), +SOC_SINGLE_RANGE_TLV("IN1L Volume", ARIZONA_IN1L_CONTROL, + ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN1R Volume", ARIZONA_IN1R_CONTROL, + ARIZONA_IN1R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN2L Volume", ARIZONA_IN2L_CONTROL, + ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN2R Volume", ARIZONA_IN2R_CONTROL, + ARIZONA_IN2R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN3L Volume", ARIZONA_IN3L_CONTROL, + ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN3R Volume", ARIZONA_IN3R_CONTROL, + ARIZONA_IN3R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), + +SOC_SINGLE("IN1L Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_IN1L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN1R Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_1R, + ARIZONA_IN1R_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN2L Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_IN2L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN2R Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_2R, + ARIZONA_IN2R_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN3L Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_3L, + ARIZONA_IN3L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN3R Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_3R, + ARIZONA_IN3R_MUTE_SHIFT, 1, 1), + +SOC_SINGLE_TLV("IN1L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_IN1L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN1R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1R, + ARIZONA_IN1R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN2L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_IN2L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN2R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2R, + ARIZONA_IN2R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN3L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_3L, + ARIZONA_IN3L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN3R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_3R, + ARIZONA_IN3R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), SOC_ENUM("Input Ramp Up", arizona_in_vi_ramp), SOC_ENUM("Input Ramp Down", arizona_in_vd_ramp), -- cgit v1.2.3 From 02482da46ec13856c9244812db13d4d1be61bff7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 14 Dec 2012 15:18:30 +0900 Subject: ASoC: wm5110: Split input PGA controls Though the controls are named as stereo controls in the part the main use case for the analogue inputs to the WM5102 is mono. Reflect this in the controls exposed to userspace, providing a series of mono controls rather than stereo ones. Signed-off-by: Mark Brown --- sound/soc/codecs/wm5110.c | 77 ++++++++++++++++++++++++++++------------------- 1 file changed, 46 insertions(+), 31 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index ae80c8c2853..9cdac35914f 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -52,37 +52,52 @@ SOC_SINGLE("IN3 High Performance Switch", ARIZONA_IN3L_CONTROL, SOC_SINGLE("IN4 High Performance Switch", ARIZONA_IN4L_CONTROL, ARIZONA_IN4_OSR_SHIFT, 1, 0), -SOC_DOUBLE_R_RANGE_TLV("IN1 Volume", ARIZONA_IN1L_CONTROL, - ARIZONA_IN1R_CONTROL, - ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), -SOC_DOUBLE_R_RANGE_TLV("IN2 Volume", ARIZONA_IN2L_CONTROL, - ARIZONA_IN2R_CONTROL, - ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), -SOC_DOUBLE_R_RANGE_TLV("IN3 Volume", ARIZONA_IN3L_CONTROL, - ARIZONA_IN3R_CONTROL, - ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), - -SOC_DOUBLE_R("IN1 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_1L, - ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_MUTE_SHIFT, 1, 1), -SOC_DOUBLE_R("IN2 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_2L, - ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_MUTE_SHIFT, 1, 1), -SOC_DOUBLE_R("IN3 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_3L, - ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_MUTE_SHIFT, 1, 1), -SOC_DOUBLE_R("IN4 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_4L, - ARIZONA_ADC_DIGITAL_VOLUME_4R, ARIZONA_IN4L_MUTE_SHIFT, 1, 1), - -SOC_DOUBLE_R_TLV("IN1 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L, - ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_DIG_VOL_SHIFT, - 0xbf, 0, digital_tlv), -SOC_DOUBLE_R_TLV("IN2 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2L, - ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_DIG_VOL_SHIFT, - 0xbf, 0, digital_tlv), -SOC_DOUBLE_R_TLV("IN3 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_3L, - ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_DIG_VOL_SHIFT, - 0xbf, 0, digital_tlv), -SOC_DOUBLE_R_TLV("IN4 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_4L, - ARIZONA_ADC_DIGITAL_VOLUME_4R, ARIZONA_IN4L_DIG_VOL_SHIFT, - 0xbf, 0, digital_tlv), +SOC_SINGLE_RANGE_TLV("IN1L Volume", ARIZONA_IN1L_CONTROL, + ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN1R Volume", ARIZONA_IN1R_CONTROL, + ARIZONA_IN1R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN2L Volume", ARIZONA_IN2L_CONTROL, + ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN2R Volume", ARIZONA_IN2R_CONTROL, + ARIZONA_IN2R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN3L Volume", ARIZONA_IN3L_CONTROL, + ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN3R Volume", ARIZONA_IN3R_CONTROL, + ARIZONA_IN3R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), + +SOC_SINGLE("IN1L Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_IN1L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN1R Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_1R, + ARIZONA_IN1R_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN2L Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_IN2L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN2R Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_2R, + ARIZONA_IN2R_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN3L Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_3L, + ARIZONA_IN3L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN3R Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_3R, + ARIZONA_IN3R_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN4L Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_4L, + ARIZONA_IN4L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN4R Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_4R, + ARIZONA_IN4R_MUTE_SHIFT, 1, 1), + +SOC_SINGLE_TLV("IN1L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_IN1L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN1R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1R, + ARIZONA_IN1R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN2L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_IN2L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN2R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2R, + ARIZONA_IN2R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN3L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_3L, + ARIZONA_IN3L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN3R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_3R, + ARIZONA_IN3R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN4L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_4L, + ARIZONA_IN4L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN4R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_4R, + ARIZONA_IN4R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), SOC_ENUM("Input Ramp Up", arizona_in_vi_ramp), SOC_ENUM("Input Ramp Down", arizona_in_vd_ramp), -- cgit v1.2.3 From 845571cce6aebffd6de9b9024daf789f333a734d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 18 Dec 2012 13:47:57 +0000 Subject: ASoC: arizona: Add noise gate hold time enumeration Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 9 +++++++++ sound/soc/codecs/arizona.h | 2 ++ 2 files changed, 11 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index adf397b9d0e..819920dfa90 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -275,6 +275,15 @@ const struct soc_enum arizona_lhpf4_mode = arizona_lhpf_mode_text); EXPORT_SYMBOL_GPL(arizona_lhpf4_mode); +static const char *arizona_ng_hold_text[] = { + "30ms", "120ms", "250ms", "500ms", +}; + +const struct soc_enum arizona_ng_hold = + SOC_ENUM_SINGLE(ARIZONA_NOISE_GATE_CONTROL, ARIZONA_NGATE_HOLD_SHIFT, + 4, arizona_ng_hold_text); +EXPORT_SYMBOL_GPL(arizona_ng_hold); + int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 41dae1ed3b7..3d083d00579 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -176,6 +176,8 @@ extern const struct soc_enum arizona_lhpf2_mode; extern const struct soc_enum arizona_lhpf3_mode; extern const struct soc_enum arizona_lhpf4_mode; +extern const struct soc_enum arizona_ng_hold; + extern int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); -- cgit v1.2.3 From 50571263722452a57a67edee23b892b2607dc374 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 18 Dec 2012 13:48:36 +0000 Subject: ASoC: wm5102: Add noise gate control The references used for the noise gates and parameters for their triggering are configurable, expose that to users. Signed-off-by: Mark Brown --- sound/soc/codecs/wm5102.c | 28 ++++++++++++++++++++++++++++ 1 file changed, 28 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index a16239f594e..e64b968a8c9 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -42,6 +42,7 @@ static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0); static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0); static DECLARE_TLV_DB_SCALE(noise_tlv, 0, 600, 0); +static DECLARE_TLV_DB_SCALE(ng_tlv, -10200, 600, 0); static const struct wm_adsp_region wm5102_dsp1_regions[] = { { .type = WMFW_ADSP2_PM, .base = 0x100000 }, @@ -600,6 +601,17 @@ static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w, return 0; } +#define WM5102_NG_SRC(name, base) \ + SOC_SINGLE(name " NG HPOUT1L Switch", base, 0, 1, 0), \ + SOC_SINGLE(name " NG HPOUT1R Switch", base, 1, 1, 0), \ + SOC_SINGLE(name " NG HPOUT2L Switch", base, 2, 1, 0), \ + SOC_SINGLE(name " NG HPOUT2R Switch", base, 3, 1, 0), \ + SOC_SINGLE(name " NG EPOUT Switch", base, 4, 1, 0), \ + SOC_SINGLE(name " NG SPKOUTL Switch", base, 6, 1, 0), \ + SOC_SINGLE(name " NG SPKOUTR Switch", base, 7, 1, 0), \ + SOC_SINGLE(name " NG SPKDAT1L Switch", base, 8, 1, 0), \ + SOC_SINGLE(name " NG SPKDAT1R Switch", base, 9, 1, 0) + static const struct snd_kcontrol_new wm5102_snd_controls[] = { SOC_SINGLE("IN1 High Performance Switch", ARIZONA_IN1L_CONTROL, ARIZONA_IN1_OSR_SHIFT, 1, 0), @@ -783,6 +795,22 @@ SOC_ENUM("Output Ramp Down", arizona_out_vd_ramp), SOC_DOUBLE("SPKDAT1 Switch", ARIZONA_PDM_SPK1_CTRL_1, ARIZONA_SPK1L_MUTE_SHIFT, ARIZONA_SPK1R_MUTE_SHIFT, 1, 1), +SOC_SINGLE("Noise Gate Switch", ARIZONA_NOISE_GATE_CONTROL, + ARIZONA_NGATE_ENA_SHIFT, 1, 0), +SOC_SINGLE_TLV("Noise Gate Threshold Volume", ARIZONA_NOISE_GATE_CONTROL, + ARIZONA_NGATE_THR_SHIFT, 7, 1, ng_tlv), +SOC_ENUM("Noise Gate Hold", arizona_ng_hold), + +WM5102_NG_SRC("HPOUT1L", ARIZONA_NOISE_GATE_SELECT_1L), +WM5102_NG_SRC("HPOUT1R", ARIZONA_NOISE_GATE_SELECT_1R), +WM5102_NG_SRC("HPOUT2L", ARIZONA_NOISE_GATE_SELECT_2L), +WM5102_NG_SRC("HPOUT2R", ARIZONA_NOISE_GATE_SELECT_2R), +WM5102_NG_SRC("EPOUT", ARIZONA_NOISE_GATE_SELECT_3L), +WM5102_NG_SRC("SPKOUTL", ARIZONA_NOISE_GATE_SELECT_4L), +WM5102_NG_SRC("SPKOUTR", ARIZONA_NOISE_GATE_SELECT_4R), +WM5102_NG_SRC("SPKDAT1L", ARIZONA_NOISE_GATE_SELECT_5L), +WM5102_NG_SRC("SPKDAT1R", ARIZONA_NOISE_GATE_SELECT_5R), + ARIZONA_MIXER_CONTROLS("AIF1TX1", ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("AIF1TX2", ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("AIF1TX3", ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE), -- cgit v1.2.3 From bd7fe24bc43bc75255cebb7aed5ed675579531d3 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 18 Dec 2012 13:53:26 +0000 Subject: ASoC: wm5110: Add noise gate control The references used for the noise gates and parameters for their triggering are configurable, expose that to users. Signed-off-by: Mark Brown --- sound/soc/codecs/wm5110.c | 34 ++++++++++++++++++++++++++++++++++ 1 file changed, 34 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 9cdac35914f..8f7081e1417 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -41,6 +41,21 @@ static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0); static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0); static DECLARE_TLV_DB_SCALE(noise_tlv, 0, 600, 0); +static DECLARE_TLV_DB_SCALE(ng_tlv, -10200, 600, 0); + +#define WM5110_NG_SRC(name, base) \ + SOC_SINGLE(name " NG HPOUT1L Switch", base, 0, 1, 0), \ + SOC_SINGLE(name " NG HPOUT1R Switch", base, 1, 1, 0), \ + SOC_SINGLE(name " NG HPOUT2L Switch", base, 2, 1, 0), \ + SOC_SINGLE(name " NG HPOUT2R Switch", base, 3, 1, 0), \ + SOC_SINGLE(name " NG HPOUT3L Switch", base, 4, 1, 0), \ + SOC_SINGLE(name " NG HPOUT3R Switch", base, 5, 1, 0), \ + SOC_SINGLE(name " NG SPKOUTL Switch", base, 6, 1, 0), \ + SOC_SINGLE(name " NG SPKOUTR Switch", base, 7, 1, 0), \ + SOC_SINGLE(name " NG SPKDAT1L Switch", base, 8, 1, 0), \ + SOC_SINGLE(name " NG SPKDAT1R Switch", base, 9, 1, 0), \ + SOC_SINGLE(name " NG SPKDAT2L Switch", base, 10, 1, 0), \ + SOC_SINGLE(name " NG SPKDAT2R Switch", base, 11, 1, 0) static const struct snd_kcontrol_new wm5110_snd_controls[] = { SOC_SINGLE("IN1 High Performance Switch", ARIZONA_IN1L_CONTROL, @@ -278,6 +293,25 @@ SOC_DOUBLE("SPKDAT2 Switch", ARIZONA_PDM_SPK2_CTRL_1, ARIZONA_SPK2L_MUTE_SHIFT, SOC_ENUM("Output Ramp Up", arizona_out_vi_ramp), SOC_ENUM("Output Ramp Down", arizona_out_vd_ramp), +SOC_SINGLE("Noise Gate Switch", ARIZONA_NOISE_GATE_CONTROL, + ARIZONA_NGATE_ENA_SHIFT, 1, 0), +SOC_SINGLE_TLV("Noise Gate Threshold Volume", ARIZONA_NOISE_GATE_CONTROL, + ARIZONA_NGATE_THR_SHIFT, 7, 1, ng_tlv), +SOC_ENUM("Noise Gate Hold", arizona_ng_hold), + +WM5110_NG_SRC("HPOUT1L", ARIZONA_NOISE_GATE_SELECT_1L), +WM5110_NG_SRC("HPOUT1R", ARIZONA_NOISE_GATE_SELECT_1R), +WM5110_NG_SRC("HPOUT2L", ARIZONA_NOISE_GATE_SELECT_2L), +WM5110_NG_SRC("HPOUT2R", ARIZONA_NOISE_GATE_SELECT_2R), +WM5110_NG_SRC("HPOUT3L", ARIZONA_NOISE_GATE_SELECT_3L), +WM5110_NG_SRC("HPOUT3R", ARIZONA_NOISE_GATE_SELECT_3R), +WM5110_NG_SRC("SPKOUTL", ARIZONA_NOISE_GATE_SELECT_4L), +WM5110_NG_SRC("SPKOUTR", ARIZONA_NOISE_GATE_SELECT_4R), +WM5110_NG_SRC("SPKDAT1L", ARIZONA_NOISE_GATE_SELECT_5L), +WM5110_NG_SRC("SPKDAT1R", ARIZONA_NOISE_GATE_SELECT_5R), +WM5110_NG_SRC("SPKDAT2L", ARIZONA_NOISE_GATE_SELECT_6L), +WM5110_NG_SRC("SPKDAT2R", ARIZONA_NOISE_GATE_SELECT_6R), + ARIZONA_MIXER_CONTROLS("AIF1TX1", ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("AIF1TX2", ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("AIF1TX3", ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE), -- cgit v1.2.3 From 01df259f592147db97293b90d03e5fd8075cbeb3 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 12 Dec 2012 16:22:08 +0900 Subject: ASoC: arizona: Implement tristate support Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 819920dfa90..e55fa034eec 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -803,11 +803,27 @@ static int arizona_dai_set_sysclk(struct snd_soc_dai *dai, return snd_soc_dapm_sync(&codec->dapm); } +static int arizona_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct snd_soc_codec *codec = dai->codec; + int base = dai->driver->base; + unsigned int reg; + + if (tristate) + reg = ARIZONA_AIF1_TRI; + else + reg = 0; + + return snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL, + ARIZONA_AIF1_TRI, reg); +} + const struct snd_soc_dai_ops arizona_dai_ops = { .startup = arizona_startup, .set_fmt = arizona_set_fmt, .hw_params = arizona_hw_params, .set_sysclk = arizona_dai_set_sysclk, + .set_tristate = arizona_set_tristate, }; EXPORT_SYMBOL_GPL(arizona_dai_ops); -- cgit v1.2.3 From 514cfd6dd72508b79030c8504764a73a7261b713 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 13 Dec 2012 17:29:00 +0900 Subject: ASoC: wm2000: Integrate with clock API Request MCLK as a clock and then enable it when carrying out a state transtion and while ANC is active, minimising system power consumption in idle modes. Signed-off-by: Mark Brown --- sound/soc/codecs/wm2000.c | 23 ++++++++++++++++++++++- 1 file changed, 22 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 1cbe88f01d6..0aba8ce424c 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -26,6 +26,7 @@ #include #include #include +#include #include #include #include @@ -62,6 +63,7 @@ enum wm2000_anc_mode { struct wm2000_priv { struct i2c_client *i2c; struct regmap *regmap; + struct clk *mclk; struct regulator_bulk_data supplies[WM2000_NUM_SUPPLIES]; @@ -550,6 +552,15 @@ static int wm2000_anc_transition(struct wm2000_priv *wm2000, return -EINVAL; } + /* Maintain clock while active */ + if (anc_transitions[i].source == ANC_OFF) { + ret = clk_prepare_enable(wm2000->mclk); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to enable MCLK: %d\n", ret); + return ret; + } + } + for (j = 0; j < ARRAY_SIZE(anc_transitions[j].step); j++) { if (!anc_transitions[i].step[j]) break; @@ -559,7 +570,10 @@ static int wm2000_anc_transition(struct wm2000_priv *wm2000, return ret; } - return 0; + if (anc_transitions[i].dest == ANC_OFF) + clk_disable_unprepare(wm2000->mclk); + + return ret; } static int wm2000_anc_set_mode(struct wm2000_priv *wm2000) @@ -823,6 +837,13 @@ static int wm2000_i2c_probe(struct i2c_client *i2c, reg = wm2000_read(i2c, WM2000_REG_REVISON); dev_info(&i2c->dev, "revision %c\n", reg + 'A'); + wm2000->mclk = devm_clk_get(&i2c->dev, "MCLK"); + if (IS_ERR(wm2000->mclk)) { + ret = PTR_ERR(wm2000->mclk); + dev_err(&i2c->dev, "Failed to get MCLK: %d\n", ret); + goto err_supplies; + } + filename = "wm2000_anc.bin"; pdata = dev_get_platdata(&i2c->dev); if (pdata) { -- cgit v1.2.3 From d61100bbd18e8b3fc9406be55354dabd5e7525ec Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 14 Dec 2012 15:16:58 +0900 Subject: ASoC: wm2000: Use clock API integration to configure MCLK divisor Since we are now using the clock API integration to manage MCLK we can now use clk_get_rate() to determine if we need to divide MCLK without relying on platform data. Signed-off-by: Mark Brown --- sound/soc/codecs/wm2000.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 0aba8ce424c..85550dc319d 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -73,7 +73,6 @@ struct wm2000_priv { unsigned int anc_eng_ena:1; unsigned int spk_ena:1; - unsigned int mclk_div:1; unsigned int speech_clarity:1; int anc_download_size; @@ -133,6 +132,7 @@ static int wm2000_poll_bit(struct i2c_client *i2c, static int wm2000_power_up(struct i2c_client *i2c, int analogue) { struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); + unsigned long rate; int ret; BUG_ON(wm2000->anc_mode != ANC_OFF); @@ -145,7 +145,8 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue) return ret; } - if (!wm2000->mclk_div) { + rate = clk_get_rate(wm2000->mclk); + if (rate <= 13500000) { dev_dbg(&i2c->dev, "Disabling MCLK divider\n"); wm2000_write(i2c, WM2000_REG_SYS_CTL2, WM2000_MCLK_DIV2_ENA_CLR); @@ -847,7 +848,6 @@ static int wm2000_i2c_probe(struct i2c_client *i2c, filename = "wm2000_anc.bin"; pdata = dev_get_platdata(&i2c->dev); if (pdata) { - wm2000->mclk_div = pdata->mclkdiv2; wm2000->speech_clarity = !pdata->speech_enh_disable; if (pdata->download_file) -- cgit v1.2.3 From 153f5a18e4a88d8a3c2d1a19658a3ce34559b473 Mon Sep 17 00:00:00 2001 From: Joachim Eastwood Date: Sat, 8 Dec 2012 14:23:22 +0100 Subject: ASoC: atmel-soc: make it buildable on other architectures Not very useful on non AT91/AVR32 platforms but it provides more build coverage and prepares for ARM multiplatform. Also fixes a couple of format type warnings. Signed-off-by: Joachim Eastwood Acked-by: Nicolas Ferre Signed-off-by: Mark Brown --- sound/soc/atmel/Kconfig | 6 +++--- sound/soc/atmel/atmel-pcm-pdc.c | 4 ++-- sound/soc/atmel/atmel-pcm.c | 2 +- sound/soc/atmel/atmel_ssc_dai.c | 2 -- 4 files changed, 6 insertions(+), 8 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index d1b691bf8e2..3fdd87fa18a 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -1,6 +1,6 @@ config SND_ATMEL_SOC tristate "SoC Audio for the Atmel System-on-Chip" - depends on ARCH_AT91 + depends on HAS_IOMEM help Say Y or M if you want to add support for codecs attached to the ATMEL SSC interface. You will also need @@ -24,7 +24,7 @@ config SND_ATMEL_SOC_SSC config SND_AT91_SOC_SAM9G20_WM8731 tristate "SoC Audio support for WM8731-based At91sam9g20 evaluation board" - depends on ATMEL_SSC && SND_ATMEL_SOC && AT91_PROGRAMMABLE_CLOCKS + depends on ARCH_AT91 && ATMEL_SSC && SND_ATMEL_SOC && AT91_PROGRAMMABLE_CLOCKS select SND_ATMEL_SOC_PDC select SND_ATMEL_SOC_SSC select SND_SOC_WM8731 @@ -34,7 +34,7 @@ config SND_AT91_SOC_SAM9G20_WM8731 config SND_AT91_SOC_AFEB9260 tristate "SoC Audio support for AFEB9260 board" - depends on ATMEL_SSC && ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC + depends on ARCH_AT91 && ATMEL_SSC && ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC select SND_ATMEL_SOC_PDC select SND_ATMEL_SOC_SSC select SND_SOC_TLV320AIC23 diff --git a/sound/soc/atmel/atmel-pcm-pdc.c b/sound/soc/atmel/atmel-pcm-pdc.c index 6a293c713a3..054ea4d9326 100644 --- a/sound/soc/atmel/atmel-pcm-pdc.c +++ b/sound/soc/atmel/atmel-pcm-pdc.c @@ -159,7 +159,7 @@ static int atmel_pcm_hw_params(struct snd_pcm_substream *substream, pr_debug("atmel-pcm: " "hw_params: DMA for %s initialized " - "(dma_bytes=%u, period_size=%u)\n", + "(dma_bytes=%zu, period_size=%zu)\n", prtd->params->name, runtime->dma_bytes, prtd->period_size); @@ -201,7 +201,7 @@ static int atmel_pcm_trigger(struct snd_pcm_substream *substream, int ret = 0; pr_debug("atmel-pcm:buffer_size = %ld," - "dma_area = %p, dma_bytes = %u\n", + "dma_area = %p, dma_bytes = %zu\n", rtd->buffer_size, rtd->dma_area, rtd->dma_bytes); switch (cmd) { diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c index e99f1811300..3109db7b901 100644 --- a/sound/soc/atmel/atmel-pcm.c +++ b/sound/soc/atmel/atmel-pcm.c @@ -49,7 +49,7 @@ static int atmel_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, buf->private_data = NULL; buf->area = dma_alloc_coherent(pcm->card->dev, size, &buf->addr, GFP_KERNEL); - pr_debug("atmel-pcm: alloc dma buffer: area=%p, addr=%p, size=%d\n", + pr_debug("atmel-pcm: alloc dma buffer: area=%p, addr=%p, size=%zu\n", (void *)buf->area, (void *)buf->addr, size); if (!buf->area) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 1c766342205..2755750bda4 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -42,8 +42,6 @@ #include #include -#include - #include "atmel-pcm.h" #include "atmel_ssc_dai.h" -- cgit v1.2.3 From fd23fb9f6bfd43a6e62b2646d18d5ca3edc3ebe3 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 10 Dec 2012 10:30:04 +0100 Subject: ALSA: ASoC: cs4271: add optional soft reset workaround The CS4271 requires its LRCLK and MCLK to be stable before its RESET line is de-asserted. That also means that clocks cannot be changed without putting the chip back into hardware reset, which also requires a complete re-initialization of all registers. One (undocumented) workaround is to assert and de-assert the PDN bit in the MODE2 register. This patch adds a new flag to both the DT bindings as well as to the platform data to enable that workaround. Signed-off-by: Daniel Mack Acked-by: Alexander Sverdlin Signed-off-by: Mark Brown --- sound/soc/codecs/cs4271.c | 34 ++++++++++++++++++++++++++++++++++ 1 file changed, 34 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index ac8742a1f25..2415a4118db 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -167,6 +167,8 @@ struct cs4271_private { int gpio_nreset; /* GPIO that disable serial bus, if any */ int gpio_disable; + /* enable soft reset workaround */ + bool enable_soft_reset; }; /* @@ -325,6 +327,33 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, int i, ret; unsigned int ratio, val; + if (cs4271->enable_soft_reset) { + /* + * Put the codec in soft reset and back again in case it's not + * currently streaming data. This way of bringing the codec in + * sync to the current clocks is not explicitly documented in + * the data sheet, but it seems to work fine, and in contrast + * to a read hardware reset, we don't have to sync back all + * registers every time. + */ + + if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK && + !dai->capture_active) || + (substream->stream == SNDRV_PCM_STREAM_CAPTURE && + !dai->playback_active)) { + ret = snd_soc_update_bits(codec, CS4271_MODE2, + CS4271_MODE2_PDN, + CS4271_MODE2_PDN); + if (ret < 0) + return ret; + + ret = snd_soc_update_bits(codec, CS4271_MODE2, + CS4271_MODE2_PDN, 0); + if (ret < 0) + return ret; + } + } + cs4271->rate = params_rate(params); /* Configure DAC */ @@ -484,6 +513,10 @@ static int cs4271_probe(struct snd_soc_codec *codec) if (of_get_property(codec->dev->of_node, "cirrus,amutec-eq-bmutec", NULL)) amutec_eq_bmutec = true; + + if (of_get_property(codec->dev->of_node, + "cirrus,enable-soft-reset", NULL)) + cs4271->enable_soft_reset = true; } #endif @@ -492,6 +525,7 @@ static int cs4271_probe(struct snd_soc_codec *codec) gpio_nreset = cs4271plat->gpio_nreset; amutec_eq_bmutec = cs4271plat->amutec_eq_bmutec; + cs4271->enable_soft_reset = cs4271plat->enable_soft_reset; } if (gpio_nreset >= 0) -- cgit v1.2.3 From 1edbd35667c5a7f7cf9140b2009027ac9546e8fd Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 12 Dec 2012 00:53:13 -0200 Subject: ASoC: wm8804: Remove redundant check The condition "if (!freq_in || !freq_out)" has already been tested previously, so no need to do it again. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/wm8804.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index d321a875b02..1704b1e119c 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -395,9 +395,6 @@ static int wm8804_set_pll(struct snd_soc_dai *dai, int pll_id, /* power down the PLL before reprogramming it */ snd_soc_update_bits(codec, WM8804_PWRDN, 0x1, 0x1); - if (!freq_in || !freq_out) - return 0; - /* set PLLN and PRESCALE */ snd_soc_update_bits(codec, WM8804_PLL4, 0xf | 0x10, pll_div.n | (pll_div.prescale << 4)); -- cgit v1.2.3 From 6757d8cc0c6ff2e61972d84797d358a2f69f2217 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 12 Dec 2012 01:24:40 -0200 Subject: ASoC: wm8993: Refactor set_pll code to avoid GCC warnings Refactor set_pll code to avoid the following warnings: sound/soc/codecs/wm8983.c:873:40: warning: 'pll_div.k' may be used uninitialized in this function [-Wuninitialized] sound/soc/codecs/wm8983.c:870:9: warning: 'pll_div.n' may be used uninitialized in this function [-Wuninitialized] sound/soc/codecs/wm8983.c:869:23: warning: 'pll_div.div2' may be used uninitialized in this function [-Wuninitialized] Do the same as in commit 86ce6c9a (ASoC: WM8804: Refactor set_pll code to avoid GCC warnings). Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/wm8983.c | 41 ++++++++++++++++++++++------------------- 1 file changed, 22 insertions(+), 19 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c index 9fe1e041da4..c9c707b8698 100644 --- a/sound/soc/codecs/wm8983.c +++ b/sound/soc/codecs/wm8983.c @@ -851,30 +851,33 @@ static int wm8983_set_pll(struct snd_soc_dai *dai, int pll_id, struct pll_div pll_div; codec = dai->codec; - if (freq_in && freq_out) { + if (!freq_in || !freq_out) { + /* disable the PLL */ + snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1, + WM8983_PLLEN_MASK, 0); + return 0; + } else { ret = pll_factors(&pll_div, freq_out * 4 * 2, freq_in); if (ret) return ret; - } - - /* disable the PLL before re-programming it */ - snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1, - WM8983_PLLEN_MASK, 0); - if (!freq_in || !freq_out) - return 0; + /* disable the PLL before re-programming it */ + snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1, + WM8983_PLLEN_MASK, 0); + + /* set PLLN and PRESCALE */ + snd_soc_write(codec, WM8983_PLL_N, + (pll_div.div2 << WM8983_PLL_PRESCALE_SHIFT) + | pll_div.n); + /* set PLLK */ + snd_soc_write(codec, WM8983_PLL_K_3, pll_div.k & 0x1ff); + snd_soc_write(codec, WM8983_PLL_K_2, (pll_div.k >> 9) & 0x1ff); + snd_soc_write(codec, WM8983_PLL_K_1, (pll_div.k >> 18)); + /* enable the PLL */ + snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1, + WM8983_PLLEN_MASK, WM8983_PLLEN); + } - /* set PLLN and PRESCALE */ - snd_soc_write(codec, WM8983_PLL_N, - (pll_div.div2 << WM8983_PLL_PRESCALE_SHIFT) - | pll_div.n); - /* set PLLK */ - snd_soc_write(codec, WM8983_PLL_K_3, pll_div.k & 0x1ff); - snd_soc_write(codec, WM8983_PLL_K_2, (pll_div.k >> 9) & 0x1ff); - snd_soc_write(codec, WM8983_PLL_K_1, (pll_div.k >> 18)); - /* enable the PLL */ - snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1, - WM8983_PLLEN_MASK, WM8983_PLLEN); return 0; } -- cgit v1.2.3 From 6cbdbffba19620db77de38094f407b6f21d3f10c Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Sun, 16 Dec 2012 22:12:21 -0800 Subject: ASoC: fsi: remove platform depended .set_rate() callback support ab6f6d85210c4d0265cf48e9958c04e08595055a (ASoC: fsi: add master clock control functions) added driver level clock control functions. And now, platform depended .set_rate() is no longer needed. This patch removed unnecessary .set_rate() platform callback support. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 131 +++++------------------------------------------------ 1 file changed, 12 insertions(+), 119 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index a606d0f93d1..5cb1332e043 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -131,8 +131,6 @@ #define FSI_FMTS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE) -typedef int (*set_rate_func)(struct device *dev, int rate, int enable); - /* * bus options * @@ -244,8 +242,7 @@ struct fsi_clk { struct clk *ick; struct clk *div; int (*set_rate)(struct device *dev, - struct fsi_priv *fsi, - unsigned long rate); + struct fsi_priv *fsi); unsigned long rate; unsigned int count; @@ -270,8 +267,6 @@ struct fsi_priv { int enable_stream:1; int bit_clk_inv:1; int lr_clk_inv:1; - - long rate; }; struct fsi_stream_handler { @@ -431,14 +426,6 @@ static struct fsi_priv *fsi_get_priv(struct snd_pcm_substream *substream) return fsi_get_priv_frm_dai(fsi_get_dai(substream)); } -static set_rate_func fsi_get_info_set_rate(struct fsi_priv *fsi) -{ - if (!fsi->info) - return NULL; - - return fsi->info->set_rate; -} - static u32 fsi_get_info_flags(struct fsi_priv *fsi) { if (!fsi->info) @@ -757,8 +744,7 @@ static int fsi_clk_init(struct device *dev, int ick, int div, int (*set_rate)(struct device *dev, - struct fsi_priv *fsi, - unsigned long rate)) + struct fsi_priv *fsi)) { struct fsi_clk *clock = &fsi->clock; int is_porta = fsi_is_port_a(fsi); @@ -829,8 +815,7 @@ static int fsi_clk_is_valid(struct fsi_priv *fsi) } static int fsi_clk_enable(struct device *dev, - struct fsi_priv *fsi, - unsigned long rate) + struct fsi_priv *fsi) { struct fsi_clk *clock = &fsi->clock; int ret = -EINVAL; @@ -839,7 +824,7 @@ static int fsi_clk_enable(struct device *dev, return ret; if (0 == clock->count) { - ret = clock->set_rate(dev, fsi, rate); + ret = clock->set_rate(dev, fsi); if (ret < 0) { fsi_clk_invalid(fsi); return ret; @@ -946,11 +931,11 @@ static int fsi_clk_set_ackbpf(struct device *dev, } static int fsi_clk_set_rate_external(struct device *dev, - struct fsi_priv *fsi, - unsigned long rate) + struct fsi_priv *fsi) { struct clk *xck = fsi->clock.xck; struct clk *ick = fsi->clock.ick; + unsigned long rate = fsi->clock.rate; unsigned long xrate; int ackmd, bpfmd; int ret = 0; @@ -978,11 +963,11 @@ static int fsi_clk_set_rate_external(struct device *dev, } static int fsi_clk_set_rate_cpg(struct device *dev, - struct fsi_priv *fsi, - unsigned long rate) + struct fsi_priv *fsi) { struct clk *ick = fsi->clock.ick; struct clk *div = fsi->clock.div; + unsigned long rate = fsi->clock.rate; unsigned long target = 0; /* 12288000 or 11289600 */ unsigned long actual, cout; unsigned long diff, min; @@ -1063,85 +1048,6 @@ static int fsi_clk_set_rate_cpg(struct device *dev, return ret; } -static int fsi_set_master_clk(struct device *dev, struct fsi_priv *fsi, - long rate, int enable) -{ - set_rate_func set_rate = fsi_get_info_set_rate(fsi); - int ret; - - /* - * CAUTION - * - * set_rate will be deleted - */ - if (!set_rate) { - if (enable) - return fsi_clk_enable(dev, fsi, rate); - else - return fsi_clk_disable(dev, fsi); - } - - ret = set_rate(dev, rate, enable); - if (ret < 0) /* error */ - return ret; - - if (!enable) - return 0; - - if (ret > 0) { - u32 data = 0; - - switch (ret & SH_FSI_ACKMD_MASK) { - default: - /* FALL THROUGH */ - case SH_FSI_ACKMD_512: - data |= (0x0 << 12); - break; - case SH_FSI_ACKMD_256: - data |= (0x1 << 12); - break; - case SH_FSI_ACKMD_128: - data |= (0x2 << 12); - break; - case SH_FSI_ACKMD_64: - data |= (0x3 << 12); - break; - case SH_FSI_ACKMD_32: - data |= (0x4 << 12); - break; - } - - switch (ret & SH_FSI_BPFMD_MASK) { - default: - /* FALL THROUGH */ - case SH_FSI_BPFMD_32: - data |= (0x0 << 8); - break; - case SH_FSI_BPFMD_64: - data |= (0x1 << 8); - break; - case SH_FSI_BPFMD_128: - data |= (0x2 << 8); - break; - case SH_FSI_BPFMD_256: - data |= (0x3 << 8); - break; - case SH_FSI_BPFMD_512: - data |= (0x4 << 8); - break; - case SH_FSI_BPFMD_16: - data |= (0x7 << 8); - break; - } - - fsi_reg_mask_set(fsi, CKG1, (ACKMD_MASK | BPFMD_MASK) , data); - udelay(10); - ret = 0; - } - - return ret; -} - /* * pio data transfer handler */ @@ -1698,7 +1604,7 @@ static int fsi_hw_startup(struct fsi_priv *fsi, /* start master clock */ if (fsi_is_clk_master(fsi)) - return fsi_set_master_clk(dev, fsi, fsi->rate, 1); + return fsi_clk_enable(dev, fsi); return 0; } @@ -1708,7 +1614,7 @@ static int fsi_hw_shutdown(struct fsi_priv *fsi, { /* stop master clock */ if (fsi_is_clk_master(fsi)) - return fsi_set_master_clk(dev, fsi, fsi->rate, 0); + return fsi_clk_disable(dev, fsi); return 0; } @@ -1719,7 +1625,6 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream, struct fsi_priv *fsi = fsi_get_priv(substream); fsi_clk_invalid(fsi); - fsi->rate = 0; return 0; } @@ -1730,7 +1635,6 @@ static void fsi_dai_shutdown(struct snd_pcm_substream *substream, struct fsi_priv *fsi = fsi_get_priv(substream); fsi_clk_invalid(fsi); - fsi->rate = 0; } static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd, @@ -1795,7 +1699,6 @@ static int fsi_set_fmt_spdif(struct fsi_priv *fsi) static int fsi_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct fsi_priv *fsi = fsi_get_priv_frm_dai(dai); - set_rate_func set_rate = fsi_get_info_set_rate(fsi); int ret; /* set master/slave audio interface */ @@ -1831,14 +1734,6 @@ static int fsi_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) } if (fsi_is_clk_master(fsi)) { - /* - * CAUTION - * - * set_rate will be deleted - */ - if (set_rate) - dev_warn(dai->dev, "set_rate will be removed soon\n"); - if (fsi->clk_cpg) fsi_clk_init(dai->dev, fsi, 0, 1, 1, fsi_clk_set_rate_cpg); @@ -1862,10 +1757,8 @@ static int fsi_dai_hw_params(struct snd_pcm_substream *substream, { struct fsi_priv *fsi = fsi_get_priv(substream); - if (fsi_is_clk_master(fsi)) { - fsi->rate = params_rate(params); - fsi_clk_valid(fsi, fsi->rate); - } + if (fsi_is_clk_master(fsi)) + fsi_clk_valid(fsi, params_rate(params)); return 0; } -- cgit v1.2.3 From abca75814a82c0c53c0a8ec7fa1300c133bc4f01 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Sun, 16 Dec 2012 22:12:42 -0800 Subject: ASoC: fsi: remove SH_FSI_xxx_INV flags 3449f5fab8c51e37a8a48bc2516588c615373191 (ASoC: fsi: add SND_SOC_DAIFMT_INV_xxx support) added clock inversion support via snd_soc_dai_set_fmt(). Thus, this patch removed SH_FSI_xxx_INV and fsi_get_info() from fsi driver, and modified platform settings to use new style. Then, it cleaned up meaningless settings from platform. Signed-off-by: Kuninori Morimoto Acked-by: Simon Horman Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 25 ------------------------- 1 file changed, 25 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 5cb1332e043..f14c611b38c 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -251,7 +251,6 @@ struct fsi_clk { struct fsi_priv { void __iomem *base; struct fsi_master *master; - struct sh_fsi_port_info *info; struct fsi_stream playback; struct fsi_stream capture; @@ -426,14 +425,6 @@ static struct fsi_priv *fsi_get_priv(struct snd_pcm_substream *substream) return fsi_get_priv_frm_dai(fsi_get_dai(substream)); } -static u32 fsi_get_info_flags(struct fsi_priv *fsi) -{ - if (!fsi->info) - return 0; - - return fsi->info->flags; -} - static u32 fsi_get_port_shift(struct fsi_priv *fsi, struct fsi_stream *io) { int is_play = fsi_stream_is_play(fsi, io); @@ -1543,7 +1534,6 @@ static int fsi_hw_startup(struct fsi_priv *fsi, struct fsi_stream *io, struct device *dev) { - u32 flags = fsi_get_info_flags(fsi); u32 data = 0; /* clock setting */ @@ -1560,19 +1550,6 @@ static int fsi_hw_startup(struct fsi_priv *fsi, data |= (1 << 4); if (fsi_is_clk_master(fsi)) data <<= 8; - /* FIXME - * - * SH_FSI_xxx_INV style will be removed - */ - if (SH_FSI_LRM_INV & flags) - data |= 1 << 12; - if (SH_FSI_BRM_INV & flags) - data |= 1 << 8; - if (SH_FSI_LRS_INV & flags) - data |= 1 << 4; - if (SH_FSI_BRS_INV & flags) - data |= 1 << 0; - fsi_reg_write(fsi, CKG2, data); /* spdif ? */ @@ -1988,7 +1965,6 @@ static int fsi_probe(struct platform_device *pdev) fsi = &master->fsia; fsi->base = master->base; fsi->master = master; - fsi->info = pinfo; fsi_port_info_init(fsi, pinfo); fsi_handler_init(fsi, pinfo); ret = fsi_stream_probe(fsi, &pdev->dev); @@ -2002,7 +1978,6 @@ static int fsi_probe(struct platform_device *pdev) fsi = &master->fsib; fsi->base = master->base + 0x40; fsi->master = master; - fsi->info = pinfo; fsi_port_info_init(fsi, pinfo); fsi_handler_init(fsi, pinfo); ret = fsi_stream_probe(fsi, &pdev->dev); -- cgit v1.2.3 From 919ad49c214adcc80578c7a02efd3fe8460e0797 Mon Sep 17 00:00:00 2001 From: Lucas Stach Date: Thu, 20 Dec 2012 00:17:33 +0100 Subject: ASoC: tegra: add function to set ac97 rate AC97 uses a fixed rate, unrelated to the sample rate. Add a function to make the setup more trivial. Signed-off-by: Lucas Stach Acked-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_asoc_utils.c | 53 ++++++++++++++++++++++++++++++++++++++ sound/soc/tegra/tegra_asoc_utils.h | 1 + 2 files changed, 54 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/tegra/tegra_asoc_utils.c b/sound/soc/tegra/tegra_asoc_utils.c index 6872c77a119..ba419f86384 100644 --- a/sound/soc/tegra/tegra_asoc_utils.c +++ b/sound/soc/tegra/tegra_asoc_utils.c @@ -112,6 +112,59 @@ int tegra_asoc_utils_set_rate(struct tegra_asoc_utils_data *data, int srate, } EXPORT_SYMBOL_GPL(tegra_asoc_utils_set_rate); +int tegra_asoc_utils_set_ac97_rate(struct tegra_asoc_utils_data *data) +{ + const int pll_rate = 73728000; + const int ac97_rate = 24576000; + int err; + + clk_disable_unprepare(data->clk_cdev1); + clk_disable_unprepare(data->clk_pll_a_out0); + clk_disable_unprepare(data->clk_pll_a); + + /* + * AC97 rate is fixed at 24.576MHz and is used for both the host + * controller and the external codec + */ + err = clk_set_rate(data->clk_pll_a, pll_rate); + if (err) { + dev_err(data->dev, "Can't set pll_a rate: %d\n", err); + return err; + } + + err = clk_set_rate(data->clk_pll_a_out0, ac97_rate); + if (err) { + dev_err(data->dev, "Can't set pll_a_out0 rate: %d\n", err); + return err; + } + + /* Don't set cdev1/extern1 rate; it's locked to pll_a_out0 */ + + err = clk_prepare_enable(data->clk_pll_a); + if (err) { + dev_err(data->dev, "Can't enable pll_a: %d\n", err); + return err; + } + + err = clk_prepare_enable(data->clk_pll_a_out0); + if (err) { + dev_err(data->dev, "Can't enable pll_a_out0: %d\n", err); + return err; + } + + err = clk_prepare_enable(data->clk_cdev1); + if (err) { + dev_err(data->dev, "Can't enable cdev1: %d\n", err); + return err; + } + + data->set_baseclock = pll_rate; + data->set_mclk = ac97_rate; + + return 0; +} +EXPORT_SYMBOL_GPL(tegra_asoc_utils_set_ac97_rate); + int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data, struct device *dev) { diff --git a/sound/soc/tegra/tegra_asoc_utils.h b/sound/soc/tegra/tegra_asoc_utils.h index 44db1dbb8f2..974c9f8830f 100644 --- a/sound/soc/tegra/tegra_asoc_utils.h +++ b/sound/soc/tegra/tegra_asoc_utils.h @@ -43,6 +43,7 @@ struct tegra_asoc_utils_data { int tegra_asoc_utils_set_rate(struct tegra_asoc_utils_data *data, int srate, int mclk); +int tegra_asoc_utils_set_ac97_rate(struct tegra_asoc_utils_data *data); int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data, struct device *dev); void tegra_asoc_utils_fini(struct tegra_asoc_utils_data *data); -- cgit v1.2.3 From 15fab585070ebdd6b31880b3a9a848389d302dd2 Mon Sep 17 00:00:00 2001 From: Lucas Stach Date: Thu, 20 Dec 2012 00:17:32 +0100 Subject: ASoC: tegra: setup DAP3<->DAC3 connection by default This connection is used by the AC97 controller. Signed-off-by: Lucas Stach Acked-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra20_das.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/tegra/tegra20_das.c b/sound/soc/tegra/tegra20_das.c index 65431848387..e72392927bd 100644 --- a/sound/soc/tegra/tegra20_das.c +++ b/sound/soc/tegra/tegra20_das.c @@ -191,6 +191,19 @@ static int tegra20_das_probe(struct platform_device *pdev) goto err; } + ret = tegra20_das_connect_dap_to_dac(TEGRA20_DAS_DAP_ID_3, + TEGRA20_DAS_DAP_SEL_DAC3); + if (ret) { + dev_err(&pdev->dev, "Can't set up DAS DAP connection\n"); + goto err; + } + ret = tegra20_das_connect_dac_to_dap(TEGRA20_DAS_DAC_ID_3, + TEGRA20_DAS_DAC_SEL_DAP3); + if (ret) { + dev_err(&pdev->dev, "Can't set up DAS DAC connection\n"); + goto err; + } + platform_set_drvdata(pdev, das); return 0; -- cgit v1.2.3 From 8a47ca957a6bf86f81893edd9401b91c00be4848 Mon Sep 17 00:00:00 2001 From: Tejun Heo Date: Fri, 21 Dec 2012 17:57:03 -0800 Subject: ASoC: wm8350: don't use [delayed_]work_pending() There's no need to test whether a (delayed) work item in pending before queueing, flushing or cancelling it. Most uses are unnecessary and quite a few of them are buggy. Remove unnecessary pending tests from wm8350. Only compile tested. Signed-off-by: Tejun Heo Signed-off-by: Mark Brown --- sound/soc/codecs/wm8350.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index fb92fb47d63..ec0efc1443b 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -283,18 +283,16 @@ static int pga_event(struct snd_soc_dapm_widget *w, out->ramp = WM8350_RAMP_UP; out->active = 1; - if (!delayed_work_pending(&codec->dapm.delayed_work)) - schedule_delayed_work(&codec->dapm.delayed_work, - msecs_to_jiffies(1)); + schedule_delayed_work(&codec->dapm.delayed_work, + msecs_to_jiffies(1)); break; case SND_SOC_DAPM_PRE_PMD: out->ramp = WM8350_RAMP_DOWN; out->active = 0; - if (!delayed_work_pending(&codec->dapm.delayed_work)) - schedule_delayed_work(&codec->dapm.delayed_work, - msecs_to_jiffies(1)); + schedule_delayed_work(&codec->dapm.delayed_work, + msecs_to_jiffies(1)); break; } -- cgit v1.2.3 From e958f8b806c3953fcadce3929400638ae0dc796f Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 25 Dec 2012 13:28:26 +0800 Subject: ASoC: cs42l52: Convert to devm_input_allocate_device() Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l52.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 99bb1c69499..73a804931cf 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -1040,7 +1040,7 @@ static void cs42l52_init_beep(struct snd_soc_codec *codec) struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); int ret; - cs42l52->beep = input_allocate_device(); + cs42l52->beep = devm_input_allocate_device(codec->dev); if (!cs42l52->beep) { dev_err(codec->dev, "Failed to allocate beep device\n"); return; @@ -1061,7 +1061,6 @@ static void cs42l52_init_beep(struct snd_soc_codec *codec) ret = input_register_device(cs42l52->beep); if (ret != 0) { - input_free_device(cs42l52->beep); cs42l52->beep = NULL; dev_err(codec->dev, "Failed to register beep device\n"); } @@ -1078,7 +1077,6 @@ static void cs42l52_free_beep(struct snd_soc_codec *codec) struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); device_remove_file(codec->dev, &dev_attr_beep); - input_unregister_device(cs42l52->beep); cancel_work_sync(&cs42l52->beep_work); cs42l52->beep = NULL; -- cgit v1.2.3 From 5f3d25c08dee44a40229f1f9e8934f3217478a67 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Mon, 24 Dec 2012 15:55:37 -0200 Subject: ASoC: wm8985: Refactor set_pll code to avoid gcc warnings Refactor set_pll code to avoid the following warnings: sound/soc/codecs/wm8985.c:852:50: warning: 'pll_div.k' may be used uninitialized in this function sound/soc/codecs/wm8985.c:849:9: warning: 'pll_div.n' may be used uninitialized in this function sound/soc/codecs/wm8985.c:848:23: warning: 'pll_div.div2' may be used uninitialized in this function Do the same as in commit 86ce6c9a (ASoC: WM8804: Refactor set_pll code to avoid GCC warnings). Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/wm8985.c | 43 ++++++++++++++++++++----------------------- 1 file changed, 20 insertions(+), 23 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c index ab3782657ac..dd6ce3bc01c 100644 --- a/sound/soc/codecs/wm8985.c +++ b/sound/soc/codecs/wm8985.c @@ -830,33 +830,30 @@ static int wm8985_set_pll(struct snd_soc_dai *dai, int pll_id, struct pll_div pll_div; codec = dai->codec; - if (freq_in && freq_out) { + if (!freq_in || !freq_out) { + /* disable the PLL */ + snd_soc_update_bits(codec, WM8985_POWER_MANAGEMENT_1, + WM8985_PLLEN_MASK, 0); + } else { ret = pll_factors(&pll_div, freq_out * 4 * 2, freq_in); if (ret) return ret; - } - /* disable the PLL before reprogramming it */ - snd_soc_update_bits(codec, WM8985_POWER_MANAGEMENT_1, - WM8985_PLLEN_MASK, 0); - - if (!freq_in || !freq_out) - return 0; - - /* set PLLN and PRESCALE */ - snd_soc_write(codec, WM8985_PLL_N, - (pll_div.div2 << WM8985_PLL_PRESCALE_SHIFT) - | pll_div.n); - /* set PLLK */ - snd_soc_write(codec, WM8985_PLL_K_3, pll_div.k & 0x1ff); - snd_soc_write(codec, WM8985_PLL_K_2, (pll_div.k >> 9) & 0x1ff); - snd_soc_write(codec, WM8985_PLL_K_1, (pll_div.k >> 18)); - /* set the source of the clock to be the PLL */ - snd_soc_update_bits(codec, WM8985_CLOCK_GEN_CONTROL, - WM8985_CLKSEL_MASK, WM8985_CLKSEL); - /* enable the PLL */ - snd_soc_update_bits(codec, WM8985_POWER_MANAGEMENT_1, - WM8985_PLLEN_MASK, WM8985_PLLEN); + /* set PLLN and PRESCALE */ + snd_soc_write(codec, WM8985_PLL_N, + (pll_div.div2 << WM8985_PLL_PRESCALE_SHIFT) + | pll_div.n); + /* set PLLK */ + snd_soc_write(codec, WM8985_PLL_K_3, pll_div.k & 0x1ff); + snd_soc_write(codec, WM8985_PLL_K_2, (pll_div.k >> 9) & 0x1ff); + snd_soc_write(codec, WM8985_PLL_K_1, (pll_div.k >> 18)); + /* set the source of the clock to be the PLL */ + snd_soc_update_bits(codec, WM8985_CLOCK_GEN_CONTROL, + WM8985_CLKSEL_MASK, WM8985_CLKSEL); + /* enable the PLL */ + snd_soc_update_bits(codec, WM8985_POWER_MANAGEMENT_1, + WM8985_PLLEN_MASK, WM8985_PLLEN); + } return 0; } -- cgit v1.2.3 From f89983ef61677afc828ed32b3c42ef5a71eb0686 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 25 Dec 2012 22:52:33 -0800 Subject: ASoC: simple-card: use struct device pointer for dev_xxx() Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index b4b4cab3023..bc050ec8680 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -50,9 +50,10 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) static int asoc_simple_card_probe(struct platform_device *pdev) { struct asoc_simple_card_info *cinfo = pdev->dev.platform_data; + struct device *dev = &pdev->dev; if (!cinfo) { - dev_err(&pdev->dev, "no info for asoc-simple-card\n"); + dev_err(dev, "no info for asoc-simple-card\n"); return -EINVAL; } @@ -62,7 +63,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) !cinfo->codec || !cinfo->platform || !cinfo->codec_dai) { - dev_err(&pdev->dev, "insufficient asoc_simple_card_info settings\n"); + dev_err(dev, "insufficient asoc_simple_card_info settings\n"); return -EINVAL; } -- cgit v1.2.3 From 927a77476ed37080793f0e9d3211359f2d480d4d Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 31 Dec 2012 11:51:42 +0100 Subject: ASoC: twl4030: Correct the support for Voice port In order to be able to use the Voice port of twl4030 three bits need to be handled in VOICE_IF register: VIF_EN: to enable the voice port (needed for both playback and capture) VIF_DIN_EN: Need to be enabled for playback only (input to the codec) VIF_DOUT_EN: Need to be enabled for capture only (output from codec) Use DAPM_SUPPLY for the VIF_EN bit and add DAPM_AIF_IO/OUT widget to handle the playback/capture bit. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 17 ++++++++++++----- 1 file changed, 12 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 63b280b0603..79b2f86f61c 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1306,6 +1306,9 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_DAC("DAC Left2", NULL, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC("DAC Voice", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("VAIFIN", "Voice Playback", 0, + TWL4030_REG_VOICE_IF, 6, 0), + /* Analog bypasses */ SND_SOC_DAPM_SWITCH("Right1 Analog Loopback", SND_SOC_NOPM, 0, 0, &twl4030_dapm_abypassr1_control), @@ -1438,6 +1441,9 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_ADC("ADC Virtual Left2", NULL, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_ADC("ADC Virtual Right2", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("VAIFOUT", "Voice Capture", 0, + TWL4030_REG_VOICE_IF, 5, 0), + /* Analog/Digital mic path selection. TX1 Left/Right: either analog Left/Right or Digimic0 TX2 Left/Right: either analog Left/Right or Digimic1 */ @@ -1477,6 +1483,7 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_MICBIAS("Mic Bias 2", TWL4030_REG_MICBIAS_CTL, 1, 0), SND_SOC_DAPM_MICBIAS("Headset Mic Bias", TWL4030_REG_MICBIAS_CTL, 2, 0), + SND_SOC_DAPM_SUPPLY("VIF Enable", TWL4030_REG_VOICE_IF, 0, 0, NULL, 0), }; static const struct snd_soc_dapm_route intercon[] = { @@ -1485,17 +1492,16 @@ static const struct snd_soc_dapm_route intercon[] = { {"DAC Left1", NULL, "HiFi Playback"}, {"DAC Right2", NULL, "HiFi Playback"}, {"DAC Left2", NULL, "HiFi Playback"}, - {"DAC Voice", NULL, "Voice Playback"}, + {"DAC Voice", NULL, "VAIFIN"}, /* ADC -> Stream mapping */ {"HiFi Capture", NULL, "ADC Virtual Left1"}, {"HiFi Capture", NULL, "ADC Virtual Right1"}, {"HiFi Capture", NULL, "ADC Virtual Left2"}, {"HiFi Capture", NULL, "ADC Virtual Right2"}, - {"Voice Capture", NULL, "ADC Virtual Left1"}, - {"Voice Capture", NULL, "ADC Virtual Right1"}, - {"Voice Capture", NULL, "ADC Virtual Left2"}, - {"Voice Capture", NULL, "ADC Virtual Right2"}, + {"VAIFOUT", NULL, "ADC Virtual Left2"}, + {"VAIFOUT", NULL, "ADC Virtual Right2"}, + {"VAIFOUT", NULL, "VIF Enable"}, {"Digital L1 Playback Mixer", NULL, "DAC Left1"}, {"Digital R1 Playback Mixer", NULL, "DAC Right1"}, @@ -1510,6 +1516,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"DAC Right1", NULL, "AIF Enable"}, {"DAC Left2", NULL, "AIF Enable"}, {"DAC Right1", NULL, "AIF Enable"}, + {"DAC Voice", NULL, "VIF Enable"}, {"Digital R2 Playback Mixer", NULL, "AIF Enable"}, {"Digital L2 Playback Mixer", NULL, "AIF Enable"}, -- cgit v1.2.3 From 01df26edaf4d15e1af3300f8a52b11d3f06c5096 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 31 Dec 2012 11:51:43 +0100 Subject: ASoC: zoom2: No need to configure the Voice port anymore The codec driver takes care of these bits. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/zoom2.c | 14 -------------- 1 file changed, 14 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c index 771bff27ac3..5845d487d27 100644 --- a/sound/soc/omap/zoom2.c +++ b/sound/soc/omap/zoom2.c @@ -110,19 +110,6 @@ static int zoom2_twl4030_init(struct snd_soc_pcm_runtime *rtd) return 0; } -static int zoom2_twl4030_voice_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - unsigned short reg; - - /* Enable voice interface */ - reg = codec->driver->read(codec, TWL4030_REG_VOICE_IF); - reg |= TWL4030_VIF_DIN_EN | TWL4030_VIF_DOUT_EN | TWL4030_VIF_EN; - codec->driver->write(codec, TWL4030_REG_VOICE_IF, reg); - - return 0; -} - /* Digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link zoom2_dai[] = { { @@ -146,7 +133,6 @@ static struct snd_soc_dai_link zoom2_dai[] = { .codec_name = "twl4030-codec", .dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBM_CFM, - .init = zoom2_twl4030_voice_init, .ops = &zoom2_ops, }, }; -- cgit v1.2.3 From 57296cc28cf8aab2e5e63d8f9964718ae29cc229 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 31 Dec 2012 11:51:44 +0100 Subject: ASoC: sdp3430: No need to configure the Voice port anymore The codec driver takes care of these bits. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/sdp3430.c | 15 --------------- 1 file changed, 15 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index b462a2c9385..86e77e9d24c 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -167,20 +167,6 @@ static int sdp3430_twl4030_init(struct snd_soc_pcm_runtime *rtd) return ret; } -static int sdp3430_twl4030_voice_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - unsigned short reg; - - /* Enable voice interface */ - reg = codec->driver->read(codec, TWL4030_REG_VOICE_IF); - reg |= TWL4030_VIF_DIN_EN | TWL4030_VIF_DOUT_EN | TWL4030_VIF_EN; - codec->driver->write(codec, TWL4030_REG_VOICE_IF, reg); - - return 0; -} - - /* Digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link sdp3430_dai[] = { { @@ -204,7 +190,6 @@ static struct snd_soc_dai_link sdp3430_dai[] = { .codec_name = "twl4030-codec", .dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBM_CFM, - .init = sdp3430_twl4030_voice_init, .ops = &sdp3430_ops, }, }; -- cgit v1.2.3 From e04d6e55fe02351b1ac338d9e8082fcc434610ce Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 31 Dec 2012 11:51:45 +0100 Subject: ASoC: twl4030: Convert MICBIAS to SUPPLY widget In order to avoid breakage update the machine drivers at the same time using twl4030: omap3pandora, sdp3430 and zoom2 Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 10 +++++++--- sound/soc/omap/omap3pandora.c | 8 ++++---- sound/soc/omap/sdp3430.c | 14 +++++++------- sound/soc/omap/zoom2.c | 14 +++++++------- 4 files changed, 25 insertions(+), 21 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 79b2f86f61c..5dd7c81623a 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1479,9 +1479,13 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("micbias2 select", TWL4030_REG_MICBIAS_CTL, 6, 0, NULL, 0), - SND_SOC_DAPM_MICBIAS("Mic Bias 1", TWL4030_REG_MICBIAS_CTL, 0, 0), - SND_SOC_DAPM_MICBIAS("Mic Bias 2", TWL4030_REG_MICBIAS_CTL, 1, 0), - SND_SOC_DAPM_MICBIAS("Headset Mic Bias", TWL4030_REG_MICBIAS_CTL, 2, 0), + /* Microphone bias */ + SND_SOC_DAPM_SUPPLY("Mic Bias 1", + TWL4030_REG_MICBIAS_CTL, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Mic Bias 2", + TWL4030_REG_MICBIAS_CTL, 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headset Mic Bias", + TWL4030_REG_MICBIAS_CTL, 2, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("VIF Enable", TWL4030_REG_VOICE_IF, 0, 0, NULL, 0), }; diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index 43d950a79ff..805512f2555 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -144,11 +144,11 @@ static const struct snd_soc_dapm_route omap3pandora_in_map[] = { {"AUXL", NULL, "Line In"}, {"AUXR", NULL, "Line In"}, - {"MAINMIC", NULL, "Mic Bias 1"}, - {"Mic Bias 1", NULL, "Mic (internal)"}, + {"MAINMIC", NULL, "Mic (internal)"}, + {"Mic (internal)", NULL, "Mic Bias 1"}, - {"SUBMIC", NULL, "Mic Bias 2"}, - {"Mic Bias 2", NULL, "Mic (external)"}, + {"SUBMIC", NULL, "Mic (external)"}, + {"Mic (external)", NULL, "Mic Bias 2"}, }; static int omap3pandora_out_init(struct snd_soc_pcm_runtime *rtd) diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index 86e77e9d24c..f2e26512d2a 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -104,19 +104,19 @@ static const struct snd_soc_dapm_widget sdp3430_twl4030_dapm_widgets[] = { }; static const struct snd_soc_dapm_route audio_map[] = { - /* External Mics: MAINMIC, SUBMIC with bias*/ - {"MAINMIC", NULL, "Mic Bias 1"}, - {"SUBMIC", NULL, "Mic Bias 2"}, - {"Mic Bias 1", NULL, "Ext Mic"}, - {"Mic Bias 2", NULL, "Ext Mic"}, + /* External Mics: MAINMIC, SUBMIC with bias */ + {"MAINMIC", NULL, "Ext Mic"}, + {"SUBMIC", NULL, "Ext Mic"}, + {"Ext Mic", NULL, "Mic Bias 1"}, + {"Ext Mic", NULL, "Mic Bias 2"}, /* External Speakers: HFL, HFR */ {"Ext Spk", NULL, "HFL"}, {"Ext Spk", NULL, "HFR"}, /* Headset Mic: HSMIC with bias */ - {"HSMIC", NULL, "Headset Mic Bias"}, - {"Headset Mic Bias", NULL, "Headset Mic"}, + {"HSMIC", NULL, "Headset Mic"}, + {"Headset Mic", NULL, "Headset Mic Bias"}, /* Headset Stereophone (Headphone): HSOL, HSOR */ {"Headset Stereophone", NULL, "HSOL"}, diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c index 5845d487d27..62a6b02d9b3 100644 --- a/sound/soc/omap/zoom2.c +++ b/sound/soc/omap/zoom2.c @@ -69,11 +69,11 @@ static const struct snd_soc_dapm_widget zoom2_twl4030_dapm_widgets[] = { }; static const struct snd_soc_dapm_route audio_map[] = { - /* External Mics: MAINMIC, SUBMIC with bias*/ - {"MAINMIC", NULL, "Mic Bias 1"}, - {"SUBMIC", NULL, "Mic Bias 2"}, - {"Mic Bias 1", NULL, "Ext Mic"}, - {"Mic Bias 2", NULL, "Ext Mic"}, + /* External Mics: MAINMIC, SUBMIC with bias */ + {"MAINMIC", NULL, "Ext Mic"}, + {"SUBMIC", NULL, "Ext Mic"}, + {"Ext Mic", NULL, "Mic Bias 1"}, + {"Ext Mic", NULL, "Mic Bias 2"}, /* External Speakers: HFL, HFR */ {"Ext Spk", NULL, "HFL"}, @@ -84,8 +84,8 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Headset Stereophone", NULL, "HSOR"}, /* Headset Mic: HSMIC with bias */ - {"HSMIC", NULL, "Headset Mic Bias"}, - {"Headset Mic Bias", NULL, "Headset Mic"}, + {"HSMIC", NULL, "Headset Mic"}, + {"Headset Mic", NULL, "Headset Mic Bias"}, /* Aux In: AUXL, AUXR */ {"Aux In", NULL, "AUXL"}, -- cgit v1.2.3 From 5712ded9cf0b6981825af3c86e0059be93e5e17c Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 31 Dec 2012 11:51:46 +0100 Subject: ASoC: twl4030: Configure extmute pinmux when the dedicated pin is in use When HS extmute is enabled without custom GPIO we should configure the mux to allow the pin to be used as extmute signal. