April 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Fixed single output empty pin_sense issue for alc283
by Kailang 03 Apr '14

03 Apr '14

Hi Takashi, Some machine with single output. It has only one HP out on it. alc283_init and alc283_shutup will get empty hp_pin value. I add the check for the attachment patch. Thanks. BR, Kailang

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[alsa-devel] [PATCH] ASoC: fsl_sai: Add imx6sx platform support
by Nicolin Chen 03 Apr '14

03 Apr '14

The next coming i.MX6 Solo X SoC also contains SAI module while we use imp_pcm_init() for i.MX platform. So this patch adds one compatible route for imx6sx and updates the DT doc accordingly. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- Documentation/devicetree/bindings/sound/fsl-sai.txt | 2 +- sound/soc/fsl/fsl_sai.c | 12 ++++++++++-- sound/soc/fsl/fsl_sai.h | 1 + 3 files changed, 12 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt index 98611a6..35c09fe 100644 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt @@ -7,7 +7,7 @@ codec/DSP interfaces. Required properties: -- compatible: Compatible list, contains "fsl,vf610-sai". +- compatible: Compatible list, contains "fsl,vf610-sai" or "fsl,imx6sx-sai". - reg: Offset and length of the register set for the device. - clocks: Must contain an entry for each entry in clock-names. - clock-names : Must include the "sai" entry. diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index d64c33f..9ed6795 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -22,6 +22,7 @@ #include <sound/pcm_params.h> #include "fsl_sai.h" +#include "imx-pcm.h" #define FSL_SAI_FLAGS (FSL_SAI_CSR_SEIE |\ FSL_SAI_CSR_FEIE) @@ -592,6 +593,9 @@ static int fsl_sai_probe(struct platform_device *pdev) sai->pdev = pdev; + if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai")) + sai->sai_on_imx = true; + sai->big_endian_regs = of_property_read_bool(np, "big-endian-regs"); if (sai->big_endian_regs) fsl_sai_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG; @@ -634,12 +638,16 @@ static int fsl_sai_probe(struct platform_device *pdev) if (ret) return ret; - return devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, - SND_DMAENGINE_PCM_FLAG_NO_RESIDUE); + if (sai->sai_on_imx) + return imx_pcm_dma_init(pdev); + else + return devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, + SND_DMAENGINE_PCM_FLAG_NO_RESIDUE); } static const struct of_device_id fsl_sai_ids[] = { { .compatible = "fsl,vf610-sai", }, + { .compatible = "fsl,imx6sx-sai", }, { /* sentinel */ } }; diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index be26d46..677670d 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -130,6 +130,7 @@ struct fsl_sai { bool big_endian_regs; bool big_endian_data; bool is_dsp_mode; + bool sai_on_imx; struct snd_dmaengine_dai_dma_data dma_params_rx; struct snd_dmaengine_dai_dma_data dma_params_tx; -- 1.8.4

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[alsa-devel] [PATCH v3] ASoC: sta350: Add codec driver
by Daniel Mack 02 Apr '14

