From c05d9a8c7f55a901d9e8ec2a5f0730137bbfea4a Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 25 Jun 2015 09:35:11 +0100 Subject: ASoC: arizona: Implement stability check for EQ coefficients Specifying unstable coefficients for the EQ can have a severe impact on the audio. This patchs adds a stability check on the coefficients written to the EQ, for this it is necessary to merge the mode control and the coefficients as some coefficients may only be unstable with a certain mode setting so it is ideal if these are always updated in sync. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 59 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/arizona.h | 10 ++++++++ sound/soc/codecs/wm5102.c | 12 ++++------ sound/soc/codecs/wm5110.c | 12 ++++------ sound/soc/codecs/wm8997.c | 12 ++++------ 5 files changed, 81 insertions(+), 24 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 802e05eae3e9..39967863d2ec 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -2313,6 +2313,65 @@ const struct snd_kcontrol_new arizona_adsp2_rate_controls[] = { }; EXPORT_SYMBOL_GPL(arizona_adsp2_rate_controls); +static bool arizona_eq_filter_unstable(bool mode, __be16 _a, __be16 _b) +{ + s16 a = be16_to_cpu(_a); + s16 b = be16_to_cpu(_b); + + if (!mode) { + return abs(a) >= 4096; + } else { + if (abs(b) >= 4096) + return true; + + return (abs((a << 16) / (4096 - b)) >= 4096 << 4); + } +} + +int arizona_eq_coeff_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct arizona *arizona = dev_get_drvdata(codec->dev->parent); + struct soc_bytes *params = (void *)kcontrol->private_value; + unsigned int val; + __be16 *data; + int len; + int ret; + + len = params->num_regs * regmap_get_val_bytes(arizona->regmap); + + data = kmemdup(ucontrol->value.bytes.data, len, GFP_KERNEL | GFP_DMA); + if (!data) + return -ENOMEM; + + data[0] &= cpu_to_be16(ARIZONA_EQ1_B1_MODE); + + if (arizona_eq_filter_unstable(!!data[0], data[1], data[2]) || + arizona_eq_filter_unstable(true, data[4], data[5]) || + arizona_eq_filter_unstable(true, data[8], data[9]) || + arizona_eq_filter_unstable(true, data[12], data[13]) || + arizona_eq_filter_unstable(false, data[16], data[17])) { + dev_err(arizona->dev, "Rejecting unstable EQ coefficients\n"); + ret = -EINVAL; + goto out; + } + + ret = regmap_read(arizona->regmap, params->base, &val); + if (ret != 0) + goto out; + + val &= ~ARIZONA_EQ1_B1_MODE; + data[0] |= cpu_to_be16(val); + + ret = regmap_raw_write(arizona->regmap, params->base, data, len); + +out: + kfree(data); + return ret; +} +EXPORT_SYMBOL_GPL(arizona_eq_coeff_put); + 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 43deb0462309..ebe50cf2f878 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -194,6 +194,13 @@ extern int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS]; ARIZONA_MIXER_ROUTES(name " Preloader", name "L"), \ ARIZONA_MIXER_ROUTES(name " Preloader", name "R") +#define ARIZONA_EQ_CONTROL(xname, xbase) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_bytes_info, .get = snd_soc_bytes_get, \ + .put = arizona_eq_coeff_put, .private_value = \ + ((unsigned long)&(struct soc_bytes) { .base = xbase, \ + .num_regs = 20, .mask = ~ARIZONA_EQ1_B1_MODE }) } + #define ARIZONA_RATE_ENUM_SIZE 4 extern const char *arizona_rate_text[ARIZONA_RATE_ENUM_SIZE]; extern const int arizona_rate_val[ARIZONA_RATE_ENUM_SIZE]; @@ -229,6 +236,9 @@ extern int arizona_hp_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); +extern int arizona_eq_coeff_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); + extern int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id, int source, unsigned int freq, int dir); diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index d097f09e50f2..556bd9840e91 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -788,8 +788,7 @@ ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), -SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 19), -SOC_SINGLE("EQ1 Mode Switch", ARIZONA_EQ1_2, ARIZONA_EQ1_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ1 Coefficients", ARIZONA_EQ1_2), SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT, @@ -801,8 +800,7 @@ SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 19), -SOC_SINGLE("EQ2 Mode Switch", ARIZONA_EQ2_2, ARIZONA_EQ2_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ2 Coefficients", ARIZONA_EQ2_2), SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT, @@ -814,8 +812,7 @@ SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 19), -SOC_SINGLE("EQ3 Mode Switch", ARIZONA_EQ3_2, ARIZONA_EQ3_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ3 Coefficients", ARIZONA_EQ3_2), SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT, @@ -827,8 +824,7 @@ SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 19), -SOC_SINGLE("EQ4 Mode Switch", ARIZONA_EQ4_2, ARIZONA_EQ4_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ4 Coefficients", ARIZONA_EQ4_2), SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT, diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 709fcc6169d8..dfa4d4bad89b 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -247,8 +247,7 @@ ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), -SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 