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

[alsa-devel] [PATCH v4 2/4] mfd: Add Samsung Exynos Low Power Audio Subsystem driver
by Sylwester Nawrocki 10 Aug '16

10 Aug '16

This patch adds common driver for the Top block of the Samsung Exynos SoC Low Power Audio Subsystem. This is a minimal driver which prepares resources for IP blocks like I2S, audio DMA and UART and exposes a regmap for the Top block registers. Also system power ops are added to ensure the Audio Subsystem is operational after system suspend/resume cycle. Signed-off-by: Inha Song <ideal.song(a)samsung.com> Signed-off-by: Beomho Seo <beomho.seo(a)samsung.com> Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- Changes since v3: - moved from sound/soc/samsung and rewritten as a MFD driver, - PMU register definitions moved to include/linux/mfd/syscon/exynos5-pmu.h, - added regmap for LPASS Top SFR region, - cleaned up register bit field defintions. Changes since v2: - move misplaced SND_SAMSUNG_AUDSS Kconfig symbol addition to this patch. --- drivers/mfd/Kconfig | 8 ++ drivers/mfd/Makefile | 1 + drivers/mfd/exynos-lpass.c | 185 +++++++++++++++++++++++++++++++++ include/linux/mfd/syscon/exynos5-pmu.h | 4 +- 4 files changed, 197 insertions(+), 1 deletion(-) create mode 100644 drivers/mfd/exynos-lpass.c diff --git a/drivers/mfd/Kconfig b/drivers/mfd/Kconfig index 1bcf601..fc2dffb 100644 --- a/drivers/mfd/Kconfig +++ b/drivers/mfd/Kconfig @@ -270,6 +270,14 @@ config MFD_DLN2 etc. must be enabled in order to use the functionality of the device. +config MFD_EXYNOS_LPASS + tristate "Samsung Exynos SoC Low Power Audio Subsystem" + select MFD_CORE + select REGMAP_MMIO + help + Select this option to enable support for Samsung Exynos Low Power + Audio Subsystem. + config MFD_MC13XXX tristate depends on (SPI_MASTER || I2C) diff --git a/drivers/mfd/Makefile b/drivers/mfd/Makefile index 42a66e1..5c4be31 100644 --- a/drivers/mfd/Makefile +++ b/drivers/mfd/Makefile @@ -13,6 +13,7 @@ obj-$(CONFIG_MFD_BCM590XX) += bcm590xx.o obj-$(CONFIG_MFD_CROS_EC) += cros_ec.o obj-$(CONFIG_MFD_CROS_EC_I2C) += cros_ec_i2c.o obj-$(CONFIG_MFD_CROS_EC_SPI) += cros_ec_spi.o +obj-$(CONFIG_MFD_EXYNOS_LPASS) += exynos-lpass.o rtsx_pci-objs := rtsx_pcr.o rts5209.o rts5229.o rtl8411.o rts5227.o rts5249.o obj-$(CONFIG_MFD_RTSX_PCI) += rtsx_pci.o diff --git a/drivers/mfd/exynos-lpass.c b/drivers/mfd/exynos-lpass.c new file mode 100644 index 0000000..956ce1f --- /dev/null +++ b/drivers/mfd/exynos-lpass.c @@ -0,0 +1,185 @@ +/* + * Copyright (C) 2015 - 2016 Samsung Electronics Co., Ltd. + * + * Authors: Inha Song <ideal.song(a)samsung.com> + * Sylwester Nawrocki <s.nawrocki(a)samsung.com> + * + * Samsung Exynos SoC series Low Power Audio Subsystem driver. + * + * This module provides regmap for the Top SFR region and instantiates + * devices for IP blocks like DMAC, I2S, UART. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 and + * only version 2 as published by the Free Software Foundation. + */ + +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/module.h> +#include <linux/mfd/syscon.h> +#include <linux/mfd/syscon/exynos5-pmu.h> +#include <linux/of.h> +#include <linux/of_platform.h> +#include <linux/platform_device.h> +#include <linux/regmap.h> +#include <linux/types.h> + +/* LPASS Top register definitions */ +#define SFR_LPASS_CORE_SW_RESET (0x08) +#define LPASS_SB_SW_RESET (1 << 11) +#define LPASS_UART_SW_RESET (1 << 10) +#define LPASS_PCM_SW_RESET (1 << 9) +#define LPASS_I2S_SW_RESET (1 << 8) +#define LPASS_WDT1_SW_RESET (1 << 4) +#define LPASS_WDT0_SW_RESET (1 << 3) +#define LPASS_TIMER_SW_RESET (1 << 2) +#define LPASS_MEM_SW_RESET (1 << 1) +#define LPASS_DMA_SW_RESET (1 << 0) + +#define SFR_LPASS_INTR_CA5_MASK (0x48) +#define SFR_LPASS_INTR_CPU_MASK (0x58) +#define LPASS_INTR_APM (1 << 9) +#define LPASS_INTR_MIF (1 << 8) +#define LPASS_INTR_TIMER (1 << 7) +#define LPASS_INTR_DMA (1 << 6) +#define LPASS_INTR_GPIO (1 << 5) +#define LPASS_INTR_I2S (1 << 4) +#define LPASS_INTR_PCM (1 << 3) +#define LPASS_INTR_SLIMBUS (1 << 2) +#define LPASS_INTR_UART (1 << 1) +#define LPASS_INTR_SFR (1 << 0) + +struct exynos_lpass { + struct regmap *pmu; + struct regmap *top; +}; + +static void exynos_lpass_core_sw_reset(struct exynos_lpass *lpass, int mask) +{ + unsigned int val = 0; + + regmap_read(lpass->top, SFR_LPASS_CORE_SW_RESET, &val); + + val &= ~mask; + regmap_write(lpass->top, SFR_LPASS_CORE_SW_RESET, val); + + usleep_range(100, 150); + + val |= mask; + regmap_write(lpass->top, SFR_LPASS_CORE_SW_RESET, val); +} + +static void exynos_lpass_enable(struct exynos_lpass *lpass) +{ + /* Unmask SFR, DMA and I2S interrupt */ + regmap_write(lpass->top, SFR_LPASS_INTR_CA5_MASK, + LPASS_INTR_SFR | LPASS_INTR_DMA | LPASS_INTR_I2S); + + regmap_write(lpass->top, SFR_LPASS_INTR_CPU_MASK, + LPASS_INTR_SFR | LPASS_INTR_DMA | LPASS_INTR_I2S); + + /* Activate related PADs from retention state */ + regmap_write(lpass->pmu, EXYNOS5433_PAD_RETENTION_AUD_OPTION, + EXYNOS5433_PAD_INITIATE_WAKEUP_FROM_LOWPWR); + + exynos_lpass_core_sw_reset(lpass, LPASS_I2S_SW_RESET); + exynos_lpass_core_sw_reset(lpass, LPASS_DMA_SW_RESET); + exynos_lpass_core_sw_reset(lpass, LPASS_MEM_SW_RESET); +} + +static void exynos_lpass_disable(struct exynos_lpass *lpass) +{ + /* Mask any unmasked IP interrupt sources */ + regmap_write(lpass->top, SFR_LPASS_INTR_CPU_MASK, 0); + regmap_write(lpass->top, SFR_LPASS_INTR_CA5_MASK, 0); + + /* Deactivate related PADs from retention state */ + regmap_write(lpass->pmu, EXYNOS5433_PAD_RETENTION_AUD_OPTION, 0); +} + +static const struct regmap_config exynos_lpass_reg_conf = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = 0xfc, + .fast_io = true, +}; + +static int exynos_lpass_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct exynos_lpass *lpass; + void __iomem *base_top; + struct resource *res; + + if (!dev->of_node) + return -ENODEV; + + lpass = devm_kzalloc(dev, sizeof(*lpass), GFP_KERNEL); + if (!lpass) + return -ENOMEM; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + + base_top = devm_ioremap_resource(dev, res); + if (IS_ERR(base_top)) + return PTR_ERR(base_top); + + lpass->top = regmap_init_mmio(dev, base_top, + &exynos_lpass_reg_conf); + if (IS_ERR(lpass->top)) { + dev_err(dev, "LPASS top regmap initialization failed\n"); + return PTR_ERR(lpass->top); + } + + lpass->pmu = syscon_regmap_lookup_by_phandle(dev->of_node, + "samsung,pmu-syscon"); + if (IS_ERR(lpass->pmu)) { + dev_err(dev, "Failed to lookup PMU regmap\n"); + return PTR_ERR(lpass->pmu); + } + + platform_set_drvdata(pdev, lpass); + exynos_lpass_enable(lpass); + + return of_platform_populate(dev->of_node, NULL, NULL, dev); +} + +static int exynos_lpass_suspend(struct device *dev) +{ + struct exynos_lpass *lpass = dev_get_drvdata(dev); + + exynos_lpass_disable(lpass); + return 0; +} + +static int exynos_lpass_resume(struct device *dev) +{ + struct exynos_lpass *lpass = dev_get_drvdata(dev); + + exynos_lpass_enable(lpass); + return 0; +} + +static const struct of_device_id exynos_lpass_of_match[] = { + { .compatible = "samsung,exynos5433-lpass" }, + { }, +}; +MODULE_DEVICE_TABLE(of, exynos_lpass_of_match); + +static SIMPLE_DEV_PM_OPS(lpass_pm_ops, exynos_lpass_suspend, + exynos_lpass_resume); + +static struct platform_driver exynos_lpass_driver = { + .driver = { + .name = "exynos-lpass", + .pm = &lpass_pm_ops, + .of_match_table = exynos_lpass_of_match, + }, + .probe = exynos_lpass_probe, +}; +module_platform_driver(exynos_lpass_driver); + +MODULE_DESCRIPTION("Samsung Low Power Audio Subsystem driver"); +MODULE_LICENSE("GPL v2"); diff --git a/include/linux/mfd/syscon/exynos5-pmu.h b/include/linux/mfd/syscon/exynos5-pmu.h index 76f30f9..c28ff21 100644 --- a/include/linux/mfd/syscon/exynos5-pmu.h +++ b/include/linux/mfd/syscon/exynos5-pmu.h @@ -43,8 +43,10 @@ #define EXYNOS5433_MIPI_PHY2_CONTROL (0x718) #define EXYNOS5_PHY_ENABLE BIT(0) - #define EXYNOS5_MIPI_PHY_S_RESETN BIT(1) #define EXYNOS5_MIPI_PHY_M_RESETN BIT(2) +#define EXYNOS5433_PAD_RETENTION_AUD_OPTION (0x3028) +#define EXYNOS5433_PAD_INITIATE_WAKEUP_FROM_LOWPWR BIT(28) + #endif /* _LINUX_MFD_SYSCON_PMU_EXYNOS5_H_ */ -- 1.9.1