02 Apr '14

From: Sven Brandau <brandau(a)gmx.de> The TI STA350 is an integrated 2.1-channel power amplifier that is controllable over I2C. This patch adds an ASoC driver for it. At a glance, this chip is very similar to the STA320 for which a driver already exists. In details, however, the register maps contain subtle differences which made a whole new driver easier to write and maintain. [daniel(a)zonque.org: cleanups, DT property rework, rebased on asoc-next] Signed-off-by: Sven Brandau <brandau(a)gmx.de> Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- v2 -> v3: * Renamed the GPIO DT properties to *-gpios * Use DECLARE_TLV_DB_RANGE for automatic TLV_DB_RANGE_HEAD calculation * Use the new gpiod API * Some more cleanups v1 -> v2: * Added my S-o-b * Made the Kconfig symbol visible * Documented regulator notes in DT bindings doc * Fixed spelling and other style nits * Converted to SOC_ENUM_SINGLE_DECL * Added a lock around the logic in sta350_coefficient_get() * Added a DAPM route from "DAC" to "Playback" * Use dev_dbg() rather than pr_debug() * Changed the code to make use of regmap_update_bits rather than open-coding the same functionality * Cleaned up the reset sequence logic * If a gpio was requested, the claiming has to succeed (handles EPROBE_DEFER cases) * Kick codec->control_data initialization * Skip initialization of default values in regcache * Apply power before running init sequences * Parse the DT without consulting of_match_device() .../devicetree/bindings/sound/st,sta350.txt | 107 ++ include/sound/sta350.h | 52 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sta350.c | 1266 ++++++++++++++++++++ sound/soc/codecs/sta350.h | 228 ++++ 6 files changed, 1660 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/st,sta350.txt create mode 100644 include/sound/sta350.h create mode 100644 sound/soc/codecs/sta350.c create mode 100644 sound/soc/codecs/sta350.h diff --git a/Documentation/devicetree/bindings/sound/st,sta350.txt b/Documentation/devicetree/bindings/sound/st,sta350.txt new file mode 100644 index 0000000..9501888 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/st,sta350.txt @@ -0,0 +1,107 @@ +STA350 audio CODEC + +The driver for this device only supports I2C. + +Required properties: + + - compatible: "st,sta350" + - reg: the I2C address of the device for I2C + - reset-gpios: a GPIO spec for the reset pin. If specified, it will be + deasserted before communication to the codec starts. + + - power-down-gpios: a GPIO spec for the power down pin. If specified, + it will be deasserted before communication to the codec + starts. + + - vdd-dig-supply: regulator spec, providing 3.3V + - vdd-pll-supply: regulator spec, providing 3.3V + - vcc-supply: regulator spec, providing 5V - 26V + +Optional properties: + + - st,output-conf: number, Selects the output configuration: + 0: 2-channel (full-bridge) power, 2-channel data-out + 1: 2 (half-bridge). 1 (full-bridge) on-board power + 2: 2 Channel (Full-Bridge) Power, 1 Channel FFX + 3: 1 Channel Mono-Parallel + If parameter is missing, mode 0 will be enabled. + + - st,ch1-output-mapping: Channel 1 output mapping + - st,ch2-output-mapping: Channel 2 output mapping + - st,ch3-output-mapping: Channel 3 output mapping + 0: Channel 1 + 1: Channel 2 + 2: Channel 3 + If parameter is missing, channel 1 is choosen. + + - st,thermal-warning-recover: + If present, thermal warning recovery is enabled. + + - st,thermal-warning-adjustment: + If present, thermal warning adjustment is enabled. + + - st,fault-detect-recovery: + If present, then fault recovery will be enabled. + + - st,ffx-power-output-mode: string + The FFX power output mode selects how the FFX output timing is + configured. Must be one of these values: + - "drop-compensation" + - "tapered-compensation" + - "full-power-mode" + - "variable-drop-compensation" (default) + + - st,drop-compensation-ns: number + Only required for "st,ffx-power-output-mode" == + "variable-drop-compensation". + Specifies the drop compensation in nanoseconds. + The value must be in the range of 0..300, and only + multiples of 20 are allowed. Default is 140ns. + + - st,overcurrent-warning-adjustment: + If present, overcurrent warning adjustment is enabled. + + - st,max-power-use-mpcc: + If present, then MPCC bits are used for MPC coefficients, + otherwise standard MPC coefficients are used. + + - st,max-power-corr: + If present, power bridge correction for THD reduction near maximum + power output is enabled. + + - st,am-reduction-mode: + If present, FFX mode runs in AM reduction mode, otherwise normal + FFX mode is used. + + - st,odd-pwm-speed-mode: + If present, PWM speed mode run on odd speed mode (341.3 kHz) on all + channels. If not present, normal PWM spped mode (384 kHz) will be used. + + - st,distortion-compensation: + If present, distortion compensation variable uses DCC coefficient. + If not present, preset DC coefficient is used. + + - st,invalid-input-detect-mute: + If not present, automatic invalid input detect mute is enabled. + + + +Example: + +codec: sta350@38 { + compatible = "st,sta350"; + reg = <0x1c>; + reset-gpios = <&gpio1 19 0>; + power-down-gpios = <&gpio1 16 0>; + st,output-conf = <0x3>; // set output to 2-channel + // (full-bridge) power, + // 2-channel data-out + st,ch1-output-mapping = <0>; // set channel 1 output ch 1 + st,ch2-output-mapping = <0>; // set channel 2 output ch 1 + st,ch3-output-mapping = <0>; // set channel 3 output ch 1 + st,max-power-correction; // enables power bridge + // correction for THD reduction + // near maximum power output + st,invalid-input-detect-mute; // mute if no valid digital + // audio signal is provided. +}; diff --git a/include/sound/sta350.h b/include/sound/sta350.h new file mode 100644 index 0000000..3a329810 --- /dev/null +++ b/include/sound/sta350.h @@ -0,0 +1,52 @@ +/* + * Platform data for ST STA350 ASoC codec driver. + * + * Copyright: 2014 Raumfeld GmbH + * Author: Sven Brandau <info(a)brandau.biz> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ +#ifndef __LINUX_SND__STA350_H +#define __LINUX_SND__STA350_H + +#define STA350_OCFG_2CH 0 +#define STA350_OCFG_2_1CH 1 +#define STA350_OCFG_1CH 3 + +#define STA350_OM_CH1 0 +#define STA350_OM_CH2 1 +#define STA350_OM_CH3 2 + +#define STA350_THERMAL_ADJUSTMENT_ENABLE 1 +#define STA350_THERMAL_RECOVERY_ENABLE 2 +#define STA350_FAULT_DETECT_RECOVERY_BYPASS 1 + +#define STA350_FFX_PM_DROP_COMP 0 +#define STA350_FFX_PM_TAPERED_COMP 1 +#define STA350_FFX_PM_FULL_POWER 2 +#define STA350_FFX_PM_VARIABLE_DROP_COMP 3 + + +struct sta350_platform_data { + u8 output_conf; + u8 ch1_output_mapping; + u8 ch2_output_mapping; + u8 ch3_output_mapping; + u8 ffx_power_output_mode; + u8 drop_compensation_ns; + unsigned int thermal_warning_recovery:1; + unsigned int thermal_warning_adjustment:1; + unsigned int fault_detect_recovery:1; + unsigned int oc_warning_adjustment:1; + unsigned int max_power_use_mpcc:1; + unsigned int max_power_correction:1; + unsigned int am_reduction_mode:1; + unsigned int odd_pwm_speed_mode:1; + unsigned int distortion_compensation:1; + unsigned int invalid_input_detect_mute:1; +}; + +#endif /* __LINUX_SND__STA350_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f0e8401..c7b853f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -80,6 +80,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_SSM2602_SPI if SPI_MASTER select SND_SOC_SSM2602_I2C if I2C select SND_SOC_STA32X if I2C + select SND_SOC_STA350 if I2C select SND_SOC_STA529 if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TAS5086 if I2C @@ -435,6 +436,10 @@ config SND_SOC_SSM2602_I2C config SND_SOC_STA32X tristate +config SND_SOC_STA350 + tristate "STA350 speaker amplifier" + depends on I2C + config SND_SOC_STA529 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 3c4d275..efdb4d0 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -74,6 +74,7 @@ snd-soc-ssm2602-objs := ssm2602.o snd-soc-ssm2602-spi-objs := ssm2602-spi.o snd-soc-ssm2602-i2c-objs := ssm2602-i2c.o snd-soc-sta32x-objs := sta32x.o +snd-soc-sta350-objs := sta350.o snd-soc-sta529-objs := sta529.o snd-soc-stac9766-objs := stac9766.o snd-soc-tas5086-objs := tas5086.o @@ -221,6 +222,7 @@ obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_SSM2602_SPI) += snd-soc-ssm2602-spi.o obj-$(CONFIG_SND_SOC_SSM2602_I2C) += snd-soc-ssm2602-i2c.o obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o +obj-$(CONFIG_SND_SOC_STA350) += snd-soc-sta350.o obj-$(CONFIG_SND_SOC_STA529) += snd-soc-sta529.o obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o diff --git a/sound/soc/codecs/sta350.c b/sound/soc/codecs/sta350.c new file mode 100644 index 0000000..552e92a --- /dev/null +++ b/sound/soc/codecs/sta350.c @@ -0,0 +1,1266 @@ +/* + * Codec driver for ST STA350 2.1-channel high-efficiency digital audio system + * + * Copyright: 2014 Raumfeld GmbH + * Author: Sven Brandau <info(a)brandau.biz> + * + * based on code from: + * Raumfeld GmbH + * Johannes Stezenbach <js(a)sig21.net> + * Wolfson Microelectronics PLC. + * Mark Brown <broonie(a)opensource.wolfsonmicro.com> + * Freescale Semiconductor, Inc. + * Timur Tabi <timur(a)freescale.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#define pr_fmt(fmt) KBUILD_MODNAME ":%s:%d: " fmt, __func__, __LINE__ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/of_device.h> +#include <linux/of_gpio.h> +#include <linux/regmap.h> +#include <linux/regulator/consumer.h> +#include <linux/gpio/consumer.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include <sound/sta350.h> +#include "sta350.h" + +#define STA350_RATES (SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_88200 | \ + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_176400 | \ + SNDRV_PCM_RATE_192000) + +#define STA350_FORMATS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE | \ + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE) + +/* Power-up register defaults */ +static const struct reg_default sta350_regs[] = { + { 0x0, 0x63 }, + { 0x1, 0x80 }, + { 0x2, 0xdf }, + { 0x3, 0x40 }, + { 0x4, 0xc2 }, + { 0x5, 0x5c }, + { 0x6, 0x00 }, + { 0x7, 0xff }, + { 0x8, 0x60 }, + { 0x9, 0x60 }, + { 0xa, 0x60 }, + { 0xb, 0x00 }, + { 0xc, 0x00 }, + { 0xd, 0x00 }, + { 0xe, 0x00 }, + { 0xf, 0x40 }, + { 0x10, 0x80 }, + { 0x11, 0x77 }, + { 0x12, 0x6a }, + { 0x13, 0x69 }, + { 0x14, 0x6a }, + { 0x15, 0x69 }, + { 0x16, 0x00 }, + { 0x17, 0x00 }, + { 0x18, 0x00 }, + { 0x19, 0x00 }, + { 0x1a, 0x00 }, + { 0x1b, 0x00 }, + { 0x1c, 0x00 }, + { 0x1d, 0x00 }, + { 0x1e, 0x00 }, + { 0x1f, 0x00 }, + { 0x20, 0x00 }, + { 0x21, 0x00 }, + { 0x22, 0x00 }, + { 0x23, 0x00 }, + { 0x24, 0x00 }, + { 0x25, 0x00 }, + { 0x26, 0x00 }, + { 0x27, 0x2a }, + { 0x28, 0xc0 }, + { 0x29, 0xf3 }, + { 0x2a, 0x33 }, + { 0x2b, 0x00 }, + { 0x2c, 0x0c }, + { 0x31, 0x00 }, + { 0x36, 0x00 }, + { 0x37, 0x00 }, + { 0x38, 0x00 }, + { 0x39, 0x01 }, + { 0x3a, 0xee }, + { 0x3b, 0xff }, + { 0x3c, 0x7e }, + { 0x3d, 0xc0 }, + { 0x3e, 0x26 }, + { 0x3f, 0x00 }, + { 0x48, 0x00 }, + { 0x49, 0x00 }, + { 0x4a, 0x00 }, + { 0x4b, 0x04 }, + { 0x4c, 0x00 }, +}; + +static const struct regmap_range sta350_write_regs_range[] = { + regmap_reg_range(STA350_CONFA, STA350_AUTO2), + regmap_reg_range(STA350_C1CFG, STA350_FDRC2), + regmap_reg_range(STA350_EQCFG, STA350_EVOLRES), + regmap_reg_range(STA350_NSHAPE, STA350_MISC2), +}; + +static const struct regmap_range sta350_read_regs_range[] = { + regmap_reg_range(STA350_CONFA, STA350_AUTO2), + regmap_reg_range(STA350_C1CFG, STA350_STATUS), + regmap_reg_range(STA350_EQCFG, STA350_EVOLRES), + regmap_reg_range(STA350_NSHAPE, STA350_MISC2), +}; + +static const struct regmap_range sta350_volatile_regs_range[] = { + regmap_reg_range(STA350_CFADDR2, STA350_CFUD), + regmap_reg_range(STA350_STATUS, STA350_STATUS), +}; + +static const struct regmap_access_table sta350_write_regs = { + .yes_ranges = sta350_write_regs_range, + .n_yes_ranges = ARRAY_SIZE(sta350_write_regs_range), +}; + +static const struct regmap_access_table sta350_read_regs = { + .yes_ranges = sta350_read_regs_range, + .n_yes_ranges = ARRAY_SIZE(sta350_read_regs_range), +}; + +static const struct regmap_access_table sta350_volatile_regs = { + .yes_ranges = sta350_volatile_regs_range, + .n_yes_ranges = ARRAY_SIZE(sta350_volatile_regs_range), +}; + +/* regulator power supply names */ +static const char * const sta350_supply_names[] = { + "vdd-dig", /* digital supply, 3.3V */ + "vdd-pll", /* pll supply, 3.3V */ + "vcc" /* power amp supply, 5V - 26V */ +}; + +/* codec