May 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/2] ASoC: rt298: fix null deref on acpi driver data
by Vinod Koul 13 May '16

13 May '16

ACPI driver data can be NULL so we need to check that before dereference the driver data. Signed-off-by: Senthilnathan Veppur <senthilnathanx.veppur(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/codecs/rt298.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index 68cf8d5a174f..45dc5fa24961 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -1179,7 +1179,7 @@ static int rt298_i2c_probe(struct i2c_client *i2c, /* enable jack combo mode on supported devices */ acpiid = acpi_match_device(dev->driver->acpi_match_table, dev); - if (acpiid) { + if (acpiid && acpiid->driver_data) { rt298->pdata = *(struct rt298_platform_data *) acpiid->driver_data; } -- 1.9.1

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[alsa-devel] [PATCH] rcar: src: skip disabled-SRC nodes
by Sergei Shtylyov 13 May '16

13 May '16

The current device tree representation of the R-Car Sample Rate Converters (SRC) assumes that they are numbered consecutively, starting from 0. Alas, this is not the case with the R8A7794 SoC where SRC0 isn't present. In order to keep the existing device trees working, I'm suggesting to use a disabled node for SRC0. Teach the SRC probe to just skip disabled nodes. Signed-off-by: Sergei Shtylyov <sergei.shtylyov(a)cogentembedded.com> --- The patch is against the 'for-next' branch of Mark Brown's 'sound.git' repo. sound/soc/sh/rcar/src.c | 4 ++++ sound/soc/sh/rcar/src.c | 4 ++++ 1 file changed, 4 insertions(+) Index: sound/sound/soc/sh/rcar/src.c =================================================================== --- sound.orig/sound/soc/sh/rcar/src.c +++ sound/sound/soc/sh/rcar/src.c @@ -572,6 +572,9 @@ int rsnd_src_probe(struct rsnd_priv *pri i = 0; for_each_child_of_node(node, np) { + if (!of_device_is_available(np)) + goto skip; + src = rsnd_src_get(priv, i); snprintf(name, RSND_SRC_NAME_SIZE, "%s.%d", @@ -595,6 +598,7 @@ int rsnd_src_probe(struct rsnd_priv *pri if (ret) goto rsnd_src_probe_done; +skip: i++; }

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[alsa-devel] [PATCH] ASoC: rsnd: count .probe/.remove for rsnd_mod_call()
by Kuninori Morimoto 13 May '16

13 May '16

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current rsnd_mod_call is counting its calling count to avoid unbalanced function pair calling for error cases (ex init <-> quit). SSI parent is now controlled as "mod" on current rsnd driver. Because of this reason, SSI .remove function will be called twice if it was used as SSI parent when user tried unbind. But probe/remove pair were not counted. This patch counts probe/remove functions to avoid it. Special thans Hiep Reported-by: Hiep Cao Minh <cm-hiep(a)jinso.co.jp> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/rsnd.h | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index fc89a67..a8f61d7 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -276,8 +276,9 @@ struct rsnd_mod { /* * status * - * 0xH0000CB0 + * 0xH0000CBA * + * A 0: probe 1: remove * B 0: init 1: quit * C 0: start 1: stop * @@ -287,19 +288,19 @@ struct rsnd_mod { * H 0: fallback * H 0: hw_params */ +#define __rsnd_mod_shift_probe 0 +#define __rsnd_mod_shift_remove 0 #define __rsnd_mod_shift_init 4 #define __rsnd_mod_shift_quit 4 #define __rsnd_mod_shift_start 8 #define __rsnd_mod_shift_stop 8 -#define __rsnd_mod_shift_probe 28 /* always called */ -#define __rsnd_mod_shift_remove 28 /* always called */ #define __rsnd_mod_shift_irq 28 /* always called */ #define __rsnd_mod_shift_pcm_new 28 /* always called */ #define __rsnd_mod_shift_fallback 28 /* always called */ #define __rsnd_mod_shift_hw_params 28 /* always called */ -#define __rsnd_mod_add_probe 0 -#define __rsnd_mod_add_remove 0 +#define __rsnd_mod_add_probe 1 +#define __rsnd_mod_add_remove -1 #define __rsnd_mod_add_init 1 #define __rsnd_mod_add_quit -1 #define __rsnd_mod_add_start 1 @@ -310,7 +311,7 @@ struct rsnd_mod { #define __rsnd_mod_add_hw_params 0 #define __rsnd_mod_call_probe 0 -#define __rsnd_mod_call_remove 0 +#define __rsnd_mod_call_remove 1 #define __rsnd_mod_call_init 0 #define __rsnd_mod_call_quit 1 #define __rsnd_mod_call_start 0 -- 1.9.1

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[alsa-devel] [PATCH v2] ASoC: MAX9860: new driver
by Peter Rosin 12 May '16