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 44 +++++++++++++++++++++++++++++++------------- 1 file changed, 31 insertions(+), 13 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 5dd7c81623a..7bfabe5750e 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -41,6 +41,11 @@ /* Register descriptions are here */ #include +/* TWL4030 PMBR1 Register */ +#define TWL4030_PMBR1_REG 0x0D +/* TWL4030 PMBR1 Register GPIO6 mux bits */ +#define TWL4030_GPIO6_PWM0_MUTE(value) ((value & 0x03) << 2) + /* Shadow register used by the audio driver */ #define TWL4030_REG_SW_SHADOW 0x4A #define TWL4030_CACHEREGNUM (TWL4030_REG_SW_SHADOW + 1) @@ -348,19 +353,32 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) pdata = twl4030_get_pdata(codec); - if (pdata && pdata->hs_extmute && - gpio_is_valid(pdata->hs_extmute_gpio)) { - int ret; - - if (!pdata->hs_extmute_gpio) - dev_warn(codec->dev, - "Extmute GPIO is 0 is this correct?\n"); - - ret = gpio_request_one(pdata->hs_extmute_gpio, - GPIOF_OUT_INIT_LOW, "hs_extmute"); - if (ret) { - dev_err(codec->dev, "Failed to get hs_extmute GPIO\n"); - pdata->hs_extmute_gpio = -1; + if (pdata && pdata->hs_extmute) { + if (gpio_is_valid(pdata->hs_extmute_gpio)) { + int ret; + + if (!pdata->hs_extmute_gpio) + dev_warn(codec->dev, + "Extmute GPIO is 0 is this correct?\n"); + + ret = gpio_request_one(pdata->hs_extmute_gpio, + GPIOF_OUT_INIT_LOW, + "hs_extmute"); + if (ret) { + dev_err(codec->dev, + "Failed to get hs_extmute GPIO\n"); + pdata->hs_extmute_gpio = -1; + } + } else { + u8 pin_mux; + + /* Set TWL4030 GPIO6 as EXTMUTE signal */ + twl_i2c_read_u8(TWL4030_MODULE_INTBR, &pin_mux, + TWL4030_PMBR1_REG); + pin_mux &= ~TWL4030_GPIO6_PWM0_MUTE(0x03); + pin_mux |= TWL4030_GPIO6_PWM0_MUTE(0x02); + twl_i2c_write_u8(TWL4030_MODULE_INTBR, pin_mux, + TWL4030_PMBR1_REG); } } -- cgit v1.2.3 From fff3dd40132d6106d4d2a61e70e782f82394fd17 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 31 Dec 2012 11:51:47 +0100 Subject: ASoC: sdp3430: No need to configure pin mux for extmute The codec driver takes care of this. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/sdp3430.c | 18 ------------------ 1 file changed, 18 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index f2e26512d2a..216cbdd569e 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -24,7 +24,6 @@ #include #include -#include #include #include #include @@ -32,20 +31,12 @@ #include #include -#include -/* Register descriptions for twl4030 codec part */ -#include #include #include "omap-mcbsp.h" #include "omap-pcm.h" -/* TWL4030 PMBR1 Register */ -#define TWL4030_INTBR_PMBR1 0x0D -/* TWL4030 PMBR1 Register GPIO6 mux bit */ -#define TWL4030_GPIO6_PWM0_MUTE(value) (value << 2) - static struct snd_soc_card snd_soc_sdp3430; static int sdp3430_hw_params(struct snd_pcm_substream *substream, @@ -212,7 +203,6 @@ static struct platform_device *sdp3430_snd_device; static int __init sdp3430_soc_init(void) { int ret; - u8 pin_mux; if (!machine_is_omap_3430sdp()) return -ENODEV; @@ -226,14 +216,6 @@ static int __init sdp3430_soc_init(void) platform_set_drvdata(sdp3430_snd_device, &snd_soc_sdp3430); - /* Set TWL4030 GPIO6 as EXTMUTE signal */ - twl_i2c_read_u8(TWL4030_MODULE_INTBR, &pin_mux, - TWL4030_INTBR_PMBR1); - pin_mux &= ~TWL4030_GPIO6_PWM0_MUTE(0x03); - pin_mux |= TWL4030_GPIO6_PWM0_MUTE(0x02); - twl_i2c_write_u8(TWL4030_MODULE_INTBR, pin_mux, - TWL4030_INTBR_PMBR1); - ret = platform_device_add(sdp3430_snd_device); if (ret) goto err1; -- cgit v1.2.3 From bd0b286e838ef1ca19bbe1cb55f0ec7e0135de1f Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 31 Dec 2012 11:51:48 +0100 Subject: ASoC: omap-twl4030: Add support for routing, voice port and jack detect Update the common machine driver to support more boards including Zoom2 and SDP3430. - Support for voice port of twl4030 - HS jack plug detection support - The audio routing can be fine tuned via pdata or via provided routing table from DT. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/Kconfig | 2 + sound/soc/omap/omap-twl4030.c | 204 +++++++++++++++++++++++++++++++++++++++++- 2 files changed, 204 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 7048137f9a3..e8d2a2f983b 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -91,6 +91,8 @@ config SND_OMAP_SOC_OMAP_TWL4030 - Gumstix Overo or CompuLab CM-T35/CM-T3730 - IGEP v2 - OMAP3EVM + - SDP3430 + - Zoom2 config SND_OMAP_SOC_OMAP_ABE_TWL6040 tristate "SoC Audio support for OMAP boards using ABE and twl6040 codec" diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c index 4541d28b531..fd98509d0f4 100644 --- a/sound/soc/omap/omap-twl4030.c +++ b/sound/soc/omap/omap-twl4030.c @@ -11,6 +11,8 @@ * omap3evm (Author: Anuj Aggarwal ) * overo (Author: Steve Sakoman ) * igep0020 (Author: Enric Balletbo i Serra ) + * zoom2 (Author: Misael Lopez Cruz ) + * sdp3430 (Author: Misael Lopez Cruz ) * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License @@ -32,14 +34,22 @@ #include #include #include +#include +#include #include #include #include +#include #include "omap-mcbsp.h" #include "omap-pcm.h" +struct omap_twl4030 { + int jack_detect; /* board can detect jack events */ + struct snd_soc_jack hs_jack; +}; + static int omap_twl4030_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -87,17 +97,164 @@ static struct snd_soc_ops omap_twl4030_ops = { .hw_params = omap_twl4030_hw_params, }; +static const struct snd_soc_dapm_widget dapm_widgets[] = { + SND_SOC_DAPM_SPK("Earpiece Spk", NULL), + SND_SOC_DAPM_SPK("Handsfree Spk", NULL), + SND_SOC_DAPM_HP("Headset Stereophone", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_SPK("Carkit Spk", NULL), + + SND_SOC_DAPM_MIC("Main Mic", NULL), + SND_SOC_DAPM_MIC("Sub Mic", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Carkit Mic", NULL), + SND_SOC_DAPM_MIC("Digital0 Mic", NULL), + SND_SOC_DAPM_MIC("Digital1 Mic", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* Headset Stereophone: HSOL, HSOR */ + {"Headset Stereophone", NULL, "HSOL"}, + {"Headset Stereophone", NULL, "HSOR"}, + /* External Speakers: HFL, HFR */ + {"Handsfree Spk", NULL, "HFL"}, + {"Handsfree Spk", NULL, "HFR"}, + /* External Speakers: PredrivL, PredrivR */ + {"Ext Spk", NULL, "PREDRIVEL"}, + {"Ext Spk", NULL, "PREDRIVER"}, + /* Carkit speakers: CARKITL, CARKITR */ + {"Carkit Spk", NULL, "CARKITL"}, + {"Carkit Spk", NULL, "CARKITR"}, + /* Earpiece */ + {"Earpiece Spk", NULL, "EARPIECE"}, + + /* External Mics: MAINMIC, SUBMIC with bias */ + {"MAINMIC", NULL, "Main Mic"}, + {"Main Mic", NULL, "Mic Bias 1"}, + {"SUBMIC", NULL, "Sub Mic"}, + {"Sub Mic", NULL, "Mic Bias 2"}, + /* Headset Mic: HSMIC with bias */ + {"HSMIC", NULL, "Headset Mic"}, + {"Headset Mic", NULL, "Headset Mic Bias"}, + /* Digital Mics: DIGIMIC0, DIGIMIC1 with bias */ + {"DIGIMIC0", NULL, "Digital0 Mic"}, + {"Digital0 Mic", NULL, "Mic Bias 1"}, + {"DIGIMIC1", NULL, "Digital1 Mic"}, + {"Digital1 Mic", NULL, "Mic Bias 2"}, + /* Carkit In: CARKITMIC */ + {"CARKITMIC", NULL, "Carkit Mic"}, + /* Aux In: AUXL, AUXR */ + {"AUXL", NULL, "Line In"}, + {"AUXR", NULL, "Line In"}, +}; + +/* Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin hs_jack_pins[] = { + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headset Stereophone", + .mask = SND_JACK_HEADPHONE, + }, +}; + +/* Headset jack detection gpios */ +static struct snd_soc_jack_gpio hs_jack_gpios[] = { + { + .name = "hsdet-gpio", + .report = SND_JACK_HEADSET, + .debounce_time = 200, + }, +}; + +static inline void twl4030_disconnect_pin(struct snd_soc_dapm_context *dapm, + int connected, char *pin) +{ + if (!connected) + snd_soc_dapm_disable_pin(dapm, pin); +} + +static int omap_twl4030_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_card *card = codec->card; + struct snd_soc_dapm_context *dapm = &codec->dapm; + struct omap_tw4030_pdata *pdata = dev_get_platdata(card->dev); + struct omap_twl4030 *priv = snd_soc_card_get_drvdata(card); + int ret = 0; + + /* Headset jack detection only if it is supported */ + if (priv->jack_detect > 0) { + hs_jack_gpios[0].gpio = priv->jack_detect; + + ret = snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET, + &priv->hs_jack); + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(&priv->hs_jack, + ARRAY_SIZE(hs_jack_pins), + hs_jack_pins); + if (ret) + return ret; + + ret = snd_soc_jack_add_gpios(&priv->hs_jack, + ARRAY_SIZE(hs_jack_gpios), + hs_jack_gpios); + if (ret) + return ret; + } + + /* + * NULL pdata means we booted with DT. In this case the routing is + * provided and the card is fully routed, no need to mark pins. + */ + if (!pdata || !pdata->custom_routing) + return ret; + + /* Disable not connected paths if not used */ + twl4030_disconnect_pin(dapm, pdata->has_ear, "Earpiece Spk"); + twl4030_disconnect_pin(dapm, pdata->has_hf, "Handsfree Spk"); + twl4030_disconnect_pin(dapm, pdata->has_hs, "Headset Stereophone"); + twl4030_disconnect_pin(dapm, pdata->has_predriv, "Ext Spk"); + twl4030_disconnect_pin(dapm, pdata->has_carkit, "Carkit Spk"); + + twl4030_disconnect_pin(dapm, pdata->has_mainmic, "Main Mic"); + twl4030_disconnect_pin(dapm, pdata->has_submic, "Sub Mic"); + twl4030_disconnect_pin(dapm, pdata->has_hsmic, "Headset Mic"); + twl4030_disconnect_pin(dapm, pdata->has_carkitmic, "Carkit Mic"); + twl4030_disconnect_pin(dapm, pdata->has_digimic0, "Digital0 Mic"); + twl4030_disconnect_pin(dapm, pdata->has_digimic1, "Digital1 Mic"); + twl4030_disconnect_pin(dapm, pdata->has_linein, "Line In"); + + return ret; +} + /* Digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link omap_twl4030_dai_links[] = { { - .name = "TWL4030", - .stream_name = "TWL4030", + .name = "TWL4030 HiFi", + .stream_name = "TWL4030 HiFi", .cpu_dai_name = "omap-mcbsp.2", .codec_dai_name = "twl4030-hifi", .platform_name = "omap-pcm-audio", .codec_name = "twl4030-codec", + .init = omap_twl4030_init, .ops = &omap_twl4030_ops, }, + { + .name = "TWL4030 Voice", + .stream_name = "TWL4030 Voice", + .cpu_dai_name = "omap-mcbsp.3", + .codec_dai_name = "twl4030-voice", + .platform_name = "omap-pcm-audio", + .codec_name = "twl4030-codec", + .dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBM_CFM, + }, }; /* Audio machine driver */ @@ -105,6 +262,11 @@ static struct snd_soc_card omap_twl4030_card = { .owner = THIS_MODULE, .dai_link = omap_twl4030_dai_links, .num_links = ARRAY_SIZE(omap_twl4030_dai_links), + + .dapm_widgets = dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; static int omap_twl4030_probe(struct platform_device *pdev) @@ -112,12 +274,18 @@ static int omap_twl4030_probe(struct platform_device *pdev) struct omap_tw4030_pdata *pdata = dev_get_platdata(&pdev->dev); struct device_node *node = pdev->dev.of_node; struct snd_soc_card *card = &omap_twl4030_card; + struct omap_twl4030 *priv; int ret = 0; card->dev = &pdev->dev; + priv = devm_kzalloc(&pdev->dev, sizeof(struct omap_twl4030), GFP_KERNEL); + if (priv == NULL) + return -ENOMEM; + if (node) { struct device_node *dai_node; + struct property *prop; if (snd_soc_of_parse_card_name(card, "ti,model")) { dev_err(&pdev->dev, "Card name is not provided\n"); @@ -132,6 +300,27 @@ static int omap_twl4030_probe(struct platform_device *pdev) omap_twl4030_dai_links[0].cpu_dai_name = NULL; omap_twl4030_dai_links[0].cpu_of_node = dai_node; + dai_node = of_parse_phandle(node, "ti,mcbsp-voice", 0); + if (!dai_node) { + card->num_links = 1; + } else { + omap_twl4030_dai_links[1].cpu_dai_name = NULL; + omap_twl4030_dai_links[1].cpu_of_node = dai_node; + } + + priv->jack_detect = of_get_named_gpio(node, + "ti,jack-det-gpio", 0); + + /* Optional: audio routing can be provided */ + prop = of_find_property(node, "ti,audio-routing", NULL); + if (prop) { + ret = snd_soc_of_parse_audio_routing(card, + "ti,audio-routing"); + if (ret) + return ret; + + card->fully_routed = 1; + } } else if (pdata) { if (pdata->card_name) { card->name = pdata->card_name; @@ -139,11 +328,17 @@ static int omap_twl4030_probe(struct platform_device *pdev) dev_err(&pdev->dev, "Card name is not provided\n"); return -ENODEV; } + + if (!pdata->voice_connected) + card->num_links = 1; + + priv->jack_detect = pdata->jack_detect; } else { dev_err(&pdev->dev, "Missing pdata\n"); return -ENODEV; } + snd_soc_card_set_drvdata(card, priv); ret = snd_soc_register_card(card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", @@ -157,7 +352,12 @@ static int omap_twl4030_probe(struct platform_device *pdev) static int omap_twl4030_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); + struct omap_twl4030 *priv = snd_soc_card_get_drvdata(card); + if (priv->jack_detect > 0) + snd_soc_jack_free_gpios(&priv->hs_jack, + ARRAY_SIZE(hs_jack_gpios), + hs_jack_gpios); snd_soc_unregister_card(card); return 0; -- cgit v1.2.3 From 57d61b9d2d7bc618b7b28a46310d9c2f086f8f51 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 31 Dec 2012 11:51:49 +0100 Subject: ASoC: OMAP: Remove obsolete machine drivers for Zoom2 and SDP3430 These boards are using the common omap-twl4030 machine driver, no need for separate machine drivers anymore. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/Kconfig | 17 ---- sound/soc/omap/Makefile | 4 - sound/soc/omap/sdp3430.c | 245 ----------------------------------------------- sound/soc/omap/zoom2.c | 193 ------------------------------------- 4 files changed, 459 deletions(-) delete mode 100644 sound/soc/omap/sdp3430.c delete mode 100644 sound/soc/omap/zoom2.c (limited to 'sound/soc') diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index e8d2a2f983b..60259f2f3f2 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -70,15 +70,6 @@ config SND_OMAP_SOC_AM3517EVM Say Y if you want to add support for SoC audio on the OMAP3517 / AM3517 EVM. -config SND_OMAP_SOC_SDP3430 - tristate "SoC Audio support for Texas Instruments SDP3430" - depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_3430SDP - select SND_OMAP_SOC_MCBSP - select SND_SOC_TWL4030 - help - Say Y if you want to add support for SoC audio on Texas Instruments - SDP3430. - config SND_OMAP_SOC_OMAP_TWL4030 tristate "SoC Audio support for TI SoC based boards with twl4030 codec" depends on TWL4030_CORE && SND_OMAP_SOC @@ -125,11 +116,3 @@ config SND_OMAP_SOC_OMAP3_PANDORA select SND_SOC_TWL4030 help Say Y if you want to add support for SoC audio on the OMAP3 Pandora. - -config SND_OMAP_SOC_ZOOM2 - tristate "SoC Audio support for Zoom2" - depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_ZOOM2 - select SND_OMAP_SOC_MCBSP - select SND_SOC_TWL4030 - help - Say Y if you want to add support for Soc audio on Zoom2 board. diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 19637e55ea4..2b225945359 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -17,11 +17,9 @@ snd-soc-rx51-objs := rx51.o snd-soc-ams-delta-objs := ams-delta.o snd-soc-osk5912-objs := osk5912.o snd-soc-am3517evm-objs := am3517evm.o -snd-soc-sdp3430-objs := sdp3430.o snd-soc-omap-abe-twl6040-objs := omap-abe-twl6040.o snd-soc-omap-twl4030-objs := omap-twl4030.o snd-soc-omap3pandora-objs := omap3pandora.o -snd-soc-zoom2-objs := zoom2.o snd-soc-omap-hdmi-card-objs := omap-hdmi-card.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o @@ -30,9 +28,7 @@ obj-$(CONFIG_SND_OMAP_SOC_AMS_DELTA) += snd-soc-ams-delta.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o obj-$(CONFIG_SND_OMAP_SOC_OMAP2EVM) += snd-soc-omap2evm.o obj-$(CONFIG_SND_OMAP_SOC_AM3517EVM) += snd-soc-am3517evm.o -obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o obj-$(CONFIG_SND_OMAP_SOC_OMAP_ABE_TWL6040) += snd-soc-omap-abe-twl6040.o obj-$(CONFIG_SND_OMAP_SOC_OMAP_TWL4030) += snd-soc-omap-twl4030.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o -obj-$(CONFIG_SND_OMAP_SOC_ZOOM2) += snd-soc-zoom2.o obj-$(CONFIG_SND_OMAP_SOC_OMAP_HDMI) += snd-soc-omap-hdmi-card.o diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c deleted file mode 100644 index 216cbdd569e..00000000000 --- a/sound/soc/omap/sdp3430.c +++ /dev/null @@ -1,245 +0,0 @@ -/* - * sdp3430.c -- SoC audio for TI OMAP3430 SDP - * - * Author: Misael Lopez Cruz - * - * Based on: - * Author: Steve Sakoman - * - * 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 program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA - * 02110-1301 USA - * - */ - -#include -#include -#include -#include -#include -#include - -#include -#include - -#include - -#include "omap-mcbsp.h" -#include "omap-pcm.h" - -static struct snd_soc_card snd_soc_sdp3430; - -static int sdp3430_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_dai *codec_dai = rtd->codec_dai; - int ret; - - /* Set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, - SND_SOC_CLOCK_IN); - if (ret < 0) { - printk(KERN_ERR "can't set codec system clock\n"); - return ret; - } - - return 0; -} - -static struct snd_soc_ops sdp3430_ops = { - .hw_params = sdp3430_hw_params, -}; - -/* Headset jack */ -static struct snd_soc_jack hs_jack; - -/* Headset jack detection DAPM pins */ -static struct snd_soc_jack_pin hs_jack_pins[] = { - { - .pin = "Headset Mic", - .mask = SND_JACK_MICROPHONE, - }, - { - .pin = "Headset Stereophone", - .mask = SND_JACK_HEADPHONE, - }, -}; - -/* Headset jack detection gpios */ -static struct snd_soc_jack_gpio hs_jack_gpios[] = { - { - .gpio = (OMAP_MAX_GPIO_LINES + 2), - .name = "hsdet-gpio", - .report = SND_JACK_HEADSET, - .debounce_time = 200, - }, -}; - -/* SDP3430 machine DAPM */ -static const struct snd_soc_dapm_widget sdp3430_twl4030_dapm_widgets[] = { - SND_SOC_DAPM_MIC("Ext Mic", NULL), - SND_SOC_DAPM_SPK("Ext Spk", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_HP("Headset Stereophone", NULL), -}; - -static const struct snd_soc_dapm_route audio_map[] = { - /* External Mics: MAINMIC, SUBMIC with bias */ - {"MAINMIC", NULL, "Ext Mic"}, - {"SUBMIC", NULL, "Ext Mic"}, - {"Ext Mic", NULL, "Mic Bias 1"}, - {"Ext Mic", NULL, "Mic Bias 2"}, - - /* External Speakers: HFL, HFR */ - {"Ext Spk", NULL, "HFL"}, - {"Ext Spk", NULL, "HFR"}, - - /* Headset Mic: HSMIC with bias */ - {"HSMIC", NULL, "Headset Mic"}, - {"Headset Mic", NULL, "Headset Mic Bias"}, - - /* Headset Stereophone (Headphone): HSOL, HSOR */ - {"Headset Stereophone", NULL, "HSOL"}, - {"Headset Stereophone", NULL, "HSOR"}, -}; - -static int sdp3430_twl4030_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret; - - /* SDP3430 connected pins */ - snd_soc_dapm_enable_pin(dapm, "Ext Mic"); - snd_soc_dapm_enable_pin(dapm, "Ext Spk"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic"); - snd_soc_dapm_disable_pin(dapm, "Headset Stereophone"); - - /* TWL4030 not connected pins */ - snd_soc_dapm_nc_pin(dapm, "AUXL"); - snd_soc_dapm_nc_pin(dapm, "AUXR"); - snd_soc_dapm_nc_pin(dapm, "CARKITMIC"); - snd_soc_dapm_nc_pin(dapm, "DIGIMIC0"); - snd_soc_dapm_nc_pin(dapm, "DIGIMIC1"); - - snd_soc_dapm_nc_pin(dapm, "OUTL"); - snd_soc_dapm_nc_pin(dapm, "OUTR"); - snd_soc_dapm_nc_pin(dapm, "EARPIECE"); - snd_soc_dapm_nc_pin(dapm, "PREDRIVEL"); - snd_soc_dapm_nc_pin(dapm, "PREDRIVER"); - snd_soc_dapm_nc_pin(dapm, "CARKITL"); - snd_soc_dapm_nc_pin(dapm, "CARKITR"); - - /* Headset jack detection */ - ret = snd_soc_jack_new(codec, "Headset Jack", - SND_JACK_HEADSET, &hs_jack); - if (ret) - return ret; - - ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), - hs_jack_pins); - if (ret) - return ret; - - ret = snd_soc_jack_add_gpios(&hs_jack, ARRAY_SIZE(hs_jack_gpios), - hs_jack_gpios); - - return ret; -} - -/* Digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link sdp3430_dai[] = { - { - .name = "TWL4030 I2S", - .stream_name = "TWL4030 Audio", - .cpu_dai_name = "omap-mcbsp.2", - .codec_dai_name = "twl4030-hifi", - .platform_name = "omap-pcm-audio", - .codec_name = "twl4030-codec", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, - .init = sdp3430_twl4030_init, - .ops = &sdp3430_ops, - }, - { - .name = "TWL4030 PCM", - .stream_name = "TWL4030 Voice", - .cpu_dai_name = "omap-mcbsp.3", - .codec_dai_name = "twl4030-voice", - .platform_name = "omap-pcm-audio", - .codec_name = "twl4030-codec", - .dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF | - SND_SOC_DAIFMT_CBM_CFM, - .ops = &sdp3430_ops, - }, -}; - -/* Audio machine driver */ -static struct snd_soc_card snd_soc_sdp3430 = { - .name = "SDP3430", - .owner = THIS_MODULE, - .dai_link = sdp3430_dai, - .num_links = ARRAY_SIZE(sdp3430_dai), - - .dapm_widgets = sdp3430_twl4030_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(sdp3430_twl4030_dapm_widgets), - .dapm_routes = audio_map, - .num_dapm_routes = ARRAY_SIZE(audio_map), -}; - -static struct platform_device *sdp3430_snd_device; - -static int __init sdp3430_soc_init(void) -{ - int ret; - - if (!machine_is_omap_3430sdp()) - return -ENODEV; - printk(KERN_INFO "SDP3430 SoC init\n"); - - sdp3430_snd_device = platform_device_alloc("soc-audio", -1); - if (!sdp3430_snd_device) { - printk(KERN_ERR "Platform device allocation failed\n"); - return -ENOMEM; - } - - platform_set_drvdata(sdp3430_snd_device, &snd_soc_sdp3430); - - ret = platform_device_add(sdp3430_snd_device); - if (ret) - goto err1; - - return 0; - -err1: - printk(KERN_ERR "Unable to add platform device\n"); - platform_device_put(sdp3430_snd_device); - - return ret; -} -module_init(sdp3430_soc_init); - -static void __exit sdp3430_soc_exit(void) -{ - snd_soc_jack_free_gpios(&hs_jack, ARRAY_SIZE(hs_jack_gpios), - hs_jack_gpios); - - platform_device_unregister(sdp3430_snd_device); -} -module_exit(sdp3430_soc_exit); - -MODULE_AUTHOR("Misael Lopez Cruz "); -MODULE_DESCRIPTION("ALSA SoC SDP3430"); -MODULE_LICENSE("GPL"); - diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c deleted file mode 100644 index 62a6b02d9b3..00000000000 --- a/sound/soc/omap/zoom2.c +++ /dev/null @@ -1,193 +0,0 @@ -/* - * zoom2.c -- SoC audio for Zoom2 - * - * Author: Misael Lopez Cruz - * - * 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 program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA - * 02110-1301 USA - * - */ - -#include -#include -#include -#include -#include -#include - -#include -#include -#include - -/* Register descriptions for twl4030 codec part */ -#include -#include - -#include "omap-mcbsp.h" -#include "omap-pcm.h" - -static int zoom2_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_dai *codec_dai = rtd->codec_dai; - int ret; - - /* Set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, - SND_SOC_CLOCK_IN); - if (ret < 0) { - printk(KERN_ERR "can't set codec system clock\n"); - return ret; - } - - return 0; -} - -static struct snd_soc_ops zoom2_ops = { - .hw_params = zoom2_hw_params, -}; - -/* Zoom2 machine DAPM */ -static const struct snd_soc_dapm_widget zoom2_twl4030_dapm_widgets[] = { - SND_SOC_DAPM_MIC("Ext Mic", NULL), - SND_SOC_DAPM_SPK("Ext Spk", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_HP("Headset Stereophone", NULL), - SND_SOC_DAPM_LINE("Aux In", NULL), -}; - -static const struct snd_soc_dapm_route audio_map[] = { - /* External Mics: MAINMIC, SUBMIC with bias */ - {"MAINMIC", NULL, "Ext Mic"}, - {"SUBMIC", NULL, "Ext Mic"}, - {"Ext Mic", NULL, "Mic Bias 1"}, - {"Ext Mic", NULL, "Mic Bias 2"}, - - /* External Speakers: HFL, HFR */ - {"Ext Spk", NULL, "HFL"}, - {"Ext Spk", NULL, "HFR"}, - - /* Headset Stereophone: HSOL, HSOR */ - {"Headset Stereophone", NULL, "HSOL"}, - {"Headset Stereophone", NULL, "HSOR"}, - - /* Headset Mic: HSMIC with bias */ - {"HSMIC", NULL, "Headset Mic"}, - {"Headset Mic", NULL, "Headset Mic Bias"}, - - /* Aux In: AUXL, AUXR */ - {"Aux In", NULL, "AUXL"}, - {"Aux In", NULL, "AUXR"}, -}; - -static int zoom2_twl4030_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - - /* TWL4030 not connected pins */ - snd_soc_dapm_nc_pin(dapm, "CARKITMIC"); - snd_soc_dapm_nc_pin(dapm, "DIGIMIC0"); - snd_soc_dapm_nc_pin(dapm, "DIGIMIC1"); - snd_soc_dapm_nc_pin(dapm, "EARPIECE"); - snd_soc_dapm_nc_pin(dapm, "PREDRIVEL"); - snd_soc_dapm_nc_pin(dapm, "PREDRIVER"); - snd_soc_dapm_nc_pin(dapm, "CARKITL"); - snd_soc_dapm_nc_pin(dapm, "CARKITR"); - - return 0; -} - -/* Digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link zoom2_dai[] = { - { - .name = "TWL4030 I2S", - .stream_name = "TWL4030 Audio", - .cpu_dai_name = "omap-mcbsp.2", - .codec_dai_name = "twl4030-hifi", - .platform_name = "omap-pcm-audio", - .codec_name = "twl4030-codec", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, - .init = zoom2_twl4030_init, - .ops = &zoom2_ops, - }, - { - .name = "TWL4030 PCM", - .stream_name = "TWL4030 Voice", - .cpu_dai_name = "omap-mcbsp.3", - .codec_dai_name = "twl4030-voice", - .platform_name = "omap-pcm-audio", - .codec_name = "twl4030-codec", - .dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF | - SND_SOC_DAIFMT_CBM_CFM, - .ops = &zoom2_ops, - }, -}; - -/* Audio machine driver */ -static struct snd_soc_card snd_soc_zoom2 = { - .name = "Zoom2", - .owner = THIS_MODULE, - .dai_link = zoom2_dai, - .num_links = ARRAY_SIZE(zoom2_dai), - - .dapm_widgets = zoom2_twl4030_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(zoom2_twl4030_dapm_widgets), - .dapm_routes = audio_map, - .num_dapm_routes = ARRAY_SIZE(audio_map), -}; - -static struct platform_device *zoom2_snd_device; - -static int __init zoom2_soc_init(void) -{ - int ret; - - if (!machine_is_omap_zoom2()) - return -ENODEV; - printk(KERN_INFO "Zoom2 SoC init\n"); - - zoom2_snd_device = platform_device_alloc("soc-audio", -1); - if (!zoom2_snd_device) { - printk(KERN_ERR "Platform device allocation failed\n"); - return -ENOMEM; - } - - platform_set_drvdata(zoom2_snd_device, &snd_soc_zoom2); - ret = platform_device_add(zoom2_snd_device); - if (ret) - goto err1; - - return 0; - -err1: - printk(KERN_ERR "Unable to add platform device\n"); - platform_device_put(zoom2_snd_device); - - return ret; -} -module_init(zoom2_soc_init); - -static void __exit zoom2_soc_exit(void) -{ - platform_device_unregister(zoom2_snd_device); -} -module_exit(zoom2_soc_exit); - -MODULE_AUTHOR("Misael Lopez Cruz "); -MODULE_DESCRIPTION("ALSA SoC Zoom2"); -MODULE_LICENSE("GPL"); - -- cgit v1.2.3 From 016fb39c98a71539d9abd5ee66cc1103a32fbb34 Mon Sep 17 00:00:00 2001 From: Mike Dunn Date: Sat, 29 Dec 2012 10:53:17 -0800 Subject: ASoC: palm27x: fix widgets and routes in dai_link init ASoC core code now handles creation of controls and routing based on contents of struct snd_soc_card, so remove calls to snd_soc_dapm_new_controls() and snd_soc_dapm_add_routes() from the snd_soc_dai_link init function, and add widget and route definitions to struct snd_soc_card. Signed-off-by: Mike Dunn Signed-off-by: Mark Brown --- sound/soc/pxa/palm27x.c | 15 ++++----------- 1 file changed, 4 insertions(+), 11 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c index 2074e2daf9c..bb062e6b9db 100644 --- a/sound/soc/pxa/palm27x.c +++ b/sound/soc/pxa/palm27x.c @@ -79,17 +79,6 @@ static int palm27x_ac97_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dapm_context *dapm = &codec->dapm; int err; - /* add palm27x specific widgets */ - err = snd_soc_dapm_new_controls(dapm, palm27x_dapm_widgets, - ARRAY_SIZE(palm27x_dapm_widgets)); - if (err) - return err; - - /* set up palm27x specific audio path audio_map */ - err = snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - if (err) - return err; - /* connected pins */ if (machine_is_palmld()) snd_soc_dapm_enable_pin(dapm, "MIC1"); @@ -149,6 +138,10 @@ static struct snd_soc_card palm27x_asoc = { .owner = THIS_MODULE, .dai_link = palm27x_dai, .num_links = ARRAY_SIZE(palm27x_dai), + .dapm_widgets = palm27x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(palm27x_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map) }; static struct platform_device *palm27x_snd_device; -- cgit v1.2.3 From 01a61f490cab732542753db69e17e5db657d185a Mon Sep 17 00:00:00 2001 From: Mike Dunn Date: Sat, 29 Dec 2012 10:53:18 -0800 Subject: ASoC: palm27x: register card in platform_driver probe Remove creation of an soc-audio device from the machine platform_driver probe function, and add a call to snd_soc_register_card() instead. The current code still works, but this mechanism has been deprecated, if I'm not mistaken. The ASoC core code produces the warning "ASoC: machine Palm/PXA27x should use snd_soc_register_card()" Signed-off-by: Mike Dunn Signed-off-by: Mark Brown --- sound/soc/pxa/palm27x.c | 23 ++++++----------------- 1 file changed, 6 insertions(+), 17 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c index bb062e6b9db..e1ffcdd9a64 100644 --- a/sound/soc/pxa/palm27x.c +++ b/sound/soc/pxa/palm27x.c @@ -144,8 +144,6 @@ static struct snd_soc_card palm27x_asoc = { .num_dapm_routes = ARRAY_SIZE(audio_map) }; -static struct platform_device *palm27x_snd_device; - static int palm27x_asoc_probe(struct platform_device *pdev) { int ret; @@ -162,27 +160,18 @@ static int palm27x_asoc_probe(struct platform_device *pdev) hs_jack_gpios[0].gpio = ((struct palm27x_asoc_info *) (pdev->dev.platform_data))->jack_gpio; - palm27x_snd_device = platform_device_alloc("soc-audio", -1); - if (!palm27x_snd_device) - return -ENOMEM; - - platform_set_drvdata(palm27x_snd_device, &palm27x_asoc); - ret = platform_device_add(palm27x_snd_device); - - if (ret != 0) - goto put_device; - - return 0; - -put_device: - platform_device_put(palm27x_snd_device); + palm27x_asoc.dev = &pdev->dev; + ret = snd_soc_register_card(&palm27x_asoc); + if (ret) + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); return ret; } static int palm27x_asoc_remove(struct platform_device *pdev) { - platform_device_unregister(palm27x_snd_device); + snd_soc_unregister_card(&palm27x_asoc); return 0; } -- cgit v1.2.3 From fd974e52dbbdb27910d40d8feea7000f26545683 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 27 Dec 2012 19:15:08 -0800 Subject: ASoC: fsi: don't use platform info pointer on probe() Current FSI driver is using platform info pointer, but it is not good design for DT support. This patch made it not to use platform info pointer. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 19 ++++++++----------- 1 file changed, 8 insertions(+), 11 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index f14c611b38c..ef34ef8e92e 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1918,16 +1918,15 @@ static int fsi_probe(struct platform_device *pdev) { struct fsi_master *master; const struct platform_device_id *id_entry; - struct sh_fsi_platform_info *info = pdev->dev.platform_data; - struct sh_fsi_port_info nul_info, *pinfo; + struct sh_fsi_platform_info info; struct fsi_priv *fsi; struct resource *res; unsigned int irq; int ret; - nul_info.flags = 0; - nul_info.tx_id = 0; - nul_info.rx_id = 0; + memset(&info, 0, sizeof(info)); + if (pdev->dev.platform_data) + memcpy(&info, pdev->dev.platform_data, sizeof(info)); id_entry = pdev->id_entry; if (!id_entry) { @@ -1961,12 +1960,11 @@ static int fsi_probe(struct platform_device *pdev) spin_lock_init(&master->lock); /* FSI A setting */ - pinfo = (info) ? &info->port_a : &nul_info; fsi = &master->fsia; fsi->base = master->base; fsi->master = master; - fsi_port_info_init(fsi, pinfo); - fsi_handler_init(fsi, pinfo); + fsi_port_info_init(fsi, &info.port_a); + fsi_handler_init(fsi, &info.port_a); ret = fsi_stream_probe(fsi, &pdev->dev); if (ret < 0) { dev_err(&pdev->dev, "FSIA stream probe failed\n"); @@ -1974,12 +1972,11 @@ static int fsi_probe(struct platform_device *pdev) } /* FSI B setting */ - pinfo = (info) ? &info->port_b : &nul_info; fsi = &master->fsib; fsi->base = master->base + 0x40; fsi->master = master; - fsi_port_info_init(fsi, pinfo); - fsi_handler_init(fsi, pinfo); + fsi_port_info_init(fsi, &info.port_b); + fsi_handler_init(fsi, &info.port_b); ret = fsi_stream_probe(fsi, &pdev->dev); if (ret < 0) { dev_err(&pdev->dev, "FSIB stream probe failed\n"); -- cgit v1.2.3 From 91660bd65c26b71c35772004c686ed437a1e2cf1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 5 Dec 2012 20:35:24 +0900 Subject: ASoC: wm5102: Implement routing and power management for ISRCs Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 48 +++++++++++++++++++++++++++++ sound/soc/codecs/arizona.h | 2 +- sound/soc/codecs/wm5102.c | 76 ++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 125 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index e55fa034eec..27c6a52442a 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -141,6 +141,30 @@ const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS] = { "ASRC1R", "ASRC2L", "ASRC2R", + "ISRC1INT1", + "ISRC1INT2", + "ISRC1INT3", + "ISRC1INT4", + "ISRC1DEC1", + "ISRC1DEC2", + "ISRC1DEC3", + "ISRC1DEC4", + "ISRC2INT1", + "ISRC2INT2", + "ISRC2INT3", + "ISRC2INT4", + "ISRC2DEC1", + "ISRC2DEC2", + "ISRC2DEC3", + "ISRC2DEC4", + "ISRC3INT1", + "ISRC3INT2", + "ISRC3INT3", + "ISRC3INT4", + "ISRC3DEC1", + "ISRC3DEC2", + "ISRC3DEC3", + "ISRC3DEC4", }; EXPORT_SYMBOL_GPL(arizona_mixer_texts); @@ -220,6 +244,30 @@ int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS] = { 0x91, 0x92, 0x93, + 0xa0, /* ISRC1INT1 */ + 0xa1, + 0xa2, + 0xa3, + 0xa4, /* ISRC1DEC1 */ + 0xa5, + 0xa6, + 0xa7, + 0xa8, /* ISRC2DEC1 */ + 0xa9, + 0xaa, + 0xab, + 0xac, /* ISRC2INT1 */ + 0xad, + 0xae, + 0xaf, + 0xb0, /* ISRC3DEC1 */ + 0xb1, + 0xb2, + 0xb3, + 0xb4, /* ISRC3INT1 */ + 0xb5, + 0xb6, + 0xb7, }; EXPORT_SYMBOL_GPL(arizona_mixer_values); diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 3d083d00579..7f22b4f8436 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -66,7 +66,7 @@ struct arizona_priv { struct arizona_dai_priv dai[ARIZONA_MAX_DAI]; }; -#define ARIZONA_NUM_MIXER_INPUTS 75 +#define ARIZONA_NUM_MIXER_INPUTS 99 extern const unsigned int arizona_mixer_tlv[]; extern const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS]; diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index e64b968a8c9..04ceb6d587c 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -876,6 +876,18 @@ ARIZONA_MUX_ENUMS(ASRC1R, ARIZONA_ASRC1RMIX_INPUT_1_SOURCE); ARIZONA_MUX_ENUMS(ASRC2L, ARIZONA_ASRC2LMIX_INPUT_1_SOURCE); ARIZONA_MUX_ENUMS(ASRC2R, ARIZONA_ASRC2RMIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC1INT1, ARIZONA_ISRC1INT1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC1INT2, ARIZONA_ISRC1INT2MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC1DEC1, ARIZONA_ISRC1DEC1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC1DEC2, ARIZONA_ISRC1DEC2MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC2INT1, ARIZONA_ISRC2INT1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC2INT2, ARIZONA_ISRC2INT2MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC2DEC1, ARIZONA_ISRC2DEC1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC2DEC2, ARIZONA_ISRC2DEC2MIX_INPUT_1_SOURCE); + ARIZONA_MIXER_ENUMS(DSP1L, ARIZONA_DSP1LMIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(DSP1R, ARIZONA_DSP1RMIX_INPUT_1_SOURCE); @@ -999,6 +1011,26 @@ SND_SOC_DAPM_PGA("ASRC2L", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2L_ENA_SHIFT, 0, SND_SOC_DAPM_PGA("ASRC2R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2R_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1INT1", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_INT0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1INT2", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_INT1_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC1DEC1", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_DEC0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1DEC2", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_DEC1_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC2INT1", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_INT0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC2INT2", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_INT1_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC2DEC1", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_DEC0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC2DEC2", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_DEC1_ENA_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX1", NULL, 0, ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX1_ENA_SHIFT, 0), SND_SOC_DAPM_AIF_OUT("AIF1TX2", NULL, 0, @@ -1135,6 +1167,18 @@ ARIZONA_MUX_WIDGETS(ASRC1R, "ASRC1R"), ARIZONA_MUX_WIDGETS(ASRC2L, "ASRC2L"), ARIZONA_MUX_WIDGETS(ASRC2R, "ASRC2R"), +ARIZONA_MUX_WIDGETS(ISRC1DEC1, "ISRC1DEC1"), +ARIZONA_MUX_WIDGETS(ISRC1DEC2, "ISRC1DEC2"), + +ARIZONA_MUX_WIDGETS(ISRC1INT1, "ISRC1INT1"), +ARIZONA_MUX_WIDGETS(ISRC1INT2, "ISRC1INT2"), + +ARIZONA_MUX_WIDGETS(ISRC2DEC1, "ISRC2DEC1"), +ARIZONA_MUX_WIDGETS(ISRC2DEC2, "ISRC2DEC2"), + +ARIZONA_MUX_WIDGETS(ISRC2INT1, "ISRC2INT1"), +ARIZONA_MUX_WIDGETS(ISRC2INT2, "ISRC2INT2"), + WM_ADSP2("DSP1", 0), SND_SOC_DAPM_OUTPUT("HPOUT1L"), @@ -1190,6 +1234,14 @@ SND_SOC_DAPM_OUTPUT("SPKDAT1R"), { name, "ASRC1R", "ASRC1R" }, \ { name, "ASRC2L", "ASRC2L" }, \ { name, "ASRC2R", "ASRC2R" }, \ + { name, "ISRC1DEC1", "ISRC1DEC1" }, \ + { name, "ISRC1DEC2", "ISRC1DEC2" }, \ + { name, "ISRC1INT1", "ISRC1INT1" }, \ + { name, "ISRC1INT2", "ISRC1INT2" }, \ + { name, "ISRC2DEC1", "ISRC2DEC1" }, \ + { name, "ISRC2DEC2", "ISRC2DEC2" }, \ + { name, "ISRC2INT1", "ISRC2INT1" }, \ + { name, "ISRC2INT2", "ISRC2INT2" }, \ { name, "DSP1.1", "DSP1" }, \ { name, "DSP1.2", "DSP1" }, \ { name, "DSP1.3", "DSP1" }, \ @@ -1286,6 +1338,18 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { { "ASRC2L", NULL, "ASRC2L Input" }, { "ASRC2R", NULL, "ASRC2R Input" }, + { "ISRC1DEC1", NULL, "ISRC1DEC1 Input" }, + { "ISRC1DEC2", NULL, "ISRC1DEC2 Input" }, + + { "ISRC1INT1", NULL, "ISRC1INT1 Input" }, + { "ISRC1INT2", NULL, "ISRC1INT2 Input" }, + + { "ISRC2DEC1", NULL, "ISRC2DEC1 Input" }, + { "ISRC2DEC2", NULL, "ISRC2DEC2 Input" }, + + { "ISRC2INT1", NULL, "ISRC2INT1 Input" }, + { "ISRC2INT2", NULL, "ISRC2INT2 Input" }, + ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"), ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"), ARIZONA_MIXER_ROUTES("OUT2L", "HPOUT2L"), @@ -1333,6 +1397,18 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { ARIZONA_MUX_ROUTES("ASRC2L"), ARIZONA_MUX_ROUTES("ASRC2R"), + ARIZONA_MUX_ROUTES("ISRC1INT1"), + ARIZONA_MUX_ROUTES("ISRC1INT2"), + + ARIZONA_MUX_ROUTES("ISRC1DEC1"), + ARIZONA_MUX_ROUTES("ISRC1DEC2"), + + ARIZONA_MUX_ROUTES("ISRC2INT1"), + ARIZONA_MUX_ROUTES("ISRC2INT2"), + + ARIZONA_MUX_ROUTES("ISRC2DEC1"), + ARIZONA_MUX_ROUTES("ISRC2DEC2"), + ARIZONA_DSP_ROUTES("DSP1"), { "AEC Loopback", "HPOUT1L", "OUT1L" }, -- cgit v1.2.3 From bc9ab6d31c4fde1016ecc6606913ed0d52b3ed76 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 4 Jan 2013 19:31:00 +0000 Subject: ASoC: arizona: Allow runtime reconfiguration of the output mode Some systems use external analogue switches to connect more analogue devices to the CODEC than are supported by the device. In some systems this requires changing the switched output from single ended to differential mode dynamically at runtime. Add a new function arizona_set_output_mode() to support this. Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 34 ++++++++++++++++++++++++++++++++++ sound/soc/codecs/arizona.h | 3 +++ 2 files changed, 37 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 27c6a52442a..bcd225853eb 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1164,6 +1164,40 @@ int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq, } EXPORT_SYMBOL_GPL(arizona_init_fll); +/** + * arizona_set_output_mode - Set the mode of the specified output + * + * @codec: Device to configure + * @output: Output number + * @diff: True to set the output to differential mode + * + * Some systems use external analogue switches to connect more + * analogue devices to the CODEC than are supported by the device. In + * some systems this requires changing the switched output from single + * ended to differential mode dynamically at runtime, an operation + * supported using this function. + * + * Most systems have a single static configuration and should use + * platform data instead. + */ +int arizona_set_output_mode(struct snd_soc_codec *codec, int output, bool diff) +{ + unsigned int reg, val; + + if (output < 1 || output > 6) + return -EINVAL; + + reg = ARIZONA_OUTPUT_PATH_CONFIG_1L + (output - 1) * 8; + + if (diff) + val = ARIZONA_OUT1_MONO; + else + val = 0; + + return snd_soc_update_bits(codec, reg, ARIZONA_OUT1_MONO, val); +} +EXPORT_SYMBOL_GPL(arizona_set_output_mode); + MODULE_DESCRIPTION("ASoC Wolfson Arizona class device support"); MODULE_AUTHOR("Mark Brown "); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 7f22b4f8436..0973fd9bd9a 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -213,4 +213,7 @@ extern int arizona_set_fll(struct arizona_fll *fll, int source, extern int arizona_init_dai(struct arizona_priv *priv, int dai); +int arizona_set_output_mode(struct snd_soc_codec *codec, int output, + bool diff); + #endif -- cgit v1.2.3 From b272efc8600a7bbf2dd91d0eba8a3b8949e84497 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 10 Oct 2012 15:10:08 +0900 Subject: ASoC: arizona: Factor out rate selection code In preparation for more advanced sample rate managment move the existing code out of the main hw_params() function. Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 89 +++++++++++++++++++++++++++------------------- 1 file changed, 53 insertions(+), 36 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 5afc8110ae3..6d98cf41f72 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -702,17 +702,63 @@ static int arizona_startup(struct snd_pcm_substream *substream, constraint); } +static int arizona_hw_params_rate(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona_dai_priv *dai_priv = &priv->dai[dai->id - 1]; + int base = dai->driver->base; + int i, sr_val; + + /* + * We will need to be more flexible than this in future, + * currently we use a single sample rate for SYSCLK. + */ + for (i = 0; i < ARRAY_SIZE(arizona_sr_vals); i++) + if (arizona_sr_vals[i] == params_rate(params)) + break; + if (i == ARRAY_SIZE(arizona_sr_vals)) { + arizona_aif_err(dai, "Unsupported sample rate %dHz\n", + params_rate(params)); + return -EINVAL; + } + sr_val = i; + + switch (dai_priv->clk) { + case ARIZONA_CLK_SYSCLK: + snd_soc_update_bits(codec, ARIZONA_SAMPLE_RATE_1, + ARIZONA_SAMPLE_RATE_1_MASK, sr_val); + if (base) + snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL, + ARIZONA_AIF1_RATE_MASK, 0); + break; + case ARIZONA_CLK_ASYNCCLK: + snd_soc_update_bits(codec, ARIZONA_ASYNC_SAMPLE_RATE_1, + ARIZONA_ASYNC_SAMPLE_RATE_MASK, sr_val); + if (base) + snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL, + ARIZONA_AIF1_RATE_MASK, + 8 << ARIZONA_AIF1_RATE_SHIFT); + break; + default: + arizona_aif_err(dai, "Invalid clock %d\n", dai_priv->clk); + return -EINVAL; + } + + return 0; +} + static int arizona_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; - struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); - struct arizona_dai_priv *dai_priv = &priv->dai[dai->id - 1]; int base = dai->driver->base; const int *rates; - int i; - int bclk, lrclk, wl, frame, sr_val; + int i, ret; + int bclk, lrclk, wl, frame; if (params_rate(params) % 8000) rates = &arizona_44k1_bclk_rates[0]; @@ -732,16 +778,6 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - for (i = 0; i < ARRAY_SIZE(arizona_sr_vals); i++) - if (arizona_sr_vals[i] == params_rate(params)) - break; - if (i == ARRAY_SIZE(arizona_sr_vals)) { - arizona_aif_err(dai, "Unsupported sample rate %dHz\n", - params_rate(params)); - return -EINVAL; - } - sr_val = i; - lrclk = snd_soc_params_to_bclk(params) / params_rate(params); arizona_aif_dbg(dai, "BCLK %dHz LRCLK %dHz\n", @@ -750,28 +786,9 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, wl = snd_pcm_format_width(params_format(params)); frame = wl << ARIZONA_AIF1TX_WL_SHIFT | wl; - /* - * We will need to be more flexible than this in future, - * currently we use a single sample rate for SYSCLK. - */ - switch (dai_priv->clk) { - case ARIZONA_CLK_SYSCLK: - snd_soc_update_bits(codec, ARIZONA_SAMPLE_RATE_1, - ARIZONA_SAMPLE_RATE_1_MASK, sr_val); - snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL, - ARIZONA_AIF1_RATE_MASK, 0); - break; - case ARIZONA_CLK_ASYNCCLK: - snd_soc_update_bits(codec, ARIZONA_ASYNC_SAMPLE_RATE_1, - ARIZONA_ASYNC_SAMPLE_RATE_MASK, sr_val); - snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL, - ARIZONA_AIF1_RATE_MASK, - 8 << ARIZONA_AIF1_RATE_SHIFT); - break; - default: - arizona_aif_err(dai, "Invalid clock %d\n", dai_priv->clk); - return -EINVAL; - } + ret = arizona_hw_params_rate(substream, params, dai); + if (ret != 0) + return ret; snd_soc_update_bits(codec, base + ARIZONA_AIF_BCLK_CTRL, ARIZONA_AIF1_BCLK_FREQ_MASK, bclk); -- cgit v1.2.3 From 4498a3cae5012979bbf3be2064c5ca00fe29109b Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 14 Nov 2012 18:28:58 -0200 Subject: ASoC: mxs-saif: Remove platform data All MXS users have been converted to device tree and the board files have been removed. No need to keep platform data in the driver. Signed-off-by: Fabio Estevam Acked-by: Dong Aisheng Acked-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-saif.c | 44 ++++++++++++++++---------------------------- 1 file changed, 16 insertions(+), 28 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 365d9d27a32..752675da065 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -32,7 +32,6 @@ #include #include #include -#include #include #include #include @@ -662,43 +661,32 @@ static int mxs_saif_probe(struct platform_device *pdev) struct device_node *np = pdev->dev.of_node; struct resource *iores, *dmares; struct mxs_saif *saif; - struct mxs_saif_platform_data *pdata; struct pinctrl *pinctrl; int ret = 0; + struct device_node *master; - - if (!np && pdev->id >= ARRAY_SIZE(mxs_saif)) + if (!np) return -EINVAL; saif = devm_kzalloc(&pdev->dev, sizeof(*saif), GFP_KERNEL); if (!saif) return -ENOMEM; - if (np) { - struct device_node *master; - saif->id = of_alias_get_id(np, "saif"); - if (saif->id < 0) - return saif->id; - /* - * If there is no "fsl,saif-master" phandle, it's a saif - * master. Otherwise, it's a slave and its phandle points - * to the master. - */ - master = of_parse_phandle(np, "fsl,saif-master", 0); - if (!master) { - saif->master_id = saif->id; - } else { - saif->master_id = of_alias_get_id(master, "saif"); - if (saif->master_id < 0) - return saif->master_id; - } + saif->id = of_alias_get_id(np, "saif"); + if (saif->id < 0) + return saif->id; + /* + * If there is no "fsl,saif-master" phandle, it's a saif + * master. Otherwise, it's a slave and its phandle points + * to the master. + */ + master = of_parse_phandle(np, "fsl,saif-master", 0); + if (!master) { + saif->master_id = saif->id; } else { - saif->id = pdev->id; - pdata = pdev->dev.platform_data; - if (pdata && !pdata->master_mode) - saif->master_id = pdata->master_id; - else - saif->master_id = saif->id; + saif->master_id = of_alias_get_id(master, "saif"); + if (saif->master_id < 0) + return saif->master_id; } if (saif->master_id < 0 || saif->master_id >= ARRAY_SIZE(mxs_saif)) { -- cgit v1.2.3 From 324a7fb02b9e7a226916e3593f756cd85312bae8 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Tue, 8 Jan 2013 10:45:04 -0200 Subject: ASoC: mxs-saif: Use a signed integer for error value saif->id and saif->master_id are unsigned, so they can not be negative. Fix the following warning when building with W=1 option: sound/soc/mxs/mxs-saif.c: In function 'mxs_saif_probe': sound/soc/mxs/mxs-saif.c:676:2: warning: comparison of unsigned expression < 0 is always false [-Wtype-limits] sound/soc/mxs/mxs-saif.c:688:3: warning: comparison of unsigned expression < 0 is always false [-Wtype-limits] sound/soc/mxs/mxs-saif.c:692:2: warning: comparison of unsigned expression < 0 is always false [-Wtype-limits] Use a signed variable 'ret' to handle the error values. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-saif.c | 19 ++++++++++++------- 1 file changed, 12 insertions(+), 7 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 752675da065..e70e6c844f9 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -672,9 +672,12 @@ static int mxs_saif_probe(struct platform_device *pdev) if (!saif) return -ENOMEM; - saif->id = of_alias_get_id(np, "saif"); - if (saif->id < 0) - return saif->id; + ret = of_alias_get_id(np, "saif"); + if (ret < 0) + return ret; + else + saif->id = ret; + /* * If there is no "fsl,saif-master" phandle, it's a saif * master. Otherwise, it's a slave and its phandle points @@ -684,12 +687,14 @@ static int mxs_saif_probe(struct platform_device *pdev) if (!master) { saif->master_id = saif->id; } else { - saif->master_id = of_alias_get_id(master, "saif"); - if (saif->master_id < 0) - return saif->master_id; + ret = of_alias_get_id(master, "saif"); + if (ret < 0) + return ret; + else + saif->master_id = ret; } - if (saif->master_id < 0 || saif->master_id >= ARRAY_SIZE(mxs_saif)) { + if (saif->master_id >= ARRAY_SIZE(mxs_saif)) { dev_err(&pdev->dev, "get wrong master id\n"); return -EINVAL; } -- cgit v1.2.3 From db40517c75e1a33a886c8cadfa2d95f0fe5f5f4c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 26 Oct 2012 19:30:40 +0100 Subject: ASoC: wm_adsp: Add support for parsing algorithms ADSP devices report information on the algorithms loaded on them. Parse this data and use it to allow coefficients to be configured for specific algorithms. Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 143 +++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 143 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index ffc89fab96f..990403b162f 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -350,6 +350,141 @@ out: return ret; } +static int wm_adsp_setup_algs(struct wm_adsp *dsp) +{ + struct regmap *regmap = dsp->regmap; + struct wmfw_adsp1_id_hdr adsp1_id; + struct wmfw_adsp2_id_hdr adsp2_id; + struct wmfw_adsp1_alg_hdr *adsp1_alg; + struct wmfw_adsp2_alg_hdr *adsp2_alg; + void *alg; + const struct wm_adsp_region *mem; + unsigned int pos, term; + size_t algs; + __be32 val; + int i, ret; + + switch (dsp->type) { + case WMFW_ADSP1: + mem = wm_adsp_find_region(dsp, WMFW_ADSP1_DM); + break; + case WMFW_ADSP2: + mem = wm_adsp_find_region(dsp, WMFW_ADSP2_XM); + break; + default: + mem = NULL; + break; + } + + if (mem == NULL) { + BUG_ON(mem != NULL); + return -EINVAL; + } + + switch (dsp->type) { + case WMFW_ADSP1: + ret = regmap_raw_read(regmap, mem->base, &adsp1_id, + sizeof(adsp1_id)); + if (ret != 0) { + adsp_err(dsp, "Failed to read algorithm info: %d\n", + ret); + return ret; + } + + algs = be32_to_cpu(adsp1_id.algs); + adsp_info(dsp, "Firmware: %x v%d.%d.%d, %zu algorithms\n", + be32_to_cpu(adsp1_id.fw.id), + (be32_to_cpu(adsp1_id.fw.ver) & 0xff0000) >> 16, + (be32_to_cpu(adsp1_id.fw.ver) & 0xff00) >> 8, + be32_to_cpu(adsp1_id.fw.ver) & 0xff, + algs); + + pos = sizeof(adsp1_id) / 2; + term = pos + ((sizeof(*adsp1_alg) * algs) / 2); + break; + + case WMFW_ADSP2: + ret = regmap_raw_read(regmap, mem->base, &adsp2_id, + sizeof(adsp2_id)); + if (ret != 0) { + adsp_err(dsp, "Failed to read algorithm info: %d\n", + ret); + return ret; + } + + algs = be32_to_cpu(adsp2_id.algs); + adsp_info(dsp, "Firmware: %x v%d.%d.%d, %zu algorithms\n", + be32_to_cpu(adsp2_id.fw.id), + (be32_to_cpu(adsp2_id.fw.ver) & 0xff0000) >> 16, + (be32_to_cpu(adsp2_id.fw.ver) & 0xff00) >> 8, + be32_to_cpu(adsp2_id.fw.ver) & 0xff, + algs); + + pos = sizeof(adsp2_id) / 2; + term = pos + ((sizeof(*adsp2_alg) * algs) / 2); + break; + + default: + BUG_ON(NULL == "Unknown DSP type"); + return -EINVAL; + } + + if (algs == 0) { + adsp_err(dsp, "No algorithms\n"); + return -EINVAL; + } + + /* Read the terminator first to validate the length */ + ret = regmap_raw_read(regmap, mem->base + term, &val, sizeof(val)); + if (ret != 0) { + adsp_err(dsp, "Failed to read algorithm list end: %d\n", + ret); + return ret; + } + + if (be32_to_cpu(val) != 0xbedead) + adsp_warn(dsp, "Algorithm list end %x 0x%x != 0xbeadead\n", + term, be32_to_cpu(val)); + + alg = kzalloc((term - pos) * 2, GFP_KERNEL); + if (!alg) + return -ENOMEM; + + ret = regmap_raw_read(regmap, mem->base + pos, alg, (term - pos) * 2); + if (ret != 0) { + adsp_err(dsp, "Failed to read algorithm list: %d\n", + ret); + goto out; + } + + adsp1_alg = alg; + adsp2_alg = alg; + + for (i = 0; i < algs; i++) { + switch (dsp->type) { + case WMFW_ADSP1: + adsp_info(dsp, "%d: ID %x v%d.%d.%d\n", + i, be32_to_cpu(adsp1_alg[i].alg.id), + (be32_to_cpu(adsp1_alg[i].alg.ver) & 0xff0000) >> 16, + (be32_to_cpu(adsp1_alg[i].alg.ver) & 0xff00) >> 8, + be32_to_cpu(adsp1_alg[i].alg.ver) & 0xff); + break; + + case WMFW_ADSP2: + adsp_info(dsp, "%d: ID %x v%d.%d.%d\n", + i, be32_to_cpu(adsp2_alg[i].alg.id), + (be32_to_cpu(adsp2_alg[i].alg.ver) & 0xff0000) >> 16, + (be32_to_cpu(adsp2_alg[i].alg.ver) & 0xff00) >> 8, + be32_to_cpu(adsp2_alg[i].alg.ver) & 0xff); + break; + } + } + +out: + kfree(alg); + return ret; +} + static int wm_adsp_load_coeff(struct wm_adsp *dsp) { struct regmap *regmap = dsp->regmap; @@ -468,6 +603,10 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; + ret = wm_adsp_setup_algs(dsp); + if (ret != 0) + goto err; + ret = wm_adsp_load_coeff(dsp); if (ret != 0) goto err; @@ -604,6 +743,10 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; + ret = wm_adsp_setup_algs(dsp); + if (ret != 0) + goto err; + ret = wm_adsp_load_coeff(dsp); if (ret != 0) goto err; -- cgit v1.2.3 From 45b9ee72d0e70c11a01152ef4bec92718b55906b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 8 Jan 2013 16:02:06 +0000 Subject: ASoC: wm_adsp: Factor out calculation of memory base addresses Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 30 +++++++++++++++++++++++++----- 1 file changed, 25 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 990403b162f..bcc0d0ffe72 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -156,6 +156,26 @@ static struct wm_adsp_region const *wm_adsp_find_region(struct wm_adsp *dsp, return NULL; } +static unsigned int wm_adsp_region_to_reg(struct wm_adsp_region const *region, + unsigned int offset) +{ + switch (region->type) { + case WMFW_ADSP1_PM: + return region->base + (offset * 3); + case WMFW_ADSP1_DM: + return region->base + (offset * 2); + case WMFW_ADSP2_XM: + return region->base + (offset * 2); + case WMFW_ADSP2_YM: + return region->base + (offset * 2); + case WMFW_ADSP1_ZM: + return region->base + (offset * 2); + default: + WARN_ON(NULL != "Unknown memory region type"); + return offset; + } +} + static int wm_adsp_load(struct wm_adsp *dsp) { const struct firmware *firmware; @@ -282,27 +302,27 @@ static int wm_adsp_load(struct wm_adsp *dsp) case WMFW_ADSP1_PM: BUG_ON(!mem); region_name = "PM"; - reg = mem->base + (offset * 3); + reg = wm_adsp_region_to_reg(mem, offset); break; case WMFW_ADSP1_DM: BUG_ON(!mem); region_name = "DM"; - reg = mem->base + (offset * 2); + reg = wm_adsp_region_to_reg(mem, offset); break; case WMFW_ADSP2_XM: BUG_ON(!mem); region_name = "XM"; - reg = mem->base + (offset * 2); + reg = wm_adsp_region_to_reg(mem, offset); break; case WMFW_ADSP2_YM: BUG_ON(!mem); region_name = "YM"; - reg = mem->base + (offset * 2); + reg = wm_adsp_region_to_reg(mem, offset); break; case WMFW_ADSP1_ZM: BUG_ON(!mem); region_name = "ZM"; - reg = mem->base + (offset * 2); + reg = wm_adsp_region_to_reg(mem, offset); break; default: adsp_warn(dsp, -- cgit v1.2.3 From d62f4bc665f3efd2a3d20134f874608b8afce7a1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 19 Dec 2012 14:00:30 +0000 Subject: ASoC: wm_asdp: Validate sanity of algorithm count If we run into I/O problems the algorithm count may be crazy, validate it before we proceed and dump the read data for diagnostic purposes. Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 17 +++++++++++++++-- 1 file changed, 15 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index bcc0d0ffe72..017d5101e78 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -377,10 +377,10 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) struct wmfw_adsp2_id_hdr adsp2_id; struct wmfw_adsp1_alg_hdr *adsp1_alg; struct wmfw_adsp2_alg_hdr *adsp2_alg; - void *alg; + void *alg, *buf; const struct wm_adsp_region *mem; unsigned int pos, term; - size_t algs; + size_t algs, buf_size; __be32 val; int i, ret; @@ -411,6 +411,9 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) return ret; } + buf = &adsp1_id; + buf_size = sizeof(adsp1_id); + algs = be32_to_cpu(adsp1_id.algs); adsp_info(dsp, "Firmware: %x v%d.%d.%d, %zu algorithms\n", be32_to_cpu(adsp1_id.fw.id), @@ -432,6 +435,9 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) return ret; } + buf = &adsp2_id; + buf_size = sizeof(adsp2_id); + algs = be32_to_cpu(adsp2_id.algs); adsp_info(dsp, "Firmware: %x v%d.%d.%d, %zu algorithms\n", be32_to_cpu(adsp2_id.fw.id), @@ -454,6 +460,13 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) return -EINVAL; } + if (algs > 1024) { + adsp_err(dsp, "Algorithm count %zx excessive\n", algs); + print_hex_dump_bytes(dev_name(dsp->dev), DUMP_PREFIX_OFFSET, + buf, buf_size); + return -EINVAL; + } + /* Read the terminator first to validate the length */ ret = regmap_raw_read(regmap, mem->base + term, &val, sizeof(val)); if (ret != 0) { -- cgit v1.2.3 From 471f488583c62e4daca7d24fc7c937a39de7d95f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 8 Jan 2013 16:09:31 +0000 Subject: ASoC: wm_adsp: Implement support for algorithm-specific coefficient blocks WMDR coefficient files can specify coefficients in terms of algorithm specific data regions. Record the start addresses of these regions while parsing the algorithms and then use them to handle coefficients with these formats. Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 114 +++++++++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/wm_adsp.h | 9 ++++ 2 files changed, 119 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 017d5101e78..5015ff287c3 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -378,6 +378,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) struct wmfw_adsp1_alg_hdr *adsp1_alg; struct wmfw_adsp2_alg_hdr *adsp2_alg; void *alg, *buf; + struct wm_adsp_alg_region *region; const struct wm_adsp_region *mem; unsigned int pos, term; size_t algs, buf_size; @@ -496,19 +497,80 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) for (i = 0; i < algs; i++) { switch (dsp->type) { case WMFW_ADSP1: - adsp_info(dsp, "%d: ID %x v%d.%d.%d\n", + adsp_info(dsp, "%d: ID %x v%d.%d.%d DM@%x ZM@%x\n", i, be32_to_cpu(adsp1_alg[i].alg.id), (be32_to_cpu(adsp1_alg[i].alg.ver) & 0xff0000) >> 16, (be32_to_cpu(adsp1_alg[i].alg.ver) & 0xff00) >> 8, - be32_to_cpu(adsp1_alg[i].alg.ver) & 0xff); + be32_to_cpu(adsp1_alg[i].alg.ver) & 0xff, + be32_to_cpu(adsp1_alg[i].dm), + be32_to_cpu(adsp1_alg[i].zm)); + + if (adsp1_alg[i].dm) { + region = kzalloc(sizeof(*region), GFP_KERNEL); + if (!region) + return -ENOMEM; + region->type = WMFW_ADSP1_DM; + region->alg = be32_to_cpu(adsp1_alg[i].alg.id); + region->base = be32_to_cpu(adsp1_alg[i].dm); + list_add_tail(®ion->list, + &dsp->alg_regions); + } + + if (adsp1_alg[i].zm) { + region = kzalloc(sizeof(*region), GFP_KERNEL); + if (!region) + return -ENOMEM; + region->type = WMFW_ADSP1_ZM; + region->alg = be32_to_cpu(adsp1_alg[i].alg.id); + region->base = be32_to_cpu(adsp1_alg[i].zm); + list_add_tail(®ion->list, + &dsp->alg_regions); + } break; case WMFW_ADSP2: - adsp_info(dsp, "%d: ID %x v%d.%d.%d\n", + adsp_info(dsp, + "%d: ID %x v%d.%d.%d XM@%x YM@%x ZM@%x\n", i, be32_to_cpu(adsp2_alg[i].alg.id), (be32_to_cpu(adsp2_alg[i].alg.ver) & 0xff0000) >> 16, (be32_to_cpu(adsp2_alg[i].alg.ver) & 0xff00) >> 8, - be32_to_cpu(adsp2_alg[i].alg.ver) & 0xff); + be32_to_cpu(adsp2_alg[i].alg.ver) & 0xff, + be32_to_cpu(adsp2_alg[i].xm), + be32_to_cpu(adsp2_alg[i].ym), + be32_to_cpu(adsp2_alg[i].zm)); + + if (adsp2_alg[i].xm) { + region = kzalloc(sizeof(*region), GFP_KERNEL); + if (!region) + return -ENOMEM; + region->type = WMFW_ADSP2_XM; + region->alg = be32_to_cpu(adsp2_alg[i].alg.id); + region->base = be32_to_cpu(adsp2_alg[i].xm); + list_add_tail(®ion->list, + &dsp->alg_regions); + } + + if (adsp2_alg[i].ym) { + region = kzalloc(sizeof(*region), GFP_KERNEL); + if (!region) + return -ENOMEM; + region->type = WMFW_ADSP2_YM; + region->alg = be32_to_cpu(adsp2_alg[i].alg.id); + region->base = be32_to_cpu(adsp2_alg[i].ym); + list_add_tail(®ion->list, + &dsp->alg_regions); + } + + if (adsp2_alg[i].zm) { + region = kzalloc(sizeof(*region), GFP_KERNEL); + if (!region) + return -ENOMEM; + region->type = WMFW_ADSP2_ZM; + region->alg = be32_to_cpu(adsp2_alg[i].alg.id); + region->base = be32_to_cpu(adsp2_alg[i].zm); + list_add_tail(®ion->list, + &dsp->alg_regions); + } break; } } @@ -524,6 +586,8 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) struct wmfw_coeff_hdr *hdr; struct wmfw_coeff_item *blk; const struct firmware *firmware; + const struct wm_adsp_region *mem; + struct wm_adsp_alg_region *alg_region; const char *region_name; int ret, pos, blocks, type, offset, reg; char *file; @@ -588,6 +652,37 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) region_name = "register"; reg = offset; break; + + case WMFW_ADSP1_DM: + case WMFW_ADSP1_ZM: + case WMFW_ADSP2_XM: + case WMFW_ADSP2_YM: + adsp_dbg(dsp, "%s.%d: %d bytes in %x for %x\n", + file, blocks, le32_to_cpu(blk->len), + type, le32_to_cpu(blk->id)); + + mem = wm_adsp_find_region(dsp, type); + if (!mem) { + adsp_err(dsp, "No base for region %x\n", type); + break; + } + + reg = 0; + list_for_each_entry(alg_region, + &dsp->alg_regions, list) { + if (le32_to_cpu(blk->id) == alg_region->alg && + type == alg_region->type) { + reg = alg_region->base + offset; + reg = wm_adsp_region_to_reg(mem, + reg); + } + } + + if (reg == 0) + adsp_err(dsp, "No %x for algorithm %x\n", + type, le32_to_cpu(blk->id)); + break; + default: adsp_err(dsp, "Unknown region type %x\n", type); break; @@ -711,6 +806,7 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, struct snd_soc_codec *codec = w->codec; struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); struct wm_adsp *dsp = &dsps[w->shift]; + struct wm_adsp_alg_region *alg_region; unsigned int val; int ret; @@ -811,6 +907,14 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, "Failed to enable supply: %d\n", ret); } + + while (!list_empty(&dsp->alg_regions)) { + alg_region = list_first_entry(&dsp->alg_regions, + struct wm_adsp_alg_region, + list); + list_del(&alg_region->list); + kfree(alg_region); + } break; default: @@ -840,6 +944,8 @@ int wm_adsp2_init(struct wm_adsp *adsp, bool dvfs) return ret; } + INIT_LIST_HEAD(&adsp->alg_regions); + if (dvfs) { adsp->dvfs = devm_regulator_get(adsp->dev, "DCVDD"); if (IS_ERR(adsp->dvfs)) { diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index ffd29a4609e..48814198e24 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -25,6 +25,13 @@ struct wm_adsp_region { unsigned int base; }; +struct wm_adsp_alg_region { + struct list_head list; + unsigned int alg; + int type; + unsigned int base; +}; + struct wm_adsp { const char *part; int num; @@ -34,6 +41,8 @@ struct wm_adsp { int base; + struct list_head alg_regions; + const struct wm_adsp_region *mem; int num_mems; -- cgit v1.2.3 From bbf1453e28e4e3ee2cf5a0c34a20469b4d465f0f Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 10 Jan 2013 00:29:11 -0800 Subject: ASoC: ak4642: add Device Tree support Support for loading the ak4642 codec module via devicetree. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/codecs/ak4642.c | 33 +++++++++++++++++++++++++++++++-- 1 file changed, 31 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 1f0cdab0329..c78794dc4b6 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -26,6 +26,7 @@ #include #include #include +#include #include #include #include @@ -513,12 +514,31 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4648 = { }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static struct of_device_id ak4642_of_match[]; static int ak4642_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { + struct device_node *np = i2c->dev.of_node; + const struct snd_soc_codec_driver *driver; + + driver = NULL; + if (np) { + const struct of_device_id *of_id; + + of_id = of_match_device(ak4642_of_match, &i2c->dev); + if (of_id) + driver = of_id->data; + } else { + driver = (struct snd_soc_codec_driver *)id->driver_data; + } + + if (!driver) { + dev_err(&i2c->dev, "no driver\n"); + return -EINVAL; + } + return snd_soc_register_codec(&i2c->dev, - (struct snd_soc_codec_driver *)id->driver_data, - &ak4642_dai, 1); + driver, &ak4642_dai, 1); } static int ak4642_i2c_remove(struct i2c_client *client) @@ -527,6 +547,14 @@ static int ak4642_i2c_remove(struct i2c_client *client) return 0; } +static struct of_device_id ak4642_of_match[] __devinitconst = { + { .compatible = "asahi-kasei,ak4642", .data = &soc_codec_dev_ak4642}, + { .compatible = "asahi-kasei,ak4643", .data = &soc_codec_dev_ak4642}, + { .compatible = "asahi-kasei,ak4648", .data = &soc_codec_dev_ak4648}, + {}, +}; +MODULE_DEVICE_TABLE(of, ak4642_of_match); + static const struct i2c_device_id ak4642_i2c_id[] = { { "ak4642", (kernel_ulong_t)&soc_codec_dev_ak4642 }, { "ak4643", (kernel_ulong_t)&soc_codec_dev_ak4642 }, @@ -539,6 +567,7 @@ static struct i2c_driver ak4642_i2c_driver = { .driver = { .name = "ak4642-codec", .owner = THIS_MODULE, + .of_match_table = ak4642_of_match, }, .probe = ak4642_i2c_probe, .remove = ak4642_i2c_remove, -- cgit v1.2.3 From 8d61f4901f83461e1f04df7743777e9db5f541aa Mon Sep 17 00:00:00 2001 From: Misael Lopez Cruz Date: Fri, 11 Jan 2013 17:00:59 +0100 Subject: ASoC: twl6040: Convert PLUGINT to no-suspend irq Convert headset PLUGINT interrupt to NO_SUSPEND type in order to allow handling of insertion/removal events while device is suspended. Signed-off-by: Misael Lopez Cruz Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 3fc3fc64dd8..ef31ace41d5 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1174,7 +1174,7 @@ static int twl6040_probe(struct snd_soc_codec *codec) mutex_init(&priv->mutex); ret = request_threaded_irq(priv->plug_irq, NULL, twl6040_audio_handler, - 0, "twl6040_irq_plug", codec); + IRQF_NO_SUSPEND, "twl6040_irq_plug", codec); if (ret) { dev_err(codec->dev, "PLUG IRQ request failed: %d\n", ret); goto plugirq_err; -- cgit v1.2.3 From 09a8b6719caecff6c08d505c1f7126b7be68663a Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 11 Jan 2013 17:01:41 +0100 Subject: ASoC: twl4030: Remove suspend/resume soc driver operations With idle_bias_off these are no longer needed. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 14 -------------- 1 file changed, 14 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 7bfabe5750e..8e6e5b01602 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -2296,18 +2296,6 @@ static struct snd_soc_dai_driver twl4030_dai[] = { }, }; -static int twl4030_soc_suspend(struct snd_soc_codec *codec) -{ - twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int twl4030_soc_resume(struct snd_soc_codec *codec) -{ - twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; -} - static int twl4030_soc_probe(struct snd_soc_codec *codec) { struct twl4030_priv *twl4030; @@ -2345,8 +2333,6 @@ static int twl4030_soc_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_twl4030 = { .probe = twl4030_soc_probe, .remove = twl4030_soc_remove, - .suspend = twl4030_soc_suspend, - .resume = twl4030_soc_resume, .read = twl4030_read_reg_cache, .write = twl4030_write, .set_bias_level = twl4030_set_bias_level, -- cgit v1.2.3 From 156db9f3bb3c210cdf905172f6063c90c4a62c3c Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 11 Jan 2013 17:01:00 +0100 Subject: ASoC: twl6040: Only set the bias_level once in twl6040_resume() No need to set the bias_level twice to _STANDBY - since this is the only state the device could be at suspend time. The driver do not support idle_bias_off yet. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index ef31ace41d5..86f12a498f3 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1115,7 +1115,6 @@ static int twl6040_suspend(struct snd_soc_codec *codec) static int twl6040_resume(struct snd_soc_codec *codec) { twl6040_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - twl6040_set_bias_level(codec, codec->dapm.suspend_bias_level); return 0; } -- cgit v1.2.3 From 9523fcdcc02e812f3a0f4849b3af1b295ad50470 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 11 Jan 2013 17:01:01 +0100 Subject: ASoC: twl6040: Convert to use devm_* when possible In this way we can clean up the probe and remove paths Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 27 ++++++++++----------------- 1 file changed, 10 insertions(+), 17 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 86f12a498f3..90b721e437a 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1131,9 +1131,10 @@ static int twl6040_probe(struct snd_soc_codec *codec) struct platform_device, dev); int ret = 0; - priv = kzalloc(sizeof(struct twl6040_data), GFP_KERNEL); + priv = devm_kzalloc(codec->dev, sizeof(*priv), GFP_KERNEL); if (priv == NULL) return -ENOMEM; + snd_soc_codec_set_drvdata(codec, priv); priv->codec = codec; @@ -1158,25 +1159,23 @@ static int twl6040_probe(struct snd_soc_codec *codec) priv->plug_irq = platform_get_irq(pdev, 0); if (priv->plug_irq < 0) { dev_err(codec->dev, "invalid irq\n"); - ret = -EINVAL; - goto work_err; + return -EINVAL; } priv->workqueue = alloc_workqueue("twl6040-codec", 0, 0); - if (!priv->workqueue) { - ret = -ENOMEM; - goto work_err; - } + if (!priv->workqueue) + return -ENOMEM; INIT_DELAYED_WORK(&priv->hs_jack.work, twl6040_accessory_work); mutex_init(&priv->mutex); - ret = request_threaded_irq(priv->plug_irq, NULL, twl6040_audio_handler, - IRQF_NO_SUSPEND, "twl6040_irq_plug", codec); + ret = devm_request_threaded_irq(codec->dev, priv->plug_irq, NULL, + twl6040_audio_handler, IRQF_NO_SUSPEND, + "twl6040_irq_plug", codec); if (ret) { dev_err(codec->dev, "PLUG IRQ request failed: %d\n", ret); - goto plugirq_err; + goto err; } twl6040_init_chip(codec); @@ -1186,12 +1185,8 @@ static int twl6040_probe(struct snd_soc_codec *codec) if (!ret) return 0; - /* Error path */ - free_irq(priv->plug_irq, codec); -plugirq_err: +err: destroy_workqueue(priv->workqueue); -work_err: - kfree(priv); return ret; } @@ -1200,9 +1195,7 @@ static int twl6040_remove(struct snd_soc_codec *codec) struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); twl6040_set_bias_level(codec, SND_SOC_BIAS_OFF); - free_irq(priv->plug_irq, codec); destroy_workqueue(priv->workqueue); - kfree(priv); return 0; } -- cgit v1.2.3 From da2107d1e464811407675880e9ab78f057b7c9d6 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 11 Jan 2013 17:01:02 +0100 Subject: ASoC: twl6040: Switch to use system workqueue for jack reporting There's no need to create a queue for this anymore Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 21 +++------------------ 1 file changed, 3 insertions(+), 18 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 90b721e437a..984911ba580 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -75,7 +75,6 @@ struct twl6040_data { u16 hf_right_step; struct twl6040_jack_data hs_jack; struct snd_soc_codec *codec; - struct workqueue_struct *workqueue; struct mutex mutex; }; @@ -404,8 +403,7 @@ static irqreturn_t twl6040_audio_handler(int irq, void *data) struct snd_soc_codec *codec = data; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); - queue_delayed_work(priv->workqueue, &priv->hs_jack.work, - msecs_to_jiffies(200)); + schedule_delayed_work(&priv->hs_jack.work, msecs_to_jiffies(200)); return IRQ_HANDLED; } @@ -1162,10 +1160,6 @@ static int twl6040_probe(struct snd_soc_codec *codec) return -EINVAL; } - priv->workqueue = alloc_workqueue("twl6040-codec", 0, 0); - if (!priv->workqueue) - return -ENOMEM; - INIT_DELAYED_WORK(&priv->hs_jack.work, twl6040_accessory_work); mutex_init(&priv->mutex); @@ -1175,27 +1169,18 @@ static int twl6040_probe(struct snd_soc_codec *codec) "twl6040_irq_plug", codec); if (ret) { dev_err(codec->dev, "PLUG IRQ request failed: %d\n", ret); - goto err; + return ret; } twl6040_init_chip(codec); /* power on device */ - ret = twl6040_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - if (!ret) - return 0; - -err: - destroy_workqueue(priv->workqueue); - return ret; + return twl6040_set_bias_level(codec, SND_SOC_BIAS_STANDBY); } static int twl6040_remove(struct snd_soc_codec *codec) { - struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); - twl6040_set_bias_level(codec, SND_SOC_BIAS_OFF); - destroy_workqueue(priv->workqueue); return 0; } -- cgit v1.2.3 From 85becda62c77d0951a1079bb45d0b5cbe6012252 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 11 Jan 2013 17:01:03 +0100 Subject: ASoC: twl6040: Remove leftover code from hs/hf ramp implementation The code to do the ramp has been removed a long time ago. Remove the remaining code as well since this is not needed. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 21 --------------------- 1 file changed, 21 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 984911ba580..9b9a6e58761 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -69,10 +69,6 @@ struct twl6040_data { int hs_power_mode_locked; unsigned int clk_in; unsigned int sysclk; - u16 hs_left_step; - u16 hs_right_step; - u16 hf_left_step; - u16 hf_right_step; struct twl6040_jack_data hs_jack; struct snd_soc_codec *codec; struct mutex mutex; @@ -1124,7 +1120,6 @@ static int twl6040_resume(struct snd_soc_codec *codec) static int twl6040_probe(struct snd_soc_codec *codec) { struct twl6040_data *priv; - struct twl6040_codec_data *pdata = dev_get_platdata(codec->dev); struct platform_device *pdev = container_of(codec->dev, struct platform_device, dev); int ret = 0; @@ -1138,22 +1133,6 @@ static int twl6040_probe(struct snd_soc_codec *codec) priv->codec = codec; codec->control_data = dev_get_drvdata(codec->dev->parent); - if (pdata && pdata->hs_left_step && pdata->hs_right_step) { - priv->hs_left_step = pdata->hs_left_step; - priv->hs_right_step = pdata->hs_right_step; - } else { - priv->hs_left_step = 1; - priv->hs_right_step = 1; - } - - if (pdata && pdata->hf_left_step && pdata->hf_right_step) { - priv->hf_left_step = pdata->hf_left_step; - priv->hf_right_step = pdata->hf_right_step; - } else { - priv->hf_left_step = 1; - priv->hf_right_step = 1; - } - priv->plug_irq = platform_get_irq(pdev, 0); if (priv->plug_irq < 0) { dev_err(codec->dev, "invalid irq\n"); -- cgit v1.2.3 From a88fedfd342badff3653edcc6b6423e892d1773f Mon Sep 17 00:00:00 2001 From: Ricardo Neri Date: Thu, 10 Jan 2013 18:32:13 -0600 Subject: ASoC: OMAP: HDMI: Initialize IEC-60958 channel status word As the IEC-60958 channel status word is set by ANDing and ORing with the appropriate definitions, the word bytes need to be initialized to zero to avoid misconfiguration due to previous hw_params calls. Signed-off-by: Ricardo Neri Signed-off-by: Mark Brown --- sound/soc/omap/omap-hdmi.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/omap/omap-hdmi.c b/sound/soc/omap/omap-hdmi.c index 7ea24819d57..32fa840c493 100644 --- a/sound/soc/omap/omap-hdmi.c +++ b/sound/soc/omap/omap-hdmi.c @@ -110,6 +110,8 @@ static int omap_hdmi_dai_hw_params(struct snd_pcm_substream *substream, /* * fill the IEC-60958 channel status word */ + /* initialize the word bytes */ + memset(iec->status, 0, sizeof(iec->status)); /* specify IEC-60958-3 (commercial use) */ iec->status[0] &= ~IEC958_AES0_PROFESSIONAL; -- cgit v1.2.3 From fffc0ca29fdf3a786e74082c698c701d6ebdf720 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Thu, 10 Jan 2013 11:59:57 +0300 Subject: ASoC: pcm: delete some dead code I've removed several unreachable returns. Signed-off-by: Dan Carpenter Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 12 +++--------- 1 file changed, 3 insertions(+), 9 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index d7711fce119..6006b371b79 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1728,20 +1728,16 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) /* startup must always be called for new BEs */ ret = dpcm_be_dai_startup(fe, stream); - if (ret < 0) { + if (ret < 0) goto disconnect; - return ret; - } /* keep going if FE state is > open */ if (fe->dpcm[stream].state == SND_SOC_DPCM_STATE_OPEN) return 0; ret = dpcm_be_dai_hw_params(fe, stream); - if (ret < 0) { + if (ret < 0) goto close; - return ret; - } /* keep going if FE state is > hw_params */ if (fe->dpcm[stream].state == SND_SOC_DPCM_STATE_HW_PARAMS) @@ -1749,10 +1745,8 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) ret = dpcm_be_dai_prepare(fe, stream); - if (ret < 0) { + if (ret < 0) goto hw_free; - return ret; - } /* run the stream event for each BE */ dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_NOP); -- cgit v1.2.3 From 1023dbd90c1e3e87921198939917c1f50b4b6af7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 11 Jan 2013 22:58:28 +0000 Subject: ASoC: wm_adsp: Add basic firmware selection support There are many firmwares available for ADSP devices. Add basic support for selecting between them, including a couple of feature sets in the set of available firmware to start off with. Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 75 ++++++++++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/wm_adsp.h | 5 ++++ 2 files changed, 78 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 5015ff287c3..1f8e8e2a1a6 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -143,6 +143,71 @@ #define ADSP2_RAM_RDY_SHIFT 0 #define ADSP2_RAM_RDY_WIDTH 1 +#define WM_ADSP_NUM_FW 3 + +static const char *wm_adsp_fw_text[WM_ADSP_NUM_FW] = { + "MBC/VSS", "Tx", "Rx ANC" +}; + +static struct { + const char *file; +} wm_adsp_fw[WM_ADSP_NUM_FW] = { + { .file = "mbc-vss" }, + { .file = "tx" }, + { .file = "rx-anc" }, +}; + +static int wm_adsp_fw_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + struct wm_adsp *adsp = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = adsp[e->shift_l].fw; + + return 0; +} + +static int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + struct wm_adsp *adsp = snd_soc_codec_get_drvdata(codec); + + if (ucontrol->value.integer.value[0] == adsp[e->shift_l].fw) + return 0; + + if (ucontrol->value.integer.value[0] >= WM_ADSP_NUM_FW) + return -EINVAL; + + if (adsp[e->shift_l].running) + return -EBUSY; + + adsp->fw = ucontrol->value.integer.value[0]; + + return 0; +} + +static const struct soc_enum wm_adsp_fw_enum[] = { + SOC_ENUM_SINGLE(0, 0, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text), + SOC_ENUM_SINGLE(0, 1, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text), + SOC_ENUM_SINGLE(0, 2, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text), + SOC_ENUM_SINGLE(0, 3, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text), +}; + +const struct snd_kcontrol_new wm_adsp_fw_controls[] = { + SOC_ENUM_EXT("DSP1 Firmware", wm_adsp_fw_enum[0], + wm_adsp_fw_get, wm_adsp_fw_put), + SOC_ENUM_EXT("DSP2 Firmware", wm_adsp_fw_enum[1], + wm_adsp_fw_get, wm_adsp_fw_put), + SOC_ENUM_EXT("DSP3 Firmware", wm_adsp_fw_enum[2], + wm_adsp_fw_get, wm_adsp_fw_put), + SOC_ENUM_EXT("DSP4 Firmware", wm_adsp_fw_enum[3], + wm_adsp_fw_get, wm_adsp_fw_put), +}; +EXPORT_SYMBOL_GPL(wm_adsp_fw_controls); static struct wm_adsp_region const *wm_adsp_find_region(struct wm_adsp *dsp, int type) @@ -197,7 +262,8 @@ static int wm_adsp_load(struct wm_adsp *dsp) if (file == NULL) return -ENOMEM; - snprintf(file, PAGE_SIZE, "%s-dsp%d.wmfw", dsp->part, dsp->num); + snprintf(file, PAGE_SIZE, "%s-dsp%d-%s.wmfw", dsp->part, dsp->num, + wm_adsp_fw[dsp->fw].file); file[PAGE_SIZE - 1] = '\0'; ret = request_firmware(&firmware, file, dsp->dev); @@ -596,7 +662,8 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) if (file == NULL) return -ENOMEM; - snprintf(file, PAGE_SIZE, "%s-dsp%d.bin", dsp->part, dsp->num); + snprintf(file, PAGE_SIZE, "%s-dsp%d-%s.bin", dsp->part, dsp->num, + wm_adsp_fw[dsp->fw].file); file[PAGE_SIZE - 1] = '\0'; ret = request_firmware(&firmware, file, dsp->dev); @@ -886,9 +953,13 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, ADSP2_CORE_ENA | ADSP2_START); if (ret != 0) goto err; + + dsp->running = true; break; case SND_SOC_DAPM_PRE_PMD: + dsp->running = false; + regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL, ADSP2_SYS_ENA | ADSP2_CORE_ENA | ADSP2_START, 0); diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 48814198e24..5e71410f8b0 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -46,6 +46,9 @@ struct wm_adsp { const struct wm_adsp_region *mem; int num_mems; + int fw; + bool running; + struct regulator *dvfs; }; @@ -59,6 +62,8 @@ struct wm_adsp { .shift = num, .event = wm_adsp2_event, \ .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD } +extern const struct snd_kcontrol_new wm_adsp_fw_controls[]; + int wm_adsp2_init(struct wm_adsp *adsp, bool dvfs); int wm_adsp1_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); -- cgit v1.2.3 From f4d8ada2a0d4634d6b334ec85328f04c2d01fc98 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 11 Jan 2013 17:01:50 +0100 Subject: ASoC: tlv320dac33: Remove suspend/resume soc driver operations With idle_bias_off these are no longer needed. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320dac33.c | 16 ---------------- 1 file changed, 16 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 782b0cded2e..4f358393d6d 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1452,20 +1452,6 @@ static int dac33_soc_remove(struct snd_soc_codec *codec) return 0; } -static int dac33_soc_suspend(struct snd_soc_codec *codec) -{ - dac33_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} - -static int dac33_soc_resume(struct snd_soc_codec *codec) -{ - dac33_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} - static struct snd_soc_codec_driver soc_codec_dev_tlv320dac33 = { .read = dac33_read_reg_cache, .write = dac33_write_locked, @@ -1476,8 +1462,6 @@ static struct snd_soc_codec_driver soc_codec_dev_tlv320dac33 = { .reg_cache_default = dac33_reg, .probe = dac33_soc_probe, .remove = dac33_soc_remove, - .suspend = dac33_soc_suspend, - .resume = dac33_soc_resume, .controls = dac33_snd_controls, .num_controls = ARRAY_SIZE(dac33_snd_controls), -- cgit v1.2.3 From a4a2992c531f6ca0aa00ce0deb31e51c1b7ae69b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 10 Jan 2013 16:49:11 -0800 Subject: ASoC: simple-card: add asoc_simple_dai for initializing Current simple-card driver calls asoc_simple_card_dai_init() if platform had a asoc_simple_card_dai_init pointer. And, this initialization function works only when platform has an applicable initial value for each dai settings. And basically, almost all sound card requires certain initialization. This means that almost all platform has initialization settings, and driver do nothing if it doesn't have settings. And additionally, current simple-card supports sysclk settings but it was only for codec. In order to abolish deviation between cpu and codec, and in order to simplify processing, this patch adds asoc_simple_dai, and removed pointless struct asoc_simple_dai_init_info which was trigger of calling asoc_simple_card_dai_init(). Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 58 +++++++++++++++++++++-------------------- 1 file changed, 30 insertions(+), 28 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index bc050ec8680..6cf8355a854 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -16,33 +16,38 @@ #define asoc_simple_get_card_info(p) \ container_of(p->dai_link, struct asoc_simple_card_info, snd_link) +static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, + struct asoc_simple_dai *set, + unsigned int daifmt) +{ + int ret = 0; + + daifmt |= set->fmt; + + if (!ret && daifmt) + ret = snd_soc_dai_set_fmt(dai, daifmt); + + if (!ret && set->sysclk) + ret = snd_soc_dai_set_sysclk(dai, 0, set->sysclk, 0); + + return ret; +} + static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) { - struct asoc_simple_card_info *cinfo = asoc_simple_get_card_info(rtd); - struct asoc_simple_dai_init_info *iinfo = cinfo->init; + struct asoc_simple_card_info *info = asoc_simple_get_card_info(rtd); struct snd_soc_dai *codec = rtd->codec_dai; struct snd_soc_dai *cpu = rtd->cpu_dai; - unsigned int cpu_daifmt = iinfo->fmt | iinfo->cpu_daifmt; - unsigned int codec_daifmt = iinfo->fmt | iinfo->codec_daifmt; + unsigned int daifmt = info->daifmt; int ret; - if (codec_daifmt) { - ret = snd_soc_dai_set_fmt(codec, codec_daifmt); - if (ret < 0) - return ret; - } - - if (iinfo->sysclk) { - ret = snd_soc_dai_set_sysclk(codec, 0, iinfo->sysclk, 0); - if (ret < 0) - return ret; - } + ret = __asoc_simple_card_dai_init(codec, &info->codec_dai, daifmt); + if (ret < 0) + return ret; - if (cpu_daifmt) { - ret = snd_soc_dai_set_fmt(cpu, cpu_daifmt); - if (ret < 0) - return ret; - } + ret = __asoc_simple_card_dai_init(cpu, &info->cpu_dai, daifmt); + if (ret < 0) + return ret; return 0; } @@ -59,10 +64,10 @@ static int asoc_simple_card_probe(struct platform_device *pdev) if (!cinfo->name || !cinfo->card || - !cinfo->cpu_dai || !cinfo->codec || !cinfo->platform || - !cinfo->codec_dai) { + !cinfo->cpu_dai.name || + !cinfo->codec_dai.name) { dev_err(dev, "insufficient asoc_simple_card_info settings\n"); return -EINVAL; } @@ -72,14 +77,11 @@ static int asoc_simple_card_probe(struct platform_device *pdev) */ cinfo->snd_link.name = cinfo->name; cinfo->snd_link.stream_name = cinfo->name; - cinfo->snd_link.cpu_dai_name = cinfo->cpu_dai; + cinfo->snd_link.cpu_dai_name = cinfo->cpu_dai.name; cinfo->snd_link.platform_name = cinfo->platform; cinfo->snd_link.codec_name = cinfo->codec; - cinfo->snd_link.codec_dai_name = cinfo->codec_dai; - - /* enable snd_link.init if cinfo has settings */ - if (cinfo->init) - cinfo->snd_link.init = asoc_simple_card_dai_init; + cinfo->snd_link.codec_dai_name = cinfo->codec_dai.name; + cinfo->snd_link.init = asoc_simple_card_dai_init; /* * init snd_soc_card -- cgit v1.2.3 From 609dad9bdf970da0952cea29a4442318cd4a090e Mon Sep 17 00:00:00 2001 From: Lucas Stach Date: Sat, 5 Jan 2013 02:18:43 +0100 Subject: ASoC: tegra: add ac97 host driver This adds the driver for the Tegra 2x AC97 host controller. Signed-off-by: Lucas Stach Reviewed-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/Kconfig | 10 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra20_ac97.c | 480 +++++++++++++++++++++++++++++++++++++++++ sound/soc/tegra/tegra20_ac97.h | 95 ++++++++ 4 files changed, 587 insertions(+) create mode 100644 sound/soc/tegra/tegra20_ac97.c create mode 100644 sound/soc/tegra/tegra20_ac97.h (limited to 'sound/soc') diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 19e5fe7cc40..4b3a2b8cb78 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -6,6 +6,16 @@ config SND_SOC_TEGRA help Say Y or M here if you want support for SoC audio on Tegra. +config SND_SOC_TEGRA20_AC97 + tristate + depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC + select SND_SOC_AC97_BUS + select SND_SOC_TEGRA20_DAS + help + Say Y or M if you want to add support for codecs attached to the + Tegra20 AC97 interface. You will also need to select the individual + machine drivers to support below. + config SND_SOC_TEGRA20_DAS tristate depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile index 391e78a34c0..02513d9edf2 100644 --- a/sound/soc/tegra/Makefile +++ b/sound/soc/tegra/Makefile @@ -1,6 +1,7 @@ # Tegra platform Support snd-soc-tegra-pcm-objs := tegra_pcm.o snd-soc-tegra-utils-objs += tegra_asoc_utils.o +snd-soc-tegra20-ac97-objs := tegra20_ac97.o snd-soc-tegra20-das-objs := tegra20_das.o snd-soc-tegra20-i2s-objs := tegra20_i2s.o snd-soc-tegra20-spdif-objs := tegra20_spdif.o @@ -9,6 +10,7 @@ snd-soc-tegra30-i2s-objs := tegra30_i2s.o obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-pcm.o obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-utils.o +obj-$(CONFIG_SND_SOC_TEGRA20_AC97) += snd-soc-tegra20-ac97.o obj-$(CONFIG_SND_SOC_TEGRA20_DAS) += snd-soc-tegra20-das.o obj-$(CONFIG_SND_SOC_TEGRA20_I2S) += snd-soc-tegra20-i2s.o obj-$(CONFIG_SND_SOC_TEGRA20_SPDIF) += snd-soc-tegra20-spdif.o diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c new file mode 100644 index 00000000000..1bae73bf1c8 --- /dev/null +++ b/sound/soc/tegra/tegra20_ac97.c @@ -0,0 +1,480 @@ +/* + * tegra20_ac97.c - Tegra20 AC97 platform driver + * + * Copyright (c) 2012 Lucas Stach + * + * Partly based on code copyright/by: + * + * Copyright (c) 2011,2012 Toradex Inc. + * + * 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 program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "tegra_asoc_utils.h" +#include "tegra20_ac97.h" + +#define DRV_NAME "tegra20-ac97" + +static struct tegra20_ac97 *workdata; + +static void tegra20_ac97_codec_reset(struct snd_ac97 *ac97) +{ + u32 readback; + unsigned long timeout; + + /* reset line is not driven by DAC pad group, have to toggle GPIO */ + gpio_set_value(workdata->reset_gpio, 0); + udelay(2); + + gpio_set_value(workdata->reset_gpio, 1); + udelay(2); + + timeout = jiffies + msecs_to_jiffies(100); + + do { + regmap_read(workdata->regmap, TEGRA20_AC97_STATUS1, &readback); + if (readback & TEGRA20_AC97_STATUS1_CODEC1_RDY) + break; + usleep_range(1000, 2000); + } while (!time_after(jiffies, timeout)); +} + +static void tegra20_ac97_codec_warm_reset(struct snd_ac97 *ac97) +{ + u32 readback; + unsigned long timeout; + + /* + * although sync line is driven by the DAC pad group warm reset using + * the controller cmd is not working, have to toggle sync line + * manually. + */ + gpio_request(workdata->sync_gpio, "codec-sync"); + + gpio_direction_output(workdata->sync_gpio, 1); + + udelay(2); + gpio_set_value(workdata->sync_gpio, 0); + udelay(2); + gpio_free(workdata->sync_gpio); + + timeout = jiffies + msecs_to_jiffies(100); + + do { + regmap_read(workdata->regmap, TEGRA20_AC97_STATUS1, &readback); + if (readback & TEGRA20_AC97_STATUS1_CODEC1_RDY) + break; + usleep_range(1000, 2000); + } while (!time_after(jiffies, timeout)); +} + +static unsigned short tegra20_ac97_codec_read(struct snd_ac97 *ac97_snd, + unsigned short reg) +{ + u32 readback; + unsigned long timeout; + + regmap_write(workdata->regmap, TEGRA20_AC97_CMD, + (((reg | 0x80) << TEGRA20_AC97_CMD_CMD_ADDR_SHIFT) & + TEGRA20_AC97_CMD_CMD_ADDR_MASK) | + TEGRA20_AC97_CMD_BUSY); + + timeout = jiffies + msecs_to_jiffies(100); + + do { + regmap_read(workdata->regmap, TEGRA20_AC97_STATUS1, &readback); + if (readback & TEGRA20_AC97_STATUS1_STA_VALID1) + break; + usleep_range(1000, 2000); + } while (!time_after(jiffies, timeout)); + + return ((readback & TEGRA20_AC97_STATUS1_STA_DATA1_MASK) >> + TEGRA20_AC97_STATUS1_STA_DATA1_SHIFT); +} + +static void tegra20_ac97_codec_write(struct snd_ac97 *ac97_snd, + unsigned short reg, unsigned short val) +{ + u32 readback; + unsigned long timeout; + + regmap_write(workdata->regmap, TEGRA20_AC97_CMD, + ((reg << TEGRA20_AC97_CMD_CMD_ADDR_SHIFT) & + TEGRA20_AC97_CMD_CMD_ADDR_MASK) | + ((val << TEGRA20_AC97_CMD_CMD_DATA_SHIFT) & + TEGRA20_AC97_CMD_CMD_DATA_MASK) | + TEGRA20_AC97_CMD_BUSY); + + timeout = jiffies + msecs_to_jiffies(100); + + do { + regmap_read(workdata->regmap, TEGRA20_AC97_CMD, &readback); + if (!(readback & TEGRA20_AC97_CMD_BUSY)) + break; + usleep_range(1000, 2000); + } while (!time_after(jiffies, timeout)); +} + +struct snd_ac97_bus_ops soc_ac97_ops = { + .read = tegra20_ac97_codec_read, + .write = tegra20_ac97_codec_write, + .reset = tegra20_ac97_codec_reset, + .warm_reset = tegra20_ac97_codec_warm_reset, +}; +EXPORT_SYMBOL_GPL(soc_ac97_ops); + +static inline void tegra20_ac97_start_playback(struct tegra20_ac97 *ac97) +{ + regmap_update_bits(ac97->regmap, TEGRA20_AC97_FIFO1_SCR, + TEGRA20_AC97_FIFO_SCR_PB_QRT_MT_EN, + TEGRA20_AC97_FIFO_SCR_PB_QRT_MT_EN); + + regmap_update_bits(ac97->regmap, TEGRA20_AC97_CTRL, + TEGRA20_AC97_CTRL_PCM_DAC_EN | + TEGRA20_AC97_CTRL_STM_EN, + TEGRA20_AC97_CTRL_PCM_DAC_EN | + TEGRA20_AC97_CTRL_STM_EN); +} + +static inline void tegra20_ac97_stop_playback(struct tegra20_ac97 *ac97) +{ + regmap_update_bits(ac97->regmap, TEGRA20_AC97_FIFO1_SCR, + TEGRA20_AC97_FIFO_SCR_PB_QRT_MT_EN, 0); + + regmap_update_bits(ac97->regmap, TEGRA20_AC97_CTRL, + TEGRA20_AC97_CTRL_PCM_DAC_EN, 0); +} + +static inline void tegra20_ac97_start_capture(struct tegra20_ac97 *ac97) +{ + regmap_update_bits(ac97->regmap, TEGRA20_AC97_FIFO1_SCR, + TEGRA20_AC97_FIFO_SCR_REC_FULL_EN, + TEGRA20_AC97_FIFO_SCR_REC_FULL_EN); +} + +static inline void tegra20_ac97_stop_capture(struct tegra20_ac97 *ac97) +{ + regmap_update_bits(ac97->regmap, TEGRA20_AC97_FIFO1_SCR, + TEGRA20_AC97_FIFO_SCR_REC_FULL_EN, 0); +} + +static int tegra20_ac97_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct tegra20_ac97 *ac97 = snd_soc_dai_get_drvdata(dai); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + tegra20_ac97_start_playback(ac97); + else + tegra20_ac97_start_capture(ac97); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + tegra20_ac97_stop_playback(ac97); + else + tegra20_ac97_stop_capture(ac97); + break; + default: + return -EINVAL; + } + + return 0; +} + +static const struct snd_soc_dai_ops tegra20_ac97_dai_ops = { + .trigger = tegra20_ac97_trigger, +}; + +static int tegra20_ac97_probe(struct snd_soc_dai *dai) +{ + struct tegra20_ac97 *ac97 = snd_soc_dai_get_drvdata(dai); + + dai->capture_dma_data = &ac97->capture_dma_data; + dai->playback_dma_data = &ac97->playback_dma_data; + + return 0; +} + +static struct snd_soc_dai_driver tegra20_ac97_dai = { + .name = "tegra-ac97-pcm", + .ac97_control = 1, + .probe = tegra20_ac97_probe, + .playback = { + .stream_name = "PCM Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "PCM Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &tegra20_ac97_dai_ops, +}; + +static bool tegra20_ac97_wr_rd_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TEGRA20_AC97_CTRL: + case TEGRA20_AC97_CMD: + case TEGRA20_AC97_STATUS1: + case TEGRA20_AC97_FIFO1_SCR: + case TEGRA20_AC97_FIFO_TX1: + case TEGRA20_AC97_FIFO_RX1: + return true; + default: + break; + } + + return false; +} + +static bool tegra20_ac97_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TEGRA20_AC97_STATUS1: + case TEGRA20_AC97_FIFO1_SCR: + case TEGRA20_AC97_FIFO_TX1: + case TEGRA20_AC97_FIFO_RX1: + return true; + default: + break; + } + + return false; +} + +static bool tegra20_ac97_precious_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TEGRA20_AC97_FIFO_TX1: + case TEGRA20_AC97_FIFO_RX1: + return true; + default: + break; + } + + return false; +} + +static const struct regmap_config tegra20_ac97_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = TEGRA20_AC97_FIFO_RX1, + .writeable_reg = tegra20_ac97_wr_rd_reg, + .readable_reg = tegra20_ac97_wr_rd_reg, + .volatile_reg = tegra20_ac97_volatile_reg, + .precious_reg = tegra20_ac97_precious_reg, + .cache_type = REGCACHE_RBTREE, +}; + +static int tegra20_ac97_platform_probe(struct platform_device *pdev) +{ + struct tegra20_ac97 *ac97; + struct resource *mem, *memregion; + u32 of_dma[2]; + void __iomem *regs; + int ret = 0; + + ac97 = devm_kzalloc(&pdev->dev, sizeof(struct tegra20_ac97), + GFP_KERNEL); + if (!ac97) { + dev_err(&pdev->dev, "Can't allocate tegra20_ac97\n"); + ret = -ENOMEM; + goto err; + } + dev_set_drvdata(&pdev->dev, ac97); + + ac97->clk_ac97 = clk_get(&pdev->dev, NULL); + if (IS_ERR(ac97->clk_ac97)) { + dev_err(&pdev->dev, "Can't retrieve ac97 clock\n"); + ret = PTR_ERR(ac97->clk_ac97); + goto err; + } + + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!mem) { + dev_err(&pdev->dev, "No memory resource\n"); + ret = -ENODEV; + goto err_clk_put; + } + + memregion = devm_request_mem_region(&pdev->dev, mem->start, + resource_size(mem), DRV_NAME); + if (!memregion) { + dev_err(&pdev->dev, "Memory region already claimed\n"); + ret = -EBUSY; + goto err_clk_put; + } + + regs = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); + if (!regs) { + dev_err(&pdev->dev, "ioremap failed\n"); + ret = -ENOMEM; + goto err_clk_put; + } + + ac97->regmap = devm_regmap_init_mmio(&pdev->dev, regs, + &tegra20_ac97_regmap_config); + if (IS_ERR(ac97->regmap)) { + dev_err(&pdev->dev, "regmap init failed\n"); + ret = PTR_ERR(ac97->regmap); + goto err_clk_put; + } + + if (of_property_read_u32_array(pdev->dev.of_node, + "nvidia,dma-request-selector", + of_dma, 2) < 0) { + dev_err(&pdev->dev, "No DMA resource\n"); + ret = -ENODEV; + goto err_clk_put; + } + + ac97->reset_gpio = of_get_named_gpio(pdev->dev.of_node, + "nvidia,codec-reset-gpio", 0); + if (gpio_is_valid(ac97->reset_gpio)) { + ret = devm_gpio_request_one(&pdev->dev, ac97->reset_gpio, + GPIOF_OUT_INIT_HIGH, "codec-reset"); + if (ret) { + dev_err(&pdev->dev, "could not get codec-reset GPIO\n"); + goto err_clk_put; + } + } else { + dev_err(&pdev->dev, "no codec-reset GPIO supplied\n"); + goto err_clk_put; + } + + ac97->sync_gpio = of_get_named_gpio(pdev->dev.of_node, + "nvidia,codec-sync-gpio", 0); + if (!gpio_is_valid(ac97->sync_gpio)) { + dev_err(&pdev->dev, "no codec-sync GPIO supplied\n"); + goto err_clk_put; + } + + ac97->capture_dma_data.addr = mem->start + TEGRA20_AC97_FIFO_RX1; + ac97->capture_dma_data.wrap = 4; + ac97->capture_dma_data.width = 32; + ac97->capture_dma_data.req_sel = of_dma[1]; + + ac97->playback_dma_data.addr = mem->start + TEGRA20_AC97_FIFO_TX1; + ac97->playback_dma_data.wrap = 4; + ac97->playback_dma_data.width = 32; + ac97->playback_dma_data.req_sel = of_dma[1]; + + ret = snd_soc_register_dais(&pdev->dev, &tegra20_ac97_dai, 1); + if (ret) { + dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); + ret = -ENOMEM; + goto err_clk_put; + } + + ret = tegra_pcm_platform_register(&pdev->dev); + if (ret) { + dev_err(&pdev->dev, "Could not register PCM: %d\n", ret); + goto err_unregister_dai; + } + + ret = tegra_asoc_utils_init(&ac97->util_data, &pdev->dev); + if (ret) + goto err_unregister_pcm; + + ret = tegra_asoc_utils_set_ac97_rate(&ac97->util_data); + if (ret) + goto err_asoc_utils_fini; + + ret = clk_prepare_enable(ac97->clk_ac97); + if (ret) { + dev_err(&pdev->dev, "clk_enable failed: %d\n", ret); + goto err_asoc_utils_fini; + } + + /* XXX: crufty ASoC AC97 API - only one AC97 codec allowed */ + workdata = ac97; + + return 0; + +err_asoc_utils_fini: + tegra_asoc_utils_fini(&ac97->util_data); +err_unregister_pcm: + tegra_pcm_platform_unregister(&pdev->dev); +err_unregister_dai: + snd_soc_unregister_dai(&pdev->dev); +err_clk_put: + clk_put(ac97->clk_ac97); +err: + return ret; +} + +static int tegra20_ac97_platform_remove(struct platform_device *pdev) +{ + struct tegra20_ac97 *ac97 = dev_get_drvdata(&pdev->dev); + + tegra_pcm_platform_unregister(&pdev->dev); + snd_soc_unregister_dai(&pdev->dev); + + tegra_asoc_utils_fini(&ac97->util_data); + + clk_disable_unprepare(ac97->clk_ac97); + clk_put(ac97->clk_ac97); + + return 0; +} + +static const struct of_device_id tegra20_ac97_of_match[] __devinitconst = { + { .compatible = "nvidia,tegra20-ac97", }, + {}, +}; + +static struct platform_driver tegra20_ac97_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .of_match_table = tegra20_ac97_of_match, + }, + .probe = tegra20_ac97_platform_probe, + .remove = tegra20_ac97_platform_remove, +}; +module_platform_driver(tegra20_ac97_driver); + +MODULE_AUTHOR("Lucas Stach"); +MODULE_DESCRIPTION("Tegra20 AC97 ASoC driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, tegra20_ac97_of_match); diff --git a/sound/soc/tegra/tegra20_ac97.h b/sound/soc/tegra/tegra20_ac97.h new file mode 100644 index 00000000000..dddc6828004 --- /dev/null +++ b/sound/soc/tegra/tegra20_ac97.h @@ -0,0 +1,95 @@ +/* + * tegra20_ac97.h - Definitions for the Tegra20 AC97 controller driver + * + * Copyright (c) 2012 Lucas Stach + * + * Partly based on code copyright/by: + * + * Copyright (c) 2011,2012 Toradex Inc. + * + * 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 program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + */ + +#ifndef __TEGRA20_AC97_H__ +#define __TEGRA20_AC97_H__ + +#include "tegra_pcm.h" + +#define TEGRA20_AC97_CTRL 0x00 +#define TEGRA20_AC97_CMD 0x04 +#define TEGRA20_AC97_STATUS1 0x08 +/* ... */ +#define TEGRA20_AC97_FIFO1_SCR 0x1c +/* ... */ +#define TEGRA20_AC97_FIFO_TX1 0x40 +#define TEGRA20_AC97_FIFO_RX1 0x80 + +/* TEGRA20_AC97_CTRL */ +#define TEGRA20_AC97_CTRL_STM2_EN (1 << 16) +#define TEGRA20_AC97_CTRL_DOUBLE_SAMPLING_EN (1 << 11) +#define TEGRA20_AC97_CTRL_IO_CNTRL_EN (1 << 10) +#define TEGRA20_AC97_CTRL_HSET_DAC_EN (1 << 9) +#define TEGRA20_AC97_CTRL_LINE2_DAC_EN (1 << 8) +#define TEGRA20_AC97_CTRL_PCM_LFE_EN (1 << 7) +#define TEGRA20_AC97_CTRL_PCM_SUR_EN (1 << 6) +#define TEGRA20_AC97_CTRL_PCM_CEN_DAC_EN (1 << 5) +#define TEGRA20_AC97_CTRL_LINE1_DAC_EN (1 << 4) +#define TEGRA20_AC97_CTRL_PCM_DAC_EN (1 << 3) +#define TEGRA20_AC97_CTRL_COLD_RESET (1 << 2) +#define TEGRA20_AC97_CTRL_WARM_RESET (1 << 1) +#define TEGRA20_AC97_CTRL_STM_EN (1 << 0) + +/* TEGRA20_AC97_CMD */ +#define TEGRA20_AC97_CMD_CMD_ADDR_SHIFT 24 +#define TEGRA20_AC97_CMD_CMD_ADDR_MASK (0xff << TEGRA20_AC97_CMD_CMD_ADDR_SHIFT) +#define TEGRA20_AC97_CMD_CMD_DATA_SHIFT 8 +#define TEGRA20_AC97_CMD_CMD_DATA_MASK (0xffff << TEGRA20_AC97_CMD_CMD_DATA_SHIFT) +#define TEGRA20_AC97_CMD_CMD_ID_SHIFT 2 +#define TEGRA20_AC97_CMD_CMD_ID_MASK (0x3 << TEGRA20_AC97_CMD_CMD_ID_SHIFT) +#define TEGRA20_AC97_CMD_BUSY (1 << 0) + +/* TEGRA20_AC97_STATUS1 */ +#define TEGRA20_AC97_STATUS1_STA_ADDR1_SHIFT 24 +#define TEGRA20_AC97_STATUS1_STA_ADDR1_MASK (0xff << TEGRA20_AC97_STATUS1_STA_ADDR1_SHIFT) +#define TEGRA20_AC97_STATUS1_STA_DATA1_SHIFT 8 +#define TEGRA20_AC97_STATUS1_STA_DATA1_MASK (0xffff << TEGRA20_AC97_STATUS1_STA_DATA1_SHIFT) +#define TEGRA20_AC97_STATUS1_STA_VALID1 (1 << 2) +#define TEGRA20_AC97_STATUS1_STANDBY1 (1 << 1) +#define TEGRA20_AC97_STATUS1_CODEC1_RDY (1 << 0) + +/* TEGRA20_AC97_FIFO1_SCR */ +#define TEGRA20_AC97_FIFO_SCR_REC_MT_CNT_SHIFT 27 +#define TEGRA20_AC97_FIFO_SCR_REC_MT_CNT_MASK (0x1f << TEGRA20_AC97_FIFO_SCR_REC_MT_CNT_SHIFT) +#define TEGRA20_AC97_FIFO_SCR_PB_MT_CNT_SHIFT 22 +#define TEGRA20_AC97_FIFO_SCR_PB_MT_CNT_MASK (0x1f << TEGRA20_AC97_FIFO_SCR_PB_MT_CNT_SHIFT) +#define TEGRA20_AC97_FIFO_SCR_REC_OVERRUN_INT_STA (1 << 19) +#define TEGRA20_AC97_FIFO_SCR_PB_UNDERRUN_INT_STA (1 << 18) +#define TEGRA20_AC97_FIFO_SCR_REC_FORCE_MT (1 << 17) +#define TEGRA20_AC97_FIFO_SCR_PB_FORCE_MT (1 << 16) +#define TEGRA20_AC97_FIFO_SCR_REC_FULL_EN (1 << 15) +#define TEGRA20_AC97_FIFO_SCR_REC_3QRT_FULL_EN (1 << 14) +#define TEGRA20_AC97_FIFO_SCR_REC_QRT_FULL_EN (1 << 13) +#define TEGRA20_AC97_FIFO_SCR_REC_EMPTY_EN (1 << 12) +#define TEGRA20_AC97_FIFO_SCR_PB_NOT_FULL_EN (1 << 11) +#define TEGRA20_AC97_FIFO_SCR_PB_QRT_MT_EN (1 << 10) +#define TEGRA20_AC97_FIFO_SCR_PB_3QRT_MT_EN (1 << 9) +#define TEGRA20_AC97_FIFO_SCR_PB_EMPTY_MT_EN (1 << 8) + +struct tegra20_ac97 { + struct clk *clk_ac97; + struct tegra_pcm_dma_params capture_dma_data; + struct tegra_pcm_dma_params playback_dma_data; + struct regmap *regmap; + int reset_gpio; + int sync_gpio; + struct tegra_asoc_utils_data util_data; +}; +#endif /* __TEGRA20_AC97_H__ */ -- cgit v1.2.3 From 9e7b6d60d880a463b17e4eae0d61c9f9a12f22bb Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 10 Jan 2013 00:34:08 -0800 Subject: ASoC: fsi: add device tree support Support for loading the Renesas FSI driver via devicetree. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 71 ++++++++++++++++++++++++++++++++++++++++++++++++------ 1 file changed, 63 insertions(+), 8 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index ef34ef8e92e..91576120cd4 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -16,6 +16,8 @@ #include #include #include +#include +#include #include #include #include @@ -297,7 +299,7 @@ struct fsi_master { int irq; struct fsi_priv fsia; struct fsi_priv fsib; - struct fsi_core *core; + const struct fsi_core *core; spinlock_t lock; }; @@ -1887,6 +1889,33 @@ static struct snd_soc_platform_driver fsi_soc_platform = { /* * platform function */ +static void fsi_of_parse(char *name, + struct device_node *np, + struct sh_fsi_port_info *info, + struct device *dev) +{ + int i; + char prop[128]; + unsigned long flags = 0; + struct { + char *name; + unsigned int val; + } of_parse_property[] = { + { "spdif-connection", SH_FSI_FMT_SPDIF }, + { "stream-mode-support", SH_FSI_ENABLE_STREAM_MODE }, + { "use-internal-clock", SH_FSI_CLK_CPG }, + }; + + for (i = 0; i < ARRAY_SIZE(of_parse_property); i++) { + sprintf(prop, "%s,%s", name, of_parse_property[i].name); + if (of_get_property(np, prop, NULL)) + flags |= of_parse_property[i].val; + } + info->flags = flags; + + dev_dbg(dev, "%s flags : %lx\n", name, info->flags); +} + static void fsi_port_info_init(struct fsi_priv *fsi, struct sh_fsi_port_info *info) { @@ -1914,22 +1943,40 @@ static void fsi_handler_init(struct fsi_priv *fsi, } } +static struct of_device_id fsi_of_match[]; static int fsi_probe(struct platform_device *pdev) { struct fsi_master *master; - const struct platform_device_id *id_entry; + struct device_node *np = pdev->dev.of_node; struct sh_fsi_platform_info info; + const struct fsi_core *core; struct fsi_priv *fsi; struct resource *res; unsigned int irq; int ret; memset(&info, 0, sizeof(info)); - if (pdev->dev.platform_data) - memcpy(&info, pdev->dev.platform_data, sizeof(info)); - id_entry = pdev->id_entry; - if (!id_entry) { + core = NULL; + if (np) { + const struct of_device_id *of_id; + + of_id = of_match_device(fsi_of_match, &pdev->dev); + if (of_id) { + core = of_id->data; + fsi_of_parse("fsia", np, &info.port_a, &pdev->dev); + fsi_of_parse("fsib", np, &info.port_b, &pdev->dev); + } + } else { + const struct platform_device_id *id_entry = pdev->id_entry; + if (id_entry) + core = (struct fsi_core *)id_entry->driver_data; + + if (pdev->dev.platform_data) + memcpy(&info, pdev->dev.platform_data, sizeof(info)); + } + + if (!core) { dev_err(&pdev->dev, "unknown fsi device\n"); return -ENODEV; } @@ -1956,7 +2003,7 @@ static int fsi_probe(struct platform_device *pdev) /* master setting */ master->irq = irq; - master->core = (struct fsi_core *)id_entry->driver_data; + master->core = core; spin_lock_init(&master->lock); /* FSI A setting */ @@ -1987,7 +2034,7 @@ static int fsi_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, master); ret = devm_request_irq(&pdev->dev, irq, &fsi_interrupt, 0, - id_entry->name, master); + dev_name(&pdev->dev), master); if (ret) { dev_err(&pdev->dev, "irq request err\n"); goto exit_fsib; @@ -2113,6 +2160,13 @@ static struct fsi_core fsi2_core = { .b_mclk = B_MST_CTLR, }; +static struct of_device_id fsi_of_match[] __devinitconst = { + { .compatible = "renesas,sh_fsi", .data = &fsi1_core}, + { .compatible = "renesas,sh_fsi2", .data = &fsi2_core}, + {}, +}; +MODULE_DEVICE_TABLE(of, fsi_of_match); + static struct platform_device_id fsi_id_table[] = { { "sh_fsi", (kernel_ulong_t)&fsi1_core }, { "sh_fsi2", (kernel_ulong_t)&fsi2_core }, @@ -2124,6 +2178,7 @@ static struct platform_driver fsi_driver = { .driver = { .name = "fsi-pcm-audio", .pm = &fsi_pm_ops, + .of_match_table = fsi_of_match, }, .probe = fsi_probe, .remove = fsi_remove, -- cgit v1.2.3 From c890caee54cd1fa2dd63d7f0c51d6b43eca63db3 Mon Sep 17 00:00:00 2001 From: Stephen Rothwell Date: Mon, 14 Jan 2013 12:04:00 +1100 Subject: ASoC: ak4642: remove __devinitconst annotation CONFIG_HOTPLUG is always true now and the __dev* macros have been removed. Signed-off-by: Stephen Rothwell Signed-off-by: Mark Brown --- sound/soc/codecs/ak4642.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index c78794dc4b6..2d037870970 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -547,7 +547,7 @@ static int ak4642_i2c_remove(struct i2c_client *client) return 0; } -static struct of_device_id ak4642_of_match[] __devinitconst = { +static struct of_device_id ak4642_of_match[] = { { .compatible = "asahi-kasei,ak4642", .data = &soc_codec_dev_ak4642}, { .compatible = "asahi-kasei,ak4643", .data = &soc_codec_dev_ak4642}, { .compatible = "asahi-kasei,ak4648", .data = &soc_codec_dev_ak4648}, -- cgit v1.2.3 From 5e7a7a221fbae313a8635411b557e736ba044c98 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 16 Jan 2013 10:03:56 +0900 Subject: ASoC: wm_adsp: Add initialisation function for ADSP1 Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 8 ++++++++ sound/soc/codecs/wm_adsp.h | 1 + 2 files changed, 9 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 1f8e8e2a1a6..58cac071456 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -780,6 +780,14 @@ out: return 0; } +int wm_adsp1_init(struct wm_adsp *adsp) +{ + INIT_LIST_HEAD(&adsp->alg_regions); + + return 0; +} +EXPORT_SYMBOL_GPL(wm_adsp1_init); + int wm_adsp1_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 5e71410f8b0..41206d79e03 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -64,6 +64,7 @@ struct wm_adsp { extern const struct snd_kcontrol_new wm_adsp_fw_controls[]; +int wm_adsp1_init(struct wm_adsp *adsp); int wm_adsp2_init(struct wm_adsp *adsp, bool dvfs); int wm_adsp1_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); -- cgit v1.2.3 From 5851cb3daf31a7865983ac131be87af92ab4ff7a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 16 Jan 2013 10:04:57 +0900 Subject: ASoC: wm2200: Initialise the ADSPs Signed-off-by: Mark Brown --- sound/soc/codecs/wm2200.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index afcf31df77e..90aae49350e 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -2224,6 +2224,9 @@ static int wm2200_i2c_probe(struct i2c_client *i2c, wm2200->dsp[1].mem = wm2200_dsp2_regions; wm2200->dsp[1].num_mems = ARRAY_SIZE(wm2200_dsp2_regions); + for (i = 0; i < ARRAY_SIZE(wm2200->dsp); i++) + wm_adsp1_init(&wm2200->dsp[i]); + if (pdata) wm2200->pdata = *pdata; -- cgit v1.2.3 From c98137bfcb4afd4415864258dd8ecf05c2bb5843 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 16 Jan 2013 15:03:52 +0900 Subject: ASoC: arizona: Don't request FLL lock IRQ We only log the result and since the interrupt triggers on loss of lock during shutdown this may lead to spurious interrupts during shutdown delaying the process. Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 19 ------------------- sound/soc/codecs/arizona.h | 1 - 2 files changed, 20 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index f919a3ae288..a12dd166988 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -897,17 +897,6 @@ int arizona_init_dai(struct arizona_priv *priv, int id) } EXPORT_SYMBOL_GPL(arizona_init_dai); -static irqreturn_t arizona_fll_lock(int irq, void *data) -{ - struct arizona_fll *fll = data; - - arizona_fll_dbg(fll, "Lock status changed\n"); - - complete(&fll->lock); - - return IRQ_HANDLED; -} - static irqreturn_t arizona_fll_clock_ok(int irq, void *data) { struct arizona_fll *fll = data; @@ -1147,7 +1136,6 @@ int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq, { int ret; - init_completion(&fll->lock); init_completion(&fll->ok); fll->id = id; @@ -1158,13 +1146,6 @@ int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq, snprintf(fll->clock_ok_name, sizeof(fll->clock_ok_name), "FLL%d clock OK", id); - ret = arizona_request_irq(arizona, lock_irq, fll->lock_name, - arizona_fll_lock, fll); - if (ret != 0) { - dev_err(arizona->dev, "Failed to get FLL%d lock IRQ: %d\n", - id, ret); - } - ret = arizona_request_irq(arizona, ok_irq, fll->clock_ok_name, arizona_fll_clock_ok, fll); if (ret != 0) { diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 697ff6fcaba..116372c91f5 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -197,7 +197,6 @@ struct arizona_fll { int id; unsigned int base; unsigned int vco_mult; - struct completion lock; struct completion ok; unsigned int fref; unsigned int fout; -- cgit v1.2.3 From e43fc6af2538a7a30e695227928eed8a29aa4f63 Mon Sep 17 00:00:00 2001 From: Thierry Reding Date: Wed, 16 Jan 2013 12:09:45 +0100 Subject: ASoC: fsi: Remove __devinitconst __devinitconst and friends have recently been removed and must not be used anymore. Signed-off-by: Thierry Reding Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 91576120cd4..c724026a246 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -2160,7 +2160,7 @@ static struct fsi_core fsi2_core = { .b_mclk = B_MST_CTLR, }; -static struct of_device_id fsi_of_match[] __devinitconst = { +static struct of_device_id fsi_of_match[] = { { .compatible = "renesas,sh_fsi", .data = &fsi1_core}, { .compatible = "renesas,sh_fsi2", .data = &fsi2_core}, {}, -- cgit v1.2.3 From 6995b8cb9622bf574ac6f309e69288e7d0f76ece Mon Sep 17 00:00:00 2001 From: Lucas Stach Date: Wed, 16 Jan 2013 13:05:12 +0100 Subject: ASoC: tegra: add tegra machine driver using wm9712 codec This adds a very simple machine driver using the Wolfson wm9712 AC97 codec. Signed-off-by: Lucas Stach Reviewed-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/Kconfig | 9 +++ sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra_wm9712.c | 176 +++++++++++++++++++++++++++++++++++++++++ 3 files changed, 187 insertions(+) create mode 100644 sound/soc/tegra/tegra_wm9712.c (limited to 'sound/soc') diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 4b3a2b8cb78..dbc27ce1d4d 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -80,6 +80,15 @@ config SND_SOC_TEGRA_WM8903 boards using the WM8093 codec. Currently, the supported boards are Harmony, Ventana, Seaboard, Kaen, and Aebl. +config SND_SOC_TEGRA_WM9712 + tristate "SoC Audio support for Tegra boards using a WM9712 codec" + depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC + select SND_SOC_TEGRA20_AC97 + select SND_SOC_WM9712 + help + Say Y or M here if you want to add support for SoC audio on Tegra + boards using the WM9712 (or compatible) codec. + config SND_SOC_TEGRA_TRIMSLICE tristate "SoC Audio support for TrimSlice board" depends on SND_SOC_TEGRA && I2C diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile index 02513d9edf2..416a14bde41 100644 --- a/sound/soc/tegra/Makefile +++ b/sound/soc/tegra/Makefile @@ -20,10 +20,12 @@ obj-$(CONFIG_SND_SOC_TEGRA30_I2S) += snd-soc-tegra30-i2s.o # Tegra machine Support snd-soc-tegra-wm8753-objs := tegra_wm8753.o snd-soc-tegra-wm8903-objs := tegra_wm8903.o +snd-soc-tegra-wm9712-objs := tegra_wm9712.o snd-soc-tegra-trimslice-objs := trimslice.o snd-soc-tegra-alc5632-objs := tegra_alc5632.o obj-$(CONFIG_SND_SOC_TEGRA_WM8753) += snd-soc-tegra-wm8753.o obj-$(CONFIG_SND_SOC_TEGRA_WM8903) += snd-soc-tegra-wm8903.o +obj-$(CONFIG_SND_SOC_TEGRA_WM9712) += snd-soc-tegra-wm9712.o obj-$(CONFIG_SND_SOC_TEGRA_TRIMSLICE) += snd-soc-tegra-trimslice.o obj-$(CONFIG_SND_SOC_TEGRA_ALC5632) += snd-soc-tegra-alc5632.o diff --git a/sound/soc/tegra/tegra_wm9712.c b/sound/soc/tegra/tegra_wm9712.c new file mode 100644 index 00000000000..cdbd2f0a23b --- /dev/null +++ b/sound/soc/tegra/tegra_wm9712.c @@ -0,0 +1,176 @@ +/* + * tegra20_wm9712.c - Tegra machine ASoC driver for boards using WM9712 codec. + * + * Copyright 2012 Lucas Stach + * + * Partly based on code copyright/by: + * Copyright 2011,2012 Toradex Inc. + * + * 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 program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + */ + +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +#define DRV_NAME "tegra-snd-wm9712" + +struct tegra_wm9712 { + struct platform_device *codec; +}; + +static const struct snd_soc_dapm_widget tegra_wm9712_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_LINE("LineIn", NULL), + SND_SOC_DAPM_MIC("Mic", NULL), +}; + +static int tegra_wm9712_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = codec_dai->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + + snd_soc_dapm_force_enable_pin(dapm, "Mic Bias"); + + return snd_soc_dapm_sync(dapm); +} + +static struct snd_soc_dai_link tegra_wm9712_dai = { + .name = "AC97 HiFi", + .stream_name = "AC97 HiFi", + .cpu_dai_name = "tegra-ac97-pcm", + .codec_dai_name = "wm9712-hifi", + .codec_name = "wm9712-codec", + .init = tegra_wm9712_init, +}; + +static struct snd_soc_card snd_soc_tegra_wm9712 = { + .name = "tegra-wm9712", + .owner = THIS_MODULE, + .dai_link = &tegra_wm9712_dai, + .num_links = 1, + + .dapm_widgets = tegra_wm9712_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tegra_wm9712_dapm_widgets), + .fully_routed = true, +}; + +static int tegra_wm9712_driver_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct snd_soc_card *card = &snd_soc_tegra_wm9712; + struct tegra_wm9712 *machine; + int ret; + + if (!pdev->dev.of_node) { + dev_err(&pdev->dev, "No platform data supplied\n"); + return -EINVAL; + } + + machine = devm_kzalloc(&pdev->dev, sizeof(struct tegra_wm9712), + GFP_KERNEL); + if (!machine) { + dev_err(&pdev->dev, "Can't allocate tegra_wm9712 struct\n"); + return -ENOMEM; + } + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + snd_soc_card_set_drvdata(card, machine); + + machine->codec = platform_device_alloc("wm9712-codec", -1); + if (!machine->codec) { + dev_err(&pdev->dev, "Can't allocate wm9712 platform device\n"); + return -ENOMEM; + } + + ret = platform_device_add(machine->codec); + if (ret) + goto codec_put; + + ret = snd_soc_of_parse_card_name(card, "nvidia,model"); + if (ret) + goto codec_unregister; + + ret = snd_soc_of_parse_audio_routing(card, "nvidia,audio-routing"); + if (ret) + goto codec_unregister; + + tegra_wm9712_dai.cpu_of_node = of_parse_phandle(np, + "nvidia,ac97-controller", 0); + if (!tegra_wm9712_dai.cpu_of_node) { + dev_err(&pdev->dev, + "Property 'nvidia,ac97-controller' missing or invalid\n"); + ret = -EINVAL; + goto codec_unregister; + } + + tegra_wm9712_dai.platform_of_node = tegra_wm9712_dai.cpu_of_node; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", + ret); + goto codec_unregister; + } + + return 0; + +codec_unregister: + platform_device_del(machine->codec); +codec_put: + platform_device_put(machine->codec); + return ret; +} + +static int tegra_wm9712_driver_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct tegra_wm9712 *machine = snd_soc_card_get_drvdata(card); + + snd_soc_unregister_card(card); + + platform_device_unregister(machine->codec); + + return 0; +} + +static const struct of_device_id tegra_wm9712_of_match[] __devinitconst = { + { .compatible = "nvidia,tegra-audio-wm9712", }, + {}, +}; + +static struct platform_driver tegra_wm9712_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + .of_match_table = tegra_wm9712_of_match, + }, + .probe = tegra_wm9712_driver_probe, + .remove = tegra_wm9712_driver_remove, +}; +module_platform_driver(tegra_wm9712_driver); + +MODULE_AUTHOR("Lucas Stach"); +MODULE_DESCRIPTION("Tegra+WM9712 machine ASoC driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, tegra_wm9712_of_match); -- cgit v1.2.3 From c712326d6c1e74678791d5864cd2ed283e1cc572 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 16 Jan 2013 16:59:04 +0900 Subject: ASoC: wm_adsp: Implement support for coefficeint file format 1 Implement support for a new revision of the coefficeint file format for ADSP cores. Since coefficient file format 0 has not been widely deployed and is very unlikely to ever be used with this driver code support for it has been removed. Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 20 +++++++++++++++----- sound/soc/codecs/wmfw.h | 15 ++++++++++----- 2 files changed, 25 insertions(+), 10 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 58cac071456..5841285c213 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -686,6 +686,16 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) return -EINVAL; } + switch (be32_to_cpu(hdr->rev) & 0xff) { + case 1: + break; + default: + adsp_err(dsp, "%s: Unsupported coefficient file format %d\n", + file, be32_to_cpu(hdr->rev) & 0xff); + ret = -EINVAL; + goto out_fw; + } + adsp_dbg(dsp, "%s: v%d.%d.%d\n", file, (le32_to_cpu(hdr->ver) >> 16) & 0xff, (le32_to_cpu(hdr->ver) >> 8) & 0xff, @@ -698,8 +708,8 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) pos - firmware->size > sizeof(*blk)) { blk = (void*)(&firmware->data[pos]); - type = be32_to_cpu(blk->type) & 0xff; - offset = le32_to_cpu(blk->offset) & 0xffffff; + type = le16_to_cpu(blk->type); + offset = le16_to_cpu(blk->offset); adsp_dbg(dsp, "%s.%d: %x v%d.%d.%d\n", file, blocks, le32_to_cpu(blk->id), @@ -712,10 +722,10 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) reg = 0; region_name = "Unknown"; switch (type) { - case WMFW_NAME_TEXT: - case WMFW_INFO_TEXT: + case (WMFW_NAME_TEXT << 8): + case (WMFW_INFO_TEXT << 8): break; - case WMFW_ABSOLUTE: + case (WMFW_ABSOLUTE << 8): region_name = "register"; reg = offset; break; diff --git a/sound/soc/codecs/wmfw.h b/sound/soc/codecs/wmfw.h index 5632ded67fd..ef163360a74 100644 --- a/sound/soc/codecs/wmfw.h +++ b/sound/soc/codecs/wmfw.h @@ -93,15 +93,20 @@ struct wmfw_adsp2_alg_hdr { struct wmfw_coeff_hdr { u8 magic[4]; __le32 len; - __le32 ver; + union { + __be32 rev; + __le32 ver; + }; + union { + __be32 core; + __le32 core_ver; + }; u8 data[]; } __packed; struct wmfw_coeff_item { - union { - __be32 type; - __le32 offset; - }; + __le16 offset; + __le16 type; __le32 id; __le32 ver; __le32 sr; -- cgit v1.2.3 From 82e993fac48674b0231b835516e0fdae94285b9b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 11 Jan 2013 22:59:48 +0000 Subject: ASoC: wm2200: Add controls for firmware enumeration Signed-off-by: Mark Brown --- sound/soc/codecs/wm2200.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index 90aae49350e..d400291110b 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1551,6 +1551,10 @@ static int wm2200_probe(struct snd_soc_codec *codec) return ret; } + ret = snd_soc_add_codec_controls(codec, wm_adsp_fw_controls, 2); + if (ret != 0) + return ret; + return ret; } -- cgit v1.2.3 From e5ddd303215d1e5e89d06056d17673b18219ace3 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 11 Jan 2013 22:59:35 +0000 Subject: ASoC: wm5102: Add controls for firmware selection Signed-off-by: Mark Brown --- sound/soc/codecs/wm5102.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 2adcfbae4d2..5e85b645f2e 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1579,6 +1579,10 @@ static int wm5102_codec_probe(struct snd_soc_codec *codec) if (ret != 0) return ret; + ret = snd_soc_add_codec_controls(codec, wm_adsp_fw_controls, 1); + if (ret != 0) + return ret; + snd_soc_dapm_disable_pin(&codec->dapm, "HAPTICS"); priv->core.arizona->dapm = &codec->dapm; -- cgit v1.2.3 From 94e205bfb73b6d19028dbd40404219fdeb27175e Mon Sep 17 00:00:00 2001 From: Chris Rattray Date: Fri, 18 Jan 2013 08:43:09 +0000 Subject: ASoC: wm_adsp: Set ADSP1 clock rate to match sys clock Sets the ADSP1 clock rate to match the system clock rate. To support this the codec driver provides details of register containing the system clock control bits. Signed-off-by: Chris Rattray Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 33 +++++++++++++++++++++++++++++++++ sound/soc/codecs/wm_adsp.h | 3 +++ 2 files changed, 36 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 5841285c213..9e311622190 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -103,6 +103,13 @@ #define ADSP1_START_SHIFT 0 /* DSP1_START */ #define ADSP1_START_WIDTH 1 /* DSP1_START */ +/* + * ADSP1 Control 31 + */ +#define ADSP1_CLK_SEL_MASK 0x0007 /* CLK_SEL_ENA */ +#define ADSP1_CLK_SEL_SHIFT 0 /* CLK_SEL_ENA */ +#define ADSP1_CLK_SEL_WIDTH 3 /* CLK_SEL_ENA */ + #define ADSP2_CONTROL 0 #define ADSP2_CLOCKING 1 #define ADSP2_STATUS1 4 @@ -806,12 +813,38 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); struct wm_adsp *dsp = &dsps[w->shift]; int ret; + int val; switch (event) { case SND_SOC_DAPM_POST_PMU: regmap_update_bits(dsp->regmap, dsp->base + ADSP1_CONTROL_30, ADSP1_SYS_ENA, ADSP1_SYS_ENA); + /* + * For simplicity set the DSP clock rate to be the + * SYSCLK rate rather than making it configurable. + */ + if(dsp->sysclk_reg) { + ret = regmap_read(dsp->regmap, dsp->sysclk_reg, &val); + if (ret != 0) { + adsp_err(dsp, "Failed to read SYSCLK state: %d\n", + ret); + return ret; + } + + val = (val & dsp->sysclk_mask) + >> dsp->sysclk_shift; + + ret = regmap_update_bits(dsp->regmap, + dsp->base + ADSP1_CONTROL_31, + ADSP1_CLK_SEL_MASK, val); + if (ret != 0) { + adsp_err(dsp, "Failed to set clock rate: %d\n", + ret); + return ret; + } + } + ret = wm_adsp_load(dsp); if (ret != 0) goto err; diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 41206d79e03..cb8871a3ec0 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -40,6 +40,9 @@ struct wm_adsp { struct regmap *regmap; int base; + int sysclk_reg; + int sysclk_mask; + int sysclk_shift; struct list_head alg_regions; -- cgit v1.2.3 From 20fc48632f6de2f8cb35cd7a4de2b4c1649fcac7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 20 Jan 2013 21:56:21 +0900 Subject: ASoC: wm5100: Implement DRC, EQ and LHPF coefficient configuration Signed-off-by: Mark Brown --- sound/soc/codecs/wm5100.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 54397a50807..ac1745d030d 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -563,6 +563,19 @@ SOC_DOUBLE_R("IN3 Switch", WM5100_ADC_DIGITAL_VOLUME_3L, SOC_DOUBLE_R("IN4 Switch", WM5100_ADC_DIGITAL_VOLUME_4L, WM5100_ADC_DIGITAL_VOLUME_4R, WM5100_IN4L_MUTE_SHIFT, 1, 1), +SND_SOC_BYTES_MASK("EQ1 Coefficients", WM5100_EQ1_1, 20, WM5100_EQ1_ENA), +SND_SOC_BYTES_MASK("EQ2 Coefficients", WM5100_EQ2_1, 20, WM5100_EQ2_ENA), +SND_SOC_BYTES_MASK("EQ3 Coefficients", WM5100_EQ3_1, 20, WM5100_EQ3_ENA), +SND_SOC_BYTES_MASK("EQ4 Coefficients", WM5100_EQ4_1, 20, WM5100_EQ4_ENA), + +SND_SOC_BYTES_MASK("DRC Coefficients", WM5100_DRC1_CTRL1, 5, + WM5100_DRCL_ENA | WM5100_DRCR_ENA), + +SND_SOC_BYTES("LHPF1 Coefficeints", WM5100_HPLPF1_2, 1), +SND_SOC_BYTES("LHPF2 Coefficeints", WM5100_HPLPF2_2, 1), +SND_SOC_BYTES("LHPF3 Coefficeints", WM5100_HPLPF3_2, 1), +SND_SOC_BYTES("LHPF4 Coefficeints", WM5100_HPLPF4_2, 1), + SOC_SINGLE("HPOUT1 High Performance Switch", WM5100_OUT_VOLUME_1L, WM5100_OUT1_OSR_SHIFT, 1, 0), SOC_SINGLE("HPOUT2 High Performance Switch", WM5100_OUT_VOLUME_2L, -- cgit v1.2.3 From 908a5741abc40f46cbb51704031d92ef43413df0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 20 Jan 2013 21:55:55 +0900 Subject: ASoC: wm2200: Implement EQ and LHPF coefficient configuration Signed-off-by: Mark Brown --- sound/soc/codecs/wm2200.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index d400291110b..0e11184a373 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1144,6 +1144,12 @@ SOC_DOUBLE_R_TLV("IN3 Digital Volume", WM2200_ADC_DIGITAL_VOLUME_3L, WM2200_ADC_DIGITAL_VOLUME_3R, WM2200_IN3L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SND_SOC_BYTES_MASK("EQL Coefficients", WM2200_EQL_1, 20, WM2200_EQL_ENA), +SND_SOC_BYTES_MASK("EQR Coefficients", WM2200_EQR_1, 20, WM2200_EQR_ENA), + +SND_SOC_BYTES("LHPF1 Coefficeints", WM2200_HPLPF1_2, 1), +SND_SOC_BYTES("LHPF2 Coefficeints", WM2200_HPLPF2_2, 1), + SOC_SINGLE("OUT1 High Performance Switch", WM2200_DAC_DIGITAL_VOLUME_1L, WM2200_OUT1_OSR_SHIFT, 1, 0), SOC_SINGLE("OUT2 High Performance Switch", WM2200_DAC_DIGITAL_VOLUME_2L, -- cgit v1.2.3 From 25c62f7e7014d64c0308d0dedb99282ed8ade2ce Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 20 Jan 2013 19:02:19 +0900 Subject: ASoC: wm_adsp: Make region identification errors more informative Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 9e311622190..084ea5f4aed 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -768,7 +768,8 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) break; default: - adsp_err(dsp, "Unknown region type %x\n", type); + adsp_err(dsp, "%s.%d: Unknown region type %x at %d\n", + file, blocks, type, pos); break; } -- cgit v1.2.3 From f2a93e2a4c048dfb5c7ee3f159a4a1d1cb61b4b5 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 20 Jan 2013 22:17:30 +0900 Subject: ASoC: wm_adsp: Use GFP_DMA for algorithm readback Normally kmalloc() returns things that are DMA safe so not visible on all platforms but we do need to explicitly request DMA safe memory. Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 084ea5f4aed..edb67138d54 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -553,7 +553,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) adsp_warn(dsp, "Algorithm list end %x 0x%x != 0xbeadead\n", term, be32_to_cpu(val)); - alg = kzalloc((term - pos) * 2, GFP_KERNEL); + alg = kzalloc((term - pos) * 2, GFP_KERNEL | GFP_DMA); if (!alg) return -ENOMEM; -- cgit v1.2.3 From f2c26d48d9c6e91c6b9c914f215952400e6f889b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 21 Jan 2013 16:09:36 +0900 Subject: ASoC: arizona: Support clearing clocks Some systems may wish to support switching between telephony and CD audio clock rates but this is restricted by enforcement of constraints on the current DAI clock. Support setting clocks to zero and don't enforce any constraints in that case in order to facilitate this use case. Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 316f074e686..845d25630ba 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -474,6 +474,10 @@ int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id, case 147456000: val |= 6 << ARIZONA_SYSCLK_FREQ_SHIFT; break; + case 0: + dev_dbg(arizona->dev, "%s cleared\n", name); + *clk = freq; + return 0; default: return -EINVAL; } @@ -692,6 +696,9 @@ static int arizona_startup(struct snd_pcm_substream *substream, return 0; } + if (base_rate == 0) + return 0; + if (base_rate % 8000) constraint = &arizona_44k1_constraint; else -- cgit v1.2.3 From 20da6d5ac05905833db4b339b57b9dcbd21b6152 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 12 Jan 2013 19:58:17 +0000 Subject: ASoC: wm_adsp: Provide explicit trace of coefficient writes Helpful for debugging. Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 76ca176eac0..bc5e3838599 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -793,6 +793,9 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) return -ENOMEM; } + adsp_dbg(dsp, "%s.%d: Writing %d bytes at %x\n", + file, blocks, le32_to_cpu(blk->len), + reg); ret = regmap_raw_write(regmap, reg, blk->data, le32_to_cpu(blk->len)); if (ret != 0) { -- cgit v1.2.3 From c94aa30edac4d328674e9c127918317009d30c1a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 17 Jan 2013 16:35:14 +0900 Subject: ASoC: arizona: Allow number of channels clocked to be restricted Place a cap on the number of channels clocks are generated for. This is intended for use with systems which have the WM5102 master an I2S bus with multiple data lines. Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 14 ++++++++++++-- 1 file changed, 12 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 845d25630ba..d855a6c098d 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -762,18 +762,28 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; int base = dai->driver->base; const int *rates; int i, ret; - int bclk, lrclk, wl, frame; + int chan_limit = arizona->pdata.max_channels_clocked[dai->id - 1]; + int bclk, lrclk, wl, frame, bclk_target; if (params_rate(params) % 8000) rates = &arizona_44k1_bclk_rates[0]; else rates = &arizona_48k_bclk_rates[0]; + bclk_target = snd_soc_params_to_bclk(params); + if (chan_limit && chan_limit < params_channels(params)) { + arizona_aif_dbg(dai, "Limiting to %d channels\n", chan_limit); + bclk_target /= params_channels(params); + bclk_target *= chan_limit; + } + for (i = 0; i < ARRAY_SIZE(arizona_44k1_bclk_rates); i++) { - if (rates[i] >= snd_soc_params_to_bclk(params) && + if (rates[i] >= bclk_target && rates[i] % params_rate(params) == 0) { bclk = i; break; -- cgit v1.2.3 From b5a8fe439ab343631d905a51438db3de54ba94df Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 20 Jan 2013 21:42:22 +0900 Subject: ASoC: core: Ensure SND_SOC_BYTES writes are from DMA safe memory With some buses the transfers may DMAed, especially for larger blocks. Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 12 +++++------- 1 file changed, 5 insertions(+), 7 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e0e8ce0c031..02d826efd7f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3097,9 +3097,12 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, if (!codec->using_regmap) return -EINVAL; - data = ucontrol->value.bytes.data; len = params->num_regs * codec->val_bytes; + data = kmemdup(ucontrol->value.bytes.data, len, GFP_KERNEL | GFP_DMA); + if (!data) + return -ENOMEM; + /* * If we've got a mask then we need to preserve the register * bits. We shouldn't modify the incoming data so take a @@ -3112,10 +3115,6 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, val &= params->mask; - data = kmemdup(data, len, GFP_KERNEL); - if (!data) - return -ENOMEM; - switch (codec->val_bytes) { case 1: ((u8 *)data)[0] &= ~params->mask; @@ -3137,8 +3136,7 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, ret = regmap_raw_write(codec->control_data, params->base, data, len); - if (params->mask) - kfree(data); + kfree(data); return ret; } -- cgit v1.2.3 From f63d944b71e8c4df9daaf44c7a9756549735140c Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Mon, 21 Jan 2013 09:02:31 +0000 Subject: ASoC: wm_adsp: Release firmware on error This patch correctly releases the firmware if the magic string in the firmware header does not match. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index ffc89fab96f..046b70be6be 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -384,7 +384,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) hdr = (void*)&firmware->data[0]; if (memcmp(hdr->magic, "WMDR", 4) != 0) { adsp_err(dsp, "%s: invalid magic\n", file); - return -EINVAL; + goto out_fw; } adsp_dbg(dsp, "%s: v%d.%d.%d\n", file, -- cgit v1.2.3 From 338c5188f6fbccf3ef34a6ae46ada8f64565e1d0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 24 Jan 2013 00:35:48 +0800 Subject: ASoC: wm_adsp: Correct handling of some coefficeint blocks Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index bc5e3838599..55a0089d775 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -768,9 +768,10 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) &dsp->alg_regions, list) { if (le32_to_cpu(blk->id) == alg_region->alg && type == alg_region->type) { - reg = alg_region->base + offset; + reg = alg_region->base; reg = wm_adsp_region_to_reg(mem, reg); + reg += offset; } } -- cgit v1.2.3 From 0098389564fbdfbeb08181f772205d87a18f5b37 Mon Sep 17 00:00:00 2001 From: Chris Rattray Date: Thu, 17 Jan 2013 13:11:47 +0000 Subject: ASoC: wm2200: Set system clock control register is adsp structs Allows ADSP control code to set the dsp clock rate to match the sys clock rate. Signed-off-by: Chris Rattray Signed-off-by: Mark Brown --- sound/soc/codecs/wm2200.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index 0e11184a373..fc05553cc9c 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -2224,6 +2224,9 @@ static int wm2200_i2c_probe(struct i2c_client *i2c, wm2200->dsp[i].num = i + 1; wm2200->dsp[i].dev = &i2c->dev; wm2200->dsp[i].regmap = wm2200->regmap; + wm2200->dsp[i].sysclk_reg = WM2200_CLOCKING_3; + wm2200->dsp[i].sysclk_mask = WM2200_SYSCLK_FREQ_MASK; + wm2200->dsp[i].sysclk_shift = WM2200_SYSCLK_FREQ_SHIFT; } wm2200->dsp[0].base = WM2200_DSP1_CONTROL_1; -- cgit v1.2.3 From 7c62eebbf7d1cdaec68ab9d2d4017007f9312391 Mon Sep 17 00:00:00 2001 From: Padmavathi Venna Date: Fri, 18 Jan 2013 17:17:00 +0530 Subject: ASoC: samsung: Rename samsung i2s secondary device name All Samsung SoCs has max 3 i2s controllers. So the i2s secondary fifo interface device id was named as samsung-i2s.4. Renaming this to "samsung-i2s-sec" to support device tree in i2s driver. Signed-off-by: Padmavathi Venna Signed-off-by: Mark Brown --- sound/soc/samsung/i2s.c | 31 ++++++++++++++++++++++++++++--- sound/soc/samsung/i2s.h | 7 ------- sound/soc/samsung/smdk_wm8580.c | 7 ++----- sound/soc/samsung/smdk_wm8994.c | 4 ++-- 4 files changed, 32 insertions(+), 17 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index d2d124f1dd1..ed5eeae6a48 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -29,6 +29,11 @@ #define msecs_to_loops(t) (loops_per_jiffy / 1000 * HZ * t) +enum samsung_dai_type { + TYPE_PRI, + TYPE_SEC, +}; + struct i2s_dai { /* Platform device for this DAI */ struct platform_device *pdev; @@ -981,8 +986,7 @@ static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec) i2s->i2s_dai_drv.capture.formats = SAMSUNG_I2S_FMTS; } else { /* Create a new platform_device for Secondary */ i2s->pdev = platform_device_register_resndata(NULL, - pdev->name, pdev->id + SAMSUNG_I2S_SECOFF, - NULL, 0, NULL, 0); + "samsung-i2s-sec", -1, NULL, 0, NULL, 0); if (IS_ERR(i2s->pdev)) return NULL; } @@ -993,6 +997,11 @@ static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec) return i2s; } +static inline int samsung_i2s_get_driver_data(struct platform_device *pdev) +{ + return platform_get_device_id(pdev)->driver_data; +} + static int samsung_i2s_probe(struct platform_device *pdev) { u32 dma_pl_chan, dma_cp_chan, dma_pl_sec_chan; @@ -1001,10 +1010,13 @@ static int samsung_i2s_probe(struct platform_device *pdev) struct samsung_i2s *i2s_cfg; struct resource *res; u32 regs_base, quirks; + enum samsung_dai_type samsung_dai_type; int ret = 0; /* Call during Seconday interface registration */ - if (pdev->id >= SAMSUNG_I2S_SECOFF) { + samsung_dai_type = samsung_i2s_get_driver_data(pdev); + + if (samsung_dai_type == TYPE_SEC) { sec_dai = dev_get_drvdata(&pdev->dev); snd_soc_register_dai(&sec_dai->pdev->dev, &sec_dai->i2s_dai_drv); @@ -1143,9 +1155,22 @@ static int samsung_i2s_remove(struct platform_device *pdev) return 0; } +static struct platform_device_id samsung_i2s_driver_ids[] = { + { + .name = "samsung-i2s", + .driver_data = TYPE_PRI, + }, { + .name = "samsung-i2s-sec", + .driver_data = TYPE_SEC, + }, + {}, +}; +MODULE_DEVICE_TABLE(platform, samsung-i2s-driver-ids); + static struct platform_driver samsung_i2s_driver = { .probe = samsung_i2s_probe, .remove = samsung_i2s_remove, + .id_table = samsung_i2s_driver_ids, .driver = { .name = "samsung-i2s", .owner = THIS_MODULE, diff --git a/sound/soc/samsung/i2s.h b/sound/soc/samsung/i2s.h index d420a7ca56c..7966afc934d 100644 --- a/sound/soc/samsung/i2s.h +++ b/sound/soc/samsung/i2s.h @@ -13,13 +13,6 @@ #ifndef __SND_SOC_SAMSUNG_I2S_H #define __SND_SOC_SAMSUNG_I2S_H -/* - * Maximum number of I2S blocks that any SoC can have. - * The secondary interface of a CPU dai(if there exists any), - * is indexed at [cpu-dai's ID + SAMSUNG_I2S_SECOFF] - */ -#define SAMSUNG_I2S_SECOFF 4 - #define SAMSUNG_I2S_DIV_BCLK 1 #define SAMSUNG_I2S_RCLKSRC_0 0 diff --git a/sound/soc/samsung/smdk_wm8580.c b/sound/soc/samsung/smdk_wm8580.c index 7e2b710763b..7a16b32ed67 100644 --- a/sound/soc/samsung/smdk_wm8580.c +++ b/sound/soc/samsung/smdk_wm8580.c @@ -193,9 +193,9 @@ static struct snd_soc_dai_link smdk_dai[] = { [SEC_PLAYBACK] = { /* Sec_Fifo Playback i/f */ .name = "Sec_FIFO TX", .stream_name = "Playback", - .cpu_dai_name = "samsung-i2s.x", + .cpu_dai_name = "samsung-i2s-sec", .codec_dai_name = "wm8580-hifi-playback", - .platform_name = "samsung-i2s.x", + .platform_name = "samsung-i2s-sec", .codec_name = "wm8580.0-001b", .ops = &smdk_ops, }, @@ -223,9 +223,6 @@ static int __init smdk_audio_init(void) if (machine_is_smdkc100() || machine_is_smdkv210() || machine_is_smdkc110()) { smdk.num_links = 3; - /* Secondary is at offset SAMSUNG_I2S_SECOFF from Primary */ - str = (char *)smdk_dai[SEC_PLAYBACK].cpu_dai_name; - str[strlen(str) - 1] = '0' + SAMSUNG_I2S_SECOFF; } else if (machine_is_smdk6410()) { str = (char *)smdk_dai[PRI_PLAYBACK].cpu_dai_name; str[strlen(str) - 1] = '2'; diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c index b0d0ab8bff5..cc2f407e9f1 100644 --- a/sound/soc/samsung/smdk_wm8994.c +++ b/sound/soc/samsung/smdk_wm8994.c @@ -134,9 +134,9 @@ static struct snd_soc_dai_link smdk_dai[] = { }, { /* Sec_Fifo Playback i/f */ .name = "Sec_FIFO TX", .stream_name = "Sec_Dai", - .cpu_dai_name = "samsung-i2s.4", + .cpu_dai_name = "samsung-i2s-sec", .codec_dai_name = "wm8994-aif1", - .platform_name = "samsung-i2s.4", + .platform_name = "samsung-i2s-sec", .codec_name = "wm8994-codec", .ops = &smdk_ops, }, -- cgit v1.2.3 From 33e7546e199f123808699c65274283606114e225 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 22 Jan 2013 15:51:08 +0900 Subject: ASoC: wm2000: Expose some additional registers Signed-off-by: Mark Brown --- sound/soc/codecs/wm2000.c | 3 +++ sound/soc/codecs/wm2000.h | 3 +++ 2 files changed, 6 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 85550dc319d..627c4549939 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -717,6 +717,9 @@ static bool wm2000_readable_reg(struct device *dev, unsigned int reg) { switch (reg) { case WM2000_REG_SYS_START: + case WM2000_REG_ANC_GAIN_CTRL: + case WM2000_REG_MSE_TH1: + case WM2000_REG_MSE_TH2: case WM2000_REG_SPEECH_CLARITY: case WM2000_REG_SYS_WATCHDOG: case WM2000_REG_ANA_VMID_PD_TIME: diff --git a/sound/soc/codecs/wm2000.h b/sound/soc/codecs/wm2000.h index abcd82a9399..fb812cd9e77 100644 --- a/sound/soc/codecs/wm2000.h +++ b/sound/soc/codecs/wm2000.h @@ -10,6 +10,9 @@ #define _WM2000_H #define WM2000_REG_SYS_START 0x8000 +#define WM2000_REG_ANC_GAIN_CTRL 0x8fa2 +#define WM2000_REG_MSE_TH2 0x8fdf +#define WM2000_REG_MSE_TH1 0x8fe0 #define WM2000_REG_SPEECH_CLARITY 0x8fef #define WM2000_REG_SYS_WATCHDOG 0x8ff6 #define WM2000_REG_ANA_VMID_PD_TIME 0x8ff7 -- cgit v1.2.3 From 8c3d2aa4cfeaba66be68ef8c351b2e099e50c25b Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 24 Jan 2013 09:44:28 +0000 Subject: ASoC: soc-compress: Add missing brackets around else Signed-off-by: Charles Keepax Acked-by: Vinod Koul Tested-by: Jeeja KP Signed-off-by: Mark Brown --- sound/soc/soc-compress.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 5fbfb06e808..80040f03b02 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -112,10 +112,11 @@ static int soc_compr_free(struct snd_compr_stream *cstream) snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, SND_SOC_DAPM_STREAM_STOP); - } else + } else { rtd->pop_wait = 1; schedule_delayed_work(&rtd->delayed_work, msecs_to_jiffies(rtd->pmdown_time)); + } } else { /* capture streams can be powered down now */ snd_soc_dapm_stream_event(rtd, -- cgit v1.2.3 From 15e2e6194a3ae13ffeea9b7c368b54b143f31594 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 24 Jan 2013 09:44:29 +0000 Subject: ASoC: soc-compress: Serialise compressed ops Use the pcm_mutex to serialise the compressed ops. Signed-off-by: Charles Keepax Acked-by: Vinod Koul Tested-by: Jeeja KP Signed-off-by: Mark Brown --- sound/soc/soc-compress.c | 37 ++++++++++++++++++++++++++++++++++--- 1 file changed, 34 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 80040f03b02..c48db6391ad 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -33,6 +33,8 @@ static int soc_compr_open(struct snd_compr_stream *cstream) struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret = 0; + mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + if (platform->driver->compr_ops && platform->driver->compr_ops->open) { ret = platform->driver->compr_ops->open(cstream); if (ret < 0) { @@ -61,12 +63,15 @@ static int soc_compr_open(struct snd_compr_stream *cstream) codec_dai->active++; rtd->codec->active++; + mutex_unlock(&rtd->pcm_mutex); + return 0; machine_err: if (platform->driver->compr_ops && platform->driver->compr_ops->free) platform->driver->compr_ops->free(cstream); out: + mutex_unlock(&rtd->pcm_mutex); return ret; } @@ -78,6 +83,8 @@ static int soc_compr_free(struct snd_compr_stream *cstream) struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_codec *codec = rtd->codec; + mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + if (cstream->direction == SND_COMPRESS_PLAYBACK) { cpu_dai->playback_active--; codec_dai->playback_active--; @@ -124,6 +131,7 @@ static int soc_compr_free(struct snd_compr_stream *cstream) SND_SOC_DAPM_STREAM_STOP); } + mutex_unlock(&rtd->pcm_mutex); return 0; } @@ -135,10 +143,12 @@ static int soc_compr_trigger(struct snd_compr_stream *cstream, int cmd) struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret = 0; + mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + if (platform->driver->compr_ops && platform->driver->compr_ops->trigger) { ret = platform->driver->compr_ops->trigger(cstream, cmd); if (ret < 0) - return ret; + goto out; } if (cmd == SNDRV_PCM_TRIGGER_START) @@ -146,6 +156,8 @@ static int soc_compr_trigger(struct snd_compr_stream *cstream, int cmd) else if (cmd == SNDRV_PCM_TRIGGER_STOP) snd_soc_dai_digital_mute(codec_dai, 1); +out: + mutex_unlock(&rtd->pcm_mutex); return ret; } @@ -156,6 +168,8 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream, struct snd_soc_platform *platform = rtd->platform; int ret = 0; + mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + /* first we call set_params for the platform driver * this should configure the soc side * if the machine has compressed ops then we call that as well @@ -165,18 +179,20 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream, if (platform->driver->compr_ops && platform->driver->compr_ops->set_params) { ret = platform->driver->compr_ops->set_params(cstream, params); if (ret < 0) - return ret; + goto out; } if (rtd->dai_link->compr_ops && rtd->dai_link->compr_ops->set_params) { ret = rtd->dai_link->compr_ops->set_params(cstream); if (ret < 0) - return ret; + goto out; } snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, SND_SOC_DAPM_STREAM_START); +out: + mutex_unlock(&rtd->pcm_mutex); return ret; } @@ -187,9 +203,12 @@ static int soc_compr_get_params(struct snd_compr_stream *cstream, struct snd_soc_platform *platform = rtd->platform; int ret = 0; + mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + if (platform->driver->compr_ops && platform->driver->compr_ops->get_params) ret = platform->driver->compr_ops->get_params(cstream, params); + mutex_unlock(&rtd->pcm_mutex); return ret; } @@ -200,9 +219,12 @@ static int soc_compr_get_caps(struct snd_compr_stream *cstream, struct snd_soc_platform *platform = rtd->platform; int ret = 0; + mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + if (platform->driver->compr_ops && platform->driver->compr_ops->get_caps) ret = platform->driver->compr_ops->get_caps(cstream, caps); + mutex_unlock(&rtd->pcm_mutex); return ret; } @@ -213,9 +235,12 @@ static int soc_compr_get_codec_caps(struct snd_compr_stream *cstream, struct snd_soc_platform *platform = rtd->platform; int ret = 0; + mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + if (platform->driver->compr_ops && platform->driver->compr_ops->get_codec_caps) ret = platform->driver->compr_ops->get_codec_caps(cstream, codec); + mutex_unlock(&rtd->pcm_mutex); return ret; } @@ -225,9 +250,12 @@ static int soc_compr_ack(struct snd_compr_stream *cstream, size_t bytes) struct snd_soc_platform *platform = rtd->platform; int ret = 0; + mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + if (platform->driver->compr_ops && platform->driver->compr_ops->ack) ret = platform->driver->compr_ops->ack(cstream, bytes); + mutex_unlock(&rtd->pcm_mutex); return ret; } @@ -237,9 +265,12 @@ static int soc_compr_pointer(struct snd_compr_stream *cstream, struct snd_soc_pcm_runtime *rtd = cstream->private_data; struct snd_soc_platform *platform = rtd->platform; + mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + if (platform->driver->compr_ops && platform->driver->compr_ops->pointer) platform->driver->compr_ops->pointer(cstream, tstamp); + mutex_unlock(&rtd->pcm_mutex); return 0; } -- cgit v1.2.3 From 202c8f7082e87e09f861d06b1a03501047c017b5 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 24 Jan 2013 09:44:30 +0000 Subject: ASoC: soc-compress: Initialise delayed work to power down audio Delayed work was scheduled but not initialised, this patch adds the actual work and initialises it. Signed-off-by: Charles Keepax Acked-by: Vinod Koul Tested-by: Jeeja KP Signed-off-by: Mark Brown --- sound/soc/soc-compress.c | 31 +++++++++++++++++++++++++++++++ 1 file changed, 31 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index c48db6391ad..3ea79564a99 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -75,6 +75,34 @@ out: return ret; } +/* + * Power down the audio subsystem pmdown_time msecs after close is called. + * This is to ensure there are no pops or clicks in between any music tracks + * due to DAPM power cycling. + */ +static void close_delayed_work(struct work_struct *work) +{ + struct snd_soc_pcm_runtime *rtd = + container_of(work, struct snd_soc_pcm_runtime, delayed_work.work); + struct snd_soc_dai *codec_dai = rtd->codec_dai; + + mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + + dev_dbg(rtd->dev, "ASoC: pop wq checking: %s status: %s waiting: %s\n", + codec_dai->driver->playback.stream_name, + codec_dai->playback_active ? "active" : "inactive", + rtd->pop_wait ? "yes" : "no"); + + /* are we waiting on this codec DAI stream */ + if (rtd->pop_wait == 1) { + rtd->pop_wait = 0; + snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, + SND_SOC_DAPM_STREAM_STOP); + } + + mutex_unlock(&rtd->pcm_mutex); +} + static int soc_compr_free(struct snd_compr_stream *cstream) { struct snd_soc_pcm_runtime *rtd = cstream->private_data; @@ -317,6 +345,9 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) return ret; } + /* DAPM dai link stream work */ + INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); + rtd->compr = compr; compr->private_data = rtd; -- cgit v1.2.3 From b10fedf89269a444ddfc92371acc9721749e8b41 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Thu, 24 Jan 2013 14:51:15 +0530 Subject: ASoC: tegra_wm9712: Remove __devinitconst attribute MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This has been removed from the kernel recently and gives following build errors: sound/soc/tegra/tegra_wm9712.c:155:58: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘__devinitconst’ sound/soc/tegra/tegra_wm9712.c:165:21: error: ‘tegra_wm9712_of_match’ undeclared here (not in a function) Cc: Lucas Stach Signed-off-by: Sachin Kamat Reviewed-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_wm9712.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/tegra/tegra_wm9712.c b/sound/soc/tegra/tegra_wm9712.c index cdbd2f0a23b..68d42403d9b 100644 --- a/sound/soc/tegra/tegra_wm9712.c +++ b/sound/soc/tegra/tegra_wm9712.c @@ -152,7 +152,7 @@ static int tegra_wm9712_driver_remove(struct platform_device *pdev) return 0; } -static const struct of_device_id tegra_wm9712_of_match[] __devinitconst = { +static const struct of_device_id tegra_wm9712_of_match[] = { { .compatible = "nvidia,tegra-audio-wm9712", }, {}, }; -- cgit v1.2.3 From d58579e3c3f8b6ce623de0ca26580375852b574f Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Thu, 24 Jan 2013 14:51:16 +0530 Subject: ASoC: tegra20_ac97: Remove __devinitconst attribute __devinitconst has been removed from the kernel and gives the following build errors: sound/soc/tegra/tegra20_ac97.c:460:58: error: Expected ; at end of declaration sound/soc/tegra/tegra20_ac97.c:460:58: error: got __devinitconst Cc: Lucas Stach Signed-off-by: Sachin Kamat Reviewed-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra20_ac97.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index 1bae73bf1c8..336dcdd3e8a 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -457,7 +457,7 @@ static int tegra20_ac97_platform_remove(struct platform_device *pdev) return 0; } -static const struct of_device_id tegra20_ac97_of_match[] __devinitconst = { +static const struct of_device_id tegra20_ac97_of_match[] = { { .compatible = "nvidia,tegra20-ac97", }, {}, }; -- cgit v1.2.3 From ecb2c174346f9e1b2cb671b261a16f1b9aaa66c4 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Thu, 24 Jan 2013 14:51:17 +0530 Subject: ASoC: tegra: Use NULL instead of 0 for pointers Fixes the following sparse warnings: sound/soc/tegra/tegra30_ahub.c:583:16: warning: Using plain integer as NULL pointer sound/soc/tegra/tegra30_ahub.c:600:16: warning: Using plain integer as NULL pointer Signed-off-by: Sachin Kamat Reviewed-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra30_ahub.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c index f354dc390a0..dd146f10fef 100644 --- a/sound/soc/tegra/tegra30_ahub.c +++ b/sound/soc/tegra/tegra30_ahub.c @@ -580,7 +580,7 @@ err_clk_put_apbif: clk_put(ahub->clk_apbif); err_clk_put_d_audio: clk_put(ahub->clk_d_audio); - ahub = 0; + ahub = NULL; err: return ret; } @@ -597,7 +597,7 @@ static int tegra30_ahub_remove(struct platform_device *pdev) clk_put(ahub->clk_apbif); clk_put(ahub->clk_d_audio); - ahub = 0; + ahub = NULL; return 0; } -- cgit v1.2.3 From ec05cc554eb7cfb7ca0d6d93a3afabff6bc80033 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Thu, 24 Jan 2013 14:51:18 +0530 Subject: ASoC: tegra: Staticize some functions in tegra30_i2s.c 'tegra30_i2s_startup' and 'tegra30_i2s_shutdown' are used only in this file and hence made static. Fixes the following sparse warnings: sound/soc/tegra/tegra30_i2s.c:74:5: warning: symbol 'tegra30_i2s_startup' was not declared. Should it be static? sound/soc/tegra/tegra30_i2s.c:101:6: warning: symbol 'tegra30_i2s_shutdown' was not declared. Should it be static? Signed-off-by: Sachin Kamat Reviewed-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra30_i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index 27e91dd0b91..f4e1ce82750 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -71,7 +71,7 @@ static int tegra30_i2s_runtime_resume(struct device *dev) return 0; } -int tegra30_i2s_startup(struct snd_pcm_substream *substream, +static int tegra30_i2s_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai); @@ -98,7 +98,7 @@ int tegra30_i2s_startup(struct snd_pcm_substream *substream, return ret; } -void tegra30_i2s_shutdown(struct snd_pcm_substream *substream, +static void tegra30_i2s_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai); -- cgit v1.2.3 From 36e8fe990153d015d9bb1d36f5bd70ec5b205204 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 25 Jan 2013 17:47:48 +0800 Subject: ASoC: wm_adsp: Add speaker Tx as a firmware option Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 55a0089d775..99f00f879c5 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -150,10 +150,10 @@ #define ADSP2_RAM_RDY_SHIFT 0 #define ADSP2_RAM_RDY_WIDTH 1 -#define WM_ADSP_NUM_FW 3 +#define WM_ADSP_NUM_FW 4 static const char *wm_adsp_fw_text[WM_ADSP_NUM_FW] = { - "MBC/VSS", "Tx", "Rx ANC" + "MBC/VSS", "Tx", "Tx Speaker", "Rx ANC" }; static struct { @@ -161,6 +161,7 @@ static struct { } wm_adsp_fw[WM_ADSP_NUM_FW] = { { .file = "mbc-vss" }, { .file = "tx" }, + { .file = "tx-spk" }, { .file = "rx-anc" }, }; -- cgit v1.2.3 From 7480800ea682b70b0a15cda00eed7eedc90ceb9c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 26 Jan 2013 00:29:51 +0800 Subject: ASoC: wm_adsp: Accept 0 as a parameter block address Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 85 +++++++++++++++++++--------------------------- 1 file changed, 35 insertions(+), 50 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 99f00f879c5..88831648454 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -590,27 +590,21 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) be32_to_cpu(adsp1_alg[i].dm), be32_to_cpu(adsp1_alg[i].zm)); - if (adsp1_alg[i].dm) { - region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; - region->type = WMFW_ADSP1_DM; - region->alg = be32_to_cpu(adsp1_alg[i].alg.id); - region->base = be32_to_cpu(adsp1_alg[i].dm); - list_add_tail(®ion->list, - &dsp->alg_regions); - } + region = kzalloc(sizeof(*region), GFP_KERNEL); + if (!region) + return -ENOMEM; + region->type = WMFW_ADSP1_DM; + region->alg = be32_to_cpu(adsp1_alg[i].alg.id); + region->base = be32_to_cpu(adsp1_alg[i].dm); + list_add_tail(®ion->list, &dsp->alg_regions); - if (adsp1_alg[i].zm) { - region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; - region->type = WMFW_ADSP1_ZM; - region->alg = be32_to_cpu(adsp1_alg[i].alg.id); - region->base = be32_to_cpu(adsp1_alg[i].zm); - list_add_tail(®ion->list, - &dsp->alg_regions); - } + region = kzalloc(sizeof(*region), GFP_KERNEL); + if (!region) + return -ENOMEM; + region->type = WMFW_ADSP1_ZM; + region->alg = be32_to_cpu(adsp1_alg[i].alg.id); + region->base = be32_to_cpu(adsp1_alg[i].zm); + list_add_tail(®ion->list, &dsp->alg_regions); break; case WMFW_ADSP2: @@ -624,38 +618,29 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) be32_to_cpu(adsp2_alg[i].ym), be32_to_cpu(adsp2_alg[i].zm)); - if (adsp2_alg[i].xm) { - region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; - region->type = WMFW_ADSP2_XM; - region->alg = be32_to_cpu(adsp2_alg[i].alg.id); - region->base = be32_to_cpu(adsp2_alg[i].xm); - list_add_tail(®ion->list, - &dsp->alg_regions); - } + region = kzalloc(sizeof(*region), GFP_KERNEL); + if (!region) + return -ENOMEM; + region->type = WMFW_ADSP2_XM; + region->alg = be32_to_cpu(adsp2_alg[i].alg.id); + region->base = be32_to_cpu(adsp2_alg[i].xm); + list_add_tail(®ion->list, &dsp->alg_regions); - if (adsp2_alg[i].ym) { - region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; - region->type = WMFW_ADSP2_YM; - region->alg = be32_to_cpu(adsp2_alg[i].alg.id); - region->base = be32_to_cpu(adsp2_alg[i].ym); - list_add_tail(®ion->list, - &dsp->alg_regions); - } + region = kzalloc(sizeof(*region), GFP_KERNEL); + if (!region) + return -ENOMEM; + region->type = WMFW_ADSP2_YM; + region->alg = be32_to_cpu(adsp2_alg[i].alg.id); + region->base = be32_to_cpu(adsp2_alg[i].ym); + list_add_tail(®ion->list, &dsp->alg_regions); - if (adsp2_alg[i].zm) { - region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; - region->type = WMFW_ADSP2_ZM; - region->alg = be32_to_cpu(adsp2_alg[i].alg.id); - region->base = be32_to_cpu(adsp2_alg[i].zm); - list_add_tail(®ion->list, - &dsp->alg_regions); - } + region = kzalloc(sizeof(*region), GFP_KERNEL); + if (!region) + return -ENOMEM; + region->type = WMFW_ADSP2_ZM; + region->alg = be32_to_cpu(adsp2_alg[i].alg.id); + region->base = be32_to_cpu(adsp2_alg[i].zm); + list_add_tail(®ion->list, &dsp->alg_regions); break; } } -- cgit v1.2.3 From 2d77828d9904494d3c7424189ee38cc07950df5e Mon Sep 17 00:00:00 2001 From: Padmavathi Venna Date: Thu, 24 Jan 2013 18:05:31 +0530 Subject: ASoC: Samsung: Add I2S S/W RST in startup function I2S module need to be reset after S2R. Keeping the S/W rst control part in resume didn't help in playing audio after resume. So this patch adds S/W RST control part in startup function which gets triggered for every new audio stream playback. Signed-off-by: Padmavathi Venna Signed-off-by: R. Chandrasekar Signed-off-by: Mark Brown --- sound/soc/samsung/i2s.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index ed5eeae6a48..808df74c324 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -656,6 +656,9 @@ static int i2s_startup(struct snd_pcm_substream *substream, /* Enforce set_sysclk in Master mode */ i2s->rclk_srcrate = 0; + if (!any_active(i2s) && (i2s->quirks & QUIRK_NEED_RSTCLR)) + writel(CON_RSTCLR, i2s->addr + I2SCON); + spin_unlock_irqrestore(&lock, flags); return 0; -- cgit v1.2.3 From 0099d24c6bb584d3bac76ca81bb55f3fd8ddfb21 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Fri, 25 Jan 2013 09:43:43 +0300 Subject: ASoC: dwc: fix support for more than two channels MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit There were missing break statements so everything used TWO_CHANNEL_SUPPORT. Also I added a return statement to silence a GCC warning: sound/soc/dwc/designware_i2s.c: In function ‘dw_i2s_hw_params’: sound/soc/dwc/designware_i2s.c:236:32: warning: ‘ch_reg’ may be used uninitialized in this function [-Wmaybe-uninitialized] Signed-off-by: Dan Carpenter Acked-by: Rajeev Kumar Signed-off-by: Mark Brown --- sound/soc/dwc/designware_i2s.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index 1aa51300c56..deb30d59965 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -210,15 +210,19 @@ static int dw_i2s_hw_params(struct snd_pcm_substream *substream, switch (config->chan_nr) { case EIGHT_CHANNEL_SUPPORT: ch_reg = 3; + break; case SIX_CHANNEL_SUPPORT: ch_reg = 2; + break; case FOUR_CHANNEL_SUPPORT: ch_reg = 1; + break; case TWO_CHANNEL_SUPPORT: ch_reg = 0; break; default: dev_err(dev->dev, "channel not supported\n"); + return -EINVAL; } i2s_disable_channels(dev, substream->stream); -- cgit v1.2.3 From a7930ed458afeacb029cee2b22f77b2a15472ad6 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 14 Jan 2013 18:36:04 -0800 Subject: ASoC: add snd_soc_of_parse_daifmt() for DeviceTree This patch adds snd_soc_of_parse_daifmt() and supports below style on DT. [prefix]format = "i2c"; [prefix]clock-gating = "continuous"; [prefix]bitclock-inversion; [prefix]bitclock-master; [prefix]frame-master; Each driver can use specific [prefix] (ex simple-card,cpu,dai,format = xxx;) This sample will be SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CONT | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBM_CFM Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 115 +++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 115 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2370063b582..9d07dc03a1d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4208,6 +4208,121 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(snd_soc_of_parse_audio_routing); +unsigned int snd_soc_of_parse_daifmt(struct device_node *np, + const char *prefix) +{ + int ret, i; + char prop[128]; + unsigned int format = 0; + int bit, frame; + const char *str; + struct { + char *name; + unsigned int val; + } of_fmt_table[] = { + { "i2s", SND_SOC_DAIFMT_I2S }, + { "right_j", SND_SOC_DAIFMT_RIGHT_J }, + { "left_j", SND_SOC_DAIFMT_LEFT_J }, + { "dsp_a", SND_SOC_DAIFMT_DSP_A }, + { "dsp_b", SND_SOC_DAIFMT_DSP_B }, + { "ac97", SND_SOC_DAIFMT_AC97 }, + { "pdm", SND_SOC_DAIFMT_PDM}, + { "msb", SND_SOC_DAIFMT_MSB }, + { "lsb", SND_SOC_DAIFMT_LSB }, + }, of_clock_table[] = { + { "continuous", SND_SOC_DAIFMT_CONT }, + { "gated", SND_SOC_DAIFMT_GATED }, + }; + + if (!prefix) + prefix = ""; + + /* + * check "[prefix]format = xxx" + * SND_SOC_DAIFMT_FORMAT_MASK area + */ + snprintf(prop, sizeof(prop), "%sformat", prefix); + ret = of_property_read_string(np, prop, &str); + if (ret == 0) { + for (i = 0; i < ARRAY_SIZE(of_fmt_table); i++) { + if (strcmp(str, of_fmt_table[i].name) == 0) { + format |= of_fmt_table[i].val; + break; + } + } + } + + /* + * check "[prefix]clock-gating = xxx" + * SND_SOC_DAIFMT_CLOCK_MASK area + */ + snprintf(prop, sizeof(prop), "%sclock-gating", prefix); + ret = of_property_read_string(np, prop, &str); + if (ret == 0) { + for (i = 0; i < ARRAY_SIZE(of_clock_table); i++) { + if (strcmp(str, of_clock_table[i].name) == 0) { + format |= of_clock_table[i].val; + break; + } + } + } + + /* + * check "[prefix]bitclock-inversion" + * check "[prefix]frame-inversion" + * SND_SOC_DAIFMT_INV_MASK area + */ + snprintf(prop, sizeof(prop), "%sbitclock-inversion", prefix); + bit = !!of_get_property(np, prop, NULL); + + snprintf(prop, sizeof(prop), "%sframe-inversion", prefix); + frame = !!of_get_property(np, prop, NULL); + + switch ((bit << 4) + frame) { + case 0x11: + format |= SND_SOC_DAIFMT_IB_IF; + break; + case 0x10: + format |= SND_SOC_DAIFMT_IB_NF; + break; + case 0x01: + format |= SND_SOC_DAIFMT_NB_IF; + break; + default: + /* SND_SOC_DAIFMT_NB_NF is default */ + break; + } + + /* + * check "[prefix]bitclock-master" + * check "[prefix]frame-master" + * SND_SOC_DAIFMT_MASTER_MASK area + */ + snprintf(prop, sizeof(prop), "%sbitclock-master", prefix); + bit = !!of_get_property(np, prop, NULL); + + snprintf(prop, sizeof(prop), "%sframe-master", prefix); + frame = !!of_get_property(np, prop, NULL); + + switch ((bit << 4) + frame) { + case 0x11: + format |= SND_SOC_DAIFMT_CBM_CFM; + break; + case 0x10: + format |= SND_SOC_DAIFMT_CBM_CFS; + break; + case 0x01: + format |= SND_SOC_DAIFMT_CBS_CFM; + break; + default: + format |= SND_SOC_DAIFMT_CBS_CFS; + break; + } + + return format; +} +EXPORT_SYMBOL_GPL(snd_soc_of_parse_daifmt); + static int __init snd_soc_init(void) { #ifdef CONFIG_DEBUG_FS -- cgit v1.2.3 From dde109fb46256f61642ba9bf0199418dfaaca12b Mon Sep 17 00:00:00 2001 From: Michal Bachraty Date: Fri, 18 Jan 2013 10:17:00 +0100 Subject: ASoC: McASP: Fix data rotation for playback. Enables 24bit audio playback u32 rotate = (32 - word_length) / 4; This implementation is wrong, but it works only for 16, or 32 bit audio data. (rotation for 16 or 32 bit is same as in code I present) Mcasp rotated data in 4 bits (max value 0x7)and then masks them . That data are sended to i2s bus. For 24 bit or 20 bit or other data formats, this code rotates data badly and you hear somethink like noise. You need to use u32 rotate = (word_length / 4) & 0x7; to proper data rotation. Signed-off-by: Michal Bachraty Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 55e2bf652be..9321e5c9d8c 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -626,7 +626,7 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, int word_length) { u32 fmt; - u32 rotate = (32 - word_length) / 4; + u32 rotate = (word_length / 4) & 0x7; u32 mask = (1ULL << word_length) - 1; /* -- cgit v1.2.3 From 2d30b5751d3fdcc589985929ffb3969520f6c828 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 28 Jan 2013 20:18:17 +0800 Subject: ASoC: wm_adsp: Ensure ADSP2 DMAs are quiesced when DSP is halted Maximise robustness for the widest range of firmwares by ensuring the DSP is in a consistent state when halted. Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index b6b65483758..93d03bc0661 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -103,9 +103,12 @@ #define ADSP1_START_SHIFT 0 /* DSP1_START */ #define ADSP1_START_WIDTH 1 /* DSP1_START */ -#define ADSP2_CONTROL 0 -#define ADSP2_CLOCKING 1 -#define ADSP2_STATUS1 4 +#define ADSP2_CONTROL 0x0 +#define ADSP2_CLOCKING 0x1 +#define ADSP2_STATUS1 0x4 +#define ADSP2_WDMA_CONFIG_1 0x30 +#define ADSP2_WDMA_CONFIG_2 0x31 +#define ADSP2_RDMA_CONFIG_1 0x34 /* * ADSP2 Control @@ -642,6 +645,11 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, ADSP2_SYS_ENA | ADSP2_CORE_ENA | ADSP2_START, 0); + /* Make sure DMAs are quiesced */ + regmap_write(dsp->regmap, dsp->base + ADSP2_WDMA_CONFIG_1, 0); + regmap_write(dsp->regmap, dsp->base + ADSP2_WDMA_CONFIG_2, 0); + regmap_write(dsp->regmap, dsp->base + ADSP2_RDMA_CONFIG_1, 0); + if (dsp->dvfs) { ret = regulator_set_voltage(dsp->dvfs, 1200000, 1800000); -- cgit v1.2.3 From 40476f61897933d524b7069a6df65629a469d922 Mon Sep 17 00:00:00 2001 From: Padmavathi Venna Date: Fri, 18 Jan 2013 17:17:01 +0530 Subject: ASoC: samsung: Add DT support for i2s Add support for device based discovery. Signed-off-by: Padmavathi Venna Signed-off-by: Mark Brown --- sound/soc/samsung/dma.c | 3 +- sound/soc/samsung/dma.h | 1 + sound/soc/samsung/i2s.c | 209 +++++++++++++++++++++++++++++++++++++----------- 3 files changed, 167 insertions(+), 46 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index db87628d763..21b79262010 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -174,7 +174,8 @@ static int dma_hw_params(struct snd_pcm_substream *substream, config.width = prtd->params->dma_size; config.fifo = prtd->params->dma_addr; prtd->params->ch = prtd->params->ops->request( - prtd->params->channel, &req); + prtd->params->channel, &req, rtd->cpu_dai->dev, + prtd->params->ch_name); prtd->params->ops->config(prtd->params->ch, &config); } diff --git a/sound/soc/samsung/dma.h b/sound/soc/samsung/dma.h index 73d8c7c8a1e..189a7a6d502 100644 --- a/sound/soc/samsung/dma.h +++ b/sound/soc/samsung/dma.h @@ -19,6 +19,7 @@ struct s3c_dma_params { int dma_size; /* Size of the DMA transfer */ unsigned ch; struct samsung_dma_ops *ops; + char *ch_name; }; int asoc_dma_platform_register(struct device *dev); diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 808df74c324..2fc42f9bf96 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -15,11 +15,15 @@ #include #include #include +#include +#include #include #include #include +#include + #include #include "dma.h" @@ -34,6 +38,10 @@ enum samsung_dai_type { TYPE_SEC, }; +struct samsung_i2s_dai_data { + int dai_type; +}; + struct i2s_dai { /* Platform device for this DAI */ struct platform_device *pdev; @@ -71,6 +79,7 @@ struct i2s_dai { u32 suspend_i2smod; u32 suspend_i2scon; u32 suspend_i2spsr; + unsigned long gpios[7]; /* i2s gpio line numbers */ }; /* Lock for cross i/f checks */ @@ -1000,19 +1009,76 @@ static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec) return i2s; } +#ifdef CONFIG_OF +static int samsung_i2s_parse_dt_gpio(struct i2s_dai *i2s) +{ + struct device *dev = &i2s->pdev->dev; + int index, gpio, ret; + + for (index = 0; index < 7; index++) { + gpio = of_get_gpio(dev->of_node, index); + if (!gpio_is_valid(gpio)) { + dev_err(dev, "invalid gpio[%d]: %d\n", index, gpio); + goto free_gpio; + } + + ret = gpio_request(gpio, dev_name(dev)); + if (ret) { + dev_err(dev, "gpio [%d] request failed\n", gpio); + goto free_gpio; + } + i2s->gpios[index] = gpio; + } + return 0; + +free_gpio: + while (--index >= 0) + gpio_free(i2s->gpios[index]); + return -EINVAL; +} + +static void samsung_i2s_dt_gpio_free(struct i2s_dai *i2s) +{ + unsigned int index; + for (index = 0; index < 7; index++) + gpio_free(i2s->gpios[index]); +} +#else +static int samsung_i2s_parse_dt_gpio(struct i2s_dai *dai) +{ + return -EINVAL; +} + +static void samsung_i2s_dt_gpio_free(struct i2s_dai *dai) +{ +} + +#endif + +static const struct of_device_id exynos_i2s_match[]; + static inline int samsung_i2s_get_driver_data(struct platform_device *pdev) { - return platform_get_device_id(pdev)->driver_data; +#ifdef CONFIG_OF + struct samsung_i2s_dai_data *data; + if (pdev->dev.of_node) { + const struct of_device_id *match; + match = of_match_node(exynos_i2s_match, pdev->dev.of_node); + data = (struct samsung_i2s_dai_data *) match->data; + return data->dai_type; + } else +#endif + return platform_get_device_id(pdev)->driver_data; } static int samsung_i2s_probe(struct platform_device *pdev) { - u32 dma_pl_chan, dma_cp_chan, dma_pl_sec_chan; struct i2s_dai *pri_dai, *sec_dai = NULL; - struct s3c_audio_pdata *i2s_pdata; - struct samsung_i2s *i2s_cfg; + struct s3c_audio_pdata *i2s_pdata = pdev->dev.platform_data; + struct samsung_i2s *i2s_cfg = NULL; struct resource *res; - u32 regs_base, quirks; + u32 regs_base, quirks = 0, idma_addr = 0; + struct device_node *np = pdev->dev.of_node; enum samsung_dai_type samsung_dai_type; int ret = 0; @@ -1027,31 +1093,60 @@ static int samsung_i2s_probe(struct platform_device *pdev) return 0; } - i2s_pdata = pdev->dev.platform_data; - if (i2s_pdata == NULL) { - dev_err(&pdev->dev, "Can't work without s3c_audio_pdata\n"); - return -EINVAL; + pri_dai = i2s_alloc_dai(pdev, false); + if (!pri_dai) { + dev_err(&pdev->dev, "Unable to alloc I2S_pri\n"); + return -ENOMEM; } - res = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!res) { - dev_err(&pdev->dev, "Unable to get I2S-TX dma resource\n"); - return -ENXIO; - } - dma_pl_chan = res->start; + if (!np) { + res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!res) { + dev_err(&pdev->dev, + "Unable to get I2S-TX dma resource\n"); + return -ENXIO; + } + pri_dai->dma_playback.channel = res->start; - res = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!res) { - dev_err(&pdev->dev, "Unable to get I2S-RX dma resource\n"); - return -ENXIO; - } - dma_cp_chan = res->start; + res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!res) { + dev_err(&pdev->dev, + "Unable to get I2S-RX dma resource\n"); + return -ENXIO; + } + pri_dai->dma_capture.channel = res->start; - res = platform_get_resource(pdev, IORESOURCE_DMA, 2); - if (res) - dma_pl_sec_chan = res->start; - else - dma_pl_sec_chan = 0; + if (i2s_pdata == NULL) { + dev_err(&pdev->dev, "Can't work without s3c_audio_pdata\n"); + return -EINVAL; + } + + if (&i2s_pdata->type) + i2s_cfg = &i2s_pdata->type.i2s; + + if (i2s_cfg) { + quirks = i2s_cfg->quirks; + idma_addr = i2s_cfg->idma_addr; + } + } else { + if (of_find_property(np, "samsung,supports-6ch", NULL)) + quirks |= QUIRK_PRI_6CHAN; + + if (of_find_property(np, "samsung,supports-secdai", NULL)) + quirks |= QUIRK_SEC_DAI; + + if (of_find_property(np, "samsung,supports-rstclr", NULL)) + quirks |= QUIRK_NEED_RSTCLR; + + if (of_property_read_u32(np, "samsung,idma-addr", + &idma_addr)) { + if (quirks & QUIRK_SEC_DAI) { + dev_err(&pdev->dev, "idma address is not"\ + "specified"); + return -EINVAL; + } + } + } res = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!res) { @@ -1066,24 +1161,14 @@ static int samsung_i2s_probe(struct platform_device *pdev) } regs_base = res->start; - i2s_cfg = &i2s_pdata->type.i2s; - quirks = i2s_cfg->quirks; - - pri_dai = i2s_alloc_dai(pdev, false); - if (!pri_dai) { - dev_err(&pdev->dev, "Unable to alloc I2S_pri\n"); - ret = -ENOMEM; - goto err; - } - pri_dai->dma_playback.dma_addr = regs_base + I2STXD; pri_dai->dma_capture.dma_addr = regs_base + I2SRXD; pri_dai->dma_playback.client = (struct s3c2410_dma_client *)&pri_dai->dma_playback; + pri_dai->dma_playback.ch_name = "tx"; pri_dai->dma_capture.client = (struct s3c2410_dma_client *)&pri_dai->dma_capture; - pri_dai->dma_playback.channel = dma_pl_chan; - pri_dai->dma_capture.channel = dma_cp_chan; + pri_dai->dma_capture.ch_name = "rx"; pri_dai->dma_playback.dma_size = 4; pri_dai->dma_capture.dma_size = 4; pri_dai->base = regs_base; @@ -1102,20 +1187,34 @@ static int samsung_i2s_probe(struct platform_device *pdev) sec_dai->dma_playback.dma_addr = regs_base + I2STXDS; sec_dai->dma_playback.client = (struct s3c2410_dma_client *)&sec_dai->dma_playback; - /* Use iDMA always if SysDMA not provided */ - sec_dai->dma_playback.channel = dma_pl_sec_chan ? : -1; + sec_dai->dma_playback.ch_name = "tx-sec"; + + if (!np) { + res = platform_get_resource(pdev, IORESOURCE_DMA, 2); + if (res) + sec_dai->dma_playback.channel = res->start; + } + sec_dai->dma_playback.dma_size = 4; sec_dai->base = regs_base; sec_dai->quirks = quirks; - sec_dai->idma_playback.dma_addr = i2s_cfg->idma_addr; + sec_dai->idma_playback.dma_addr = idma_addr; sec_dai->pri_dai = pri_dai; pri_dai->sec_dai = sec_dai; } - if (i2s_pdata->cfg_gpio && i2s_pdata->cfg_gpio(pdev)) { - dev_err(&pdev->dev, "Unable to configure gpio\n"); - ret = -EINVAL; - goto err; + if (np) { + if (samsung_i2s_parse_dt_gpio(pri_dai)) { + dev_err(&pdev->dev, "Unable to configure gpio\n"); + ret = -EINVAL; + goto err; + } + } else { + if (i2s_pdata->cfg_gpio && i2s_pdata->cfg_gpio(pdev)) { + dev_err(&pdev->dev, "Unable to configure gpio\n"); + ret = -EINVAL; + goto err; + } } snd_soc_register_dai(&pri_dai->pdev->dev, &pri_dai->i2s_dai_drv); @@ -1135,10 +1234,14 @@ static int samsung_i2s_remove(struct platform_device *pdev) { struct i2s_dai *i2s, *other; struct resource *res; + struct s3c_audio_pdata *i2s_pdata = pdev->dev.platform_data; i2s = dev_get_drvdata(&pdev->dev); other = i2s->pri_dai ? : i2s->sec_dai; + if (!i2s_pdata->cfg_gpio && pdev->dev.of_node) + samsung_i2s_dt_gpio_free(i2s->pri_dai); + if (other) { other->pri_dai = NULL; other->sec_dai = NULL; @@ -1170,6 +1273,21 @@ static struct platform_device_id samsung_i2s_driver_ids[] = { }; MODULE_DEVICE_TABLE(platform, samsung-i2s-driver-ids); +#ifdef CONFIG_OF +static struct samsung_i2s_dai_data samsung_i2s_dai_data_array[] = { + [TYPE_PRI] = { TYPE_PRI }, + [TYPE_SEC] = { TYPE_SEC }, +}; + +static const struct of_device_id exynos_i2s_match[] = { + { .compatible = "samsung,i2s-v5", + .data = &samsung_i2s_dai_data_array[TYPE_PRI], + }, + {}, +}; +MODULE_DEVICE_TABLE(of, exynos_i2s_match); +#endif + static struct platform_driver samsung_i2s_driver = { .probe = samsung_i2s_probe, .remove = samsung_i2s_remove, @@ -1177,6 +1295,7 @@ static struct platform_driver samsung_i2s_driver = { .driver = { .name = "samsung-i2s", .owner = THIS_MODULE, + .of_match_table = of_match_ptr(exynos_i2s_match), }, }; -- cgit v1.2.3 From 28a480583361b8e67b0a7f4898180725b71cceec Mon Sep 17 00:00:00 2001 From: Padmavathi Venna Date: Fri, 18 Jan 2013 17:17:06 +0530 Subject: ASoC: SMDK: WM8994: Add device tree support for machine file Add the basic device tree based lookup. Signed-off-by: Padmavathi Venna Signed-off-by: Mark Brown --- sound/soc/samsung/smdk_wm8994.c | 26 ++++++++++++++++++++++++++ 1 file changed, 26 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c index cc2f407e9f1..581ea4a06fc 100644 --- a/sound/soc/samsung/smdk_wm8994.c +++ b/sound/soc/samsung/smdk_wm8994.c @@ -10,6 +10,7 @@ #include "../codecs/wm8994.h" #include #include +#include /* * Default CFG switch settings to use this driver: @@ -153,9 +154,25 @@ static struct snd_soc_card smdk = { static int smdk_audio_probe(struct platform_device *pdev) { int ret; + struct device_node *np = pdev->dev.of_node; struct snd_soc_card *card = &smdk; card->dev = &pdev->dev; + + if (np) { + smdk_dai[0].cpu_dai_name = NULL; + smdk_dai[0].cpu_of_node = of_parse_phandle(np, + "samsung,i2s-controller", 0); + if (!smdk_dai[0].cpu_of_node) { + dev_err(&pdev->dev, + "Property 'samsung,i2s-controller' missing or invalid\n"); + ret = -EINVAL; + } + + smdk_dai[0].platform_name = NULL; + smdk_dai[0].platform_of_node = smdk_dai[0].cpu_of_node; + } + ret = snd_soc_register_card(card); if (ret) @@ -173,10 +190,19 @@ static int smdk_audio_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_OF +static const struct of_device_id samsung_wm8994_of_match[] = { + { .compatible = "samsung,smdk-wm8994", }, + {}, +}; +MODULE_DEVICE_TABLE(of, samsung_wm8994_of_match); +#endif /* CONFIG_OF */ + static struct platform_driver smdk_audio_driver = { .driver = { .name = "smdk-audio", .owner = THIS_MODULE, + .of_match_table = of_match_ptr(samsung_wm8994_of_match), }, .probe = smdk_audio_probe, .remove = smdk_audio_remove, -- cgit v1.2.3 From 313367e7bfa9de924245ad65c83c32c073b2fdfc Mon Sep 17 00:00:00 2001 From: Padmavathi Venna Date: Fri, 18 Jan 2013 17:17:08 +0530 Subject: ASoC: Samsung: Update Kconfig for I2S,SPDIF and PCM audio Update Kconfig file to enable I2S,PCM audio for wm8994 and spdif on all samsung platforms. Signed-off-by: Sangsu Park Signed-off-by: Sangbeom Kim Signed-off-by: Padmavathi Venna Signed-off-by: Mark Brown --- sound/soc/samsung/Kconfig | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 3c7c3a59ed3..90e7e665323 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -63,7 +63,7 @@ config SND_SOC_SAMSUNG_SMDK_WM8580 config SND_SOC_SAMSUNG_SMDK_WM8994 tristate "SoC I2S Audio support for WM8994 on SMDK" - depends on SND_SOC_SAMSUNG && (MACH_SMDKV310 || MACH_SMDKC210 || MACH_SMDK4212) + depends on SND_SOC_SAMSUNG depends on I2C=y && GENERIC_HARDIRQS select MFD_WM8994 select SND_SOC_WM8994 @@ -162,7 +162,7 @@ config SND_SOC_GONI_AQUILA_WM8994 config SND_SOC_SAMSUNG_SMDK_SPDIF tristate "SoC S/PDIF Audio support for SMDK" - depends on SND_SOC_SAMSUNG && (MACH_SMDKC100 || MACH_SMDKC110 || MACH_SMDKV210 || MACH_SMDKV310 || MACH_SMDK4212) + depends on SND_SOC_SAMSUNG select SND_SAMSUNG_SPDIF help Say Y if you want to add support for SoC S/PDIF audio on the SMDK. @@ -177,7 +177,7 @@ config SND_SOC_SMDK_WM8580_PCM config SND_SOC_SMDK_WM8994_PCM tristate "SoC PCM Audio support for WM8994 on SMDK" - depends on SND_SOC_SAMSUNG && (MACH_SMDKC210 || MACH_SMDKV310 || MACH_SMDK4212) + depends on SND_SOC_SAMSUNG depends on I2C=y && GENERIC_HARDIRQS select MFD_WM8994 select SND_SOC_WM8994 -- cgit v1.2.3 From 06378da45d58fc3703ce5243c7b278aa8cfadb61 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Beno=C3=AEt=20Th=C3=A9baudeau?