19), -SOC_SINGLE("EQ1 Mode Switch", ARIZONA_EQ1_2, ARIZONA_EQ1_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ1 Coefficients", ARIZONA_EQ1_2), SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT, @@ -260,8 +259,7 @@ SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 19), -SOC_SINGLE("EQ2 Mode Switch", ARIZONA_EQ2_2, ARIZONA_EQ2_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ2 Coefficients", ARIZONA_EQ2_2), SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT, @@ -273,8 +271,7 @@ SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 19), -SOC_SINGLE("EQ3 Mode Switch", ARIZONA_EQ3_2, ARIZONA_EQ3_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ3 Coefficients", ARIZONA_EQ3_2), SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT, @@ -286,8 +283,7 @@ SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 19), -SOC_SINGLE("EQ4 Mode Switch", ARIZONA_EQ4_2, ARIZONA_EQ4_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ4 Coefficients", ARIZONA_EQ4_2), SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT, diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 4134dc7e1243..53e0a467d8de 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -174,8 +174,7 @@ ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), -SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 19), -SOC_SINGLE("EQ1 Mode Switch", ARIZONA_EQ1_2, ARIZONA_EQ1_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ1 Coefficients", ARIZONA_EQ1_2), SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT, @@ -187,8 +186,7 @@ SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 19), -SOC_SINGLE("EQ2 Mode Switch", ARIZONA_EQ2_2, ARIZONA_EQ2_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ2 Coefficients", ARIZONA_EQ2_2), SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT, @@ -200,8 +198,7 @@ SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 19), -SOC_SINGLE("EQ3 Mode Switch", ARIZONA_EQ3_2, ARIZONA_EQ3_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ3 Coefficients", ARIZONA_EQ3_2), SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT, @@ -213,8 +210,7 @@ SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 19), -SOC_SINGLE("EQ4 Mode Switch", ARIZONA_EQ4_2, ARIZONA_EQ4_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ4 Coefficients", ARIZONA_EQ4_2), SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT, -- cgit v1.2.3-59-g8ed1b From 5f8e671a49e1d608fcf52c8944ea7818cd4c99a9 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 25 Jun 2015 09:35:12 +0100 Subject: ASoC: arizona: Implement stability check for LHPF coefficients Specifying unstable coefficients for the low/high pass filters can have a severe impact on the audio. This patchs adds a stability check on the coefficients written to the low/high pass filter block to prevent this. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 17 +++++++++++++++++ sound/soc/codecs/arizona.h | 9 +++++++++ sound/soc/codecs/wm5102.c | 8 ++++---- sound/soc/codecs/wm5110.c | 8 ++++---- sound/soc/codecs/wm8997.c | 8 ++++---- 5 files changed, 38 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 39967863d2ec..a88202d768f4 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -2372,6 +2372,23 @@ out: } EXPORT_SYMBOL_GPL(arizona_eq_coeff_put); +int arizona_lhpf_coeff_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct arizona *arizona = dev_get_drvdata(codec->dev->parent); + __be16 *data = (__be16 *)ucontrol->value.bytes.data; + s16 val = be16_to_cpu(*data); + + if (abs(val) >= 4096) { + dev_err(arizona->dev, "Rejecting unstable LHPF coefficients\n"); + return -EINVAL; + } + + return snd_soc_bytes_put(kcontrol, ucontrol); +} +EXPORT_SYMBOL_GPL(arizona_lhpf_coeff_put); + 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 ebe50cf2f878..13598b0572ee 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -201,6 +201,13 @@ extern int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS]; ((unsigned long)&(struct soc_bytes) { .base = xbase, \ .num_regs = 20, .mask = ~ARIZONA_EQ1_B1_MODE }) } +#define ARIZONA_LHPF_CONTROL(xname, xbase) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_bytes_info, .get = snd_soc_bytes_get, \ + .put = arizona_lhpf_coeff_put, .private_value = \ + ((unsigned long)&(struct soc_bytes) { .base = xbase, \ + .num_regs = 1 }) } + #define ARIZONA_RATE_ENUM_SIZE 4 extern const char *arizona_rate_text[ARIZONA_RATE_ENUM_SIZE]; extern const int arizona_rate_val[ARIZONA_RATE_ENUM_SIZE]; @@ -238,6 +245,8 @@ extern int arizona_hp_ev(struct snd_soc_dapm_widget *w, extern int arizona_eq_coeff_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +extern int arizona_lhpf_coeff_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); extern int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id, int source, unsigned int freq, int dir); diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 556bd9840e91..6ddee999bc51 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -847,10 +847,10 @@ ARIZONA_MIXER_CONTROLS("LHPF2", ARIZONA_HPLP2MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("LHPF3", ARIZONA_HPLP3MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("LHPF4", ARIZONA_HPLP4MIX_INPUT_1_SOURCE), -SND_SOC_BYTES("LHPF1 