12 May '16

This is a driver for the MAX9860 Mono Audio Voice Codec. https://datasheets.maximintegrated.com/en/ds/MAX9860.pdf This driver does not support sidetone since the DVST register field is backwards with the mute near the maximum level instead of the minimum. Signed-off-by: Peter Rosin <peda(a)axentia.se> --- MAINTAINERS | 7 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max9860.c | 761 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/max9860.h | 162 ++++++++++ 5 files changed, 938 insertions(+) create mode 100644 sound/soc/codecs/max9860.c create mode 100644 sound/soc/codecs/max9860.h I have yet to find a good spot to add scanning for a mute TLV value needed for the backwards sidetone register field (DVST). Hence, still no sidetone support... New in v2: - Add comment about fall through when Integer Clock Mode is not possible. - Drop export of max9860_probe. - Ignore clk docs and read the mclk rate w/o enabling the clock. - Manage the DVDDIO supply. Cheers, Peter diff --git a/MAINTAINERS b/MAINTAINERS index a727d9959ecd..0803396b10f4 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -7000,6 +7000,13 @@ F: Documentation/devicetree/bindings/i2c/max6697.txt F: drivers/hwmon/max6697.c F: include/linux/platform_data/max6697.h +MAX9860 MONO AUDIO VOICE CODEC DRIVER +M: Peter Rosin <peda(a)axentia.se> +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained +F: Documentation/devicetree/bindings/sound/max9860.txt +F: sound/soc/codecs/max9860* + MAXIM MUIC CHARGER DRIVERS FOR EXYNOS BASED BOARDS M: Krzysztof Kozlowski <k.kozlowski(a)samsung.com> L: linux-pm(a)vger.kernel.org diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7ef3a0c16478..241a23c9aa9f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -83,6 +83,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98925 if I2C select SND_SOC_MAX98926 if I2C select SND_SOC_MAX9850 if I2C + select SND_SOC_MAX9860 if I2C select SND_SOC_MAX9768 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_MC13783 if MFD_MC13XXX @@ -534,6 +535,11 @@ config SND_SOC_MAX98926 config SND_SOC_MAX9850 tristate +config SND_SOC_MAX9860 + tristate "Maxim MAX9860 Mono Audio Voice Codec" + depends on I2C + select REGMAP_I2C + config SND_SOC_PCM1681 tristate "Texas Instruments PCM1681 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 185a712a7fe7..e435f4df1787 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -78,6 +78,7 @@ snd-soc-max9867-objs := max9867.o snd-soc-max98925-objs := max98925.o snd-soc-max98926-objs := max98926.o snd-soc-max9850-objs := max9850.o +snd-soc-max9860-objs := max9860.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o snd-soc-nau8825-objs := nau8825.o @@ -287,6 +288,7 @@ obj-$(CONFIG_SND_SOC_MAX9867) += snd-soc-max9867.o obj-$(CONFIG_SND_SOC_MAX98925) += snd-soc-max98925.o obj-$(CONFIG_SND_SOC_MAX98926) += snd-soc-max98926.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o +obj-$(CONFIG_SND_SOC_MAX9860) += snd-soc-max9860.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o diff --git a/sound/soc/codecs/max9860.c b/sound/soc/codecs/max9860.c new file mode 100644 index 000000000000..ee30832b0afb --- /dev/null +++ b/sound/soc/codecs/max9860.c @@ -0,0 +1,761 @@ +/* + * Driver for the MAX9860 Mono Audio Voice Codec + * + * https://datasheets.maximintegrated.com/en/ds/MAX9860.pdf + * + * The driver does not support sidetone since the DVST register field is + * backwards with the mute near the maximum level instead of the minimum. + * + * Author: Peter Rosin <peda(a)axentia.s> + * Copyright 2016 Axentia Technologies + * + * 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 <linux/init.h> +#include <linux/module.h> +#include <linux/clk.h> +#include <linux/kernel.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/i2c.h> +#include <linux/regulator/consumer.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/pcm_params.h> +#include <sound/tlv.h> + +#include "max9860.h" + +struct max9860_priv { + struct regmap *regmap; + struct regulator *dvddio; + struct notifier_block dvddio_nb; + u8 psclk; + unsigned long pclk_rate; + int fmt; +}; + +static int max9860_dvddio_event(struct notifier_block *nb, + unsigned long event, void *data) +{ + struct max9860_priv *max9860 = container_of(nb, struct max9860_priv, + dvddio_nb); + if (event & REGULATOR_EVENT_DISABLE) { + regcache_mark_dirty(max9860->regmap); + regcache_cache_only(max9860->regmap, true); + } + + return 0; +} + +static const struct reg_default max9860_reg_defaults[] = { + { MAX9860_PWRMAN, 0x00 }, + { MAX9860_INTEN, 0x00 }, + { MAX9860_SYSCLK, 0x00 }, + { MAX9860_AUDIOCLKHIGH, 0x00 }, + { MAX9860_AUDIOCLKLOW, 0x00 }, + { MAX9860_IFC1A, 0x00 }, + { MAX9860_IFC1B, 0x00 }, + { MAX9860_VOICEFLTR, 0x00 }, + { MAX9860_DACATTN, 0x00 }, + { MAX9860_ADCLEVEL, 0x00 }, + { MAX9860_DACGAIN, 0x00 }, + { MAX9860_MICGAIN, 0x00 }, + { MAX9860_MICADC, 0x00 }, + { MAX9860_NOISEGATE, 0x00 }, +}; + +static bool max9860_readable(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX9860_INTRSTATUS ... MAX9860_MICGAIN: + case MAX9860_MICADC ... MAX9860_PWRMAN: + case MAX9860_REVISION: + return true; + } + + return false; +} + +static bool max9860_writeable(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX9860_INTEN ... MAX9860_MICGAIN: + case MAX9860_MICADC ... MAX9860_PWRMAN: + return true; + } + + return false; +} + +static bool max9860_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX9860_INTRSTATUS: + case MAX9860_MICREADBACK: + return true; + } + + return false; +} + +static bool max9860_precious(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX9860_INTRSTATUS: + return true; + } + + return false; +} + +const struct regmap_config max9860_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .readable_reg = max9860_readable, + .writeable_reg = max9860_writeable, + .volatile_reg = max9860_volatile, + .precious_reg = max9860_precious, + + .max_register = MAX9860_MAX_REGISTER, + .reg_defaults = max9860_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(max9860_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; + +static const DECLARE_TLV_DB_SCALE(dva_tlv, -9100, 100, 1); +static const DECLARE_TLV_DB_SCALE(dvg_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(adc_tlv, -1200, 100, 0); +static const DECLARE_TLV_DB_RANGE(pam_tlv, + 0, MAX9860_PAM_MAX - 1, TLV_DB_SCALE_ITEM(-2000, 2000, 1), + MAX9860_PAM_MAX, MAX9860_PAM_MAX, TLV_DB_SCALE_ITEM(3000, 0, 0)); +static const DECLARE_TLV_DB_SCALE(pgam_tlv, 0, 100, 0); +static const DECLARE_TLV_DB_SCALE(anth_tlv, -7600, 400, 1); +static const DECLARE_TLV_DB_SCALE(agcth_tlv, -1800, 100, 0); + +static const char * const agchld_text[] = { + "AGC Disabled", "50ms", "100ms", "400ms" +}; + +static SOC_ENUM_SINGLE_DECL(agchld_enum, MAX9860_MICADC, + MAX9860_AGCHLD_SHIFT, agchld_text); + +static const char * const agcsrc_text[] = { + "Left ADC", "Left/Right ADC" +}; + +static SOC_ENUM_SINGLE_DECL(agcsrc_enum, MAX9860_MICADC, + MAX9860_AGCSRC_SHIFT, agcsrc_text); + +static const char * const agcatk_text[] = { + "3ms", "12ms", "50ms", "200ms" +}; + +static SOC_ENUM_SINGLE_DECL(agcatk_enum, MAX9860_MICADC, + MAX9860_AGCATK_SHIFT, agcatk_text); + +static const char * const agcrls_text[] = { + "78ms", "156ms", "312ms", "625ms", + "1.25s", "2.5s", "5s", "10s" +}; + +static SOC_ENUM_SINGLE_DECL(agcrls_enum, MAX9860_MICADC, + MAX9860_AGCRLS_SHIFT, agcrls_text); + +static const char * const filter_text[] = { + "Disabled", + "Elliptical HP 217Hz notch (16kHz)", + "Butterworth HP 500Hz (16kHz)", + "Elliptical HP 217Hz notch (8kHz)", + "Butterworth HP 500Hz (8kHz)", + "Butterworth HP 200Hz (48kHz)" +}; + +static SOC_ENUM_SINGLE_DECL(avflt_enum, MAX9860_VOICEFLTR, + MAX9860_AVFLT_SHIFT, filter_text); + +static SOC_ENUM_SINGLE_DECL(dvflt_enum, MAX9860_VOICEFLTR, + MAX9860_DVFLT_SHIFT, filter_text); + +static const struct snd_kcontrol_new max9860_controls[] = { +SOC_SINGLE_TLV("Master Playback Volume", MAX9860_DACATTN, + MAX9860_DVA_SHIFT, MAX9860_DVA_MUTE, 1, dva_tlv), +SOC_SINGLE_TLV("DAC Gain Volume", MAX9860_DACGAIN, + MAX9860_DVG_SHIFT, MAX9860_DVG_MAX, 0, dvg_tlv), +SOC_DOUBLE_TLV("Line Capture Volume", MAX9860_ADCLEVEL, + MAX9860_ADCLL_SHIFT, MAX9860_ADCRL_SHIFT, MAX9860_ADCxL_MIN, 1, + adc_tlv), + +SOC_ENUM("AGC Hold Time", agchld_enum), +SOC_ENUM("AGC/Noise Gate Source", agcsrc_enum), +SOC_ENUM("AGC Attack Time", agcatk_enum), +SOC_ENUM("AGC Release Time", agcrls_enum), + +SOC_SINGLE_TLV("Noise Gate Threshold Volume", MAX9860_NOISEGATE, + MAX9860_ANTH_SHIFT, MAX9860_ANTH_MAX, 0, anth_tlv), +SOC_SINGLE_TLV("AGC Signal Threshold Volume", MAX9860_NOISEGATE, + MAX9860_AGCTH_SHIFT, MAX9860_AGCTH_MIN, 1, agcth_tlv), + +SOC_SINGLE_TLV("Mic PGA Volume", MAX9860_MICGAIN, + MAX9860_PGAM_SHIFT, MAX9860_PGAM_MIN, 1, pgam_tlv), +SOC_SINGLE_TLV("Mic Preamp Volume", MAX9860_MICGAIN, + MAX9860_PAM_SHIFT, MAX9860_PAM_MAX, 0, pam_tlv), + +SOC_ENUM("ADC Filter", avflt_enum), +SOC_ENUM("DAC Filter", dvflt_enum), +}; + +static const struct snd_soc_dapm_widget max9860_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("MICL"), +SND_SOC_DAPM_INPUT("MICR"), + +SND_SOC_DAPM_ADC("ADCL", NULL, MAX9860_PWRMAN, MAX9860_ADCLEN_SHIFT, 0), +SND_SOC_DAPM_ADC("ADCR", NULL, MAX9860_PWRMAN, MAX9860_ADCREN_SHIFT, 0), + +SND_SOC_DAPM_AIF_OUT("AIFOUTL", "Capture", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_OUT("AIFOUTR", "Capture", 1, SND_SOC_NOPM, 0, 0), + +SND_SOC_DAPM_AIF_IN("AIFINL", "Playback", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_IN("AIFINR", "Playback", 1, SND_SOC_NOPM, 0, 0), + +SND_SOC_DAPM_DAC("DAC", NULL, MAX9860_PWRMAN, MAX9860_DACEN_SHIFT, 0), + +SND_SOC_DAPM_OUTPUT("OUT"), + +SND_SOC_DAPM_SUPPLY("Supply", SND_SOC_NOPM, 0, 0, + NULL, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_REGULATOR_SUPPLY("AVDD", 0, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("DVDD", 0, 0), +SND_SOC_DAPM_CLOCK_SUPPLY("mclk"), +}; + +static const struct snd_soc_dapm_route max9860_dapm_routes[] = { + { "ADCL", NULL, "MICL" }, + { "ADCR", NULL, "MICR" }, + { "AIFOUTL", NULL, "ADCL" }, + { "AIFOUTR", NULL, "ADCR" }, + + { "DAC", NULL, "AIFINL" }, + { "DAC", NULL, "AIFINR" }, + { "OUT", NULL, "DAC" }, + + { "Supply", NULL, "AVDD" }, + { "Supply", NULL, "DVDD" }, + { "Supply", NULL, "mclk" }, + + { "DAC", NULL, "Supply" }, + { "ADCL", NULL, "Supply" }, + { "ADCR", NULL, "Supply" }, +}; + +static int max9860_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max9860_priv *max9860 = snd_soc_codec_get_drvdata(codec); + + switch (max9860->fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBS_CFS: + return 0; + + default: + return -EINVAL; + } +} + +static int max9860_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 max9860_priv *max9860 = snd_soc_codec_get_drvdata(codec); + u8 master; + u8 ifc1a = 0; + u8 ifc1b = 0; + u8 sysclk = 0; + unsigned long n; + int ret; + + dev_dbg(codec->dev, "hw_params %u Hz, %u channels\n", + params_rate(params), + params_channels(params)); + + if (params_channels(params) == 2) + ifc1b |= MAX9860_ST; + + switch (max9860->fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + master = 0; + break; + case SND_SOC_DAIFMT_CBM_CFM: + master = MAX9860_MASTER; + break; + default: + return -EINVAL; + } + ifc1a |= master; + + if (master) { + if (params_width(params) * params_channels(params) > 48) + ifc1b |= MAX9860_BSEL_64X; + else + ifc1b |= MAX9860_BSEL_48X; + } + + switch (max9860->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ifc1a |= MAX9860_DDLY; + ifc1b |= MAX9860_ADLY; + break; + case SND_SOC_DAIFMT_LEFT_J: + ifc1a |= MAX9860_WCI; + break; + case SND_SOC_DAIFMT_DSP_A: + if (params_width(params) != 16) { + dev_err(codec->dev, + "DSP_A works for 16 bits per sample only.\n"); + return -EINVAL; + } + ifc1a |= MAX9860_DDLY | MAX9860_WCI | MAX9860_HIZ | MAX9860_TDM; + ifc1b |= MAX9860_ADLY; + break; + case SND_SOC_DAIFMT_DSP_B: + if (params_width(params) != 16) { + dev_err(codec->dev, + "DSP_B works for 16 bits per sample only.