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[alsa-devel] [PATCH] ucm: Add ucm files for DB410c board.
by Riku Voipio 10 Aug '16

10 Aug '16

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> DB410c board has support for both Digital and Analog audio. Digital audio is over HDMI and analog is over codec chip integrated inside the APQ8016 SOC. It can support: - 3 Microphones: Primary Mic(Handset mic), Headset Mic and Secondary - 2 Digital Microphones. - Earpiece. - Headset. - Loud Speaker. - HDMI. [Riku: squashed Srinivas's patches together and converted spaces to tabs] Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Signed-off-by: Riku Voipio <riku.voipio(a)linaro.org> Cc: nicolas.dechesne(a)linaro.org --- configure.ac | 1 + src/conf/ucm/DB410c/DB410c.conf | 8 ++ src/conf/ucm/DB410c/HDMI | 31 +++++++ src/conf/ucm/DB410c/HiFi | 185 ++++++++++++++++++++++++++++++++++++++++ src/conf/ucm/DB410c/Makefile.am | 4 + src/conf/ucm/Makefile.am | 2 +- 6 files changed, 230 insertions(+), 1 deletion(-) create mode 100644 src/conf/ucm/DB410c/DB410c.conf create mode 100644 src/conf/ucm/DB410c/HDMI create mode 100644 src/conf/ucm/DB410c/HiFi create mode 100644 src/conf/ucm/DB410c/Makefile.am diff --git a/configure.ac b/configure.ac index f592e8b..014af5f 100644 --- a/configure.ac +++ b/configure.ac @@ -675,6 +675,7 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/conf/ucm/broadwell-rt286/Makefile \ src/conf/ucm/VEYRON-I2S/Makefile \ src/conf/ucm/chtrt5645/Makefile \ + src/conf/ucm/DB410c/Makefile \ src/conf/topology/Makefile \ src/conf/topology/broadwell/Makefile \ modules/Makefile modules/mixer/Makefile modules/mixer/simple/Makefile \ diff --git a/src/conf/ucm/DB410c/DB410c.conf b/src/conf/ucm/DB410c/DB410c.conf new file mode 100644 index 0000000..590278f --- /dev/null +++ b/src/conf/ucm/DB410c/DB410c.conf @@ -0,0 +1,8 @@ +SectionUseCase."HiFi" { + File "HiFi" + Comment "Play HiFi quality Music." +} +SectionUseCase."HDMI" { + File "HDMI" + Comment "HDMI output." +} diff --git a/src/conf/ucm/DB410c/HDMI b/src/conf/ucm/DB410c/HDMI new file mode 100644 index 0000000..8112bc0 --- /dev/null +++ b/src/conf/ucm/DB410c/HDMI @@ -0,0 +1,31 @@ +# Use case configuration for DB410c board. +# Author: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> + +SectionVerb { + EnableSequence [ + ] + + DisableSequence [ + ] + Value { + TQ "HiFi" + PlaybackPCM "plughw:0,0" + } +} + +SectionDevice."HDMI-stereo" { + #Name "HDMI-stereo" + Comment "HDMI Digital Stereo Output" + + EnableSequence [ + cdev "hw:0" + ] + + DisableSequence [ + cdev "hw:0" + ] + + Value { + PlaybackChannels "2" + } +} diff --git a/src/conf/ucm/DB410c/HiFi b/src/conf/ucm/DB410c/HiFi new file mode 100644 index 0000000..f9cbcf0 --- /dev/null +++ b/src/conf/ucm/DB410c/HiFi @@ -0,0 +1,185 @@ +# Use case configuration for DB410c board. +# Author: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> + +SectionVerb { + + EnableSequence [ + ] + + DisableSequence [ + ] + + # ALSA PCM + Value { + # ALSA PCM device for HiFi + PlaybackPCM "plughw:0,1" + CapturePCM "plughw:0,2" + } +} + + +SectionDevice."Speaker" { + Comment "Speaker playback" + + ConflictingDevice [ + "Headphones" + "Earpiece" + ] + + EnableSequence [ + cdev "hw:0" + cset "name='SPK DAC Switch' 1" + cset "name='RX3 MIX1 INP1' RX1" + ## gain to 0dB + cset "name='RX3 Digital Volume' 128" + ] + + DisableSequence [ + cdev "hw:0" + cset "name='SPK DAC Switch' 0" + cset "name='RX3 MIX1 INP1' ZERO" + ] + + Value { + PlaybackChannels "2" + } +} + + +SectionDevice."Headphones" { + Comment "Headphones playback" + + ConflictingDevice [ + "Speaker" + "Earpiece" + ] + EnableSequence [ + cdev "hw:0" + cset "name='RX1 MIX1 INP1' RX1" + cset "name='RX2 MIX1 INP1' RX2" + cset "name='RDAC2 MUX' RX2" + cset "name='HPHL' 1" + cset "name='HPHR' 1" + ## gain to 0dB + cset "name='RX1 Digital Volume' 128" + ## gain to 0dB + cset "name='RX2 Digital Volume' 128" + ] + + DisableSequence [ + cdev "hw:0" + cset "name='RX1 Digital Volume' 0" + cset "name='RX2 Digital Volume' 0" + cset "name='HPHL' 0" + cset "name='HPHR' 0" + cset "name='RDAC2 MUX' ZERO" + cset "name='RX1 MIX1 INP1' ZERO" + cset "name='RX2 MIX1 INP1' ZERO" + ] + + Value { + PlaybackChannels "2" + } +} + +SectionDevice."Earpiece" { + Comment "Earpiece playback" + + ConflictingDevice [ + "Speaker" + "Headphones" + ] + EnableSequence [ + cdev "hw:0" + ] + + DisableSequence [ + cdev "hw:0" + ] + + Value { + PlaybackChannels "2" + } +} + +SectionDevice."Handset" { + Comment "Headset Microphone" + + EnableSequence [ + cdev "hw:0" + cset "name='DEC1 MUX' ADC2" + cset "name='ADC2 Volume' 8" + cset "name='ADC2 MUX' INP2" + ] + + DisableSequence [ + cdev "hw:0" + cset "name='ADC2 MUX' ZERO" + cset "name='ADC2 Volume' 0" + cset "name='DEC1 MUX' ZERO" + ] + + Value { + CaptureChannels "2" + } +} + +SectionDevice."Primarymic" { + Comment "Primary Microphone" + + EnableSequence [ + cdev "hw:0" + cset "name='DEC1 MUX' ADC1" + cset "name='ADC1 Volume' 8" + ] + + DisableSequence [ + cdev "hw:0" + cset "name='DEC1 MUX' ZERO" + cset "name='ADC1 Volume' 0" + ] + + Value { + CaptureChannels "2" + } +} + +SectionDevice."Secondarymic" { + Comment "Secondary Microphone" + + EnableSequence [ + cdev "hw:0" + cset "name='DEC1 MUX' ADC2" + cset "name='ADC2 Volume' 8" + cset "name='ADC2 MUX' INP2" + ] + + DisableSequence [ + cdev "hw:0" + cset "name='DEC1 MUX' ZERO" + cset "name='ADC2 Volume' 0" + cset "name='ADC2 MUX' ZERO" + ] + + Value { + CaptureChannels "2" + } +} + +SectionDevice."DMIC" { + Comment "Digital Microphone" + + EnableSequence [ + cdev "hw:0" + cset "name='DEC1 MUX' DMIC1" + ] + + DisableSequence [ + cdev "hw:0" + cset "name='DEC1 MUX' ZERO" + ] + + Value { + CaptureChannels "2" + } +} diff --git a/src/conf/ucm/DB410c/Makefile.am b/src/conf/ucm/DB410c/Makefile.am new file mode 100644 index 0000000..e10a136 --- /dev/null +++ b/src/conf/ucm/DB410c/Makefile.am @@ -0,0 +1,4 @@ +alsaconfigdir = @ALSA_CONFIG_DIR@ +ucmdir = $(alsaconfigdir)/ucm/DB410c +ucm_DATA = DB410c.conf HDMI HiFi +EXTRA_DIST = $(ucm_DATA) diff --git a/src/conf/ucm/Makefile.am b/src/conf/ucm/Makefile.am index 8865770..7bf432e 100644 --- a/src/conf/ucm/Makefile.am +++ b/src/conf/ucm/Makefile.am @@ -1 +1 @@ -SUBDIRS=DAISY-I2S PandaBoard PandaBoardES SDP4430 tegraalc5632 PAZ00 GoogleNyan broadwell-rt286 VEYRON-I2S chtrt5645 +SUBDIRS=DAISY-I2S PandaBoard PandaBoardES SDP4430 tegraalc5632 PAZ00 GoogleNyan broadwell-rt286 VEYRON-I2S chtrt5645 DB410c -- 2.1.4