= Date: Tue, 29 Jan 2013 21:31:48 +0100 Subject: ASoC: tlv320aic3x: Remove mono support MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Playing a mono stream on the TLV320AIC3x results in too fast playback rate. Remove mono support so that mono streams can be played correctly on this codec. Tested with imx-ssi (i.MX25) and TLV320AIC3104. Signed-off-by: Benoît Thébaudeau Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 5708a973a77..49891432af7 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1210,13 +1210,13 @@ static struct snd_soc_dai_driver aic3x_dai = { .name = "tlv320aic3x-hifi", .playback = { .stream_name = "Playback", - .channels_min = 1, + .channels_min = 2, .channels_max = 2, .rates = AIC3X_RATES, .formats = AIC3X_FORMATS,}, .capture = { .stream_name = "Capture", - .channels_min = 1, + .channels_min = 2, .channels_max = 2, .rates = AIC3X_RATES, .formats = AIC3X_FORMATS,}, -- cgit v1.2.3 From 8c2d6a9f9cfa59acfa63ee88e70d58f0ba3eaf21 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 29 Jan 2013 21:03:36 -0800 Subject: ASoC: clock gating is decided by bool on snd_soc_of_parse_daifmt() ASoC clock gate settings are continuous/gated only. This patch decides it as bool, then, gated clock will be default. Special thanks to Stephen Cc: Stephen Warren Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 20 ++++++-------------- 1 file changed, 6 insertions(+), 14 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 9d07dc03a1d..b3d75d2c966 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4229,9 +4229,6 @@ unsigned int snd_soc_of_parse_daifmt(struct device_node *np, { "pdm", SND_SOC_DAIFMT_PDM}, { "msb", SND_SOC_DAIFMT_MSB }, { "lsb", SND_SOC_DAIFMT_LSB }, - }, of_clock_table[] = { - { "continuous", SND_SOC_DAIFMT_CONT }, - { "gated", SND_SOC_DAIFMT_GATED }, }; if (!prefix) @@ -4253,19 +4250,14 @@ unsigned int snd_soc_of_parse_daifmt(struct device_node *np, } /* - * check "[prefix]clock-gating = xxx" + * check "[prefix]continuous-clock" * SND_SOC_DAIFMT_CLOCK_MASK area */ - snprintf(prop, sizeof(prop), "%sclock-gating", prefix); - ret = of_property_read_string(np, prop, &str); - if (ret == 0) { - for (i = 0; i < ARRAY_SIZE(of_clock_table); i++) { - if (strcmp(str, of_clock_table[i].name) == 0) { - format |= of_clock_table[i].val; - break; - } - } - } + snprintf(prop, sizeof(prop), "%scontinuous-clock", prefix); + if (of_get_property(np, prop, NULL)) + format |= SND_SOC_DAIFMT_CONT; + else + format |= SND_SOC_DAIFMT_GATED; /* * check "[prefix]bitclock-inversion" -- cgit v1.2.3 From 31522764c6b57e41b79220156efc5d208f2f841a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 30 Jan 2013 20:11:01 +0800 Subject: ASoC: wm_adsp: Fix support for firmware switching on DSP2 and higher Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index f3a9b559e0f..5487a94f459 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -196,7 +196,7 @@ static int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, if (adsp[e->shift_l].running) return -EBUSY; - adsp->fw = ucontrol->value.integer.value[0]; + adsp[e->shift_l].fw = ucontrol->value.integer.value[0]; return 0; } -- cgit v1.2.3 From 8e9bb423a2f70eb5f15ba5cc4423b0b6f75133e2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 23 Jan 2013 18:38:54 +0800 Subject: ASoC: wm2000: Lock state machine updates Need to ensure we don't get confused by simultaneous updates. Signed-off-by: Mark Brown --- sound/soc/codecs/wm2000.c | 31 ++++++++++++++++++++++++++++--- 1 file changed, 28 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 12bcae63a7f..eb96b876809 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -76,6 +76,8 @@ struct wm2000_priv { int anc_download_size; char *anc_download; + + struct mutex lock; }; static int wm2000_write(struct i2c_client *i2c, unsigned int reg, @@ -599,13 +601,20 @@ static int wm2000_anc_mode_put(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev); int anc_active = ucontrol->value.enumerated.item[0]; + int ret; if (anc_active > 1) return -EINVAL; + mutex_lock(&wm2000->lock); + wm2000->anc_active = anc_active; - return wm2000_anc_set_mode(wm2000); + ret = wm2000_anc_set_mode(wm2000); + + mutex_unlock(&wm2000->lock); + + return ret; } static int wm2000_speaker_get(struct snd_kcontrol *kcontrol, @@ -625,13 +634,20 @@ static int wm2000_speaker_put(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev); int val = ucontrol->value.enumerated.item[0]; + int ret; if (val > 1) return -EINVAL; + mutex_lock(&wm2000->lock); + wm2000->spk_ena = val; - return wm2000_anc_set_mode(wm2000); + ret = wm2000_anc_set_mode(wm2000); + + mutex_unlock(&wm2000->lock); + + return ret; } static const struct snd_kcontrol_new wm2000_controls[] = { @@ -648,6 +664,9 @@ static int wm2000_anc_power_event(struct snd_soc_dapm_widget *w, { struct snd_soc_codec *codec = w->codec; struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev); + int ret; + + mutex_lock(&wm2000->lock); if (SND_SOC_DAPM_EVENT_ON(event)) wm2000->anc_eng_ena = 1; @@ -655,7 +674,11 @@ static int wm2000_anc_power_event(struct snd_soc_dapm_widget *w, if (SND_SOC_DAPM_EVENT_OFF(event)) wm2000->anc_eng_ena = 0; - return wm2000_anc_set_mode(wm2000); + ret = wm2000_anc_set_mode(wm2000); + + mutex_unlock(&wm2000->lock); + + return ret; } static const struct snd_soc_dapm_widget wm2000_dapm_widgets[] = { @@ -782,6 +805,8 @@ static int wm2000_i2c_probe(struct i2c_client *i2c, return -ENOMEM; } + mutex_init(&wm2000->lock); + dev_set_drvdata(&i2c->dev, wm2000); wm2000->regmap = devm_regmap_init_i2c(i2c, &wm2000_regmap); -- cgit v1.2.3 From 3f3af6eeff3e048e27eb864519e779aedc194386 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 30 Jan 2013 21:35:44 +0800 Subject: ASoC: wm2000: Expose ANC gain adjustment No TLV information since it's not actually a direct gain control. Signed-off-by: Mark Brown --- sound/soc/codecs/wm2000.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 627c4549939..cb7fa3c7867 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -650,6 +650,7 @@ static int wm2000_speaker_put(struct snd_kcontrol *kcontrol, } static const struct snd_kcontrol_new wm2000_controls[] = { + SOC_SINGLE("ANC Volume", WM2000_REG_ANC_GAIN_CTRL, 0, 255, 0), SOC_SINGLE_BOOL_EXT("WM2000 ANC Switch", 0, wm2000_anc_mode_get, wm2000_anc_mode_put), @@ -755,6 +756,8 @@ static int wm2000_probe(struct snd_soc_codec *codec) { struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev); + snd_soc_codec_set_cache_io(codec, 16, 8, SND_SOC_REGMAP); + /* This will trigger a transition to standby mode by default */ wm2000_anc_set_mode(wm2000); -- cgit v1.2.3 From 58f5a7151efe2a3039b097883e6ba9fb24e42536 Mon Sep 17 00:00:00 2001 From: Alexey Galakhov Date: Wed, 30 Jan 2013 15:35:22 +0600 Subject: ASoC: samsung: Fix compilation error on S3C2440 The commit a08485d8 ("ASoC: Samsung: Do not register samsung audio dma device as pdev") introduced compilation error. Combination of CONFIG_CPU_S3C2440 and CONFIG_SND_S3C24XX_I2S caused undefined symbols 's3c_i2sv2_register_dai' and 's3c2412_i2s_dai' in sound/soc/samsung/s3c24xx-i2s.c. This patch fixes the problem and makes S3C2440 I2S usable again. It does not affect S3C2412 (aka I2S-v2). Signed-off-by: Alexey Galakhov Signed-off-by: Mark Brown --- sound/soc/samsung/s3c24xx-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index ee10e8704e9..13f6dd1ceb0 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -469,7 +469,7 @@ static int s3c24xx_iis_dev_probe(struct platform_device *pdev) { int ret = 0; - ret = s3c_i2sv2_register_dai(&pdev->dev, -1, &s3c2412_i2s_dai); + ret = snd_soc_register_dai(&pdev->dev, &s3c24xx_i2s_dai); if (ret) { pr_err("failed to register the dai\n"); return ret; -- cgit v1.2.3 From 5b1d3c3472f1941ab1a78575fe9ada718a7c0c25 Mon Sep 17 00:00:00 2001 From: "R. Chandrasekar" Date: Wed, 30 Jan 2013 17:41:04 +0530 Subject: ASoC: Samsung: I2S: Add support for runtime S2R This patch adds runtime suspend to resume support for I2S. I2S clk is disabled at suspend and enabled at resume. Signed-off-by: R. Chandrasekar Signed-off-by: Padmavathi Venna Signed-off-by: Mark Brown --- sound/soc/samsung/i2s.c | 26 ++++++++++++++++++++++++++ 1 file changed, 26 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 2fc42f9bf96..d7231e336a7 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1071,6 +1071,26 @@ static inline int samsung_i2s_get_driver_data(struct platform_device *pdev) return platform_get_device_id(pdev)->driver_data; } +#ifdef CONFIG_PM_RUNTIME +static int i2s_runtime_suspend(struct device *dev) +{ + struct i2s_dai *i2s = dev_get_drvdata(dev); + + clk_disable_unprepare(i2s->clk); + + return 0; +} + +static int i2s_runtime_resume(struct device *dev) +{ + struct i2s_dai *i2s = dev_get_drvdata(dev); + + clk_prepare_enable(i2s->clk); + + return 0; +} +#endif /* CONFIG_PM_RUNTIME */ + static int samsung_i2s_probe(struct platform_device *pdev) { struct i2s_dai *pri_dai, *sec_dai = NULL; @@ -1288,6 +1308,11 @@ static const struct of_device_id exynos_i2s_match[] = { MODULE_DEVICE_TABLE(of, exynos_i2s_match); #endif +static const struct dev_pm_ops samsung_i2s_pm = { + SET_RUNTIME_PM_OPS(i2s_runtime_suspend, + i2s_runtime_resume, NULL) +}; + static struct platform_driver samsung_i2s_driver = { .probe = samsung_i2s_probe, .remove = samsung_i2s_remove, @@ -1296,6 +1321,7 @@ static struct platform_driver samsung_i2s_driver = { .name = "samsung-i2s", .owner = THIS_MODULE, .of_match_table = of_match_ptr(exynos_i2s_match), + .pm = &samsung_i2s_pm, }, }; -- cgit v1.2.3 From 9fc4cd828107d0196b1ac0f5dec1f3b747542470 Mon Sep 17 00:00:00 2001 From: Lee Jones Date: Thu, 31 Jan 2013 12:34:00 +0000 Subject: ASoC: Ux500: Fix build error pertaining to missing include file Some compilers complain when building sound for MOP500: sound/soc/ux500/mop500.c:27:27: fatal error: mop500_ab8500.h: No such file or directory That file is actually in the local directory rather than include. Signed-off-by: Lee Jones Signed-off-by: Mark Brown --- sound/soc/ux500/mop500.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c index ae699073878..204b899c231 100644 --- a/sound/soc/ux500/mop500.c +++ b/sound/soc/ux500/mop500.c @@ -24,7 +24,7 @@ #include "ux500_pcm.h" #include "ux500_msp_dai.h" -#include +#include "mop500_ab8500.h" /* Define the whole MOP500 soundcard, linking platform to the codec-drivers */ struct snd_soc_dai_link mop500_dai_links[] = { -- cgit v1.2.3 From 4c97e8fedf1a7ef52e23dc06ec7a15be54ed18a4 Mon Sep 17 00:00:00 2001 From: Chris Rattray Date: Fri, 1 Feb 2013 15:40:41 +0000 Subject: ASoC: wm2200: Add Rx ANC input select control Signed-off-by: Chris Rattray Signed-off-by: Mark Brown --- sound/soc/codecs/wm2200.c | 11 +++++++++++ 1 file changed, 11 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index fc05553cc9c..fee1a18e309 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1112,6 +1112,16 @@ static int wm2200_mixer_values[] = { static WM2200_MUX_CTL_DECL(name##_aux5); \ static WM2200_MUX_CTL_DECL(name##_aux6); +static const char *wm2200_rxanc_input_sel_texts[] = { + "None", "IN1", "IN2", "IN3", +}; + +static const struct soc_enum wm2200_rxanc_input_sel = + SOC_ENUM_SINGLE(WM2200_RXANC_SRC, + WM2200_IN_RXANC_SEL_SHIFT, + ARRAY_SIZE(wm2200_rxanc_input_sel_texts), + wm2200_rxanc_input_sel_texts); + static const struct snd_kcontrol_new wm2200_snd_controls[] = { SOC_SINGLE("IN1 High Performance Switch", WM2200_IN1L_CONTROL, WM2200_IN1_OSR_SHIFT, 1, 0), @@ -1171,6 +1181,7 @@ SOC_DOUBLE_R_TLV("OUT2 Digital Volume", WM2200_DAC_DIGITAL_VOLUME_2L, digital_tlv), SOC_DOUBLE("OUT2 Switch", WM2200_PDM_1, WM2200_SPK1L_MUTE_SHIFT, WM2200_SPK1R_MUTE_SHIFT, 1, 1), +SOC_ENUM("RxANC Src", wm2200_rxanc_input_sel), }; WM2200_MIXER_ENUMS(OUT1L, WM2200_OUT1LMIX_INPUT_1_SOURCE); -- cgit v1.2.3 From 01f58153aefc158fd690b337d29ad140e963959d Mon Sep 17 00:00:00 2001 From: Ryo Tsutsui Date: Sun, 3 Feb 2013 17:18:00 +0900 Subject: ASoC: arizona: Fixed a bug in FLL fractional calculation Previously arizona_calc_fll() was checking if the target frequency is exactly divisible by reference frequency, but should have been product of the ratio and the reference frequency. Also scale down the Lamba and Theta coefficients be under 16-bits in order to match the registers. Signed-off-by: Ryo Tsutsui Signed-off-by: Charles Keepax Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/arizona.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index ef62c435848..2899cb909d1 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -910,7 +910,7 @@ static int arizona_calc_fll(struct arizona_fll *fll, cfg->n = target / (ratio * Fref); - if (target % Fref) { + if (target % (ratio * Fref)) { gcd_fll = gcd(target, ratio * Fref); arizona_fll_dbg(fll, "GCD=%u\n", gcd_fll); @@ -922,6 +922,15 @@ static int arizona_calc_fll(struct arizona_fll *fll, cfg->lambda = 0; } + /* Round down to 16bit range with cost of accuracy lost. + * Denominator must be bigger than numerator so we only + * take care of it. + */ + while (cfg->lambda >= (1 << 16)) { + cfg->theta >>= 1; + cfg->lambda >>= 1; + } + arizona_fll_dbg(fll, "N=%x THETA=%x LAMBDA=%x\n", cfg->n, cfg->theta, cfg->lambda); arizona_fll_dbg(fll, "FRATIO=%x(%d) OUTDIV=%x REFCLK_DIV=%x\n", -- cgit v1.2.3 From d8976cfd8257cd9539f19cd7fe512be468ed8118 Mon Sep 17 00:00:00 2001 From: Bo Shen Date: Thu, 31 Jan 2013 11:53:37 +0800 Subject: ASoC: atmel_pcm: make it buildable as module When build as module, it reports following error, using this patch fix it sound/soc/atmel/atmel-pcm-pdc.c:387: error: redefinition of 'atmel_pcm_pdc_platform_register' sound/soc/atmel/atmel-pcm.h:95: note: previous definition of 'atmel_pcm_pdc_platform_register' was here sound/soc/atmel/atmel-pcm-pdc.c:393: error: redefinition of 'atmel_pcm_pdc_platform_unregister' sound/soc/atmel/atmel-pcm.h:99: note: previous definition of 'atmel_pcm_pdc_platform_unregister' was here Signed-off-by: Bo Shen Signed-off-by: Mark Brown --- sound/soc/atmel/atmel-pcm.h | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/atmel/atmel-pcm.h b/sound/soc/atmel/atmel-pcm.h index bb45d20e725..12ae814eff2 100644 --- a/sound/soc/atmel/atmel-pcm.h +++ b/sound/soc/atmel/atmel-pcm.h @@ -88,7 +88,8 @@ void atmel_pcm_free(struct snd_pcm *pcm); int atmel_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma); -#ifdef CONFIG_SND_ATMEL_SOC_PDC +#if defined(CONFIG_SND_ATMEL_SOC_PDC) || \ + defined(CONFIG_SND_ATMEL_SOC_PDC_MODULE) int atmel_pcm_pdc_platform_register(struct device *dev); void atmel_pcm_pdc_platform_unregister(struct device *dev); #else @@ -101,7 +102,8 @@ static inline void atmel_pcm_pdc_platform_unregister(struct device *dev) } #endif -#ifdef CONFIG_SND_ATMEL_SOC_DMA +#if defined(CONFIG_SND_ATMEL_SOC_DMA) || \ + defined(CONFIG_SND_ATMEL_SOC_DMA_MODULE) int atmel_pcm_dma_platform_register(struct device *dev); void atmel_pcm_dma_platform_unregister(struct device *dev); #else -- cgit v1.2.3 From e08b273c38a7c049eefd72b9bbb2dcecb28f3956 Mon Sep 17 00:00:00 2001 From: Bo Shen Date: Thu, 31 Jan 2013 11:53:38 +0800 Subject: ASoC: atmel_ssc_dai: remove error set private data ssc private data has been set in ssc driver, this cause the error private data set to ssc, remove it Signed-off-by: Bo Shen Signed-off-by: Mark Brown --- sound/soc/atmel/atmel_ssc_dai.c | 10 ---------- 1 file changed, 10 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 2755750bda4..5cb849825a0 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -677,15 +677,6 @@ static int atmel_ssc_resume(struct snd_soc_dai *cpu_dai) # define atmel_ssc_resume NULL #endif /* CONFIG_PM */ -static int atmel_ssc_probe(struct snd_soc_dai *dai) -{ - struct atmel_ssc_info *ssc_p = &ssc_info[dai->id]; - - snd_soc_dai_set_drvdata(dai, ssc_p); - - return 0; -} - #define ATMEL_SSC_RATES (SNDRV_PCM_RATE_8000_96000) #define ATMEL_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ @@ -701,7 +692,6 @@ static const struct snd_soc_dai_ops atmel_ssc_dai_ops = { }; static struct snd_soc_dai_driver atmel_ssc_dai = { - .probe = atmel_ssc_probe, .suspend = atmel_ssc_suspend, .resume = atmel_ssc_resume, .playback = { -- cgit v1.2.3 From 69706028b94f10a2dc0a28af65e84ec6fd38054b Mon Sep 17 00:00:00 2001 From: Bo Shen Date: Thu, 31 Jan 2013 11:53:39 +0800 Subject: ASoC: atmel_ssc_dai: correct sequence when unload correct the sequence when unload this module Signed-off-by: Bo Shen Signed-off-by: Mark Brown --- sound/soc/atmel/atmel_ssc_dai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 5cb849825a0..e13580d6c47 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -778,8 +778,8 @@ void atmel_ssc_put_audio(int ssc_id) { struct ssc_device *ssc = ssc_info[ssc_id].ssc; - ssc_free(ssc); asoc_ssc_exit(&ssc->pdev->dev); + ssc_free(ssc); } EXPORT_SYMBOL_GPL(atmel_ssc_put_audio); -- cgit v1.2.3 From 151edfc757babd94d8b6c64d84a9ccf8f3c53320 Mon Sep 17 00:00:00 2001 From: Bo Shen Date: Thu, 31 Jan 2013 11:53:40 +0800 Subject: ASoC: sam9g20_wm8731: disable clock and correct sequence when unload disable clock and correct sequence when unload Signed-off-by: Bo Shen Signed-off-by: Mark Brown --- sound/soc/atmel/sam9g20_wm8731.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index da976291da9..2d6fbd0125b 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -305,10 +305,10 @@ static int at91sam9g20ek_audio_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); - atmel_ssc_put_audio(0); - snd_soc_unregister_card(card); - clk_put(mclk); + clk_disable(mclk); mclk = NULL; + snd_soc_unregister_card(card); + atmel_ssc_put_audio(0); return 0; } -- cgit v1.2.3 From e2e8bfdf61573c98162d1112b971d8d00f00fcf8 Mon Sep 17 00:00:00 2001 From: Hebbar Gururaja Date: Thu, 31 Jan 2013 18:23:04 +0530 Subject: ASoC: tlv320aic3x: Convert mic bias to a supply widget Convert MicBias widgets to supply widget. On tlv320aic3x, Mic bias power on/off shares the same register bits with output mic bias voltage. So, when power on mic bias, we need reclaim it to voltage value. Provide a new platform data so that the micbias voltage can be sent according to board requirement. Now since tlv320aic3x codec driver is DT aware, update dt files and functions to handle this new "micbias-vg" platform data. Because of sharing of bits, when enabling the micbias, voltage also needs to be updated. So use SND_SOC_DAPM_POST_PMU & SND_SOC_DAPM_PRE_PMD macro to create an event to handle this. Since micbias is converted to supply widget, updated machine drivers as well. This change is runtime tested on da850-evm with audio loopback (arecord|aplay) for confirmation. Signed-off-by: Hebbar Gururaja Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 83 ++++++++++++++++++++++++++++++++++++++--- sound/soc/codecs/tlv320aic3x.h | 4 ++ sound/soc/davinci/davinci-evm.c | 6 +-- sound/soc/omap/n810.c | 4 +- sound/soc/omap/rx51.c | 8 ++-- 5 files changed, 90 insertions(+), 15 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 5708a973a77..ba82ba2a713 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -85,6 +85,9 @@ struct aic3x_priv { #define AIC3X_MODEL_33 1 #define AIC3X_MODEL_3007 2 u16 model; + + /* Selects the micbias voltage */ + enum aic3x_micbias_voltage micbias_vg; }; /* @@ -195,6 +198,37 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, return ret; } +/* + * mic bias power on/off share the same register bits with + * output voltage of mic bias. when power on mic bias, we + * need reclaim it to voltage value. + * 0x0 = Powered off + * 0x1 = MICBIAS output is powered to 2.0V, + * 0x2 = MICBIAS output is powered to 2.5V + * 0x3 = MICBIAS output is connected to AVDD + */ +static int mic_bias_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* change mic bias voltage to user defined */ + snd_soc_update_bits(codec, MICBIAS_CTRL, + MICBIAS_LEVEL_MASK, + aic3x->micbias_vg << MICBIAS_LEVEL_SHIFT); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, MICBIAS_CTRL, + MICBIAS_LEVEL_MASK, 0); + break; + } + return 0; +} + static const char *aic3x_left_dac_mux[] = { "DAC_L1", "DAC_L3", "DAC_L2" }; static const char *aic3x_right_dac_mux[] = { "DAC_R1", "DAC_R3", "DAC_R2" }; static const char *aic3x_left_hpcom_mux[] = @@ -596,12 +630,9 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { 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), - 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), + SND_SOC_DAPM_SUPPLY("Mic Bias", MICBIAS_CTRL, 6, 0, + mic_bias_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), /* Output mixers */ SND_SOC_DAPM_MIXER("Left Line Mixer", SND_SOC_NOPM, 0, 0, @@ -1386,6 +1417,24 @@ static int aic3x_probe(struct snd_soc_codec *codec) if (aic3x->model == AIC3X_MODEL_3007) snd_soc_add_codec_controls(codec, &aic3x_classd_amp_gain_ctrl, 1); + /* set mic bias voltage */ + switch (aic3x->micbias_vg) { + case AIC3X_MICBIAS_2_0V: + case AIC3X_MICBIAS_2_5V: + case AIC3X_MICBIAS_AVDDV: + snd_soc_update_bits(codec, MICBIAS_CTRL, + MICBIAS_LEVEL_MASK, + (aic3x->micbias_vg) << MICBIAS_LEVEL_SHIFT); + break; + case AIC3X_MICBIAS_OFF: + /* + * noting to do. target won't enter here. This is just to avoid + * compile time warning "warning: enumeration value + * 'AIC3X_MICBIAS_OFF' not handled in switch" + */ + break; + } + aic3x_add_widgets(codec); list_add(&aic3x->list, &reset_list); @@ -1461,6 +1510,7 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, struct aic3x_setup_data *ai3x_setup; struct device_node *np = i2c->dev.of_node; int ret; + u32 value; aic3x = devm_kzalloc(&i2c->dev, sizeof(struct aic3x_priv), GFP_KERNEL); if (aic3x == NULL) { @@ -1474,6 +1524,7 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, if (pdata) { aic3x->gpio_reset = pdata->gpio_reset; aic3x->setup = pdata->setup; + aic3x->micbias_vg = pdata->micbias_vg; } else if (np) { ai3x_setup = devm_kzalloc(&i2c->dev, sizeof(*ai3x_setup), GFP_KERNEL); @@ -1493,6 +1544,26 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, aic3x->setup = ai3x_setup; } + if (!of_property_read_u32(np, "ai3x-micbias-vg", &value)) { + switch (value) { + case 1 : + aic3x->micbias_vg = AIC3X_MICBIAS_2_0V; + break; + case 2 : + aic3x->micbias_vg = AIC3X_MICBIAS_2_5V; + break; + case 3 : + aic3x->micbias_vg = AIC3X_MICBIAS_AVDDV; + break; + default : + aic3x->micbias_vg = AIC3X_MICBIAS_OFF; + dev_err(&i2c->dev, "Unsuitable MicBias voltage " + "found in DT\n"); + } + } else { + aic3x->micbias_vg = AIC3X_MICBIAS_OFF; + } + } else { aic3x->gpio_reset = -1; } diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index 6db3c41b016..e521ac3ddde 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -238,6 +238,10 @@ /* Default input volume */ #define DEFAULT_GAIN 0x20 +/* MICBIAS Control Register */ +#define MICBIAS_LEVEL_SHIFT (6) +#define MICBIAS_LEVEL_MASK (3 << 6) + /* headset detection / button API */ /* The AIC3x supports detection of stereo headsets (GND + left + right signal) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index d55e6477bff..484b22c5df5 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -116,9 +116,9 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Line Out", NULL, "RLOUT"}, /* Mic connected to (MIC3L | MIC3R) */ - {"MIC3L", NULL, "Mic Bias 2V"}, - {"MIC3R", NULL, "Mic Bias 2V"}, - {"Mic Bias 2V", NULL, "Mic Jack"}, + {"MIC3L", NULL, "Mic Bias"}, + {"MIC3R", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "Mic Jack"}, /* Line In connected to (LINE1L | LINE2L), (LINE1R | LINE2R) */ {"LINE1L", NULL, "Line In"}, diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 230b8c14484..ee7cd53aa3e 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -230,8 +230,8 @@ 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"}, + {"DMic Rate 64", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "DMic"}, }; static const char *spk_function[] = {"Off", "On"}; diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index d921ddbe3ec..3cd52574897 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -248,16 +248,16 @@ static const struct snd_soc_dapm_route audio_map[] = { {"FM Transmitter", NULL, "LLOUT"}, {"FM Transmitter", NULL, "RLOUT"}, - {"DMic Rate 64", NULL, "Mic Bias 2V"}, - {"Mic Bias 2V", NULL, "DMic"}, + {"DMic Rate 64", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "DMic"}, }; static const struct snd_soc_dapm_route audio_mapb[] = { {"b LINE2R", NULL, "MONO_LOUT"}, {"Earphone", NULL, "b HPLOUT"}, - {"LINE1L", NULL, "b Mic Bias 2.5V"}, - {"b Mic Bias 2.5V", NULL, "HS Mic"} + {"LINE1L", NULL, "b Mic Bias"}, + {"b Mic Bias", NULL, "HS Mic"} }; static const char *spk_function[] = {"Off", "On"}; -- cgit v1.2.3 From 0d2b6422529a26ac4dee06196524ba9da70cf735 Mon Sep 17 00:00:00 2001 From: Chris Rattray Date: Fri, 1 Feb 2013 15:51:41 +0000 Subject: ASoC: wm2200: correct IN2L and IN3L digital mute Signed-off-by: Chris Rattray Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm2200.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index d8c65f57465..d5371e0d890 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1126,9 +1126,9 @@ SOC_DOUBLE_R_TLV("IN3 Volume", WM2200_IN3L_CONTROL, WM2200_IN3R_CONTROL, SOC_DOUBLE_R("IN1 Digital Switch", WM2200_ADC_DIGITAL_VOLUME_1L, WM2200_ADC_DIGITAL_VOLUME_1R, WM2200_IN1L_MUTE_SHIFT, 1, 1), -SOC_DOUBLE_R("IN2 Digital Switch", WM2200_ADC_DIGITAL_VOLUME_1L, +SOC_DOUBLE_R("IN2 Digital Switch", WM2200_ADC_DIGITAL_VOLUME_2L, WM2200_ADC_DIGITAL_VOLUME_2R, WM2200_IN2L_MUTE_SHIFT, 1, 1), -SOC_DOUBLE_R("IN3 Digital Switch", WM2200_ADC_DIGITAL_VOLUME_1L, +SOC_DOUBLE_R("IN3 Digital Switch", WM2200_ADC_DIGITAL_VOLUME_3L, WM2200_ADC_DIGITAL_VOLUME_3R, WM2200_IN3L_MUTE_SHIFT, 1, 1), SOC_DOUBLE_R_TLV("IN1 Digital Volume", WM2200_ADC_DIGITAL_VOLUME_1L, -- cgit v1.2.3 From cf17c83c4ac2de13a7b158c1c27fffb30ce109c3 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 30 Jan 2013 14:37:23 +0800 Subject: ASoC: wm_adsp: Use asynchronous I/O to write firmware and coefficients Allow the regmap API to use asynchronous I/O where supported to minimise the delay between transfers, reducing firmware download times. Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 82 +++++++++++++++++++++++++++++++++++++--------- 1 file changed, 66 insertions(+), 16 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 5487a94f459..be45e2b8b59 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -15,6 +15,7 @@ #include #include #include +#include #include #include #include @@ -153,6 +154,43 @@ #define ADSP2_RAM_RDY_SHIFT 0 #define ADSP2_RAM_RDY_WIDTH 1 +struct wm_adsp_buf { + struct list_head list; + void *buf; +}; + +static struct wm_adsp_buf *wm_adsp_buf_alloc(const void *src, size_t len, + struct list_head *list) +{ + struct wm_adsp_buf *buf = kzalloc(sizeof(*buf), GFP_KERNEL); + + if (buf == NULL) + return NULL; + + buf->buf = kmemdup(src, len, GFP_KERNEL | GFP_DMA); + if (!buf->buf) { + kfree(buf); + return NULL; + } + + if (list) + list_add_tail(&buf->list, list); + + return buf; +} + +static void wm_adsp_buf_free(struct list_head *list) +{ + while (!list_empty(list)) { + struct wm_adsp_buf *buf = list_first_entry(list, + struct wm_adsp_buf, + list); + list_del(&buf->list); + kfree(buf->buf); + kfree(buf); + } +} + #define WM_ADSP_NUM_FW 4 static const char *wm_adsp_fw_text[WM_ADSP_NUM_FW] = { @@ -254,6 +292,7 @@ static unsigned int wm_adsp_region_to_reg(struct wm_adsp_region const *region, static int wm_adsp_load(struct wm_adsp *dsp) { + LIST_HEAD(buf_list); const struct firmware *firmware; struct regmap *regmap = dsp->regmap; unsigned int pos = 0; @@ -265,7 +304,7 @@ static int wm_adsp_load(struct wm_adsp *dsp) const struct wm_adsp_region *mem; const char *region_name; char *file, *text; - void *buf; + struct wm_adsp_buf *buf; unsigned int reg; int regions = 0; int ret, offset, type, sizes; @@ -420,18 +459,16 @@ static int wm_adsp_load(struct wm_adsp *dsp) } if (reg) { - buf = kmemdup(region->data, le32_to_cpu(region->len), - GFP_KERNEL | GFP_DMA); + buf = wm_adsp_buf_alloc(region->data, + le32_to_cpu(region->len), + &buf_list); if (!buf) { adsp_err(dsp, "Out of memory\n"); return -ENOMEM; } - ret = regmap_raw_write(regmap, reg, buf, - le32_to_cpu(region->len)); - - kfree(buf); - + ret = regmap_raw_write_async(regmap, reg, buf->buf, + le32_to_cpu(region->len)); if (ret != 0) { adsp_err(dsp, "%s.%d: Failed to write %d bytes at %d in %s: %d\n", @@ -445,12 +482,20 @@ static int wm_adsp_load(struct wm_adsp *dsp) pos += le32_to_cpu(region->len) + sizeof(*region); regions++; } - + + ret = regmap_async_complete(regmap); + if (ret != 0) { + adsp_err(dsp, "Failed to complete async write: %d\n", ret); + goto out_fw; + } + if (pos > firmware->size) adsp_warn(dsp, "%s.%d: %zu bytes at end of file\n", file, regions, pos - firmware->size); out_fw: + regmap_async_complete(regmap); + wm_adsp_buf_free(&buf_list); release_firmware(firmware); out: kfree(file); @@ -655,6 +700,7 @@ out: static int wm_adsp_load_coeff(struct wm_adsp *dsp) { + LIST_HEAD(buf_list); struct regmap *regmap = dsp->regmap; struct wmfw_coeff_hdr *hdr; struct wmfw_coeff_item *blk; @@ -664,7 +710,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) const char *region_name; int ret, pos, blocks, type, offset, reg; char *file; - void *buf; + struct wm_adsp_buf *buf; file = kzalloc(PAGE_SIZE, GFP_KERNEL); if (file == NULL) @@ -776,8 +822,9 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) } if (reg) { - buf = kmemdup(blk->data, le32_to_cpu(blk->len), - GFP_KERNEL | GFP_DMA); + buf = wm_adsp_buf_alloc(blk->data, + le32_to_cpu(blk->len), + &buf_list); if (!buf) { adsp_err(dsp, "Out of memory\n"); return -ENOMEM; @@ -786,27 +833,30 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) adsp_dbg(dsp, "%s.%d: Writing %d bytes at %x\n", file, blocks, le32_to_cpu(blk->len), reg); - ret = regmap_raw_write(regmap, reg, blk->data, - le32_to_cpu(blk->len)); + ret = regmap_raw_write_async(regmap, reg, buf->buf, + le32_to_cpu(blk->len)); if (ret != 0) { adsp_err(dsp, "%s.%d: Failed to write to %x in %s\n", file, blocks, reg, region_name); } - - kfree(buf); } pos += le32_to_cpu(blk->len) + sizeof(*blk); blocks++; } + ret = regmap_async_complete(regmap); + if (ret != 0) + adsp_err(dsp, "Failed to complete async write: %d\n", ret); + if (pos > firmware->size) adsp_warn(dsp, "%s.%d: %zu bytes at end of file\n", file, blocks, pos - firmware->size); out_fw: release_firmware(firmware); + wm_adsp_buf_free(&buf_list); out: kfree(file); return 0; -- cgit v1.2.3 From a92b53179d7d2d004f0379e70d41c56c4f570c5c Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 5 Feb 2013 13:43:39 +0100 Subject: ASoC: omap-pcm: No need to set constraint at open time The same constraint is going to be set in the snd_dmaengine_pcm_open() function, so there is no need to set it here as well. Signed-off-by: Peter Ujfalusi Reviewed-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/omap/omap-pcm.c | 14 +++----------- 1 file changed, 3 insertions(+), 11 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 47bdbd415ad..c722c2ef966 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -174,23 +174,15 @@ static snd_pcm_uframes_t omap_pcm_pointer(struct snd_pcm_substream *substream) static int omap_pcm_open(struct snd_pcm_substream *substream) { - struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct omap_pcm_dma_data *dma_data; - int ret; snd_soc_set_runtime_hwparams(substream, &omap_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) - return ret; - dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - ret = snd_dmaengine_pcm_open(substream, omap_dma_filter_fn, - &dma_data->dma_req); - return ret; + + return snd_dmaengine_pcm_open(substream, omap_dma_filter_fn, + &dma_data->dma_req); } static int omap_pcm_close(struct snd_pcm_substream *substream) -- cgit v1.2.3 From 1f88eb0f0660f8b58a1fe9011f3d3a350c7dd194 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 5 Feb 2013 10:41:47 +0000 Subject: ASoC: soc-compress: Add support for not memory mapped DSPs The ASoC compressed API did not implement the copy callback in its compressed ops which is required for DSPs that are not memory mapped. This patch creates a local copy of the compress ops for each runtime and modifies them with a copy callback as appropriate. Signed-off-by: Charles Keepax Acked-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/soc-compress.c | 38 +++++++++++++++++++++++++++++++++++--- 1 file changed, 35 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 3ea79564a99..c81aeecec93 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -302,6 +302,22 @@ static int soc_compr_pointer(struct snd_compr_stream *cstream, return 0; } +static int soc_compr_copy(struct snd_compr_stream *cstream, + const char __user *buf, size_t count) +{ + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_platform *platform = rtd->platform; + int ret = 0; + + mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + + if (platform->driver->compr_ops && platform->driver->compr_ops->copy) + ret = platform->driver->compr_ops->copy(cstream, buf, count); + + mutex_unlock(&rtd->pcm_mutex); + return ret; +} + /* ASoC Compress operations */ static struct snd_compr_ops soc_compr_ops = { .open = soc_compr_open, @@ -319,6 +335,7 @@ static struct snd_compr_ops soc_compr_ops = { int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) { struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_compr *compr; @@ -335,14 +352,25 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) return -ENOMEM; } - compr->ops = &soc_compr_ops; + compr->ops = devm_kzalloc(rtd->card->dev, sizeof(soc_compr_ops), + GFP_KERNEL); + if (compr->ops == NULL) { + dev_err(rtd->card->dev, "Cannot allocate compressed ops\n"); + ret = -ENOMEM; + goto compr_err; + } + memcpy(compr->ops, &soc_compr_ops, sizeof(soc_compr_ops)); + + /* Add copy callback for not memory mapped DSPs */ + if (platform->driver->compr_ops && platform->driver->compr_ops->copy) + compr->ops->copy = soc_compr_copy; + mutex_init(&compr->lock); ret = snd_compress_new(rtd->card->snd_card, num, direction, compr); if (ret < 0) { pr_err("compress asoc: can't create compress for codec %s\n", codec->name); - kfree(compr); - return ret; + goto compr_err; } /* DAPM dai link stream work */ @@ -354,4 +382,8 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) printk(KERN_INFO "compress asoc: %s <-> %s mapping ok\n", codec_dai->name, cpu_dai->name); return ret; + +compr_err: + kfree(compr); + return ret; } -- cgit v1.2.3 From 1a786243235b8a8f4762ee57f185dadd97794fa4 Mon Sep 17 00:00:00 2001 From: Chris Rattray Date: Tue, 5 Feb 2013 14:40:44 +0000 Subject: ASoC: wm2200: Provide platform data for MICBIAS configuration Signed-off-by: Chris Rattray Signed-off-by: Mark Brown --- sound/soc/codecs/wm2200.c | 31 +++++++++++++++++++++++++++++++ 1 file changed, 31 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index fee1a18e309..31d29c8f11b 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -2212,6 +2212,7 @@ static int wm2200_i2c_probe(struct i2c_client *i2c, struct wm2200_priv *wm2200; unsigned int reg; int ret, i; + int val; wm2200 = devm_kzalloc(&i2c->dev, sizeof(struct wm2200_priv), GFP_KERNEL); @@ -2362,6 +2363,36 @@ static int wm2200_i2c_probe(struct i2c_client *i2c, regmap_write(wm2200->regmap, WM2200_AUDIO_IF_1_16 + i, i); } + for (i = 0; i < WM2200_MAX_MICBIAS; i++) { + if (!wm2200->pdata.micbias[i].mb_lvl && + !wm2200->pdata.micbias[i].bypass) + continue; + + /* Apply default for bypass mode */ + if (!wm2200->pdata.micbias[i].mb_lvl) + wm2200->pdata.micbias[i].mb_lvl + = WM2200_MBIAS_LVL_1V5; + + val = (wm2200->pdata.micbias[i].mb_lvl -1) + << WM2200_MICB1_LVL_SHIFT; + + if (wm2200->pdata.micbias[i].discharge) + val |= WM2200_MICB1_DISCH; + + if (wm2200->pdata.micbias[i].fast_start) + val |= WM2200_MICB1_RATE; + + if (wm2200->pdata.micbias[i].bypass) + val |= WM2200_MICB1_MODE; + + regmap_update_bits(wm2200->regmap, + WM2200_MIC_BIAS_CTRL_1 + i, + WM2200_MICB1_LVL_MASK | + WM2200_MICB1_DISCH | + WM2200_MICB1_MODE | + WM2200_MICB1_RATE, val); + } + for (i = 0; i < ARRAY_SIZE(wm2200->pdata.in_mode); i++) { regmap_update_bits(wm2200->regmap, wm2200_mic_ctrl_reg[i], WM2200_IN1_MODE_MASK | -- cgit v1.2.3 From 9092a6ea1627d739ce2a98c2b4a7eadb23e07021 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 6 Feb 2013 17:58:57 +0000 Subject: ASoC: arizona: Fix debug logging level for FLLs and AIFs Use _dbg for debug messages. Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 2899cb909d1..3b8e8c70b79 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -56,14 +56,14 @@ #define arizona_fll_warn(_fll, fmt, ...) \ dev_warn(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__) #define arizona_fll_dbg(_fll, fmt, ...) \ - dev_err(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__) + dev_dbg(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__) #define arizona_aif_err(_dai, fmt, ...) \ dev_err(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__) #define arizona_aif_warn(_dai, fmt, ...) \ dev_warn(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__) #define arizona_aif_dbg(_dai, fmt, ...) \ - dev_err(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__) + dev_dbg(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__) const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS] = { "None", -- cgit v1.2.3 From 7e3bb169ff20972b7c238a9fda108b94e7cb9df0 Mon Sep 17 00:00:00 2001 From: Jerry Wong Date: Wed, 6 Feb 2013 11:02:33 -0800 Subject: ASoC: Replace max98090 Device Driver This patch removes the existing max98090 driver prior to installing a more complete one. Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 - sound/soc/codecs/Makefile | 2 - sound/soc/codecs/max98090.c | 577 -------------------------------------------- 3 files changed, 583 deletions(-) delete mode 100644 sound/soc/codecs/max98090.c (limited to 'sound/soc') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3a847828932..0e368d46976 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -44,7 +44,6 @@ config SND_SOC_ALL_CODECS select SND_SOC_LM4857 if I2C select SND_SOC_LM49453 if I2C select SND_SOC_MAX98088 if I2C - select SND_SOC_MAX98090 if I2C select SND_SOC_MAX98095 if I2C select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9768 if I2C @@ -268,9 +267,6 @@ config SND_SOC_LM49453 config SND_SOC_MAX98088 tristate -config SND_SOC_MAX98090 - tristate - config SND_SOC_MAX98095 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f6e8e36cceb..aa563122016 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -34,7 +34,6 @@ snd-soc-lm4857-objs := lm4857.o snd-soc-lm49453-objs := lm49453.o snd-soc-max9768-objs := max9768.o snd-soc-max98088-objs := max98088.o -snd-soc-max98090-objs := max98090.o snd-soc-max98095-objs := max98095.o snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o @@ -158,7 +157,6 @@ obj-$(CONFIG_SND_SOC_LM4857) += snd-soc-lm4857.o obj-$(CONFIG_SND_SOC_LM49453) += snd-soc-lm49453.o obj-$(CONFIG_SND_SOC_MAX9768) += snd-soc-max9768.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o -obj-$(CONFIG_SND_SOC_MAX98090) += snd-soc-max98090.o obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c deleted file mode 100644 index c9772ca3da4..00000000000 --- a/sound/soc/codecs/max98090.c +++ /dev/null @@ -1,577 +0,0 @@ -/* - * max98090.c -- MAX98090 ALSA SoC Audio driver - * based on Rev0p8 datasheet - * - * Copyright (C) 2012 Renesas Solutions Corp. - * Kuninori Morimoto - * - * Based on - * - * max98095.c - * Copyright 2011 Maxim Integrated Products - * - * https://github.com/hardkernel/linux/commit/\ - * 3417d7166b17113b3b33b0a337c74d1c7cc313df#sound/soc/codecs/max98090.c - * Copyright 2011 Maxim Integrated Products - * - * 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 - -/* - * - * MAX98090 Registers Definition - * - */ - -/* RESET / STATUS / INTERRUPT REGISTERS */ -#define MAX98090_0x00_SW_RESET 0x00 -#define MAX98090_0x01_INT_STS 0x01 -#define MAX98090_0x02_JACK_STS 0x02 -#define MAX98090_0x03_INT_MASK 0x03 - -/* QUICK SETUP REGISTERS */ -#define MAX98090_0x04_SYS_CLK 0x04 -#define MAX98090_0x05_SAMPLE_RATE 0x05 -#define MAX98090_0x06_DAI_IF 0x06 -#define MAX98090_0x07_DAC_PATH 0x07 -#define MAX98090_0x08_MIC_TO_ADC 0x08 -#define MAX98090_0x09_LINE_TO_ADC 0x09 -#define MAX98090_0x0A_ANALOG_MIC_LOOP 0x0A -#define MAX98090_0x0B_ANALOG_LINE_LOOP 0x0B - -/* ANALOG INPUT CONFIGURATION REGISTERS */ -#define MAX98090_0x0D_INPUT_CONFIG 0x0D -#define MAX98090_0x0E_LINE_IN_LVL 0x0E -#define MAX98090_0x0F_LINI_IN_CFG 0x0F -#define MAX98090_0x10_MIC1_IN_LVL 0x10 -#define MAX98090_0x11_MIC2_IN_LVL 0x11 - -/* MICROPHONE CONFIGURATION REGISTERS */ -#define MAX98090_0x12_MIC_BIAS_VOL 0x12 -#define MAX98090_0x13_DIGITAL_MIC_CFG 0x13 -#define MAX98090_0x14_DIGITAL_MIC_MODE 0x14 - -/* ADC PATH AND CONFIGURATION REGISTERS */ -#define MAX98090_0x15_L_ADC_MIX 0x15 -#define MAX98090_0x16_R_ADC_MIX 0x16 -#define MAX98090_0x17_L_ADC_LVL 0x17 -#define MAX98090_0x18_R_ADC_LVL 0x18 -#define MAX98090_0x19_ADC_BIQUAD_LVL 0x19 -#define MAX98090_0x1A_ADC_SIDETONE 0x1A - -/* CLOCK CONFIGURATION REGISTERS */ -#define MAX98090_0x1B_SYS_CLK 0x1B -#define MAX98090_0x1C_CLK_MODE 0x1C -#define MAX98090_0x1D_ANY_CLK1 0x1D -#define MAX98090_0x1E_ANY_CLK2 0x1E -#define MAX98090_0x1F_ANY_CLK3 0x1F -#define MAX98090_0x20_ANY_CLK4 0x20 -#define MAX98090_0x21_MASTER_MODE 0x21 - -/* INTERFACE CONTROL REGISTERS */ -#define MAX98090_0x22_DAI_IF_FMT 0x22 -#define MAX98090_0x23_DAI_TDM_FMT1 0x23 -#define MAX98090_0x24_DAI_TDM_FMT2 0x24 -#define MAX98090_0x25_DAI_IO_CFG 0x25 -#define MAX98090_0x26_FILTER_CFG 0x26 -#define MAX98090_0x27_DAI_PLAYBACK_LVL 0x27 -#define MAX98090_0x28_EQ_PLAYBACK_LVL 0x28 - -/* HEADPHONE CONTROL REGISTERS */ -#define MAX98090_0x29_L_HP_MIX 0x29 -#define MAX98090_0x2A_R_HP_MIX 0x2A -#define MAX98090_0x2B_HP_CTR 0x2B -#define MAX98090_0x2C_L_HP_VOL 0x2C -#define MAX98090_0x2D_R_HP_VOL 0x2D - -/* SPEAKER CONFIGURATION REGISTERS */ -#define MAX98090_0x2E_L_SPK_MIX 0x2E -#define MAX98090_0x2F_R_SPK_MIX 0x2F -#define MAX98090_0x30_SPK_CTR 0x30 -#define MAX98090_0x31_L_SPK_VOL 0x31 -#define MAX98090_0x32_R_SPK_VOL 0x32 - -/* ALC CONFIGURATION REGISTERS */ -#define MAX98090_0x33_ALC_TIMING 0x33 -#define MAX98090_0x34_ALC_COMPRESSOR 0x34 -#define MAX98090_0x35_ALC_EXPANDER 0x35 -#define MAX98090_0x36_ALC_GAIN 0x36 - -/* RECEIVER AND LINE_OUTPUT REGISTERS */ -#define MAX98090_0x37_RCV_LOUT_L_MIX 0x37 -#define MAX98090_0x38_RCV_LOUT_L_CNTL 0x38 -#define MAX98090_0x39_RCV_LOUT_L_VOL 0x39 -#define MAX98090_0x3A_LOUT_R_MIX 0x3A -#define MAX98090_0x3B_LOUT_R_CNTL 0x3B -#define MAX98090_0x3C_LOUT_R_VOL 0x3C - -/* JACK DETECT AND ENABLE REGISTERS */ -#define MAX98090_0x3D_JACK_DETECT 0x3D -#define MAX98090_0x3E_IN_ENABLE 0x3E -#define MAX98090_0x3F_OUT_ENABLE 0x3F -#define MAX98090_0x40_LVL_CTR 0x40 -#define MAX98090_0x41_DSP_FILTER_ENABLE 0x41 - -/* BIAS AND POWER MODE CONFIGURATION REGISTERS */ -#define MAX98090_0x42_BIAS_CTR 0x42 -#define MAX98090_0x43_DAC_CTR 0x43 -#define MAX98090_0x44_ADC_CTR 0x44 -#define MAX98090_0x45_DEV_SHUTDOWN 0x45 - -/* REVISION ID REGISTER */ -#define MAX98090_0xFF_REV_ID 0xFF - -#define MAX98090_REG_MAX_CACHED 0x45 -#define MAX98090_REG_END 0xFF - -/* - * - * MAX98090 Registers Bit Fields - * - */ - -/* MAX98090_0x06_DAI_IF */ -#define MAX98090_DAI_IF_MASK 0x3F -#define MAX98090_RJ_M (1 << 5) -#define MAX98090_RJ_S (1 << 4) -#define MAX98090_LJ_M (1 << 3) -#define MAX98090_LJ_S (1 << 2) -#define MAX98090_I2S_M (1 << 1) -#define MAX98090_I2S_S (1 << 0) - -/* MAX98090_0x45_DEV_SHUTDOWN */ -#define MAX98090_SHDNRUN (1 << 7) - -/* codec private data */ -struct max98090_priv { - struct regmap *regmap; -}; - -static const struct reg_default max98090_reg_defaults[] = { - /* RESET / STATUS / INTERRUPT REGISTERS */ - {MAX98090_0x00_SW_RESET, 0x00}, - {MAX98090_0x01_INT_STS, 0x00}, - {MAX98090_0x02_JACK_STS, 0x00}, - {MAX98090_0x03_INT_MASK, 0x04}, - - /* QUICK SETUP REGISTERS */ - {MAX98090_0x04_SYS_CLK, 0x00}, - {MAX98090_0x05_SAMPLE_RATE, 0x00}, - {MAX98090_0x06_DAI_IF, 0x00}, - {MAX98090_0x07_DAC_PATH, 0x00}, - {MAX98090_0x08_MIC_TO_ADC, 0x00}, - {MAX98090_0x09_LINE_TO_ADC, 0x00}, - {MAX98090_0x0A_ANALOG_MIC_LOOP, 0x00}, - {MAX98090_0x0B_ANALOG_LINE_LOOP, 0x00}, - - /* ANALOG INPUT CONFIGURATION REGISTERS */ - {MAX98090_0x0D_INPUT_CONFIG, 0x00}, - {MAX98090_0x0E_LINE_IN_LVL, 0x1B}, - {MAX98090_0x0F_LINI_IN_CFG, 0x00}, - {MAX98090_0x10_MIC1_IN_LVL, 0x11}, - {MAX98090_0x11_MIC2_IN_LVL, 0x11}, - - /* MICROPHONE CONFIGURATION REGISTERS */ - {MAX98090_0x12_MIC_BIAS_VOL, 0x00}, - {MAX98090_0x13_DIGITAL_MIC_CFG, 0x00}, - {MAX98090_0x14_DIGITAL_MIC_MODE, 0x00}, - - /* ADC PATH AND CONFIGURATION REGISTERS */ - {MAX98090_0x15_L_ADC_MIX, 0x00}, - {MAX98090_0x16_R_ADC_MIX, 0x00}, - {MAX98090_0x17_L_ADC_LVL, 0x03}, - {MAX98090_0x18_R_ADC_LVL, 0x03}, - {MAX98090_0x19_ADC_BIQUAD_LVL, 0x00}, - {MAX98090_0x1A_ADC_SIDETONE, 0x00}, - - /* CLOCK CONFIGURATION REGISTERS */ - {MAX98090_0x1B_SYS_CLK, 0x00}, - {MAX98090_0x1C_CLK_MODE, 0x00}, - {MAX98090_0x1D_ANY_CLK1, 0x00}, - {MAX98090_0x1E_ANY_CLK2, 0x00}, - {MAX98090_0x1F_ANY_CLK3, 0x00}, - {MAX98090_0x20_ANY_CLK4, 0x00}, - {MAX98090_0x21_MASTER_MODE, 0x00}, - - /* INTERFACE CONTROL REGISTERS */ - {MAX98090_0x22_DAI_IF_FMT, 0x00}, - {MAX98090_0x23_DAI_TDM_FMT1, 0x00}, - {MAX98090_0x24_DAI_TDM_FMT2, 0x00}, - {MAX98090_0x25_DAI_IO_CFG, 0x00}, - {MAX98090_0x26_FILTER_CFG, 0x80}, - {MAX98090_0x27_DAI_PLAYBACK_LVL, 0x00}, - {MAX98090_0x28_EQ_PLAYBACK_LVL, 0x00}, - - /* HEADPHONE CONTROL REGISTERS */ - {MAX98090_0x29_L_HP_MIX, 0x00}, - {MAX98090_0x2A_R_HP_MIX, 0x00}, - {MAX98090_0x2B_HP_CTR, 0x00}, - {MAX98090_0x2C_L_HP_VOL, 0x1A}, - {MAX98090_0x2D_R_HP_VOL, 0x1A}, - - /* SPEAKER CONFIGURATION REGISTERS */ - {MAX98090_0x2E_L_SPK_MIX, 0x00}, - {MAX98090_0x2F_R_SPK_MIX, 0x00}, - {MAX98090_0x30_SPK_CTR, 0x00}, - {MAX98090_0x31_L_SPK_VOL, 0x2C}, - {MAX98090_0x32_R_SPK_VOL, 0x2C}, - - /* ALC CONFIGURATION REGISTERS */ - {MAX98090_0x33_ALC_TIMING, 0x00}, - {MAX98090_0x34_ALC_COMPRESSOR, 0x00}, - {MAX98090_0x35_ALC_EXPANDER, 0x00}, - {MAX98090_0x36_ALC_GAIN, 0x00}, - - /* RECEIVER AND LINE_OUTPUT REGISTERS */ - {MAX98090_0x37_RCV_LOUT_L_MIX, 0x00}, - {MAX98090_0x38_RCV_LOUT_L_CNTL, 0x00}, - {MAX98090_0x39_RCV_LOUT_L_VOL, 0x15}, - {MAX98090_0x3A_LOUT_R_MIX, 0x00}, - {MAX98090_0x3B_LOUT_R_CNTL, 0x00}, - {MAX98090_0x3C_LOUT_R_VOL, 0x15}, - - /* JACK DETECT AND ENABLE REGISTERS */ - {MAX98090_0x3D_JACK_DETECT, 0x00}, - {MAX98090_0x3E_IN_ENABLE, 0x00}, - {MAX98090_0x3F_OUT_ENABLE, 0x00}, - {MAX98090_0x40_LVL_CTR, 0x00}, - {MAX98090_0x41_DSP_FILTER_ENABLE, 0x00}, - - /* BIAS AND POWER MODE CONFIGURATION REGISTERS */ - {MAX98090_0x42_BIAS_CTR, 0x00}, - {MAX98090_0x43_DAC_CTR, 0x00}, - {MAX98090_0x44_ADC_CTR, 0x06}, - {MAX98090_0x45_DEV_SHUTDOWN, 0x00}, -}; - -static const unsigned int max98090_hp_tlv[] = { - TLV_DB_RANGE_HEAD(5), - 0x0, 0x6, TLV_DB_SCALE_ITEM(-6700, 400, 0), - 0x7, 0xE, TLV_DB_SCALE_ITEM(-4000, 300, 0), - 0xF, 0x15, TLV_DB_SCALE_ITEM(-1700, 200, 0), - 0x16, 0x1B, TLV_DB_SCALE_ITEM(-400, 100, 0), - 0x1C, 0x1F, TLV_DB_SCALE_ITEM(150, 50, 0), -}; - -static struct snd_kcontrol_new max98090_snd_controls[] = { - SOC_DOUBLE_R_TLV("Headphone Volume", MAX98090_0x2C_L_HP_VOL, - MAX98090_0x2D_R_HP_VOL, 0, 31, 0, max98090_hp_tlv), -}; - -/* Left HeadPhone Mixer Switch */ -static struct snd_kcontrol_new max98090_left_hp_mixer_controls[] = { - SOC_DAPM_SINGLE("DACR Switch", MAX98090_0x29_L_HP_MIX, 1, 1, 0), - SOC_DAPM_SINGLE("DACL Switch", MAX98090_0x29_L_HP_MIX, 0, 1, 0), -}; - -/* Right HeadPhone Mixer Switch */ -static struct snd_kcontrol_new max98090_right_hp_mixer_controls[] = { - SOC_DAPM_SINGLE("DACR Switch", MAX98090_0x2A_R_HP_MIX, 1, 1, 0), - SOC_DAPM_SINGLE("DACL Switch", MAX98090_0x2A_R_HP_MIX, 0, 1, 0), -}; - -static struct snd_soc_dapm_widget max98090_dapm_widgets[] = { - /* Output */ - SND_SOC_DAPM_OUTPUT("HPL"), - SND_SOC_DAPM_OUTPUT("HPR"), - - /* PGA */ - SND_SOC_DAPM_PGA("HPL Out", MAX98090_0x3F_OUT_ENABLE, 7, 0, NULL, 0), - SND_SOC_DAPM_PGA("HPR Out", MAX98090_0x3F_OUT_ENABLE, 6, 0, NULL, 0), - - /* Mixer */ - SND_SOC_DAPM_MIXER("HPL Mixer", SND_SOC_NOPM, 0, 0, - max98090_left_hp_mixer_controls, - ARRAY_SIZE(max98090_left_hp_mixer_controls)), - - SND_SOC_DAPM_MIXER("HPR Mixer", SND_SOC_NOPM, 0, 0, - max98090_right_hp_mixer_controls, - ARRAY_SIZE(max98090_right_hp_mixer_controls)), - - /* DAC */ - SND_SOC_DAPM_DAC("DACL", "Hifi Playback", MAX98090_0x3F_OUT_ENABLE, 0, 0), - SND_SOC_DAPM_DAC("DACR", "Hifi Playback", MAX98090_0x3F_OUT_ENABLE, 1, 0), -}; - -static struct snd_soc_dapm_route max98090_audio_map[] = { - /* Output */ - {"HPL", NULL, "HPL Out"}, - {"HPR", NULL, "HPR Out"}, - - /* PGA */ - {"HPL Out", NULL, "HPL Mixer"}, - {"HPR Out", NULL, "HPR Mixer"}, - - /* Mixer*/ - {"HPL Mixer", "DACR Switch", "DACR"}, - {"HPL Mixer", "DACL Switch", "DACL"}, - - {"HPR Mixer", "DACR Switch", "DACR"}, - {"HPR Mixer", "DACL Switch", "DACL"}, -}; - -static bool max98090_volatile(struct device *dev, unsigned int reg) -{ - if ((reg == MAX98090_0x01_INT_STS) || - (reg == MAX98090_0x02_JACK_STS) || - (reg > MAX98090_REG_MAX_CACHED)) - return true; - - return false; -} - -static int max98090_dai_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_soc_codec *codec = dai->codec; - unsigned int val; - - switch (params_rate(params)) { - case 96000: - val = 1 << 5; - break; - case 32000: - val = 1 << 4; - break; - case 48000: - val = 1 << 3; - break; - case 44100: - val = 1 << 2; - break; - case 16000: - val = 1 << 1; - break; - case 8000: - val = 1 << 0; - break; - default: - dev_err(codec->dev, "unsupported rate\n"); - return -EINVAL; - } - snd_soc_update_bits(codec, MAX98090_0x05_SAMPLE_RATE, 0x03F, val); - - return 0; -} - -static int max98090_dai_set_sysclk(struct snd_soc_dai *dai, - int clk_id, unsigned int freq, int dir) -{ - struct snd_soc_codec *codec = dai->codec; - unsigned int val; - - snd_soc_update_bits(codec, MAX98090_0x45_DEV_SHUTDOWN, - MAX98090_SHDNRUN, 0); - - switch (freq) { - case 26000000: - val = 1 << 7; - break; - case 19200000: - val = 1 << 6; - break; - case 13000000: - val = 1 << 5; - break; - case 12288000: - val = 1 << 4; - break; - case 12000000: - val = 1 << 3; - break; - case 11289600: - val = 1 << 2; - break; - default: - dev_err(codec->dev, "Invalid master clock frequency\n"); - return -EINVAL; - } - snd_soc_update_bits(codec, MAX98090_0x04_SYS_CLK, 0xFD, val); - - snd_soc_update_bits(codec, MAX98090_0x45_DEV_SHUTDOWN, - MAX98090_SHDNRUN, MAX98090_SHDNRUN); - - dev_dbg(dai->dev, "sysclk is %uHz\n", freq); - - return 0; -} - -static int max98090_dai_set_fmt(struct snd_soc_dai *dai, - unsigned int fmt) -{ - struct snd_soc_codec *codec = dai->codec; - int is_master; - u8 val; - - /* master/slave mode */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - is_master = 1; - break; - case SND_SOC_DAIFMT_CBS_CFS: - is_master = 0; - break; - default: - dev_err(codec->dev, "unsupported clock\n"); - return -EINVAL; - } - - /* format */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_I2S: - val = (is_master) ? MAX98090_I2S_M : MAX98090_I2S_S; - break; - case SND_SOC_DAIFMT_RIGHT_J: - val = (is_master) ? MAX98090_RJ_M : MAX98090_RJ_S; - break; - case SND_SOC_DAIFMT_LEFT_J: - val = (is_master) ? MAX98090_LJ_M : MAX98090_LJ_S; - break; - default: - dev_err(codec->dev, "unsupported format\n"); - return -EINVAL; - } - snd_soc_update_bits(codec, MAX98090_0x06_DAI_IF, - MAX98090_DAI_IF_MASK, val); - - return 0; -} - -#define MAX98090_RATES SNDRV_PCM_RATE_8000_96000 -#define MAX98090_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) - -static struct snd_soc_dai_ops max98090_dai_ops = { - .set_sysclk = max98090_dai_set_sysclk, - .set_fmt = max98090_dai_set_fmt, - .hw_params = max98090_dai_hw_params, -}; - -static struct snd_soc_dai_driver max98090_dai = { - .name = "max98090-Hifi", - .playback = { - .stream_name = "Playback", - .channels_min = 1, - .channels_max = 2, - .rates = MAX98090_RATES, - .formats = MAX98090_FORMATS, - }, - .ops = &max98090_dai_ops, -}; - -static int max98090_probe(struct snd_soc_codec *codec) -{ - struct max98090_priv *priv = snd_soc_codec_get_drvdata(codec); - struct device *dev = codec->dev; - int ret; - - codec->control_data = priv->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - - /* Device active */ - snd_soc_update_bits(codec, MAX98090_0x45_DEV_SHUTDOWN, - MAX98090_SHDNRUN, MAX98090_SHDNRUN); - - return 0; -} - -static int max98090_remove(struct snd_soc_codec *codec) -{ - return 0; -} - -static struct snd_soc_codec_driver soc_codec_dev_max98090 = { - .probe = max98090_probe, - .remove = max98090_remove, - .controls = max98090_snd_controls, - .num_controls = ARRAY_SIZE(max98090_snd_controls), - .dapm_widgets = max98090_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(max98090_dapm_widgets), - .dapm_routes = max98090_audio_map, - .num_dapm_routes = ARRAY_SIZE(max98090_audio_map), -}; - -static const struct regmap_config max98090_regmap = { - .reg_bits = 8, - .val_bits = 8, - .max_register = MAX98090_REG_END, - .volatile_reg = max98090_volatile, - .cache_type = REGCACHE_RBTREE, - .reg_defaults = max98090_reg_defaults, - .num_reg_defaults = ARRAY_SIZE(max98090_reg_defaults), -}; - -static int max98090_i2c_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) -{ - struct max98090_priv *priv; - struct device *dev = &i2c->dev; - unsigned int val; - int ret; - - priv = devm_kzalloc(dev, sizeof(struct max98090_priv), - GFP_KERNEL); - if (!priv) - return -ENOMEM; - - priv->regmap = devm_regmap_init_i2c(i2c, &max98090_regmap); - if (IS_ERR(priv->regmap)) { - ret = PTR_ERR(priv->regmap); - dev_err(dev, "Failed to init regmap: %d\n", ret); - return ret; - } - - i2c_set_clientdata(i2c, priv); - - ret = regmap_read(priv->regmap, MAX98090_0xFF_REV_ID, &val); - if (ret < 0) { - dev_err(dev, "Failed to read device revision: %d\n", ret); - return ret; - } - dev_info(dev, "revision 0x%02x\n", val); - - ret = snd_soc_register_codec(dev, - &soc_codec_dev_max98090, - &max98090_dai, 1); - - return ret; -} - -static int max98090_i2c_remove(struct i2c_client *client) -{ - snd_soc_unregister_codec(&client->dev); - return 0; -} - -static const struct i2c_device_id max98090_i2c_id[] = { - { "max98090", 0 }, - { } -}; -MODULE_DEVICE_TABLE(i2c, max98090_i2c_id); - -static struct i2c_driver max98090_i2c_driver = { - .driver = { - .name = "max98090", - .owner = THIS_MODULE, - }, - .probe = max98090_i2c_probe, - .remove = max98090_i2c_remove, - .id_table = max98090_i2c_id, -}; -module_i2c_driver(max98090_i2c_driver); - -MODULE_DESCRIPTION("ALSA SoC MAX98090 driver"); -MODULE_AUTHOR("Peter Hsiang, Kuninori Morimoto"); -MODULE_LICENSE("GPL"); -- cgit v1.2.3 From 685e42154dcf3f6c0a52c115bd15e3d28ad8621b Mon Sep 17 00:00:00 2001 From: Jerry Wong Date: Wed, 6 Feb 2013 11:06:37 -0800 Subject: ASoC: Replace max98090 Device Driver This patch completes the replacement of the existing max98090 driver, by installing a more complete driver. Signed-off-by: Jerry Wong Tested-by: Matthew Mowdy Reviewed-by: Ralph Birt Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max98090.c | 2398 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/max98090.h | 1549 ++++++++++++++++++++++++++++ 4 files changed, 3953 insertions(+) create mode 100755 sound/soc/codecs/max98090.c create mode 100755 sound/soc/codecs/max98090.h (limited to 'sound/soc') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0e368d46976..3a847828932 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -44,6 +44,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_LM4857 if I2C select SND_SOC_LM49453 if I2C select SND_SOC_MAX98088 if I2C + select SND_SOC_MAX98090 if I2C select SND_SOC_MAX98095 if I2C select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9768 if I2C @@ -267,6 +268,9 @@ config SND_SOC_LM49453 config SND_SOC_MAX98088 tristate +config SND_SOC_MAX98090 + tristate + config SND_SOC_MAX98095 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index aa563122016..f6e8e36cceb 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -34,6 +34,7 @@ snd-soc-lm4857-objs := lm4857.o snd-soc-lm49453-objs := lm49453.o snd-soc-max9768-objs := max9768.o snd-soc-max98088-objs := max98088.o +snd-soc-max98090-objs := max98090.o snd-soc-max98095-objs := max98095.o snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o @@ -157,6 +158,7 @@ obj-$(CONFIG_SND_SOC_LM4857) += snd-soc-lm4857.o obj-$(CONFIG_SND_SOC_LM49453) += snd-soc-lm49453.o obj-$(CONFIG_SND_SOC_MAX9768) += snd-soc-max9768.