Coefficients", ARIZONA_HPLPF1_2, 1), -SND_SOC_BYTES("LHPF2 Coefficients", ARIZONA_HPLPF2_2, 1), -SND_SOC_BYTES("LHPF3 Coefficients", ARIZONA_HPLPF3_2, 1), -SND_SOC_BYTES("LHPF4 Coefficients", ARIZONA_HPLPF4_2, 1), +ARIZONA_LHPF_CONTROL("LHPF1 Coefficients", ARIZONA_HPLPF1_2), +ARIZONA_LHPF_CONTROL("LHPF2 Coefficients", ARIZONA_HPLPF2_2), +ARIZONA_LHPF_CONTROL("LHPF3 Coefficients", ARIZONA_HPLPF3_2), +ARIZONA_LHPF_CONTROL("LHPF4 Coefficients", ARIZONA_HPLPF4_2), ARIZONA_MIXER_CONTROLS("DSP1L", ARIZONA_DSP1LMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DSP1R", ARIZONA_DSP1RMIX_INPUT_1_SOURCE), diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index dfa4d4bad89b..05aa5bca97cf 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -310,10 +310,10 @@ ARIZONA_MIXER_CONTROLS("LHPF2", ARIZONA_HPLP2MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("LHPF3", ARIZONA_HPLP3MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("LHPF4", ARIZONA_HPLP4MIX_INPUT_1_SOURCE), -SND_SOC_BYTES("LHPF1 Coefficients", ARIZONA_HPLPF1_2, 1), -SND_SOC_BYTES("LHPF2 Coefficients", ARIZONA_HPLPF2_2, 1), -SND_SOC_BYTES("LHPF3 Coefficients", ARIZONA_HPLPF3_2, 1), -SND_SOC_BYTES("LHPF4 Coefficients", ARIZONA_HPLPF4_2, 1), +ARIZONA_LHPF_CONTROL("LHPF1 Coefficients", ARIZONA_HPLPF1_2), +ARIZONA_LHPF_CONTROL("LHPF2 Coefficients", ARIZONA_HPLPF2_2), +ARIZONA_LHPF_CONTROL("LHPF3 Coefficients", ARIZONA_HPLPF3_2), +ARIZONA_LHPF_CONTROL("LHPF4 Coefficients", ARIZONA_HPLPF4_2), SOC_ENUM("LHPF1 Mode", arizona_lhpf1_mode), SOC_ENUM("LHPF2 Mode", arizona_lhpf2_mode), diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 53e0a467d8de..b4dba3a02aba 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -238,10 +238,10 @@ SOC_ENUM("LHPF2 Mode", arizona_lhpf2_mode), SOC_ENUM("LHPF3 Mode", arizona_lhpf3_mode), SOC_ENUM("LHPF4 Mode", arizona_lhpf4_mode), -SND_SOC_BYTES("LHPF1 Coefficients", ARIZONA_HPLPF1_2, 1), -SND_SOC_BYTES("LHPF2 Coefficients", ARIZONA_HPLPF2_2, 1), -SND_SOC_BYTES("LHPF3 Coefficients", ARIZONA_HPLPF3_2, 1), -SND_SOC_BYTES("LHPF4 Coefficients", ARIZONA_HPLPF4_2, 1), +ARIZONA_LHPF_CONTROL("LHPF1 Coefficients", ARIZONA_HPLPF1_2), +ARIZONA_LHPF_CONTROL("LHPF2 Coefficients", ARIZONA_HPLPF2_2), +ARIZONA_LHPF_CONTROL("LHPF3 Coefficients", ARIZONA_HPLPF3_2), +ARIZONA_LHPF_CONTROL("LHPF4 Coefficients", ARIZONA_HPLPF4_2), SOC_ENUM("ISRC1 FSL", arizona_isrc_fsl[0]), SOC_ENUM("ISRC2 FSL", arizona_isrc_fsl[1]), -- cgit v1.2.3-59-g8ed1b From 93ec3a1ad5b86aef8ca90d2b149ded0f6bb689f5 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 29 Jun 2015 11:15:23 +0800 Subject: ASoC: 88pm860x: Don't change pm860x->dir setting if pm860x_set_dai_sysclk fails 88pm860x does not support slave mode, so it returns -EINVAL for PM860X_CLK_DIR_IN. Current code changes pm860x->dir setting before return error, so it has impact on the logic of pm860x_pcm_set_dai_fmt. This patch adds comment for the reason to return -EINVAL for PM860X_CLK_DIR_IN, and avoid changing pm860x->dir setting if pm860x_set_dai_sysclk fails. Signed-off-by: Axel Lin Acked-by: Haojian Zhuang Signed-off-by: Mark Brown --- sound/soc/codecs/88pm860x-codec.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 38b3dad9d48a..4d91a6aa696b 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -1028,10 +1028,8 @@ static int pm860x_set_dai_sysclk(struct snd_soc_dai *codec_dai, if (dir == PM860X_CLK_DIR_OUT) pm860x->dir = PM860X_CLK_DIR_OUT; - else { - pm860x->dir = PM860X_CLK_DIR_IN; + else /* Slave mode is not supported */ return -EINVAL; - } return 0; } -- cgit v1.2.3-59-g8ed1b From f8ea6cebcfa6499949392da71fc427567c9e5a0e Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 1 Jul 2015 00:56:36 +0800 Subject: ASoC: ak4642: Fix up max_register setting The max_register setting for ak4642, ak4643 and ak4648 are wrong, fix it. According to the datasheet: the maximum valid register for ak4642 is 0x1f the maximum valid register for ak4643 is 0x24 the maximum valid register for ak4648 is 0x27 The default settings for ak4642 and ak4643 are the same for 0x0 ~ 0x1f registers, so it's fine to use the same reg_default table with differnt num_reg_defaults setting. Signed-off-by: Axel Lin Tested-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/codecs/ak4642.c | 33 ++++++++++++++++++++++++++------- 1 file changed, 26 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 7c0f6552c229..fe963e17ceb4 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -64,12 +64,15 @@ #define FIL1_0 0x1c #define FIL1_1 0x1d #define FIL1_2 0x1e -#define FIL1_3 0x1f +#define FIL1_3 0x1f /* The maximum valid register for ak4642 */ #define PW_MGMT4 0x20 #define MD_CTL5 0x21 #define LO_MS 0x22 #define HP_MS 0x23 -#define SPK_MS 0x24 +#define SPK_MS 0x24 /* The maximum valid register for ak4643 */ +#define EQ_FBEQAB 0x25 +#define EQ_FBEQCD 0x26 +#define EQ_FBEQE 0x27 /* The maximum valid register for ak4648 */ /* PW_MGMT1*/ #define PMVCM (1 << 6) /* VCOM Power Management */ @@ -241,7 +244,7 @@ static const struct snd_soc_dapm_route ak4642_intercon[] = { /* * ak4642 register cache */ -static const struct reg_default ak4642_reg[] = { +static const struct reg_default ak4643_reg[] = { { 0, 0x00 }, { 1, 0x00 }, { 2, 0x01 }, { 3, 0x00 }, { 4, 0x02 }, { 5, 0x00 }, { 6, 0x00 }, { 7, 0x00 }, { 8, 0xe1 }, { 9, 0xe1 }, { 10, 0x18 }, { 11, 0x00 }, @@ -254,6 +257,14 @@ static const struct reg_default ak4642_reg[] = { { 36, 0x00 }, }; +/* The default settings for 0x0 ~ 0x1f registers are the same for ak4642 + and ak4643. So we reuse the ak4643 reg_default for ak4642. + The valid registers for ak4642 are 0x0 ~ 0x1f which is a subset of ak4643, + so define NUM_AK4642_REG_DEFAULTS for ak4642. +*/ +#define ak4642_reg ak4643_reg +#define NUM_AK4642_REG_DEFAULTS (FIL1_3 + 1) + static const struct reg_default ak4648_reg[] = { { 0, 0x00 }, { 1, 0x00 }, { 2, 0x01 }, { 3, 0x00 }, { 4, 0x02 }, { 5, 0x00 }, { 6, 0x00 }, { 7, 0x00 }, @@ -535,15 +546,23 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4642 = { static const struct regmap_config ak4642_regmap = { .reg_bits = 8, .val_bits = 8, - .max_register = ARRAY_SIZE(ak4642_reg) + 1, + .max_register = FIL1_3, .reg_defaults = ak4642_reg, - .num_reg_defaults = ARRAY_SIZE(ak4642_reg), + .num_reg_defaults = NUM_AK4642_REG_DEFAULTS, +}; + +static const struct regmap_config ak4643_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = SPK_MS, + .reg_defaults = ak4643_reg, + .num_reg_defaults = ARRAY_SIZE(ak4643_reg), }; static const struct regmap_config ak4648_regmap = { .reg_bits = 8, .val_bits = 8, - .max_register = ARRAY_SIZE(ak4648_reg) + 1, + .max_register = EQ_FBEQE, .reg_defaults = ak4648_reg, .num_reg_defaults = ARRAY_SIZE(ak4648_reg), }; @@ -553,7 +572,7 @@ static const struct ak4642_drvdata ak4642_drvdata = { }; static const struct ak4642_drvdata ak4643_drvdata = { - .regmap_config = &ak4642_regmap, + .regmap_config = &ak4643_regmap, }; static const struct ak4642_drvdata ak4648_drvdata = { -- cgit v1.2.3-59-g8ed1b From 5ed68f0a28f96c5127246d1743dd57b58a090f07 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 9 Jul 2015 11:28:46 +0100 Subject: ASoC: arizona: Use the more idiomatic params_width Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index a88202d768f4..c246fbd70881 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1504,7 +1504,7 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, else rates = &arizona_48k_bclk_rates[0]; - wl = snd_pcm_format_width(params_format(params)); + wl = params_width(params); if (tdm_slots) { arizona_aif_dbg(dai, "Configuring for %d %d bit TDM slots\n", -- cgit v1.2.3-59-g8ed1b From d90c6cc242045115c5c4973114cc93e1c20e2ea8 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 10 Jul 2015 14:37:25 +0100 Subject: ASoC: arizona: Fix error path in codec probe If we fail to add some DSPs or fail to add the controls we should call wm_adsp2_codec_remove for all the cores we have already added. This patch fixes this up on the wm5102 and wm5110. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm5102.c | 7 ++++++- sound/soc/codecs/wm5110.c | 10 ++++++++-- 2 files changed, 14 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 6ddee999bc51..64637d1cf4e5 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1879,7 +1879,7 @@ static int wm5102_codec_probe(struct snd_soc_codec *codec) ret = snd_soc_add_codec_controls(codec, arizona_adsp2_rate_controls, 1); if (ret) - return ret; + goto err_adsp2_codec_probe; arizona_init_spk(codec); arizona_init_gpio(codec); @@ -1889,6 +1889,11 @@ static int wm5102_codec_probe(struct snd_soc_codec *codec) priv->core.arizona->dapm = dapm; return 0; + +err_adsp2_codec_probe: + wm_adsp2_codec_remove(&priv->core.adsp[0], codec); + + return ret; } static int wm5102_codec_remove(struct snd_soc_codec *codec) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 05aa5bca97cf..2d1168c768d9 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -1607,18 +1607,24 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec) for (i = 0; i < WM5110_NUM_ADSP; ++i) { ret = wm_adsp2_codec_probe(&priv->core.adsp[i], codec); if (ret) - return ret; + goto err_adsp2_codec_probe; } ret = snd_soc_add_codec_controls(codec, arizona_adsp2_rate_controls, WM5110_NUM_ADSP); if (ret) - return ret; + goto err_adsp2_codec_probe; snd_soc_dapm_disable_pin(dapm, "HAPTICS"); return 0; + +err_adsp2_codec_probe: + for (--i; i >= 0; --i) + wm_adsp2_codec_remove(&priv->core.adsp[i], codec); + + return ret; } static int wm5110_codec_remove(struct snd_soc_codec *codec) -- cgit v1.2.3-59-g8ed1b From 5f1d980ee9b6353f18765bfa6774a5a08d6cb944 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 21 Jul 2015 21:53:00 +0200 Subject: ALSA: ac97: Add helper function to reset the AC97 device There is currently a lot of code duplication in ASoC drivers regarding the reset handling of devices. This patch introduces a new generic reset function in the generic AC'97 framework that can be used to replace most the custom reset functions. Signed-off-by: Lars-Peter Clausen Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- include/sound/ac97_codec.h | 2 ++ sound/ac97_bus.c | 62 ++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 64 insertions(+) diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h index 0e9d75b49bed..74bc85473b58 100644 --- a/include/sound/ac97_codec.h +++ b/include/sound/ac97_codec.h @@ -584,6 +584,8 @@ static inline int snd_ac97_update_power(struct snd_ac97 *ac97, int reg, void