\n"); + return -EINVAL; + } + ifc1a |= MAX9860_WCI | MAX9860_HIZ | MAX9860_TDM; + break; + default: + return -EINVAL; + } + + switch (max9860->fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + switch (max9860->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + return -EINVAL; + } + ifc1a ^= MAX9860_WCI; + break; + case SND_SOC_DAIFMT_IB_IF: + switch (max9860->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + return -EINVAL; + } + ifc1a ^= MAX9860_WCI; + /* fall through */ + case SND_SOC_DAIFMT_IB_NF: + ifc1a ^= MAX9860_DBCI; + ifc1b ^= MAX9860_ABCI; + break; + default: + return -EINVAL; + } + + dev_dbg(codec->dev, "IFC1A %02x\n", ifc1a); + ret = regmap_write(max9860->regmap, MAX9860_IFC1A, ifc1a); + if (ret) { + dev_err(codec->dev, "Failed to set IFC1A: %d\n", ret); + return ret; + } + dev_dbg(codec->dev, "IFC1B %02x\n", ifc1b); + ret = regmap_write(max9860->regmap, MAX9860_IFC1B, ifc1b); + if (ret) { + dev_err(codec->dev, "Failed to set IFC1B: %d\n", ret); + return ret; + } + + /* + * Check if Integer Clock Mode is possible, but avoid it in slave mode + * since we then do not know if lrclk is derived from pclk and the + * datasheet mentions that the frequencies have to match exactly in + * order for this to work. + */ + if (params_rate(params) == 8000 || params_rate(params) == 16000) { + if (master) { + switch (max9860->pclk_rate) { + case 12000000: + sysclk = MAX9860_FREQ_12MHZ; + break; + case 13000000: + sysclk = MAX9860_FREQ_13MHZ; + break; + case 19200000: + sysclk = MAX9860_FREQ_19_2MHZ; + break; + default: + /* + * Integer Clock Mode not possible. Leave + * sysclk at zero and fall through to the + * code below for PLL mode. + */ + break; + } + + if (sysclk && params_rate(params) == 16000) + sysclk |= MAX9860_16KHZ; + } + } + + /* + * Largest possible n: + * 65536 * 96 * 48kHz / 10MHz -> 30199 + * Smallest possible n: + * 65536 * 96 * 8kHz / 20MHz -> 2517 + * Both fit nicely in the available 15 bits, no need to apply any mask. + */ + n = DIV_ROUND_CLOSEST_ULL(65536ULL * 96 * params_rate(params), + max9860->pclk_rate); + + if (!sysclk) { + /* PLL mode */ + if (params_rate(params) > 24000) + sysclk |= MAX9860_16KHZ; + + if (!master) + n |= 1; /* trigger rapid pll lock mode */ + } + + sysclk |= max9860->psclk; + dev_dbg(codec->dev, "SYSCLK %02x\n", sysclk); + ret = regmap_write(max9860->regmap, + MAX9860_SYSCLK, sysclk); + if (ret) { + dev_err(codec->dev, "Failed to set SYSCLK: %d\n", ret); + return ret; + } + dev_dbg(codec->dev, "N %lu\n", n); + ret = regmap_write(max9860->regmap, + MAX9860_AUDIOCLKHIGH, n >> 8); + if (ret) { + dev_err(codec->dev, "Failed to set NHI: %d\n", ret); + return ret; + } + ret = regmap_write(max9860->regmap, + MAX9860_AUDIOCLKLOW, n & 0xff); + if (ret) { + dev_err(codec->dev, "Failed to set NLO: %d\n", ret); + return ret; + } + + if (!master) { + dev_dbg(codec->dev, "Enable PLL\n"); + ret = regmap_update_bits(max9860->regmap, MAX9860_AUDIOCLKHIGH, + MAX9860_PLL, MAX9860_PLL); + if (ret) { + dev_err(codec->dev, "Failed to enable PLL: %d\n", ret); + return ret; + } + } + + return 0; +} + +static int max9860_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct max9860_priv *max9860 = snd_soc_codec_get_drvdata(codec); + + max9860->fmt = fmt; + return 0; +} + +static const struct snd_soc_dai_ops max9860_dai_ops = { + .startup = max9860_dai_startup, + .hw_params = max9860_hw_params, + .set_fmt = max9860_set_fmt, +}; + +static struct snd_soc_dai_driver max9860_dai = { + .name = "max9860-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &max9860_dai_ops, + .symmetric_rates = 1, +}; + +static int max9860_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct max9860_priv *max9860 = dev_get_drvdata(codec->dev); + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + ret = regmap_update_bits(max9860->regmap, MAX9860_PWRMAN, + MAX9860_SHDN, MAX9860_SHDN); + if (ret) { + dev_err(codec->dev, "Failed to remove SHDN: %d\n", ret); + return ret; + } + break; + + case SND_SOC_BIAS_OFF: + ret = regmap_update_bits(max9860->regmap, MAX9860_PWRMAN, + MAX9860_SHDN, 0); + if (ret) { + dev_err(codec->dev, "Failed to request SHDN: %d\n", + ret); + return ret; + } + break; + } + + return 0; +} + +static struct snd_soc_codec_driver max9860_codec_driver = { + .set_bias_level = max9860_set_bias_level, + .idle_bias_off = true, + + .controls = max9860_controls, + .num_controls = ARRAY_SIZE(max9860_controls), + .dapm_widgets = max9860_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max9860_dapm_widgets), + .dapm_routes = max9860_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(max9860_dapm_routes), +}; + +#ifdef CONFIG_PM +static int max9860_suspend(struct device *dev) +{ + struct max9860_priv *max9860 = dev_get_drvdata(dev); + int ret; + + ret = regmap_update_bits(max9860->regmap, MAX9860_SYSCLK, + MAX9860_PSCLK, MAX9860_PSCLK_OFF); + if (ret) { + dev_err(dev, "Failed to disable clock: %d\n", ret); + return ret; + } + + regulator_disable(max9860->dvddio); + + return 0; +} + +static int max9860_resume(struct device *dev) +{ + struct max9860_priv *max9860 = dev_get_drvdata(dev); + int ret; + + ret = regulator_enable(max9860->dvddio); + if (ret) { + dev_err(dev, "Failed to enable DVDDIO: %d\n", ret); + return ret; + } + + regcache_cache_only(max9860->regmap, false); + ret = regcache_sync(max9860->regmap); + if (ret) { + dev_err(dev, "Failed to sync cache: %d\n", ret); + return ret; + } + + ret = regmap_update_bits(max9860->regmap, MAX9860_SYSCLK, + MAX9860_PSCLK, max9860->psclk); + if (ret) { + dev_err(dev, "Failed to enable clock: %d\n", ret); + return ret; + } + + return 0; +} +#endif + +const struct dev_pm_ops max9860_pm_ops = { + SET_RUNTIME_PM_OPS(max9860_suspend, max9860_resume, NULL) +}; + +static int max9860_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct max9860_priv *max9860; + int ret; + struct clk *mclk; + unsigned long mclk_rate; + int i; + int intr; + + max9860 = devm_kzalloc(dev, sizeof(struct max9860_priv), GFP_KERNEL); + if (!max9860) + return -ENOMEM; + + max9860->dvddio = devm_regulator_get(dev, "DVDDIO"); + if (IS_ERR(max9860->dvddio)) { + ret = PTR_ERR(max9860->dvddio); + dev_err(dev, "Failed to get DVDDIO supply: %d\n", ret); + return ret; + } + + max9860->dvddio_nb.notifier_call = max9860_dvddio_event; + + ret = regulator_register_notifier(max9860->dvddio, &max9860->dvddio_nb); + if (ret) + dev_err(dev, "Failed to register DVDDIO notifier: %d\n", ret); + + ret = regulator_enable(max9860->dvddio); + if (ret != 0) { + dev_err(dev, "Failed to enable DVDDIO: %d\n", ret); + return ret; + } + + max9860->regmap = devm_regmap_init_i2c(i2c, &max9860_regmap); + if (IS_ERR(max9860->regmap)) { + ret = PTR_ERR(max9860->regmap); + goto err_regulator; + } + + dev_set_drvdata(dev, max9860); + + /* + * mclk has to be in the 10MHz to 60MHz range. + * psclk is used to scale mclk into pclk so that + * pclk is in the 10MHz to 20MHz range. + */ + mclk = clk_get(dev, "mclk"); + + if (IS_ERR(mclk)) { + ret = PTR_ERR(mclk); + dev_err(dev, "Failed to get MCLK: %d\n", ret); + goto err_regulator; + } + + mclk_rate = clk_get_rate(mclk); + clk_put(mclk); + + if (mclk_rate > 60000000 || mclk_rate < 10000000) { + dev_err(dev, "Bad mclk %luHz (needs 10MHz - 60MHz)\n", + mclk_rate); + ret = -EINVAL; + goto err_regulator; + } + if (mclk_rate >= 40000000) + max9860->psclk = 3; + else if (mclk_rate >= 20000000) + max9860->psclk = 2; + else + max9860->psclk = 1; + max9860->pclk_rate = mclk_rate >> (max9860->psclk - 1); + max9860->psclk <<= MAX9860_PSCLK_SHIFT; + dev_dbg(dev, "mclk %lu pclk %lu\n", mclk_rate, max9860->pclk_rate); + + regcache_cache_bypass(max9860->regmap, true); + for (i = 0; i < max9860_regmap.num_reg_defaults; ++i) { + ret = regmap_write(max9860->regmap, + max9860_regmap.reg_defaults[i].reg, + max9860_regmap.reg_defaults[i].def); + if (ret) { + dev_err(dev, "Failed to initialize register %u: %d\n", + max9860_regmap.reg_defaults[i].reg, ret); + goto err_regulator; + } + } + regcache_cache_bypass(max9860->regmap, false); + + ret = regmap_read(max9860->regmap, MAX9860_INTRSTATUS, &intr); + if (ret) { + dev_err(dev, "Failed to clear INTRSTATUS: %d\n", ret); + goto err_regulator; + } + + pm_runtime_set_active(dev); + pm_runtime_enable(dev); + pm_runtime_idle(dev); + + ret = snd_soc_register_codec(dev, &max9860_codec_driver, + &max9860_dai, 1); + if (ret) { + dev_err(dev, "Failed to register CODEC: %d\n", ret); + goto err_pm; + } + + return 0; + +err_pm: + pm_runtime_disable(dev); +err_regulator: + regulator_disable(max9860->dvddio); + return ret; +} + +static int max9860_remove(struct i2c_client *i2c) +{ + struct device *dev = &i2c->dev; + struct max9860_priv *max9860 = dev_get_drvdata(dev); + + snd_soc_unregister_codec(dev); + pm_runtime_disable(dev); + regulator_disable(max9860->dvddio); + return 0; +} + +static const struct i2c_device_id max9860_i2c_id[] = { + { "max9860", }, + { } +}; +MODULE_DEVICE_TABLE(i2c, max9860_i2c_id); + +static const struct of_device_id max9860_of_match[] = { + { .compatible = "maxim,max9860", }, + { } +}; +MODULE_DEVICE_TABLE(of, max9860_of_match); + +static struct i2c_driver max9860_i2c_driver = { + .probe = max9860_probe, + .remove = max9860_remove, + .id_table = max9860_i2c_id, + .driver = { + .name = "max9860", + .of_match_table = max9860_of_match, + .pm = &max9860_pm_ops, + }, +}; + +module_i2c_driver(max9860_i2c_driver); + +MODULE_DESCRIPTION("ASoC MAX9860 Mono Audio Voice Codec driver"); +MODULE_AUTHOR("Peter Rosin <peda(a)axentia.se>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/max9860.h b/sound/soc/codecs/max9860.h new file mode 100644 index 000000000000..22041bd67a7d --- /dev/null +++ b/sound/soc/codecs/max9860.h @@ -0,0 +1,162 @@ +/* + * Driver for the MAX9860 Mono Audio Voice Codec + * + * Author: Peter Rosin <peda(a)axentia.s> + * Copyright 2016 Axentia Technologies + * + * 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 _SND_SOC_MAX9860 +#define _SND_SOC_MAX9860 + +#define MAX9860_INTRSTATUS 0x00 +#define MAX9860_MICREADBACK 0x01 +#define MAX9860_INTEN 0x02 +#define MAX9860_SYSCLK 0x03 +#define MAX9860_AUDIOCLKHIGH 0x04 +#define MAX9860_AUDIOCLKLOW 0x05 +#define MAX9860_IFC1A 0x06 +#define MAX9860_IFC1B 0x07 +#define MAX9860_VOICEFLTR 0x08 +#define MAX9860_DACATTN 0x09 +#define MAX9860_ADCLEVEL 0x0a +#define MAX9860_DACGAIN 0x0b +#define MAX9860_MICGAIN 0x0c +#define MAX9860_RESERVED 0x0d +#define MAX9860_MICADC 0x0e +#define MAX9860_NOISEGATE 0x0f +#define MAX9860_PWRMAN 0x10 +#define MAX9860_REVISION 0xff + +#define MAX9860_MAX_REGISTER 0xff + +/* INTRSTATUS */ +#define MAX9860_CLD 0x80 +#define MAX9860_SLD 0x40 +#define MAX9860_ULK 0x20 + +/* MICREADBACK */ +#define MAX9860_NG 0xe0 +#define MAX9860_AGC 0x1f + +/* INTEN */ +#define MAX9860_ICLD 0x80 +#define MAX9860_ISLD 0x40 +#define MAX9860_IULK 0x20 + +/* SYSCLK */ +#define MAX9860_PSCLK 0x30 +#define MAX9860_PSCLK_OFF 0x00 +#define MAX9860_PSCLK_SHIFT 4 +#define MAX9860_FREQ 0x06 +#define MAX9860_FREQ_NORMAL 0x00 +#define MAX9860_FREQ_12MHZ 0x02 +#define MAX9860_FREQ_13MHZ 0x04 +#define MAX9860_FREQ_19_2MHZ 0x06 +#define MAX9860_16KHZ 0x01 + +/* AUDIOCLKHIGH */ +#define MAX9860_PLL 0x80 +#define MAX9860_NHI 0x7f + +/* AUDIOCLKLOW */ +#define MAX9860_NLO 0xff + +/* IFC1A */ +#define MAX9860_MASTER 0x80 +#define MAX9860_WCI 0x40 +#define MAX9860_DBCI 0x20 +#define MAX9860_DDLY 0x10 +#define MAX9860_HIZ 0x08 +#define MAX9860_TDM 0x04 + +/* IFC1B */ +#define MAX9860_ABCI 0x20 +#define MAX9860_ADLY 0x10 +#define MAX9860_ST 0x08 +#define MAX9860_BSEL 0x07 +#define MAX9860_BSEL_OFF 0x00 +#define MAX9860_BSEL_64X 0x01 +#define MAX9860_BSEL_48X 0x02 +#define MAX9860_BSEL_PCLK_2 0x04 +#define MAX9860_BSEL_PCLK_4 0x05 +#define MAX9860_BSEL_PCLK_8 0x06 +#define MAX9860_BSEL_PCLK_16 0x07 + +/* VOICEFLTR */ +#define MAX9860_AVFLT 0xf0 +#define MAX9860_AVFLT_SHIFT 4 +#define MAX9860_AVFLT_COUNT 6 +#define MAX9860_DVFLT 0x0f +#define MAX9860_DVFLT_SHIFT 0 +#define MAX9860_DVFLT_COUNT 6 + +/* DACATTN */ +#define MAX9860_DVA 0xfe +#define MAX9860_DVA_SHIFT 1 +#define MAX9860_DVA_MUTE 0x5e + +/* ADCLEVEL */ +#define MAX9860_ADCRL 0xf0 +#define MAX9860_ADCRL_SHIFT 4 +#define MAX9860_ADCLL 0x0f +#define MAX9860_ADCLL_SHIFT 0 +#define MAX9860_ADCxL_MIN 15 + +/* DACGAIN */ +#define MAX9860_DVG 0x60 +#define MAX9860_DVG_SHIFT 5 +#define MAX9860_DVG_MAX 3 +#define MAX9860_DVST 0x1f +#define MAX9860_DVST_SHIFT 0 +#define MAX9860_DVST_MIN 31 + +/* MICGAIN */ +#define MAX9860_PAM 0x60 +#define MAX9860_PAM_SHIFT 5 +#define MAX9860_PAM_MAX 3 +#define MAX9860_PGAM 0x1f +#define MAX9860_PGAM_SHIFT 0 +#define MAX9860_PGAM_MIN 20 + +/* MICADC */ +#define MAX9860_AGCSRC 0x80 +#define MAX9860_AGCSRC_SHIFT 7 +#define MAX9860_AGCSRC_COUNT 2 +#define MAX9860_AGCRLS 0x70 +#define MAX9860_AGCRLS_SHIFT 4 +#define MAX9860_AGCRLS_COUNT 8 +#define MAX9860_AGCATK 0x0c +#define MAX9860_AGCATK_SHIFT 2 +#define MAX9860_AGCATK_COUNT 4 +#define MAX9860_AGCHLD 0x03 +#define MAX9860_AGCHLD_OFF 0x00 +#define MAX9860_AGCHLD_SHIFT 0 +#define MAX9860_AGCHLD_COUNT 4 + +/* NOISEGATE */ +#define MAX9860_ANTH 0xf0 +#define MAX9860_ANTH_SHIFT 4 +#define MAX9860_ANTH_MAX 15 +#define MAX9860_AGCTH 0x0f +#define MAX9860_AGCTH_SHIFT 0 +#define MAX9860_AGCTH_MIN 15 + +/* PWRMAN */ +#define MAX9860_SHDN 0x80 +#define MAX9860_DACEN 0x08 +#define MAX9860_DACEN_SHIFT 3 +#define MAX9860_ADCLEN 0x02 +#define MAX9860_ADCLEN_SHIFT 1 +#define MAX9860_ADCREN 0x01 +#define MAX9860_ADCREN_SHIFT 0 + +#endif /* _SND_SOC_MAX9860 */ -- 2.1.4