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[alsa-devel] [PATCH] sound: soc: atmel: Don't unconditionally reset SSC on stream startup
by Peter Meerwald 10 Aug '16

10 Aug '16

From: Christoph Huber <c.huber(a)bct-electronic.com> commit cbaadf0f90d6 (ASoC: atmel_ssc_dai: refactor the startup and shutdown) refactored code such that the SSC is reset on every startup; this breaks duplex audio (e.g. first start audio playback, then start record, causing the playback to stop/hang) Signed-off-by: Christoph Huber <c.huber(a)bct-electronic.com> Signed-off-by: Peter Meerwald-Stadler <p.meerwald(a)bct-electronic.com> Cc: Songjun Wu <songjun.wu(a)microchip.com> Cc: Nicolas Ferre <nicolas.ferre(a)atmel.com> --- sound/soc/atmel/atmel_ssc_dai.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 841d059..b3e9f0c 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -298,8 +298,9 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, clk_enable(ssc_p->ssc->clk); ssc_p->mck_rate = clk_get_rate(ssc_p->ssc->clk); - /* Reset the SSC to keep it at a clean status */ - ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); + /* Reset the SSC if it is not initialized to keep it at a clean status */ + if(!ssc_p->initialized) + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { dir = 0; -- 2.1.4

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[alsa-devel] [PATCH v5 1/2] mfd: Add DT bindings documentation for Samsung Exynos LPASS
by Sylwester Nawrocki 10 Aug '16