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o +obj-$(CONFIG_SND_SOC_MAX98090) += snd-soc-max98090.o obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c new file mode 100755 index 00000000000..fc176044994 --- /dev/null +++ b/sound/soc/codecs/max98090.c @@ -0,0 +1,2398 @@ +/* + * max98090.c -- MAX98090 ALSA SoC Audio driver + * + * Copyright 2011-2012 Maxim Integrated Products + * + * 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 "max98090.h" + +#include + +#define DEBUG +#define EXTMIC_METHOD +#define EXTMIC_METHOD_TEST + +/* Allows for sparsely populated register maps */ +static struct reg_default max98090_reg[] = { + { 0x00, 0x00 }, /* 00 Software Reset */ + { 0x03, 0x04 }, /* 03 Interrupt Masks */ + { 0x04, 0x00 }, /* 04 System Clock Quick */ + { 0x05, 0x00 }, /* 05 Sample Rate Quick */ + { 0x06, 0x00 }, /* 06 DAI Interface Quick */ + { 0x07, 0x00 }, /* 07 DAC Path Quick */ + { 0x08, 0x00 }, /* 08 Mic/Direct to ADC Quick */ + { 0x09, 0x00 }, /* 09 Line to ADC Quick */ + { 0x0A, 0x00 }, /* 0A Analog Mic Loop Quick */ + { 0x0B, 0x00 }, /* 0B Analog Line Loop Quick */ + { 0x0C, 0x00 }, /* 0C Reserved */ + { 0x0D, 0x00 }, /* 0D Input Config */ + { 0x0E, 0x1B }, /* 0E Line Input Level */ + { 0x0F, 0x00 }, /* 0F Line Config */ + + { 0x10, 0x14 }, /* 10 Mic1 Input Level */ + { 0x11, 0x14 }, /* 11 Mic2 Input Level */ + { 0x12, 0x00 }, /* 12 Mic Bias Voltage */ + { 0x13, 0x00 }, /* 13 Digital Mic Config */ + { 0x14, 0x00 }, /* 14 Digital Mic Mode */ + { 0x15, 0x00 }, /* 15 Left ADC Mixer */ + { 0x16, 0x00 }, /* 16 Right ADC Mixer */ + { 0x17, 0x03 }, /* 17 Left ADC Level */ + { 0x18, 0x03 }, /* 18 Right ADC Level */ + { 0x19, 0x00 }, /* 19 ADC Biquad Level */ + { 0x1A, 0x00 }, /* 1A ADC Sidetone */ + { 0x1B, 0x00 }, /* 1B System Clock */ + { 0x1C, 0x00 }, /* 1C Clock Mode */ + { 0x1D, 0x00 }, /* 1D Any Clock 1 */ + { 0x1E, 0x00 }, /* 1E Any Clock 2 */ + { 0x1F, 0x00 }, /* 1F Any Clock 3 */ + + { 0x20, 0x00 }, /* 20 Any Clock 4 */ + { 0x21, 0x00 }, /* 21 Master Mode */ + { 0x22, 0x00 }, /* 22 Interface Format */ + { 0x23, 0x00 }, /* 23 TDM Format 1*/ + { 0x24, 0x00 }, /* 24 TDM Format 2*/ + { 0x25, 0x00 }, /* 25 I/O Configuration */ + { 0x26, 0x80 }, /* 26 Filter Config */ + { 0x27, 0x00 }, /* 27 DAI Playback Level */ + { 0x28, 0x00 }, /* 28 EQ Playback Level */ + { 0x29, 0x00 }, /* 29 Left HP Mixer */ + { 0x2A, 0x00 }, /* 2A Right HP Mixer */ + { 0x2B, 0x00 }, /* 2B HP Control */ + { 0x2C, 0x1A }, /* 2C Left HP Volume */ + { 0x2D, 0x1A }, /* 2D Right HP Volume */ + { 0x2E, 0x00 }, /* 2E Left Spk Mixer */ + { 0x2F, 0x00 }, /* 2F Right Spk Mixer */ + + { 0x30, 0x00 }, /* 30 Spk Control */ + { 0x31, 0x2C }, /* 31 Left Spk Volume */ + { 0x32, 0x2C }, /* 32 Right Spk Volume */ + { 0x33, 0x00 }, /* 33 ALC Timing */ + { 0x34, 0x00 }, /* 34 ALC Compressor */ + { 0x35, 0x00 }, /* 35 ALC Expander */ + { 0x36, 0x00 }, /* 36 ALC Gain */ + { 0x37, 0x00 }, /* 37 Rcv/Line OutL Mixer */ + { 0x38, 0x00 }, /* 38 Rcv/Line OutL Control */ + { 0x39, 0x15 }, /* 39 Rcv/Line OutL Volume */ + { 0x3A, 0x00 }, /* 3A Line OutR Mixer */ + { 0x3B, 0x00 }, /* 3B Line OutR Control */ + { 0x3C, 0x15 }, /* 3C Line OutR Volume */ + { 0x3D, 0x00 }, /* 3D Jack Detect */ + { 0x3E, 0x00 }, /* 3E Input Enable */ + { 0x3F, 0x00 }, /* 3F Output Enable */ + + { 0x40, 0x00 }, /* 40 Level Control */ + { 0x41, 0x00 }, /* 41 DSP Filter Enable */ + { 0x42, 0x00 }, /* 42 Bias Control */ + { 0x43, 0x00 }, /* 43 DAC Control */ + { 0x44, 0x06 }, /* 44 ADC Control */ + { 0x45, 0x00 }, /* 45 Device Shutdown */ + { 0x46, 0x00 }, /* 46 Equalizer Band 1 Coefficient B0 */ + { 0x47, 0x00 }, /* 47 Equalizer Band 1 Coefficient B0 */ + { 0x48, 0x00 }, /* 48 Equalizer Band 1 Coefficient B0 */ + { 0x49, 0x00 }, /* 49 Equalizer Band 1 Coefficient B1 */ + { 0x4A, 0x00 }, /* 4A Equalizer Band 1 Coefficient B1 */ + { 0x4B, 0x00 }, /* 4B Equalizer Band 1 Coefficient B1 */ + { 0x4C, 0x00 }, /* 4C Equalizer Band 1 Coefficient B2 */ + { 0x4D, 0x00 }, /* 4D Equalizer Band 1 Coefficient B2 */ + { 0x4E, 0x00 }, /* 4E Equalizer Band 1 Coefficient B2 */ + { 0x4F, 0x00 }, /* 4F Equalizer Band 1 Coefficient A1 */ + + { 0x50, 0x00 }, /* 50 Equalizer Band 1 Coefficient A1 */ + { 0x51, 0x00 }, /* 51 Equalizer Band 1 Coefficient A1 */ + { 0x52, 0x00 }, /* 52 Equalizer Band 1 Coefficient A2 */ + { 0x53, 0x00 }, /* 53 Equalizer Band 1 Coefficient A2 */ + { 0x54, 0x00 }, /* 54 Equalizer Band 1 Coefficient A2 */ + { 0x55, 0x00 }, /* 55 Equalizer Band 2 Coefficient B0 */ + { 0x56, 0x00 }, /* 56 Equalizer Band 2 Coefficient B0 */ + { 0x57, 0x00 }, /* 57 Equalizer Band 2 Coefficient B0 */ + { 0x58, 0x00 }, /* 58 Equalizer Band 2 Coefficient B1 */ + { 0x59, 0x00 }, /* 59 Equalizer Band 2 Coefficient B1 */ + { 0x5A, 0x00 }, /* 5A Equalizer Band 2 Coefficient B1 */ + { 0x5B, 0x00 }, /* 5B Equalizer Band 2 Coefficient B2 */ + { 0x5C, 0x00 }, /* 5C Equalizer Band 2 Coefficient B2 */ + { 0x5D, 0x00 }, /* 5D Equalizer Band 2 Coefficient B2 */ + { 0x5E, 0x00 }, /* 5E Equalizer Band 2 Coefficient A1 */ + { 0x5F, 0x00 }, /* 5F Equalizer Band 2 Coefficient A1 */ + + { 0x60, 0x00 }, /* 60 Equalizer Band 2 Coefficient A1 */ + { 0x61, 0x00 }, /* 61 Equalizer Band 2 Coefficient A2 */ + { 0x62, 0x00 }, /* 62 Equalizer Band 2 Coefficient A2 */ + { 0x63, 0x00 }, /* 63 Equalizer Band 2 Coefficient A2 */ + { 0x64, 0x00 }, /* 64 Equalizer Band 3 Coefficient B0 */ + { 0x65, 0x00 }, /* 65 Equalizer Band 3 Coefficient B0 */ + { 0x66, 0x00 }, /* 66 Equalizer Band 3 Coefficient B0 */ + { 0x67, 0x00 }, /* 67 Equalizer Band 3 Coefficient B1 */ + { 0x68, 0x00 }, /* 68 Equalizer Band 3 Coefficient B1 */ + { 0x69, 0x00 }, /* 69 Equalizer Band 3 Coefficient B1 */ + { 0x6A, 0x00 }, /* 6A Equalizer Band 3 Coefficient B2 */ + { 0x6B, 0x00 }, /* 6B Equalizer Band 3 Coefficient B2 */ + { 0x6C, 0x00 }, /* 6C Equalizer Band 3 Coefficient B2 */ + { 0x6D, 0x00 }, /* 6D Equalizer Band 3 Coefficient A1 */ + { 0x6E, 0x00 }, /* 6E Equalizer Band 3 Coefficient A1 */ + { 0x6F, 0x00 }, /* 6F Equalizer Band 3 Coefficient A1 */ + + { 0x70, 0x00 }, /* 70 Equalizer Band 3 Coefficient A2 */ + { 0x71, 0x00 }, /* 71 Equalizer Band 3 Coefficient A2 */ + { 0x72, 0x00 }, /* 72 Equalizer Band 3 Coefficient A2 */ + { 0x73, 0x00 }, /* 73 Equalizer Band 4 Coefficient B0 */ + { 0x74, 0x00 }, /* 74 Equalizer Band 4 Coefficient B0 */ + { 0x75, 0x00 }, /* 75 Equalizer Band 4 Coefficient B0 */ + { 0x76, 0x00 }, /* 76 Equalizer Band 4 Coefficient B1 */ + { 0x77, 0x00 }, /* 77 Equalizer Band 4 Coefficient B1 */ + { 0x78, 0x00 }, /* 78 Equalizer Band 4 Coefficient B1 */ + { 0x79, 0x00 }, /* 79 Equalizer Band 4 Coefficient B2 */ + { 0x7A, 0x00 }, /* 7A Equalizer Band 4 Coefficient B2 */ + { 0x7B, 0x00 }, /* 7B Equalizer Band 4 Coefficient B2 */ + { 0x7C, 0x00 }, /* 7C Equalizer Band 4 Coefficient A1 */ + { 0x7D, 0x00 }, /* 7D Equalizer Band 4 Coefficient A1 */ + { 0x7E, 0x00 }, /* 7E Equalizer Band 4 Coefficient A1 */ + { 0x7F, 0x00 }, /* 7F Equalizer Band 4 Coefficient A2 */ + + { 0x80, 0x00 }, /* 80 Equalizer Band 4 Coefficient A2 */ + { 0x81, 0x00 }, /* 81 Equalizer Band 4 Coefficient A2 */ + { 0x82, 0x00 }, /* 82 Equalizer Band 5 Coefficient B0 */ + { 0x83, 0x00 }, /* 83 Equalizer Band 5 Coefficient B0 */ + { 0x84, 0x00 }, /* 84 Equalizer Band 5 Coefficient B0 */ + { 0x85, 0x00 }, /* 85 Equalizer Band 5 Coefficient B1 */ + { 0x86, 0x00 }, /* 86 Equalizer Band 5 Coefficient B1 */ + { 0x87, 0x00 }, /* 87 Equalizer Band 5 Coefficient B1 */ + { 0x88, 0x00 }, /* 88 Equalizer Band 5 Coefficient B2 */ + { 0x89, 0x00 }, /* 89 Equalizer Band 5 Coefficient B2 */ + { 0x8A, 0x00 }, /* 8A Equalizer Band 5 Coefficient B2 */ + { 0x8B, 0x00 }, /* 8B Equalizer Band 5 Coefficient A1 */ + { 0x8C, 0x00 }, /* 8C Equalizer Band 5 Coefficient A1 */ + { 0x8D, 0x00 }, /* 8D Equalizer Band 5 Coefficient A1 */ + { 0x8E, 0x00 }, /* 8E Equalizer Band 5 Coefficient A2 */ + { 0x8F, 0x00 }, /* 8F Equalizer Band 5 Coefficient A2 */ + + { 0x90, 0x00 }, /* 90 Equalizer Band 5 Coefficient A2 */ + { 0x91, 0x00 }, /* 91 Equalizer Band 6 Coefficient B0 */ + { 0x92, 0x00 }, /* 92 Equalizer Band 6 Coefficient B0 */ + { 0x93, 0x00 }, /* 93 Equalizer Band 6 Coefficient B0 */ + { 0x94, 0x00 }, /* 94 Equalizer Band 6 Coefficient B1 */ + { 0x95, 0x00 }, /* 95 Equalizer Band 6 Coefficient B1 */ + { 0x96, 0x00 }, /* 96 Equalizer Band 6 Coefficient B1 */ + { 0x97, 0x00 }, /* 97 Equalizer Band 6 Coefficient B2 */ + { 0x98, 0x00 }, /* 98 Equalizer Band 6 Coefficient B2 */ + { 0x99, 0x00 }, /* 99 Equalizer Band 6 Coefficient B2 */ + { 0x9A, 0x00 }, /* 9A Equalizer Band 6 Coefficient A1 */ + { 0x9B, 0x00 }, /* 9B Equalizer Band 6 Coefficient A1 */ + { 0x9C, 0x00 }, /* 9C Equalizer Band 6 Coefficient A1 */ + { 0x9D, 0x00 }, /* 9D Equalizer Band 6 Coefficient A2 */ + { 0x9E, 0x00 }, /* 9E Equalizer Band 6 Coefficient A2 */ + { 0x9F, 0x00 }, /* 9F Equalizer Band 6 Coefficient A2 */ + + { 0xA0, 0x00 }, /* A0 Equalizer Band 7 Coefficient B0 */ + { 0xA1, 0x00 }, /* A1 Equalizer Band 7 Coefficient B0 */ + { 0xA2, 0x00 }, /* A2 Equalizer Band 7 Coefficient B0 */ + { 0xA3, 0x00 }, /* A3 Equalizer Band 7 Coefficient B1 */ + { 0xA4, 0x00 }, /* A4 Equalizer Band 7 Coefficient B1 */ + { 0xA5, 0x00 }, /* A5 Equalizer Band 7 Coefficient B1 */ + { 0xA6, 0x00 }, /* A6 Equalizer Band 7 Coefficient B2 */ + { 0xA7, 0x00 }, /* A7 Equalizer Band 7 Coefficient B2 */ + { 0xA8, 0x00 }, /* A8 Equalizer Band 7 Coefficient B2 */ + { 0xA9, 0x00 }, /* A9 Equalizer Band 7 Coefficient A1 */ + { 0xAA, 0x00 }, /* AA Equalizer Band 7 Coefficient A1 */ + { 0xAB, 0x00 }, /* AB Equalizer Band 7 Coefficient A1 */ + { 0xAC, 0x00 }, /* AC Equalizer Band 7 Coefficient A2 */ + { 0xAD, 0x00 }, /* AD Equalizer Band 7 Coefficient A2 */ + { 0xAE, 0x00 }, /* AE Equalizer Band 7 Coefficient A2 */ + { 0xAF, 0x00 }, /* AF ADC Biquad Coefficient B0 */ + + { 0xB0, 0x00 }, /* B0 ADC Biquad Coefficient B0 */ + { 0xB1, 0x00 }, /* B1 ADC Biquad Coefficient B0 */ + { 0xB2, 0x00 }, /* B2 ADC Biquad Coefficient B1 */ + { 0xB3, 0x00 }, /* B3 ADC Biquad Coefficient B1 */ + { 0xB4, 0x00 }, /* B4 ADC Biquad Coefficient B1 */ + { 0xB5, 0x00 }, /* B5 ADC Biquad Coefficient B2 */ + { 0xB6, 0x00 }, /* B6 ADC Biquad Coefficient B2 */ + { 0xB7, 0x00 }, /* B7 ADC Biquad Coefficient B2 */ + { 0xB8, 0x00 }, /* B8 ADC Biquad Coefficient A1 */ + { 0xB9, 0x00 }, /* B9 ADC Biquad Coefficient A1 */ + { 0xBA, 0x00 }, /* BA ADC Biquad Coefficient A1 */ + { 0xBB, 0x00 }, /* BB ADC Biquad Coefficient A2 */ + { 0xBC, 0x00 }, /* BC ADC Biquad Coefficient A2 */ + { 0xBD, 0x00 }, /* BD ADC Biquad Coefficient A2 */ + { 0xBE, 0x00 }, /* BE Digital Mic 3 Volume */ + { 0xBF, 0x00 }, /* BF Digital Mic 4 Volume */ + + { 0xC0, 0x00 }, /* C0 Digital Mic 34 Biquad Pre Atten */ + { 0xC1, 0x00 }, /* C1 Record TDM Slot */ + { 0xC2, 0x00 }, /* C2 Sample Rate */ + { 0xC3, 0x00 }, /* C3 Digital Mic 34 Biquad Coefficient C3 */ + { 0xC4, 0x00 }, /* C4 Digital Mic 34 Biquad Coefficient C4 */ + { 0xC5, 0x00 }, /* C5 Digital Mic 34 Biquad Coefficient C5 */ + { 0xC6, 0x00 }, /* C6 Digital Mic 34 Biquad Coefficient C6 */ + { 0xC7, 0x00 }, /* C7 Digital Mic 34 Biquad Coefficient C7 */ + { 0xC8, 0x00 }, /* C8 Digital Mic 34 Biquad Coefficient C8 */ + { 0xC9, 0x00 }, /* C9 Digital Mic 34 Biquad Coefficient C9 */ + { 0xCA, 0x00 }, /* CA Digital Mic 34 Biquad Coefficient CA */ + { 0xCB, 0x00 }, /* CB Digital Mic 34 Biquad Coefficient CB */ + { 0xCC, 0x00 }, /* CC Digital Mic 34 Biquad Coefficient CC */ + { 0xCD, 0x00 }, /* CD Digital Mic 34 Biquad Coefficient CD */ + { 0xCE, 0x00 }, /* CE Digital Mic 34 Biquad Coefficient CE */ + { 0xCF, 0x00 }, /* CF Digital Mic 34 Biquad Coefficient CF */ + + { 0xD0, 0x00 }, /* D0 Digital Mic 34 Biquad Coefficient D0 */ + { 0xD1, 0x00 }, /* D1 Digital Mic 34 Biquad Coefficient D1 */ +}; + +static bool max98090_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case M98090_REG_DEVICE_STATUS: + case M98090_REG_JACK_STATUS: + case M98090_REG_REVISION_ID: + return true; + default: + return false; + } +} + +static bool max98090_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case M98090_REG_DEVICE_STATUS: + case M98090_REG_JACK_STATUS: + case M98090_REG_INTERRUPT_S: + case M98090_REG_RESERVED: + case M98090_REG_LINE_INPUT_CONFIG: + case M98090_REG_LINE_INPUT_LEVEL: + case M98090_REG_INPUT_MODE: + case M98090_REG_MIC1_INPUT_LEVEL: + case M98090_REG_MIC2_INPUT_LEVEL: + case M98090_REG_MIC_BIAS_VOLTAGE: + case M98090_REG_DIGITAL_MIC_ENABLE: + case M98090_REG_DIGITAL_MIC_CONFIG: + case M98090_REG_LEFT_ADC_MIXER: + case M98090_REG_RIGHT_ADC_MIXER: + case M98090_REG_LEFT_ADC_LEVEL: + case M98090_REG_RIGHT_ADC_LEVEL: + case M98090_REG_ADC_BIQUAD_LEVEL: + case M98090_REG_ADC_SIDETONE: + case M98090_REG_SYSTEM_CLOCK: + case M98090_REG_CLOCK_MODE: + case M98090_REG_CLOCK_RATIO_NI_MSB: + case M98090_REG_CLOCK_RATIO_NI_LSB: + case M98090_REG_CLOCK_RATIO_MI_MSB: + case M98090_REG_CLOCK_RATIO_MI_LSB: + case M98090_REG_MASTER_MODE: + case M98090_REG_INTERFACE_FORMAT: + case M98090_REG_TDM_CONTROL: + case M98090_REG_TDM_FORMAT: + case M98090_REG_IO_CONFIGURATION: + case M98090_REG_FILTER_CONFIG: + case M98090_REG_DAI_PLAYBACK_LEVEL: + case M98090_REG_DAI_PLAYBACK_LEVEL_EQ: + case M98090_REG_LEFT_HP_MIXER: + case M98090_REG_RIGHT_HP_MIXER: + case M98090_REG_HP_CONTROL: + case M98090_REG_LEFT_HP_VOLUME: + case M98090_REG_RIGHT_HP_VOLUME: + case M98090_REG_LEFT_SPK_MIXER: + case M98090_REG_RIGHT_SPK_MIXER: + case M98090_REG_SPK_CONTROL: + case M98090_REG_LEFT_SPK_VOLUME: + case M98090_REG_RIGHT_SPK_VOLUME: + case M98090_REG_DRC_TIMING: + case M98090_REG_DRC_COMPRESSOR: + case M98090_REG_DRC_EXPANDER: + case M98090_REG_DRC_GAIN: + case M98090_REG_RCV_LOUTL_MIXER: + case M98090_REG_RCV_LOUTL_CONTROL: + case M98090_REG_RCV_LOUTL_VOLUME: + case M98090_REG_LOUTR_MIXER: + case M98090_REG_LOUTR_CONTROL: + case M98090_REG_LOUTR_VOLUME: + case M98090_REG_JACK_DETECT: + case M98090_REG_INPUT_ENABLE: + case M98090_REG_OUTPUT_ENABLE: + case M98090_REG_LEVEL_CONTROL: + case M98090_REG_DSP_FILTER_ENABLE: + case M98090_REG_BIAS_CONTROL: + case M98090_REG_DAC_CONTROL: + case M98090_REG_ADC_CONTROL: + case M98090_REG_DEVICE_SHUTDOWN: + case M98090_REG_EQUALIZER_BASE ... M98090_REG_EQUALIZER_BASE + 0x68: + case M98090_REG_RECORD_BIQUAD_BASE ... M98090_REG_RECORD_BIQUAD_BASE + 0x0E: + case M98090_REG_DMIC3_VOLUME: + case M98090_REG_DMIC4_VOLUME: + case M98090_REG_DMIC34_BQ_PREATTEN: + case M98090_REG_RECORD_TDM_SLOT: + case M98090_REG_SAMPLE_RATE: + case M98090_REG_DMIC34_BIQUAD_BASE ... M98090_REG_DMIC34_BIQUAD_BASE + 0x0E: + return true; + default: + return false; + } +} + +static int max98090_reset(struct max98090_priv *max98090) +{ + int ret; + + /* Reset the codec by writing to this write-only reset register */ + ret = regmap_write(max98090->regmap, M98090_REG_SOFTWARE_RESET, + M98090_SWRESET_MASK); + if (ret < 0) { + dev_err(max98090->codec->dev, + "Failed to reset codec: %d\n", ret); + return ret; + } + + msleep(20); + return ret; +} + +static const unsigned int max98090_micboost_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 1, TLV_DB_SCALE_ITEM(0, 2000, 0), + 2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0), +}; + +static const DECLARE_TLV_DB_SCALE(max98090_mic_tlv, 0, 100, 0); + +static const DECLARE_TLV_DB_SCALE(max98090_line_single_ended_tlv, + -600, 600, 0); + +static const unsigned int max98090_line_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 3, TLV_DB_SCALE_ITEM(-600, 300, 0), + 4, 5, TLV_DB_SCALE_ITEM(1400, 600, 0), +}; + +static const DECLARE_TLV_DB_SCALE(max98090_avg_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(max98090_av_tlv, -1200, 100, 0); + +static const DECLARE_TLV_DB_SCALE(max98090_dvg_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(max98090_dv_tlv, -1500, 100, 0); + +static const DECLARE_TLV_DB_SCALE(max98090_sidetone_tlv, -6050, 200, 0); + +static const DECLARE_TLV_DB_SCALE(max98090_alc_tlv, -1500, 100, 0); +static const DECLARE_TLV_DB_SCALE(max98090_alcmakeup_tlv, 0, 100, 0); +static const DECLARE_TLV_DB_SCALE(max98090_alccomp_tlv, -3100, 100, 0); +static const DECLARE_TLV_DB_SCALE(max98090_drcexp_tlv, -6600, 100, 0); + +static const unsigned int max98090_mixout_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 1, TLV_DB_SCALE_ITEM(-1200, 250, 0), + 2, 3, TLV_DB_SCALE_ITEM(-600, 600, 0), +}; + +static const unsigned int max98090_hp_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 6, TLV_DB_SCALE_ITEM(-6700, 400, 0), + 7, 14, TLV_DB_SCALE_ITEM(-4000, 300, 0), + 15, 21, TLV_DB_SCALE_ITEM(-1700, 200, 0), + 22, 27, TLV_DB_SCALE_ITEM(-400, 100, 0), + 28, 31, TLV_DB_SCALE_ITEM(150, 50, 0), +}; + +static const unsigned int max98090_spk_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 4, TLV_DB_SCALE_ITEM(-4800, 400, 0), + 5, 10, TLV_DB_SCALE_ITEM(-2900, 300, 0), + 11, 14, TLV_DB_SCALE_ITEM(-1200, 200, 0), + 15, 29, TLV_DB_SCALE_ITEM(-500, 100, 0), + 30, 39, TLV_DB_SCALE_ITEM(950, 50, 0), +}; + +static const unsigned int max98090_rcv_lout_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 6, TLV_DB_SCALE_ITEM(-6200, 400, 0), + 7, 14, TLV_DB_SCALE_ITEM(-3500, 300, 0), + 15, 21, TLV_DB_SCALE_ITEM(-1200, 200, 0), + 22, 27, TLV_DB_SCALE_ITEM(100, 100, 0), + 28, 31, TLV_DB_SCALE_ITEM(650, 50, 0), +}; + +static int max98090_get_enab_tlv(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int mask = (1 << fls(mc->max)) - 1; + unsigned int val = snd_soc_read(codec, mc->reg); + unsigned int *select; + + switch (mc->reg) { + case M98090_REG_MIC1_INPUT_LEVEL: + select = &(max98090->pa1en); + break; + case M98090_REG_MIC2_INPUT_LEVEL: + select = &(max98090->pa2en); + break; + case M98090_REG_ADC_SIDETONE: + select = &(max98090->sidetone); + break; + default: + return -EINVAL; + } + + val = (val >> mc->shift) & mask; + + if (val >= 1) { + /* If on, return the volume */ + val = val - 1; + *select = val; + } else { + /* If off, return last stored value */ + val = *select; + } + + ucontrol->value.integer.value[0] = val; + return 0; +} + +static int max98090_put_enab_tlv(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int mask = (1 << fls(mc->max)) - 1; + unsigned int sel = ucontrol->value.integer.value[0]; + unsigned int val = snd_soc_read(codec, mc->reg); + unsigned int *select; + + switch (mc->reg) { + case M98090_REG_MIC1_INPUT_LEVEL: + select = &(max98090->pa1en); + break; + case M98090_REG_MIC2_INPUT_LEVEL: + select = &(max98090->pa2en); + break; + case M98090_REG_ADC_SIDETONE: + select = &(max98090->sidetone); + break; + default: + return -EINVAL; + } + + val = (val >> mc->shift) & mask; + + *select = sel; + + /* Setting a volume is only valid if it is already On */ + if (val >= 1) { + sel = sel + 1; + } else { + /* Write what was already there */ + sel = val; + } + + snd_soc_update_bits(codec, mc->reg, + mask << mc->shift, + sel << mc->shift); + + return 0; +} + +static const char * max98090_perf_pwr_text[] = + { "High Performance", "Low Power" }; +static const char * max98090_pwr_perf_text[] = + { "Low Power", "High Performance" }; + +static const struct soc_enum max98090_vcmbandgap_enum = + SOC_ENUM_SINGLE(M98090_REG_BIAS_CONTROL, M98090_VCM_MODE_SHIFT, + ARRAY_SIZE(max98090_pwr_perf_text), max98090_pwr_perf_text); + +static const char * max98090_osr128_text[] = { "64*fs", "128*fs" }; + +static const struct soc_enum max98090_osr128_enum = + SOC_ENUM_SINGLE(M98090_REG_ADC_CONTROL, M98090_OSR128_SHIFT, + ARRAY_SIZE(max98090_osr128_text), max98090_osr128_text); + +static const char *max98090_mode_text[] = { "Voice", "Music" }; + +static const struct soc_enum max98090_mode_enum = + SOC_ENUM_SINGLE(M98090_REG_FILTER_CONFIG, M98090_MODE_SHIFT, + ARRAY_SIZE(max98090_mode_text), max98090_mode_text); + +static const struct soc_enum max98090_filter_dmic34mode_enum = + SOC_ENUM_SINGLE(M98090_REG_FILTER_CONFIG, + M98090_FLT_DMIC34MODE_SHIFT, + ARRAY_SIZE(max98090_mode_text), max98090_mode_text); + +static const char * max98090_drcatk_text[] = + { "0.5ms", "1ms", "5ms", "10ms", "25ms", "50ms", "100ms", "200ms" }; + +static const struct soc_enum max98090_drcatk_enum = + SOC_ENUM_SINGLE(M98090_REG_DRC_TIMING, M98090_DRCATK_SHIFT, + ARRAY_SIZE(max98090_drcatk_text), max98090_drcatk_text); + +static const char * max98090_drcrls_text[] = + { "8s", "4s", "2s", "1s", "0.5s", "0.25s", "0.125s", "0.0625s" }; + +static const struct soc_enum max98090_drcrls_enum = + SOC_ENUM_SINGLE(M98090_REG_DRC_TIMING, M98090_DRCRLS_SHIFT, + ARRAY_SIZE(max98090_drcrls_text), max98090_drcrls_text); + +static const char * max98090_alccmp_text[] = + { "1:1", "1:1.5", "1:2", "1:4", "1:INF" }; + +static const struct soc_enum max98090_alccmp_enum = + SOC_ENUM_SINGLE(M98090_REG_DRC_COMPRESSOR, M98090_DRCCMP_SHIFT, + ARRAY_SIZE(max98090_alccmp_text), max98090_alccmp_text); + +static const char * max98090_drcexp_text[] = { "1:1", "2:1", "3:1" }; + +static const struct soc_enum max98090_drcexp_enum = + SOC_ENUM_SINGLE(M98090_REG_DRC_EXPANDER, M98090_DRCEXP_SHIFT, + ARRAY_SIZE(max98090_drcexp_text), max98090_drcexp_text); + +static const struct soc_enum max98090_dac_perfmode_enum = + SOC_ENUM_SINGLE(M98090_REG_DAC_CONTROL, M98090_PERFMODE_SHIFT, + ARRAY_SIZE(max98090_perf_pwr_text), max98090_perf_pwr_text); + +static const struct soc_enum max98090_dachp_enum = + SOC_ENUM_SINGLE(M98090_REG_DAC_CONTROL, M98090_DACHP_SHIFT, + ARRAY_SIZE(max98090_pwr_perf_text), max98090_pwr_perf_text); + +static const struct soc_enum max98090_adchp_enum = + SOC_ENUM_SINGLE(M98090_REG_ADC_CONTROL, M98090_ADCHP_SHIFT, + ARRAY_SIZE(max98090_pwr_perf_text), max98090_pwr_perf_text); + +static const struct snd_kcontrol_new max98090_snd_controls[] = { + SOC_ENUM("MIC Bias VCM Bandgap", max98090_vcmbandgap_enum), + + SOC_SINGLE("DMIC MIC Comp Filter Config", M98090_REG_DIGITAL_MIC_CONFIG, + M98090_DMIC_COMP_SHIFT, M98090_DMIC_COMP_NUM - 1, 0), + + SOC_SINGLE_EXT_TLV("MIC1 Boost Volume", + M98090_REG_MIC1_INPUT_LEVEL, M98090_MIC_PA1EN_SHIFT, + M98090_MIC_PA1EN_NUM - 1, 0, max98090_get_enab_tlv, + max98090_put_enab_tlv, max98090_micboost_tlv), + + SOC_SINGLE_EXT_TLV("MIC2 Boost Volume", + M98090_REG_MIC2_INPUT_LEVEL, M98090_MIC_PA2EN_SHIFT, + M98090_MIC_PA2EN_NUM - 1, 0, max98090_get_enab_tlv, + max98090_put_enab_tlv, max98090_micboost_tlv), + + SOC_SINGLE_TLV("MIC1 Volume", M98090_REG_MIC1_INPUT_LEVEL, + M98090_MIC_PGAM1_SHIFT, M98090_MIC_PGAM1_NUM - 1, 1, + max98090_mic_tlv), + + SOC_SINGLE_TLV("MIC2 Volume", M98090_REG_MIC2_INPUT_LEVEL, + M98090_MIC_PGAM2_SHIFT, M98090_MIC_PGAM2_NUM - 1, 1, + max98090_mic_tlv), + + SOC_SINGLE_RANGE_TLV("LINEA Single Ended Volume", + M98090_REG_LINE_INPUT_LEVEL, M98090_MIXG135_SHIFT, 0, + M98090_MIXG135_NUM - 1, 1, max98090_line_single_ended_tlv), + + SOC_SINGLE_RANGE_TLV("LINEB Single Ended Volume", + M98090_REG_LINE_INPUT_LEVEL, M98090_MIXG246_SHIFT, 0, + M98090_MIXG246_NUM - 1, 1, max98090_line_single_ended_tlv), + + SOC_SINGLE_RANGE_TLV("LINEA Volume", M98090_REG_LINE_INPUT_LEVEL, + M98090_LINAPGA_SHIFT, 0, M98090_LINAPGA_NUM - 1, 1, + max98090_line_tlv), + + SOC_SINGLE_RANGE_TLV("LINEB Volume", M98090_REG_LINE_INPUT_LEVEL, + M98090_LINBPGA_SHIFT, 0, M98090_LINBPGA_NUM - 1, 1, + max98090_line_tlv), + + SOC_SINGLE("LINEA Ext Resistor Gain Mode", M98090_REG_INPUT_MODE, + M98090_EXTBUFA_SHIFT, M98090_EXTBUFA_NUM - 1, 0), + SOC_SINGLE("LINEB Ext Resistor Gain Mode", M98090_REG_INPUT_MODE, + M98090_EXTBUFB_SHIFT, M98090_EXTBUFB_NUM - 1, 0), + + SOC_SINGLE_TLV("ADCL Boost Volume", M98090_REG_LEFT_ADC_LEVEL, + M98090_AVLG_SHIFT, M98090_AVLG_NUM - 1, 0, + max98090_avg_tlv), + SOC_SINGLE_TLV("ADCR Boost Volume", M98090_REG_RIGHT_ADC_LEVEL, + M98090_AVRG_SHIFT, M98090_AVLG_NUM - 1, 0, + max98090_avg_tlv), + + SOC_SINGLE_TLV("ADCL Volume", M98090_REG_LEFT_ADC_LEVEL, + M98090_AVL_SHIFT, M98090_AVL_NUM - 1, 1, + max98090_av_tlv), + SOC_SINGLE_TLV("ADCR Volume", M98090_REG_RIGHT_ADC_LEVEL, + M98090_AVR_SHIFT, M98090_AVR_NUM - 1, 1, + max98090_av_tlv), + + SOC_ENUM("ADC Oversampling Rate", max98090_osr128_enum), + SOC_SINGLE("ADC Quantizer Dither", M98090_REG_ADC_CONTROL, + M98090_ADCDITHER_SHIFT, M98090_ADCDITHER_NUM - 1, 0), + SOC_ENUM("ADC High Performance Mode", max98090_adchp_enum), + + SOC_SINGLE("DAC Mono Mode", M98090_REG_IO_CONFIGURATION, + M98090_DMONO_SHIFT, M98090_DMONO_NUM - 1, 0), + SOC_SINGLE("SDIN Mode", M98090_REG_IO_CONFIGURATION, + M98090_SDIEN_SHIFT, M98090_SDIEN_NUM - 1, 0), + SOC_SINGLE("SDOUT Mode", M98090_REG_IO_CONFIGURATION, + M98090_SDOEN_SHIFT, M98090_SDOEN_NUM - 1, 0), + SOC_SINGLE("SDOUT Hi-Z Mode", M98090_REG_IO_CONFIGURATION, + M98090_HIZOFF_SHIFT, M98090_HIZOFF_NUM - 1, 1), + SOC_ENUM("Filter Mode", max98090_mode_enum), + SOC_SINGLE("Record Path DC Blocking", M98090_REG_FILTER_CONFIG, + M98090_AHPF_SHIFT, M98090_AHPF_NUM - 1, 0), + SOC_SINGLE("Playback Path DC Blocking", M98090_REG_FILTER_CONFIG, + M98090_DHPF_SHIFT, M98090_DHPF_NUM - 1, 0), + SOC_SINGLE_TLV("Digital BQ Volume", M98090_REG_ADC_BIQUAD_LEVEL, + M98090_AVBQ_SHIFT, M98090_AVBQ_NUM - 1, 1, max98090_dv_tlv), + SOC_SINGLE_EXT_TLV("Digital Sidetone Volume", + M98090_REG_ADC_SIDETONE, M98090_DVST_SHIFT, + M98090_DVST_NUM - 1, 1, max98090_get_enab_tlv, + max98090_put_enab_tlv, max98090_micboost_tlv), + SOC_SINGLE_TLV("Digital Coarse Volume", M98090_REG_DAI_PLAYBACK_LEVEL, + M98090_DVG_SHIFT, M98090_DVG_NUM - 1, 0, + max98090_dvg_tlv), + SOC_SINGLE_TLV("Digital Volume", M98090_REG_DAI_PLAYBACK_LEVEL, + M98090_DV_SHIFT, M98090_DV_NUM - 1, 1, + max98090_dv_tlv), + SND_SOC_BYTES("EQ Coefficients", M98090_REG_EQUALIZER_BASE, 105), + SOC_SINGLE("Digital EQ 3 Band Switch", M98090_REG_DSP_FILTER_ENABLE, + M98090_EQ3BANDEN_SHIFT, M98090_EQ3BANDEN_NUM - 1, 0), + SOC_SINGLE("Digital EQ 5 Band Switch", M98090_REG_DSP_FILTER_ENABLE, + M98090_EQ5BANDEN_SHIFT, M98090_EQ5BANDEN_NUM - 1, 0), + SOC_SINGLE("Digital EQ 7 Band Switch", M98090_REG_DSP_FILTER_ENABLE, + M98090_EQ7BANDEN_SHIFT, M98090_EQ7BANDEN_NUM - 1, 0), + SOC_SINGLE("Digital EQ Clipping Detection", M98090_REG_DAI_PLAYBACK_LEVEL_EQ, + M98090_EQCLPN_SHIFT, M98090_EQCLPN_NUM - 1, + 1), + SOC_SINGLE_TLV("Digital EQ Volume", M98090_REG_DAI_PLAYBACK_LEVEL_EQ, + M98090_DVEQ_SHIFT, M98090_DVEQ_NUM - 1, 1, + max98090_dv_tlv), + + SOC_SINGLE("ALC Enable", M98090_REG_DRC_TIMING, + M98090_DRCEN_SHIFT, M98090_DRCEN_NUM - 1, 0), + SOC_ENUM("ALC Attack Time", max98090_drcatk_enum), + SOC_ENUM("ALC Release Time", max98090_drcrls_enum), + SOC_SINGLE_TLV("ALC Make Up Volume", M98090_REG_DRC_GAIN, + M98090_DRCG_SHIFT, M98090_DRCG_NUM - 1, 0, + max98090_alcmakeup_tlv), + SOC_ENUM("ALC Compression Ratio", max98090_alccmp_enum), + SOC_ENUM("ALC Expansion Ratio", max98090_drcexp_enum), + SOC_SINGLE_TLV("ALC Compression Threshold Volume", + M98090_REG_DRC_COMPRESSOR, M98090_DRCTHC_SHIFT, + M98090_DRCTHC_NUM - 1, 1, max98090_alccomp_tlv), + SOC_SINGLE_TLV("ALC Expansion Threshold Volume", + M98090_REG_DRC_EXPANDER, M98090_DRCTHE_SHIFT, + M98090_DRCTHE_NUM - 1, 1, max98090_drcexp_tlv), + + SOC_ENUM("DAC HP Playback Performance Mode", + max98090_dac_perfmode_enum), + SOC_ENUM("DAC High Performance Mode", max98090_dachp_enum), + + SOC_SINGLE_TLV("Headphone Left Mixer Volume", + M98090_REG_HP_CONTROL, M98090_MIXHPLG_SHIFT, + M98090_MIXHPLG_NUM - 1, 1, max98090_mixout_tlv), + SOC_SINGLE_TLV("Headphone Right Mixer Volume", + M98090_REG_HP_CONTROL, M98090_MIXHPRG_SHIFT, + M98090_MIXHPRG_NUM - 1, 1, max98090_mixout_tlv), + + SOC_SINGLE_TLV("Speaker Left Mixer Volume", + M98090_REG_SPK_CONTROL, M98090_MIXSPLG_SHIFT, + M98090_MIXSPLG_NUM - 1, 1, max98090_mixout_tlv), + SOC_SINGLE_TLV("Speaker Right Mixer Volume", + M98090_REG_SPK_CONTROL, M98090_MIXSPRG_SHIFT, + M98090_MIXSPRG_NUM - 1, 1, max98090_mixout_tlv), + + SOC_SINGLE_TLV("Receiver Left Mixer Volume", + M98090_REG_RCV_LOUTL_CONTROL, M98090_MIXRCVLG_SHIFT, + M98090_MIXRCVLG_NUM - 1, 1, max98090_mixout_tlv), + SOC_SINGLE_TLV("Receiver Right Mixer Volume", + M98090_REG_LOUTR_CONTROL, M98090_MIXRCVRG_SHIFT, + M98090_MIXRCVRG_NUM - 1, 1, max98090_mixout_tlv), + + SOC_DOUBLE_R_TLV("Headphone Volume", M98090_REG_LEFT_HP_VOLUME, + M98090_REG_RIGHT_HP_VOLUME, M98090_HPVOLL_SHIFT, + M98090_HPVOLL_NUM - 1, 0, max98090_hp_tlv), + + SOC_DOUBLE_R_RANGE_TLV("Speaker Volume", + M98090_REG_LEFT_SPK_VOLUME, M98090_REG_RIGHT_SPK_VOLUME, + M98090_SPVOLL_SHIFT, 24, M98090_SPVOLL_NUM - 1 + 24, + 0, max98090_spk_tlv), + + SOC_DOUBLE_R_TLV("Receiver Volume", M98090_REG_RCV_LOUTL_VOLUME, + M98090_REG_LOUTR_VOLUME, M98090_RCVLVOL_SHIFT, + M98090_RCVLVOL_NUM - 1, 0, max98090_rcv_lout_tlv), + + SOC_SINGLE("Headphone Left Switch", M98090_REG_LEFT_HP_VOLUME, + M98090_HPLM_SHIFT, 1, 1), + SOC_SINGLE("Headphone Right Switch", M98090_REG_RIGHT_HP_VOLUME, + M98090_HPRM_SHIFT, 1, 1), + + SOC_SINGLE("Speaker Left Switch", M98090_REG_LEFT_SPK_VOLUME, + M98090_SPLM_SHIFT, 1, 1), + SOC_SINGLE("Speaker Right Switch", M98090_REG_RIGHT_SPK_VOLUME, + M98090_SPRM_SHIFT, 1, 1), + + SOC_SINGLE("Receiver Left Switch", M98090_REG_RCV_LOUTL_VOLUME, + M98090_RCVLM_SHIFT, 1, 1), + SOC_SINGLE("Receiver Right Switch", M98090_REG_LOUTR_VOLUME, + M98090_RCVRM_SHIFT, 1, 1), + + SOC_SINGLE("Zero-Crossing Detection", M98090_REG_LEVEL_CONTROL, + M98090_ZDENN_SHIFT, M98090_ZDENN_NUM - 1, 1), + SOC_SINGLE("Enhanced Vol Smoothing", M98090_REG_LEVEL_CONTROL, + M98090_VS2ENN_SHIFT, M98090_VS2ENN_NUM - 1, 1), + SOC_SINGLE("Volume Adjustment Smoothing", M98090_REG_LEVEL_CONTROL, + M98090_VSENN_SHIFT, M98090_VSENN_NUM - 1, 1), + + SND_SOC_BYTES("Biquad Coefficients", M98090_REG_RECORD_BIQUAD_BASE, 15), + SOC_SINGLE("Biquad Switch", M98090_REG_DSP_FILTER_ENABLE, + M98090_ADCBQEN_SHIFT, M98090_ADCBQEN_NUM - 1, 0), +}; + +static const struct snd_kcontrol_new max98091_snd_controls[] = { + + SOC_SINGLE("DMIC34 Zeropad", M98090_REG_SAMPLE_RATE, + M98090_DMIC34_ZEROPAD_SHIFT, + M98090_DMIC34_ZEROPAD_NUM - 1, 0), + + SOC_ENUM("Filter DMIC34 Mode", max98090_filter_dmic34mode_enum), + SOC_SINGLE("DMIC34 DC Blocking", M98090_REG_FILTER_CONFIG, + M98090_FLT_DMIC34HPF_SHIFT, + M98090_FLT_DMIC34HPF_NUM - 1, 0), + + SOC_SINGLE_TLV("DMIC3 Boost Volume", M98090_REG_DMIC3_VOLUME, + M98090_DMIC_AV3G_SHIFT, M98090_DMIC_AV3G_NUM - 1, 0, + max98090_avg_tlv), + SOC_SINGLE_TLV("DMIC4 Boost Volume", M98090_REG_DMIC4_VOLUME, + M98090_DMIC_AV4G_SHIFT, M98090_DMIC_AV4G_NUM - 1, 0, + max98090_avg_tlv), + + SOC_SINGLE_TLV("DMIC3 Volume", M98090_REG_DMIC3_VOLUME, + M98090_DMIC_AV3_SHIFT, M98090_DMIC_AV3_NUM - 1, 1, + max98090_av_tlv), + SOC_SINGLE_TLV("DMIC4 Volume", M98090_REG_DMIC4_VOLUME, + M98090_DMIC_AV4_SHIFT, M98090_DMIC_AV4_NUM - 1, 1, + max98090_av_tlv), + + SND_SOC_BYTES("DMIC34 Biquad Coefficients", + M98090_REG_DMIC34_BIQUAD_BASE, 15), + SOC_SINGLE("DMIC34 Biquad Switch", M98090_REG_DSP_FILTER_ENABLE, + M98090_DMIC34BQEN_SHIFT, M98090_DMIC34BQEN_NUM - 1, 0), + + SOC_SINGLE_TLV("DMIC34 BQ PreAttenuation Volume", + M98090_REG_DMIC34_BQ_PREATTEN, M98090_AV34BQ_SHIFT, + M98090_AV34BQ_NUM - 1, 1, max98090_dv_tlv), +}; + +static int max98090_micinput_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + + unsigned int val = snd_soc_read(codec, w->reg); + + if (w->reg == M98090_REG_MIC1_INPUT_LEVEL) + val = (val & M98090_MIC_PA1EN_MASK) >> M98090_MIC_PA1EN_SHIFT; + else + val = (val & M98090_MIC_PA2EN_MASK) >> M98090_MIC_PA2EN_SHIFT; + + + if (val >= 1) { + if (w->reg == M98090_REG_MIC1_INPUT_LEVEL) { + max98090->pa1en = val - 1; /* Update for volatile */ + } else { + max98090->pa2en = val - 1; /* Update for volatile */ + } + } + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* If turning on, set to most recently selected volume */ + if (w->reg == M98090_REG_MIC1_INPUT_LEVEL) + val = max98090->pa1en + 1; + else + val = max98090->pa2en + 1; + break; + case SND_SOC_DAPM_POST_PMD: + /* If turning off, turn off */ + val = 0; + break; + default: + return -EINVAL; + } + + if (w->reg == M98090_REG_MIC1_INPUT_LEVEL) + snd_soc_update_bits(codec, w->reg, M98090_MIC_PA1EN_MASK, + val << M98090_MIC_PA1EN_SHIFT); + else + snd_soc_update_bits(codec, w->reg, M98090_MIC_PA2EN_MASK, + val << M98090_MIC_PA2EN_SHIFT); + + return 0; +} + +static const char *mic1_mux_text[] = { "IN12", "IN56" }; + +static const struct soc_enum mic1_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_INPUT_MODE, M98090_EXTMIC1_SHIFT, + ARRAY_SIZE(mic1_mux_text), mic1_mux_text); + +static const struct snd_kcontrol_new max98090_mic1_mux = + SOC_DAPM_ENUM("MIC1 Mux", mic1_mux_enum); + +static const char *mic2_mux_text[] = { "IN34", "IN56" }; + +static const struct soc_enum mic2_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_INPUT_MODE, M98090_EXTMIC2_SHIFT, + ARRAY_SIZE(mic2_mux_text), mic2_mux_text); + +static const struct snd_kcontrol_new max98090_mic2_mux = + SOC_DAPM_ENUM("MIC2 Mux", mic2_mux_enum); + +static const char * max98090_micpre_text[] = { "Off", "On" }; + +static const struct soc_enum max98090_pa1en_enum = + SOC_ENUM_SINGLE(M98090_REG_MIC1_INPUT_LEVEL, M98090_MIC_PA1EN_SHIFT, + ARRAY_SIZE(max98090_micpre_text), max98090_micpre_text); + +static const struct soc_enum max98090_pa2en_enum = + SOC_ENUM_SINGLE(M98090_REG_MIC2_INPUT_LEVEL, M98090_MIC_PA2EN_SHIFT, + ARRAY_SIZE(max98090_micpre_text), max98090_micpre_text); + +/* LINEA mixer switch */ +static const struct snd_kcontrol_new max98090_linea_mixer_controls[] = { + SOC_DAPM_SINGLE("IN1 Switch", M98090_REG_LINE_INPUT_CONFIG, + M98090_IN1SEEN_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN3 Switch", M98090_REG_LINE_INPUT_CONFIG, + M98090_IN3SEEN_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN5 Switch", M98090_REG_LINE_INPUT_CONFIG, + M98090_IN5SEEN_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN34 Switch", M98090_REG_LINE_INPUT_CONFIG, + M98090_IN34DIFF_SHIFT, 1, 0), +}; + +/* LINEB mixer switch */ +static const struct snd_kcontrol_new max98090_lineb_mixer_controls[] = { + SOC_DAPM_SINGLE("IN2 Switch", M98090_REG_LINE_INPUT_CONFIG, + M98090_IN2SEEN_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN4 Switch", M98090_REG_LINE_INPUT_CONFIG, + M98090_IN4SEEN_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN6 Switch", M98090_REG_LINE_INPUT_CONFIG, + M98090_IN6SEEN_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN56 Switch", M98090_REG_LINE_INPUT_CONFIG, + M98090_IN56DIFF_SHIFT, 1, 0), +}; + +/* Left ADC mixer switch */ +static const struct snd_kcontrol_new max98090_left_adc_mixer_controls[] = { + SOC_DAPM_SINGLE("IN12 Switch", M98090_REG_LEFT_ADC_MIXER, + M98090_MIXADL_IN12DIFF_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN34 Switch", M98090_REG_LEFT_ADC_MIXER, + M98090_MIXADL_IN34DIFF_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN56 Switch", M98090_REG_LEFT_ADC_MIXER, + M98090_MIXADL_IN65DIFF_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_LEFT_ADC_MIXER, + M98090_MIXADL_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_LEFT_ADC_MIXER, + M98090_MIXADL_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_LEFT_ADC_MIXER, + M98090_MIXADL_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_LEFT_ADC_MIXER, + M98090_MIXADL_MIC2_SHIFT, 1, 0), +}; + +/* Right ADC mixer switch */ +static const struct snd_kcontrol_new max98090_right_adc_mixer_controls[] = { + SOC_DAPM_SINGLE("IN12 Switch", M98090_REG_RIGHT_ADC_MIXER, + M98090_MIXADR_IN12DIFF_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN34 Switch", M98090_REG_RIGHT_ADC_MIXER, + M98090_MIXADR_IN34DIFF_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN56 Switch", M98090_REG_RIGHT_ADC_MIXER, + M98090_MIXADR_IN65DIFF_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_RIGHT_ADC_MIXER, + M98090_MIXADR_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_RIGHT_ADC_MIXER, + M98090_MIXADR_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_RIGHT_ADC_MIXER, + M98090_MIXADR_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_RIGHT_ADC_MIXER, + M98090_MIXADR_MIC2_SHIFT, 1, 0), +}; + +static const char *lten_mux_text[] = { "Normal", "Loopthrough" }; + +static const struct soc_enum ltenl_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_IO_CONFIGURATION, M98090_LTEN_SHIFT, + ARRAY_SIZE(lten_mux_text), lten_mux_text); + +static const struct soc_enum ltenr_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_IO_CONFIGURATION, M98090_LTEN_SHIFT, + ARRAY_SIZE(lten_mux_text), lten_mux_text); + +static const struct snd_kcontrol_new max98090_ltenl_mux = + SOC_DAPM_ENUM("LTENL Mux", ltenl_mux_enum); + +static const struct snd_kcontrol_new max98090_ltenr_mux = + SOC_DAPM_ENUM("LTENR Mux", ltenr_mux_enum); + +static const char *lben_mux_text[] = { "Normal", "Loopback" }; + +static const struct soc_enum lbenl_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_IO_CONFIGURATION, M98090_LBEN_SHIFT, + ARRAY_SIZE(lben_mux_text), lben_mux_text); + +static const struct soc_enum lbenr_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_IO_CONFIGURATION, M98090_LBEN_SHIFT, + ARRAY_SIZE(lben_mux_text), lben_mux_text); + +static const struct snd_kcontrol_new max98090_lbenl_mux = + SOC_DAPM_ENUM("LBENL Mux", lbenl_mux_enum); + +static const struct snd_kcontrol_new max98090_lbenr_mux = + SOC_DAPM_ENUM("LBENR Mux", lbenr_mux_enum); + +static const char *stenl_mux_text[] = { "Normal", "Sidetone Left" }; + +static const char *stenr_mux_text[] = { "Normal", "Sidetone Right" }; + +static const struct soc_enum stenl_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_ADC_SIDETONE, M98090_DSTSL_SHIFT, + ARRAY_SIZE(stenl_mux_text), stenl_mux_text); + +static const struct soc_enum stenr_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_ADC_SIDETONE, M98090_DSTSR_SHIFT, + ARRAY_SIZE(stenr_mux_text), stenr_mux_text); + +static const struct snd_kcontrol_new max98090_stenl_mux = + SOC_DAPM_ENUM("STENL Mux", stenl_mux_enum); + +static const struct snd_kcontrol_new max98090_stenr_mux = + SOC_DAPM_ENUM("STENR Mux", stenr_mux_enum); + +/* Left speaker mixer switch */ +static const struct + snd_kcontrol_new max98090_left_speaker_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC Switch", M98090_REG_LEFT_SPK_MIXER, + M98090_MIXSPL_DACL_SHIFT, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", M98090_REG_LEFT_SPK_MIXER, + M98090_MIXSPL_DACR_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_LEFT_SPK_MIXER, + M98090_MIXSPL_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_LEFT_SPK_MIXER, + M98090_MIXSPL_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_LEFT_SPK_MIXER, + M98090_MIXSPL_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_LEFT_SPK_MIXER, + M98090_MIXSPL_MIC2_SHIFT, 1, 0), +}; + +/* Right speaker mixer switch */ +static const struct + snd_kcontrol_new max98090_right_speaker_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC Switch", M98090_REG_RIGHT_SPK_MIXER, + M98090_MIXSPR_DACL_SHIFT, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", M98090_REG_RIGHT_SPK_MIXER, + M98090_MIXSPR_DACR_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_RIGHT_SPK_MIXER, + M98090_MIXSPR_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_RIGHT_SPK_MIXER, + M98090_MIXSPR_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_RIGHT_SPK_MIXER, + M98090_MIXSPR_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_RIGHT_SPK_MIXER, + M98090_MIXSPR_MIC2_SHIFT, 1, 0), +}; + +/* Left headphone mixer switch */ +static const struct snd_kcontrol_new max98090_left_hp_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC Switch", M98090_REG_LEFT_HP_MIXER, + M98090_MIXHPL_DACL_SHIFT, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", M98090_REG_LEFT_HP_MIXER, + M98090_MIXHPL_DACR_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_LEFT_HP_MIXER, + M98090_MIXHPL_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_LEFT_HP_MIXER, + M98090_MIXHPL_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_LEFT_HP_MIXER, + M98090_MIXHPL_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_LEFT_HP_MIXER, + M98090_MIXHPL_MIC2_SHIFT, 1, 0), +}; + +/* Right headphone mixer switch */ +static const struct snd_kcontrol_new max98090_right_hp_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC Switch", M98090_REG_RIGHT_HP_MIXER, + M98090_MIXHPR_DACL_SHIFT, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", M98090_REG_RIGHT_HP_MIXER, + M98090_MIXHPR_DACR_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_RIGHT_HP_MIXER, + M98090_MIXHPR_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_RIGHT_HP_MIXER, + M98090_MIXHPR_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_RIGHT_HP_MIXER, + M98090_MIXHPR_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_RIGHT_HP_MIXER, + M98090_MIXHPR_MIC2_SHIFT, 1, 0), +}; + +/* Left receiver mixer switch */ +static const struct snd_kcontrol_new max98090_left_rcv_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC Switch", M98090_REG_RCV_LOUTL_MIXER, + M98090_MIXRCVL_DACL_SHIFT, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", M98090_REG_RCV_LOUTL_MIXER, + M98090_MIXRCVL_DACR_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_RCV_LOUTL_MIXER, + M98090_MIXRCVL_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_RCV_LOUTL_MIXER, + M98090_MIXRCVL_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_RCV_LOUTL_MIXER, + M98090_MIXRCVL_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_RCV_LOUTL_MIXER, + M98090_MIXRCVL_MIC2_SHIFT, 1, 0), +}; + +/* Right receiver mixer switch */ +static const struct snd_kcontrol_new max98090_right_rcv_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC Switch", M98090_REG_LOUTR_MIXER, + M98090_MIXRCVR_DACL_SHIFT, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", M98090_REG_LOUTR_MIXER, + M98090_MIXRCVR_DACR_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_LOUTR_MIXER, + M98090_MIXRCVR_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_LOUTR_MIXER, + M98090_MIXRCVR_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_LOUTR_MIXER, + M98090_MIXRCVR_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_LOUTR_MIXER, + M98090_MIXRCVR_MIC2_SHIFT, 1, 0), +}; + +static const char *linmod_mux_text[] = { "Left Only", "Left and Right" }; + +static const struct soc_enum linmod_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_LOUTR_MIXER, M98090_LINMOD_SHIFT, + ARRAY_SIZE(linmod_mux_text), linmod_mux_text); + +static const struct snd_kcontrol_new max98090_linmod_mux = + SOC_DAPM_ENUM("LINMOD Mux", linmod_mux_enum); + +static const char *mixhpsel_mux_text[] = { "DAC Only", "HP Mixer" }; + +/* + * This is a mux as it selects the HP output, but to DAPM it is a Mixer enable + */ +static const struct soc_enum mixhplsel_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_HP_CONTROL, M98090_MIXHPLSEL_SHIFT, + ARRAY_SIZE(mixhpsel_mux_text), mixhpsel_mux_text); + +static const struct snd_kcontrol_new max98090_mixhplsel_mux = + SOC_DAPM_ENUM("MIXHPLSEL Mux", mixhplsel_mux_enum); + +static const struct soc_enum mixhprsel_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_HP_CONTROL, M98090_MIXHPRSEL_SHIFT, + ARRAY_SIZE(mixhpsel_mux_text), mixhpsel_mux_text); + +static const struct snd_kcontrol_new max98090_mixhprsel_mux = + SOC_DAPM_ENUM("MIXHPRSEL Mux", mixhprsel_mux_enum); + +static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = { + + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("DMICL"), + SND_SOC_DAPM_INPUT("DMICR"), + SND_SOC_DAPM_INPUT("IN1"), + SND_SOC_DAPM_INPUT("IN2"), + SND_SOC_DAPM_INPUT("IN3"), + SND_SOC_DAPM_INPUT("IN4"), + SND_SOC_DAPM_INPUT("IN5"), + SND_SOC_DAPM_INPUT("IN6"), + SND_SOC_DAPM_INPUT("IN12"), + SND_SOC_DAPM_INPUT("IN34"), + SND_SOC_DAPM_INPUT("IN56"), + + SND_SOC_DAPM_SUPPLY("MICBIAS", M98090_REG_INPUT_ENABLE, + M98090_MBEN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("SHDN", M98090_REG_DEVICE_SHUTDOWN, + M98090_SHDNN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("SDIEN", M98090_REG_IO_CONFIGURATION, + M98090_SDIEN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("SDOEN", M98090_REG_IO_CONFIGURATION, + M98090_SDOEN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMICL_ENA", M98090_REG_DIGITAL_MIC_ENABLE, + M98090_DIGMICL_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMICR_ENA", M98090_REG_DIGITAL_MIC_ENABLE, + M98090_DIGMICR_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("AHPF", M98090_REG_FILTER_CONFIG, + M98090_AHPF_SHIFT, 0, NULL, 0), + +/* + * Note: Sysclk and misc power supplies are taken care of by SHDN + */ + + SND_SOC_DAPM_MUX("MIC1 Mux", SND_SOC_NOPM, + 0, 0, &max98090_mic1_mux), + + SND_SOC_DAPM_MUX("MIC2 Mux", SND_SOC_NOPM, + 0, 0, &max98090_mic2_mux), + + SND_SOC_DAPM_PGA_E("MIC1 Input", M98090_REG_MIC1_INPUT_LEVEL, + M98090_MIC_PA1EN_SHIFT, 0, NULL, 0, max98090_micinput_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_PGA_E("MIC2 Input", M98090_REG_MIC2_INPUT_LEVEL, + M98090_MIC_PA2EN_SHIFT, 0, NULL, 0, max98090_micinput_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_MIXER("LINEA Mixer", SND_SOC_NOPM, 0, 0, + &max98090_linea_mixer_controls[0], + ARRAY_SIZE(max98090_linea_mixer_controls)), + + SND_SOC_DAPM_MIXER("LINEB Mixer", SND_SOC_NOPM, 0, 0, + &max98090_lineb_mixer_controls[0], + ARRAY_SIZE(max98090_lineb_mixer_controls)), + + SND_SOC_DAPM_PGA("LINEA Input", M98090_REG_INPUT_ENABLE, + M98090_LINEAEN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("LINEB Input", M98090_REG_INPUT_ENABLE, + M98090_LINEBEN_SHIFT, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("Left ADC Mixer", SND_SOC_NOPM, 0, 0, + &max98090_left_adc_mixer_controls[0], + ARRAY_SIZE(max98090_left_adc_mixer_controls)), + + SND_SOC_DAPM_MIXER("Right ADC Mixer", SND_SOC_NOPM, 0, 0, + &max98090_right_adc_mixer_controls[0], + ARRAY_SIZE(max98090_right_adc_mixer_controls)), + + SND_SOC_DAPM_ADC("ADCL", NULL, M98090_REG_INPUT_ENABLE, + M98090_ADLEN_SHIFT, 0), + SND_SOC_DAPM_ADC("ADCR", NULL, M98090_REG_INPUT_ENABLE, + M98090_ADREN_SHIFT, 0), + + SND_SOC_DAPM_AIF_OUT("AIFOUTL", "HiFi Capture", 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIFOUTR", "HiFi Capture", 1, + SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_MUX("LBENL Mux", SND_SOC_NOPM, + 0, 0, &max98090_lbenl_mux), + + SND_SOC_DAPM_MUX("LBENR Mux", SND_SOC_NOPM, + 0, 0, &max98090_lbenr_mux), + + SND_SOC_DAPM_MUX("LTENL Mux", SND_SOC_NOPM, + 0, 0, &max98090_ltenl_mux), + + SND_SOC_DAPM_MUX("LTENR Mux", SND_SOC_NOPM, + 0, 0, &max98090_ltenr_mux), + + SND_SOC_DAPM_MUX("STENL Mux", SND_SOC_NOPM, + 0, 0, &max98090_stenl_mux), + + SND_SOC_DAPM_MUX("STENR Mux", SND_SOC_NOPM, + 0, 0, &max98090_stenr_mux), + + SND_SOC_DAPM_AIF_IN("AIFINL", "HiFi Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIFINR", "HiFi Playback", 1, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_DAC("DACL", NULL, M98090_REG_OUTPUT_ENABLE, + M98090_DALEN_SHIFT, 0), + SND_SOC_DAPM_DAC("DACR", NULL, M98090_REG_OUTPUT_ENABLE, + M98090_DAREN_SHIFT, 0), + + SND_SOC_DAPM_MIXER("Left Headphone Mixer", SND_SOC_NOPM, 0, 0, + &max98090_left_hp_mixer_controls[0], + ARRAY_SIZE(max98090_left_hp_mixer_controls)), + + SND_SOC_DAPM_MIXER("Right Headphone Mixer", SND_SOC_NOPM, 0, 0, + &max98090_right_hp_mixer_controls[0], + ARRAY_SIZE(max98090_right_hp_mixer_controls)), + + SND_SOC_DAPM_MIXER("Left Speaker Mixer", SND_SOC_NOPM, 0, 0, + &max98090_left_speaker_mixer_controls[0], + ARRAY_SIZE(max98090_left_speaker_mixer_controls)), + + SND_SOC_DAPM_MIXER("Right Speaker Mixer", SND_SOC_NOPM, 0, 0, + &max98090_right_speaker_mixer_controls[0], + ARRAY_SIZE(max98090_right_speaker_mixer_controls)), + + SND_SOC_DAPM_MIXER("Left Receiver Mixer", SND_SOC_NOPM, 0, 0, + &max98090_left_rcv_mixer_controls[0], + ARRAY_SIZE(max98090_left_rcv_mixer_controls)), + + SND_SOC_DAPM_MIXER("Right Receiver Mixer", SND_SOC_NOPM, 0, 0, + &max98090_right_rcv_mixer_controls[0], + ARRAY_SIZE(max98090_right_rcv_mixer_controls)), + + SND_SOC_DAPM_MUX("LINMOD Mux", M98090_REG_LOUTR_MIXER, + M98090_LINMOD_SHIFT, 0, &max98090_linmod_mux), + + SND_SOC_DAPM_MUX("MIXHPLSEL Mux", M98090_REG_HP_CONTROL, + M98090_MIXHPLSEL_SHIFT, 0, &max98090_mixhplsel_mux), + + SND_SOC_DAPM_MUX("MIXHPRSEL Mux", M98090_REG_HP_CONTROL, + M98090_MIXHPRSEL_SHIFT, 0, &max98090_mixhprsel_mux), + + SND_SOC_DAPM_PGA("HP Left Out", M98090_REG_OUTPUT_ENABLE, + M98090_HPLEN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("HP Right Out", M98090_REG_OUTPUT_ENABLE, + M98090_HPREN_SHIFT, 0, NULL, 0), + + SND_SOC_DAPM_PGA("SPK Left Out", M98090_REG_OUTPUT_ENABLE, + M98090_SPLEN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("SPK Right Out", M98090_REG_OUTPUT_ENABLE, + M98090_SPREN_SHIFT, 0, NULL, 0), + + SND_SOC_DAPM_PGA("RCV Left Out", M98090_REG_OUTPUT_ENABLE, + M98090_RCVLEN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("RCV Right Out", M98090_REG_OUTPUT_ENABLE, + M98090_RCVREN_SHIFT, 0, NULL, 0), + + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), + SND_SOC_DAPM_OUTPUT("SPKL"), + SND_SOC_DAPM_OUTPUT("SPKR"), + SND_SOC_DAPM_OUTPUT("RCVL"), + SND_SOC_DAPM_OUTPUT("RCVR"), +}; + +static const struct snd_soc_dapm_widget max98091_dapm_widgets[] = { + + SND_SOC_DAPM_INPUT("DMIC3"), + SND_SOC_DAPM_INPUT("DMIC4"), + + SND_SOC_DAPM_SUPPLY("DMIC3_ENA", M98090_REG_DIGITAL_MIC_ENABLE, + M98090_DIGMIC3_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC4_ENA", M98090_REG_DIGITAL_MIC_ENABLE, + M98090_DIGMIC4_SHIFT, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_route max98090_dapm_routes[] = { + + {"MIC1 Input", NULL, "MIC1"}, + {"MIC2 Input", NULL, "MIC2"}, + + {"DMICL", NULL, "DMICL_ENA"}, + {"DMICR", NULL, "DMICR_ENA"}, + {"DMICL", NULL, "AHPF"}, + {"DMICR", NULL, "AHPF"}, + + /* MIC1 input mux */ + {"MIC1 Mux", "IN12", "IN12"}, + {"MIC1 Mux", "IN56", "IN56"}, + + /* MIC2 input mux */ + {"MIC2 Mux", "IN34", "IN34"}, + {"MIC2 Mux", "IN56", "IN56"}, + + {"MIC1 Input", NULL, "MIC1 Mux"}, + {"MIC2 Input", NULL, "MIC2 Mux"}, + + /* Left ADC input mixer */ + {"Left ADC Mixer", "IN12 Switch", "IN12"}, + {"Left ADC Mixer", "IN34 Switch", "IN34"}, + {"Left ADC Mixer", "IN56 Switch", "IN56"}, + {"Left ADC Mixer", "LINEA Switch", "LINEA Input"}, + {"Left ADC Mixer", "LINEB Switch", "LINEB Input"}, + {"Left ADC Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Left ADC Mixer", "MIC2 Switch", "MIC2 Input"}, + + /* Right ADC input mixer */ + {"Right ADC Mixer", "IN12 Switch", "IN12"}, + {"Right ADC Mixer", "IN34 Switch", "IN34"}, + {"Right ADC Mixer", "IN56 Switch", "IN56"}, + {"Right ADC Mixer", "LINEA Switch", "LINEA Input"}, + {"Right ADC Mixer", "LINEB Switch", "LINEB Input"}, + {"Right ADC Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Right ADC Mixer", "MIC2 Switch", "MIC2 Input"}, + + /* Line A input mixer */ + {"LINEA Mixer", "IN1 Switch", "IN1"}, + {"LINEA Mixer", "IN3 Switch", "IN3"}, + {"LINEA Mixer", "IN5 Switch", "IN5"}, + {"LINEA Mixer", "IN34 Switch", "IN34"}, + + /* Line B input mixer */ + {"LINEB Mixer", "IN2 Switch", "IN2"}, + {"LINEB Mixer", "IN4 Switch", "IN4"}, + {"LINEB Mixer", "IN6 Switch", "IN6"}, + {"LINEB Mixer", "IN56 Switch", "IN56"}, + + {"LINEA Input", NULL, "LINEA Mixer"}, + {"LINEB Input", NULL, "LINEB Mixer"}, + + /* Inputs */ + {"ADCL", NULL, "Left ADC Mixer"}, + {"ADCR", NULL, "Right ADC Mixer"}, + {"ADCL", NULL, "SHDN"}, + {"ADCR", NULL, "SHDN"}, + + {"LBENL Mux", "Normal", "ADCL"}, + {"LBENL Mux", "Normal", "DMICL"}, + {"LBENL Mux", "Loopback", "LTENL Mux"}, + {"LBENR Mux", "Normal", "ADCR"}, + {"LBENR Mux", "Normal", "DMICR"}, + {"LBENR Mux", "Loopback", "LTENR Mux"}, + + {"AIFOUTL", NULL, "LBENL Mux"}, + {"AIFOUTR", NULL, "LBENR Mux"}, + {"AIFOUTL", NULL, "SHDN"}, + {"AIFOUTR", NULL, "SHDN"}, + {"AIFOUTL", NULL, "SDOEN"}, + {"AIFOUTR", NULL, "SDOEN"}, + + {"LTENL Mux", "Normal", "AIFINL"}, + {"LTENL Mux", "Loopthrough", "LBENL Mux"}, + {"LTENR Mux", "Normal", "AIFINR"}, + {"LTENR Mux", "Loopthrough", "LBENR Mux"}, + + {"DACL", NULL, "LTENL Mux"}, + {"DACR", NULL, "LTENR Mux"}, + + {"STENL Mux", "Sidetone Left", "ADCL"}, + {"STENL Mux", "Sidetone Left", "DMICL"}, + {"STENR Mux", "Sidetone Right", "ADCR"}, + {"STENR Mux", "Sidetone Right", "DMICR"}, + {"DACL", "NULL", "STENL Mux"}, + {"DACR", "NULL", "STENL Mux"}, + + {"AIFINL", NULL, "SHDN"}, + {"AIFINR", NULL, "SHDN"}, + {"AIFINL", NULL, "SDIEN"}, + {"AIFINR", NULL, "SDIEN"}, + {"DACL", NULL, "SHDN"}, + {"DACR", NULL, "SHDN"}, + + /* Left headphone output mixer */ + {"Left Headphone Mixer", "Left DAC Switch", "DACL"}, + {"Left Headphone Mixer", "Right DAC Switch", "DACR"}, + {"Left Headphone Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Left Headphone Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Left Headphone Mixer", "LINEA Switch", "LINEA Input"}, + {"Left Headphone Mixer", "LINEB Switch", "LINEB Input"}, + + /* Right headphone output mixer */ + {"Right Headphone Mixer", "Left DAC Switch", "DACL"}, + {"Right Headphone Mixer", "Right DAC Switch", "DACR"}, + {"Right Headphone Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Right Headphone Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Right Headphone Mixer", "LINEA Switch", "LINEA Input"}, + {"Right Headphone Mixer", "LINEB Switch", "LINEB Input"}, + + /* Left speaker output mixer */ + {"Left Speaker Mixer", "Left DAC Switch", "DACL"}, + {"Left Speaker Mixer", "Right DAC Switch", "DACR"}, + {"Left Speaker Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Left Speaker Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Left Speaker Mixer", "LINEA Switch", "LINEA Input"}, + {"Left Speaker Mixer", "LINEB Switch", "LINEB Input"}, + + /* Right speaker output mixer */ + {"Right Speaker Mixer", "Left DAC Switch", "DACL"}, + {"Right Speaker Mixer", "Right DAC Switch", "DACR"}, + {"Right Speaker Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Right Speaker Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Right Speaker Mixer", "LINEA Switch", "LINEA Input"}, + {"Right Speaker Mixer", "LINEB Switch", "LINEB Input"}, + + /* Left Receiver output mixer */ + {"Left Receiver Mixer", "Left DAC Switch", "DACL"}, + {"Left Receiver Mixer", "Right DAC Switch", "DACR"}, + {"Left Receiver Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Left Receiver Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Left Receiver Mixer", "LINEA Switch", "LINEA Input"}, + {"Left Receiver Mixer", "LINEB Switch", "LINEB Input"}, + + /* Right Receiver output mixer */ + {"Right Receiver Mixer", "Left DAC Switch", "DACL"}, + {"Right Receiver Mixer", "Right DAC Switch", "DACR"}, + {"Right Receiver Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Right Receiver Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Right Receiver Mixer", "LINEA Switch", "LINEA Input"}, + {"Right Receiver Mixer", "LINEB Switch", "LINEB Input"}, + + {"MIXHPLSEL Mux", "HP Mixer", "Left Headphone Mixer"}, + + /* + * Disable this for lowest power if bypassing + * the DAC with an analog signal + */ + {"HP Left Out", NULL, "DACL"}, + {"HP Left Out", NULL, "MIXHPLSEL Mux"}, + + {"MIXHPRSEL Mux", "HP Mixer", "Right Headphone Mixer"}, + + /* + * Disable this for lowest power if bypassing + * the DAC with an analog signal + */ + {"HP Right Out", NULL, "DACR"}, + {"HP Right Out", NULL, "MIXHPRSEL Mux"}, + + {"SPK Left Out", NULL, "Left Speaker Mixer"}, + {"SPK Right Out", NULL, "Right Speaker Mixer"}, + {"RCV Left Out", NULL, "Left Receiver Mixer"}, + + {"LINMOD Mux", "Left and Right", "Right Receiver Mixer"}, + {"LINMOD Mux", "Left Only", "Left Receiver Mixer"}, + {"RCV Right Out", NULL, "LINMOD Mux"}, + + {"HPL", NULL, "HP Left Out"}, + {"HPR", NULL, "HP Right Out"}, + {"SPKL", NULL, "SPK Left Out"}, + {"SPKR", NULL, "SPK Right Out"}, + {"RCVL", NULL, "RCV Left Out"}, + {"RCVR", NULL, "RCV Right Out"}, + +}; + +static const struct snd_soc_dapm_route max98091_dapm_routes[] = { + + /* DMIC inputs */ + {"DMIC3", NULL, "DMIC3_ENA"}, + {"DMIC4", NULL, "DMIC4_ENA"}, + {"DMIC3", NULL, "AHPF"}, + {"DMIC4", NULL, "AHPF"}, + +}; + +static int max98090_add_widgets(struct snd_soc_codec *codec) +{ + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = &codec->dapm; + + snd_soc_add_codec_controls(codec, max98090_snd_controls, + ARRAY_SIZE(max98090_snd_controls)); + + if (max98090->devtype == MAX98091) { + snd_soc_add_codec_controls(codec, max98091_snd_controls, + ARRAY_SIZE(max98091_snd_controls)); + } + + snd_soc_dapm_new_controls(dapm, max98090_dapm_widgets, + ARRAY_SIZE(max98090_dapm_widgets)); + + snd_soc_dapm_add_routes(dapm, max98090_dapm_routes, + ARRAY_SIZE(max98090_dapm_routes)); + + if (max98090->devtype == MAX98091) { + snd_soc_dapm_new_controls(dapm, max98091_dapm_widgets, + ARRAY_SIZE(max98091_dapm_widgets)); + + snd_soc_dapm_add_routes(dapm, max98091_dapm_routes, + ARRAY_SIZE(max98091_dapm_routes)); + + } + + return 0; +} + +static const int pclk_rates[] = { + 12000000, 12000000, 13000000, 13000000, + 16000000, 16000000, 19200000, 19200000 +}; + +static const int lrclk_rates[] = { + 8000, 16000, 8000, 16000, + 8000, 16000, 8000, 16000 +}; + +static const int user_pclk_rates[] = { + 13000000, 13000000 +}; + +static const int user_lrclk_rates[] = { + 44100, 48000 +}; + +static const unsigned long long ni_value[] = { + 3528, 768 +}; + +static const unsigned long long mi_value[] = { + 8125, 1625 +}; + +static void max98090_configure_bclk(struct snd_soc_codec *codec) +{ + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + unsigned long long ni; + int i; + + if (!max98090->sysclk) { + dev_err(codec->dev, "No SYSCLK configured\n"); + return; + } + + if (!max98090->bclk || !max98090->lrclk) { + dev_err(codec->dev, "No audio clocks configured\n"); + return; + } + + /* Skip configuration when operating as slave */ + if (!