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[alsa-devel] [PATCH] ALSA: au88x0: Fix zero clear of stream->resources
by Takashi Iwai 12 May '16

12 May '16

There are a few calls of memset() to stream->resources, but they all are called in a wrong size, sizeof(unsigned char) * VORTEX_RESOURCE_LAST, while this field is a u32 array. This may leave the memories not zero-cleared. Fix it by replacing them with a simpler sizeof(stream->resources) instead. Reported-by: David Binderman <dcb314(a)hotmail.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/au88x0/au88x0_core.c | 14 +++++--------- 1 file changed, 5 insertions(+), 9 deletions(-) diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c index 4667c3232b7f..4a054d720112 100644 --- a/sound/pci/au88x0/au88x0_core.c +++ b/sound/pci/au88x0/au88x0_core.c @@ -2151,8 +2151,7 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, stream->resources, en, VORTEX_RESOURCE_SRC)) < 0) { memset(stream->resources, 0, - sizeof(unsigned char) * - VORTEX_RESOURCE_LAST); + sizeof(stream->resources)); return -EBUSY; } if (stream->type != VORTEX_PCM_A3D) { @@ -2162,7 +2161,7 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, VORTEX_RESOURCE_MIXIN)) < 0) { memset(stream->resources, 0, - sizeof(unsigned char) * VORTEX_RESOURCE_LAST); + sizeof(stream->resources)); return -EBUSY; } } @@ -2175,8 +2174,7 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, stream->resources, en, VORTEX_RESOURCE_A3D)) < 0) { memset(stream->resources, 0, - sizeof(unsigned char) * - VORTEX_RESOURCE_LAST); + sizeof(stream->resources)); dev_err(vortex->card->dev, "out of A3D sources. Sorry\n"); return -EBUSY; @@ -2290,8 +2288,7 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, VORTEX_RESOURCE_MIXOUT)) < 0) { memset(stream->resources, 0, - sizeof(unsigned char) * - VORTEX_RESOURCE_LAST); + sizeof(stream->resources)); return -EBUSY; } if ((src[i] = @@ -2299,8 +2296,7 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, stream->resources, en, VORTEX_RESOURCE_SRC)) < 0) { memset(stream->resources, 0, - sizeof(unsigned char) * - VORTEX_RESOURCE_LAST); + sizeof(stream->resources)); return -EBUSY; } } -- 2.8.2

1 0

[alsa-devel] sound/pci/au88x0/au88x0_core.c: many calls to memset with not-plausible size ?
by David Binderman 12 May '16

12 May '16

Hello there, sound/pci/au88x0/au88x0_core.c:2153:6: warning: ‘memset’ used with length equal to number of elements without multiplication by element size [-Wmemset-elt-size] Source code is memset(stream->resources, 0, sizeof(unsigned char) * VORTEX_RESOURCE_LAST); but sound/pci/au88x0/au88x0.h: u32 resources[5]; Something like memset(stream->resources, 0, sizeof(stream->resources[0]) * VORTEX_RESOURCE_LAST); Some duplicates: sound/pci/au88x0/au88x0_core.c:2163:7: warning: ‘memset’ used with length equal to number of elements without multiplication by element size [-Wmemset-elt-size] sound/pci/au88x0/au88x0_core.c:2177:5: warning: ‘memset’ used with length equal to number of elements without multiplication by element size [-Wmemset-elt-size] sound/pci/au88x0/au88x0_core.c:2292:5: warning: ‘memset’ used with length equal to number of elements without multiplication by element size [-Wmemset-elt-size] sound/pci/au88x0/au88x0_core.c:2301:5: warning: ‘memset’ used with length equal to number of elements without multiplication by element size [-Wmemset-elt-size] Regards David Binderman

2 2

[alsa-devel] [PATCH] ASoC: wm8985: rework and fix the clock calculation
by Petr Kulhavy 12 May '16

12 May '16

The clock calculation has several issues: 1) if PLL is used in master mode the BCLK output runs at double the speed 2) de-facto only 44.1kHz and 48kHz sampling rates are supported, other rates like 8kHz, 12kHz, 24kHz fail to find the proper BCLK divider 3) the wm8985->sysclk variable has a misleading name and is used wrongly in the clock calculation in wm8985_hw_params() which is the root cause for (1) 4) wm8985->bclk is used only in wm8985_hw_params() and therefore no need to store it in the wm8985_priv structure Therefore the clock calculation is rewritten in more clean and proper way: - move wm8985_priv->bclk as a local variable into mw8985_hw_params() - new variable wm8985_priv->pllout holds the actual frequency that is input to the MCLKDIV post-divider - move wm8985_priv->sysclk as a local variable into mw8985_hw_params() - sysclk is now always calculated as 256 * fs - the MCLKDIV is looked up as pllout/sysclk - fs_ratios[] is replaced by simpler mclk_divs[] lookup table With this patch all rates: 8, 11.025, 12, 16, 22.05, 24, 32, 44.1 and 48kHz work properly and generate the correct BCLK. Signed-off-by: Petr Kulhavy <petr(a)barix.com> --- sound/soc/codecs/wm8985.c | 59 ++++++++++++++++++++++++----------------------- 1 file changed, 30 insertions(+), 29 deletions(-) diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c index 18f2babe1090..628127aa3c96 100644 --- a/sound/soc/codecs/wm8985.c +++ b/sound/soc/codecs/wm8985.c @@ -181,22 +181,11 @@ static const int volume_update_regs[] = { struct wm8985_priv { struct regmap *regmap; struct regulator_bulk_data supplies[WM8985_NUM_SUPPLIES]; - unsigned int sysclk; - unsigned int bclk; + unsigned int pllout; /* input rate to the MCLKDIV divider */ }; -static const struct { - int div; - int ratio; -} fs_ratios[] = { - { 10, 128 }, - { 15, 192 }, - { 20, 256 }, - { 30, 384 }, - { 40, 512 }, - { 60, 768 }, - { 80, 1024 }, - { 120, 1536 } +static const int mclk_divs[] = { + 10, 15, 20, 30, 40, 60, 80, 120 }; static const int srates[] = { 48000, 32000, 24000, 16000, 12000, 8000 }; @@ -693,15 +682,18 @@ static int wm8985_hw_params(struct snd_pcm_substream *substream, struct snd_soc_codec *codec; struct wm8985_priv *wm8985; u16 blen, srate_idx; - unsigned int tmp; int srate_best; + unsigned int bclk; /* bit clock matching the current params */ + unsigned int sysclk; /* the actual sysclk after post division */ codec = dai->codec; wm8985 = snd_soc_codec_get_drvdata(codec); - wm8985->bclk = snd_soc_params_to_bclk(params); - if ((int)wm8985->bclk < 0) - return wm8985->bclk; + /* always use 256 * fs in order to get best filter quality */ + sysclk = 256 * params_rate(params); + bclk = snd_soc_params_to_bclk(params); + if ((int)bclk < 0) + return bclk; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: @@ -742,29 +734,28 @@ static int wm8985_hw_params(struct snd_pcm_substream *substream, snd_soc_update_bits(codec, WM8985_ADDITIONAL_CONTROL, WM8985_SR_MASK, srate_idx << WM8985_SR_SHIFT); - dev_dbg(dai->dev, "Target BCLK = %uHz\n", wm8985->bclk); - dev_dbg(dai->dev, "SYSCLK = %uHz\n", wm8985->sysclk); + dev_dbg(dai->dev, "Target BCLK = %uHz\n", bclk); + dev_dbg(dai->dev, "SYSCLK = %uHz\n", sysclk); - for (i = 0; i < ARRAY_SIZE(fs_ratios); ++i) { - if (wm8985->sysclk / params_rate(params) - == fs_ratios[i].ratio) + /* select the appropriate mclk divider */ + for (i = 0; i < ARRAY_SIZE(mclk_divs); ++i) { + if (wm8985->pllout / mclk_divs[i] * 10 + == sysclk) break; } - if (i == ARRAY_SIZE(fs_ratios)) { + if (i == ARRAY_SIZE(mclk_divs)) { dev_err(dai->dev, "Unable to configure MCLK ratio %u/%u\n", - wm8985->sysclk, params_rate(params)); + wm8985->pllout, sysclk); return -EINVAL; } - dev_dbg(dai->dev, "MCLK ratio = %dfs\n", fs_ratios[i].ratio); snd_soc_update_bits(codec, WM8985_CLOCK_GEN_CONTROL, WM8985_MCLKDIV_MASK, i << WM8985_MCLKDIV_SHIFT); /* select the appropriate bclk divider */ - tmp = (wm8985->sysclk / fs_ratios[i].div) * 10; for (i = 0; i < ARRAY_SIZE(bclk_divs); ++i) { - if (wm8985->bclk == tmp / bclk_divs[i]) + if (bclk == sysclk / bclk_divs[i]) break; } @@ -786,6 +777,10 @@ struct pll_div { }; #define FIXED_PLL_SIZE ((1ULL << 24) * 10) +/* + * source = MCLK input of the chip + * target = the f2 coming out of the PLL before /4 divider + */ static int pll_factors(struct pll_div *pll_div, unsigned int target, unsigned int source) { @@ -872,17 +867,23 @@ static int wm8985_set_sysclk(struct snd_soc_dai *dai, WM8985_CLKSEL_MASK, 0); snd_soc_update_bits(codec, WM8985_POWER_MANAGEMENT_1, WM8985_PLLEN_MASK, 0); + wm8985->pllout = freq; break; case WM8985_CLKSRC_PLL: snd_soc_update_bits(codec, WM8985_CLOCK_GEN_CONTROL, WM8985_CLKSEL_MASK, WM8985_CLKSEL); + /* + * in order to run the PLL within the recommended 90MHz + * operating range the wm8985_set_pll() configures the PLL + * to output double the required frequency + */ + wm8985->pllout = 2 * freq; break; default: dev_err(dai->dev, "Unknown clock source %d\n", clk_id); return -EINVAL; } - wm8985->sysclk = freq; return 0; } -- 1.9.1

2 4

[alsa-devel] [PATCH 1/2] ASoC: nau8825: non-clock jack detection for power saving at standby
by John Hsu 12 May '16