10 Aug '16

This patch adds documentation of the DT bindings for the Samsung Exynos SoC Low Power Audio Subsystem. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Acked-by: Rob Herring <robh(a)kernel.org> --- Changes since v4: - text reformatting, added VIC to the list of sub-devices, relative paths used for the reference DT binding documentation files, added example dts. Changes since v3: - moved to Documentation/devicetree/bindings/mfd, - added #address-cells, #size-cells, ranges as required properties, - added description of the IP block subnodes and links to their binding's documentation. Changes since v2: - none. --- .../bindings/mfd/samsung,exynos5433-lpass.txt | 70 ++++++++++++++++++++++ 1 file changed, 70 insertions(+) create mode 100644 Documentation/devicetree/bindings/mfd/samsung,exynos5433-lpass.txt diff --git a/Documentation/devicetree/bindings/mfd/samsung,exynos5433-lpass.txt b/Documentation/devicetree/bindings/mfd/samsung,exynos5433-lpass.txt new file mode 100644 index 0000000..c110e11 --- /dev/null +++ b/Documentation/devicetree/bindings/mfd/samsung,exynos5433-lpass.txt @@ -0,0 +1,70 @@ +Samsung Exynos SoC Low Power Audio Subsystem (LPASS) + +Required properties: + + - compatible : "samsung,exynos5433-lpass" + - reg : should contain the LPASS top SFR region location + and size + - samsung,pmu-syscon : the phandle to the Power Management Unit node + - #address-cells : should be 1 + - #size-cells : should be 1 + - ranges : must be present + +Each IP block of the Low Power Audio Subsystem should be specified as +an optional sub-node. For "samsung,exynos5433-lpass" compatible this includes: +UART, SLIMBUS, PCM, I2S, DMAC, Timers 0...4, VIC, WDT 0...1 devices. + +Bindings of the sub-nodes are described in: + ../serial/samsung_uart.txt + ../sound/samsung-i2s.txt + ../dma/arm-pl330.txt + + +Example: + +audio-subsystem { + compatible = "samsung,exynos5433-lpass"; + reg = <0x11400000 0x100>, <0x11500000 0x08>; + samsung,pmu-syscon = <&pmu_system_controller>; + #address-cells = <1>; + #size-cells = <1>; + ranges; + + adma: adma@11420000 { + compatible = "arm,pl330", "arm,primecell"; + reg = <0x11420000 0x1000>; + interrupts = <0 73 0>; + clocks = <&cmu_aud CLK_ACLK_DMAC>; + clock-names = "apb_pclk"; + #dma-cells = <1>; + #dma-channels = <8>; + #dma-requests = <32>; + }; + + i2s0: i2s0@11440000 { + compatible = "samsung,exynos7-i2s"; + reg = <0x11440000 0x100>; + dmas = <&adma 0 &adma 2>; + dma-names = "tx", "rx"; + interrupts = <0 70 0>; + clocks = <&cmu_aud CLK_PCLK_AUD_I2S>, + <&cmu_aud CLK_SCLK_AUD_I2S>, + <&cmu_aud CLK_SCLK_I2S_BCLK>; + clock-names = "iis", "i2s_opclk0", "i2s_opclk1"; + pinctrl-names = "default"; + pinctrl-0 = <&i2s0_bus>; + status = "disabled"; + }; + + serial_3: serial@11460000 { + compatible = "samsung,exynos5433-uart"; + reg = <0x11460000 0x100>; + interrupts = <0 67 0>; + clocks = <&cmu_aud CLK_PCLK_AUD_UART>, + <&cmu_aud CLK_SCLK_AUD_UART>; + clock-names = "uart", "clk_uart_baud0"; + pinctrl-names = "default"; + pinctrl-0 = <&uart_aud_bus>; + status = "disabled"; + }; + }; -- 1.9.1

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[alsa-devel] Underrun with 2 periods on skylake HDA driver
by Dharageswari R 10 Aug '16