(snd_soc_read(codec, M98090_REG_MASTER_MODE) & + M98090_MAS_MASK)) { + return; + } + + /* Check for supported PCLK to LRCLK ratios */ + for (i = 0; i < ARRAY_SIZE(pclk_rates); i++) { + if ((pclk_rates[i] == max98090->sysclk) && + (lrclk_rates[i] == max98090->lrclk)) { + dev_dbg(codec->dev, + "Found supported PCLK to LRCLK rates 0x%x\n", + i + 0x8); + + snd_soc_update_bits(codec, M98090_REG_CLOCK_MODE, + M98090_FREQ_MASK, + (i + 0x8) << M98090_FREQ_SHIFT); + snd_soc_update_bits(codec, M98090_REG_CLOCK_MODE, + M98090_USE_M1_MASK, 0); + return; + } + } + + /* Check for user calculated MI and NI ratios */ + for (i = 0; i < ARRAY_SIZE(user_pclk_rates); i++) { + if ((user_pclk_rates[i] == max98090->sysclk) && + (user_lrclk_rates[i] == max98090->lrclk)) { + dev_dbg(codec->dev, + "Found user supported PCLK to LRCLK rates\n"); + dev_dbg(codec->dev, "i %d ni %lld mi %lld\n", + i, ni_value[i], mi_value[i]); + + snd_soc_update_bits(codec, M98090_REG_CLOCK_MODE, + M98090_FREQ_MASK, 0); + snd_soc_update_bits(codec, M98090_REG_CLOCK_MODE, + M98090_USE_M1_MASK, + 1 << M98090_USE_M1_SHIFT); + + snd_soc_write(codec, M98090_REG_CLOCK_RATIO_NI_MSB, + (ni_value[i] >> 8) & 0x7F); + snd_soc_write(codec, M98090_REG_CLOCK_RATIO_NI_LSB, + ni_value[i] & 0xFF); + snd_soc_write(codec, M98090_REG_CLOCK_RATIO_MI_MSB, + (mi_value[i] >> 8) & 0x7F); + snd_soc_write(codec, M98090_REG_CLOCK_RATIO_MI_LSB, + mi_value[i] & 0xFF); + + return; + } + } + + /* + * Calculate based on MI = 65536 (not as good as either method above) + */ + snd_soc_update_bits(codec, M98090_REG_CLOCK_MODE, + M98090_FREQ_MASK, 0); + snd_soc_update_bits(codec, M98090_REG_CLOCK_MODE, + M98090_USE_M1_MASK, 0); + + /* + * Configure NI when operating as master + * Note: There is a small, but significant audio quality improvement + * by calculating ni and mi. + */ + ni = 65536ULL * (max98090->lrclk < 50000 ? 96ULL : 48ULL) + * (unsigned long long int)max98090->lrclk; + do_div(ni, (unsigned long long int)max98090->sysclk); + dev_info(codec->dev, "No better method found\n"); + dev_info(codec->dev, "Calculating ni %lld with mi 65536\n", ni); + snd_soc_write(codec, M98090_REG_CLOCK_RATIO_NI_MSB, + (ni >> 8) & 0x7F); + snd_soc_write(codec, M98090_REG_CLOCK_RATIO_NI_LSB, ni & 0xFF); +} + +static int max98090_dai_set_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct max98090_cdata *cdata; + u8 regval; + + max98090->dai_fmt = fmt; + cdata = &max98090->dai[0]; + + if (fmt != cdata->fmt) { + cdata->fmt = fmt; + + regval = 0; + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + /* Set to slave mode PLL - MAS mode off */ + snd_soc_write(codec, + M98090_REG_CLOCK_RATIO_NI_MSB, 0x00); + snd_soc_write(codec, + M98090_REG_CLOCK_RATIO_NI_LSB, 0x00); + snd_soc_update_bits(codec, M98090_REG_CLOCK_MODE, + M98090_USE_M1_MASK, 0); + break; + case SND_SOC_DAIFMT_CBM_CFM: + /* Set to master mode */ + if (max98090->tdm_slots == 4) { + /* TDM */ + regval |= M98090_MAS_MASK | + M98090_BSEL_64; + } else if (max98090->tdm_slots == 3) { + /* TDM */ + regval |= M98090_MAS_MASK | + M98090_BSEL_48; + } else { + /* Few TDM slots, or No TDM */ + regval |= M98090_MAS_MASK | + M98090_BSEL_32; + } + break; + case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_CBM_CFS: + default: + dev_err(codec->dev, "DAI clock mode unsupported"); + return -EINVAL; + } + snd_soc_write(codec, M98090_REG_MASTER_MODE, regval); + + regval = 0; + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + regval |= M98090_DLY_MASK; + break; + case SND_SOC_DAIFMT_LEFT_J: + break; + case SND_SOC_DAIFMT_RIGHT_J: + regval |= M98090_RJ_MASK; + break; + case SND_SOC_DAIFMT_DSP_A: + /* Not supported mode */ + default: + dev_err(codec->dev, "DAI format unsupported"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + regval |= M98090_WCI_MASK; + break; + case SND_SOC_DAIFMT_IB_NF: + regval |= M98090_BCI_MASK; + break; + case SND_SOC_DAIFMT_IB_IF: + regval |= M98090_BCI_MASK|M98090_WCI_MASK; + break; + default: + dev_err(codec->dev, "DAI invert mode unsupported"); + return -EINVAL; + } + + /* + * This accommodates an inverted logic in the MAX98090 chip + * for Bit Clock Invert (BCI). The inverted logic is only + * seen for the case of TDM mode. The remaining cases have + * normal logic. + */ + if (max98090->tdm_slots > 1) { + regval ^= M98090_BCI_MASK; + } + + snd_soc_write(codec, + M98090_REG_INTERFACE_FORMAT, regval); + } + + return 0; +} + +static int max98090_set_tdm_slot(struct snd_soc_dai *codec_dai, + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct max98090_cdata *cdata; + cdata = &max98090->dai[0]; + + if (slots < 0 || slots > 4) + return -EINVAL; + + max98090->tdm_slots = slots; + max98090->tdm_width = slot_width; + + if (max98090->tdm_slots > 1) { + /* SLOTL SLOTR SLOTDLY */ + snd_soc_write(codec, M98090_REG_TDM_FORMAT, + 0 << M98090_TDM_SLOTL_SHIFT | + 1 << M98090_TDM_SLOTR_SHIFT | + 0 << M98090_TDM_SLOTDLY_SHIFT); + + /* FSW TDM */ + snd_soc_update_bits(codec, M98090_REG_TDM_CONTROL, + M98090_TDM_MASK, + M98090_TDM_MASK); + } + + /* + * Normally advisable to set TDM first, but this permits either order + */ + cdata->fmt = 0; + max98090_dai_set_fmt(codec_dai, max98090->dai_fmt); + + return 0; +} + +static int max98090_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = regcache_sync(max98090->regmap); + + if (ret != 0) { + dev_err(codec->dev, + "Failed to sync cache: %d\n", ret); + return ret; + } + } + + if (max98090->jack_state == M98090_JACK_STATE_HEADSET) { + /* + * Set to normal bias level. + */ + snd_soc_update_bits(codec, M98090_REG_MIC_BIAS_VOLTAGE, + M98090_MBVSEL_MASK, M98090_MBVSEL_2V8); + } + break; + + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + case SND_SOC_BIAS_OFF: + /* Set internal pull-up to lowest power mode */ + snd_soc_update_bits(codec, M98090_REG_JACK_DETECT, + M98090_JDWK_MASK, M98090_JDWK_MASK); + regcache_mark_dirty(max98090->regmap); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static const int comp_pclk_rates[] = { + 11289600, 12288000, 12000000, 13000000, 19200000 +}; + +static const int dmic_micclk[] = { + 2, 2, 2, 2, 4, 2 +}; + +static const int comp_lrclk_rates[] = { + 8000, 16000, 32000, 44100, 48000, 96000 +}; + +static const int dmic_comp[6][6] = { + {7, 8, 3, 3, 3, 3}, + {7, 8, 3, 3, 3, 3}, + {7, 8, 3, 3, 3, 3}, + {7, 8, 3, 1, 1, 1}, + {7, 8, 3, 1, 2, 2}, + {7, 8, 3, 3, 3, 3} +}; + +static int max98090_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct max98090_cdata *cdata; + int i, j; + + cdata = &max98090->dai[0]; + max98090->bclk = snd_soc_params_to_bclk(params); + if (params_channels(params) == 1) + max98090->bclk *= 2; + + max98090->lrclk = params_rate(params); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + snd_soc_update_bits(codec, M98090_REG_INTERFACE_FORMAT, + M98090_WS_MASK, 0); + break; + default: + return -EINVAL; + } + + max98090_configure_bclk(codec); + + cdata->rate = max98090->lrclk; + + /* Update filter mode */ + if (max98090->lrclk < 24000) + snd_soc_update_bits(codec, M98090_REG_FILTER_CONFIG, + M98090_MODE_MASK, 0); + else + snd_soc_update_bits(codec, M98090_REG_FILTER_CONFIG, + M98090_MODE_MASK, M98090_MODE_MASK); + + /* Update sample rate mode */ + if (max98090->lrclk < 50000) + snd_soc_update_bits(codec, M98090_REG_FILTER_CONFIG, + M98090_DHF_MASK, 0); + else + snd_soc_update_bits(codec, M98090_REG_FILTER_CONFIG, + M98090_DHF_MASK, M98090_DHF_MASK); + + /* Check for supported PCLK to LRCLK ratios */ + for (j = 0; j < ARRAY_SIZE(comp_pclk_rates); j++) { + if (comp_pclk_rates[j] == max98090->sysclk) { + break; + } + } + + for (i = 0; i < ARRAY_SIZE(comp_lrclk_rates) - 1; i++) { + if (max98090->lrclk <= (comp_lrclk_rates[i] + + comp_lrclk_rates[i + 1]) / 2) { + break; + } + } + + snd_soc_update_bits(codec, M98090_REG_DIGITAL_MIC_ENABLE, + M98090_MICCLK_MASK, + dmic_micclk[j] << M98090_MICCLK_SHIFT); + + snd_soc_update_bits(codec, M98090_REG_DIGITAL_MIC_CONFIG, + M98090_DMIC_COMP_MASK, + dmic_comp[j][i] << M98090_DMIC_COMP_SHIFT); + + return 0; +} + +/* + * PLL / Sysclk + */ +static int max98090_dai_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + + /* Requested clock frequency is already setup */ + if (freq == max98090->sysclk) + return 0; + + /* Setup clocks for slave mode, and using the PLL + * PSCLK = 0x01 (when master clk is 10MHz to 20MHz) + * 0x02 (when master clk is 20MHz to 40MHz).. + * 0x03 (when master clk is 40MHz to 60MHz).. + */ + if ((freq >= 10000000) && (freq < 20000000)) { + snd_soc_write(codec, M98090_REG_SYSTEM_CLOCK, + M98090_PSCLK_DIV1); + } else if ((freq >= 20000000) && (freq < 40000000)) { + snd_soc_write(codec, M98090_REG_SYSTEM_CLOCK, + M98090_PSCLK_DIV2); + } else if ((freq >= 40000000) && (freq < 60000000)) { + snd_soc_write(codec, M98090_REG_SYSTEM_CLOCK, + M98090_PSCLK_DIV4); + } else { + dev_err(codec->dev, "Invalid master clock frequency\n"); + return -EINVAL; + } + + max98090->sysclk = freq; + + max98090_configure_bclk(codec); + + return 0; +} + +static int max98090_dai_digital_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + int regval; + + regval = mute ? M98090_DVM_MASK : 0; + snd_soc_update_bits(codec, M98090_REG_DAI_PLAYBACK_LEVEL, + M98090_DVM_MASK, regval); + + return 0; +} + +static void max98090_jack_work(struct work_struct *work) +{ + struct max98090_priv *max98090 = container_of(work, + struct max98090_priv, + jack_work.work); + struct snd_soc_codec *codec = max98090->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + int status = 0; + int reg; + + /* Read a second time */ + if (max98090->jack_state == M98090_JACK_STATE_NO_HEADSET) { + + /* Strong pull up allows mic detection */ + snd_soc_update_bits(codec, M98090_REG_JACK_DETECT, + M98090_JDWK_MASK, 0); + + msleep(50); + + reg = snd_soc_read(codec, M98090_REG_JACK_STATUS); + + /* Weak pull up allows only insertion detection */ + snd_soc_update_bits(codec, M98090_REG_JACK_DETECT, + M98090_JDWK_MASK, M98090_JDWK_MASK); + } else { + reg = snd_soc_read(codec, M98090_REG_JACK_STATUS); + } + + reg = snd_soc_read(codec, M98090_REG_JACK_STATUS); + + switch (reg & (M98090_LSNS_MASK | M98090_JKSNS_MASK)) { + case M98090_LSNS_MASK | M98090_JKSNS_MASK: + dev_dbg(codec->dev, "No Headset Detected\n"); + + max98090->jack_state = M98090_JACK_STATE_NO_HEADSET; + + status |= 0; + + break; + + case 0: + if (max98090->jack_state == + M98090_JACK_STATE_HEADSET) { + + dev_dbg(codec->dev, + "Headset Button Down Detected\n"); + + /* + * max98090_headset_button_event(codec) + * could be defined, then called here. + */ + + status |= SND_JACK_HEADSET; + status |= SND_JACK_BTN_0; + + break; + } + + /* Line is reported as Headphone */ + /* Nokia Headset is reported as Headphone */ + /* Mono Headphone is reported as Headphone */ + dev_dbg(codec->dev, "Headphone Detected\n"); + + max98090->jack_state = M98090_JACK_STATE_HEADPHONE; + + status |= SND_JACK_HEADPHONE; + + break; + + case M98090_JKSNS_MASK: + dev_dbg(codec->dev, "Headset Detected\n"); + + max98090->jack_state = M98090_JACK_STATE_HEADSET; + + status |= SND_JACK_HEADSET; + + break; + + default: + dev_dbg(codec->dev, "Unrecognized Jack Status\n"); + break; + } + + snd_soc_jack_report(max98090->jack, status, + SND_JACK_HEADSET | SND_JACK_BTN_0); + + snd_soc_dapm_sync(dapm); +} + +static irqreturn_t max98090_interrupt(int irq, void *data) +{ + struct snd_soc_codec *codec = data; + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + int ret; + unsigned int mask; + unsigned int active; + + dev_dbg(codec->dev, "***** max98090_interrupt *****\n"); + + ret = regmap_read(max98090->regmap, M98090_REG_INTERRUPT_S, &mask); + + if (ret != 0) { + dev_err(codec->dev, + "failed to read M98090_REG_INTERRUPT_S: %d\n", + ret); + return IRQ_NONE; + } + + ret = regmap_read(max98090->regmap, M98090_REG_DEVICE_STATUS, &active); + + if (ret != 0) { + dev_err(codec->dev, + "failed to read M98090_REG_DEVICE_STATUS: %d\n", + ret); + return IRQ_NONE; + } + + dev_dbg(codec->dev, "active=0x%02x mask=0x%02x -> active=0x%02x\n", + active, mask, active & mask); + + active &= mask; + + if (!active) + return IRQ_NONE; + + if (active & M98090_CLD_MASK) { + dev_err(codec->dev, "M98090_CLD_MASK\n"); + } + + if (active & M98090_SLD_MASK) { + dev_dbg(codec->dev, "M98090_SLD_MASK\n"); + } + + if (active & M98090_ULK_MASK) { + dev_err(codec->dev, "M98090_ULK_MASK\n"); + } + + if (active & M98090_JDET_MASK) { + dev_dbg(codec->dev, "M98090_JDET_MASK\n"); + + pm_wakeup_event(codec->dev, 100); + + schedule_delayed_work(&max98090->jack_work, + msecs_to_jiffies(100)); + } + + if (active & M98090_DRCACT_MASK) { + dev_dbg(codec->dev, "M98090_DRCACT_MASK\n"); + } + + if (active & M98090_DRCCLP_MASK) { + dev_err(codec->dev, "M98090_DRCCLP_MASK\n"); + } + + return IRQ_HANDLED; +} + +/** + * max98090_mic_detect - Enable microphone detection via the MAX98090 IRQ + * + * @codec: MAX98090 codec + * @jack: jack to report detection events on + * + * Enable microphone detection via IRQ on the MAX98090. If GPIOs are + * being used to bring out signals to the processor then only platform + * data configuration is needed for MAX98090 and processor GPIOs should + * be configured using snd_soc_jack_add_gpios() instead. + * + * If no jack is supplied detection will be disabled. + */ +int max98090_mic_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack) +{ + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "max98090_mic_detect\n"); + + max98090->jack = jack; + if (jack) { + snd_soc_update_bits(codec, M98090_REG_INTERRUPT_S, + M98090_IJDET_MASK, + 1 << M98090_IJDET_SHIFT); + } else { + snd_soc_update_bits(codec, M98090_REG_INTERRUPT_S, + M98090_IJDET_MASK, + 0); + } + + /* Send an initial empty report */ + snd_soc_jack_report(max98090->jack, 0, + SND_JACK_HEADSET | SND_JACK_BTN_0); + + schedule_delayed_work(&max98090->jack_work, + msecs_to_jiffies(100)); + + return 0; +} +EXPORT_SYMBOL_GPL(max98090_mic_detect); + +#define MAX98090_RATES SNDRV_PCM_RATE_8000_96000 +#define MAX98090_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops max98090_dai_ops = { + .set_sysclk = max98090_dai_set_sysclk, + .set_fmt = max98090_dai_set_fmt, + .set_tdm_slot = max98090_set_tdm_slot, + .hw_params = max98090_dai_hw_params, + .digital_mute = max98090_dai_digital_mute, +}; + +static struct snd_soc_dai_driver max98090_dai[] = { +{ + .name = "HiFi", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 2, + .channels_max = 2, + .rates = MAX98090_RATES, + .formats = MAX98090_FORMATS, + }, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 1, + .channels_max = 2, + .rates = MAX98090_RATES, + .formats = MAX98090_FORMATS, + }, + .ops = &max98090_dai_ops, +} +}; + +static void max98090_handle_pdata(struct snd_soc_codec *codec) +{ + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct max98090_pdata *pdata = max98090->pdata; + + if (!pdata) { + dev_err(codec->dev, "No platform data\n"); + return; + } + +} + +static int max98090_probe(struct snd_soc_codec *codec) +{ + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct max98090_cdata *cdata; + int ret = 0; + + dev_dbg(codec->dev, "max98090_probe\n"); + + max98090->codec = codec; + + codec->control_data = max98090->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + if (ret != 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + /* Reset the codec, the DSP core, and disable all interrupts */ + max98090_reset(max98090); + + /* Initialize private data */ + + max98090->sysclk = (unsigned)-1; + + cdata = &max98090->dai[0]; + cdata->rate = (unsigned)-1; + cdata->fmt = (unsigned)-1; + + max98090->lin_state = 0; + max98090->pa1en = 0; + max98090->pa2en = 0; + max98090->extmic_mux = 0; + + ret = snd_soc_read(codec, M98090_REG_REVISION_ID); + if (ret < 0) { + dev_err(codec->dev, "Failed to read device revision: %d\n", + ret); + goto err_access; + } + + if ((ret >= M98090_REVA) && (ret <= M98090_REVA + 0x0f)) { + max98090->devtype = MAX98090; + dev_info(codec->dev, "MAX98090 REVID=0x%02x\n", ret); + } else if ((ret >= M98091_REVA) && (ret <= M98091_REVA + 0x0f)) { + max98090->devtype = MAX98091; + dev_info(codec->dev, "MAX98091 REVID=0x%02x\n", ret); + } else { + max98090->devtype = MAX98090; + dev_err(codec->dev, "Unrecognized revision 0x%02x\n", ret); + } + + max98090->jack_state = M98090_JACK_STATE_NO_HEADSET; + + INIT_DELAYED_WORK(&max98090->jack_work, max98090_jack_work); + + /* Enable jack detection */ + snd_soc_write(codec, M98090_REG_JACK_DETECT, + M98090_JDETEN_MASK | M98090_JDEB_25MS); + + /* Register for interrupts */ + dev_dbg(codec->dev, "irq = %d\n", max98090->irq); + + ret = request_threaded_irq(max98090->irq, NULL, + max98090_interrupt, IRQF_TRIGGER_FALLING, + "max98090_interrupt", codec); + if (ret < 0) { + dev_err(codec->dev, "request_irq failed: %d\n", + ret); + } + + /* + * Clear any old interrupts. + * An old interrupt ocurring prior to installing the ISR + * can keep a new interrupt from generating a trigger. + */ + snd_soc_read(codec, M98090_REG_DEVICE_STATUS); + + /* High Performance is default */ + snd_soc_update_bits(codec, M98090_REG_DAC_CONTROL, + M98090_DACHP_MASK, + 1 << M98090_DACHP_SHIFT); + snd_soc_update_bits(codec, M98090_REG_DAC_CONTROL, + M98090_PERFMODE_MASK, + 0 << M98090_PERFMODE_SHIFT); + snd_soc_update_bits(codec, M98090_REG_ADC_CONTROL, + M98090_ADCHP_MASK, + 1 << M98090_ADCHP_SHIFT); + + /* Turn on VCM bandgap reference */ + snd_soc_write(codec, M98090_REG_BIAS_CONTROL, + M98090_VCM_MODE_MASK); + + max98090_handle_pdata(codec); + + max98090_add_widgets(codec); + +err_access: + return ret; +} + +static int max98090_remove(struct snd_soc_codec *codec) +{ + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + + cancel_delayed_work_sync(&max98090->jack_work); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_max98090 = { + .probe = max98090_probe, + .remove = max98090_remove, + .set_bias_level = max98090_set_bias_level, +}; + +static const struct regmap_config max98090_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = MAX98090_MAX_REGISTER, + .reg_defaults = max98090_reg, + .num_reg_defaults = ARRAY_SIZE(max98090_reg), + .volatile_reg = max98090_volatile_register, + .readable_reg = max98090_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int max98090_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct max98090_priv *max98090; + int ret; + + pr_debug("max98090_i2c_probe\n"); + + max98090 = devm_kzalloc(&i2c->dev, sizeof(struct max98090_priv), + GFP_KERNEL); + if (max98090 == NULL) + return -ENOMEM; + + max98090->devtype = id->driver_data; + i2c_set_clientdata(i2c, max98090); + max98090->control_data = i2c; + max98090->pdata = i2c->dev.platform_data; + max98090->irq = i2c->irq; + + max98090->regmap = regmap_init_i2c(i2c, &max98090_regmap); + if (IS_ERR(max98090->regmap)) { + ret = PTR_ERR(max98090->regmap); + dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret); + goto err_enable; + } + + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_max98090, max98090_dai, + ARRAY_SIZE(max98090_dai)); + if (ret < 0) + regmap_exit(max98090->regmap); + +err_enable: + return ret; +} + +static int max98090_i2c_remove(struct i2c_client *client) +{ + struct max98090_priv *max98090 = dev_get_drvdata(&client->dev); + snd_soc_unregister_codec(&client->dev); + regmap_exit(max98090->regmap); + return 0; +} + +static int max98090_runtime_resume(struct device *dev) +{ + struct max98090_priv *max98090 = dev_get_drvdata(dev); + + regcache_cache_only(max98090->regmap, false); + + regcache_sync(max98090->regmap); + + return 0; +} + +static int max98090_runtime_suspend(struct device *dev) +{ + struct max98090_priv *max98090 = dev_get_drvdata(dev); + + regcache_cache_only(max98090->regmap, true); + + return 0; +} + +static struct dev_pm_ops max98090_pm = { + SET_RUNTIME_PM_OPS(max98090_runtime_suspend, + max98090_runtime_resume, NULL) +}; + +static const struct i2c_device_id max98090_i2c_id[] = { + { "max98090", MAX98090 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, max98090_i2c_id); + +static struct i2c_driver max98090_i2c_driver = { + .driver = { + .name = "max98090", + .owner = THIS_MODULE, + .pm = &max98090_pm, + }, + .probe = max98090_i2c_probe, + .remove = max98090_i2c_remove, + .id_table = max98090_i2c_id, +}; + +module_i2c_driver(max98090_i2c_driver); + +MODULE_DESCRIPTION("ALSA SoC MAX98090 driver"); +MODULE_AUTHOR("Peter Hsiang, Jesse Marroqin, Jerry Wong"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max98090.h b/sound/soc/codecs/max98090.h new file mode 100755 index 00000000000..7e103f24905 --- /dev/null +++ b/sound/soc/codecs/max98090.h @@ -0,0 +1,1549 @@ +/* + * max98090.h -- MAX98090 ALSA SoC Audio driver + * + * Copyright 2011-2012 Maxim Integrated Products + * + * 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 _MAX98090_H +#define _MAX98090_H + +#include + +/* One can override the Linux version here with an explicit version number */ +#define M98090_LINUX_VERSION LINUX_VERSION_CODE + +/* + * MAX98090 Register Definitions + */ + +#define M98090_REG_SOFTWARE_RESET 0x00 +#define M98090_REG_DEVICE_STATUS 0x01 +#define M98090_REG_JACK_STATUS 0x02 +#define M98090_REG_INTERRUPT_S 0x03 +#define M98090_REG_QUICK_SYSTEM_CLOCK 0x04 +#define M98090_REG_QUICK_SAMPLE_RATE 0x05 +#define M98090_REG_DAI_INTERFACE 0x06 +#define M98090_REG_DAC_PATH 0x07 +#define M98090_REG_MIC_DIRECT_TO_ADC 0x08 +#define M98090_REG_LINE_TO_ADC 0x09 +#define M98090_REG_ANALOG_MIC_LOOP 0x0A +#define M98090_REG_ANALOG_LINE_LOOP 0x0B +#define M98090_REG_RESERVED 0x0C +#define M98090_REG_LINE_INPUT_CONFIG 0x0D +#define M98090_REG_LINE_INPUT_LEVEL 0x0E +#define M98090_REG_INPUT_MODE 0x0F +#define M98090_REG_MIC1_INPUT_LEVEL 0x10 +#define M98090_REG_MIC2_INPUT_LEVEL 0x11 +#define M98090_REG_MIC_BIAS_VOLTAGE 0x12 +#define M98090_REG_DIGITAL_MIC_ENABLE 0x13 +#define M98090_REG_DIGITAL_MIC_CONFIG 0x14 +#define M98090_REG_LEFT_ADC_MIXER 0x15 +#define M98090_REG_RIGHT_ADC_MIXER 0x16 +#define M98090_REG_LEFT_ADC_LEVEL 0x17 +#define M98090_REG_RIGHT_ADC_LEVEL 0x18 +#define M98090_REG_ADC_BIQUAD_LEVEL 0x19 +#define M98090_REG_ADC_SIDETONE 0x1A +#define M98090_REG_SYSTEM_CLOCK 0x1B +#define M98090_REG_CLOCK_MODE 0x1C +#define M98090_REG_CLOCK_RATIO_NI_MSB 0x1D +#define M98090_REG_CLOCK_RATIO_NI_LSB 0x1E +#define M98090_REG_CLOCK_RATIO_MI_MSB 0x1F +#define M98090_REG_CLOCK_RATIO_MI_LSB 0x20 +#define M98090_REG_MASTER_MODE 0x21 +#define M98090_REG_INTERFACE_FORMAT 0x22 +#define M98090_REG_TDM_CONTROL 0x23 +#define M98090_REG_TDM_FORMAT 0x24 +#define M98090_REG_IO_CONFIGURATION 0x25 +#define M98090_REG_FILTER_CONFIG 0x26 +#define M98090_REG_DAI_PLAYBACK_LEVEL 0x27 +#define M98090_REG_DAI_PLAYBACK_LEVEL_EQ 0x28 +#define M98090_REG_LEFT_HP_MIXER 0x29 +#define M98090_REG_RIGHT_HP_MIXER 0x2A +#define M98090_REG_HP_CONTROL 0x2B +#define M98090_REG_LEFT_HP_VOLUME 0x2C +#define M98090_REG_RIGHT_HP_VOLUME 0x2D +#define M98090_REG_LEFT_SPK_MIXER 0x2E +#define M98090_REG_RIGHT_SPK_MIXER 0x2F +#define M98090_REG_SPK_CONTROL 0x30 +#define M98090_REG_LEFT_SPK_VOLUME 0x31 +#define M98090_REG_RIGHT_SPK_VOLUME 0x32 +#define M98090_REG_DRC_TIMING 0x33 +#define M98090_REG_DRC_COMPRESSOR 0x34 +#define M98090_REG_DRC_EXPANDER 0x35 +#define M98090_REG_DRC_GAIN 0x36 +#define M98090_REG_RCV_LOUTL_MIXER 0x37 +#define M98090_REG_RCV_LOUTL_CONTROL 0x38 +#define M98090_REG_RCV_LOUTL_VOLUME 0x39 +#define M98090_REG_LOUTR_MIXER 0x3A +#define M98090_REG_LOUTR_CONTROL 0x3B +#define M98090_REG_LOUTR_VOLUME 0x3C +#define M98090_REG_JACK_DETECT 0x3D +#define M98090_REG_INPUT_ENABLE 0x3E +#define M98090_REG_OUTPUT_ENABLE 0x3F +#define M98090_REG_LEVEL_CONTROL 0x40 +#define M98090_REG_DSP_FILTER_ENABLE 0x41 +#define M98090_REG_BIAS_CONTROL 0x42 +#define M98090_REG_DAC_CONTROL 0x43 +#define M98090_REG_ADC_CONTROL 0x44 +#define M98090_REG_DEVICE_SHUTDOWN 0x45 +#define M98090_REG_EQUALIZER_BASE 0x46 +#define M98090_REG_RECORD_BIQUAD_BASE 0xAF +#define M98090_REG_DMIC3_VOLUME 0xBE +#define M98090_REG_DMIC4_VOLUME 0xBF +#define M98090_REG_DMIC34_BQ_PREATTEN 0xC0 +#define M98090_REG_RECORD_TDM_SLOT 0xC1 +#define M98090_REG_SAMPLE_RATE 0xC2 +#define M98090_REG_DMIC34_BIQUAD_BASE 0xC3 +#define M98090_REG_REVISION_ID 0xFF + +#define M98090_REG_CNT (0xFF+1) +#define MAX98090_MAX_REGISTER 0xFF + +/* MAX98090 Register Bit Fields */ + +/* + * M98090_REG_SOFTWARE_RESET + */ +#define M98090_SWRESET_MASK (1<<7) +#define M98090_SWRESET_SHIFT 7 +#define M98090_SWRESET_WIDTH 1 + +/* + * M98090_REG_DEVICE_STATUS + */ +#define M98090_CLD_MASK (1<<7) +#define M98090_CLD_SHIFT 7 +#define M98090_CLD_WIDTH 1 +#define M98090_SLD_MASK (1<<6) +#define M98090_SLD_SHIFT 6 +#define M98090_SLD_WIDTH 1 +#define M98090_ULK_MASK (1<<5) +#define M98090_ULK_SHIFT 5 +#define M98090_ULK_WIDTH 1 +#define M98090_JDET_MASK (1<<2) +#define M98090_JDET_SHIFT 2 +#define M98090_JDET_WIDTH 1 +#define M98090_DRCACT_MASK (1<<1) +#define M98090_DRCACT_SHIFT 1 +#define M98090_DRCACT_WIDTH 1 +#define M98090_DRCCLP_MASK (1<<0) +#define M98090_DRCCLP_SHIFT 0 +#define M98090_DRCCLP_WIDTH 1 + +/* + * M98090_REG_JACK_STATUS + */ +#define M98090_LSNS_MASK (1<<2) +#define M98090_LSNS_SHIFT 2 +#define M98090_LSNS_WIDTH 1 +#define M98090_JKSNS_MASK (1<<1) +#define M98090_JKSNS_SHIFT 1 +#define M98090_JKSNS_WIDTH 1 + +/* + * M98090_REG_INTERRUPT_S + */ +#define M98090_ICLD_MASK (1<<7) +#define M98090_ICLD_SHIFT 7 +#define M98090_ICLD_WIDTH 1 +#define M98090_ISLD_MASK (1<<6) +#define M98090_ISLD_SHIFT 6 +#define M98090_ISLD_WIDTH 1 +#define M98090_IULK_MASK (1<<5) +#define M98090_IULK_SHIFT 5 +#define M98090_IULK_WIDTH 1 +#define M98090_IJDET_MASK (1<<2) +#define M98090_IJDET_SHIFT 2 +#define M98090_IJDET_WIDTH 1 +#define M98090_IDRCACT_MASK (1<<1) +#define M98090_IDRCACT_SHIFT 1 +#define M98090_IDRCACT_WIDTH 1 +#define M98090_IDRCCLP_MASK (1<<0) +#define M98090_IDRCCLP_SHIFT 0 +#define M98090_IDRCCLP_WIDTH 1 + +/* + * M98090_REG_QUICK_SYSTEM_CLOCK + */ +#define M98090_26M_MASK (1<<7) +#define M98090_26M_SHIFT 7 +#define M98090_26M_WIDTH 1 +#define M98090_19P2M_MASK (1<<6) +#define M98090_19P2M_SHIFT 6 +#define M98090_19P2M_WIDTH 1 +#define M98090_13M_MASK (1<<5) +#define M98090_13M_SHIFT 5 +#define M98090_13M_WIDTH 1 +#define M98090_12P288M_MASK (1<<4) +#define M98090_12P288M_SHIFT 4 +#define M98090_12P288M_WIDTH 1 +#define M98090_12M_MASK (1<<3) +#define M98090_12M_SHIFT 3 +#define M98090_12M_WIDTH 1 +#define M98090_11P2896M_MASK (1<<2) +#define M98090_11P2896M_SHIFT 2 +#define M98090_11P2896M_WIDTH 1 +#define M98090_256FS_MASK (1<<0) +#define M98090_256FS_SHIFT 0 +#define M98090_256FS_WIDTH 1 +#define M98090_CLK_ALL_SHIFT 0 +#define M98090_CLK_ALL_WIDTH 8 +#define M98090_CLK_ALL_NUM (1<> 8) & 0xff) +#define M98090_BYTE0(w) (w & 0xff) + +/* Silicon revision number */ +#define M98090_REVA 0x40 +#define M98091_REVA 0x50 + +enum max98090_type { + MAX98090, + MAX98091, +}; + +struct max98090_cdata { + unsigned int rate; + unsigned int fmt; +}; + +struct max98090_priv { + struct regmap *regmap; + struct snd_soc_codec *codec; + enum max98090_type devtype; + void *control_data; + struct max98090_pdata *pdata; + unsigned int sysclk; + unsigned int bclk; + unsigned int lrclk; + struct max98090_cdata dai[1]; + int irq; + int jack_state; + struct delayed_work jack_work; + struct snd_soc_jack *jack; + unsigned int dai_fmt; + int tdm_slots; + int tdm_width; + u8 lin_state; + unsigned int pa1en; + unsigned int pa2en; + unsigned int extmic_mux; + unsigned int sidetone; +}; + +int max98090_mic_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack); + +#endif -- cgit v1.2.3 From 262d62eb55e105b6335ec0e9ba89f6ec7247cb48 Mon Sep 17 00:00:00 2001 From: Heiko Carstens Date: Wed, 6 Feb 2013 17:24:00 +0100 Subject: ALSA: add missing HAS_IOPORT and GENERIC_HARDIRQS dependencies MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Fix these two compile errors on s390 which does not have HAS_IOPORT nor GENERIC_HARDIRQS: sound/pci/lx6464es/lx6464es.c: In function ‘snd_lx6464es_free’: sound/pci/lx6464es/lx6464es.c:565:2: error: implicit declaration of function ‘ioport_unmap’ sound/soc/codecs/wm8903.c: In function ‘wm8903_set_pdata_irq_trigger’: sound/soc/codecs/wm8903.c:1954:9: error: implicit declaration of function ‘irq_get_irq_data’ Signed-off-by: Heiko Carstens Signed-off-by: Takashi Iwai --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3a847828932..615c4788909 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -98,7 +98,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8782 select SND_SOC_WM8804 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8900 if I2C - select SND_SOC_WM8903 if I2C + select SND_SOC_WM8903 if I2C && GENERIC_HARDIRQS select SND_SOC_WM8904 if I2C select SND_SOC_WM8940 if I2C select SND_SOC_WM8955 if I2C -- cgit v1.2.3 From e38b9b7478d57701fbcbaafdde169aa1a88d0eca Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 6 Feb 2013 13:52:42 +0000 Subject: ASoC: compress: Only mute playback streams Otherwise capture activity on a compressed DAI would mute any playback on the same DAI. Signed-off-by: Mark Brown Acked-by: Vinod Koul Acked-by: Liam Girdwood --- sound/soc/soc-compress.c | 17 +++++++++++------ 1 file changed, 11 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index c81aeecec93..35726cbf1f0 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -116,13 +116,12 @@ static int soc_compr_free(struct snd_compr_stream *cstream) if (cstream->direction == SND_COMPRESS_PLAYBACK) { cpu_dai->playback_active--; codec_dai->playback_active--; + snd_soc_dai_digital_mute(codec_dai, 1); } else { cpu_dai->capture_active--; codec_dai->capture_active--; } - snd_soc_dai_digital_mute(codec_dai, 1); - cpu_dai->active--; codec_dai->active--; codec->active--; @@ -179,10 +178,16 @@ static int soc_compr_trigger(struct snd_compr_stream *cstream, int cmd) goto out; } - if (cmd == SNDRV_PCM_TRIGGER_START) - snd_soc_dai_digital_mute(codec_dai, 0); - else if (cmd == SNDRV_PCM_TRIGGER_STOP) - snd_soc_dai_digital_mute(codec_dai, 1); + if (cstream->direction == SND_COMPRESS_PLAYBACK) { + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + snd_soc_dai_digital_mute(codec_dai, 0); + break; + case SNDRV_PCM_TRIGGER_STOP: + snd_soc_dai_digital_mute(codec_dai, 1); + break; + } + } out: mutex_unlock(&rtd->pcm_mutex); -- cgit v1.2.3 From da18396f949ecaa45007d3aeb1b81bd6da092811 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 6 Feb 2013 15:44:07 +0000 Subject: ASoC: core: Allow digital mute for capture Help avoid noise from the power up of the capture path propagating through into the start of the recording (especially noise caused by the ramp of microphone biases) by keeping the capture muted until after we've finished powering things up with DAPM in the same manner we do for playback. This allows us to take advantage of soft mute support in the hardware more effectively and is more consistent. The core code using the existing digital mute operation is updated to take advantage of this. Some additional cases in the soc-pcm code and suspend will need separate handling but these are less practically relevant than the main runtime stream start/stop case. Rather than refactor the digital mute function in every single driver a new operation is added for drivers taking advantage of this functionality, the old operation should be phased out over time. Signed-off-by: Mark Brown Acked-by Vinod Koul Acked-by: Liam Girdwood --- sound/soc/soc-compress.c | 19 +++++++++---------- sound/soc/soc-core.c | 12 ++++++++++-- sound/soc/soc-dapm.c | 6 ++++-- sound/soc/soc-pcm.c | 7 +++---- 4 files changed, 26 insertions(+), 18 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 35726cbf1f0..b5b3db71e25 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -116,12 +116,13 @@ static int soc_compr_free(struct snd_compr_stream *cstream) if (cstream->direction == SND_COMPRESS_PLAYBACK) { cpu_dai->playback_active--; codec_dai->playback_active--; - snd_soc_dai_digital_mute(codec_dai, 1); } else { cpu_dai->capture_active--; codec_dai->capture_active--; } + snd_soc_dai_digital_mute(codec_dai, 1, cstream->direction); + cpu_dai->active--; codec_dai->active--; codec->active--; @@ -178,15 +179,13 @@ static int soc_compr_trigger(struct snd_compr_stream *cstream, int cmd) goto out; } - if (cstream->direction == SND_COMPRESS_PLAYBACK) { - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - snd_soc_dai_digital_mute(codec_dai, 0); - break; - case SNDRV_PCM_TRIGGER_STOP: - snd_soc_dai_digital_mute(codec_dai, 1); - break; - } + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + snd_soc_dai_digital_mute(codec_dai, 0, cstream->direction); + break; + case SNDRV_PCM_TRIGGER_STOP: + snd_soc_dai_digital_mute(codec_dai, 1, cstream->direction); + break; } out: diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2370063b582..4eac2279789 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3540,12 +3540,20 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate); * snd_soc_dai_digital_mute - configure DAI system or master clock. * @dai: DAI * @mute: mute enable + * @direction: stream to mute * * Mutes the DAI DAC. */ -int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute) +int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, + int direction) { - if (dai->driver && dai->driver->ops->digital_mute) + if (!dai->driver) + return -ENOTSUPP; + + if (dai->driver->ops->mute_stream) + return dai->driver->ops->mute_stream(dai, mute, direction); + else if (direction == SNDRV_PCM_STREAM_PLAYBACK && + dai->driver->ops->digital_mute) return dai->driver->ops->digital_mute(dai, mute); else return -ENOTSUPP; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1e36bc81e5a..4d664f3df80 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3247,14 +3247,16 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_POST_PMU: - ret = snd_soc_dai_digital_mute(sink, 0); + ret = snd_soc_dai_digital_mute(sink, 0, + SNDRV_PCM_STREAM_PLAYBACK); if (ret != 0 && ret != -ENOTSUPP) dev_warn(sink->dev, "ASoC: Failed to unmute: %d\n", ret); ret = 0; break; case SND_SOC_DAPM_PRE_PMD: - ret = snd_soc_dai_digital_mute(sink, 1); + ret = snd_soc_dai_digital_mute(sink, 1, + SNDRV_PCM_STREAM_PLAYBACK); if (ret != 0 && ret != -ENOTSUPP) dev_warn(sink->dev, "ASoC: Failed to mute: %d\n", ret); ret = 0; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index cf191e6aebb..d675b4ae0df 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -383,8 +383,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) /* Muting the DAC suppresses artifacts caused during digital * shutdown, for example from stopping clocks. */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - snd_soc_dai_digital_mute(codec_dai, 1); + snd_soc_dai_digital_mute(codec_dai, 1, substream->stream); if (cpu_dai->driver->ops->shutdown) cpu_dai->driver->ops->shutdown(substream, cpu_dai); @@ -488,7 +487,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) snd_soc_dapm_stream_event(rtd, substream->stream, SND_SOC_DAPM_STREAM_START); - snd_soc_dai_digital_mute(codec_dai, 0); + snd_soc_dai_digital_mute(codec_dai, 0, substream->stream); out: mutex_unlock(&rtd->pcm_mutex); @@ -586,7 +585,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) /* apply codec digital mute */ if (!codec->active) - snd_soc_dai_digital_mute(codec_dai, 1); + snd_soc_dai_digital_mute(codec_dai, 1, substream->stream); /* free any machine hw params */ if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) -- cgit v1.2.3 From bdaacea35960ae72e84bd481e69324da9f4d4de5 Mon Sep 17 00:00:00 2001 From: Chris Rattray Date: Fri, 8 Feb 2013 14:32:15 +0000 Subject: ASoC: wm_adsp: round to 4-byte boundary for coeff file blocks Signed-off-by: Chris Rattray Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index be45e2b8b59..f3f7e75f862 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -711,6 +711,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) int ret, pos, blocks, type, offset, reg; char *file; struct wm_adsp_buf *buf; + int tmp; file = kzalloc(PAGE_SIZE, GFP_KERNEL); if (file == NULL) @@ -842,7 +843,12 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) } } - pos += le32_to_cpu(blk->len) + sizeof(*blk); + tmp = le32_to_cpu(blk->len) % 4; + if (tmp) + pos += le32_to_cpu(blk->len) + (4 - tmp) + sizeof(*blk); + else + pos += le32_to_cpu(blk->len) + sizeof(*blk); + blocks++; } -- cgit v1.2.3 From a10807aef5cab5568e70a2d1597305999d93f85d Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Mon, 11 Feb 2013 13:39:08 -0200 Subject: ASoC: fsl: imx-audmux: Fix sparse warning Fix the following sparse warning: sound/soc/fsl/imx-audmux.c:182:3: warning: symbol 'audmux_type' was not declared. Should it be static? Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/fsl/imx-audmux.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index 251f4d981e0..fab912ea7a5 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -176,7 +176,7 @@ static inline void audmux_debugfs_remove(void) } #endif -enum imx_audmux_type { +static enum imx_audmux_type { IMX21_AUDMUX, IMX31_AUDMUX, } audmux_type; -- cgit v1.2.3 From 41c4d554e75ccbdb043878049cf2671dd49598a5 Mon Sep 17 00:00:00 2001 From: Sebastien Guiriec Date: Mon, 11 Feb 2013 14:02:42 +0100 Subject: ASoC: omap-mcpdm: Remove useless ressource get. Remove unused memory ressource get from McPDM driver. Signed-off-by: Sebastien Guiriec Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/omap-mcpdm.c | 4 ---- 1 file changed, 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index 2fe8be20945..5ca11bdac21 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -449,10 +449,6 @@ static int asoc_mcpdm_probe(struct platform_device *pdev) omap_mcpdm_dai_dma_params[0].port_addr = res->start + MCPDM_REG_DN_DATA; omap_mcpdm_dai_dma_params[1].port_addr = res->start + MCPDM_REG_UP_DATA; - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (res == NULL) - return -ENOMEM; - res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "dn_link"); if (!res) return -ENODEV; -- cgit v1.2.3 From 43cd8bf1c8d8f6e897ed0f2c4bd50a4266b5c36e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 6 Feb 2013 16:57:29 +0000 Subject: ASoC: arizona: Automatically manage input mutes For optimal performance the inputs should be kept muted until after power up. Since there are few use cases for muting inputs during capture move the mutes to automatic control. Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 17 +++++++++++++++++ sound/soc/codecs/wm5102.c | 13 ------------- sound/soc/codecs/wm5110.c | 17 ----------------- 3 files changed, 17 insertions(+), 30 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index d824c984c8a..ac948a671ea 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -335,6 +335,23 @@ EXPORT_SYMBOL_GPL(arizona_ng_hold); int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + unsigned int reg; + + if (w->shift % 2) + reg = ARIZONA_ADC_DIGITAL_VOLUME_1L + ((w->shift / 2) * 8); + else + reg = ARIZONA_ADC_DIGITAL_VOLUME_1R + ((w->shift / 2) * 8); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(w->codec, reg, ARIZONA_IN1L_MUTE, 0); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(w->codec, reg, ARIZONA_IN1L_MUTE, + ARIZONA_IN1L_MUTE); + break; + } + return 0; } EXPORT_SYMBOL_GPL(arizona_in_ev); diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 5e85b645f2e..ab69c83626c 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -636,19 +636,6 @@ SOC_SINGLE_RANGE_TLV("IN3L Volume", ARIZONA_IN3L_CONTROL, SOC_SINGLE_RANGE_TLV("IN3R Volume", ARIZONA_IN3R_CONTROL, ARIZONA_IN3R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), -SOC_SINGLE("IN1L Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_1L, - ARIZONA_IN1L_MUTE_SHIFT, 1, 1), -SOC_SINGLE("IN1R Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_1R, - ARIZONA_IN1R_MUTE_SHIFT, 1, 1), -SOC_SINGLE("IN2L Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_2L, - ARIZONA_IN2L_MUTE_SHIFT, 1, 1), -SOC_SINGLE("IN2R Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_2R, - ARIZONA_IN2R_MUTE_SHIFT, 1, 1), -SOC_SINGLE("IN3L Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_3L, - ARIZONA_IN3L_MUTE_SHIFT, 1, 1), -SOC_SINGLE("IN3R Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_3R, - ARIZONA_IN3R_MUTE_SHIFT, 1, 1), - SOC_SINGLE_TLV("IN1L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L, ARIZONA_IN1L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), SOC_SINGLE_TLV("IN1R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1R, diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 23199372d51..a1631320b44 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -80,23 +80,6 @@ SOC_SINGLE_RANGE_TLV("IN3L Volume", ARIZONA_IN3L_CONTROL, SOC_SINGLE_RANGE_TLV("IN3R Volume", ARIZONA_IN3R_CONTROL, ARIZONA_IN3R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), -SOC_SINGLE("IN1L Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_1L, - ARIZONA_IN1L_MUTE_SHIFT, 1, 1), -SOC_SINGLE("IN1R Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_1R, - ARIZONA_IN1R_MUTE_SHIFT, 1, 1), -SOC_SINGLE("IN2L Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_2L, - ARIZONA_IN2L_MUTE_SHIFT, 1, 1), -SOC_SINGLE("IN2R Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_2R, - ARIZONA_IN2R_MUTE_SHIFT, 1, 1), -SOC_SINGLE("IN3L Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_3L, - ARIZONA_IN3L_MUTE_SHIFT, 1, 1), -SOC_SINGLE("IN3R Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_3R, - ARIZONA_IN3R_MUTE_SHIFT, 1, 1), -SOC_SINGLE("IN4L Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_4L, - ARIZONA_IN4L_MUTE_SHIFT, 1, 1), -SOC_SINGLE("IN4R Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_4R, - ARIZONA_IN4R_MUTE_SHIFT, 1, 1), - SOC_SINGLE_TLV("IN1L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L, ARIZONA_IN1L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), SOC_SINGLE_TLV("IN1R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1R, -- cgit v1.2.3 From ef5c2eba2412596f1a022c11caf74428bffd9abe Mon Sep 17 00:00:00 2001 From: Adam Thomson Date: Thu, 14 Feb 2013 12:02:51 +0000 Subject: ASoC: codecs: Add da7213 codec This patch adds support for the Dialog DA7213 audio codec. Signed-off-by: Adam Thomson Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/da7213.c | 1599 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/da7213.h | 523 +++++++++++++++ 4 files changed, 2128 insertions(+) create mode 100644 sound/soc/codecs/da7213.c create mode 100644 sound/soc/codecs/da7213.h (limited to 'sound/soc') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3a847828932..751476aa781 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -36,6 +36,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI select SND_SOC_CX20442 select SND_SOC_DA7210 if I2C + select SND_SOC_DA7213 if I2C select SND_SOC_DA732X if I2C select SND_SOC_DA9055 if I2C select SND_SOC_DFBMCS320 @@ -247,6 +248,9 @@ config SND_SOC_L3 config SND_SOC_DA7210 tristate +config SND_SOC_DA7213 + tristate + config SND_SOC_DA732X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f6e8e36cceb..6a3b3c3b8b4 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -23,6 +23,7 @@ snd-soc-cs4270-objs := cs4270.o snd-soc-cs4271-objs := cs4271.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o +snd-soc-da7213-objs := da7213.o snd-soc-da732x-objs := da732x.o snd-soc-da9055-objs := da9055.o snd-soc-dfbmcs320-objs := dfbmcs320.o @@ -147,6 +148,7 @@ obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o +obj-$(CONFIG_SND_SOC_DA7213) += snd-soc-da7213.o obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o obj-$(CONFIG_SND_SOC_DFBMCS320) += snd-soc-dfbmcs320.o diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c new file mode 100644 index 00000000000..41230ad1c3e --- /dev/null +++ b/sound/soc/codecs/da7213.c @@ -0,0 +1,1599 @@ +/* + * DA7213 ALSA SoC Codec Driver + * + * Copyright (c) 2013 Dialog Semiconductor + * + * Author: Adam Thomson + * Based on DA9055 ALSA SoC codec driver. + * + * 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 "da7213.h" + + +/* Gain and Volume */ +static const unsigned int aux_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + /* -54dB */ + 0x0, 0x11, TLV_DB_SCALE_ITEM(-5400, 0, 0), + /* -52.5dB to 15dB */ + 0x12, 0x3f, TLV_DB_SCALE_ITEM(-5250, 150, 0) +}; + +static const unsigned int digital_gain_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x07, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -78dB to 12dB */ + 0x08, 0x7f, TLV_DB_SCALE_ITEM(-7800, 75, 0) +}; + +static const unsigned int alc_analog_gain_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x0, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* 0dB to 36dB */ + 0x01, 0x07, TLV_DB_SCALE_ITEM(0, 600, 0) +}; + +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(mixin_gain_tlv, -450, 150, 0); +static const DECLARE_TLV_DB_SCALE(eq_gain_tlv, -1050, 150, 0); +static const DECLARE_TLV_DB_SCALE(hp_vol_tlv, -5700, 100, 0); +static const DECLARE_TLV_DB_SCALE(lineout_vol_tlv, -4800, 100, 0); +static const DECLARE_TLV_DB_SCALE(alc_threshold_tlv, -9450, 150, 0); +static const DECLARE_TLV_DB_SCALE(alc_gain_tlv, 0, 600, 0); + +/* ADC and DAC voice mode (8kHz) high pass cutoff value */ +static const char * const da7213_voice_hpf_corner_txt[] = { + "2.5Hz", "25Hz", "50Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz" +}; + +static const struct soc_enum da7213_dac_voice_hpf_corner = + SOC_ENUM_SINGLE(DA7213_DAC_FILTERS1, DA7213_VOICE_HPF_CORNER_SHIFT, + DA7213_VOICE_HPF_CORNER_MAX, + da7213_voice_hpf_corner_txt); + +static const struct soc_enum da7213_adc_voice_hpf_corner = + SOC_ENUM_SINGLE(DA7213_ADC_FILTERS1, DA7213_VOICE_HPF_CORNER_SHIFT, + DA7213_VOICE_HPF_CORNER_MAX, + da7213_voice_hpf_corner_txt); + +/* ADC and DAC high pass filter cutoff value */ +static const char * const da7213_audio_hpf_corner_txt[] = { + "Fs/24000", "Fs/12000", "Fs/6000", "Fs/3000" +}; + +static const struct soc_enum da7213_dac_audio_hpf_corner = + SOC_ENUM_SINGLE(DA7213_DAC_FILTERS1, DA7213_AUDIO_HPF_CORNER_SHIFT, + DA7213_AUDIO_HPF_CORNER_MAX, + da7213_audio_hpf_corner_txt); + +static const struct soc_enum da7213_adc_audio_hpf_corner = + SOC_ENUM_SINGLE(DA7213_ADC_FILTERS1, DA7213_AUDIO_HPF_CORNER_SHIFT, + DA7213_AUDIO_HPF_CORNER_MAX, + da7213_audio_hpf_corner_txt); + +/* Gain ramping rate value */ +static const char * const da7213_gain_ramp_rate_txt[] = { + "nominal rate * 8", "nominal rate * 16", "nominal rate / 16", + "nominal rate / 32" +}; + +static const struct soc_enum da7213_gain_ramp_rate = + SOC_ENUM_SINGLE(DA7213_GAIN_RAMP_CTRL, DA7213_GAIN_RAMP_RATE_SHIFT, + DA7213_GAIN_RAMP_RATE_MAX, da7213_gain_ramp_rate_txt); + +/* DAC noise gate setup time value */ +static const char * const da7213_dac_ng_setup_time_txt[] = { + "256 samples", "512 samples", "1024 samples", "2048 samples" +}; + +static const struct soc_enum da7213_dac_ng_setup_time = + SOC_ENUM_SINGLE(DA7213_DAC_NG_SETUP_TIME, + DA7213_DAC_NG_SETUP_TIME_SHIFT, + DA7213_DAC_NG_SETUP_TIME_MAX, + da7213_dac_ng_setup_time_txt); + +/* DAC noise gate rampup rate value */ +static const char * const da7213_dac_ng_rampup_txt[] = { + "0.02 ms/dB", "0.16 ms/dB" +}; + +static const struct soc_enum da7213_dac_ng_rampup_rate = + SOC_ENUM_SINGLE(DA7213_DAC_NG_SETUP_TIME, + DA7213_DAC_NG_RAMPUP_RATE_SHIFT, + DA7213_DAC_NG_RAMP_RATE_MAX, + da7213_dac_ng_rampup_txt); + +/* DAC noise gate rampdown rate value */ +static const char * const da7213_dac_ng_rampdown_txt[] = { + "0.64 ms/dB", "20.48 ms/dB" +}; + +static const struct soc_enum da7213_dac_ng_rampdown_rate = + SOC_ENUM_SINGLE(DA7213_DAC_NG_SETUP_TIME, + DA7213_DAC_NG_RAMPDN_RATE_SHIFT, + DA7213_DAC_NG_RAMP_RATE_MAX, + da7213_dac_ng_rampdown_txt); + +/* DAC soft mute rate value */ +static const char * const da7213_dac_soft_mute_rate_txt[] = { + "1", "2", "4", "8", "16", "32", "64" +}; + +static const struct soc_enum da7213_dac_soft_mute_rate = + SOC_ENUM_SINGLE(DA7213_DAC_FILTERS5, DA7213_DAC_SOFTMUTE_RATE_SHIFT, + DA7213_DAC_SOFTMUTE_RATE_MAX, + da7213_dac_soft_mute_rate_txt); + +/* ALC Attack Rate select */ +static const char * const da7213_alc_attack_rate_txt[] = { + "44/fs", "88/fs", "176/fs", "352/fs", "704/fs", "1408/fs", "2816/fs", + "5632/fs", "11264/fs", "22528/fs", "45056/fs", "90112/fs", "180224/fs" +}; + +static const struct soc_enum da7213_alc_attack_rate = + SOC_ENUM_SINGLE(DA7213_ALC_CTRL2, DA7213_ALC_ATTACK_SHIFT, + DA7213_ALC_ATTACK_MAX, da7213_alc_attack_rate_txt); + +/* ALC Release Rate select */ +static const char * const da7213_alc_release_rate_txt[] = { + "176/fs", "352/fs", "704/fs", "1408/fs", "2816/fs", "5632/fs", + "11264/fs", "22528/fs", "45056/fs", "90112/fs", "180224/fs" +}; + +static const struct soc_enum da7213_alc_release_rate = + SOC_ENUM_SINGLE(DA7213_ALC_CTRL2, DA7213_ALC_RELEASE_SHIFT, + DA7213_ALC_RELEASE_MAX, da7213_alc_release_rate_txt); + +/* ALC Hold Time select */ +static const char * const da7213_alc_hold_time_txt[] = { + "62/fs", "124/fs", "248/fs", "496/fs", "992/fs", "1984/fs", "3968/fs", + "7936/fs", "15872/fs", "31744/fs", "63488/fs", "126976/fs", + "253952/fs", "507904/fs", "1015808/fs", "2031616/fs" +}; + +static const struct soc_enum da7213_alc_hold_time = + SOC_ENUM_SINGLE(DA7213_ALC_CTRL3, DA7213_ALC_HOLD_SHIFT, + DA7213_ALC_HOLD_MAX, da7213_alc_hold_time_txt); + +/* ALC Input Signal Tracking rate select */ +static const char * const da7213_alc_integ_rate_txt[] = { + "1/4", "1/16", "1/256", "1/65536" +}; + +static const struct soc_enum da7213_alc_integ_attack_rate = + SOC_ENUM_SINGLE(DA7213_ALC_CTRL3, DA7213_ALC_INTEG_ATTACK_SHIFT, + DA7213_ALC_INTEG_MAX, da7213_alc_integ_rate_txt); + +static const struct soc_enum da7213_alc_integ_release_rate = + SOC_ENUM_SINGLE(DA7213_ALC_CTRL3, DA7213_ALC_INTEG_RELEASE_SHIFT, + DA7213_ALC_INTEG_MAX, da7213_alc_integ_rate_txt); + + +/* + * Control Functions + */ + +static int da7213_get_alc_data(struct snd_soc_codec *codec, u8 reg_val) +{ + int mid_data, top_data; + int sum = 0; + u8 iteration; + + for (iteration = 0; iteration < DA7213_ALC_AVG_ITERATIONS; + iteration++) { + /* Select the left or right channel and capture data */ + snd_soc_write(codec, DA7213_ALC_CIC_OP_LVL_CTRL, reg_val); + + /* Select middle 8 bits for read back from data register */ + snd_soc_write(codec, DA7213_ALC_CIC_OP_LVL_CTRL, + reg_val | DA7213_ALC_DATA_MIDDLE); + mid_data = snd_soc_read(codec, DA7213_ALC_CIC_OP_LVL_DATA); + + /* Select top 8 bits for read back from data register */ + snd_soc_write(codec, DA7213_ALC_CIC_OP_LVL_CTRL, + reg_val | DA7213_ALC_DATA_TOP); + top_data = snd_soc_read(codec, DA7213_ALC_CIC_OP_LVL_DATA); + + sum += ((mid_data << 8) | (top_data << 16)); + } + + return sum / DA7213_ALC_AVG_ITERATIONS; +} + +static void da7213_alc_calib_man(struct snd_soc_codec *codec) +{ + u8 reg_val; + int avg_left_data, avg_right_data, offset_l, offset_r; + + /* Calculate average for Left and Right data */ + /* Left Data */ + avg_left_data = da7213_get_alc_data(codec, + DA7213_ALC_CIC_OP_CHANNEL_LEFT); + /* Right Data */ + avg_right_data = da7213_get_alc_data(codec, + DA7213_ALC_CIC_OP_CHANNEL_RIGHT); + + /* Calculate DC offset */ + offset_l = -avg_left_data; + offset_r = -avg_right_data; + + reg_val = (offset_l & DA7213_ALC_OFFSET_15_8) >> 8; + snd_soc_write(codec, DA7213_ALC_OFFSET_MAN_M_L, reg_val); + reg_val = (offset_l & DA7213_ALC_OFFSET_19_16) >> 16; + snd_soc_write(codec, DA7213_ALC_OFFSET_MAN_U_L, reg_val); + + reg_val = (offset_r & DA7213_ALC_OFFSET_15_8) >> 8; + snd_soc_write(codec, DA7213_ALC_OFFSET_MAN_M_R, reg_val); + reg_val = (offset_r & DA7213_ALC_OFFSET_19_16) >> 16; + snd_soc_write(codec, DA7213_ALC_OFFSET_MAN_U_R, reg_val); + + /* Enable analog/digital gain mode & offset cancellation */ + snd_soc_update_bits(codec, DA7213_ALC_CTRL1, + DA7213_ALC_OFFSET_EN | DA7213_ALC_SYNC_MODE, + DA7213_ALC_OFFSET_EN | DA7213_ALC_SYNC_MODE); +} + +static void da7213_alc_calib_auto(struct snd_soc_codec *codec) +{ + u8 alc_ctrl1; + + /* Begin auto calibration and wait for completion */ + snd_soc_update_bits(codec, DA7213_ALC_CTRL1, DA7213_ALC_AUTO_CALIB_EN, + DA7213_ALC_AUTO_CALIB_EN); + do { + alc_ctrl1 = snd_soc_read(codec, DA7213_ALC_CTRL1); + } while (alc_ctrl1 & DA7213_ALC_AUTO_CALIB_EN); + + /* If auto calibration fails, fall back to digital gain only mode */ + if (alc_ctrl1 & DA7213_ALC_CALIB_OVERFLOW) { + dev_warn(codec->dev, + "ALC auto calibration failed with overflow\n"); + snd_soc_update_bits(codec, DA7213_ALC_CTRL1, + DA7213_ALC_OFFSET_EN | DA7213_ALC_SYNC_MODE, + 0); + } else { + /* Enable analog/digital gain mode & offset cancellation */ + snd_soc_update_bits(codec, DA7213_ALC_CTRL1, + DA7213_ALC_OFFSET_EN | DA7213_ALC_SYNC_MODE, + DA7213_ALC_OFFSET_EN | DA7213_ALC_SYNC_MODE); + } + +} + +static void da7213_alc_calib(struct snd_soc_codec *codec) +{ + struct da7213_priv *da7213 = snd_soc_codec_get_drvdata(codec); + u8 adc_l_ctrl, adc_r_ctrl; + u8 mixin_l_sel, mixin_r_sel; + u8 mic_1_ctrl, mic_2_ctrl; + + /* Save current values from ADC control registers */ + adc_l_ctrl = snd_soc_read(codec, DA7213_ADC_L_CTRL); + adc_r_ctrl = snd_soc_read(codec, DA7213_ADC_R_CTRL); + + /* Save current values from MIXIN_L/R_SELECT registers */ + mixin_l_sel = snd_soc_read(codec, DA7213_MIXIN_L_SELECT); + mixin_r_sel = snd_soc_read(codec, DA7213_MIXIN_R_SELECT); + + /* Save current values from MIC control registers */ + mic_1_ctrl = snd_soc_read(codec, DA7213_MIC_1_CTRL); + mic_2_ctrl = snd_soc_read(codec, DA7213_MIC_2_CTRL); + + /* Enable ADC Left and Right */ + snd_soc_update_bits(codec, DA7213_ADC_L_CTRL, DA7213_ADC_EN, + DA7213_ADC_EN); + snd_soc_update_bits(codec, DA7213_ADC_R_CTRL, DA7213_ADC_EN, + DA7213_ADC_EN); + + /* Enable MIC paths */ + snd_soc_update_bits(codec, DA7213_MIXIN_L_SELECT, + DA7213_MIXIN_L_MIX_SELECT_MIC_1 | + DA7213_MIXIN_L_MIX_SELECT_MIC_2, + DA7213_MIXIN_L_MIX_SELECT_MIC_1 | + DA7213_MIXIN_L_MIX_SELECT_MIC_2); + snd_soc_update_bits(codec, DA7213_MIXIN_R_SELECT, + DA7213_MIXIN_R_MIX_SELECT_MIC_2 | + DA7213_MIXIN_R_MIX_SELECT_MIC_1, + DA7213_MIXIN_R_MIX_SELECT_MIC_2 | + DA7213_MIXIN_R_MIX_SELECT_MIC_1); + + /* Mute MIC PGAs */ + snd_soc_update_bits(codec, DA7213_MIC_1_CTRL, DA7213_MUTE_EN, + DA7213_MUTE_EN); + snd_soc_update_bits(codec, DA7213_MIC_2_CTRL, DA7213_MUTE_EN, + DA7213_MUTE_EN); + + /* Perform calibration */ + if (da7213->alc_calib_auto) + da7213_alc_calib_auto(codec); + else + da7213_alc_calib_man(codec); + + /* Restore MIXIN_L/R_SELECT registers to their original states */ + snd_soc_write(codec, DA7213_MIXIN_L_SELECT, mixin_l_sel); + snd_soc_write(codec, DA7213_MIXIN_R_SELECT, mixin_r_sel); + + /* Restore ADC control registers to their original states */ + snd_soc_write(codec, DA7213_ADC_L_CTRL, adc_l_ctrl); + snd_soc_write(codec, DA7213_ADC_R_CTRL, adc_r_ctrl); + + /* Restore original values of MIC control registers */ + snd_soc_write(codec, DA7213_MIC_1_CTRL, mic_1_ctrl); + snd_soc_write(codec, DA7213_MIC_2_CTRL, mic_2_ctrl); +} + +static int da7213_put_mixin_gain(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct da7213_priv *da7213 = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = snd_soc_put_volsw_2r(kcontrol, ucontrol); + + /* If ALC in operation, make sure calibrated offsets are updated */ + if ((!ret) && (da7213->alc_en)) + da7213_alc_calib(codec); + + return ret; +} + +static int da7213_put_alc_sw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct da7213_priv *da7213 = snd_soc_codec_get_drvdata(codec); + + /* Force ALC offset calibration if enabling ALC */ + if (ucontrol->value.integer.value[0] || + ucontrol->value.integer.value[1]) { + if (!da7213->alc_en) { + da7213_alc_calib(codec); + da7213->alc_en = true; + } + } else { + da7213->alc_en = false; + } + + return snd_soc_put_volsw(kcontrol, ucontrol); +} + + +/* + * KControls + */ + +static const struct snd_kcontrol_new da7213_snd_controls[] = { + + /* Volume controls */ + SOC_SINGLE_TLV("Mic 1 Volume", DA7213_MIC_1_GAIN, + DA7213_MIC_AMP_GAIN_SHIFT, DA7213_MIC_AMP_GAIN_MAX, + DA7213_NO_INVERT, mic_vol_tlv), + SOC_SINGLE_TLV("Mic 2 Volume", DA7213_MIC_2_GAIN, + DA7213_MIC_AMP_GAIN_SHIFT, DA7213_MIC_AMP_GAIN_MAX, + DA7213_NO_INVERT, mic_vol_tlv), + SOC_DOUBLE_R_TLV("Aux Volume", DA7213_AUX_L_GAIN, DA7213_AUX_R_GAIN, + DA7213_AUX_AMP_GAIN_SHIFT, DA7213_AUX_AMP_GAIN_MAX, + DA7213_NO_INVERT, aux_vol_tlv), + SOC_DOUBLE_R_EXT_TLV("Mixin PGA Volume", DA7213_MIXIN_L_GAIN, + DA7213_MIXIN_R_GAIN, DA7213_MIXIN_AMP_GAIN_SHIFT, + DA7213_MIXIN_AMP_GAIN_MAX, DA7213_NO_INVERT, + snd_soc_get_volsw_2r, da7213_put_mixin_gain, + mixin_gain_tlv), + SOC_DOUBLE_R_TLV("ADC Volume", DA7213_ADC_L_GAIN, DA7213_ADC_R_GAIN, + DA7213_ADC_AMP_GAIN_SHIFT, DA7213_ADC_AMP_GAIN_MAX, + DA7213_NO_INVERT, digital_gain_tlv), + SOC_DOUBLE_R_TLV("DAC Volume", DA7213_DAC_L_GAIN, DA7213_DAC_R_GAIN, + DA7213_DAC_AMP_GAIN_SHIFT, DA7213_DAC_AMP_GAIN_MAX, + DA7213_NO_INVERT, digital_gain_tlv), + SOC_DOUBLE_R_TLV("Headphone Volume", DA7213_HP_L_GAIN, DA7213_HP_R_GAIN, + DA7213_HP_AMP_GAIN_SHIFT, DA7213_HP_AMP_GAIN_MAX, + DA7213_NO_INVERT, hp_vol_tlv), + SOC_SINGLE_TLV("Lineout Volume", DA7213_LINE_GAIN, + DA7213_LINE_AMP_GAIN_SHIFT, DA7213_LINE_AMP_GAIN_MAX, + DA7213_NO_INVERT, lineout_vol_tlv), + + /* DAC Equalizer controls */ + SOC_SINGLE("DAC EQ Switch", DA7213_DAC_FILTERS4, DA7213_DAC_EQ_EN_SHIFT, + DA7213_DAC_EQ_EN_MAX, DA7213_NO_INVERT), + SOC_SINGLE_TLV("DAC EQ1 Volume", DA7213_DAC_FILTERS2, + DA7213_DAC_EQ_BAND1_SHIFT, DA7213_DAC_EQ_BAND_MAX, + DA7213_NO_INVERT, eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ2 Volume", DA7213_DAC_FILTERS2, + DA7213_DAC_EQ_BAND2_SHIFT, DA7213_DAC_EQ_BAND_MAX, + DA7213_NO_INVERT, eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ3 Volume", DA7213_DAC_FILTERS3, + DA7213_DAC_EQ_BAND3_SHIFT, DA7213_DAC_EQ_BAND_MAX, + DA7213_NO_INVERT, eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ4 Volume", DA7213_DAC_FILTERS3, + DA7213_DAC_EQ_BAND4_SHIFT, DA7213_DAC_EQ_BAND_MAX, + DA7213_NO_INVERT, eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ5 Volume", DA7213_DAC_FILTERS4, + DA7213_DAC_EQ_BAND5_SHIFT, DA7213_DAC_EQ_BAND_MAX, + DA7213_NO_INVERT, eq_gain_tlv), + + /* High Pass Filter and Voice Mode controls */ + SOC_SINGLE("ADC HPF Switch", DA7213_ADC_FILTERS1, DA7213_HPF_EN_SHIFT, + DA7213_HPF_EN_MAX, DA7213_NO_INVERT), + SOC_ENUM("ADC HPF Cutoff", da7213_adc_audio_hpf_corner), + SOC_SINGLE("ADC Voice Mode Switch", DA7213_ADC_FILTERS1, + DA7213_VOICE_EN_SHIFT, DA7213_VOICE_EN_MAX, + DA7213_NO_INVERT), + SOC_ENUM("ADC Voice Cutoff", da7213_adc_voice_hpf_corner), + + SOC_SINGLE("DAC HPF Switch", DA7213_DAC_FILTERS1, DA7213_HPF_EN_SHIFT, + DA7213_HPF_EN_MAX, DA7213_NO_INVERT), + SOC_ENUM("DAC HPF Cutoff", da7213_dac_audio_hpf_corner), + SOC_SINGLE("DAC Voice Mode Switch", DA7213_DAC_FILTERS1, + DA7213_VOICE_EN_SHIFT, DA7213_VOICE_EN_MAX, + DA7213_NO_INVERT), + SOC_ENUM("DAC Voice Cutoff", da7213_dac_voice_hpf_corner), + + /* Mute controls */ + SOC_SINGLE("Mic 1 Switch", DA7213_MIC_1_CTRL, DA7213_MUTE_EN_SHIFT, + DA7213_MUTE_EN_MAX, DA7213_INVERT), + SOC_SINGLE("Mic 2 Switch", DA7213_MIC_2_CTRL, DA7213_MUTE_EN_SHIFT, + DA7213_MUTE_EN_MAX, DA7213_INVERT), + SOC_DOUBLE_R("Aux Switch", DA7213_AUX_L_CTRL, DA7213_AUX_R_CTRL, + DA7213_MUTE_EN_SHIFT, DA7213_MUTE_EN_MAX, DA7213_INVERT), + SOC_DOUBLE_R("Mixin PGA Switch", DA7213_MIXIN_L_CTRL, + DA7213_MIXIN_R_CTRL, DA7213_MUTE_EN_SHIFT, + DA7213_MUTE_EN_MAX, DA7213_INVERT), + SOC_DOUBLE_R("ADC Switch", DA7213_ADC_L_CTRL, DA7213_ADC_R_CTRL, + DA7213_MUTE_EN_SHIFT, DA7213_MUTE_EN_MAX, DA7213_INVERT), + SOC_DOUBLE_R("Headphone Switch", DA7213_HP_L_CTRL, DA7213_HP_R_CTRL, + DA7213_MUTE_EN_SHIFT, DA7213_MUTE_EN_MAX, DA7213_INVERT), + SOC_SINGLE("Lineout Switch", DA7213_LINE_CTRL, DA7213_MUTE_EN_SHIFT, + DA7213_MUTE_EN_MAX, DA7213_INVERT), + SOC_SINGLE("DAC Soft Mute Switch", DA7213_DAC_FILTERS5, + DA7213_DAC_SOFTMUTE_EN_SHIFT, DA7213_DAC_SOFTMUTE_EN_MAX, + DA7213_NO_INVERT), + SOC_ENUM("DAC Soft Mute Rate", da7213_dac_soft_mute_rate), + + /* Zero Cross controls */ + SOC_DOUBLE_R("Aux ZC Switch", DA7213_AUX_L_CTRL, DA7213_AUX_R_CTRL, + DA7213_ZC_EN_SHIFT, DA7213_ZC_EN_MAX, DA7213_NO_INVERT), + SOC_DOUBLE_R("Mixin PGA ZC Switch", DA7213_MIXIN_L_CTRL, + DA7213_MIXIN_R_CTRL, DA7213_ZC_EN_SHIFT, DA7213_ZC_EN_MAX, + DA7213_NO_INVERT), + SOC_DOUBLE_R("Headphone ZC Switch", DA7213_HP_L_CTRL, DA7213_HP_R_CTRL, + DA7213_ZC_EN_SHIFT, DA7213_ZC_EN_MAX, DA7213_NO_INVERT), + + /* Gain Ramping controls */ + SOC_DOUBLE_R("Aux Gain Ramping Switch", DA7213_AUX_L_CTRL, + DA7213_AUX_R_CTRL, DA7213_GAIN_RAMP_EN_SHIFT, + DA7213_GAIN_RAMP_EN_MAX, DA7213_NO_INVERT), + SOC_DOUBLE_R("Mixin Gain Ramping Switch", DA7213_MIXIN_L_CTRL, + DA7213_MIXIN_R_CTRL, DA7213_GAIN_RAMP_EN_SHIFT, + DA7213_GAIN_RAMP_EN_MAX, DA7213_NO_INVERT), + SOC_DOUBLE_R("ADC Gain Ramping Switch", DA7213_ADC_L_CTRL, + DA7213_ADC_R_CTRL, DA7213_GAIN_RAMP_EN_SHIFT, + DA7213_GAIN_RAMP_EN_MAX, DA7213_NO_INVERT), + SOC_DOUBLE_R("DAC Gain Ramping Switch", DA7213_DAC_L_CTRL, + DA7213_DAC_R_CTRL, DA7213_GAIN_RAMP_EN_SHIFT, + DA7213_GAIN_RAMP_EN_MAX, DA7213_NO_INVERT), + SOC_DOUBLE_R("Headphone Gain Ramping Switch", DA7213_HP_L_CTRL, + DA7213_HP_R_CTRL, DA7213_GAIN_RAMP_EN_SHIFT, + DA7213_GAIN_RAMP_EN_MAX, DA7213_NO_INVERT), + SOC_SINGLE("Lineout Gain Ramping Switch", DA7213_LINE_CTRL, + DA7213_GAIN_RAMP_EN_SHIFT, DA7213_GAIN_RAMP_EN_MAX, + DA7213_NO_INVERT), + SOC_ENUM("Gain Ramping Rate", da7213_gain_ramp_rate), + + /* DAC Noise Gate controls */ + SOC_SINGLE("DAC NG Switch", DA7213_DAC_NG_CTRL, DA7213_DAC_NG_EN_SHIFT, + DA7213_DAC_NG_EN_MAX, DA7213_NO_INVERT), + SOC_ENUM("DAC NG Setup Time", da7213_dac_ng_setup_time), + SOC_ENUM("DAC NG Rampup Rate", da7213_dac_ng_rampup_rate), + SOC_ENUM("DAC NG Rampdown Rate", da7213_dac_ng_rampdown_rate), + SOC_SINGLE("DAC NG OFF Threshold", DA7213_DAC_NG_OFF_THRESHOLD, + DA7213_DAC_NG_THRESHOLD_SHIFT, DA7213_DAC_NG_THRESHOLD_MAX, + DA7213_NO_INVERT), + SOC_SINGLE("DAC NG ON Threshold", DA7213_DAC_NG_ON_THRESHOLD, + DA7213_DAC_NG_THRESHOLD_SHIFT, DA7213_DAC_NG_THRESHOLD_MAX, + DA7213_NO_INVERT), + + /* DAC Routing & Inversion */ + SOC_DOUBLE("DAC Mono Switch", DA7213_DIG_ROUTING_DAC, + DA7213_DAC_L_MONO_SHIFT, DA7213_DAC_R_MONO_SHIFT, + DA7213_DAC_MONO_MAX, DA7213_NO_INVERT), + SOC_DOUBLE("DAC Invert Switch", DA7213_DIG_CTRL, DA7213_DAC_L_INV_SHIFT, + DA7213_DAC_R_INV_SHIFT, DA7213_DAC_INV_MAX, + DA7213_NO_INVERT), + + /* DMIC controls */ + SOC_DOUBLE_R("DMIC Switch", DA7213_MIXIN_L_SELECT, + DA7213_MIXIN_R_SELECT, DA7213_DMIC_EN_SHIFT, + DA7213_DMIC_EN_MAX, DA7213_NO_INVERT), + + /* ALC Controls */ + SOC_DOUBLE_EXT("ALC Switch", DA7213_ALC_CTRL1, DA7213_ALC_L_EN_SHIFT, + DA7213_ALC_R_EN_SHIFT, DA7213_ALC_EN_MAX, + DA7213_NO_INVERT, snd_soc_get_volsw, da7213_put_alc_sw), + SOC_ENUM("ALC Attack Rate", da7213_alc_attack_rate), + SOC_ENUM("ALC Release Rate", da7213_alc_release_rate), + SOC_ENUM("ALC Hold Time", da7213_alc_hold_time), + /* + * Rate at which input signal envelope is tracked as the signal gets + * larger + */ + SOC_ENUM("ALC Integ Attack Rate", da7213_alc_integ_attack_rate), + /* + * Rate at which input signal envelope is tracked as the signal gets + * smaller + */ + SOC_ENUM("ALC Integ Release Rate", da7213_alc_integ_release_rate), + SOC_SINGLE_TLV("ALC Noise Threshold Volume", DA7213_ALC_NOISE, + DA7213_ALC_THRESHOLD_SHIFT, DA7213_ALC_THRESHOLD_MAX, + DA7213_INVERT, alc_threshold_tlv), + SOC_SINGLE_TLV("ALC Min Threshold Volume", DA7213_ALC_TARGET_MIN, + DA7213_ALC_THRESHOLD_SHIFT, DA7213_ALC_THRESHOLD_MAX, + DA7213_INVERT, alc_threshold_tlv), + SOC_SINGLE_TLV("ALC Max Threshold Volume", DA7213_ALC_TARGET_MAX, + DA7213_ALC_THRESHOLD_SHIFT, DA7213_ALC_THRESHOLD_MAX, + DA7213_INVERT, alc_threshold_tlv), + SOC_SINGLE_TLV("ALC Max Attenuation Volume", DA7213_ALC_GAIN_LIMITS, + DA7213_ALC_ATTEN_MAX_SHIFT, + DA7213_ALC_ATTEN_GAIN_MAX_MAX, DA7213_NO_INVERT, + alc_gain_tlv), + SOC_SINGLE_TLV("ALC Max Gain Volume", DA7213_ALC_GAIN_LIMITS, + DA7213_ALC_GAIN_MAX_SHIFT, DA7213_ALC_ATTEN_GAIN_MAX_MAX, + DA7213_NO_INVERT, alc_gain_tlv), + SOC_SINGLE_TLV("ALC Min Analog Gain Volume", DA7213_ALC_ANA_GAIN_LIMITS, + DA7213_ALC_ANA_GAIN_MIN_SHIFT, DA7213_ALC_ANA_GAIN_MAX, + DA7213_NO_INVERT, alc_analog_gain_tlv), + SOC_SINGLE_TLV("ALC Max Analog Gain Volume", DA7213_ALC_ANA_GAIN_LIMITS, + DA7213_ALC_ANA_GAIN_MAX_SHIFT, DA7213_ALC_ANA_GAIN_MAX, + DA7213_NO_INVERT, alc_analog_gain_tlv), + SOC_SINGLE("ALC Anticlip Mode Switch", DA7213_ALC_ANTICLIP_CTRL, + DA7213_ALC_ANTICLIP_EN_SHIFT, DA7213_ALC_ANTICLIP_EN_MAX, + DA7213_NO_INVERT), + SOC_SINGLE("ALC Anticlip Level", DA7213_ALC_ANTICLIP_LEVEL, + DA7213_ALC_ANTICLIP_LEVEL_SHIFT, + DA7213_ALC_ANTICLIP_LEVEL_MAX, DA7213_NO_INVERT), +}; + + +/* + * DAPM + */ + +/* + * Enums + */ + +/* MIC PGA source select */ +static const char * const da7213_mic_amp_in_sel_txt[] = { + "Differential", "MIC_P", "MIC_N" +}; + +static const struct soc_enum da7213_mic_1_amp_in_sel = + SOC_ENUM_SINGLE(DA7213_MIC_1_CTRL, DA7213_MIC_AMP_IN_SEL_SHIFT, + DA7213_MIC_AMP_IN_SEL_MAX, da7213_mic_amp_in_sel_txt); +static const struct snd_kcontrol_new da7213_mic_1_amp_in_sel_mux = + SOC_DAPM_ENUM("Mic 1 Amp Source MUX", da7213_mic_1_amp_in_sel); + +static const struct soc_enum da7213_mic_2_amp_in_sel = + SOC_ENUM_SINGLE(DA7213_MIC_2_CTRL, DA7213_MIC_AMP_IN_SEL_SHIFT, + DA7213_MIC_AMP_IN_SEL_MAX, da7213_mic_amp_in_sel_txt); +static const struct snd_kcontrol_new da7213_mic_2_amp_in_sel_mux = + SOC_DAPM_ENUM("Mic 2 Amp Source MUX", da7213_mic_2_amp_in_sel); + +/* DAI routing select */ +static const char * const da7213_dai_src_txt[] = { + "ADC Left", "ADC Right", "DAI Input Left", "DAI Input Right" +}; + +static const struct soc_enum da7213_dai_l_src = + SOC_ENUM_SINGLE(DA7213_DIG_ROUTING_DAI, DA7213_DAI_L_SRC_SHIFT, + DA7213_DAI_SRC_MAX, da7213_dai_src_txt); +static const struct snd_kcontrol_new da7213_dai_l_src_mux = + SOC_DAPM_ENUM("DAI Left Source MUX", da7213_dai_l_src); + +static const struct soc_enum da7213_dai_r_src = + SOC_ENUM_SINGLE(DA7213_DIG_ROUTING_DAI, DA7213_DAI_R_SRC_SHIFT, + DA7213_DAI_SRC_MAX, da7213_dai_src_txt); +static const struct snd_kcontrol_new da7213_dai_r_src_mux = + SOC_DAPM_ENUM("DAI Right Source MUX", da7213_dai_r_src); + +/* DAC routing select */ +static const char * const da7213_dac_src_txt[] = { + "ADC Output Left", "ADC Output Right", "DAI Input Left", + "DAI Input Right" +}; + +static const struct soc_enum da7213_dac_l_src = + SOC_ENUM_SINGLE(DA7213_DIG_ROUTING_DAC, DA7213_DAC_L_SRC_SHIFT, + DA7213_DAC_SRC_MAX, da7213_dac_src_txt); +static const struct snd_kcontrol_new da7213_dac_l_src_mux = + SOC_DAPM_ENUM("DAC Left Source MUX", da7213_dac_l_src); + +static const struct soc_enum da7213_dac_r_src = + SOC_ENUM_SINGLE(DA7213_DIG_ROUTING_DAC, DA7213_DAC_R_SRC_SHIFT, + DA7213_DAC_SRC_MAX, da7213_dac_src_txt); +static const struct snd_kcontrol_new da7213_dac_r_src_mux = + SOC_DAPM_ENUM("DAC Right Source MUX", da7213_dac_r_src); + +/* + * Mixer Controls + */ + +/* Mixin Left */ +static const struct snd_kcontrol_new da7213_dapm_mixinl_controls[] = { + SOC_DAPM_SINGLE("Aux Left Switch", DA7213_MIXIN_L_SELECT, + DA7213_MIXIN_L_MIX_SELECT_AUX_L_SHIFT, + DA7213_MIXIN_L_MIX_SELECT_MAX, DA7213_NO_INVERT), + SOC_DAPM_SINGLE("Mic 1 Switch", DA7213_MIXIN_L_SELECT, + DA7213_MIXIN_L_MIX_SELECT_MIC_1_SHIFT, + DA7213_MIXIN_L_MIX_SELECT_MAX, DA7213_NO_INVERT), + SOC_DAPM_SINGLE("Mic 2 Switch", DA7213_MIXIN_L_SELECT, + DA7213_MIXIN_L_MIX_SELECT_MIC_2_SHIFT, + DA7213_MIXIN_L_MIX_SELECT_MAX, DA7213_NO_INVERT), + SOC_DAPM_SINGLE("Mixin Right Switch", DA7213_MIXIN_L_SELECT, + DA7213_MIXIN_L_MIX_SELECT_MIXIN_R_SHIFT, + DA7213_MIXIN_L_MIX_SELECT_MAX, DA7213_NO_INVERT), +}; + +/* Mixin Right */ +static const struct snd_kcontrol_new da7213_dapm_mixinr_controls[] = { + SOC_DAPM_SINGLE("Aux Right Switch", DA7213_MIXIN_R_SELECT, + DA7213_MIXIN_R_MIX_SELECT_AUX_R_SHIFT, + DA7213_MIXIN_R_MIX_SELECT_MAX, DA7213_NO_INVERT), + SOC_DAPM_SINGLE("Mic 2 Switch", DA7213_MIXIN_R_SELECT, + DA7213_MIXIN_R_MIX_SELECT_MIC_2_SHIFT, + DA7213_MIXIN_R_MIX_SELECT_MAX, DA7213_NO_INVERT), + SOC_DAPM_SINGLE("Mic 1 Switch", DA7213_MIXIN_R_SELECT, + DA7213_MIXIN_R_MIX_SELECT_MIC_1_SHIFT, + DA7213_MIXIN_R_MIX_SELECT_MAX, DA7213_NO_INVERT), + SOC_DAPM_SINGLE("Mixin Left Switch", DA7213_MIXIN_R_SELECT, + DA7213_MIXIN_R_MIX_SELECT_MIXIN_L_SHIFT, + DA7213_MIXIN_R_MIX_SELECT_MAX, DA7213_NO_INVERT), +}; + +/* Mixout Left */ +static const struct snd_kcontrol_new da7213_dapm_mixoutl_controls[] = { + SOC_DAPM_SINGLE("Aux Left Switch", DA7213_MIXOUT_L_SELECT, + DA7213_MIXOUT_L_MIX_SELECT_AUX_L_SHIFT, + DA7213_MIXOUT_L_MIX_SELECT_MAX, DA7213_NO_INVERT), + SOC_DAPM_SINGLE("Mixin Left Switch", DA7213_MIXOUT_L_SELECT, + DA7213_MIXOUT_L_MIX_SELECT_MIXIN_L_SHIFT, + DA7213_MIXOUT_L_MIX_SELECT_MAX, DA7213_NO_INVERT), + SOC_DAPM_SINGLE("Mixin Right Switch", DA7213_MIXOUT_L_SELECT, + DA7213_MIXOUT_L_MIX_SELECT_MIXIN_R_SHIFT, + DA7213_MIXOUT_L_MIX_SELECT_MAX, DA7213_NO_INVERT), + SOC_DAPM_SINGLE("DAC Left Switch", DA7213_MIXOUT_L_SELECT, + DA7213_MIXOUT_L_MIX_SELECT_DAC_L_SHIFT, + DA7213_MIXOUT_L_MIX_SELECT_MAX, DA7213_NO_INVERT), + SOC_DAPM_SINGLE("Aux Left Invert Switch", DA7213_MIXOUT_L_SELECT, + DA7213_MIXOUT_L_MIX_SELECT_AUX_L_INVERTED_SHIFT, + DA7213_MIXOUT_L_MIX_SELECT_MAX, DA7213_NO_INVERT), + SOC_DAPM_SINGLE("Mixin Left Invert Switch", DA7213_MIXOUT_L_SELECT, + DA7213_MIXOUT_L_MIX_SELECT_MIXIN_L_INVERTED_SHIFT, + DA7213_MIXOUT_L_MIX_SELECT_MAX, DA7213_NO_INVERT), + SOC_DAPM_SINGLE("Mixin Right Invert Switch", DA7213_MIXOUT_L_SELECT, + DA7213_MIXOUT_L_MIX_SELECT_MIXIN_R_INVERTED_SHIFT, + DA7213_MIXOUT_L_MIX_SELECT_MAX, DA7213_NO_INVERT), +}; + +/* Mixout Right */ +static const struct snd_kcontrol_new da7213_dapm_mixoutr_controls[] = { + SOC_DAPM_SINGLE("Aux Right Switch", DA7213_MIXOUT_R_SELECT, + DA7213_MIXOUT_R_MIX_SELECT_AUX_R_SHIFT, + DA7213_MIXOUT_R_MIX_SELECT_MAX, DA7213_NO_INVERT), + SOC_DAPM_SINGLE("Mixin Right Switch", DA7213_MIXOUT_R_SELECT, + DA7213_MIXOUT_R_MIX_SELECT_MIXIN_R_SHIFT, + DA7213_MIXOUT_R_MIX_SELECT_MAX, DA7213_NO_INVERT), + SOC_DAPM_SINGLE("Mixin Left Switch", DA7213_MIXOUT_R_SELECT, + DA7213_MIXOUT_R_MIX_SELECT_MIXIN_L_SHIFT, + DA7213_MIXOUT_R_MIX_SELECT_MAX, DA7213_NO_INVERT), + SOC_DAPM_SINGLE("DAC Right Switch", DA7213_MIXOUT_R_SELECT, + DA7213_MIXOUT_R_MIX_SELECT_DAC_R_SHIFT, + DA7213_MIXOUT_R_MIX_SELECT_MAX, DA7213_NO_INVERT), + SOC_DAPM_SINGLE("Aux Right Invert Switch", DA7213_MIXOUT_R_SELECT, + DA7213_MIXOUT_R_MIX_SELECT_AUX_R_INVERTED_SHIFT, + DA7213_MIXOUT_R_MIX_SELECT_MAX, DA7213_NO_INVERT), + SOC_DAPM_SINGLE("Mixin Right Invert Switch", DA7213_MIXOUT_R_SELECT, + DA7213_MIXOUT_R_MIX_SELECT_MIXIN_R_INVERTED_SHIFT, + DA7213_MIXOUT_R_MIX_SELECT_MAX, DA7213_NO_INVERT), + SOC_DAPM_SINGLE("Mixin Left Invert Switch", DA7213_MIXOUT_R_SELECT, + DA7213_MIXOUT_R_MIX_SELECT_MIXIN_L_INVERTED_SHIFT, + DA7213_MIXOUT_R_MIX_SELECT_MAX, DA7213_NO_INVERT), +}; + + +/* + * DAPM widgets + */ + +static const struct snd_soc_dapm_widget da7213_dapm_widgets[] = { + /* + * Input & Output + */ + + /* Use a supply here as this controls both input & output DAIs */ + SND_SOC_DAPM_SUPPLY("DAI", DA7213_DAI_CTRL, DA7213_DAI_EN_SHIFT, + DA7213_NO_INVERT, NULL, 0), + + /* + * Input + */ + + /* Input Lines */ + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("AUXL"), + SND_SOC_DAPM_INPUT("AUXR"), + + /* MUXs for Mic PGA source selection */ + SND_SOC_DAPM_MUX("Mic 1 Amp Source MUX", SND_SOC_NOPM, 0, 0, + &da7213_mic_1_amp_in_sel_mux), + SND_SOC_DAPM_MUX("Mic 2 Amp Source MUX", SND_SOC_NOPM, 0, 0, + &da7213_mic_2_amp_in_sel_mux), + + /* Input PGAs */ + SND_SOC_DAPM_PGA("Mic 1 PGA", DA7213_MIC_1_CTRL, DA7213_AMP_EN_SHIFT, + DA7213_NO_INVERT, NULL, 0), + SND_SOC_DAPM_PGA("Mic 2 PGA", DA7213_MIC_2_CTRL, DA7213_AMP_EN_SHIFT, + DA7213_NO_INVERT, NULL, 0), + SND_SOC_DAPM_PGA("Aux Left PGA", DA7213_AUX_L_CTRL, DA7213_AMP_EN_SHIFT, + DA7213_NO_INVERT, NULL, 0), + SND_SOC_DAPM_PGA("Aux Right PGA", DA7213_AUX_R_CTRL, + DA7213_AMP_EN_SHIFT, DA7213_NO_INVERT, NULL, 0), + SND_SOC_DAPM_PGA("Mixin Left PGA", DA7213_MIXIN_L_CTRL, + DA7213_AMP_EN_SHIFT, DA7213_NO_INVERT, NULL, 0), + SND_SOC_DAPM_PGA("Mixin Right PGA", DA7213_MIXIN_R_CTRL, + DA7213_AMP_EN_SHIFT, DA7213_NO_INVERT, NULL, 0), + + /* Mic Biases */ + SND_SOC_DAPM_SUPPLY("Mic Bias 1", DA7213_MICBIAS_CTRL, + DA7213_MICBIAS1_EN_SHIFT, DA7213_NO_INVERT, + NULL, 0), + SND_SOC_DAPM_SUPPLY("Mic Bias 2", DA7213_MICBIAS_CTRL, + DA7213_MICBIAS2_EN_SHIFT, DA7213_NO_INVERT, + NULL, 0), + + /* Input Mixers */ + SND_SOC_DAPM_MIXER("Mixin Left", SND_SOC_NOPM, 0, 0, + &da7213_dapm_mixinl_controls[0], + ARRAY_SIZE(da7213_dapm_mixinl_controls)), + SND_SOC_DAPM_MIXER("Mixin Right", SND_SOC_NOPM, 0, 0, + &da7213_dapm_mixinr_controls[0], + ARRAY_SIZE(da7213_dapm_mixinr_controls)), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC Left", NULL, DA7213_ADC_L_CTRL, + DA7213_ADC_EN_SHIFT, DA7213_NO_INVERT), + SND_SOC_DAPM_ADC("ADC Right", NULL, DA7213_ADC_R_CTRL, + DA7213_ADC_EN_SHIFT, DA7213_NO_INVERT), + + /* DAI */ + SND_SOC_DAPM_MUX("DAI Left Source MUX", SND_SOC_NOPM, 0, 0, + &da7213_dai_l_src_mux), + SND_SOC_DAPM_MUX("DAI Right Source MUX", SND_SOC_NOPM, 0, 0, + &da7213_dai_r_src_mux), + SND_SOC_DAPM_AIF_OUT("DAIOUTL", "Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("DAIOUTR", "Capture", 1, SND_SOC_NOPM, 0, 0), + + /* + * Output + */ + + /* DAI */ + SND_SOC_DAPM_AIF_IN("DAIINL", "Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("DAIINR", "Playback", 1, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_MUX("DAC Left Source MUX", SND_SOC_NOPM, 0, 0, + &da7213_dac_l_src_mux), + SND_SOC_DAPM_MUX("DAC Right Source MUX", SND_SOC_NOPM, 0, 0, + &da7213_dac_r_src_mux), + + /* DACs */ + SND_SOC_DAPM_DAC("DAC Left", NULL, DA7213_DAC_L_CTRL, + DA7213_DAC_EN_SHIFT, DA7213_NO_INVERT), + SND_SOC_DAPM_DAC("DAC Right", NULL, DA7213_DAC_R_CTRL, + DA7213_DAC_EN_SHIFT, DA7213_NO_INVERT), + + /* Output Mixers */ + SND_SOC_DAPM_MIXER("Mixout Left", SND_SOC_NOPM, 0, 0, + &da7213_dapm_mixoutl_controls[0], + ARRAY_SIZE(da7213_dapm_mixoutl_controls)), + SND_SOC_DAPM_MIXER("Mixout Right", SND_SOC_NOPM, 0, 0, + &da7213_dapm_mixoutr_controls[0], + ARRAY_SIZE(da7213_dapm_mixoutr_controls)), + + /* Output PGAs */ + SND_SOC_DAPM_PGA("Mixout Left PGA", DA7213_MIXOUT_L_CTRL, + DA7213_AMP_EN_SHIFT, DA7213_NO_INVERT, NULL, 0), + SND_SOC_DAPM_PGA("Mixout Right PGA", DA7213_MIXOUT_R_CTRL, + DA7213_AMP_EN_SHIFT, DA7213_NO_INVERT, NULL, 0), + SND_SOC_DAPM_PGA("Lineout PGA", DA7213_LINE_CTRL, DA7213_AMP_EN_SHIFT, + DA7213_NO_INVERT, NULL, 0), + SND_SOC_DAPM_PGA("Headphone Left PGA", DA7213_HP_L_CTRL, + DA7213_AMP_EN_SHIFT, DA7213_NO_INVERT, NULL, 0), + SND_SOC_DAPM_PGA("Headphone Right PGA", DA7213_HP_R_CTRL, + DA7213_AMP_EN_SHIFT, DA7213_NO_INVERT, NULL, 0), + + /* Charge Pump */ + SND_SOC_DAPM_SUPPLY("Charge Pump", DA7213_CP_CTRL, DA7213_CP_EN_SHIFT, + DA7213_NO_INVERT, NULL, 0), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), + SND_SOC_DAPM_OUTPUT("LINE"), +}; + + +/* + * DAPM audio route definition + */ + +static const struct snd_soc_dapm_route da7213_audio_map[] = { + /* Dest Connecting Widget source */ + + /* Input path */ + {"MIC1", NULL, "Mic Bias 1"}, + {"MIC2", NULL, "Mic Bias 2"}, + + {"Mic 1 Amp Source MUX", "Differential", "MIC1"}, + {"Mic 1 Amp Source MUX", "MIC_P", "MIC1"}, + {"Mic 1 Amp Source MUX", "MIC_N", "MIC1"}, + + {"Mic 2 Amp Source MUX", "Differential", "MIC2"}, + {"Mic 2 Amp Source MUX", "MIC_P", "MIC2"}, + {"Mic 2 Amp Source MUX", "MIC_N", "MIC2"}, + + {"Mic 1 PGA", NULL, "Mic 1 Amp Source MUX"}, + {"Mic 2 PGA", NULL, "Mic 2 Amp Source MUX"}, + + {"Aux Left PGA", NULL, "AUXL"}, + {"Aux Right PGA", NULL, "AUXR"}, + + {"Mixin Left", "Aux Left Switch", "Aux Left PGA"}, + {"Mixin Left", "Mic 1 Switch", "Mic 1 PGA"}, + {"Mixin Left", "Mic 2 Switch", "Mic 2 PGA"}, + {"Mixin Left", "Mixin Right Switch", "Mixin Right PGA"}, + + {"Mixin Right", "Aux Right Switch", "Aux Right PGA"}, + {"Mixin Right", "Mic 2 Switch", "Mic 2 PGA"}, + {"Mixin Right", "Mic 1 Switch", "Mic 1 PGA"}, + {"Mixin Right", "Mixin Left Switch", "Mixin Left PGA"}, + + {"Mixin Left PGA", NULL, "Mixin Left"}, + {"ADC Left", NULL, "Mixin Left PGA"}, + + {"Mixin Right PGA", NULL, "Mixin Right"}, + {"ADC Right", NULL, "Mixin Right PGA"}, + + {"DAI Left Source MUX", "ADC Left", "ADC Left"}, + {"DAI Left Source MUX", "ADC Right", "ADC Right"}, + {"DAI Left Source MUX", "DAI Input Left", "DAIINL"}, + {"DAI Left Source MUX", "DAI Input Right", "DAIINR"}, + + {"DAI Right Source MUX", "ADC Left", "ADC Left"}, + {"DAI Right Source MUX", "ADC Right", "ADC Right"}, + {"DAI Right Source MUX", "DAI Input Left", "DAIINL"}, + {"DAI Right Source MUX", "DAI Input Right", "DAIINR"}, + + {"DAIOUTL", NULL, "DAI Left Source MUX"}, + {"DAIOUTR", NULL, "DAI Right Source MUX"}, + + {"DAIOUTL", NULL, "DAI"}, + {"DAIOUTR", NULL, "DAI"}, + + /* Output path */ + {"DAIINL", NULL, "DAI"}, + {"DAIINR", NULL, "DAI"}, + + {"DAC Left Source MUX", "ADC Output Left", "ADC Left"}, + {"DAC Left Source MUX", "ADC Output Right", "ADC Right"}, + {"DAC Left Source MUX", "DAI Input Left", "DAIINL"}, + {"DAC Left Source MUX", "DAI Input Right", "DAIINR"}, + + {"DAC Right Source MUX", "ADC Output Left", "ADC Left"}, + {"DAC Right Source MUX", "ADC Output Right", "ADC Right"}, + {"DAC Right Source MUX", "DAI Input Left", "DAIINL"}, + {"DAC Right Source MUX", "DAI Input Right", "DAIINR"}, + + {"DAC Left", NULL, "DAC Left Source MUX"}, + {"DAC Right", NULL, "DAC Right Source MUX"}, + + {"Mixout Left", "Aux Left Switch", "Aux Left PGA"}, + {"Mixout Left", "Mixin Left Switch", "Mixin Left PGA"}, + {"Mixout Left", "Mixin Right Switch", "Mixin Right PGA"}, + {"Mixout Left", "DAC Left Switch", "DAC Left"}, + {"Mixout Left", "Aux Left Invert Switch", "Aux Left PGA"}, + {"Mixout Left", "Mixin Left Invert Switch", "Mixin Left PGA"}, + {"Mixout Left", "Mixin Right Invert Switch", "Mixin Right PGA"}, + + {"Mixout Right", "Aux Right Switch", "Aux Right PGA"}, + {"Mixout Right", "Mixin Right Switch", "Mixin Right PGA"}, + {"Mixout Right", "Mixin Left Switch", "Mixin Left PGA"}, + {"Mixout Right", "DAC Right Switch", "DAC Right"}, + {"Mixout Right", "Aux Right Invert Switch", "Aux Right PGA"}, + {"Mixout Right", "Mixin Right Invert Switch", "Mixin Right PGA"}, + {"Mixout Right", "Mixin Left Invert Switch", "Mixin Left PGA"}, + + {"Mixout Left PGA", NULL, "Mixout Left"}, + {"Mixout Right PGA", NULL, "Mixout Right"}, + + {"Headphone Left PGA", NULL, "Mixout Left PGA"}, + {"Headphone Left PGA", NULL, "Charge Pump"}, + {"HPL", NULL, "Headphone Left PGA"}, + + {"Headphone Right PGA", NULL, "Mixout Right PGA"}, + {"Headphone Right PGA", NULL, "Charge Pump"}, + {"HPR", NULL, "Headphone Right PGA"}, + + {"Lineout PGA", NULL, "Mixout Right PGA"}, + {"LINE", NULL, "Lineout PGA"}, +}; + +static struct reg_default da7213_reg_defaults[] = { + { DA7213_DIG_ROUTING_DAI, 0x10 }, + { DA7213_SR, 0x0A }, + { DA7213_REFERENCES, 0x80 }, + { DA7213_PLL_FRAC_TOP, 0x00 }, + { DA7213_PLL_FRAC_BOT, 0x00 }, + { DA7213_PLL_INTEGER, 0x20 }, + { DA7213_PLL_CTRL, 0x0C }, + { DA7213_DAI_CLK_MODE, 0x01 }, + { DA7213_DAI_CTRL, 0x08 }, + { DA7213_DIG_ROUTING_DAC, 0x32 }, + { DA7213_AUX_L_GAIN, 0x35 }, + { DA7213_AUX_R_GAIN, 0x35 }, + { DA7213_MIXIN_L_SELECT, 0x00 }, + { DA7213_MIXIN_R_SELECT, 0x00 }, + { DA7213_MIXIN_L_GAIN, 0x03 }, + { DA7213_MIXIN_R_GAIN, 0x03 }, + { DA7213_ADC_L_GAIN, 0x6F }, + { DA7213_ADC_R_GAIN, 0x6F }, + { DA7213_ADC_FILTERS1, 0x80 }, + { DA7213_MIC_1_GAIN, 0x01 }, + { DA7213_MIC_2_GAIN, 0x01 }, + { DA7213_DAC_FILTERS5, 0x00 }, + { DA7213_DAC_FILTERS2, 0x88 }, + { DA7213_DAC_FILTERS3, 0x88 }, + { DA7213_DAC_FILTERS4, 0x08 }, + { DA7213_DAC_FILTERS1, 0x80 }, + { DA7213_DAC_L_GAIN, 0x6F }, + { DA7213_DAC_R_GAIN, 0x6F }, + { DA7213_CP_CTRL, 0x61 }, + { DA7213_HP_L_GAIN, 0x39 }, + { DA7213_HP_R_GAIN, 0x39 }, + { DA7213_LINE_GAIN, 0x30 }, + { DA7213_MIXOUT_L_SELECT, 0x00 }, + { DA7213_MIXOUT_R_SELECT, 0x00 }, + { DA7213_SYSTEM_MODES_INPUT, 0x00 }, + { DA7213_SYSTEM_MODES_OUTPUT, 0x00 }, + { DA7213_AUX_L_CTRL, 0x44 }, + { DA7213_AUX_R_CTRL, 0x44 }, + { DA7213_MICBIAS_CTRL, 0x11 }, + { DA7213_MIC_1_CTRL, 0x40 }, + { DA7213_MIC_2_CTRL, 0x40 }, + { DA7213_MIXIN_L_CTRL, 0x40 }, + { DA7213_MIXIN_R_CTRL, 0x40 }, + { DA7213_ADC_L_CTRL, 0x40 }, + { DA7213_ADC_R_CTRL, 0x40 }, + { DA7213_DAC_L_CTRL, 0x48 }, + { DA7213_DAC_R_CTRL, 0x40 }, + { DA7213_HP_L_CTRL, 0x41 }, + { DA7213_HP_R_CTRL, 0x40 }, + { DA7213_LINE_CTRL, 0x40 }, + { DA7213_MIXOUT_L_CTRL, 0x10 }, + { DA7213_MIXOUT_R_CTRL, 0x10 }, + { DA7213_LDO_CTRL, 0x00 }, + { DA7213_IO_CTRL, 0x00 }, + { DA7213_GAIN_RAMP_CTRL, 0x00}, + { DA7213_MIC_CONFIG, 0x00 }, + { DA7213_PC_COUNT, 0x00 }, + { DA7213_CP_VOL_THRESHOLD1, 0x32 }, + { DA7213_CP_DELAY, 0x95 }, + { DA7213_CP_DETECTOR, 0x00 }, + { DA7213_DAI_OFFSET, 0x00 }, + { DA7213_DIG_CTRL, 0x00 }, + { DA7213_ALC_CTRL2, 0x00 }, + { DA7213_ALC_CTRL3, 0x00 }, + { DA7213_ALC_NOISE, 0x3F }, + { DA7213_ALC_TARGET_MIN, 0x3F }, + { DA7213_ALC_TARGET_MAX, 0x00 }, + { DA7213_ALC_GAIN_LIMITS, 0xFF }, + { DA7213_ALC_ANA_GAIN_LIMITS, 0x71 }, + { DA7213_ALC_ANTICLIP_CTRL, 0x00 }, + { DA7213_ALC_ANTICLIP_LEVEL, 0x00 }, + { DA7213_ALC_OFFSET_MAN_M_L, 0x00 }, + { DA7213_ALC_OFFSET_MAN_U_L, 0x00 }, + { DA7213_ALC_OFFSET_MAN_M_R, 0x00 }, + { DA7213_ALC_OFFSET_MAN_U_R, 0x00 }, + { DA7213_ALC_CIC_OP_LVL_CTRL, 0x00 }, + { DA7213_DAC_NG_SETUP_TIME, 0x00 }, + { DA7213_DAC_NG_OFF_THRESHOLD, 0x00 }, + { DA7213_DAC_NG_ON_THRESHOLD, 0x00 }, + { DA7213_DAC_NG_CTRL, 0x00 }, +}; + +static bool da7213_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case DA7213_STATUS1: + case DA7213_PLL_STATUS: + case DA7213_AUX_L_GAIN_STATUS: + case DA7213_AUX_R_GAIN_STATUS: + case DA7213_MIC_1_GAIN_STATUS: + case DA7213_MIC_2_GAIN_STATUS: + case DA7213_MIXIN_L_GAIN_STATUS: + case DA7213_MIXIN_R_GAIN_STATUS: + case DA7213_ADC_L_GAIN_STATUS: + case DA7213_ADC_R_GAIN_STATUS: + case DA7213_DAC_L_GAIN_STATUS: + case DA7213_DAC_R_GAIN_STATUS: + case DA7213_HP_L_GAIN_STATUS: + case DA7213_HP_R_GAIN_STATUS: + case DA7213_LINE_GAIN_STATUS: + case DA7213_ALC_CTRL1: + case DA7213_ALC_OFFSET_AUTO_M_L: + case DA7213_ALC_OFFSET_AUTO_U_L: + case DA7213_ALC_OFFSET_AUTO_M_R: + case DA7213_ALC_OFFSET_AUTO_U_R: + case DA7213_ALC_CIC_OP_LVL_DATA: + return 1; + default: + return 0; + } +} + +static int da7213_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + u8 dai_ctrl = 0; + u8 fs; + + /* Set DAI format */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + dai_ctrl |= DA7213_DAI_WORD_LENGTH_S16_LE; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + dai_ctrl |= DA7213_DAI_WORD_LENGTH_S20_LE; + break; + case SNDRV_PCM_FORMAT_S24_LE: + dai_ctrl |= DA7213_DAI_WORD_LENGTH_S24_LE; + break; + case SNDRV_PCM_FORMAT_S32_LE: + dai_ctrl |= DA7213_DAI_WORD_LENGTH_S32_LE; + break; + default: + return -EINVAL; + } + + /* Set sampling rate */ + switch (params_rate(params)) { + case 8000: + fs = DA7213_SR_8000; + break; + case 11025: + fs = DA7213_SR_11025; + break; + case 12000: + fs = DA7213_SR_12000; + break; + case 16000: + fs = DA7213_SR_16000; + break; + case 22050: + fs = DA7213_SR_22050; + break; + case 32000: + fs = DA7213_SR_32000; + break; + case 44100: + fs = DA7213_SR_44100; + break; + case 48000: + fs = DA7213_SR_48000; + break; + case 88200: + fs = DA7213_SR_88200; + break; + case 96000: + fs = DA7213_SR_96000; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, DA7213_DAI_CTRL, DA7213_DAI_WORD_LENGTH_MASK, + dai_ctrl); + snd_soc_write(codec, DA7213_SR, fs); + + return 0; +} + +static int da7213_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da7213_priv *da7213 = snd_soc_codec_get_drvdata(codec); + u8 dai_clk_mode = 0, dai_ctrl = 0; + + /* Set master/slave mode */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + dai_clk_mode |= DA7213_DAI_CLK_EN_MASTER_MODE; + da7213->master = true; + break; + case SND_SOC_DAIFMT_CBS_CFS: + dai_clk_mode |= DA7213_DAI_CLK_EN_SLAVE_MODE; + da7213->master = false; + break; + default: + return -EINVAL; + } + + /* Set clock normal/inverted */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + dai_clk_mode |= DA7213_DAI_WCLK_POL_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + dai_clk_mode |= DA7213_DAI_CLK_POL_INV; + break; + case SND_SOC_DAIFMT_IB_IF: + dai_clk_mode |= DA7213_DAI_WCLK_POL_INV | DA7213_DAI_CLK_POL_INV; + break; + default: + return -EINVAL; + } + + /* Only I2S is supported */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + dai_ctrl |= DA7213_DAI_FORMAT_I2S_MODE; + break; + case SND_SOC_DAIFMT_LEFT_J: + dai_ctrl |= DA7213_DAI_FORMAT_LEFT_J; + break; + case SND_SOC_DAIFMT_RIGHT_J: + dai_ctrl |= DA7213_DAI_FORMAT_RIGHT_J; + break; + default: + return -EINVAL; + } + + /* By default only 32 BCLK per WCLK is supported */ + dai_clk_mode |= DA7213_DAI_BCLKS_PER_WCLK_32; + + snd_soc_write(codec, DA7213_DAI_CLK_MODE, dai_clk_mode); + snd_soc_update_bits(codec, DA7213_DAI_CTRL, DA7213_DAI_FORMAT_MASK, + dai_ctrl); + + return 0; +} + +static int da7213_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + + if (mute) { + snd_soc_update_bits(codec, DA7213_DAC_L_CTRL, + DA7213_MUTE_EN, DA7213_MUTE_EN); + snd_soc_update_bits(codec, DA7213_DAC_R_CTRL, + DA7213_MUTE_EN, DA7213_MUTE_EN); + } else { + snd_soc_update_bits(codec, DA7213_DAC_L_CTRL, + DA7213_MUTE_EN, 0); + snd_soc_update_bits(codec, DA7213_DAC_R_CTRL, + DA7213_MUTE_EN, 0); + } + + return 0; +} + +#define DA7213_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static int da7213_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; + struct da7213_priv *da7213 = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case DA7213_CLKSRC_MCLK: + if ((freq == 32768) || + ((freq >= 5000000) && (freq <= 54000000))) { + da7213->mclk_rate = freq; + return 0; + } else { + dev_err(codec_dai->dev, "Unsupported MCLK value %d\n", + freq); + return -EINVAL; + } + break; + default: + dev_err(codec_dai->dev, "Unknown clock source %d\n", clk_id); + return -EINVAL; + } +} + +/* Supported PLL input frequencies are 5MHz - 54MHz. */ +static int da7213_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int fref, unsigned int fout) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da7213_priv *da7213 = snd_soc_codec_get_drvdata(codec); + + u8 pll_ctrl, indiv_bits, indiv; + u8 pll_frac_top, pll_frac_bot, pll_integer; + u32 freq_ref; + u64 frac_div; + + /* Reset PLL configuration */ + snd_soc_write(codec, DA7213_PLL_CTRL, 0); + + pll_ctrl = 0; + + /* Workout input divider based on MCLK rate */ + if ((da7213->mclk_rate == 32768) && (source == DA7213_SYSCLK_PLL)) { + /* 32KHz PLL Mode */ + indiv_bits = DA7213_PLL_INDIV_10_20_MHZ; + indiv = DA7213_PLL_INDIV_10_20_MHZ_VAL; + freq_ref = 3750000; + pll_ctrl |= DA7213_PLL_32K_MODE; + } else { + /* 5 - 54MHz MCLK */ + if (da7213->mclk_rate < 5000000) { + goto pll_err; + } else if (da7213->mclk_rate <= 10000000) { + indiv_bits = DA7213_PLL_INDIV_5_10_MHZ; + indiv = DA7213_PLL_INDIV_5_10_MHZ_VAL; + } else if (da7213->mclk_rate <= 20000000) { + indiv_bits = DA7213_PLL_INDIV_10_20_MHZ; + indiv = DA7213_PLL_INDIV_10_20_MHZ_VAL; + } else if (da7213->mclk_rate <= 40000000) { + indiv_bits = DA7213_PLL_INDIV_20_40_MHZ; + indiv = DA7213_PLL_INDIV_20_40_MHZ_VAL; + } else if (da7213->mclk_rate <= 54000000) { + indiv_bits = DA7213_PLL_INDIV_40_54_MHZ; + indiv = DA7213_PLL_INDIV_40_54_MHZ_VAL; + } else { + goto pll_err; + } + freq_ref = (da7213->mclk_rate / indiv); + } + + pll_ctrl |= indiv_bits; + + /* PLL Bypass mode */ + if (source == DA7213_SYSCLK_MCLK) { + snd_soc_write(codec, DA7213_PLL_CTRL, pll_ctrl); + return 0; + } + + /* + * If Codec is slave and SRM enabled, + * freq_out is (98304000 + 90316800)/2 = 94310400 + */ + if (!da7213->master && da7213->srm_en) { + fout = DA7213_PLL_FREQ_OUT_94310400; + pll_ctrl |= DA7213_PLL_SRM_EN; + } + + /* Enable MCLK squarer if required */ + if (da7213->mclk_squarer_en) + pll_ctrl |= DA7213_PLL_MCLK_SQR_EN; + + /* Calculate dividers for PLL */ + pll_integer = fout / freq_ref; + frac_div = (u64)(fout % freq_ref) * 8192ULL; + do_div(frac_div, freq_ref); + pll_frac_top = (frac_div >> DA7213_BYTE_SHIFT) & DA7213_BYTE_MASK; + pll_frac_bot = (frac_div) & DA7213_BYTE_MASK; + + /* Write PLL dividers */ + snd_soc_write(codec, DA7213_PLL_FRAC_TOP, pll_frac_top); + snd_soc_write(codec, DA7213_PLL_FRAC_BOT, pll_frac_bot); + snd_soc_write(codec, DA7213_PLL_INTEGER, pll_integer); + + /* Enable PLL */ + pll_ctrl |= DA7213_PLL_EN; + snd_soc_write(codec, DA7213_PLL_CTRL, pll_ctrl); + + return 0; + +pll_err: + dev_err(codec_dai->dev, "Unsupported PLL input frequency %d\n", + da7213->mclk_rate); + return -EINVAL; +} + +/* DAI operations */ +static const struct snd_soc_dai_ops da7213_dai_ops = { + .hw_params = da7213_hw_params, + .set_fmt = da7213_set_dai_fmt, + .set_sysclk = da7213_set_dai_sysclk, + .set_pll = da7213_set_dai_pll, + .digital_mute = da7213_mute, +}; + +static struct snd_soc_dai_driver da7213_dai = { + .name = "da7213-hifi", + /* Playback Capabilities */ + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = DA7213_FORMATS, + }, + /* Capture Capabilities */ + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = DA7213_FORMATS, + }, + .ops = &da7213_dai_ops, + .symmetric_rates = 1, +}; + +static int da7213_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + /* Enable VMID reference & master bias */ + snd_soc_update_bits(codec, DA7213_REFERENCES, + DA7213_VMID_EN | DA7213_BIAS_EN, + DA7213_VMID_EN | DA7213_BIAS_EN); + } + break; + case SND_SOC_BIAS_OFF: + /* Disable VMID reference & master bias */ + snd_soc_update_bits(codec, DA7213_REFERENCES, + DA7213_VMID_EN | DA7213_BIAS_EN, 0); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static int da7213_probe(struct snd_soc_codec *codec) +{ + int ret; + struct da7213_priv *da7213 = snd_soc_codec_get_drvdata(codec); + struct da7213_platform_data *pdata = da7213->pdata; + + codec->control_data = da7213->regmap; + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + /* Default to using ALC auto offset calibration mode. */ + snd_soc_update_bits(codec, DA7213_ALC_CTRL1, + DA7213_ALC_CALIB_MODE_MAN, 0); + da7213->alc_calib_auto = true; + + /* Default to using SRM for slave mode */ + da7213->srm_en = true; + + /* Enable all Gain Ramps */ + snd_soc_update_bits(codec, DA7213_AUX_L_CTRL, + DA7213_GAIN_RAMP_EN, DA7213_GAIN_RAMP_EN); + snd_soc_update_bits(codec, DA7213_AUX_R_CTRL, + DA7213_GAIN_RAMP_EN, DA7213_GAIN_RAMP_EN); + snd_soc_update_bits(codec, DA7213_MIXIN_L_CTRL, + DA7213_GAIN_RAMP_EN, DA7213_GAIN_RAMP_EN); + snd_soc_update_bits(codec, DA7213_MIXIN_R_CTRL, + DA7213_GAIN_RAMP_EN, DA7213_GAIN_RAMP_EN); + snd_soc_update_bits(codec, DA7213_ADC_L_CTRL, + DA7213_GAIN_RAMP_EN, DA7213_GAIN_RAMP_EN); + snd_soc_update_bits(codec, DA7213_ADC_R_CTRL, + DA7213_GAIN_RAMP_EN, DA7213_GAIN_RAMP_EN); + snd_soc_update_bits(codec, DA7213_DAC_L_CTRL, + DA7213_GAIN_RAMP_EN, DA7213_GAIN_RAMP_EN); + snd_soc_update_bits(codec, DA7213_DAC_R_CTRL, + DA7213_GAIN_RAMP_EN, DA7213_GAIN_RAMP_EN); + snd_soc_update_bits(codec, DA7213_HP_L_CTRL, + DA7213_GAIN_RAMP_EN, DA7213_GAIN_RAMP_EN); + snd_soc_update_bits(codec, DA7213_HP_R_CTRL, + DA7213_GAIN_RAMP_EN, DA7213_GAIN_RAMP_EN); + snd_soc_update_bits(codec, DA7213_LINE_CTRL, + DA7213_GAIN_RAMP_EN, DA7213_GAIN_RAMP_EN); + + /* + * There are two separate control bits for input and output mixers as + * well as headphone and line outs. + * One to enable corresponding amplifier and other to enable its + * output. As amplifier bits are related to power control, they are + * being managed by DAPM while other (non power related) bits are + * enabled here + */ + snd_soc_update_bits(codec, DA7213_MIXIN_L_CTRL, + DA7213_MIXIN_MIX_EN, DA7213_MIXIN_MIX_EN); + snd_soc_update_bits(codec, DA7213_MIXIN_R_CTRL, + DA7213_MIXIN_MIX_EN, DA7213_MIXIN_MIX_EN); + + snd_soc_update_bits(codec, DA7213_MIXOUT_L_CTRL, + DA7213_MIXOUT_MIX_EN, DA7213_MIXOUT_MIX_EN); + snd_soc_update_bits(codec, DA7213_MIXOUT_R_CTRL, + DA7213_MIXOUT_MIX_EN, DA7213_MIXOUT_MIX_EN); + + snd_soc_update_bits(codec, DA7213_HP_L_CTRL, + DA7213_HP_AMP_OE, DA7213_HP_AMP_OE); + snd_soc_update_bits(codec, DA7213_HP_R_CTRL, + DA7213_HP_AMP_OE, DA7213_HP_AMP_OE); + + snd_soc_update_bits(codec, DA7213_LINE_CTRL, + DA7213_LINE_AMP_OE, DA7213_LINE_AMP_OE); + + /* Set platform data values */ + if (da7213->pdata) { + u8 micbias_lvl = 0, dmic_cfg = 0; + + /* Set Mic Bias voltages */ + switch (pdata->micbias1_lvl) { + case DA7213_MICBIAS_1_6V: + case DA7213_MICBIAS_2_2V: + case DA7213_MICBIAS_2_5V: + case DA7213_MICBIAS_3_0V: + micbias_lvl |= (pdata->micbias1_lvl << + DA7213_MICBIAS1_LEVEL_SHIFT); + break; + } + switch (pdata->micbias2_lvl) { + case DA7213_MICBIAS_1_6V: + case DA7213_MICBIAS_2_2V: + case DA7213_MICBIAS_2_5V: + case DA7213_MICBIAS_3_0V: + micbias_lvl |= (pdata->micbias2_lvl << + DA7213_MICBIAS2_LEVEL_SHIFT); + break; + } + snd_soc_update_bits(codec, DA7213_MICBIAS_CTRL, + DA7213_MICBIAS1_LEVEL_MASK | + DA7213_MICBIAS2_LEVEL_MASK, micbias_lvl); + + /* Set DMIC configuration */ + switch (pdata->dmic_data_sel) { + case DA7213_DMIC_DATA_LFALL_RRISE: + case DA7213_DMIC_DATA_LRISE_RFALL: + dmic_cfg |= (pdata->dmic_data_sel << + DA7213_DMIC_DATA_SEL_SHIFT); + break; + } + switch (pdata->dmic_data_sel) { + case DA7213_DMIC_SAMPLE_ON_CLKEDGE: + case DA7213_DMIC_SAMPLE_BETWEEN_CLKEDGE: + dmic_cfg |= (pdata->dmic_data_sel << + DA7213_DMIC_SAMPLEPHASE_SHIFT); + break; + } + switch (pdata->dmic_data_sel) { + case DA7213_DMIC_CLK_3_0MHZ: + case DA7213_DMIC_CLK_1_5MHZ: + dmic_cfg |= (pdata->dmic_data_sel << + DA7213_DMIC_CLK_RATE_SHIFT); + break; + } + snd_soc_update_bits(codec, DA7213_MIC_CONFIG, + DA7213_DMIC_DATA_SEL_MASK | + DA7213_DMIC_SAMPLEPHASE_MASK | + DA7213_DMIC_CLK_RATE_MASK, dmic_cfg); + + /* Set MCLK squaring */ + da7213->mclk_squarer_en = pdata->mclk_squaring; + } + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_da7213 = { + .probe = da7213_probe, + .set_bias_level = da7213_set_bias_level, + + .controls = da7213_snd_controls, + .num_controls = ARRAY_SIZE(da7213_snd_controls), + + .dapm_widgets = da7213_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(da7213_dapm_widgets), + .dapm_routes = da7213_audio_map, + .num_dapm_routes = ARRAY_SIZE(da7213_audio_map), +}; + +static const struct regmap_config da7213_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .reg_defaults = da7213_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(da7213_reg_defaults), + .volatile_reg = da7213_volatile_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int da7213_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct da7213_priv *da7213; + struct da7213_platform_data *pdata = dev_get_platdata(&i2c->dev); + int ret; + + da7213 = devm_kzalloc(&i2c->dev, sizeof(struct da7213_priv), + GFP_KERNEL); + if (!da7213) + return -ENOMEM; + + if (pdata) + da7213->pdata = pdata; + + i2c_set_clientdata(i2c, da7213); + + da7213->regmap = devm_regmap_init_i2c(i2c, &da7213_regmap_config); + if (IS_ERR(da7213->regmap)) { + ret = PTR_ERR(da7213->regmap); + dev_err(&i2c->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_da7213, &da7213_dai, 1); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to register da7213 codec: %d\n", + ret); + } + return ret; +} + +static int da7213_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id da7213_i2c_id[] = { + { "da7213", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, da7213_i2c_id); + +/* I2C codec control layer */ +static struct i2c_driver da7213_i2c_driver = { + .driver = { + .name = "da7213", + .owner = THIS_MODULE, + }, + .probe = da7213_i2c_probe, + .remove = da7213_remove, + .id_table = da7213_i2c_id, +}; + +module_i2c_driver(da7213_i2c_driver); + +MODULE_DESCRIPTION("ASoC DA7213 Codec driver"); +MODULE_AUTHOR("Adam Thomson "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/da7213.h b/sound/soc/codecs/da7213.h new file mode 100644 index 00000000000..9cb9ddd0128 --- /dev/null +++ b/sound/soc/codecs/da7213.h @@ -0,0 +1,523 @@ +/* + * da7213.h - DA7213 ASoC Codec Driver + * + * Copyright (c) 2013 Dialog Semiconductor + * + * Author: Adam Thomson + * + * 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 _DA7213_H +#define _DA7213_H + +#include +#include + +/* + * Registers + */ + +/* Status Registers */ +#define DA7213_STATUS1 0x02 +#define DA7213_PLL_STATUS 0x03 +#define DA7213_AUX_L_GAIN_STATUS 0x04 +#define DA7213_AUX_R_GAIN_STATUS 0x05 +#define DA7213_MIC_1_GAIN_STATUS 0x06 +#define DA7213_MIC_2_GAIN_STATUS 0x07 +#define DA7213_MIXIN_L_GAIN_STATUS 0x08 +#define DA7213_MIXIN_R_GAIN_STATUS 0x09 +#define DA7213_ADC_L_GAIN_STATUS 0x0A +#define DA7213_ADC_R_GAIN_STATUS 0x0B +#define DA7213_DAC_L_GAIN_STATUS 0x0C +#define DA7213_DAC_R_GAIN_STATUS 0x0D +#define DA7213_HP_L_GAIN_STATUS 0x0E +#define DA7213_HP_R_GAIN_STATUS 0x0F +#define DA7213_LINE_GAIN_STATUS 0x10 + +/* System Initialisation Registers */ +#define DA7213_DIG_ROUTING_DAI 0x21 +#define DA7213_SR 0x22 +#define DA7213_REFERENCES 0x23 +#define DA7213_PLL_FRAC_TOP 0x24 +#define DA7213_PLL_FRAC_BOT 0x25 +#define DA7213_PLL_INTEGER 0x26 +#define DA7213_PLL_CTRL 0x27 +#define DA7213_DAI_CLK_MODE 0x28 +#define DA7213_DAI_CTRL 0x29 +#define DA7213_DIG_ROUTING_DAC 0x2A +#define DA7213_ALC_CTRL1 0x2B + +/* Input - Gain, Select and Filter Registers */ +#define DA7213_AUX_L_GAIN 0x30 +#define DA7213_AUX_R_GAIN 0x31 +#define DA7213_MIXIN_L_SELECT 0x32 +#define DA7213_MIXIN_R_SELECT 0x33 +#define DA7213_MIXIN_L_GAIN 0x34 +#define DA7213_MIXIN_R_GAIN 0x35 +#define DA7213_ADC_L_GAIN 0x36 +#define DA7213_ADC_R_GAIN 0x37 +#define DA7213_ADC_FILTERS1 0x38 +#define DA7213_MIC_1_GAIN 0x39 +#define DA7213_MIC_2_GAIN 0x3A + +/* Output - Gain, Select and Filter Registers */ +#define DA7213_DAC_FILTERS5 0x40 +#define DA7213_DAC_FILTERS2 0x41 +#define DA7213_DAC_FILTERS3 0x42 +#define DA7213_DAC_FILTERS4 0x43 +#define DA7213_DAC_FILTERS1 0x44 +#define DA7213_DAC_L_GAIN 0x45 +#define DA7213_DAC_R_GAIN 0x46 +#define DA7213_CP_CTRL 0x47 +#define DA7213_HP_L_GAIN 0x48 +#define DA7213_HP_R_GAIN 0x49 +#define DA7213_LINE_GAIN 0x4A +#define DA7213_MIXOUT_L_SELECT 0x4B +#define DA7213_MIXOUT_R_SELECT 0x4C + +/* System Controller Registers */ +#define DA7213_SYSTEM_MODES_INPUT 0x50 +#define DA7213_SYSTEM_MODES_OUTPUT 0x51 + +/* Control Registers */ +#define DA7213_AUX_L_CTRL 0x60 +#define DA7213_AUX_R_CTRL 0x61 +#define DA7213_MICBIAS_CTRL 0x62 +#define DA7213_MIC_1_CTRL 0x63 +#define DA7213_MIC_2_CTRL 0x64 +#define DA7213_MIXIN_L_CTRL 0x65 +#define DA7213_MIXIN_R_CTRL 0x66 +#define DA7213_ADC_L_CTRL 0x67 +#define DA7213_ADC_R_CTRL 0x68 +#define DA7213_DAC_L_CTRL 0x69 +#define DA7213_DAC_R_CTRL 0x6A +#define DA7213_HP_L_CTRL 0x6B +#define DA7213_HP_R_CTRL 0x6C +#define DA7213_LINE_CTRL 0x6D +#define DA7213_MIXOUT_L_CTRL 0x6E +#define DA7213_MIXOUT_R_CTRL 0x6F + +/* Configuration Registers */ +#define DA7213_LDO_CTRL 0x90 +#define DA7213_IO_CTRL 0x91 +#define DA7213_GAIN_RAMP_CTRL 0x92 +#define DA7213_MIC_CONFIG 0x93 +#define DA7213_PC_COUNT 0x94 +#define DA7213_CP_VOL_THRESHOLD1 0x95 +#define DA7213_CP_DELAY 0x96 +#define DA7213_CP_DETECTOR 0x97 +#define DA7213_DAI_OFFSET 0x98 +#define DA7213_DIG_CTRL 0x99 +#define DA7213_ALC_CTRL2 0x9A +#define DA7213_ALC_CTRL3 0x9B +#define DA7213_ALC_NOISE 0x9C +#define DA7213_ALC_TARGET_MIN 0x9D +#define DA7213_ALC_TARGET_MAX 0x9E +#define DA7213_ALC_GAIN_LIMITS 0x9F +#define DA7213_ALC_ANA_GAIN_LIMITS 0xA0 +#define DA7213_ALC_ANTICLIP_CTRL 0xA1 +#define DA7213_ALC_ANTICLIP_LEVEL 0xA2 + +#define DA7213_ALC_OFFSET_AUTO_M_L 0xA3 +#define DA7213_ALC_OFFSET_AUTO_U_L 0xA4 +#define DA7213_ALC_OFFSET_MAN_M_L 0xA6 +#define DA7213_ALC_OFFSET_MAN_U_L 0xA7 +#define DA7213_ALC_OFFSET_AUTO_M_R 0xA8 +#define DA7213_ALC_OFFSET_AUTO_U_R 0xA9 +#define DA7213_ALC_OFFSET_MAN_M_R 0xAB +#define DA7213_ALC_OFFSET_MAN_U_R 0xAC +#define DA7213_ALC_CIC_OP_LVL_CTRL 0xAD +#define DA7213_ALC_CIC_OP_LVL_DATA 0xAE +#define DA7213_DAC_NG_SETUP_TIME 0xAF +#define DA7213_DAC_NG_OFF_THRESHOLD 0xB0 +#define DA7213_DAC_NG_ON_THRESHOLD 0xB1 +#define DA7213_DAC_NG_CTRL 0xB2 + + +/* + * Bit fields + */ + +/* DA7213_SR = 0x22 */ +#define DA7213_SR_8000 (0x1 << 0) +#define DA7213_SR_11025 (0x2 << 0) +#define DA7213_SR_12000 (0x3 << 0) +#define DA7213_SR_16000 (0x5 << 0) +#define DA7213_SR_22050 (0x6 << 0) +#define DA7213_SR_24000 (0x7 << 0) +#define DA7213_SR_32000 (0x9 << 0) +#define DA7213_SR_44100 (0xA << 0) +#define DA7213_SR_48000 (0xB << 0) +#define DA7213_SR_88200 (0xE << 0) +#define DA7213_SR_96000 (0xF << 0) + +/* DA7213_REFERENCES = 0x23 */ +#define DA7213_BIAS_EN (0x1 << 3) +#define DA7213_VMID_EN (0x1 << 7) + +/* DA7213_PLL_CTRL = 0x27 */ +#define DA7213_PLL_INDIV_5_10_MHZ (0x0 << 2) +#define DA7213_PLL_INDIV_10_20_MHZ (0x1 << 2) +#define DA7213_PLL_INDIV_20_40_MHZ (0x2 << 2) +#define DA7213_PLL_INDIV_40_54_MHZ (0x3 << 2) +#define DA7213_PLL_INDIV_MASK (0x3 << 2) +#define DA7213_PLL_MCLK_SQR_EN (0x1 << 4) +#define DA7213_PLL_32K_MODE (0x1 << 5) +#define DA7213_PLL_SRM_EN (0x1 << 6) +#define DA7213_PLL_EN (0x1 << 7) + +/* DA7213_DAI_CLK_MODE = 0x28 */ +#define DA7213_DAI_BCLKS_PER_WCLK_32 (0x0 << 0) +#define DA7213_DAI_BCLKS_PER_WCLK_64 (0x1 << 0) +#define DA7213_DAI_BCLKS_PER_WCLK_128 (0x2 << 0) +#define DA7213_DAI_BCLKS_PER_WCLK_256 (0x3 << 0) +#define DA7213_DAI_BCLKS_PER_WCLK_MASK (0x3 << 0) +#define DA7213_DAI_CLK_POL_INV (0x1 << 2) +#define DA7213_DAI_WCLK_POL_INV (0x1 << 3) +#define DA7213_DAI_CLK_EN_SLAVE_MODE (0x0 << 7) +#define DA7213_DAI_CLK_EN_MASTER_MODE (0x1 << 7) +#define DA7213_DAI_CLK_EN_MASK (0x1 << 7) + +/* DA7213_DAI_CTRL = 0x29 */ +#define DA7213_DAI_FORMAT_I2S_MODE (0x0 << 0) +#define DA7213_DAI_FORMAT_LEFT_J (0x1 << 0) +#define DA7213_DAI_FORMAT_RIGHT_J (0x2 << 0) +#define DA7213_DAI_FORMAT_MASK (0x3 << 0) +#define DA7213_DAI_WORD_LENGTH_S16_LE (0x0 << 2) +#define DA7213_DAI_WORD_LENGTH_S20_LE (0x1 << 2) +#define DA7213_DAI_WORD_LENGTH_S24_LE (0x2 << 2) +#define DA7213_DAI_WORD_LENGTH_S32_LE (0x3 << 2) +#define DA7213_DAI_WORD_LENGTH_MASK (0x3 << 2) +#define DA7213_DAI_EN_SHIFT 7 + +/* DA7213_DIG_ROUTING_DAI = 0x21 */ +#define DA7213_DAI_L_SRC_SHIFT 0 +#define DA7213_DAI_R_SRC_SHIFT 4 +#define DA7213_DAI_SRC_MAX 4 + +/* DA7213_DIG_ROUTING_DAC = 0x2A */ +#define DA7213_DAC_L_SRC_SHIFT 0 +#define DA7213_DAC_L_MONO_SHIFT 3 +#define DA7213_DAC_R_SRC_SHIFT 4 +#define DA7213_DAC_R_MONO_SHIFT 7 +#define DA7213_DAC_SRC_MAX 4 +#define DA7213_DAC_MONO_MAX 0x1 + +/* DA7213_ALC_CTRL1 = 0x2B */ +#define DA7213_ALC_OFFSET_EN_SHIFT 0 +#define DA7213_ALC_OFFSET_EN_MAX 0x1 +#define DA7213_ALC_OFFSET_EN (0x1 << 0) +#define DA7213_ALC_SYNC_MODE (0x1 << 1) +#define DA7213_ALC_CALIB_MODE_MAN (0x1 << 2) +#define DA7213_ALC_L_EN_SHIFT 3 +#define DA7213_ALC_AUTO_CALIB_EN (0x1 << 4) +#define DA7213_ALC_CALIB_OVERFLOW (0x1 << 5) +#define DA7213_ALC_R_EN_SHIFT 7 +#define DA7213_ALC_EN_MAX 0x1 + +/* DA7213_AUX_L/R_GAIN = 0x30/0x31 */ +#define DA7213_AUX_AMP_GAIN_SHIFT 0 +#define DA7213_AUX_AMP_GAIN_MAX 0x3F + +/* DA7213_MIXIN_L/R_SELECT = 0x32/0x33 */ +#define DA7213_DMIC_EN_SHIFT 7 +#define DA7213_DMIC_EN_MAX 0x1 + +/* DA7213_MIXIN_L_SELECT = 0x32 */ +#define DA7213_MIXIN_L_MIX_SELECT_AUX_L_SHIFT 0 +#define DA7213_MIXIN_L_MIX_SELECT_MIC_1_SHIFT 1 +#define DA7213_MIXIN_L_MIX_SELECT_MIC_1 (0x1 << 1) +#define DA7213_MIXIN_L_MIX_SELECT_MIC_2_SHIFT 2 +#define DA7213_MIXIN_L_MIX_SELECT_MIC_2 (0x1 << 2) +#define DA7213_MIXIN_L_MIX_SELECT_MIXIN_R_SHIFT 3 +#define DA7213_MIXIN_L_MIX_SELECT_MAX 0x1 + +/* DA7213_MIXIN_R_SELECT = 0x33 */ +#define DA7213_MIXIN_R_MIX_SELECT_AUX_R_SHIFT 0 +#define DA7213_MIXIN_R_MIX_SELECT_MIC_2_SHIFT 1 +#define DA7213_MIXIN_R_MIX_SELECT_MIC_2 (0x1 << 1) +#define DA7213_MIXIN_R_MIX_SELECT_MIC_1_SHIFT 2 +#define DA7213_MIXIN_R_MIX_SELECT_MIC_1 (0x1 << 2) +#define DA7213_MIXIN_R_MIX_SELECT_MIXIN_L_SHIFT 3 +#define DA7213_MIXIN_R_MIX_SELECT_MAX 0x1 +#define DA7213_MIC_BIAS_OUTPUT_SELECT_2 (0x1 << 6) + +/* DA7213_MIXIN_L/R_GAIN = 0x34/0x35 */ +#define DA7213_MIXIN_AMP_GAIN_SHIFT 0 +#define DA7213_MIXIN_AMP_GAIN_MAX 0xF + +/* DA7213_ADC_L/R_GAIN = 0x36/0x37 */ +#define DA7213_ADC_AMP_GAIN_SHIFT 0 +#define DA7213_ADC_AMP_GAIN_MAX 0x7F + +/* DA7213_ADC/DAC_FILTERS1 = 0x38/0x44 */ +#define DA7213_VOICE_HPF_CORNER_SHIFT 0 +#define DA7213_VOICE_HPF_CORNER_MAX 8 +#define DA7213_VOICE_EN_SHIFT 3 +#define DA7213_VOICE_EN_MAX 0x1 +#define DA7213_AUDIO_HPF_CORNER_SHIFT 4 +#define DA7213_AUDIO_HPF_CORNER_MAX 4 +#define DA7213_HPF_EN_SHIFT 7 +#define DA7213_HPF_EN_MAX 0x1 + +/* DA7213_MIC_1/2_GAIN = 0x39/0x3A */ +#define DA7213_MIC_AMP_GAIN_SHIFT 0 +#define DA7213_MIC_AMP_GAIN_MAX 0x7 + +/* DA7213_DAC_FILTERS5 = 0x40 */ +#define DA7213_DAC_SOFTMUTE_EN_SHIFT 7 +#define DA7213_DAC_SOFTMUTE_EN_MAX 0x1 +#define DA7213_DAC_SOFTMUTE_RATE_SHIFT 4 +#define DA7213_DAC_SOFTMUTE_RATE_MAX 7 + +/* DA7213_DAC_FILTERS2/3/4 = 0x41/0x42/0x43 */ +#define DA7213_DAC_EQ_BAND_MAX 0xF + +/* DA7213_DAC_FILTERS2 = 0x41 */ +#define DA7213_DAC_EQ_BAND1_SHIFT 0 +#define DA7213_DAC_EQ_BAND2_SHIFT 4 + +/* DA7213_DAC_FILTERS2 = 0x42 */ +#define DA7213_DAC_EQ_BAND3_SHIFT 0 +#define DA7213_DAC_EQ_BAND4_SHIFT 4 + +/* DA7213_DAC_FILTERS4 = 0x43 */ +#define DA7213_DAC_EQ_BAND5_SHIFT 0 +#define DA7213_DAC_EQ_EN_SHIFT 7 +#define DA7213_DAC_EQ_EN_MAX 0x1 + +/* DA7213_DAC_L/R_GAIN = 0x45/0x46 */ +#define DA7213_DAC_AMP_GAIN_SHIFT 0 +#define DA7213_DAC_AMP_GAIN_MAX 0x7F + +/* DA7213_HP_L/R_GAIN = 0x45/0x46 */ +#define DA7213_HP_AMP_GAIN_SHIFT 0 +#define DA7213_HP_AMP_GAIN_MAX 0x3F + +/* DA7213_CP_CTRL = 0x47 */ +#define DA7213_CP_EN_SHIFT 7 + +/* DA7213_LINE_GAIN = 0x4A */ +#define DA7213_LINE_AMP_GAIN_SHIFT 0 +#define DA7213_LINE_AMP_GAIN_MAX 0x3F + +/* DA7213_MIXOUT_L_SELECT = 0x4B */ +#define DA7213_MIXOUT_L_MIX_SELECT_AUX_L_SHIFT 0 +#define DA7213_MIXOUT_L_MIX_SELECT_MIXIN_L_SHIFT 1 +#define DA7213_MIXOUT_L_MIX_SELECT_MIXIN_R_SHIFT 2 +#define DA7213_MIXOUT_L_MIX_SELECT_DAC_L_SHIFT 3 +#define DA7213_MIXOUT_L_MIX_SELECT_AUX_L_INVERTED_SHIFT 4 +#define DA7213_MIXOUT_L_MIX_SELECT_MIXIN_L_INVERTED_SHIFT 5 +#define DA7213_MIXOUT_L_MIX_SELECT_MIXIN_R_INVERTED_SHIFT 6 +#define DA7213_MIXOUT_L_MIX_SELECT_MAX 0x1 + +/* DA7213_MIXOUT_R_SELECT = 0x4C */ +#define DA7213_MIXOUT_R_MIX_SELECT_AUX_R_SHIFT 0 +#define DA7213_MIXOUT_R_MIX_SELECT_MIXIN_R_SHIFT 1 +#define DA7213_MIXOUT_R_MIX_SELECT_MIXIN_L_SHIFT 2 +#define DA7213_MIXOUT_R_MIX_SELECT_DAC_R_SHIFT 3 +#define DA7213_MIXOUT_R_MIX_SELECT_AUX_R_INVERTED_SHIFT 4 +#define DA7213_MIXOUT_R_MIX_SELECT_MIXIN_R_INVERTED_SHIFT 5 +#define DA7213_MIXOUT_R_MIX_SELECT_MIXIN_L_INVERTED_SHIFT 6 +#define DA7213_MIXOUT_R_MIX_SELECT_MAX 0x1 + +/* + * DA7213_AUX_L/R_CTRL = 0x60/0x61, + * DA7213_MIC_1/2_CTRL = 0x63/0x64, + * DA7213_MIXIN_L/R_CTRL = 0x65/0x66, + * DA7213_ADC_L/R_CTRL = 0x65/0x66, + * DA7213_DAC_L/R_CTRL = 0x69/0x6A, + * DA7213_HP_L/R_CTRL = 0x6B/0x6C, + * DA7213_LINE_CTRL = 0x6D + */ +#define DA7213_MUTE_EN_SHIFT 6 +#define DA7213_MUTE_EN_MAX 0x1 +#define DA7213_MUTE_EN (0x1 << 6) + +/* + * DA7213_AUX_L/R_CTRL = 0x60/0x61, + * DA7213_MIXIN_L/R_CTRL = 0x65/0x66, + * DA7213_ADC_L/R_CTRL = 0x65/0x66, + * DA7213_DAC_L/R_CTRL = 0x69/0x6A, + * DA7213_HP_L/R_CTRL = 0x6B/0x6C, + * DA7213_LINE_CTRL = 0x6D + */ +#define DA7213_GAIN_RAMP_EN_SHIFT 5 +#define DA7213_GAIN_RAMP_EN_MAX 0x1 +#define DA7213_GAIN_RAMP_EN (0x1 << 5) + +/* + * DA7213_AUX_L/R_CTRL = 0x60/0x61, + * DA7213_MIXIN_L/R_CTRL = 0x65/0x66, + * DA7213_HP_L/R_CTRL = 0x6B/0x6C, + * DA7213_LINE_CTRL = 0x6D + */ +#define DA7213_ZC_EN_SHIFT 4 +#define DA7213_ZC_EN_MAX 0x1 + +/* + * DA7213_AUX_L/R_CTRL = 0x60/0x61, + * DA7213_MIC_1/2_CTRL = 0x63/0x64, + * DA7213_MIXIN_L/R_CTRL = 0x65/0x66, + * DA7213_HP_L/R_CTRL = 0x6B/0x6C, + * DA7213_MIXOUT_L/R_CTRL = 0x6E/0x6F, + * DA7213_LINE_CTRL = 0x6D + */ +#define DA7213_AMP_EN_SHIFT 7 + +/* DA7213_MIC_1/2_CTRL = 0x63/0x64 */ +#define DA7213_MIC_AMP_IN_SEL_SHIFT 2 +#define DA7213_MIC_AMP_IN_SEL_MAX 3 + +/* DA7213_MICBIAS_CTRL = 0x62 */ +#define DA7213_MICBIAS1_LEVEL_SHIFT 0 +#define DA7213_MICBIAS1_LEVEL_MASK (0x3 << 0) +#define DA7213_MICBIAS1_EN_SHIFT 3 +#define DA7213_MICBIAS2_LEVEL_SHIFT 4 +#define DA7213_MICBIAS2_LEVEL_MASK (0x3 << 4) +#define DA7213_MICBIAS2_EN_SHIFT 7 + +/* DA7213_MIXIN_L/R_CTRL = 0x65/0x66 */ +#define DA7213_MIXIN_MIX_EN (0x1 << 3) + +/* DA7213_ADC_L/R_CTRL = 0x67/0x68 */ +#define DA7213_ADC_EN_SHIFT 7 +#define DA7213_ADC_EN (0x1 << 7) + +/* DA7213_DAC_L/R_CTRL = 0x69/0x6A*/ +#define DA7213_DAC_EN_SHIFT 7 + +/* DA7213_HP_L/R_CTRL = 0x6B/0x6C */ +#define DA7213_HP_AMP_OE (0x1 << 3) + +/* DA7213_LINE_CTRL = 0x6D */ +#define DA7213_LINE_AMP_OE (0x1 << 3) + +/* DA7213_MIXOUT_L/R_CTRL = 0x6E/0x6F */ +#define DA7213_MIXOUT_MIX_EN (0x1 << 3) + +/* DA7213_GAIN_RAMP_CTRL = 0x92 */ +#define DA7213_GAIN_RAMP_RATE_SHIFT 0 +#define DA7213_GAIN_RAMP_RATE_MAX 4 + +/* DA7213_MIC_CONFIG = 0x93 */ +#define DA7213_DMIC_DATA_SEL_SHIFT 0 +#define DA7213_DMIC_DATA_SEL_MASK (0x1 << 0) +#define DA7213_DMIC_SAMPLEPHASE_SHIFT 1 +#define DA7213_DMIC_SAMPLEPHASE_MASK (0x1 << 1) +#define DA7213_DMIC_CLK_RATE_SHIFT 2 +#define DA7213_DMIC_CLK_RATE_MASK (0x1 << 2) + +/* DA7213_DIG_CTRL = 0x99 */ +#define DA7213_DAC_L_INV_SHIFT 3 +#define DA7213_DAC_R_INV_SHIFT 7 +#define DA7213_DAC_INV_MAX 0x1 + +/* DA7213_ALC_CTRL2 = 0x9A */ +#define DA7213_ALC_ATTACK_SHIFT 0 +#define DA7213_ALC_ATTACK_MAX 13 +#define DA7213_ALC_RELEASE_SHIFT 4 +#define DA7213_ALC_RELEASE_MAX 11 + +/* DA7213_ALC_CTRL3 = 0x9B */ +#define DA7213_ALC_HOLD_SHIFT 0 +#define DA7213_ALC_HOLD_MAX 16 +#define DA7213_ALC_INTEG_ATTACK_SHIFT 4 +#define DA7213_ALC_INTEG_RELEASE_SHIFT 6 +#define DA7213_ALC_INTEG_MAX 4 + +/* + * DA7213_ALC_NOISE = 0x9C, + * DA7213_ALC_TARGET_MIN/MAX = 0x9D/0x9E + */ +#define DA7213_ALC_THRESHOLD_SHIFT 0 +#define DA7213_ALC_THRESHOLD_MAX 0x3F + +/* DA7213_ALC_GAIN_LIMITS = 0x9F */ +#define DA7213_ALC_ATTEN_MAX_SHIFT 0 +#define DA7213_ALC_GAIN_MAX_SHIFT 4 +#define DA7213_ALC_ATTEN_GAIN_MAX_MAX 0xF + +/* DA7213_ALC_ANA_GAIN_LIMITS = 0xA0 */ +#define DA7213_ALC_ANA_GAIN_MIN_SHIFT 0 +#define DA7213_ALC_ANA_GAIN_MAX_SHIFT 4 +#define DA7213_ALC_ANA_GAIN_MAX 0x7 + +/* DA7213_ALC_ANTICLIP_CTRL = 0xA1 */ +#define DA7213_ALC_ANTICLIP_EN_SHIFT 7 +#define DA7213_ALC_ANTICLIP_EN_MAX 0x1 + +/* DA7213_ALC_ANTICLIP_LEVEL = 0xA2 */ +#define DA7213_ALC_ANTICLIP_LEVEL_SHIFT 0 +#define DA7213_ALC_ANTICLIP_LEVEL_MAX 0x7F + +/* DA7213_ALC_CIC_OP_LVL_CTRL = 0xAD */ +#define DA7213_ALC_DATA_MIDDLE (0x2 << 0) +#define DA7213_ALC_DATA_TOP (0x3 << 0) +#define DA7213_ALC_CIC_OP_CHANNEL_LEFT (0x0 << 7) +#define DA7213_ALC_CIC_OP_CHANNEL_RIGHT (0x1 << 7) + +/* DA7213_DAC_NG_SETUP_TIME = 0xAF */ +#define DA7213_DAC_NG_SETUP_TIME_SHIFT 0 +#define DA7213_DAC_NG_SETUP_TIME_MAX 4 +#define DA7213_DAC_NG_RAMPUP_RATE_SHIFT 2 +#define DA7213_DAC_NG_RAMPDN_RATE_SHIFT 3 +#define DA7213_DAC_NG_RAMP_RATE_MAX 2 + +/* DA7213_DAC_NG_OFF/ON_THRESH = 0xB0/0xB1 */ +#define DA7213_DAC_NG_THRESHOLD_SHIFT 0 +#define DA7213_DAC_NG_THRESHOLD_MAX 0x7 + +/* DA7213_DAC_NG_CTRL = 0xB2 */ +#define DA7213_DAC_NG_EN_SHIFT 7 +#define DA7213_DAC_NG_EN_MAX 0x1 + + +/* + * General defines + */ + +/* Register inversion */ +#define DA7213_NO_INVERT 0 +#define DA7213_INVERT 1 + +/* Byte related defines */ +#define DA7213_BYTE_SHIFT 8 +#define DA7213_BYTE_MASK 0xFF + +/* ALC related */ +#define DA7213_ALC_OFFSET_15_8 0x00FF00 +#define DA7213_ALC_OFFSET_19_16 0x0F0000 +#define DA7213_ALC_AVG_ITERATIONS 5 + +/* PLL related */ +#define DA7213_SYSCLK_MCLK 0 +#define DA7213_SYSCLK_PLL 1 +#define DA7213_PLL_FREQ_OUT_90316800 90316800 +#define DA7213_PLL_FREQ_OUT_98304000 98304000 +#define DA7213_PLL_FREQ_OUT_94310400 94310400 +#define DA7213_PLL_INDIV_5_10_MHZ_VAL 2 +#define DA7213_PLL_INDIV_10_20_MHZ_VAL 4 +#define DA7213_PLL_INDIV_20_40_MHZ_VAL 8 +#define DA7213_PLL_INDIV_40_54_MHZ_VAL 16 + +enum clk_src { + DA7213_CLKSRC_MCLK +}; + +/* Codec private data */ +struct da7213_priv { + struct regmap *regmap; + unsigned int mclk_rate; + bool master; + bool mclk_squarer_en; + bool srm_en; + bool alc_calib_auto; + bool alc_en; + struct da7213_platform_data *pdata; +}; + +#endif /* _DA7213_H */ -- cgit v1.2.3