12 May '16

The driver changes jack type detection interruption to non-clock archi- tecture for less 1mW power saving. The architecture is called manual mode jack detection. It has no hardware debounce, no jack type detection, but only detecting jack insertion. After jack insertion, the driver will switch to auto mode jack detection with internal clock which can detect microphone, jack type and do hardware debounce. The manual architecture has these main changes including codec initiation, interruption, clock control, and power management. When codec initiation or system resume, the clock is closed as jack insertion detection at man- ual mode, and bypass debounce circuit. These configurations move to resume setup function when setup bias level after resume. When jack insertion detection happens, the manual mode turns off and make configuration about jack type detection interruption at auto mode in auto irq setup function which can detect microphone and jack type. The inter- ruption will switch to manual mode again with clock free until jack ejec- tion happens. The system clock configuration adds clock disable option which can disable internal VCO clock. Before the system clock change, there is an restric- tion added to make sure clock disabled and not config any clock when no headset connected. In power management, we involve the solution about races and jack detec- tion in resume from Ben Zhang in the following patch and list his comment. [PATCH] ASoC: nau8825: Fix jack detection across suspend "Jack plug status is rechecked at resume to handle plug/unplug in S3 when the chip has no power." "Suspend/resume callbacks are moved from the i2c dev_pm_ops to snd_soc_codec_driver. soc_resume_deferred is a delayed work which may trigger nau8825_set_bias_level. The bias change races against dev_pm_ops, causing jack detection issues. soc_resume_deferred ensures bias change and snd_soc_codec_driver suspend/resume are sequenced correctly." Change SAR widget to supply type which can prevent the codec keeping at SND_SOC_BIAS_ON during suspend. The codec suspend function can just invoke normally. Before the system suspends, the driver turns off all interruptions. Keep the interruption quiet before resume setup completes. The ADC channel will be disabled which is needed for interruptions at audo mode. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> Signed-off-by: Ben Zhang <benzh(a)chromium.org> --- sound/soc/codecs/nau8825.c | 281 +++++++++++++++++++++++++++++++++++---------- sound/soc/codecs/nau8825.h | 9 +- 2 files changed, 226 insertions(+), 64 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index f35019a..9386624 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -30,10 +30,16 @@ #include "nau8825.h" + +#define NUVOTON_CODEC_DAI "nau8825-hifi" + #define NAU_FREF_MAX 13500000 #define NAU_FVCO_MAX 124000000 #define NAU_FVCO_MIN 90000000 +static int nau8825_configure_sysclk(struct nau8825 *nau8825, + int clk_id, unsigned int freq); + struct nau8825_fll { int mclk_src; int ratio; @@ -368,9 +374,12 @@ static const struct snd_soc_dapm_widget nau8825_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("ADC Power", NAU8825_REG_ANALOG_ADC_2, 6, 0, NULL, 0), - /* ADC for button press detection */ - SND_SOC_DAPM_ADC("SAR", NULL, NAU8825_REG_SAR_CTRL, - NAU8825_SAR_ADC_EN_SFT, 0), + /* ADC for button press detection. A dapm supply widget is used to + * prevent dapm_power_widgets keeping the codec at SND_SOC_BIAS_ON + * during suspend. + */ + SND_SOC_DAPM_SUPPLY("SAR", NAU8825_REG_SAR_CTRL, + NAU8825_SAR_ADC_EN_SFT, 0, NULL, 0), SND_SOC_DAPM_PGA_S("ADACL", 2, NAU8825_REG_RDAC, 12, 0, NULL, 0), SND_SOC_DAPM_PGA_S("ADACR", 2, NAU8825_REG_RDAC, 13, 0, NULL, 0), @@ -614,8 +623,10 @@ int nau8825_enable_jack_detect(struct snd_soc_codec *codec, NAU8825_HSD_AUTO_MODE | NAU8825_SPKR_DWN1R | NAU8825_SPKR_DWN1L, NAU8825_HSD_AUTO_MODE | NAU8825_SPKR_DWN1R | NAU8825_SPKR_DWN1L); - regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, - NAU8825_IRQ_HEADSET_COMPLETE_EN | NAU8825_IRQ_EJECT_EN, 0); + /* Change jack type detection interruption to non-clock architecture + * for power saving less 1mW. Move these configuration about inter- + * ruption at auto mode to auto irq setup function. + */ return 0; } @@ -642,6 +653,22 @@ static void nau8825_restart_jack_detection(struct regmap *regmap) NAU8825_JACK_DET_RESTART, 0); } +static void nau8825_int_status_clear_all(struct regmap *regmap) +{ + int active_irq, clear_irq, i; + + /* Reset the intrruption status from rightmost bit if the corres- + * ponding irq event occurs. + */ + regmap_read(regmap, NAU8825_REG_IRQ_STATUS, &active_irq); + for (i = 0; i < NAU8825_REG_DATA_LEN; i++) { + clear_irq = (0x1 << i); + if (active_irq & clear_irq) + regmap_write(regmap, + NAU8825_REG_INT_CLR_KEY_STATUS, clear_irq); + } +} + static void nau8825_eject_jack(struct nau8825 *nau8825) { struct snd_soc_dapm_context *dapm = nau8825->dapm; @@ -656,6 +683,67 @@ static void nau8825_eject_jack(struct nau8825 *nau8825) regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 0xf, 0xf); snd_soc_dapm_sync(dapm); + + /* Clear all interruption status */ + nau8825_int_status_clear_all(regmap); + + /* Mask all interruptions except jack insertion interruption */ + regmap_write(regmap, NAU8825_REG_INTERRUPT_DIS_CTRL, 0xfffe); + + /* Enable insertion interruption only and bypass de-bounce circuit */ + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_OUTPUT_EN | NAU8825_IRQ_EJECT_EN | + NAU8825_IRQ_HEADSET_COMPLETE_EN | NAU8825_IRQ_INSERT_EN, + NAU8825_IRQ_OUTPUT_EN | NAU8825_IRQ_EJECT_EN | + NAU8825_IRQ_HEADSET_COMPLETE_EN); + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_DET_DB_BYPASS, NAU8825_JACK_DET_DB_BYPASS); + + /* Disable ADC needed for interruptions at audo mode */ + regmap_update_bits(regmap, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_ADC, 0); + + /* Close clock for jack type detection at manual mode */ + nau8825_configure_sysclk(nau8825, NAU8825_CLK_DIS, 0); +} + +/* Enable audo mode interruptions with internal clock. */ +static void nau8825_setup_auto_irq(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + + /* Enable headset jack type detection complete interruption and + * jack ejection interruption. + */ + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_HEADSET_COMPLETE_EN | NAU8825_IRQ_EJECT_EN, 0); + + /* Enable internal VCO needed for interruptions */ + nau8825_configure_sysclk(nau8825, NAU8825_CLK_INTERNAL, 0); + + /* Enable ADC needed for interruptions */ + regmap_update_bits(regmap, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_ADC, NAU8825_ENABLE_ADC); + + /* Chip needs one FSCLK cycle in order to generate interruptions, + * as we cannot guarantee one will be provided by the system. Turning + * master mode on then off enables us to generate that FSCLK cycle + * with a minimum of contention on the clock bus. + */ + regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_MASTER); + regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_SLAVE); + + /* Not bypass de-bounce circuit */ + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_DET_DB_BYPASS, 0); + + /* Unmask all interruptions */ + regmap_write(regmap, NAU8825_REG_INTERRUPT_DIS_CTRL, 0); + + /* Restart the jack detection process at auto mode */ + nau8825_restart_jack_detection(regmap); } static int nau8825_button_decode(int value) @@ -753,7 +841,10 @@ static irqreturn_t nau8825_interrupt(int irq, void *data) struct regmap *regmap = nau8825->regmap; int active_irq, clear_irq = 0, event = 0, event_mask = 0; - regmap_read(regmap, NAU8825_REG_IRQ_STATUS, &active_irq); + if (regmap_read(regmap, NAU8825_REG_IRQ_STATUS, &active_irq)) { + dev_err(nau8825->dev, "failed to clear interruption\n"); + return IRQ_NONE; + } if ((active_irq & NAU8825_JACK_EJECTION_IRQ_MASK) == NAU8825_JACK_EJECTION_DETECTED) { @@ -789,6 +880,26 @@ static irqreturn_t nau8825_interrupt(int irq, void *data) event_mask |= SND_JACK_HEADSET; clear_irq = NAU8825_HEADSET_COMPLETION_IRQ; + } else if ((active_irq & NAU8825_JACK_INSERTION_IRQ_MASK) == + NAU8825_JACK_INSERTION_DETECTED) { + /* One more step to check GPIO status directly. Thus, the + * driver can confirm the real insertion interruption because + * the intrruption at manual mode has bypassed debounce + * circuit which can get rid of unstable status. + */ + if (nau8825_is_jack_inserted(regmap)) { + /* Turn off insertion interruption at manual mode */ + regmap_update_bits(regmap, + NAU8825_REG_INTERRUPT_DIS_CTRL, + NAU8825_IRQ_INSERT_DIS, + NAU8825_IRQ_INSERT_DIS); + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_INSERT_EN, NAU8825_IRQ_INSERT_EN); + /* Enable interruption for jack type detection at audo + * mode which can detect microphone and jack type. + */ + nau8825_setup_auto_irq(nau8825); + } } if (!clear_irq) @@ -938,8 +1049,8 @@ static void nau8825_init_regs(struct nau8825 *nau8825) } static const struct regmap_config nau8825_regmap_config = { - .val_bits = 16, - .reg_bits = 16, + .val_bits = NAU8825_REG_DATA_LEN, + .reg_bits = NAU8825_REG_ADDR_LEN, .max_register = NAU8825_REG_MAX, .readable_reg = nau8825_readable_reg, @@ -958,11 +1069,10 @@ static int nau8825_codec_probe(struct snd_soc_codec *codec) nau8825->dapm = dapm; - /* Unmask interruptions. Handler uses dapm object so we can enable - * interruptions only after dapm is fully initialized. + /* Change jack type detection interruption to non-clock architecture + * for power saving less 1mW. Move these configuration about inter- + * ruption at auto mode to auto irq setup function. */ - regmap_write(nau8825->regmap, NAU8825_REG_INTERRUPT_DIS_CTRL, 0); - nau8825_restart_jack_detection(nau8825->regmap); return 0; } @@ -1134,7 +1244,26 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, struct regmap *regmap = nau8825->regmap; int ret; + if (!nau8825_is_jack_inserted(nau8825->regmap) && + clk_id != NAU8825_CLK_DIS) { + /* For power saving less 1mW, the jack type detection inter- + * ruption changes to non-clock architecture. Therefore, the + * clock should be disabled and not allowed to config any clock + * when no headset connected. + */ + dev_warn(nau8825->dev, "Souldn't enable any clock when no headset connected\n"); + return 0; + } + switch (clk_id) { + case NAU8825_CLK_DIS: + /* Clock provided externally and disable internal VCO clock */ + regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, + NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_MCLK); + regmap_update_bits(regmap, NAU8825_REG_FLL6, + NAU8825_DCO_EN, 0); + + break; case NAU8825_CLK_MCLK: regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_MCLK); @@ -1209,6 +1338,30 @@ static int nau8825_set_sysclk(struct snd_soc_codec *codec, int clk_id, return nau8825_configure_sysclk(nau8825, clk_id, freq); } +static int nau8825_resume_setup(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + + /* Close clock when jack type detection at manual mode */ + nau8825_configure_sysclk(nau8825, NAU8825_CLK_DIS, 0); + + /* Clear all interruption status */ + nau8825_int_status_clear_all(regmap); + + /* Enable insertion interruption only and bypass de-bounce circuit */ + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_OUTPUT_EN | NAU8825_IRQ_EJECT_EN | + NAU8825_IRQ_HEADSET_COMPLETE_EN | NAU8825_IRQ_INSERT_EN, + NAU8825_IRQ_OUTPUT_EN | NAU8825_IRQ_EJECT_EN | + NAU8825_IRQ_HEADSET_COMPLETE_EN); + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_DET_DB_BYPASS, NAU8825_JACK_DET_DB_BYPASS); + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_DIS_CTRL, + NAU8825_IRQ_INSERT_DIS, 0); + + return 0; +} + static int nau8825_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -1238,11 +1391,22 @@ static int nau8825_set_bias_level(struct snd_soc_codec *codec, "Failed to sync cache: %d\n", ret); return ret; } + + /* Setup codec configuration after resume */ + nau8825_resume_setup(nau8825); } break; case SND_SOC_BIAS_OFF: + /* Turn off all interruptions before system shutdown. Keep the + * interruption quiet before resume setup completes. + */ + regmap_write(nau8825->regmap, + NAU8825_REG_INTERRUPT_DIS_CTRL, 0xffff); + /* Disable ADC needed for interruptions at audo mode */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_ADC, 0); if (nau8825->mclk_freq) clk_disable_unprepare(nau8825->mclk); @@ -1252,12 +1416,50 @@ static int nau8825_set_bias_level(struct snd_soc_codec *codec, return 0; } +#ifdef CONFIG_PM_SLEEP +static int nau8825_codec_suspend(struct snd_soc_codec *codec) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + disable_irq(nau8825->irq); + snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_OFF); + regcache_cache_only(nau8825->regmap, true); + + return 0; +} + +static int nau8825_codec_resume(struct snd_soc_codec *codec) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(nau8825->regmap, false); + if (!nau8825_is_jack_inserted(nau8825->regmap)) { + /* Check the jack plug status directly. If the headset is un- + * plugged during S3 when the chip has no power, there will + * be no jack detection irq event after restarting jack detec- + * tion, because the chip just thinks no headset has ever + * been plugged in. + */ + nau8825_eject_jack(nau8825); + snd_soc_jack_report(nau8825->jack, 0, SND_JACK_HEADSET); + } + enable_irq(nau8825->irq); + + return 0; +} +#else +#define nau8825_codec_suspend NULL +#define nau8825_codec_resume NULL +#endif + static struct snd_soc_codec_driver nau8825_codec_driver = { .probe = nau8825_codec_probe, .set_sysclk = nau8825_set_sysclk, .set_pll = nau8825_set_pll, .set_bias_level = nau8825_set_bias_level, .suspend_bias_off = true, + .suspend = nau8825_codec_suspend, + .resume = nau8825_codec_resume, .controls = nau8825_controls, .num_controls = ARRAY_SIZE(nau8825_controls), @@ -1351,29 +1553,13 @@ static int nau8825_read_device_properties(struct device *dev, static int nau8825_setup_irq(struct nau8825 *nau8825) { - struct regmap *regmap = nau8825->regmap; int ret; - /* IRQ Output Enable */ - regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, - NAU8825_IRQ_OUTPUT_EN, NAU8825_IRQ_OUTPUT_EN); - - /* Enable internal VCO needed for interruptions */ - nau8825_configure_sysclk(nau8825, NAU8825_CLK_INTERNAL, 0); - - /* Enable ADC needed for interrupts */ - regmap_update_bits(regmap, NAU8825_REG_ENA_CTRL, - NAU8825_ENABLE_ADC, NAU8825_ENABLE_ADC); - - /* Chip needs one FSCLK cycle in order to generate interrupts, - * as we cannot guarantee one will be provided by the system. Turning - * master mode on then off enables us to generate that FSCLK cycle - * with a minimum of contention on the clock bus. + /* Change jack type detection interruption to non-clock architecture + * for power saving less 1mW. Because resume and initiation all need + * to enable the mechanism, move these configuration to resume setup + * function. */ - regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, - NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_MASTER); - regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, - NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_SLAVE); ret = devm_request_threaded_irq(nau8825->dev, nau8825->irq, NULL, nau8825_interrupt, IRQF_TRIGGER_LOW | IRQF_ONESHOT, @@ -1442,36 +1628,6 @@ static int nau8825_i2c_remove(struct i2c_client *client) return 0; } -#ifdef CONFIG_PM_SLEEP -static int nau8825_suspend(struct device *dev) -{ - struct i2c_client *client = to_i2c_client(dev); - struct nau8825 *nau8825 = dev_get_drvdata(dev); - - disable_irq(client->irq); - regcache_cache_only(nau8825->regmap, true); - regcache_mark_dirty(nau8825->regmap); - - return 0; -} - -static int nau8825_resume(struct device *dev) -{ - struct i2c_client *client = to_i2c_client(dev); - struct nau8825 *nau8825 = dev_get_drvdata(dev); - - regcache_cache_only(nau8825->regmap, false); - regcache_sync(nau8825->regmap); - enable_irq(client->irq); - - return 0; -} -#endif - -static const struct dev_pm_ops nau8825_pm = { - SET_SYSTEM_SLEEP_PM_OPS(nau8825_suspend, nau8825_resume) -}; - static const struct i2c_device_id nau8825_i2c_ids[] = { { "nau8825", 0 }, { } @@ -1498,7 +1654,6 @@ static struct i2c_driver nau8825_driver = { .name = "nau8825", .of_match_table = of_match_ptr(nau8825_of_ids), .acpi_match_table = ACPI_PTR(nau8825_acpi_match), - .pm = &nau8825_pm, }, .probe = nau8825_i2c_probe, .remove = nau8825_i2c_remove, diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index 9e6cb62..1100385 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -93,6 +93,9 @@ #define NAU8825_REG_CHARGE_PUMP_INPUT_READ 0x81 #define NAU8825_REG_GENERAL_STATUS 0x82 #define NAU8825_REG_MAX NAU8825_REG_GENERAL_STATUS +/* 16-bit control register address, and 16-bits control register data */ +#define NAU8825_REG_ADDR_LEN 16 +#define NAU8825_REG_DATA_LEN 16 /* ENA_CTRL (0x1) */ #define NAU8825_ENABLE_DACR_SFT 10 @@ -145,6 +148,7 @@ /* JACK_DET_CTRL (0xd) */ #define NAU8825_JACK_DET_RESTART (1 << 9) +#define NAU8825_JACK_DET_DB_BYPASS (1 << 8) #define NAU8825_JACK_INSERT_DEBOUNCE_SFT 5 #define NAU8825_JACK_INSERT_DEBOUNCE_MASK (0x7 << NAU8825_JACK_INSERT_DEBOUNCE_SFT) #define NAU8825_JACK_EJECT_DEBOUNCE_SFT 2 @@ -157,6 +161,7 @@ #define NAU8825_IRQ_KEY_RELEASE_EN (1 << 7) #define NAU8825_IRQ_KEY_SHORT_PRESS_EN (1 << 5) #define NAU8825_IRQ_EJECT_EN (1 << 2) +#define NAU8825_IRQ_INSERT_EN (1 << 0) /* IRQ_STATUS (0x10) */ #define NAU8825_HEADSET_COMPLETION_IRQ (1 << 10) @@ -177,6 +182,7 @@ #define NAU8825_IRQ_KEY_RELEASE_DIS (1 << 7) #define NAU8825_IRQ_KEY_SHORT_PRESS_DIS (1 << 5) #define NAU8825_IRQ_EJECT_DIS (1 << 2) +#define NAU8825_IRQ_INSERT_DIS (1 << 0) /* SAR_CTRL (0x13) */ #define NAU8825_SAR_ADC_EN_SFT 12 @@ -333,7 +339,8 @@ /* System Clock Source */ enum { - NAU8825_CLK_MCLK = 0, + NAU8825_CLK_DIS = 0, + NAU8825_CLK_MCLK, NAU8825_CLK_INTERNAL, NAU8825_CLK_FLL_MCLK, NAU8825_CLK_FLL_BLK, -- 2.6.4