private data */ +struct sta350_priv { + struct regmap *regmap; + struct regulator_bulk_data supplies[ARRAY_SIZE(sta350_supply_names)]; + struct sta350_platform_data *pdata; + + unsigned int mclk; + unsigned int format; + + u32 coef_shadow[STA350_COEF_COUNT]; + int shutdown; + + struct gpio_desc *gpiod_nreset; + struct gpio_desc *gpiod_power_down; + + struct mutex coeff_lock; +}; + +static const DECLARE_TLV_DB_SCALE(mvol_tlv, -12750, 50, 1); +static const DECLARE_TLV_DB_SCALE(chvol_tlv, -7950, 50, 1); +static const DECLARE_TLV_DB_SCALE(tone_tlv, -1200, 200, 0); + +static const char * const sta350_drc_ac[] = { + "Anti-Clipping", "Dynamic Range Compression" +}; +static const char * const sta350_auto_gc_mode[] = { + "User", "AC no clipping", "AC limited clipping (10%)", + "DRC nighttime listening mode" +}; +static const char * const sta350_auto_xo_mode[] = { + "User", "80Hz", "100Hz", "120Hz", "140Hz", "160Hz", "180Hz", + "200Hz", "220Hz", "240Hz", "260Hz", "280Hz", "300Hz", "320Hz", + "340Hz", "360Hz" +}; +static const char * const sta350_binary_output[] = { + "FFX 3-state output - normal operation", "Binary output" +}; +static const char * const sta350_limiter_select[] = { + "Limiter Disabled", "Limiter #1", "Limiter #2" +}; +static const char * const sta350_limiter_attack_rate[] = { + "3.1584", "2.7072", "2.2560", "1.8048", "1.3536", "0.9024", + "0.4512", "0.2256", "0.1504", "0.1123", "0.0902", "0.0752", + "0.0645", "0.0564", "0.0501", "0.0451" +}; +static const char * const sta350_limiter_release_rate[] = { + "0.5116", "0.1370", "0.0744", "0.0499", "0.0360", "0.0299", + "0.0264", "0.0208", "0.0198", "0.0172", "0.0147", "0.0137", + "0.0134", "0.0117", "0.0110", "0.0104" +}; +static const char * const sta350_noise_shaper_type[] = { + "Third order", "Fourth order" +}; + +static DECLARE_TLV_DB_RANGE(sta350_limiter_ac_attack_tlv, + 0, 7, TLV_DB_SCALE_ITEM(-1200, 200, 0), + 8, 16, TLV_DB_SCALE_ITEM(300, 100, 0), +); + +static DECLARE_TLV_DB_RANGE(sta350_limiter_ac_release_tlv, + 0, 0, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(-2900, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(-2000, 0, 0), + 3, 8, TLV_DB_SCALE_ITEM(-1400, 200, 0), + 8, 16, TLV_DB_SCALE_ITEM(-700, 100, 0), +); + +static DECLARE_TLV_DB_RANGE(sta350_limiter_drc_attack_tlv, + 0, 7, TLV_DB_SCALE_ITEM(-3100, 200, 0), + 8, 13, TLV_DB_SCALE_ITEM(-1600, 100, 0), + 14, 16, TLV_DB_SCALE_ITEM(-1000, 300, 0), +); + +static DECLARE_TLV_DB_RANGE(sta350_limiter_drc_release_tlv, + 0, 0, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 0), + 1, 2, TLV_DB_SCALE_ITEM(-3800, 200, 0), + 3, 4, TLV_DB_SCALE_ITEM(-3300, 200, 0), + 5, 12, TLV_DB_SCALE_ITEM(-3000, 200, 0), + 13, 16, TLV_DB_SCALE_ITEM(-1500, 300, 0), +); + +static SOC_ENUM_SINGLE_DECL(sta350_drc_ac_enum, + STA350_CONFD, STA350_CONFD_DRC_SHIFT, + sta350_drc_ac); +static SOC_ENUM_SINGLE_DECL(sta350_noise_shaper_enum, + STA350_CONFE, STA350_CONFE_NSBW_SHIFT, + sta350_noise_shaper_type); +static SOC_ENUM_SINGLE_DECL(sta350_auto_gc_enum, + STA350_AUTO1, STA350_AUTO1_AMGC_SHIFT, + sta350_auto_gc_mode); +static SOC_ENUM_SINGLE_DECL(sta350_auto_xo_enum, + STA350_AUTO2, STA350_AUTO2_XO_SHIFT, + sta350_auto_xo_mode); +static SOC_ENUM_SINGLE_DECL(sta350_binary_output_ch1_enum, + STA350_C1CFG, STA350_CxCFG_BO_SHIFT, + sta350_binary_output); +static SOC_ENUM_SINGLE_DECL(sta350_binary_output_ch2_enum, + STA350_C2CFG, STA350_CxCFG_BO_SHIFT, + sta350_binary_output); +static SOC_ENUM_SINGLE_DECL(sta350_binary_output_ch3_enum, + STA350_C3CFG, STA350_CxCFG_BO_SHIFT, + sta350_binary_output); +static SOC_ENUM_SINGLE_DECL(sta350_limiter_ch1_enum, + STA350_C1CFG, STA350_CxCFG_LS_SHIFT, + sta350_limiter_select); +static SOC_ENUM_SINGLE_DECL(sta350_limiter_ch2_enum, + STA350_C2CFG, STA350_CxCFG_LS_SHIFT, + sta350_limiter_select); +static SOC_ENUM_SINGLE_DECL(sta350_limiter_ch3_enum, + STA350_C3CFG, STA350_CxCFG_LS_SHIFT, + sta350_limiter_select); +static SOC_ENUM_SINGLE_DECL(sta350_limiter1_attack_rate_enum, + STA350_L1AR, STA350_LxA_SHIFT, + sta350_limiter_attack_rate); +static SOC_ENUM_SINGLE_DECL(sta350_limiter2_attack_rate_enum, + STA350_L2AR, STA350_LxA_SHIFT, + sta350_limiter_attack_rate); +static SOC_ENUM_SINGLE_DECL(sta350_limiter1_release_rate_enum, + STA350_L1AR, STA350_LxR_SHIFT, + sta350_limiter_release_rate); +static SOC_ENUM_SINGLE_DECL(sta350_limiter2_release_rate_enum, + STA350_L2AR, STA350_LxR_SHIFT, + sta350_limiter_release_rate); + +/* + * byte array controls for setting biquad, mixer, scaling coefficients; + * for biquads all five coefficients need to be set in one go, + * mixer and pre/postscale coefs can be set individually; + * each coef is 24bit, the bytes are ordered in the same way + * as given in the STA350 data sheet (big endian; b1, b2, a1, a2, b0) + */ + +static int sta350_coefficient_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int numcoef = kcontrol->private_value >> 16; + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + uinfo->count = 3 * numcoef; + return 0; +} + +static int sta350_coefficient_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + int numcoef = kcontrol->private_value >> 16; + int index = kcontrol->private_value & 0xffff; + unsigned int cfud, val; + int i, ret = 0; + + mutex_lock(&sta350->coeff_lock); + + /* preserve reserved bits in STA350_CFUD */ + regmap_read(sta350->regmap, STA350_CFUD, &cfud); + cfud &= 0xf0; + /* + * chip documentation does not say if the bits are self clearing, + * so do it explicitly + */ + regmap_write(sta350->regmap, STA350_CFUD, cfud); + + regmap_write(sta350->regmap, STA350_CFADDR2, index); + if (numcoef == 1) { + regmap_write(sta350->regmap, STA350_CFUD, cfud | 0x04); + } else if (numcoef == 5) { + regmap_write(sta350->regmap, STA350_CFUD, cfud | 0x08); + } else { + ret = -EINVAL; + goto exit_unlock; + } + + for (i = 0; i < 3 * numcoef; i++) { + regmap_read(sta350->regmap, STA350_B1CF1 + i, &val); + ucontrol->value.bytes.data[i] = val; + } + +exit_unlock: + mutex_unlock(&sta350->coeff_lock); + + return ret; +} + +static int sta350_coefficient_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + int numcoef = kcontrol->private_value >> 16; + int index = kcontrol->private_value & 0xffff; + unsigned int cfud; + int i; + + /* preserve reserved bits in STA350_CFUD */ + regmap_read(sta350->regmap, STA350_CFUD, &cfud); + cfud &= 0xf0; + /* + * chip documentation does not say if the bits are self clearing, + * so do it explicitly + */ + regmap_write(sta350->regmap, STA350_CFUD, cfud); + + regmap_write(sta350->regmap, STA350_CFADDR2, index); + for (i = 0; i < numcoef && (index + i < STA350_COEF_COUNT); i++) + sta350->coef_shadow[index + i] = + (ucontrol->value.bytes.data[3 * i] << 16) + | (ucontrol->value.bytes.data[3 * i + 1] << 8) + | (ucontrol->value.bytes.data[3 * i + 2]); + for (i = 0; i < 3 * numcoef; i++) + regmap_write(sta350->regmap, STA350_B1CF1 + i, + ucontrol->value.bytes.data[i]); + if (numcoef == 1) + regmap_write(sta350->regmap, STA350_CFUD, cfud | 0x01); + else if (numcoef == 5) + regmap_write(sta350->regmap, STA350_CFUD, cfud | 0x02); + else + return -EINVAL; + + return 0; +} + +static int sta350_sync_coef_shadow(struct snd_soc_codec *codec) +{ + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + unsigned int cfud; + int i; + + /* preserve reserved bits in STA350_CFUD */ + regmap_read(sta350->regmap, STA350_CFUD, &cfud); + cfud &= 0xf0; + + for (i = 0; i < STA350_COEF_COUNT; i++) { + regmap_write(sta350->regmap, STA350_CFADDR2, i); + regmap_write(sta350->regmap, STA350_B1CF1, + (sta350->coef_shadow[i] >> 16) & 0xff); + regmap_write(sta350->regmap, STA350_B1CF2, + (sta350->coef_shadow[i] >> 8) & 0xff); + regmap_write(sta350->regmap, STA350_B1CF3, + (sta350->coef_shadow[i]) & 0xff); + /* + * chip documentation does not say if the bits are + * self-clearing, so do it explicitly + */ + regmap_write(sta350->regmap, STA350_CFUD, cfud); + regmap_write(sta350->regmap, STA350_CFUD, cfud | 0x01); + } + return 0; +} + +static int sta350_cache_sync(struct snd_soc_codec *codec) +{ + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + unsigned int mute; + int rc; + + /* mute during register sync */ + regmap_read(sta350->regmap, STA350_CFUD, &mute); + regmap_write(sta350->regmap, STA350_MMUTE, mute | STA350_MMUTE_MMUTE); + sta350_sync_coef_shadow(codec); + rc = regcache_sync(sta350->regmap); + regmap_write(sta350->regmap, STA350_MMUTE, mute); + return rc; +} + +#define SINGLE_COEF(xname, index) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = sta350_coefficient_info, \ + .get = sta350_coefficient_get,\ + .put = sta350_coefficient_put, \ + .private_value = index | (1 << 16) } + +#define BIQUAD_COEFS(xname, index) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = sta350_coefficient_info, \ + .get = sta350_coefficient_get,\ + .put = sta350_coefficient_put, \ + .private_value = index | (5 << 16) } + +static const struct snd_kcontrol_new sta350_snd_controls[] = { +SOC_SINGLE_TLV("Master Volume", STA350_MVOL, 0, 0xff, 1, mvol_tlv), +/* VOL */ +SOC_SINGLE_TLV("Ch1 Volume", STA350_C1VOL, 0, 0xff, 1, chvol_tlv), +SOC_SINGLE_TLV("Ch2 Volume", STA350_C2VOL, 0, 0xff, 1, chvol_tlv), +SOC_SINGLE_TLV("Ch3 Volume", STA350_C3VOL, 0, 0xff, 1, chvol_tlv), +/* CONFD */ +SOC_SINGLE("High Pass Filter Bypass Switch", + STA350_CONFD, STA350_CONFD_HPB_SHIFT, 1, 1), +SOC_SINGLE("De-emphasis Filter Switch", + STA350_CONFD, STA350_CONFD_DEMP_SHIFT, 1, 0), +SOC_SINGLE("DSP Bypass Switch", + STA350_CONFD, STA350_CONFD_DSPB_SHIFT, 1, 0), +SOC_SINGLE("Post-scale Link Switch", + STA350_CONFD, STA350_CONFD_PSL_SHIFT, 1, 0), +SOC_SINGLE("Biquad Coefficient Link Switch", + STA350_CONFD, STA350_CONFD_BQL_SHIFT, 1, 0), +SOC_ENUM("Compressor/Limiter Switch", sta350_drc_ac_enum), +SOC_ENUM("Noise Shaper Bandwidth", sta350_noise_shaper_enum), +SOC_SINGLE("Zero-detect Mute Enable Switch", + STA350_CONFD, STA350_CONFD_ZDE_SHIFT, 1, 0), +SOC_SINGLE("Submix Mode Switch", + STA350_CONFD, STA350_CONFD_SME_SHIFT, 1, 0), +/* CONFE */ +SOC_SINGLE("Zero Cross Switch", STA350_CONFE, STA350_CONFE_ZCE_SHIFT, 1, 0), +SOC_SINGLE("Soft Ramp Switch", STA350_CONFE, STA350_CONFE_SVE_SHIFT, 1, 0), +/* MUTE */ +SOC_SINGLE("Master Switch", STA350_MMUTE, STA350_MMUTE_MMUTE_SHIFT, 1, 1), +SOC_SINGLE("Ch1 Switch", STA350_MMUTE, STA350_MMUTE_C1M_SHIFT, 1, 1), +SOC_SINGLE("Ch2 Switch", STA350_MMUTE, STA350_MMUTE_C2M_SHIFT, 1, 1), +SOC_SINGLE("Ch3 Switch", STA350_MMUTE, STA350_MMUTE_C3M_SHIFT, 1, 1), +/* AUTOx */ +SOC_ENUM("Automode GC", sta350_auto_gc_enum), +SOC_ENUM("Automode XO", sta350_auto_xo_enum), +/* CxCFG */ +SOC_SINGLE("Ch1 Tone Control Bypass Switch", + STA350_C1CFG, STA350_CxCFG_TCB_SHIFT, 1, 0), +SOC_SINGLE("Ch2 Tone Control Bypass Switch", + STA350_C2CFG, STA350_CxCFG_TCB_SHIFT, 1, 0), +SOC_SINGLE("Ch1 EQ Bypass Switch", + STA350_C1CFG, STA350_CxCFG_EQBP_SHIFT, 1, 0), +SOC_SINGLE("Ch2 EQ Bypass Switch", + STA350_C2CFG, STA350_CxCFG_EQBP_SHIFT, 1, 0), +SOC_SINGLE("Ch1 Master Volume Bypass Switch", + STA350_C1CFG, STA350_CxCFG_VBP_SHIFT, 1, 0), +SOC_SINGLE("Ch2 Master Volume Bypass Switch", + STA350_C1CFG, STA350_CxCFG_VBP_SHIFT, 1, 0), +SOC_SINGLE("Ch3 Master Volume Bypass Switch", + STA350_C1CFG, STA350_CxCFG_VBP_SHIFT, 1, 0), +SOC_ENUM("Ch1 Binary Output Select", sta350_binary_output_ch1_enum), +SOC_ENUM("Ch2 Binary Output Select", sta350_binary_output_ch2_enum), +SOC_ENUM("Ch3 Binary Output Select", sta350_binary_output_ch3_enum), +SOC_ENUM("Ch1 Limiter Select", sta350_limiter_ch1_enum), +SOC_ENUM("Ch2 Limiter Select", sta350_limiter_ch2_enum), +SOC_ENUM("Ch3 Limiter Select", sta350_limiter_ch3_enum), +/* TONE */ +SOC_SINGLE_RANGE_TLV("Bass Tone Control Volume", + STA350_TONE, STA350_TONE_BTC_SHIFT, 1, 13, 0, tone_tlv), +SOC_SINGLE_RANGE_TLV("Treble Tone Control Volume", + STA350_TONE, STA350_TONE_TTC_SHIFT, 1, 13, 0, tone_tlv), +SOC_ENUM("Limiter1 Attack Rate (dB/ms)", sta350_limiter1_attack_rate_enum), +SOC_ENUM("Limiter2 Attack Rate (dB/ms)", sta350_limiter2_attack_rate_enum), +SOC_ENUM("Limiter1 Release Rate (dB/ms)", sta350_limiter1_release_rate_enum), +SOC_ENUM("Limiter2 Release Rate (dB/ms)", sta350_limiter2_release_rate_enum), + +/* + * depending on mode, the attack/release thresholds have + * two different enum definitions; provide both + */ +SOC_SINGLE_TLV("Limiter1 Attack Threshold (AC Mode)", + STA350_L1ATRT, STA350_LxA_SHIFT, + 16, 0, sta350_limiter_ac_attack_tlv), +SOC_SINGLE_TLV("Limiter2 Attack Threshold (AC Mode)", + STA350_L2ATRT, STA350_LxA_SHIFT, + 16, 