snd_ac97_suspend(struct snd_ac97 *ac97); void snd_ac97_resume(struct snd_ac97 *ac97); #endif +int snd_ac97_reset(struct snd_ac97 *ac97, bool try_warm, unsigned int id, + unsigned int id_mask); /* quirk types */ enum { diff --git a/sound/ac97_bus.c b/sound/ac97_bus.c index 2b50cbe6aca9..55791a0b3943 100644 --- a/sound/ac97_bus.c +++ b/sound/ac97_bus.c @@ -17,6 +17,68 @@ #include #include +/* + * snd_ac97_check_id() - Reads and checks the vendor ID of the device + * @ac97: The AC97 device to check + * @id: The ID to compare to + * @id_mask: Mask that is applied to the device ID before comparing to @id + * + * If @id is 0 this function returns true if the read device vendor ID is + * a valid ID. If @id is non 0 this functions returns true if @id + * matches the read vendor ID. Otherwise the function returns false. + */ +static bool snd_ac97_check_id(struct snd_ac97 *ac97, unsigned int id, + unsigned int id_mask) +{ + ac97->id = ac97->bus->ops->read(ac97, AC97_VENDOR_ID1) << 16; + ac97->id |= ac97->bus->ops->read(ac97, AC97_VENDOR_ID2); + + if (ac97->id == 0x0 || ac97->id == 0xffffffff) + return false; + + if (id != 0 && id != (ac97->id & id_mask)) + return false; + + return true; +} + +/** + * snd_ac97_reset() - Reset AC'97 device + * @ac97: The AC'97 device to reset + * @try_warm: Try a warm reset first + * @id: Expected device vendor ID + * @id_mask: Mask that is applied to the device ID before comparing to @id + * + * This function resets the AC'97 device. If @try_warm is true the function + * first performs a warm reset. If the warm reset is successful the function + * returns 1. Otherwise or if @try_warm is false the function issues cold reset + * followed by a warm reset. If this is successful the function returns 0, + * otherwise a negative error code. If @id is 0 any valid device ID will be + * accepted, otherwise only the ID that matches @id and @id_mask is accepted. + */ +int snd_ac97_reset(struct snd_ac97 *ac97, bool try_warm, unsigned int id, + unsigned int id_mask) +{ + struct snd_ac97_bus_ops *ops = ac97->bus->ops; + + if (try_warm && ops->warm_reset) { + ops->warm_reset(ac97); + if (snd_ac97_check_id(ac97, id, id_mask)) + return 1; + } + + if (ops->reset) + ops->reset(ac97); + if (ops->warm_reset) + ops->warm_reset(ac97); + + if (snd_ac97_check_id(ac97, id, id_mask)) + return 0; + + return -ENODEV; +} +EXPORT_SYMBOL_GPL(snd_ac97_reset); + /* * Let drivers decide whether they want to support given codec from their * probe method. Drivers have direct access to the struct snd_ac97 -- cgit v1.2.3-59-g8ed1b From 7361fbeaeaab5282bbfc88f1f6fe4cf034f7623c Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 21 Jul 2015 21:53:01 +0200 Subject: ASoC: ac97: Add support for resetting device before registration AC97 devices need to be initially reset before they can be used. Currently each driver does this on its own. Add support for resetting the device to core in snd_soc_new_ac97_codec(). If the caller supplies a device ID and device ID mask the function will reset the device and verify that it has the correct ID, if it does not a error is returned. This will allow to remove custom code with similar functionality from individual drivers. Signed-off-by: Lars-Peter Clausen Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- include/sound/soc.h | 3 ++- sound/soc/codecs/ad1980.c | 2 +- sound/soc/codecs/stac9766.c | 2 +- sound/soc/soc-ac97.c | 30 +++++++++++++++++++++++++----- 4 files changed, 29 insertions(+), 8 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 93df8bf9d54a..42d144a4b7ba 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -526,7 +526,8 @@ int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned int reg, #ifdef CONFIG_SND_SOC_AC97_BUS struct snd_ac97 *snd_soc_alloc_ac97_codec(struct snd_soc_codec *codec); -struct snd_ac97 *snd_soc_new_ac97_codec(struct snd_soc_codec *codec); +struct snd_ac97 *snd_soc_new_ac97_codec(struct snd_soc_codec *codec, + unsigned int id, unsigned int id_mask); void snd_soc_free_ac97_codec(struct snd_ac97 *ac97); int snd_soc_set_ac97_ops(struct snd_ac97_bus_ops *ops); diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 3cc69a626454..d9cb81dd64b5 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -240,7 +240,7 @@ static int ad1980_soc_probe(struct snd_soc_codec *codec) u16 vendor_id2; u16 ext_status; - ac97 = snd_soc_new_ac97_codec(codec); + ac97 = snd_soc_new_ac97_codec(codec, 0, 0); if (IS_ERR(ac97)) { ret = PTR_ERR(ac97); dev_err(codec->dev, "Failed to register AC97 codec: %d\n", ret); diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index ed4cca7f6779..c6028300c0ac 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -332,7 +332,7 @@ static int stac9766_codec_probe(struct snd_soc_codec *codec) struct snd_ac97 *ac97; int ret = 0; - ac97 = snd_soc_new_ac97_codec(codec); + ac97 = snd_soc_new_ac97_codec(codec, 0, 0); if (IS_ERR(ac97)) return PTR_ERR(ac97); diff --git a/sound/soc/soc-ac97.c b/sound/soc/soc-ac97.c index 08d7259bbaab..d40efc9fe0a9 100644 --- a/sound/soc/soc-ac97.c +++ b/sound/soc/soc-ac97.c @@ -85,10 +85,19 @@ EXPORT_SYMBOL(snd_soc_alloc_ac97_codec); /** * snd_soc_new_ac97_codec - initailise AC97 device * @codec: audio codec + * @id: The expected device