2 12

[alsa-devel] i.MX6 SGTL5000 hangs with EIO error
by Ross Wille 12 May '16

12 May '16

Hello, I am doing an ALSA capture on a Wandboard Quad (i.MX6 quad core with SGTL5000 codec chip) from the LINEIN1 jack. I'm using a 4.6.0-rc3-armv7-x0 kernel. Sometimes when the audio glitches (for example, when I plug/unplug the audio cable or adjust my signal generator) the snd_pcm_readi() function will start returning -5 (EIO). Once this happens, the only way I've been able to recover is to reboot the computer. Calling snd_pcm_close(), snd_pcm_prepare(), snd_pcm_start(), etc. doesn't help. When in this state, running arecord returns IO errors as well. It's interesting that, on rare occasions, I must do a power cycle in order to recover. When a reboot is not effective I've noticed that the capture device doesn't appear in /proc/asound/devices. I don't believe my specific ALSA settings are important, but I'm calling snd_pcm_readi() with ALSA set to a sample rate of 48000, format of S16_LE, channels=1, frames per period of 960 (20 mS periods), and 4 periods per buffer. This same problem happens on two different Wandboards, so I don't think it's a defective board or chip. It has happened on older kernels as well. Any ideas? Thank you! Wille

1 0

[alsa-devel] Intel Cherry Trail and NAU8824 codec
by TheKit 11 May '16

11 May '16

Hello, I'm trying to get sound working in Linux on Cube iWork8 Ultimate tablet. It has Windows/Android dualboot, but no kernel sources for Android, unfortunately. From dmesg I see that NAU8824 codec is used, which I can't find any drivers for, except proposed patch that wasn't merged: https://lkml.org/lkml/2015/3/4/765 But, as I understand, additional code (like https://github.com/kirananto/ZENFONE2/blob/master/sound/soc/intel/board/cht…) is needed to bridge it with Cherry Trail SoC. Any suggestions how to go about this? Relevant lines of dmesg from stock Android kernel (where sound works): <6>[ 8.665477] nau8824_i2c_probe enter <3>[ 8.666609] snd_intel_sst: sst_platform_get_resources_edk <3>[ 8.672489] snd_intel_sst: LPE base: 0x91600000 size:0x200000snd_intel_sst: IRAM base: 0x916c0000 <3>[ 8.682557] snd_intel_sst: DRAM base: 0x91700000snd_intel_sst: SHIM base: 0x91740000 <3>[ 8.691298] snd_intel_sst: Mailbox base: 0x91744000snd_intel_sst: DDR base: 0x20000000 <6>[ 8.700428] snd_intel_sst: Enter: sst_audio_platform_init <6>[ 8.700646] snd_intel_sst: audio:set_cht_platform_config <6>[ 8.700900] snd_soc_sst_platform: sst_platform_probe is deferred <6>[ 8.700918] platform sst-platform: Driver sst-platform requests probe deferral <3>[ 8.700927] snd_intel_sst: board_name:Cherry Trail CR <6>[ 8.706611] snd_intel_sst: Registering machine device cht_nau88l24 <3>[ 8.707127] snd_intel_sst: Requesting FW fw_sst_22a8.bin now... <4>[ 8.713815] intel_sst_acpi 808622A8:00: Direct firmware load failed with error -2 <4>[ 8.724734] intel_sst_acpi 808622A8:00: Falling back to user helper <6>[ 8.789741] platform cht_nau88l24: Driver cht_nau88l24 requests probe deferral <6>[ 8.789967] cht_cr_nau88l24: Cherrytrail Machine Driver cht_nau8824 registerd <6>[ 9.077571] snd_soc_sst_platform: sst_platform_probe is deferred <3>[ 10.856689] sst-platform sst-platform: ASoC: no sink widget found for Audio Capture <3>[ 10.856693] sst-platform sst-platform: ASoC: Failed to add route media3_out -> direct -> Audio Capture <6>[ 10.873799] nau8824_codec_probe enter <6>[ 11.062658] cht_nau88l24 cht_nau88l24: snd-soc-dummy-dai <-> Headset-cpu-dai mapping ok <6>[ 11.062754] cht_nau88l24 cht_nau88l24: snd-soc-dummy-dai <-> Deepbuffer-cpu-dai mapping ok <6>[ 11.062902] cht_nau88l24 cht_nau88l24: snd-soc-dummy-dai <-> Voip-cpu-dai mapping ok <6>[ 11.063035] cht_cr_nau88l24: cht_hs_detection:Check hs insertion after 500 msec <6>[ 11.063035] cht_nau88l24 cht_nau88l24: snd-soc-dummy-dai <-> Lowlatency-cpu-dai mapping ok <6>[ 11.063383] cht_nau88l24 cht_nau88l24: snd-soc-dummy-dai <-> Probe-cpu-dai mapping ok <6>[ 11.063596] cht_nau88l24 cht_nau88l24: nau8824-aif1 <-> ssp2-port mapping ok <6>[ 11.063638] cht_nau88l24 cht_nau88l24: snd-soc-dummy-dai <-> snd-soc-dummy-dai mapping ok <6>[ 11.063674] cht_nau88l24 cht_nau88l24: snd-soc-dummy-dai <-> snd-soc-dummy-dai mapping ok <3>[ 11.063742] cht_nau88l24 cht_nau88l24: ASoC: Failed to add Headset Mic Switch: -16 <6>[ 11.095628] input: cherrytrailaud Headset Jack as /devices/platform/cht_nau88l24/sound/card1/input9 <6>[ 11.096265] cht_cr_nau88l24: snd_cht_mc_probe successful

2 1

[alsa-devel] [PATCH] ALSA: firewire-lib: add context information to tracepoints
by Takashi Sakamoto 11 May '16