0, sta350_limiter_ac_attack_tlv), +SOC_SINGLE_TLV("Limiter1 Release Threshold (AC Mode)", + STA350_L1ATRT, STA350_LxR_SHIFT, + 16, 0, sta350_limiter_ac_release_tlv), +SOC_SINGLE_TLV("Limiter2 Release Threshold (AC Mode)", + STA350_L2ATRT, STA350_LxR_SHIFT, + 16, 0, sta350_limiter_ac_release_tlv), +SOC_SINGLE_TLV("Limiter1 Attack Threshold (DRC Mode)", + STA350_L1ATRT, STA350_LxA_SHIFT, + 16, 0, sta350_limiter_drc_attack_tlv), +SOC_SINGLE_TLV("Limiter2 Attack Threshold (DRC Mode)", + STA350_L2ATRT, STA350_LxA_SHIFT, + 16, 0, sta350_limiter_drc_attack_tlv), +SOC_SINGLE_TLV("Limiter1 Release Threshold (DRC Mode)", + STA350_L1ATRT, STA350_LxR_SHIFT, + 16, 0, sta350_limiter_drc_release_tlv), +SOC_SINGLE_TLV("Limiter2 Release Threshold (DRC Mode)", + STA350_L2ATRT, STA350_LxR_SHIFT, + 16, 0, sta350_limiter_drc_release_tlv), + +BIQUAD_COEFS("Ch1 - Biquad 1", 0), +BIQUAD_COEFS("Ch1 - Biquad 2", 5), +BIQUAD_COEFS("Ch1 - Biquad 3", 10), +BIQUAD_COEFS("Ch1 - Biquad 4", 15), +BIQUAD_COEFS("Ch2 - Biquad 1", 20), +BIQUAD_COEFS("Ch2 - Biquad 2", 25), +BIQUAD_COEFS("Ch2 - Biquad 3", 30), +BIQUAD_COEFS("Ch2 - Biquad 4", 35), +BIQUAD_COEFS("High-pass", 40), +BIQUAD_COEFS("Low-pass", 45), +SINGLE_COEF("Ch1 - Prescale", 50), +SINGLE_COEF("Ch2 - Prescale", 51), +SINGLE_COEF("Ch1 - Postscale", 52), +SINGLE_COEF("Ch2 - Postscale", 53), +SINGLE_COEF("Ch3 - Postscale", 54), +SINGLE_COEF("Thermal warning - Postscale", 55), +SINGLE_COEF("Ch1 - Mix 1", 56), +SINGLE_COEF("Ch1 - Mix 2", 57), +SINGLE_COEF("Ch2 - Mix 1", 58), +SINGLE_COEF("Ch2 - Mix 2", 59), +SINGLE_COEF("Ch3 - Mix 1", 60), +SINGLE_COEF("Ch3 - Mix 2", 61), +}; + +static const struct snd_soc_dapm_widget sta350_dapm_widgets[] = { +SND_SOC_DAPM_DAC("DAC", NULL, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_OUTPUT("LEFT"), +SND_SOC_DAPM_OUTPUT("RIGHT"), +SND_SOC_DAPM_OUTPUT("SUB"), +}; + +static const struct snd_soc_dapm_route sta350_dapm_routes[] = { + { "LEFT", NULL, "DAC" }, + { "RIGHT", NULL, "DAC" }, + { "SUB", NULL, "DAC" }, + { "DAC", NULL, "Playback" }, +}; + +/* MCLK interpolation ratio per fs */ +static struct { + int fs; + int ir; +} interpolation_ratios[] = { + { 32000, 0 }, + { 44100, 0 }, + { 48000, 0 }, + { 88200, 1 }, + { 96000, 1 }, + { 176400, 2 }, + { 192000, 2 }, +}; + +/* MCLK to fs clock ratios */ +static int mcs_ratio_table[3][6] = { + { 768, 512, 384, 256, 128, 576 }, + { 384, 256, 192, 128, 64, 0 }, + { 192, 128, 96, 64, 32, 0 }, +}; + +/** + * sta350_set_dai_sysclk - configure MCLK + * @codec_dai: the codec DAI + * @clk_id: the clock ID (ignored) + * @freq: the MCLK input frequency + * @dir: the clock direction (ignored) + * + * The value of MCLK is used to determine which sample rates are supported + * by the STA350, based on the mcs_ratio_table. + * + * This function must be called by the machine driver's 'startup' function, + * otherwise the list of supported sample rates will not be available in + * time for ALSA. + */ +static int sta350_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "mclk=%u\n", freq); + sta350->mclk = freq; + + return 0; +} + +/** + * sta350_set_dai_fmt - configure the codec for the selected audio format + * @codec_dai: the codec DAI + * @fmt: a SND_SOC_DAIFMT_x value indicating the data format + * + * This function takes a bitmask of SND_SOC_DAIFMT_x bits and programs the + * codec accordingly. + */ +static int sta350_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + unsigned int confb = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_RIGHT_J: + case SND_SOC_DAIFMT_LEFT_J: + sta350->format = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + confb |= STA350_CONFB_C2IM; + break; + case SND_SOC_DAIFMT_NB_IF: + confb |= STA350_CONFB_C1IM; + break; + default: + return -EINVAL; + } + + return regmap_update_bits(sta350->regmap, STA350_CONFB, + STA350_CONFB_C1IM | STA350_CONFB_C2IM, confb); +} + +/** + * sta350_hw_params - program the STA350 with the given hardware parameters. + * @substream: the audio stream + * @params: the hardware parameters to set + * @dai: the SOC DAI (ignored) + * + * This function programs the hardware with the values provided. + * Specifically, the sample rate and the data format. + */ +static int sta350_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 sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + int i, mcs = -EINVAL, ir = -EINVAL; + unsigned int confa, confb; + unsigned int rate, ratio; + int ret; + + if (!sta350->mclk) { + dev_err(codec->dev, + "sta350->mclk is unset. Unable to determine ratio\n"); + return -EIO; + } + + rate = params_rate(params); + ratio = sta350->mclk / rate; + dev_dbg(codec->dev, "rate: %u, ratio: %u\n", rate, ratio); + + for (i = 0; i < ARRAY_SIZE(interpolation_ratios); i++) { + if (interpolation_ratios[i].fs == rate) { + ir = interpolation_ratios[i].ir; + break; + } + } + + if (ir < 0) { + dev_err(codec->dev, "Unsupported samplerate: %u\n", rate); + return -EINVAL; + } + + for (i = 0; i < 6; i++) { + if (mcs_ratio_table[ir][i] == ratio) { + mcs = i; + break; + } + } + + if (mcs < 0) { + dev_err(codec->dev, "Unresolvable ratio: %u\n", ratio); + return -EINVAL; + } + + confa = (ir << STA350_CONFA_IR_SHIFT) | + (mcs << STA350_CONFA_MCS_SHIFT); + confb = 0; + + switch (params_width(params)) { + case 24: + dev_dbg(codec->dev, "24bit\n"); + /* fall through */ + case 32: + dev_dbg(codec->dev, "24bit or 32bit\n"); + switch (sta350->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x0; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0x1; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0x2; + break; + } + + break; + case 20: + dev_dbg(codec->dev, "20bit\n"); + switch (sta350->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x4; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0x5; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0x6; + break; + } + + break; + case 18: + dev_dbg(codec->dev, "18bit\n"); + switch (sta350->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x8; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0x9; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0xa; + break; + } + + break; + case 16: + dev_dbg(codec->dev, "16bit\n"); + switch (sta350->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x0; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0xd; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0xe; + break; + } + + break; + default: + return -EINVAL; + } + + ret = regmap_update_bits(sta350->regmap, STA350_CONFA, + STA350_CONFA_MCS_MASK | STA350_CONFA_IR_MASK, + confa); + if (ret < 0) + return ret; + + ret = regmap_update_bits(sta350->regmap, STA350_CONFB, + STA350_CONFB_SAI_MASK | STA350_CONFB_SAIFB, + confb); + if (ret < 0) + return ret; + + return 0; +} + +static int sta350_startup_sequence(struct sta350_priv *sta350) +{ + if (sta350->gpiod_power_down) + gpiod_set_value(sta350->gpiod_power_down, 1); + + if (sta350->gpiod_nreset) { + gpiod_set_value(sta350->gpiod_nreset, 0); + mdelay(1); + gpiod_set_value(sta350->gpiod_nreset, 1); + mdelay(1); + } + + return 0; +} + +/** + * sta350_set_bias_level - DAPM callback + * @codec: the codec device + * @level: DAPM power level + * + * This is called by ALSA to put the codec into low power mode + * or to wake it up. If the codec is powered off completely + * all registers must be restored after power on. + */ +static int sta350_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + int ret; + + dev_dbg(codec->dev, "level = %d\n", level); + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + /* Full power on */ + regmap_update_bits(sta350->regmap, STA350_CONFF, + STA350_CONFF_PWDN | STA350_CONFF_EAPD, + STA350_CONFF_PWDN | STA350_CONFF_EAPD); + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = regulator_bulk_enable( + ARRAY_SIZE(sta350->supplies), + sta350->supplies); + if (ret < 0) { + dev_err(codec->dev, + "Failed to enable supplies: %d\n", + ret); + return ret; + } + sta350_startup_sequence(sta350); + sta350_cache_sync(codec); + } + + /* Power down */ + regmap_update_bits(sta350->regmap, STA350_CONFF, + STA350_CONFF_PWDN | STA350_CONFF_EAPD, + 0); + + break; + + case SND_SOC_BIAS_OFF: + /* The chip runs through the power down sequence for us */ + regmap_update_bits(sta350->regmap, STA350_CONFF, + STA350_CONFF_PWDN | STA350_CONFF_EAPD, 0); + + /* power down: low */ + if (sta350->gpiod_power_down) + gpiod_set_value(sta350->gpiod_power_down, 0); + + if (sta350->gpiod_nreset) + gpiod_set_value(sta350->gpiod_nreset, 0); + + regulator_bulk_disable(ARRAY_SIZE(sta350->supplies), + sta350->supplies); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static const struct snd_soc_dai_ops sta350_dai_ops = { + .hw_params = sta350_hw_params, + .set_sysclk = sta350_set_dai_sysclk, + .set_fmt = sta350_set_dai_fmt, +}; + +static struct snd_soc_dai_driver sta350_dai = { + .name = "sta350-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = STA350_RATES, + .formats = STA350_FORMATS, + }, + .ops = &sta350_dai_ops, +}; + +#ifdef CONFIG_PM +static int sta350_suspend(struct snd_soc_codec *codec) +{ + sta350_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int sta350_resume(struct snd_soc_codec *codec) +{ + sta350_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} +#else +#define sta350_suspend NULL +#define sta350_resume NULL +#endif + +static int sta350_probe(struct snd_soc_codec *codec) +{ + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + struct sta350_platform_data *pdata = sta350->pdata; + int i, ret = 0, thermal = 0; + + ret = regulator_bulk_enable(ARRAY_SIZE(sta350->supplies), + sta350->supplies); + if (ret < 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + + ret = sta350_startup_sequence(sta350); + if (ret < 0) { + dev_err(codec->dev, "Failed to startup device\n"); + return ret; + } + + /* CONFA */ + if (!pdata->thermal_warning_recovery) + thermal |= STA350_CONFA_TWAB; + if (!pdata->thermal_warning_adjustment) + thermal |= STA350_CONFA_TWRB; + if (!pdata->fault_detect_recovery) + thermal |= STA350_CONFA_FDRB; + regmap_update_bits(sta350->regmap, STA350_CONFA, + STA350_CONFA_TWAB | STA350_CONFA_TWRB | + STA350_CONFA_FDRB, + thermal); + + /* CONFC */ + regmap_update_bits(sta350->regmap, STA350_CONFC, + STA350_CONFC_OM_MASK, + pdata->ffx_power_output_mode + << STA350_CONFC_OM_SHIFT); + regmap_update_bits(sta350->regmap, STA350_CONFC, + STA350_CONFC_CSZ_MASK, + pdata->drop_compensation_ns + << STA350_CONFC_CSZ_SHIFT); + regmap_update_bits(sta350->regmap, + STA350_CONFC, + STA350_CONFC_OCRB, + pdata->oc_warning_adjustment ? + STA350_CONFC_OCRB : 0); + + /* CONFE */ + regmap_update_bits(sta350->regmap, STA350_CONFE, + STA350_CONFE_MPCV, + pdata->max_power_use_mpcc ? + STA350_CONFE_MPCV : 0); + regmap_update_bits(sta350->regmap, STA350_CONFE, + STA350_CONFE_MPC, + pdata->max_power_correction ? + STA350_CONFE_MPC : 0); + regmap_update_bits(sta350->regmap, STA350_CONFE, + STA350_CONFE_AME, + pdata->am_reduction_mode ? + STA350_CONFE_AME : 0); + regmap_update_bits(sta350->regmap, STA350_CONFE, + STA350_CONFE_PWMS, + pdata->odd_pwm_speed_mode ? + STA350_CONFE_PWMS : 0); + regmap_update_bits(sta350->regmap, STA350_CONFE, + STA350_CONFE_DCCV, + pdata->distortion_compensation ? + STA350_CONFE_DCCV : 0); + /* CONFF */ + regmap_update_bits(sta350->regmap, STA350_CONFF, + STA350_CONFF_IDE, + pdata->invalid_input_detect_mute ? + STA350_CONFF_IDE : 0); + regmap_update_bits(sta350->regmap, STA350_CONFF, + STA350_CONFF_OCFG_MASK, + pdata->output_conf + << STA350_CONFF_OCFG_SHIFT); + + /* channel to output mapping */ + regmap_update_bits(sta350->regmap, STA350_C1CFG, + STA350_CxCFG_OM_MASK, + pdata->ch1_output_mapping + << STA350_CxCFG_OM_SHIFT); + regmap_update_bits(sta350->regmap, STA350_C2CFG, + STA350_CxCFG_OM_MASK, + pdata->ch2_output_mapping + << STA350_CxCFG_OM_SHIFT); + regmap_update_bits(sta350->regmap, STA350_C3CFG, + STA350_CxCFG_OM_MASK, + pdata->ch3_output_mapping + << STA350_CxCFG_OM_SHIFT); + + /* initialize coefficient shadow RAM with reset values */ + for (i = 4; i <= 49; i += 5) + sta350->coef_shadow[i] = 0x400000; + for (i = 50; i <= 54; i++) + sta350->coef_shadow[i] = 0x7fffff; + sta350->coef_shadow[55] = 0x5a9df7; + sta350->coef_shadow[56] = 0x7fffff; + sta350->coef_shadow[59] = 0x7fffff; + sta350->coef_shadow[60] = 0x400000; + sta350->coef_shadow[61] = 0x400000; + + sta350_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + /* Bias level configuration will have done an extra enable */ + regulator_bulk_disable(ARRAY_SIZE(sta350->supplies), sta350->supplies); + + return 