ID + * @id_mask: Mask that is applied to the device ID before comparing with @id * * Initialises AC97 codec resources for use by ad-hoc devices only. + * + * If @id is not 0 this function will reset the device, then read the ID from + * the device and check if it matches the expected ID. If it doesn't match an + * error will be returned and device will not be registered. + * + * Returns: A PTR_ERR() on failure or a valid snd_ac97 struct on success. */ -struct snd_ac97 *snd_soc_new_ac97_codec(struct snd_soc_codec *codec) +struct snd_ac97 *snd_soc_new_ac97_codec(struct snd_soc_codec *codec, + unsigned int id, unsigned int id_mask) { struct snd_ac97 *ac97; int ret; @@ -97,13 +106,24 @@ struct snd_ac97 *snd_soc_new_ac97_codec(struct snd_soc_codec *codec) if (IS_ERR(ac97)) return ac97; - ret = device_add(&ac97->dev); - if (ret) { - put_device(&ac97->dev); - return ERR_PTR(ret); + if (id) { + ret = snd_ac97_reset(ac97, false, id, id_mask); + if (ret < 0) { + dev_err(codec->dev, "Failed to reset AC97 device: %d\n", + ret); + goto err_put_device; + } } + ret = device_add(&ac97->dev); + if (ret) + goto err_put_device; + return ac97; + +err_put_device: + put_device(&ac97->dev); + return ERR_PTR(ret); } EXPORT_SYMBOL_GPL(snd_soc_new_ac97_codec); -- cgit v1.2.3-59-g8ed1b From 3ab3dbdfb91b70ef6bf4eb9b544bf54ff1dff51a Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 21 Jul 2015 21:53:02 +0200 Subject: ASoC: ad1980: Use core AC'97 reset helper Use the new snd_ac97_reset() helper function to perform the reset and verify the device ID. Unfortunately the reset can't be done in snd_soc_new_ac97_codec() due to the special requirements in order to support the non-standard 16-bit slot mode of the ad1980. Signed-off-by: Lars-Peter Clausen Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/ad1980.c | 34 +++++++++++----------------------- 1 file changed, 11 insertions(+), 23 deletions(-) diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index d9cb81dd64b5..9ef20dbccbe3 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -202,19 +202,21 @@ static struct snd_soc_dai_driver ad1980_dai = { .formats = SND_SOC_STD_AC97_FMTS, }, }; +#define AD1980_VENDOR_ID 0x41445300 +#define AD1980_VENDOR_MASK 0xffffff00 + static int ad1980_reset(struct snd_soc_codec *codec, int try_warm) { struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); unsigned int retry_cnt = 0; + int ret; do { - if (try_warm && soc_ac97_ops->warm_reset) { - soc_ac97_ops->warm_reset(ac97); - if (snd_soc_read(codec, AC97_RESET) == 0x0090) - return 1; - } + ret = snd_ac97_reset(ac97, true, AD1980_VENDOR_ID, + AD1980_VENDOR_MASK); + if (ret >= 0) + return 0; - soc_ac97_ops->reset(ac97); /* * Set bit 16slot in register 74h, then every slot will has only * 16 bits. This command is sent out in 20bit mode, in which @@ -223,8 +225,6 @@ static int ad1980_reset(struct snd_soc_codec *codec, int try_warm) */ snd_soc_write(codec, AC97_AD_SERIAL_CFG, 0x9900); - if (snd_soc_read(codec, AC97_RESET) == 0x0090) - return 0; } while (retry_cnt++ < 10); dev_err(codec->dev, "Failed to reset: AC97 link error\n"); @@ -260,22 +260,10 @@ static int ad1980_soc_probe(struct snd_soc_codec *codec) if (ret < 0) goto reset_err; - /* Read out vendor ID to make sure it is ad1980 */ - if (snd_soc_read(codec, AC97_VENDOR_ID1) != 0x4144) { - ret = -ENODEV; - goto reset_err; - } - vendor_id2 = snd_soc_read(codec, AC97_VENDOR_ID2); - - if (vendor_id2 != 0x5370) { - if (vendor_id2 != 0x5374) { - ret = -ENODEV; - goto reset_err; - } else { - dev_warn(codec->dev, - "Found AD1981 - only 2/2 IN/OUT Channels supported\n"); - } + if (vendor_id2 == 0x5374) { + dev_warn(codec->dev, + "Found AD1981 - only 2/2 IN/OUT Channels supported\n"); } /* unmute captures and playbacks volume */ -- cgit v1.2.3-59-g8ed1b From 017e800af9f91861dcd6e4fd8a29418de86fd884 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 21 Jul 2015 21:53:03 +0200 Subject: ASoC: stac9766: Use core reset helper Use the new snd_ac97_reset() helper and the reset functionality provided by snd_soc_new_ac97_codec() to perform the device reset rather than open-coding it. Signed-off-by: Lars-Peter Clausen Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/stac9766.c | 57 ++++++--------------------------------------- 1 file changed, 7 insertions(+), 50 deletions(-) diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index c6028300c0ac..0945c51df003 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -28,6 +28,9 @@ #include "stac9766.h" +#define STAC9766_VENDOR_ID 0x83847666 +#define STAC9766_VENDOR_ID_MASK 0xffffffff + /* * STAC9766 register cache */ @@ -239,45 +242,12 @@ static int stac9766_set_bias_level(struct snd_soc_codec *codec, return 0; } -static int stac9766_reset(struct snd_soc_codec *codec, int try_warm) -{ - struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); - - if (try_warm && soc_ac97_ops->warm_reset) { - soc_ac97_ops->warm_reset(ac97); - if (stac9766_ac97_read(codec, 0) == stac9766_reg[0]) - return 1; - } - - soc_ac97_ops->reset(ac97); - if (soc_ac97_ops->warm_reset) - soc_ac97_ops->warm_reset(ac97); - if (stac9766_ac97_read(codec, 0) != stac9766_reg[0]) - return -EIO; - return 0; -} - static int stac9766_codec_resume(struct