11 May '16

In current implementation, packet processing is done in both of software IRQ contexts of IR/IT contexts and process contexts. This is usual interrupt handling of IR/IT context for 1394 OHCI. (in hardware IRQ context) irq_handler() (drivers/firewire/ohci.c) ->tasklet_schedule() (in software IRQ context) handle_it_packet() or handle_ir_packet_per_buffer() (drivers/firewire/ohci.c) ->flush_iso_completions() ->struct fw_iso_context.callback.sc() = out_stream_callback() or in_stream_callback() However, we have another chance for packet processing. It's done in PCM frame handling via ALSA PCM interfaces. (in process context) ioctl(i.e. SNDRV_PCM_IOCTL_HWSYNC) ->snd_pcm_hwsync() (sound/core/pcm_native.c) ->snd_pcm_update_hw_ptr() (sound/core/pcm_lib.c) ->snd_pcm_update_hw_ptr0() ->struct snd_pcm_ops.pointer() = amdtp_stream_pcm_pointer() ->fw_iso_context_flush_completions() (drivers/firewire/core-iso.c) ->struct fw_card_driver.flush_iso_completions() = ohci_flush_iso_completions() (drivers/firewire/ohci.c) ->flush_iso_completions() ->struct fw_iso_context.callback.sc() = out_stream_callback() or in_stream_callback() This design is for a better granularity of PCM pointer. When ioctl(2) is executed with some commands for ALSA PCM interface, queued packets are handled at first. Then, the latest number of handled PCM frames is reported. The number can represent PCM frames transferred in most near isochronous cycle. Current tracepoints include no information to distinguish running contexts. When tracing the interval of software IRQ context, this is not good. This commit adds more information for current context. Additionally, the index of packet processed in one context is added in a case that packet processing is executed in continuous context of the same kind, As a result, the output includes 11 fields with additional two fields to commit 0c95c1d6197f ("ALSA: firewire-lib: add tracepoints to dump a part of isochronous packet data"): 17131.9186: out_packet: 07 7494 ffc0 ffc1 00 000700c0 9001a496 058 45 1 13 17131.9186: out_packet: 07 7495 ffc0 ffc1 00 000700c8 9001ba00 058 46 1 14 17131.9186: out_packet: 07 7496 ffc0 ffc1 00 000700d0 9001ffff 002 47 1 15 17131.9189: out_packet: 07 7497 ffc0 ffc1 00 000700d0 9001d36a 058 00 0 00 17131.9189: out_packet: 07 7498 ffc0 ffc1 00 000700d8 9001e8d4 058 01 0 01 17131.9189: out_packet: 07 7499 ffc0 ffc1 00 000700e0 9001023e 058 02 0 00 17131.9206: in_packet: 07 7447 ffc1 ffc0 01 3f070072 9001783d 058 32 1 00 17131.9206: in_packet: 07 7448 ffc1 ffc0 01 3f070072 90ffffff 002 33 1 01 17131.9206: in_packet: 07 7449 ffc1 ffc0 01 3f07007a 900191a8 058 34 1 02 (Here, some common fields are omitted so that a line is within 80 characters.) The legend is: - The second of cycle scheduled for the packet - The count of cycle scheduled for the packet - The ID of node as source (hex) - The ID of node as destination (hex) - The value of isochronous channel - The first quadlet of CIP header (hex) - The second quadlet of CIP header (hex) - The number of included quadlets - The index of packet in a buffer maintained by this module - 0 in process context, 1 in IRQ context - The index of packet processed in the context Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp-stream-trace.h | 28 ++++++++++++++++++++-------- sound/firewire/amdtp-stream.c | 14 ++++++++------ 2 files changed, 28 insertions(+), 14 deletions(-) diff --git a/sound/firewire/amdtp-stream-trace.h b/sound/firewire/amdtp-stream-trace.h index 425d1d7..1622579 100644 --- a/sound/firewire/amdtp-stream-trace.h +++ b/sound/firewire/amdtp-stream-trace.h @@ -14,8 +14,8 @@ #include <linux/tracepoint.h> TRACE_EVENT(in_packet, - TP_PROTO(const struct amdtp_stream *s, u32 cycles, u32 cip_header[2], unsigned int payload_quadlets), - TP_ARGS(s, cycles, cip_header, payload_quadlets), + TP_PROTO(const struct amdtp_stream *s, u32 cycles, u32 cip_header[2], unsigned int payload_quadlets, unsigned int index), + TP_ARGS(s, cycles, cip_header, payload_quadlets, index), TP_STRUCT__entry( __field(unsigned int, second) __field(unsigned int, cycle) @@ -25,6 +25,8 @@ TRACE_EVENT(in_packet, __field(u32, cip_header0) __field(u32, cip_header1) __field(unsigned int, payload_quadlets) + __field(unsigned int, packet_index) + __field(bool, irq) __field(unsigned int, index) ), TP_fast_assign( @@ -36,10 +38,12 @@ TRACE_EVENT(in_packet, __entry->cip_header0 = cip_header[0]; __entry->cip_header1 = cip_header[1]; __entry->payload_quadlets = payload_quadlets; - __entry->index = s->packet_index; + __entry->packet_index = s->packet_index; + __entry->irq = in_interrupt(); + __entry->index = index; ), TP_printk( - "%02u %04u %04x %04x %02d %08x %08x %03u %02u", + "%02u %04u %04x %04x %02d %08x %08x %03u %02u %01u %02u", __entry->second, __entry->cycle, __entry->src, @@ -48,12 +52,14 @@ TRACE_EVENT(in_packet, __entry->cip_header0, __entry->cip_header1, __entry->payload_quadlets, + __entry->packet_index, + __entry->irq, __entry->index) ); TRACE_EVENT(out_packet, - TP_PROTO(const struct amdtp_stream *s, u32 cycles, __be32 *cip_header, unsigned int payload_length), - TP_ARGS(s, cycles, cip_header, payload_length), + TP_PROTO(const struct amdtp_stream *s, u32 cycles, __be32 *cip_header, unsigned int payload_length, unsigned int index), + TP_ARGS(s, cycles, cip_header, payload_length, index), TP_STRUCT__entry( __field(unsigned int, second) __field(unsigned int, cycle) @@ -63,6 +69,8 @@ TRACE_EVENT(out_packet, __field(u32, cip_header0) __field(u32, cip_header1) __field(unsigned int, payload_quadlets) + __field(unsigned int, packet_index) + __field(bool, irq) __field(unsigned int, index) ), TP_fast_assign( @@ -74,10 +82,12 @@ TRACE_EVENT(out_packet, __entry->cip_header0 = be32_to_cpu(cip_header[0]); __entry->cip_header1 = be32_to_cpu(cip_header[1]); __entry->payload_quadlets = payload_length / 4; - __entry->index = s->packet_index; + __entry->packet_index = s->packet_index; + __entry->irq = in_interrupt(); + __entry->index = index; ), TP_printk( - "%02u %04u %04x %04x %02d %08x %08x %03u %02u", + "%02u %04u %04x %04x %02d %08x %08x %03u %02u %01u %02u", __entry->second, __entry->cycle, __entry->src, @@ -86,6 +96,8 @@ TRACE_EVENT(out_packet, __entry->cip_header0, __entry->cip_header1, __entry->payload_quadlets, + __entry->packet_index, + __entry->irq, __entry->index) ); diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index e7163bd..db30fac4 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -411,7 +411,8 @@ static inline int queue_in_packet(struct amdtp_stream *s) amdtp_stream_get_max_payload(s)); } -static int handle_out_packet(struct amdtp_stream *s, unsigned int cycle) +static int handle_out_packet(struct amdtp_stream *s, unsigned int cycle, + unsigned int index) { __be32 *buffer; unsigned int syt; @@ -436,7 +437,7 @@ static int handle_out_packet(struct amdtp_stream *s, unsigned int cycle) s->data_block_counter = (s->data_block_counter + data_blocks) & 0xff; payload_length = 8 + data_blocks * 4 * s->data_block_quadlets; - trace_out_packet(s, cycle, buffer, payload_length); + trace_out_packet(s, cycle, buffer, payload_length, index); if (queue_out_packet(s, payload_length) < 0) return -EIO; @@ -450,7 +451,8 @@ static int handle_out_packet(struct amdtp_stream *s, unsigned int cycle) } static int handle_in_packet(struct amdtp_stream *s, - unsigned int payload_quadlets, unsigned int cycle) + unsigned int payload_quadlets, unsigned int cycle, + unsigned int index) { __be32 *buffer; u32 cip_header[2]; @@ -465,7 +467,7 @@ static int handle_in_packet(struct amdtp_stream *s, cip_header[0] = be32_to_cpu(buffer[0]); cip_header[1] = be32_to_cpu(buffer[1]); - trace_in_packet(s, cycle, cip_header, payload_quadlets); + trace_in_packet(s, cycle, cip_header, payload_quadlets, index); /* * This module supports 'Two-quadlet CIP header with SYT field'. @@ -604,7 +606,7 @@ static void out_stream_callback(struct fw_iso_context *context, u32 tstamp, for (i = 0; i < packets; ++i) { cycle = increment_cycle_count(cycle, 1); - if (handle_out_packet(s, cycle) < 0) { + if (handle_out_packet(s, cycle, i) < 0) { s->packet_index = -1; amdtp_stream_pcm_abort(s); return; @@ -651,7 +653,7 @@ static void in_stream_callback(struct fw_iso_context *context, u32 tstamp, break; } - if (handle_in_packet(s, payload_quadlets, cycle) < 0) + if (handle_in_packet(s, payload_quadlets, cycle, i) < 0) break; } -- 2.7.4

2 1

[alsa-devel] [PATCH] ALSA: firewire-lib: drop skip argument from helper functions to queue a packet
by Takashi Sakamoto 11 May '16

11 May '16

On most of audio and music units on IEEE 1394 bus which ALSA firewire stack supports (or plans to support), CIP with two quadlets header is used. Thus, there's no cases to queue packets with blank payload. If such packets are going to be queued, it means that they're for skips of the cycle. This commit simplifies helper functions to queue a packet. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp-stream.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 830a95c..e7163bd 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -370,9 +370,8 @@ static void pcm_period_tasklet(unsigned long data) snd_pcm_period_elapsed(pcm); } -static int queue_packet(struct amdtp_stream *s, - unsigned int header_length, - unsigned int payload_length, bool skip) +static int queue_packet(struct amdtp_stream *s, unsigned int header_length, + unsigned int payload_length) { struct fw_iso_packet p = {0}; int err = 0; @@ -383,8 +382,10 @@ static int queue_packet(struct amdtp_stream *s, p.interrupt = IS_ALIGNED(s->packet_index + 1, INTERRUPT_INTERVAL); p.tag = TAG_CIP; p.header_length = header_length; - p.payload_length = (!skip) ? payload_length : 0; - p.skip = skip; + if (payload_length > 0) + p.payload_length = payload_length; + else + p.skip = true; err = fw_iso_context_queue(s->context, &p, &s->buffer.iso_buffer, s->buffer.packets[s->packet_index].offset); if (err < 0) { @@ -399,16 +400,15 @@ end: } static inline int queue_out_packet(struct amdtp_stream *s, - unsigned int payload_length, bool skip) + unsigned int payload_length) { - return queue_packet(s, OUT_PACKET_HEADER_SIZE, - payload_length, skip); + return queue_packet(s, OUT_PACKET_HEADER_SIZE, payload_length); } static inline int queue_in_packet(struct amdtp_stream *s) { return queue_packet(s, IN_PACKET_HEADER_SIZE, - amdtp_stream_get_max_payload(s), false); + amdtp_stream_get_max_payload(s)); } static int handle_out_packet(struct amdtp_stream *s, unsigned int cycle) @@ -438,7 +438,7 @@ static int handle_out_packet(struct amdtp_stream *s, unsigned int cycle) trace_out_packet(s, cycle, buffer, payload_length); - if (queue_out_packet(s, payload_length, false) < 0) + if (queue_out_packet(s, payload_length) < 0) return -EIO; pcm = ACCESS_ONCE(s->pcm); @@ -764,7 +764,7 @@ int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed) if (s->direction == AMDTP_IN_STREAM) err = queue_in_packet(s); else - err = queue_out_packet(s, 0, true); + err = queue_out_packet(s, 0); if (err < 0) goto err_context; } while (s->packet_index > 0); -- 2.7.4

3 3

[alsa-devel] [RFC][PATCH] ALSA: firewire-lib: permit process context only to flush queued packets for better PCM period granularity
by Takashi Sakamoto 11 May '16

11 May '16

Hi Clemens, Could I request your comment to this patch? It is to solve a race condition, but the condition is quite rare. Practically, it might have a less meanings, except for better program. And I think there's another race condition against processing each packets by calling out/in_stream_callback(), but I cannot observe the race. Software IRQ contexts of IR/IT contexts and process contexts are under the race condition, however I can see no problems related to it in my several trials in multi-core machine. I have no idea about the reason that packet sequence is processed correctly between the software IRQ contexts and the process contexts without any lock primitives. Do you have some ideas about it? Regards Takashi Sakamoto ----- 8< ----- Subject: ALSA: firewire-lib: permit process context only to flush queued packets for better PCM period granularity These three commits were merged to improve PCM pointer granularity. commit 76fb87894828 ("ALSA: firewire-lib: taskletize the snd_pcm_period_elapsed() call") commit e9148dddc3c7 ("ALSA: firewire-lib: flush completed packets when reading PCM position") commit 92b862c7d685 ("ALSA: firewire-lib: optimize packet flushing") The point of them is to handle queued packets not only in software IRQ context of IR/IT contexts, but also in process context. This idea introduced a cyclic call of 'struct snd_pcm_ops.pointer()'. The last commit adds 'pointer_flush' member to 'struct amdtp_stream' to avoid the situation. On the other hand, This solution is weak at race condition between the process context and the software IRQ context of IR/IT contexts. Practically, this race is not so critical because it influences process context to skip flushing queued packets and to get worse granularity of PCM pointer. This is quite rare because 'struct snd_pcm_ops.pointer()' is frequently called by process contexts while IT/IR contexts run less frequently. However, it's better to solve the race. The similar solution can be achieved by 'in_interrupt()' macro. This commit applies the macro to solve the race condition against 'pointer_flush'. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp-stream.c | 23 +++++++++++++++++------ sound/firewire/amdtp-stream.h | 1 - 2 files changed, 17 insertions(+), 7 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 4484242..132c7f4 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -248,7 +248,6 @@ void amdtp_stream_pcm_prepare(struct amdtp_stream *s) tasklet_kill(&s->period_tasklet); s->pcm_buffer_pointer = 0; s->pcm_period_pointer = 0; - s->pointer_flush = true; } EXPORT_SYMBOL(amdtp_stream_pcm_prepare); @@ -353,7 +352,6 @@ static void update_pcm_pointers(struct amdtp_stream *s, s->pcm_period_pointer += frames; if (s->pcm_period_pointer >= pcm->runtime->period_size) { s->pcm_period_pointer -= pcm->runtime->period_size; - s->pointer_flush = false; tasklet_hi_schedule(&s->period_tasklet); } } @@ -798,11 +796,24 @@ EXPORT_SYMBOL(amdtp_stream_start); */ unsigned long amdtp_stream_pcm_pointer(struct amdtp_stream *s) { - /* this optimization is allowed to be racy */ - if (s->pointer_flush && amdtp_stream_running(s)) + /* + * This function is called in software IRQ context of period_tasklet or + * process context. + * + * When the software IRQ context was scheduled by software IRQ context + * of IR/IT contexts, queued packets were already handled. Therefore, + * no need to flush the queue in buffer anymore. + * + * When the process context reach here, some packets wll be queued in + * the buffer. These packets should be handled immediately to keep + * better granularity of PCM pointer. + * + * Later, the process context will sometimes schedules software IRQ + * context of the period_tasklet. Then, no need to flush the queue by + * the same reason as described for IR/IT contexts. + */ + if (!in_interrupt() && amdtp_stream_running(s)) fw_iso_context_flush_completions(s->context); - else - s->pointer_flush = true; return ACCESS_ONCE(s->pcm_buffer_pointer); } diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h index 8775704..ab697cc 100644 --- a/sound/firewire/amdtp-stream.h +++ b/sound/firewire/amdtp-stream.h @@ -132,7 +132,6 @@ struct amdtp_stream { struct tasklet_struct period_tasklet; unsigned int pcm_buffer_pointer; unsigned int pcm_period_pointer; - bool pointer_flush; /* To wait for first packet. */ bool callbacked; -- 2.7.4