0; +} + +static int sta350_remove(struct snd_soc_codec *codec) +{ + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + + sta350_set_bias_level(codec, SND_SOC_BIAS_OFF); + regulator_bulk_disable(ARRAY_SIZE(sta350->supplies), sta350->supplies); + + return 0; +} + +static const struct snd_soc_codec_driver sta350_codec = { + .probe = sta350_probe, + .remove = sta350_remove, + .suspend = sta350_suspend, + .resume = sta350_resume, + .set_bias_level = sta350_set_bias_level, + .controls = sta350_snd_controls, + .num_controls = ARRAY_SIZE(sta350_snd_controls), + .dapm_widgets = sta350_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(sta350_dapm_widgets), + .dapm_routes = sta350_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(sta350_dapm_routes), +}; + +static const struct regmap_config sta350_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = STA350_MISC2, + .reg_defaults = sta350_regs, + .num_reg_defaults = ARRAY_SIZE(sta350_regs), + .cache_type = REGCACHE_RBTREE, + .wr_table = &sta350_write_regs, + .rd_table = &sta350_read_regs, + .volatile_table = &sta350_volatile_regs, +}; + +#ifdef CONFIG_OF +static const struct of_device_id st350_dt_ids[] = { + { .compatible = "st,sta350", }, + { } +}; +MODULE_DEVICE_TABLE(of, st350_dt_ids); + +static const char * const sta350_ffx_modes[] = { + [STA350_FFX_PM_DROP_COMP] = "drop-compensation", + [STA350_FFX_PM_TAPERED_COMP] = "tapered-compensation", + [STA350_FFX_PM_FULL_POWER] = "full-power-mode", + [STA350_FFX_PM_VARIABLE_DROP_COMP] = "variable-drop-compensation", +}; + +static int sta350_probe_dt(struct device *dev, struct sta350_priv *sta350) +{ + struct device_node *np = dev->of_node; + struct sta350_platform_data *pdata; + const char *ffx_power_mode; + u16 tmp; + + pdata = devm_kzalloc(dev, sizeof(*pdata), GFP_KERNEL); + if (!pdata) + return -ENOMEM; + + of_property_read_u8(np, "st,output-conf", + &pdata->output_conf); + of_property_read_u8(np, "st,ch1-output-mapping", + &pdata->ch1_output_mapping); + of_property_read_u8(np, "st,ch2-output-mapping", + &pdata->ch2_output_mapping); + of_property_read_u8(np, "st,ch3-output-mapping", + &pdata->ch3_output_mapping); + + if (of_get_property(np, "st,thermal-warning-recovery", NULL)) + pdata->thermal_warning_recovery = 1; + if (of_get_property(np, "st,thermal-warning-adjustment", NULL)) + pdata->thermal_warning_adjustment = 1; + if (of_get_property(np, "st,fault-detect-recovery", NULL)) + pdata->fault_detect_recovery = 1; + + pdata->ffx_power_output_mode = STA350_FFX_PM_VARIABLE_DROP_COMP; + if (!of_property_read_string(np, "st,ffx-power-output-mode", + &ffx_power_mode)) { + int i, mode = -EINVAL; + + for (i = 0; i < ARRAY_SIZE(sta350_ffx_modes); i++) + if (!strcasecmp(ffx_power_mode, sta350_ffx_modes[i])) + mode = i; + + if (mode < 0) + dev_warn(dev, "Unsupported ffx output mode: %s\n", + ffx_power_mode); + else + pdata->ffx_power_output_mode = mode; + } + + tmp = 140; + of_property_read_u16(np, "st,drop-compensation-ns", &tmp); + pdata->drop_compensation_ns = clamp_t(u16, tmp, 0, 300) / 20; + + if (of_get_property(np, "st,overcurrent-warning-adjustment", NULL)) + pdata->oc_warning_adjustment = 1; + + /* CONFE */ + if (of_get_property(np, "st,max-power-use-mpcc", NULL)) + pdata->max_power_use_mpcc = 1; + + if (of_get_property(np, "st,max-power-correction", NULL)) + pdata->max_power_correction = 1; + + if (of_get_property(np, "st,am-reduction-mode", NULL)) + pdata->am_reduction_mode = 1; + + if (of_get_property(np, "st,odd-pwm-speed-mode", NULL)) + pdata->odd_pwm_speed_mode = 1; + + if (of_get_property(np, "st,distortion-compensation", NULL)) + pdata->distortion_compensation = 1; + + /* CONFF */ + if (of_get_property(np, "st,invalid-input-detect-mute", NULL)) + pdata->invalid_input_detect_mute = 1; + + sta350->pdata = pdata; + + return 0; +} +#endif + +static int sta350_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct sta350_priv *sta350; + int ret, i; + + sta350 = devm_kzalloc(dev, sizeof(struct sta350_priv), GFP_KERNEL); + if (!sta350) + return -ENOMEM; + + mutex_init(&sta350->coeff_lock); + sta350->pdata = dev_get_platdata(dev); + +#ifdef CONFIG_OF + if (dev->of_node) { + ret = sta350_probe_dt(dev, sta350); + if (ret < 0) + return ret; + } +#endif + + /* GPIOs */ + sta350->gpiod_nreset = devm_gpiod_get(dev, "reset"); + if (IS_ERR(sta350->gpiod_nreset)) { + ret = PTR_ERR(sta350->gpiod_nreset); + if (ret != -ENOENT && ret != -ENOSYS) + return ret; + + sta350->gpiod_nreset = NULL; + } else { + gpiod_direction_output(sta350->gpiod_nreset, 0); + } + + sta350->gpiod_power_down = devm_gpiod_get(dev, "power-down"); + if (IS_ERR(sta350->gpiod_power_down)) { + ret = PTR_ERR(sta350->gpiod_power_down); + if (ret != -ENOENT && ret != -ENOSYS) + return ret; + + sta350->gpiod_power_down = NULL; + } else { + gpiod_direction_output(sta350->gpiod_power_down, 0); + } + + /* regulators */ + for (i = 0; i < ARRAY_SIZE(sta350->supplies); i++) + sta350->supplies[i].supply = sta350_supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(sta350->supplies), + sta350->supplies); + if (ret < 0) { + dev_err(dev, "Failed to request supplies: %d\n", ret); + return ret; + } + + sta350->regmap = devm_regmap_init_i2c(i2c, &sta350_regmap); + if (IS_ERR(sta350->regmap)) { + ret = PTR_ERR(sta350->regmap); + dev_err(dev, "Failed to init regmap: %d\n", ret); + return ret; + } + + i2c_set_clientdata(i2c, sta350); + + ret = snd_soc_register_codec(dev, &sta350_codec, &sta350_dai, 1); + if (ret < 0) + dev_err(dev, "Failed to register codec (%d)\n", ret); + + return ret; +} + +static int sta350_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id sta350_i2c_id[] = { + { "sta350", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, sta350_i2c_id); + +static struct i2c_driver sta350_i2c_driver = { + .driver = { + .name = "sta350", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(st350_dt_ids), + }, + .probe = sta350_i2c_probe, + .remove = sta350_i2c_remove, + .id_table = sta350_i2c_id, +}; + +module_i2c_driver(sta350_i2c_driver); + +MODULE_DESCRIPTION("ASoC STA350 driver"); +MODULE_AUTHOR("Sven Brandau <info(a)brandau.biz>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/sta350.h b/sound/soc/codecs/sta350.h new file mode 100644 index 0000000..c3248f0 --- /dev/null +++ b/sound/soc/codecs/sta350.h @@ -0,0 +1,228 @@ +/* + * Codec driver for ST STA350 2.1-channel high-efficiency digital audio system + * + * Copyright: 2011 Raumfeld GmbH + * Author: Sven Brandau <info(a)brandau.biz> + * + * based on code from: + * Raumfeld GmbH + * Johannes Stezenbach <js(a)sig21.net> + * Wolfson Microelectronics PLC. + * Mark Brown <broonie(a)opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ +#ifndef _ASOC_STA_350_H +#define _ASOC_STA_350_H + +/* STA50 register addresses */ + +#define STA350_REGISTER_COUNT 0x4D +#define STA350_COEF_COUNT 62 + +#define STA350_CONFA 0x00 +#define STA350_CONFB 0x01 +#define STA350_CONFC 0x02 +#define STA350_CONFD 0x03 +#define STA350_CONFE 0x04 +#define STA350_CONFF 0x05 +#define STA350_MMUTE 0x06 +#define STA350_MVOL 0x07 +#define STA350_C1VOL 0x08 +#define STA350_C2VOL 0x09 +#define STA350_C3VOL 0x0a +#define STA350_AUTO1 0x0b +#define STA350_AUTO2 0x0c +#define STA350_AUTO3 0x0d +#define STA350_C1CFG 0x0e +#define STA350_C2CFG 0x0f +#define STA350_C3CFG 0x10 +#define STA350_TONE 0x11 +#define STA350_L1AR 0x12 +#define STA350_L1ATRT 0x13 +#define STA350_L2AR 0x14 +#define STA350_L2ATRT 0x15 +#define STA350_CFADDR2 0x16 +#define STA350_B1CF1 0x17 +#define STA350_B1CF2 0x18 +#define STA350_B1CF3 0x19 +#define STA350_B2CF1 0x1a +#define STA350_B2CF2 0x1b +#define STA350_B2CF3 0x1c +#define STA350_A1CF1 0x1d +#define STA350_A1CF2 0x1e +#define STA350_A1CF3 0x1f +#define STA350_A2CF1 0x20 +#define STA350_A2CF2 0x21 +#define STA350_A2CF3 0x22 +#define STA350_B0CF1 0x23 +#define STA350_B0CF2 0x24 +#define STA350_B0CF3 0x25 +#define STA350_CFUD 0x26 +#define STA350_MPCC1 0x27 +#define STA350_MPCC2 0x28 +#define STA350_DCC1 0x29 +#define STA350_DCC2 0x2a +#define STA350_FDRC1 0x2b +#define STA350_FDRC2 0x2c +#define STA350_STATUS 0x2d +/* reserved: 0x2d - 0x30 */ +#define STA350_EQCFG 0x31 +#define STA350_EATH1 0x32 +#define STA350_ERTH1 0x33 +#define STA350_EATH2 0x34 +#define STA350_ERTH2 0x35 +#define STA350_CONFX 0x36 +#define STA350_SVCA 0x37 +#define STA350_SVCB 0x38 +#define STA350_RMS0A 0x39 +#define STA350_RMS0B 0x3a +#define STA350_RMS0C 0x3b +#define STA350_RMS1A 0x3c +#define STA350_RMS1B 0x3d +#define STA350_RMS1C 0x3e +#define STA350_EVOLRES 0x3f +/* reserved: 0x40 - 0x47 */ +#define STA350_NSHAPE 0x48 +#define STA350_CTXB4B1 0x49 +#define STA350_CTXB7B5 0x4a +#define STA350_MISC1 0x4b +#define STA350_MISC2 0x4c + +/* 0x00 CONFA */ +#define STA350_CONFA_MCS_MASK 0x03 +#define STA350_CONFA_MCS_SHIFT 0 +#define STA350_CONFA_IR_MASK 0x18 +#define STA350_CONFA_IR_SHIFT 3 +#define STA350_CONFA_TWRB BIT(5) +#define STA350_CONFA_TWAB BIT(6) +#define STA350_CONFA_FDRB BIT(7) + +/* 0x01 CONFB */ +#define STA350_CONFB_SAI_MASK 0x0f +#define STA350_CONFB_SAI_SHIFT 0 +#define STA350_CONFB_SAIFB BIT(4) +#define STA350_CONFB_DSCKE BIT(5) +#define STA350_CONFB_C1IM BIT(6) +#define STA350_CONFB_C2IM BIT(7) + +/* 0x02 CONFC */ +#define STA350_CONFC_OM_MASK 0x03 +#define STA350_CONFC_OM_SHIFT 0 +#define STA350_CONFC_CSZ_MASK 0x3c +#define STA350_CONFC_CSZ_SHIFT 2 +#define STA350_CONFC_OCRB BIT(7) + +/* 0x03 CONFD */ +#define STA350_CONFD_HPB_SHIFT 0 +#define STA350_CONFD_DEMP_SHIFT 1 +#define STA350_CONFD_DSPB_SHIFT 2 +#define STA350_CONFD_PSL_SHIFT 3 +#define STA350_CONFD_BQL_SHIFT 4 +#define STA350_CONFD_DRC_SHIFT 5 +#define STA350_CONFD_ZDE_SHIFT 6 +#define STA350_CONFD_SME_SHIFT 7 + +/* 0x04 CONFE */ +#define STA350_CONFE_MPCV BIT(0) +#define STA350_CONFE_MPCV_SHIFT 0 +#define STA350_CONFE_MPC BIT(1) +#define STA350_CONFE_MPC_SHIFT 1 +#define STA350_CONFE_NSBW BIT(2) +#define STA350_CONFE_NSBW_SHIFT 2 +#define STA350_CONFE_AME BIT(3) +#define STA350_CONFE_AME_SHIFT 3 +#define STA350_CONFE_PWMS BIT(4) +#define STA350_CONFE_PWMS_SHIFT 4 +#define STA350_CONFE_DCCV BIT(5) +#define STA350_CONFE_DCCV_SHIFT 5 +#define STA350_CONFE_ZCE BIT(6) +#define STA350_CONFE_ZCE_SHIFT 6 +#define STA350_CONFE_SVE BIT(7) +#define STA350_CONFE_SVE_SHIFT 7 + +/* 0x05 CONFF */ +#define STA350_CONFF_OCFG_MASK 0x03 +#define STA350_CONFF_OCFG_SHIFT 0 +#define STA350_CONFF_IDE BIT(2) +#define STA350_CONFF_BCLE BIT(3) +#define STA350_CONFF_LDTE BIT(4) +#define STA350_CONFF_ECLE BIT(5) +#define STA350_CONFF_PWDN BIT(6) +#define STA350_CONFF_EAPD BIT(7) + +/* 0x06 MMUTE */ +#define STA350_MMUTE_MMUTE 0x01 +#define STA350_MMUTE_MMUTE_SHIFT 0 +#define STA350_MMUTE_C1M 0x02 +#define STA350_MMUTE_C1M_SHIFT 1 +#define STA350_MMUTE_C2M 0x04 +#define STA350_MMUTE_C2M_SHIFT 2 +#define STA350_MMUTE_C3M 0x08 +#define STA350_MMUTE_C3M_SHIFT 3 +#define STA350_MMUTE_LOC_MASK 0xC0 +#define STA350_MMUTE_LOC_SHIFT 6 + +/* 0x0b AUTO1 */ +#define STA350_AUTO1_AMGC_MASK 0x30 +#define STA350_AUTO1_AMGC_SHIFT 4 + +/* 0x0c AUTO2 */ +#define STA350_AUTO2_AMAME 0x01 +#define STA350_AUTO2_AMAM_MASK 0x0e +#define STA350_AUTO2_AMAM_SHIFT 1 +#define STA350_AUTO2_XO_MASK 0xf0 +#define STA350_AUTO2_XO_SHIFT 4 + +/* 0x0d AUTO3 */ +#define STA350_AUTO3_PEQ_MASK 0x1f +#define STA350_AUTO3_PEQ_SHIFT 0 + +/* 0x0e 0x0f 0x10 CxCFG */ +#define STA350_CxCFG_TCB_SHIFT 0 +#define STA350_CxCFG_EQBP_SHIFT 1 +#define STA350_CxCFG_VBP_SHIFT 2 +#define STA350_CxCFG_BO_SHIFT 3 +#define STA350_CxCFG_LS_SHIFT 4 +#define STA350_CxCFG_OM_MASK 0xc0 +#define STA350_CxCFG_OM_SHIFT 6 + +/* 0x11 TONE */ +#define STA350_TONE_BTC_SHIFT 0 +#define STA350_TONE_TTC_SHIFT 4 + +/* 0x12 0x13 0x14 0x15 limiter attack/release */ +#define STA350_LxA_SHIFT 0 +#define STA350_LxR_SHIFT 4 + +/* 0x26 CFUD */ +#define STA350_CFUD_W1 0x01 +#define STA350_CFUD_WA 0x02 +#define STA350_CFUD_R1 0x04 +#define STA350_CFUD_RA 0x08 + + +/* biquad filter coefficient table offsets */ +#define STA350_C1_BQ_BASE 0 +#define STA350_C2_BQ_BASE 20 +#define STA350_CH_BQ_NUM 4 +#define STA350_BQ_NUM_COEF 5 +#define STA350_XO_HP_BQ_BASE 40 +#define STA350_XO_LP_BQ_BASE 45 +#define STA350_C1_PRESCALE 50 +#define STA350_C2_PRESCALE 51 +#define STA350_C1_POSTSCALE 52 +#define STA350_C2_POSTSCALE 53 +#define STA350_C3_POSTSCALE 54 +#define STA350_TW_POSTSCALE 55 +#define STA350_C1_MIX1 56 +#define STA350_C1_MIX2 57 +#define STA350_C2_MIX1 58 +#define STA350_C2_MIX2 59 +#define STA350_C3_MIX1 60 +#define STA350_C3_MIX2 61 + +#endif /* _ASOC_STA_350_H */ -- 1.8.5.3