snd_soc_codec *codec) { struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); - u16 id, reset; - reset = 0; - /* give the codec an AC97 warm reset to start the link */ -reset: - if (reset > 5) { - dev_err(codec->dev, "Failed to resume\n"); - return -EIO; - } - ac97->bus->ops->warm_reset(ac97); - id = soc_ac97_ops->read(ac97, AC97_VENDOR_ID2); - if (id != 0x4c13) { - stac9766_reset(codec, 0); - reset++; - goto reset; - } - - return 0; + return snd_ac97_reset(ac97, true, STAC9766_VENDOR_ID, + STAC9766_VENDOR_ID_MASK); } static const struct snd_soc_dai_ops stac9766_dai_ops_analog = { @@ -330,28 +300,15 @@ static struct snd_soc_dai_driver stac9766_dai[] = { static int stac9766_codec_probe(struct snd_soc_codec *codec) { struct snd_ac97 *ac97; - int ret = 0; - ac97 = snd_soc_new_ac97_codec(codec, 0, 0); + ac97 = snd_soc_new_ac97_codec(codec, STAC9766_VENDOR_ID, + STAC9766_VENDOR_ID_MASK); if (IS_ERR(ac97)) return PTR_ERR(ac97); snd_soc_codec_set_drvdata(codec, ac97); - /* do a cold reset for the controller and then try - * a warm reset followed by an optional cold reset for codec */ - stac9766_reset(codec, 0); - ret = stac9766_reset(codec, 1); - if (ret < 0) { - dev_err(codec->dev, "Failed to reset: AC97 link error\n"); - goto codec_err; - } - return 0; - -codec_err: - snd_soc_free_ac97_codec(ac97); - return ret; } static int stac9766_codec_remove(struct snd_soc_codec *codec) -- cgit v1.2.3-59-g8ed1b From 6e0b73a0a172f4d881092e388b3a3ad57ca80107 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 21 Jul 2015 21:53:04 +0200 Subject: ASoC: wm9705: Use core AC'97 reset helper Use the new snd_ac97_reset() helper and the reset functionality provided by snd_soc_new_ac97_codec() to perform the device reset rather than open-coding it. Signed-off-by: Lars-Peter Clausen Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/wm9705.c | 40 ++++++++-------------------------------- 1 file changed, 8 insertions(+), 32 deletions(-) diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index 5cc457ef8894..744842c76a60 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -22,6 +22,9 @@ #include "wm9705.h" +#define WM9705_VENDOR_ID 0x574d4c05 +#define WM9705_VENDOR_ID_MASK 0xffffffff + /* * WM9705 register cache */ @@ -293,21 +296,6 @@ static struct snd_soc_dai_driver wm9705_dai[] = { } }; -static int wm9705_reset(struct snd_soc_codec *codec) -{ - struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); - - if (soc_ac97_ops->reset) { - soc_ac97_ops->reset(ac97); - if (ac97_read(codec, 0) == wm9705_reg[0]) - return 0; /* Success */ - } - - dev_err(codec->dev, "Failed to reset: AC97 link error\n"); - - return -EIO; -} - #ifdef CONFIG_PM static int wm9705_soc_suspend(struct snd_soc_codec *codec) { @@ -324,7 +312,8 @@ static int wm9705_soc_resume(struct snd_soc_codec *codec) int i, ret; u16 *cache = codec->reg_cache; - ret = wm9705_reset(codec); + ret = snd_ac97_reset(ac97, true, WM9705_VENDOR_ID, + WM9705_VENDOR_ID_MASK); if (ret < 0) return ret; @@ -342,30 +331,17 @@ static int wm9705_soc_resume(struct snd_soc_codec *codec) static int wm9705_soc_probe(struct snd_soc_codec *codec) { struct snd_ac97 *ac97; - int ret = 0; - ac97 = snd_soc_alloc_ac97_codec(codec); + ac97 = snd_soc_new_ac97_codec(codec, WM9705_VENDOR_ID, + WM9705_VENDOR_ID_MASK); if (IS_ERR(ac97)) { - ret = PTR_ERR(ac97); dev_err(codec->dev, "Failed to register AC97 codec\n"); - return ret; + return PTR_ERR(ac97); } - ret = wm9705_reset(codec); - if (ret) - goto err_put_device; - - ret = device_add(&ac97->dev); - if (ret) - goto err_put_device; - snd_soc_codec_set_drvdata(codec, ac97); return 0; - -err_put_device: - put_device(&ac97->dev); - return ret; } static int wm9705_soc_remove(struct snd_soc_codec *codec) -- cgit v1.2.3-59-g8ed1b From a575be4cbb951244f93342487c2537f729f2239d Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 21 Jul 2015 21:53:05 +0200 Subject: ASoC: wm9712: Use core AC'97 reset helper Use the new snd_ac97_reset() helper and the reset functionality provided by snd_soc_new_ac97_codec() to perform the device reset rather than open-coding it. Signed-off-by: Lars-Peter Clausen Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/wm9712.c | 45 ++++++++------------------------------------- 1 file changed, 8 insertions(+), 37 deletions(-) diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 1fda104dfc45..488a92224249 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -23,6 +23,9 @@ #include #include "wm9712.h" +#define WM9712_VENDOR_ID 0x574d4c12 +#define WM9712_VENDOR_ID_MASK 0xffffffff + struct wm9712_priv { struct snd_ac97 *ac97; unsigned int hp_mixer[2]; @@ -613,35 +616,14 @@ static int wm9712_set_bias_level(struct snd_soc_codec *codec, return 0; } -static int wm9712_reset(struct snd_soc_codec *codec, int try_warm) -{ - struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec); - - if (try_warm && soc_ac97_ops->warm_reset) { - soc_ac97_ops->warm_reset(wm9712->ac97); - if (ac97_read(codec, 0) == wm9712_reg[0]) - return 1; - } - - soc_ac97_ops->reset(wm9712->ac97); - if (soc_ac97_ops->warm_reset) - soc_ac97_ops->warm_reset(wm9712->ac97); - if (ac97_read(codec, 0) != wm9712_reg[0]) - goto err; - return 0; - -err: - dev_err(codec->dev, "Failed