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[alsa-devel] [PATCH] ALSA: usb-audio: Yet another Phoneix Audio device quirk
by Takashi Iwai 11 May '16

11 May '16

Phoenix Audio has yet another device with another id (even a different vendor id, 0556:0014) that requires the same quirk for the sample rate. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=110221 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 7b6dca2d7488..220764cd60bd 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1137,6 +1137,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) case USB_ID(0x047F, 0x0415): /* Plantronics BT-300 */ case USB_ID(0x047F, 0xAA05): /* Plantronics DA45 */ case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ + case uSB_ID(0x0556, 0x0014): /* Phoenix Audio TMX320VC */ case USB_ID(0x074D, 0x3553): /* Outlaw RR2150 (Micronas UAC3553B) */ case USB_ID(0x1de7, 0x0013): /* Phoenix Audio MT202exe */ case USB_ID(0x1de7, 0x0014): /* Phoenix Audio TMX320 */ -- 2.8.2

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[alsa-devel] [PATCH] ASoC: twl6040: Select LPPLL during standby
by Peter Ujfalusi 11 May '16

11 May '16

When the codec is in standby we do not need to keep the HPPLL active. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/twl6040.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index bc3de2e844e6..7a744b581cc1 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -824,7 +824,7 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec, { struct twl6040 *twl6040 = codec->control_data; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); - int ret; + int ret = 0; switch (level) { case SND_SOC_BIAS_ON: @@ -832,12 +832,16 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - if (priv->codec_powered) + if (priv->codec_powered) { + /* Select low power PLL in standby */ + ret = twl6040_set_pll(twl6040, TWL6040_SYSCLK_SEL_LPPLL, + 32768, 19200000); break; + } ret = twl6040_power(twl6040, 1); if (ret) - return ret; + break; priv->codec_powered = 1; @@ -853,7 +857,7 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec, break; } - return 0; + return ret; } static int twl6040_startup(struct snd_pcm_substream *substream, -- 2.8.2

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[alsa-devel] [PATCH] ASoC: Intel: Fix printk formatting
by Joonas Lahtinen 11 May '16

11 May '16

Format number after 0x in hex. Cc: Jie Yang <yang.jie(a)linux.intel.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Signed-off-by: Joonas Lahtinen <joonas.lahtinen(a)linux.intel.com> --- sound/soc/intel/haswell/sst-haswell-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/haswell/sst-haswell-pcm.c b/sound/soc/intel/haswell/sst-haswell-pcm.c index 1aa819c..994256b 100644 --- a/sound/soc/intel/haswell/sst-haswell-pcm.c +++ b/sound/soc/intel/haswell/sst-haswell-pcm.c @@ -445,7 +445,7 @@ static int create_adsp_page_table(struct snd_pcm_substream *substream, pages = snd_sgbuf_aligned_pages(size); - dev_dbg(rtd->dev, "generating page table for %p size 0x%zu pages %d\n", + dev_dbg(rtd->dev, "generating page table for %p size 0x%zx pages %d\n", dma_area, size, pages); for (i = 0; i < pages; i++) { -- 2.5.5

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[alsa-devel] [PATCH RFT] ASoC: max98371 Remove duplicate entry in max98371_reg
by Axel Lin 11 May '16

11 May '16

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/max98371.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/max98371.c b/sound/soc/codecs/max98371.c index 21fae09..cf0a39b 100644 --- a/sound/soc/codecs/max98371.c +++ b/sound/soc/codecs/max98371.c @@ -107,7 +107,6 @@ static const struct reg_default max98371_reg[] = { { 0x39, 0x00 }, { 0x3A, 0x00 }, { 0x3B, 0x00 }, - { 0x3B, 0x00 }, { 0x3C, 0x00 }, { 0x3D, 0x00 }, { 0x3E, 0x00 }, -- 2.5.0

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[alsa-devel] [PATCH v5 0/4] ASoC: davinci-mcbsp: add binding for McBSP
by Petr Kulhavy 11 May '16

11 May '16

This series of patches adds devicetree support for the davinci McBSP audio interface. Changes in version 5 of the patch: * rebased to the current master of git://git.kernel.org/pub/scm/linux/kernel/git/nsekhar/linux-davinci.git * independent changes submitted separately (3/7 and 7/7 of the version 4) * 5/7 of the version 4 removed as already applied * corrected mcbsp0 and mcbsp1 nodes in the DTS file: using correct address and sound@ prefix

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[alsa-devel] [PATCH alsa-lib] pcm: dmix: Fix doubly resume of slave PCM
by Takashi Iwai 11 May '16

11 May '16

The dmix plugin and co may trigger the resume for each instance in snd_pcm_direct_resume(). It means that the slave PCM gets resumed or re-prepared/started by each opened dmix stream, and this may end up with the doubly triggers even though the slave PCM has been already resumed by another dmix stream. For avoiding this conflicts, check the slave PCM state and resume only when it's still in the suspended state. Meanwhile we keep the shadow state updated no matter whether the slave was triggered or not. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- src/pcm/pcm_direct.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index e28738b0de96..ac082f1a73b2 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -841,6 +841,12 @@ int snd_pcm_direct_resume(snd_pcm_t *pcm) int err; snd_pcm_direct_semaphore_down(dmix, DIRECT_IPC_SEM_CLIENT); + /* resume only when the slave PCM is still in suspended state */ + if (snd_pcm_state(dmix->spcm) != SND_PCM_STATE_SUSPENDED) { + err = 0; + goto out; + } + err = snd_pcm_resume(dmix->spcm); if (err == -ENOSYS) { /* FIXME: error handling? */ @@ -848,6 +854,7 @@ int snd_pcm_direct_resume(snd_pcm_t *pcm) snd_pcm_start(dmix->spcm); err = 0; } + out: dmix->state = snd_pcm_state(dmix->spcm); snd_pcm_direct_semaphore_up(dmix, DIRECT_IPC_SEM_CLIENT); return err; -- 2.8.2

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[alsa-devel] ASoC: DAPM and differential inputs
by Jeremy McDermond 10 May '16

10 May '16

I’ve been working on the tlv320aic32x4 driver for a while now to add some features for some projects I’m working on using the chip. I’m trying to wrap my head around DAPM and getting it to work with the variety of inputs on the aic32x4. Basically, the codec has a set of registers that control the positive and negative inputs to each of the right and left input mixers. The only setting is what value of resistor to connect between the input pin and a particular mixer (none, 10k, 20k, 40k). For example, you can have a differential pair between IN1_R and IN2_R, or between IN1_R and IN3_R, or single ended by setting the negative pin to the common. So, there’s a lot of flexibility and not just a differential “mode”. The registers would even allow you to set up two negative pins, although I’m not sure why that would be useful. I’m currently setting up DAPM so that both the positive and negative pins are wired to a single mixer. This doesn’t seem to be working since only the negative pin seems to cause the path to be enabled. Do I have to set up another “Negative Right” and “Negative Left” mixer to handle the negative pins? Am I missing something else? -- Jeremy McDermond (NH6Z) Xenotropic Systems mcdermj(a)xenotropic.com

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[alsa-devel] [PATCH] MAINTAINERS: Add myself as reviewer of FSL/NXP SoC sound drivers
by Fabio Estevam 10 May '16

10 May '16

From: Fabio Estevam <fabio.estevam(a)nxp.com> I would like to help reviewing FSL/NXP SoC sound drivers. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- MAINTAINERS | 1 + 1 file changed, 1 insertion(+) diff --git a/MAINTAINERS b/MAINTAINERS index 43b85c1..f896605 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -4846,6 +4846,7 @@ FREESCALE SOC SOUND DRIVERS M: Timur Tabi <timur(a)tabi.org> M: Nicolin Chen <nicoleotsuka(a)gmail.com> M: Xiubo Li <Xiubo.Lee(a)gmail.com> +R: Fabio Estevam <fabio.estevam(a)nxp.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) L: linuxppc-dev(a)lists.ozlabs.org S: Maintained -- 1.9.1

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[alsa-devel] [PATCH] ASoC: simple-card: Add pm callbacks to platform driver
by Peter Ujfalusi 10 May '16

10 May '16

Set snd_soc_pm_ops for the pm ops to make sure that the ASoC level of PM operations are going to happen. This is needed to get suspend/resume working correctly when the audio is using simple-card. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi Mark, we have moved most of the TI boards to use simple-card and I have noticed that we have quite bad regression because of it caused by simple-card not having the pm ops configured. This will render audio to not recover after suspend/resume when the audio IP is loosing context. Regards, Peter sound/soc/generic/simple-card.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index c6111e7a6d93..52a1e1ff3c62 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -658,6 +658,7 @@ MODULE_DEVICE_TABLE(of, asoc_simple_of_match); static struct platform_driver asoc_simple_card = { .driver = { .name = "asoc-simple-card", + .pm = &snd_soc_pm_ops, .of_match_table = asoc_simple_of_match, }, .probe = asoc_simple_card_probe, -- 2.8.1

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[alsa-devel] [PATCH] ASoC: davinci-mcasp: Do not allow multiple streams in one direction
by Peter Ujfalusi 10 May '16

10 May '16

Make sure that the user can not start multiple streams with the same direction. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index c4cd8e4fff9a..50523e744483 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1252,11 +1252,15 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, int i, dir; int tdm_slots = mcasp->tdm_slots; - if (mcasp->tdm_mask[substream->stream]) - tdm_slots = hweight32(mcasp->tdm_mask[substream->stream]); + /* Do not allow more then one stream per direction */ + if (mcasp->substreams[substream->stream]) + return -EBUSY; mcasp->substreams[substream->stream] = substream; + if (mcasp->tdm_mask[substream->stream]) + tdm_slots = hweight32(mcasp->tdm_mask[substream->stream]); + if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) return 0; -- 2.8.1

2 1

[alsa-devel] [PATCH 0/4] ASoC: davinci-mcasp: Master mode clock divider correction
by Peter Ujfalusi 10 May '16

10 May '16

Hi, If the McASP is used as clock master and the reference clock is AUXCLK we can have additional level of divider. The BCLK divider is limited to maximum 32, if the desired bclk can not be reached with this, the AUXCLK divider also needs to be used. Regards, Peter --- Peter Ujfalusi (4): ASoC: davinci-mcasp: Use defines for clkdiv IDs ASoC: davinci-mcasp: Change __davinci_mcasp_set_clkdiv() first parameter ASoC: davinci-mcasp: Restructure the davinci_mcasp_calc_clk_div() ASoC: davinci-mcasp: Calculate AUXCLK divider when setting up master clocks sound/soc/davinci/davinci-mcasp.c | 82 +++++++++++++++++++++++++-------------- sound/soc/davinci/davinci-mcasp.h | 5 +++ 2 files changed, 57 insertions(+), 30 deletions(-) -- 2.8.2

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[alsa-devel] [PATCH] ASoC: rsnd: don't use prohibited number to PDMACHCRn.SRS
by Kuninori Morimoto 10 May '16

10 May '16

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current rsnd_dmapp_get_id() returns 0xFF as error code if system used strange connection. It will be used as PDMACHCRn.SRS, but 0xFF is prohibited number. In order not to use prohibited number, this patch indicates error message and returns 0x00 (same as SSI00) in error case. Special thanks to Dung-san. Reported-by: Nguyen Viet Dung <nv-dung(a)jinso.co.jp> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/dma.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 7658e8f..6bc93cb 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -316,11 +316,15 @@ static u32 rsnd_dmapp_get_id(struct rsnd_dai_stream *io, size = ARRAY_SIZE(gen2_id_table_cmd); } - if (!entry) - return 0xFF; + if ((!entry) || (size <= id)) { + struct device *dev = rsnd_priv_to_dev(rsnd_io_to_priv(io)); - if (size <= id) - return 0xFF; + dev_err(dev, "unknown connection (%s[%d])\n", + rsnd_mod_name(mod), rsnd_mod_id(mod)); + + /* use non-prohibited SRS number as error */ + return 0x00; /* SSI00 */ + } return entry[id]; } -- 1.9.1

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