2 1

[alsa-devel] [PATCH 0/6] ASoC: davinci-mcasp: Dynamic AFIFO and DMA configuration
by Peter Ujfalusi 02 Apr '14

02 Apr '14

Hi, Apart from the first patch which is a rebased resend of a previous patch this series will change the way McASP driver will configure the AFIFO and thus the DMA burst. The AFIFO and DMA configuration will be in one function so it is going to be easier to see the code and debug if needed. The main change is that instead of static tx/rx numevt (which is provided via DT or pdata) we are going to switch to dynamic configuration. With this setup we do not need to place any constraint on the period or buffer size since the AFIFO numevt and the corresponding DMA burst size will be picked according to period size, active serializers and the preferred numevt level. The only case when the code can fail if the period size in words can not be divided evenly by the active serializers, which is unlikely to happen. Regards, Peter --- Peter Ujfalusi (6): ASoC: davinci-mcasp: Assign the dma_data earlier in dai_probe callback ASoC: davinci-mcasp: Fix debug typo in davinci_mcasp_hw_params() ASoC: davinci-mcasp: Simplify and clean up the AFIFO configuration code ASoC: davinci-mcasp: Configure the AFIFO and DMA burst size at the same place ASoC: davinic-mcasp: Adopt the AFIFO/DMA configuration to the stream (dynamic depth) ASoC: davinci-mcasp: Fine tune and correct the DMA burst configuration sound/soc/davinci/davinci-mcasp.c | 145 +++++++++++++++++++++++--------------- sound/soc/davinci/davinci-mcasp.h | 1 + 2 files changed, 89 insertions(+), 57 deletions(-) -- 1.9.1

2 7

[alsa-devel] [PATCHv2 00/11] Remove set_cache_io entirely from ASoC probe
by Xiubo Li 02 Apr '14

02 Apr '14

Change in v2: - Add .get_regmap() support for struct snd_soc_codec_driver. Xiubo Li (11): ASoC: core: Move the default regmap I/O setting to snd_soc_register_codec() ASoc: 88pm860x: Remove the set_cache_io() entirely from ASoC probe. ASoc: cq93vc: Remove the set_cache_io() entirely from ASoC probe. ASoc: mc13783: Remove the set_cache_io() entirely from ASoC probe. ASoc: si476x: Remove the set_cache_io() entirely from ASoC probe. ASoc: wm5102: Remove the set_cache_io() entirely from ASoC probe. ASoc: wm5110: Remove the set_cache_io() entirely from ASoC probe. ASoc: wm8350: Remove the set_cache_io() entirely from ASoC probe. ASoc: wm8400: Remove the set_cache_io() entirely from ASoC probe. ASoc: wm8994: Remove the set_cache_io() entirely from ASoC probe. ASoc: wm8997: Remove the set_cache_io() entirely from ASoC probe. include/sound/soc.h | 1 + sound/soc/codecs/88pm860x-codec.c | 12 ++++++++---- sound/soc/codecs/cq93vc.c | 10 ++++++++-- sound/soc/codecs/mc13783.c | 14 ++++++-------- sound/soc/codecs/si476x.c | 14 ++++++-------- sound/soc/codecs/wm5102.c | 12 ++++++++---- sound/soc/codecs/wm5110.c | 12 ++++++++---- sound/soc/codecs/wm8350.c | 10 ++++++++-- sound/soc/codecs/wm8400.c | 10 ++++++++-- sound/soc/codecs/wm8994.c | 10 ++++++++-- sound/soc/codecs/wm8997.c | 13 ++++++++----- sound/soc/soc-core.c | 28 ++++++++++++++++++---------- sound/soc/soc-io.c | 9 +++------ 13 files changed, 98 insertions(+), 57 deletions(-) -- 1.8.4

4 14

[alsa-devel] [PATCH 00/13] ASoC: Move IO and kcontrols to the component level
by Lars-Peter Clausen 02 Apr '14

02 Apr '14

Hi, This series is the first step towards full componentisation of the ASoC core. It moves both the IO abstraction layers within ASoC as well as the standard set of kcontrols to the component level. This for example means we can get rid of constructs like if (w->codec) snd_soc_read(....) else if(w->platform) snd_soc_platform_read(...) Moving the kcontrols to the component level means we can use the same implementation also for other non-CODEC components. E.g. there seems to be an increasing amount of CPU components that have basic signal processing and things like volume controls etc. whose register layout is similar to those used in CODECs. Currently each CPU component driver re-implements these controls by hand. The first two patches introduce two new helper functions which hide the actual implementation on how the CODEC or platform struct that register a control can be obtained from the control. This means that when the actual implementation is changed only the two helper functions need to be updated and not every single driver. The patches that follow that are just cleanups removing unused IO stuff and move all IO functions to soc-io.c. The next step is to make platforms also components. And then finally first the IO abstraction layers in ASoC are unified at the component level and then on top of that the kcontrol helpers are moved to the component level. The series depends on quite a few topic branches related to changes to the core and cleanups for individual drivers. It is probably best to place it on top of asoc-v3.15-2. The patch that moves the kcontrols to the component level also has a runtime dependency on the not yet applied patches that move the ams-delta and mfld_machine controls to the card level. - Lars Lars-Peter Clausen (13): ASoC: Add snd_soc_kcontrol_codec() helper function ASoC: Add snd_soc_kcontrol_platform() helper function ASoC: Prepare SOC_SINGLE_XR_SX controls for regmap ASoC: Move IO functions to soc-io.c ASoC: Drop ASoC level caching from hw_write/hw_read ASoC: Remove IO register modifier callbacks ASoC: Add helper function to cast component back to CODEC ASoC: Track which components have been registered with snd_soc_register_component() ASoC: Let snd_soc_platform subclass snd_soc_component ASoC: Move IO abstraction to the component level ASoC: Move standard kcontrol helpers to the component level ASoC: Remove snd_soc_update_bits_locked() ASoC: dapm: Rename soc_widget_update_bits_locked() to soc_widget_update_bits() include/sound/soc-dapm.h | 1 + include/sound/soc.h | 114 +++++++-- sound/soc/codecs/88pm860x-codec.c | 8 +- sound/soc/codecs/ab8500-codec.c | 12 +- sound/soc/codecs/adav80x.c | 4 +- sound/soc/codecs/ak4641.c | 4 +- sound/soc/codecs/cs4270.c | 2 +- sound/soc/codecs/cs4271.c | 4 +- sound/soc/codecs/cs42l51.c | 4 +- sound/soc/codecs/da7210.c | 4 +- sound/soc/codecs/da7213.c | 4 +- sound/soc/codecs/da732x.c | 4 +- sound/soc/codecs/da9055.c | 2 +- sound/soc/codecs/lm4857.c | 4 +- sound/soc/codecs/max9768.c | 4 +- sound/soc/codecs/max98088.c | 12 +- sound/soc/codecs/max98090.c | 4 +- sound/soc/codecs/max98095.c | 16 +- sound/soc/codecs/pcm1681.c | 4 +- sound/soc/codecs/rt5631.c | 4 +- sound/soc/codecs/sgtl5000.c | 4 +- sound/soc/codecs/sta32x.c | 4 +- sound/soc/codecs/tas5086.c | 4 +- sound/soc/codecs/tlv320aic23.c | 4 +- sound/soc/codecs/tlv320dac33.c | 4 +- sound/soc/codecs/twl4030.c | 10 +- sound/soc/codecs/twl6040.c | 8 +- sound/soc/codecs/wl1273.c | 12 +- sound/soc/codecs/wm2000.c | 8 +- sound/soc/codecs/wm8350.c | 4 +- sound/soc/codecs/wm8400.c | 2 +- sound/soc/codecs/wm8580.c | 2 +- sound/soc/codecs/wm8731.c | 4 +- sound/soc/codecs/wm8753.c | 4 +- sound/soc/codecs/wm8804.c | 4 +- sound/soc/codecs/wm8903.c | 4 +- sound/soc/codecs/wm8904.c | 14 +- sound/soc/codecs/wm8955.c | 4 +- sound/soc/codecs/wm8958-dsp2.c | 32 +-- sound/soc/codecs/wm8960.c | 4 +- sound/soc/codecs/wm8962.c | 8 +- sound/soc/codecs/wm8983.c | 4 +- sound/soc/codecs/wm8985.c | 4 +- sound/soc/codecs/wm8990.c | 2 +- sound/soc/codecs/wm8991.c | 2 +- sound/soc/codecs/wm8994.c | 10 +- sound/soc/codecs/wm8996.c | 4 +- sound/soc/codecs/wm9081.c | 4 +- sound/soc/codecs/wm_adsp.c | 4 +- sound/soc/codecs/wm_hubs.c | 2 +- sound/soc/intel/sst-haswell-pcm.c | 8 +- sound/soc/soc-cache.c | 2 - sound/soc/soc-core.c | 520 ++++++++++++++++---------------------- sound/soc/soc-dapm.c | 91 +------ sound/soc/soc-io.c | 220 +++++++++++++--- 55 files changed, 644 insertions(+), 592 deletions(-) -- 1.8.0

5 34

[alsa-devel] [PATCH v8] MIDI driver for Behringer BCD2000 USB device
by Mario Kicherer 02 Apr '14