to reset: AC97 link error\n"); - return -EIO; -} - static int wm9712_soc_resume(struct snd_soc_codec *codec) { struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec); int i, ret; u16 *cache = codec->reg_cache; - ret = wm9712_reset(codec, 1); + ret = snd_ac97_reset(wm9712->ac97, true, WM9712_VENDOR_ID, + WM9712_VENDOR_ID_MASK); if (ret < 0) return ret; @@ -663,31 +645,20 @@ static int wm9712_soc_resume(struct snd_soc_codec *codec) static int wm9712_soc_probe(struct snd_soc_codec *codec) { struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec); - int ret = 0; + int ret; - wm9712->ac97 = snd_soc_alloc_ac97_codec(codec); + wm9712->ac97 = snd_soc_new_ac97_codec(codec, WM9712_VENDOR_ID, + WM9712_VENDOR_ID_MASK); if (IS_ERR(wm9712->ac97)) { ret = PTR_ERR(wm9712->ac97); dev_err(codec->dev, "Failed to register AC97 codec: %d\n", ret); return ret; } - ret = wm9712_reset(codec, 0); - if (ret < 0) - goto err_put_device; - - ret = device_add(&wm9712->ac97->dev); - if (ret) - goto err_put_device; - /* set alc mux to none */ ac97_write(codec, AC97_VIDEO, ac97_read(codec, AC97_VIDEO) | 0x3000); return 0; - -err_put_device: - put_device(&wm9712->ac97->dev); - return ret; } static int wm9712_soc_remove(struct snd_soc_codec *codec) -- cgit v1.2.3-59-g8ed1b From 310398f5e4618e9a0f6fd0c4b152401daf06c215 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 21 Jul 2015 21:53:06 +0200 Subject: ASoC: wm9713: Use core AC'97 reset helper Use the new snd_ac97_reset() helper and the reset functionality provided by snd_soc_new_ac97_codec() to perform the device reset rather than open-coding it. Signed-off-by: Lars-Peter Clausen Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/wm9713.c | 48 ++++++++--------------------------------------- sound/soc/codecs/wm9713.h | 2 -- 2 files changed, 8 insertions(+), 42 deletions(-) diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 89cd2d6f57c0..955e6511af56 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -29,6 +29,9 @@ #include "wm9713.h" +#define WM9713_VENDOR_ID 0x574d4c13 +#define WM9713_VENDOR_ID_MASK 0xffffffff + struct wm9713_priv { struct snd_ac97 *ac97; u32 pll_in; /* PLL input frequency */ @@ -1123,28 +1126,6 @@ static struct snd_soc_dai_driver wm9713_dai[] = { }, }; -int wm9713_reset(struct snd_soc_codec *codec, int try_warm) -{ - struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec); - - if (try_warm && soc_ac97_ops->warm_reset) { - soc_ac97_ops->warm_reset(wm9713->ac97); - if (ac97_read(codec, 0) == wm9713_reg[0]) - return 1; - } - - soc_ac97_ops->reset(wm9713->ac97); - if (soc_ac97_ops->warm_reset) - soc_ac97_ops->warm_reset(wm9713->ac97); - if (ac97_read(codec, 0) != wm9713_reg[0]) { - dev_err(codec->dev, "Failed to reset: AC97 link error\n"); - return -EIO; - } - - return 0; -} -EXPORT_SYMBOL_GPL(wm9713_reset); - static int wm9713_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -1196,7 +1177,8 @@ static int wm9713_soc_resume(struct snd_soc_codec *codec) int i, ret; u16 *cache = codec->reg_cache; - ret = wm9713_reset(codec, 1); + ret = snd_ac97_reset(wm9713->ac97, true, WM9713_VENDOR_ID, + WM9713_VENDOR_ID_MASK); if (ret < 0) return ret; @@ -1222,32 +1204,18 @@ static int wm9713_soc_resume(struct snd_soc_codec *codec) static int wm9713_soc_probe(struct snd_soc_codec *codec) { struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec); - int ret = 0, reg; + int reg; - wm9713->ac97 = snd_soc_alloc_ac97_codec(codec); + wm9713->ac97 = snd_soc_new_ac97_codec(codec, WM9713_VENDOR_ID, + WM9713_VENDOR_ID_MASK); if (IS_ERR(wm9713->ac97)) return PTR_ERR(wm9713->ac97); - /* do a cold reset for the controller and then try - * a warm reset followed by an optional cold reset for codec */ - wm9713_reset(codec, 0); - ret = wm9713_reset(codec, 1); - if (ret < 0) - goto err_put_device; - - ret = device_add(&wm9713->ac97->dev); - if (ret) - goto err_put_device; - /* unmute the adc - move to kcontrol */ reg = ac97_read(codec, AC97_CD) & 0x7fff; ac97_write(codec, AC97_CD, reg); return 0; - -err_put_device: - put_device(&wm9713->ac97->dev); - return ret; } static int wm9713_soc_remove(struct snd_soc_codec *codec) diff --git a/sound/soc/codecs/wm9713.h b/sound/soc/codecs/wm9713.h index 793da863a03d..53df11b1f727 100644 --- a/sound/soc/codecs/wm9713.h +++ b/sound/soc/codecs/wm9713.h @@ -45,6 +45,4 @@ #define WM9713_DAI_AC97_AUX 1 #define WM9713_DAI_PCM_VOICE 2 -int wm9713_reset(struct snd_soc_codec *codec, int try_warm); - #endif -- cgit v1.2.3-59-g8ed1b From 56113f6e6f8d57d3c184544c6421422558a9988e Mon Sep 17 00:00:00 2001 From: Joe Perches Date: Mon, 3 Aug 2015 10:49:29 -0700 Subject: ASoC: atmel_ssc_dai: Correct misuse of 0x% Correct misuse of 0x%d in logging message. Signed-off-by: Joe Perches Signed-off-by: Mark Brown --- sound/soc/atmel/atmel_ssc_dai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 841d05946b88..ba8def5665c4 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -290,7 +290,7 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, int dir, dir_mask; int ret; - pr_debug("atmel_ssc_startup: SSC_SR=0x%u\n", + pr_debug("atmel_ssc_startup: SSC_SR=0x%x\n", ssc_readl(ssc_p->ssc->regs, SR)); /* Enable PMC peripheral clock for this SSC */ -- cgit v1.2.3-59-g8ed1b