10 Aug '16

Hello, While doing the validation of Skylake HDA driver using aplay and following parameters, we noticed underrun issues. aplay -Dhw:0,0,0 -vv s_48K_16Bit_Sunshine.wav -F 100000 -B 200000 -d 20 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 9600 period_size : 4800 period_time : 100000 tstamp_mode : NONE period_step : 1 avail_min : 4800 period_event : 0 start_threshold : 9600 stop_threshold : 9600 silence_threshold: 0 silence_size : 0 boundary : 5404319552844595200 appl_ptr : 0 hw_ptr : 0 After doing further debugging, what I found is 1.DMA complete interrupt occurs for the first period. 2.Driver reads DMA position in buffer pointer and gets value of 4784. 3.Driver calls snd_pcm_period_elapsed() function to report period elapsed interrupt to application with DMA position = 4784. 4.snd_pcm_lib_write1() is waiting for “avail_min” free space to be available. In this case the avail_min = 4800 and at this interrupt the avail = 4874 and so it does not write any data. 5.DMA complete interrupt occurs for the second period. 6.Driver reads DMA position in the buffer pointer and gets value of 9600. 7.Driver calls snd_pcm_period_elapsed() function to report period elapsed interrupt. 8.Now the snd_pcm_lib_write1() gets avail > avail_min but it’s too late because DMA has already started playing data from first period, which is not yet written by the application. 9.This causes underrun. Logs: 857.870974: hwptr: pcmC0D0p/sub0: IRQ: pos=4800, old=19200, base=19200, period=4800, buf=9600 857.871237: hwptr: pcmC0D0p/sub0: POS: pos=4800, old=24000, base=19200, period=4800, buf=9600 857.970972: hwptr: pcmC0D0p/sub0: IRQ: pos=0, old=24000, base=19200, period=4800, buf=9600 857.971101: hwptr: pcmC0D0p/sub0: POS: pos=0, old=28800, base=28800, period=4800, buf=9600 858.070971: hwptr: pcmC0D0p/sub0: IRQ: pos=4784, old=28800, base=28800, period=4800, buf=9600 858.170968: hwptr: pcmC0D0p/sub0: IRQ: pos=0, old=33584, base=28800, period=4800, buf=9600 858.170988: xrun: pcmC0D0p/sub0: XRUN: old=38400, base=38400, period=4800, buf=9600 858.172240: hwptr: pcmC0D0p/sub0: POS: pos=96, old=0, base=0, period=4800, buf=9600 858.269867: hwptr: pcmC0D0p/sub0: IRQ: pos=4798, old=96, base=0, period=4800, buf=9600 858.369902: hwptr: pcmC0D0p/sub0: IRQ: pos=0, old=4798, base=0, period=4800, buf=9600 I checked with HDA architects and they are saying that when the DMA completion interrupt occurs there can be slight delay before the DMA Position In Buffer Pointer gets updated. So driver is expected to refill data based on whatever content is reflected in the DMA Position In Buffer. What is the right way to fix this issue ? Thanks, Dhara.R --

2 1

[alsa-devel] [PATCH v4 0/5] ALSA - hda: Add support for link audio time reporting
by Vinod Koul 10 Aug '16

10 Aug '16

Skylake onwards we support GTS capability and thus can report audio link time. This series adds support for parsing the capability and then reports link time. The parsing of caps is made common and moved up from extended hda to hdac. The SKL driver and hdac extended core has been updated for this. Mark, The update of the driver needs to update Skl driver, so if you can ACK the 2nd patch, it can go along with extended audio change thru ALSA tree. Changes in v4: - fix code style - use hdac_readl/writel - add cpufeatures header Guneshwor Singh (2): ALSA - hda: Add support for parsing new HDA capabilities ALSA - hda: Add support for link audio time reporting Vinod Koul (3): ALSA: hda - move bus_parse_capabilities to core ALSA: convert users to core bus_parse_capabilities ALSA - Ext hda: remove bus_parse_capabilities include/sound/hda_register.h | 36 ++++++ include/sound/hdaudio.h | 13 +++ include/sound/hdaudio_ext.h | 12 -- sound/hda/ext/hdac_ext_controller.c | 91 ++------------- sound/hda/ext/hdac_ext_stream.c | 46 ++++---- sound/hda/hdac_controller.c | 75 ++++++++++++ sound/pci/hda/hda_controller.c | 203 ++++++++++++++++++++++++++++++++- sound/pci/hda/hda_controller.h | 3 + sound/pci/hda/hda_intel.c | 17 +++ sound/soc/intel/skylake/skl-messages.c | 4 +- sound/soc/intel/skylake/skl-pcm.c | 8 +- sound/soc/intel/skylake/skl.c | 6 +- 12 files changed, 386 insertions(+), 128 deletions(-) -- 1.9.1

3 14

[alsa-devel] [PATCH] ASoC: core: fix shift used for second item in snd_soc_get_enum_double
by jjassal 09 Aug '16

09 Aug '16

Incorrect shift value was being used to extract the second item. Signed-off-by: jjassal <jjassal(a)opensource.wolfsonmicro.com> --- sound/soc/soc-ops.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index a513a34..9fc1a7b 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -77,7 +77,7 @@ int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, 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_r) & e->mask; item = snd_soc_enum_val_to_item(e, val); ucontrol->value.enumerated.item[1] = item; } -- 2.1.4