02 Apr '14

This patch adds initial support for the Behringer BCD2000 USB DJ controller. At the moment, only the MIDI part of the device is working, i.e. knobs, buttons and LEDs. I also plan to add support for the audio part, but I assume that this will require more effort than the rather simple MIDI interface. Progress can be tracked at https://github.com/anyc/snd-usb-bcd2000. Daniel or Clemens: could I get another ACK on this? Thank you both! Best regards, Mario Changes since v7: - replaced snd_card_create with snd_card_new Changes since v6: - applied more style improvements Changes since v5: - use kernel bitmap functions for devices_used Changes since v4: - devices_used as array to support arbitrary number of SNDRV_CARDS - removed unused array "enable" Changes since v3: - applied style and snd_printk changes as suggested by Daniel Mack Changes since v2: - applied more changes from Daniel Mack and Clemens Ladisch Changes since v1: - fixed the various code style issues, thanks to Daniel Mack and checkpatch.pl. Signed-off-by: Mario Kicherer <dev(a)kicherer.org> --- sound/usb/Kconfig | 13 ++ sound/usb/Makefile | 2 +- sound/usb/bcd2000/Makefile | 3 + sound/usb/bcd2000/bcd2000.c | 477 ++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 494 insertions(+), 1 deletion(-) create mode 100644 sound/usb/bcd2000/Makefile create mode 100644 sound/usb/bcd2000/bcd2000.c diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index e05a86b..d393153 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -147,5 +147,18 @@ config SND_USB_HIFACE To compile this driver as a module, choose M here: the module will be called snd-usb-hiface. +config SND_BCD2000 + tristate "Behringer BCD2000 MIDI driver" + select SND_RAWMIDI + help + Say Y here to include MIDI support for the Behringer BCD2000 DJ + controller. + + Audio support is still work-in-progress at + https://github.com/anyc/snd-usb-bcd2000 + + To compile this driver as a module, choose M here: the module + will be called snd-bcd2000. + endif # SND_USB diff --git a/sound/usb/Makefile b/sound/usb/Makefile index abe668f..2b92f0d 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -23,4 +23,4 @@ obj-$(CONFIG_SND_USB_UA101) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_USX2Y) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_US122L) += snd-usbmidi-lib.o -obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ hiface/ +obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ hiface/ bcd2000/ diff --git a/sound/usb/bcd2000/Makefile b/sound/usb/bcd2000/Makefile new file mode 100644 index 0000000..f09ccc0 --- /dev/null +++ b/sound/usb/bcd2000/Makefile @@ -0,0 +1,3 @@ +snd-bcd2000-y := bcd2000.o + +obj-$(CONFIG_SND_BCD2000) += snd-bcd2000.o \ No newline at end of file diff --git a/sound/usb/bcd2000/bcd2000.c b/sound/usb/bcd2000/bcd2000.c new file mode 100644 index 0000000..9fda92d --- /dev/null +++ b/sound/usb/bcd2000/bcd2000.c @@ -0,0 +1,477 @@ +/* + * Behringer BCD2000 driver + * + * Copyright (C) 2014 Mario Kicherer (dev(a)kicherer.org) + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * 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/kernel.h> +#include <linux/errno.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <linux/module.h> +#include <linux/bitmap.h> +#include <linux/usb.h> +#include <linux/usb/audio.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/rawmidi.h> + +#define PREFIX "snd-bcd2000: " +#define BUFSIZE 64 + +static struct usb_device_id id_table[] = { + { USB_DEVICE(0x1397, 0x00bd) }, + { }, +}; + +static unsigned char device_cmd_prefix[] = {0x03, 0x00}; + +static unsigned char bcd2000_init_sequence[] = { + 0x07, 0x00, 0x00, 0x00, 0x78, 0x48, 0x1c, 0x81, + 0xc4, 0x00, 0x00, 0x00, 0x5e, 0x53, 0x4a, 0xf7, + 0x18, 0xfa, 0x11, 0xff, 0x6c, 0xf3, 0x90, 0xff, + 0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, + 0x18, 0xfa, 0x11, 0xff, 0x14, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0xf2, 0x34, 0x4a, 0xf7, + 0x18, 0xfa, 0x11, 0xff +}; + +struct bcd2000 { + struct usb_device *dev; + struct snd_card *card; + struct usb_interface *intf; + int card_index; + + int midi_out_active; + struct snd_rawmidi *rmidi; + struct snd_rawmidi_substream *midi_receive_substream; + struct snd_rawmidi_substream *midi_out_substream; + + unsigned char midi_in_buf[BUFSIZE]; + unsigned char midi_out_buf[BUFSIZE]; + + struct urb *midi_out_urb; + struct urb *midi_in_urb; + + struct usb_anchor anchor; +}; + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; + +static DEFINE_MUTEX(devices_mutex); +DECLARE_BITMAP(devices_used, SNDRV_CARDS); +static struct usb_driver bcd2000_driver; + +#ifdef CONFIG_SND_DEBUG +static void bcd2000_dump_buffer(const char *prefix, const char *buf, int len) +{ + print_hex_dump(KERN_DEBUG, prefix, + DUMP_PREFIX_NONE, 16, 1, + buf, len, false); +} +#else +static void bcd2000_dump_buffer(const char *prefix, const char *buf, int len) {} +#endif + +static int bcd2000_midi_input_open(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +static int bcd2000_midi_input_close(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +/* register midi substream */ +static void bcd2000_midi_input_trigger(struct snd_rawmidi_substream *substream, + int up) +{ + struct bcd2000 *bcd2k = substream->rmidi->private_data; + bcd2k->midi_receive_substream = up ? substream : NULL; +} + +static void bcd2000_midi_handle_input(struct bcd2000 *bcd2k, + const unsigned char *buf, unsigned int buf_len) +{ + unsigned int payload_length, tocopy; + struct snd_rawmidi_substream *midi_receive_substream; + + midi_receive_substream = ACCESS_ONCE(bcd2k->midi_receive_substream); + if (!midi_receive_substream) + return; + + bcd2000_dump_buffer(PREFIX "received from device: ", buf, buf_len); + + if (buf_len < 2) + return; + + payload_length = buf[0]; + + /* ignore packets without payload */ + if (payload_length == 0) + return; + + tocopy = min(payload_length, buf_len-1); + + bcd2000_dump_buffer(PREFIX "sending to userspace: ", + &buf[1], tocopy); + + snd_rawmidi_receive(midi_receive_substream, + &buf[1], tocopy); +} + +static void bcd2000_midi_send(struct bcd2000 *bcd2k) +{ + int len, ret; + struct snd_rawmidi_substream *midi_out_substream; + + BUILD_BUG_ON(sizeof(device_cmd_prefix) >= BUFSIZE); + + midi_out_substream = ACCESS_ONCE(bcd2k->midi_out_substream); + if (!midi_out_substream) + return; + + /* copy command prefix bytes */ + memcpy(bcd2k->midi_out_buf, device_cmd_prefix, + sizeof(device_cmd_prefix)); + + /* + * get MIDI packet and leave space for command prefix + * and payload length + */ + len = snd_rawmidi_transmit(midi_out_substream, + bcd2k->midi_out_buf + 3, BUFSIZE - 3); + + if (len < 0) + dev_err(&bcd2k->dev->dev, "%s: snd_rawmidi_transmit error %d\n", + __func__, len); + + if (len <= 0) + return; + + /* set payload length */ + bcd2k->midi_out_buf[2] = len; + bcd2k->midi_out_urb->transfer_buffer_length = BUFSIZE; + + bcd2000_dump_buffer(PREFIX "sending to device: ", + bcd2k->midi_out_buf, len+3); + + /* send packet to the BCD2000 */ + ret = usb_submit_urb(bcd2k->midi_out_urb, GFP_ATOMIC); + if (ret < 0) + dev_err(&bcd2k->dev->dev, PREFIX + "%s (%p): usb_submit_urb() failed, ret=%d, len=%d\n", + __func__, midi_out_substream, ret, len); + else + bcd2k->midi_out_active = 1; +} + +static int bcd2000_midi_output_open(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +static int bcd2000_midi_output_close(struct snd_rawmidi_substream *substream) +{ + struct bcd2000 *bcd2k = substream->rmidi->private_data; + + if (bcd2k->midi_out_active) { + usb_kill_urb(bcd2k->midi_out_urb); + bcd2k->midi_out_active = 0; + } + + return 0; +} + +/* register midi substream */ +static void bcd2000_midi_output_trigger(struct snd_rawmidi_substream *substream, + int up) +{ + struct bcd2000 *bcd2k = substream->rmidi->private_data; + + if (up) { + bcd2k->midi_out_substream = substream; + /* check if there is data userspace wants to send */ + if (!bcd2k->midi_out_active) + bcd2000_midi_send(bcd2k); + } else { + bcd2k->midi_out_substream = NULL; + } +} + +static void bcd2000_output_complete(struct urb *urb) +{ + struct bcd2000 *bcd2k = urb->context; + + bcd2k->midi_out_active = 0; + + if (urb->status) + dev_warn(&urb->dev->dev, + PREFIX "output urb->status: %d\n", urb->status); + + if (urb->status == -ESHUTDOWN) + return; + + /* check if there is more data userspace wants to send */ + bcd2000_midi_send(bcd2k); +} + +static void bcd2000_input_complete(struct urb *urb) +{ + int ret; + struct bcd2000 *bcd2k = urb->context; + + if (urb->status) + dev_warn(&urb->dev->dev, + PREFIX "input urb->status: %i\n", urb->status); + + if (!bcd2k || urb->status == -ESHUTDOWN) + return; + + if (urb->actual_length > 0) + bcd2000_midi_handle_input(bcd2k, urb->transfer_buffer, + urb->actual_length); + + /* acknowledge received packet */ + ret = usb_submit_urb(bcd2k->midi_in_urb, GFP_ATOMIC); + if (ret < 0) + dev_err(&bcd2k->dev->dev, PREFIX + "%s: usb_submit_urb() failed, ret=%d\n", + __func__, ret); +} + +static struct snd_rawmidi_ops bcd2000_midi_output = { + .open = bcd2000_midi_output_open, + .close = bcd2000_midi_output_close, + .trigger = bcd2000_midi_output_trigger, +}; + +static struct snd_rawmidi_ops bcd2000_midi_input = { + .open = bcd2000_midi_input_open, + .close = bcd2000_midi_input_close, + .trigger = bcd2000_midi_input_trigger, +}; + +static void bcd2000_init_device(struct bcd2000 *bcd2k) +{ + int ret; + + init_usb_anchor(&bcd2k->anchor); + usb_anchor_urb(bcd2k->midi_out_urb, &bcd2k->anchor); + usb_anchor_urb(bcd2k->midi_in_urb, &bcd2k->anchor); + + /* copy init sequence into buffer */ + memcpy(bcd2k->midi_out_buf, bcd2000_init_sequence, 52); + bcd2k->midi_out_urb->transfer_buffer_length = 52; + + /* submit sequence */ + ret = usb_submit_urb(bcd2k->midi_out_urb, GFP_KERNEL); + if (ret < 0) + dev_err(&bcd2k->dev->dev, PREFIX + "%s: usb_submit_urb() out failed, ret=%d: ", + __func__, ret); + else + bcd2k->midi_out_active = 1; + + /* send empty packet to enable button and controller events */ + ret = usb_submit_urb(bcd2k->midi_in_urb, GFP_KERNEL); + if (ret < 0) + dev_err(&bcd2k->dev->dev, PREFIX + "%s: usb_submit_urb() in failed, ret=%d: ", + __func__, ret); + + /* ensure initialization is finished */ + usb_wait_anchor_empty_timeout(&bcd2k->anchor, 1000); +} + +static int bcd2000_init_midi(struct bcd2000 *bcd2k) +{ + int ret; + struct snd_rawmidi *rmidi; + + ret = snd_rawmidi_new(bcd2k->card, bcd2k->card->shortname, 0, + 1, /* output */ + 1, /* input */ + &rmidi); + + if (ret < 0) + return ret; + + strlcpy(rmidi->name, bcd2k->card->shortname, sizeof(rmidi->name)); + + rmidi->info_flags = SNDRV_RAWMIDI_INFO_DUPLEX; + rmidi->private_data = bcd2k; + + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT; + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, + &bcd2000_midi_output); + + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_INPUT; + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, + &bcd2000_midi_input); + + bcd2k->rmidi = rmidi; + + bcd2k->midi_in_urb = usb_alloc_urb(0, GFP_KERNEL); + bcd2k->midi_out_urb = usb_alloc_urb(0, GFP_KERNEL); + + if (!bcd2k->midi_in_urb || !bcd2k->midi_out_urb) { + dev_err(&bcd2k->dev->dev, PREFIX "usb_alloc_urb failed\n"); + return -ENOMEM; + } + + usb_fill_int_urb(bcd2k->midi_in_urb, bcd2k->dev, + usb_rcvintpipe(bcd2k->dev, 0x81), + bcd2k->midi_in_buf, BUFSIZE, + bcd2000_input_complete, bcd2k, 1); + + usb_fill_int_urb(bcd2k->midi_out_urb, bcd2k->dev, + usb_sndintpipe(bcd2k->dev, 0x1), + bcd2k->midi_out_buf, BUFSIZE, + bcd2000_output_complete, bcd2k, 1); + + bcd2000_init_device(bcd2k); + + return 0; +} + +static void bcd2000_free_usb_related_resources(struct bcd2000 *bcd2k, + struct usb_interface *interface) +{ + /* usb_kill_urb not necessary, urb is aborted automatically */ + + usb_free_urb(bcd2k->midi_out_urb); + usb_free_urb(bcd2k->midi_in_urb); + + if (bcd2k->intf) { + usb_set_intfdata(bcd2k->intf, NULL); + bcd2k->intf = NULL; + } +} + +static int bcd2000_probe(struct usb_interface *interface, + const struct usb_device_id *usb_id) +{ + struct snd_card *card; + struct bcd2000 *bcd2k; + unsigned int card_index; + char usb_path[32]; + int err; + + mutex_lock(&devices_mutex); + + for (card_index = 0; card_index < SNDRV_CARDS; ++card_index) + if (!test_bit(card_index, devices_used)) + break; + + if (card_index >= SNDRV_CARDS) { + mutex_unlock(&devices_mutex); + return -ENOENT; + } + + err = snd_card_new(&interface->dev, index[card_index], id[card_index], + THIS_MODULE, sizeof(*bcd2k), &card); + if (err < 0) { + mutex_unlock(&devices_mutex); + return err; + } + + bcd2k = card->private_data; + bcd2k->dev = interface_to_usbdev(interface); + bcd2k->card = card; + bcd2k->card_index = card_index; + bcd2k->intf = interface; + + snd_card_set_dev(card, &interface->dev); + + strncpy(card->driver, "snd-bcd2000", sizeof(card->driver)); + strncpy(card->shortname, "BCD2000", sizeof(card->shortname)); + usb_make_path(bcd2k->dev, usb_path, sizeof(usb_path)); + snprintf(bcd2k->card->longname, sizeof(bcd2k->card->longname), + "Behringer BCD2000 at %s", + usb_path); + + err = bcd2000_init_midi(bcd2k); + if (err < 0) + goto probe_error; + + err = snd_card_register(card); + if (err < 0) + goto probe_error; + + usb_set_intfdata(interface, bcd2k); + set_bit(card_index, devices_used); + + mutex_unlock(&devices_mutex); + return 0; + +probe_error: + dev_info(&bcd2k->dev->dev, PREFIX "error during probing"); + bcd2000_free_usb_related_resources(bcd2k, interface); + snd_card_free(card); + mutex_unlock(&devices_mutex); + return err; +} + +static void bcd2000_disconnect(struct usb_interface *interface) +{ + struct bcd2000 *bcd2k = usb_get_intfdata(interface); + + if (!bcd2k) + return; + + mutex_lock(&devices_mutex); + + /* make sure that userspace cannot create new requests */ + snd_card_disconnect(bcd2k->card); + + bcd2000_free_usb_related_resources(bcd2k, interface); + + clear_bit(bcd2k->card_index, devices_used); + + snd_card_free_when_closed(bcd2k->card); + + mutex_unlock(&devices_mutex); +} + +static struct usb_driver bcd2000_driver = { + .name = "snd-bcd2000", + .probe = bcd2000_probe, + .disconnect = bcd2000_disconnect, + .id_table = id_table, +}; + +static int __init bcd2000_init(void) +{ + int retval = 0; + + retval = usb_register(&bcd2000_driver); + if (retval) + pr_info(PREFIX "usb_register failed. Error: %d", retval); + return retval; +} + +static void __exit bcd2000_exit(void) +{ + usb_deregister(&bcd2000_driver); +} + +module_init(bcd2000_init); +module_exit(bcd2000_exit); + +MODULE_DEVICE_TABLE(usb, id_table); +MODULE_AUTHOR("Mario Kicherer, dev(a)kicherer.org"); +MODULE_DESCRIPTION("Behringer BCD2000 driver"); +MODULE_LICENSE("GPL"); -- 1.8.3.2

2 1

Re: [alsa-devel] [PATCH] ASoC: DAPM: Add support for multi register mux
by Arun Shamanna Lakshmi 02 Apr '14

02 Apr '14

> -----Original Message----- > From: Lars-Peter Clausen [mailto:lars@metafoo.de] > Sent: Tuesday, April 01, 2014 12:48 AM > To: Arun Shamanna Lakshmi > Cc: lgirdwood(a)gmail.com; broonie(a)kernel.org; swarren(a)wwwdotorg.org; > perex(a)perex.cz; tiwai(a)suse.de; alsa- devel(a)alsa-project.org; > linux-kernel(a)vger.kernel.org; Songhee Baek > Subject: Re: [PATCH] ASoC: DAPM: Add support for multi register mux > > On 04/01/2014 08:21 AM, Arun Shamanna Lakshmi wrote: > > Modify soc_enum struct to handle pointers for reg and mask. Add dapm > > get and put APIs for multi register mux with one hot encoding. > > > > Signed-off-by: Arun Shamanna Lakshmi <aruns(a)nvidia.com> > > Signed-off-by: Songhee Baek <sbaek(a)nvidia.com> > > Looks in my opinion much better than the previous version :) Just a > few minor issues, comments inline > > > --- > > include/sound/soc-dapm.h | 10 ++++ > > include/sound/soc.h | 22 +++++-- > > sound/soc/soc-core.c | 12 ++-- > > sound/soc/soc-dapm.c | 143 > +++++++++++++++++++++++++++++++++++++++++----- > > 4 files changed, 162 insertions(+), 25 deletions(-) > > > > diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h > index > > ef78f56..983b0ab 100644 > > --- a/include/sound/soc-dapm.h > > +++ b/include/sound/soc-dapm.h > > @@ -305,6 +305,12 @@ struct device; > > .get = snd_soc_dapm_get_enum_double, \ > > .put = snd_soc_dapm_put_enum_double, \ > > .private_value = (unsigned long)&xenum } > > +#define SOC_DAPM_ENUM_WIDE(xname, xenum) \ > > maybe just call it ENUM_ONEHOT, since it doesn't actually have to be > more than one register. > > [...] > > diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index > > cd52d52..aba0094 100644 > > --- a/sound/soc/soc-core.c > > +++ b/sound/soc/soc-core.c > > @@ -2601,12 +2601,12 @@ int snd_soc_get_enum_double(struct > snd_kcontrol *kcontrol, > > unsigned int val, item; > > unsigned int reg_val; > > > > - reg_val = snd_soc_read(codec, e->reg); > > - val = (reg_val >> e->shift_l) & e->mask; > > + reg_val = snd_soc_read(codec, e->reg[0]); > > + val = (reg_val >> e->shift_l) & e->mask[0]; > > item = snd_soc_enum_val_to_item(e, val); > > ucontrol->value.enumerated.item[0] = item; > > if (e->shift_l != e->shift_r) { > > - val = (reg_val >> e->shift_l) & e->mask; > > + val = (reg_val >> e->shift_l) & e->mask[0]; > > item = snd_soc_enum_val_to_item(e, val); > > ucontrol->value.enumerated.item[1] = item; > > } > > @@ -2636,15 +2636,15 @@ int snd_soc_put_enum_double(struct > snd_kcontrol *kcontrol, > > if (item[0] >= e->items) > > return -EINVAL; > > val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l; > > - mask = e->mask << e->shift_l; > > + mask = e->mask[0] << e->shift_l; > > if (e->shift_l != e->shift_r) { > > if (item[1] >= e->items) > > return -EINVAL; > > val |= snd_soc_enum_item_to_val(e, item[1]) << e- shift_r; > > - mask |= e->mask << e->shift_r; > > + mask |= e->mask[0] << e->shift_r; > > } > > > > - return snd_soc_update_bits_locked(codec, e->reg, mask, val); > > + return snd_soc_update_bits_locked(codec, e->reg[0], mask, val); > > } > > EXPORT_SYMBOL_GPL(snd_soc_put_enum_double); > > > > diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index > > c8a780d..4d2b35c 100644 > > --- a/sound/soc/soc-dapm.c > > +++ b/sound/soc/soc-dapm.c > > @@ -514,9 +514,9 @@ static int dapm_connect_mux(struct > snd_soc_dapm_context *dapm, > > unsigned int val, item; > > int i; > > > > - if (e->reg != SND_SOC_NOPM) { > > - soc_widget_read(dest, e->reg, &val); > > - val = (val >> e->shift_l) & e->mask; > > + if (e->reg[0] != SND_SOC_NOPM) { > > + soc_widget_read(dest, e->reg[0], &val); > > + val = (val >> e->shift_l) & e->mask[0]; > > item = snd_soc_enum_val_to_item(e, val); > > This probably should handle the new enum type as well. You'll probably > need some kind of flag in the struct to distinguish between the two > enum types. Any suggestion on the flag name ? > > > } else { > > /* since a virtual mux has no backing registers to > [...] > > /** > > + * snd_soc_dapm_get_enum_wide - dapm semi enumerated multiple > > + registers > > What's a semi-enumerated register? > > > + * mixer get callback > > + * @kcontrol: mixer control > > + * @ucontrol: control element information > > + * > > + * Callback to get the value of a dapm semi enumerated multiple > > +register mixer > > + * control. > > + * > > + * semi enumerated multiple registers mixer: > > + * the mixer has multiple registers to set the enumerated items. > > +The enumerated > > + * items are referred as values. > > + * Can be used for handling bit field coded enumeration for example. > > + * > > + * Returns 0 for success. > > + */ > > +int snd_soc_dapm_get_enum_wide(struct snd_kcontrol *kcontrol, > > + struct snd_ctl_elem_value *ucontrol) { > > + struct snd_soc_codec *codec = > snd_soc_dapm_kcontrol_codec(kcontrol); > > + struct soc_enum *e = (struct soc_enum *)kcontrol- > >private_value; > > + unsigned int reg_val, val, bit_pos = 0, reg_idx; > > + > > + for (reg_idx = 0; reg_idx < e->num_regs; reg_idx++) { > > + reg_val = snd_soc_read(codec, e->reg[reg_idx]); > > + val = reg_val & e->mask[reg_idx]; > > + if (val != 0) { > > + bit_pos = ffs(val) + (e->reg_width * reg_idx); > > Should be __ffs. __ffs returns the bits zero-indexed and ffs one-indexed. > That will work better for cases where there is not additional value > table necessary, since it means bit 1 maps to value 0. > > > + break; > > + } > > + } > > + > > + ucontrol->value.enumerated.item[0] = > > + snd_soc_enum_val_to_item(e, bit_pos); > > + > > + return 0; > > +} > [...] > > +int snd_soc_dapm_put_enum_wide(struct snd_kcontrol *kcontrol, > > + struct snd_ctl_elem_value *ucontrol) { > > + struct snd_soc_codec *codec = > snd_soc_dapm_kcontrol_codec(kcontrol); > > + struct snd_soc_card *card = codec->card; > > + struct soc_enum *e = (struct soc_enum *)kcontrol- > >private_value; > > + unsigned int *item = ucontrol->value.enumerated.item; > > + unsigned int change = 0, reg_idx = 0, value, bit_pos; > > + struct snd_soc_dapm_update update; > > + int ret = 0, reg_val = 0, i; > > + > > + if (item[0] >= e->items) > > + return -EINVAL; > > + > > + value = snd_soc_enum_item_to_val(e, item[0]); > > + > > + if (value) { > > + /* get the register index and value to set */ > > + reg_idx = (value - 1) / e->reg_width; > > + bit_pos = (value - 1) % e->reg_width; > > Changing the ffs to __ffs also means you can drop the ' - 1' here. > > Also e->reg_width should be (codec->val_bytes * 8) and reg_width field > should be dropped from the enum struct. > > > + reg_val = BIT(bit_pos); > > + } > > + > > + for (i = 0; i < e->num_regs; i++) { > > + if (i == reg_idx) { > > + change = snd_soc_test_bits(codec, e->reg[i], > > + e->mask[i], > reg_val); > > + > > + } else { > > + /* accumulate the change to update the DAPM > path > > + when none is selected */ > > + change += snd_soc_test_bits(codec, e->reg[i], > > + e->mask[i], 0); > > change |= > > > + > > + /* clear the register when not selected */ > > + snd_soc_write(codec, e->reg[i], 0); > > I think this should happen as part of the DAPM update sequence like > you had earlier. Some special care should probably be take to make > sure that you de-select the previous mux input before selecting the > new one if the new one is in a different register than the previous one. I am not sure I follow this part. We are clearing the 'not selected' registers before we set the one we want. Do you want us to loop the logic of soc_dapm_mux_update_power for each register ? or do you want to change the dapm_update structure so that it takes all the regs, masks, and values together ? > > > + } > > + } > > + > > + mutex_lock_nested(&card->dapm_mutex, > SND_SOC_DAPM_CLASS_RUNTIME); > > + > [...]

3 8

[alsa-devel] [PATCH] ASoC: core: Only kmemdup binary control buffer if masking
by Charles Keepax 02 Apr '14

02 Apr '14

When writing a binary control we may apply a mask to the first register, as this requires modifying the data the buffer is duplicated, currently this is done for all binary control writes. As most binary controls don't use the mask facility and thus can freely use the original buffer, avoid the kmemdup for these cases. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 86 ++++++++++++++++++++++++------------------------- 1 files changed, 42 insertions(+), 44 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index caebd63..275bd71 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3229,67 +3229,65 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, len = params->num_regs * codec->val_bytes; - data = kmemdup(ucontrol->value.bytes.data, len, GFP_KERNEL | GFP_DMA); - if (!data) - return -ENOMEM; + if (!params->mask) + return regmap_raw_write(codec->control_data, params->base, + ucontrol->value.bytes.data, len); /* * If we've got a mask then we need to preserve the register * bits. We shouldn't modify the incoming data so take a * copy. */ - if (params->mask) { - ret = regmap_read(codec->control_data, params->base, &val); - if (ret != 0) - goto out; + data = kmemdup(ucontrol->value.bytes.data, len, + GFP_KERNEL | GFP_DMA); + if (!data) + return -ENOMEM; - val &= params->mask; + ret = regmap_read(codec->control_data, params->base, &val); + if (ret != 0) + goto out; - switch (codec->val_bytes) { - case 1: - ((u8 *)data)[0] &= ~params->mask; - ((u8 *)data)[0] |= val; - break; - case 2: - mask = ~params->mask; - ret = regmap_parse_val(codec->control_data, - &mask, &mask); - if (ret != 0) - goto out; + val &= params->mask; - ((u16 *)data)[0] &= mask; + switch (codec->val_bytes) { + case 1: + ((u8 *)data)[0] &= ~params->mask; + ((u8 *)data)[0] |= val; + break; + case 2: + mask = ~params->mask; + ret = regmap_parse_val(codec->control_data, &mask, &mask); + if (ret != 0) + goto out; - ret = regmap_parse_val(codec->control_data, - &val, &val); - if (ret != 0) - goto out; + ((u16 *)data)[0] &= mask; - ((u16 *)data)[0] |= val; - break; - case 4: - mask = ~params->mask; - ret = regmap_parse_val(codec->control_data, - &mask, &mask); - if (ret != 0) - goto out; + ret = regmap_parse_val(codec->control_data, &val, &val); + if (ret != 0) + goto out; - ((u32 *)data)[0] &= mask; + ((u16 *)data)[0] |= val; + break; + case 4: + mask = ~params->mask; + ret = regmap_parse_val(codec->control_data, &mask, &mask); + if (ret != 0) + goto out; - ret = regmap_parse_val(codec->control_data, - &val, &val); - if (ret != 0) - goto out; + ((u32 *)data)[0] &= mask; - ((u32 *)data)[0] |= val; - break; - default: - ret = -EINVAL; + ret = regmap_parse_val(codec->control_data, &val, &val); + if (ret != 0) goto out; - } + + ((u32 *)data)[0] |= val; + break; + default: + ret = -EINVAL; + goto out; } - ret = regmap_raw_write(codec->control_data, params->base, - data, len); + ret = regmap_raw_write(codec->control_data, params->base, data, len); out: kfree(data); -- 1.7.2.5

2 2

[alsa-devel] [PATCH] ASoC: core: Fail codec probe if we fail to add controls
by Nariman Poushin 02 Apr '14

02 Apr '14

Signed-off-by: Nariman Poushin <nariman(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 35c0e71..7f9ae6c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1127,9 +1127,17 @@ static int soc_probe_codec(struct snd_soc_card *card, soc_init_codec_debugfs(codec); - if (driver->dapm_widgets) - snd_soc_dapm_new_controls(&codec->dapm, driver->dapm_widgets, - driver->num_dapm_widgets); + if (driver->dapm_widgets) { + ret = snd_soc_dapm_new_controls(&codec->dapm, + driver->dapm_widgets, + driver->num_dapm_widgets); + + if (ret != 0) { + dev_err(codec->dev, + "Failed to create new controls %d\n", ret); + goto err_probe; + } + } /* Create DAPM widgets for each DAI stream */ list_for_each_entry(dai, &codec->component.dai_list, list) { -- 1.8.5.3

2 2