2 1

[alsa-devel] [PATCH 00/39] seq: obsolete change of address limit
by Takashi Sakamoto 09 Aug '16

09 Aug '16

Hi, ALSA sequencer core has two types of client; user application and kernel driver. The core allows both types of client to do relevant operations, thus it's required for the core to handle data in both user and kernel spaces. Currently, this is achieved by changing address limit of running task. This is a well-known technique to suppress address check, while it's just a suppression and unfriendly to readers or static code parsers. This patchset obsoletes the usage. In process context, data in user space is once copied to kernel space, then operated and copied to user space. As a result, actual operations for each ioctl command handle data in kernel space, and '__user' qualifier is useless. In this series, patch 1-8 just apply above design. The rest changes each operation following to the design. Therefore, static code parser like sparce generates warnings temporarily in a way to apply these patches. There's a concern of this solution. The data for ioctl is always copied to kernel space or to kernel space even when it's read-only or write-only. But this brings no severe issue as long as I read ALSA sequencer core. (This patchset is a part of my work to introduce EPIPE into ALSA rawmidi core.) Takashi Sakamoto (39): ALSA: seq: add const qualifier to table of functions for ioctl ALSA: seq: apply shorter name for file local functions ALSA: seq: fulfill callback entry for ioctl ALSA: seq: copy ioctl data from user space to kernel stack ALSA: seq: add documentation for snd_seq_kernel_client_ctl ALSA: seq: obsolete address mode in compatibility layer ALSA: seq: obsolete change of address limit in in-kernel path for ioctl ALSA: seq: obsolete address limit helper ALSA: seq: optimize pversion function to new design ALSA: seq: optimize client_id function to new design ALSA: seq: optimize system_info function to new design ALSA: seq: optimize running mode function to new design ALSA: seq: optimize client_info function to new design ALSA: seq: optimize set_client_info function to new design ALSA: seq: optimize create_port function to new design ALSA: seq: optimize delete_port function to new design ALSA: seq: optimize get_port_info function to new design ALSA: seq: optimize seq_port_info function to new design ALSA: seq: optimize subscribe_port function to new design ALSA: seq: optimize unsubscribe_port function to new design ALSA: seq: optimize create_queue function to new design ALSA: seq: optimize delete_queue function to new design ALSA: seq: optimize get_queue_info function to new design ALSA: seq: optimize seq_queue_info function to new design ALSA: seq: optimize get_named_queue function to new design ALSA: seq: optimize get_queue_status function to new design ALSA: seq: optimize get_queue_tempo function to new design ALSA: seq: optimize set_queue_tempo function to new design ALSA: seq: optimize get_queue_timer function to new design ALSA: seq: optimize seq_queue_timer function to new design ALSA: seq: optimize get_queue_client function to new design ALSA: seq: optimize set_queue_client function to new design ALSA: seq: optimize get_client_pool function to new design ALSA: seq: optimize seq_client_pool function to new design ALSA: seq: optimize remove_events function to new design ALSA: seq: optimize get_subscription function to new design ALSA: seq: optimize query_subs function to new design ALSA: seq: optimize query_next_client function to new design ALSA: seq: optimize query_next_port function to new design sound/core/seq/seq_clientmgr.c | 854 +++++++++++++++++++---------------------- sound/core/seq/seq_compat.c | 26 +- 2 files changed, 407 insertions(+), 473 deletions(-) -- 2.7.4

3 62

[alsa-devel] [PATCH v3 0/5] ALSA - hda: Add support for link audio time reporting
by Vinod Koul 09 Aug '16

09 Aug '16

Skylake onwards we support GTS capability and thus can report audio link time. This series adds support for parsing the capability and then reports link time. The parsing of caps is made common and moved up from extended hda to hdac. The SKL driver and hdac extended core has been updated for this. Mark, The update of the driver needs to update Skl driver, so if you can ACK the 2nd patch, it can go along with extended audio change thru ALSA tree. Changes in v3: moved extended hda to hdac update SKL drv to use new hdac caps update patchset for HDA drv to use new caps parsing Guneshwor Singh (2): ALSA - hda: Add support for parsing new HDA capabilities ALSA - hda: Add support for link audio time reporting Vinod Koul (3): ALSA: hda - move bus_parse_capabilities to core ALSA: convert users to core bus_parse_capabilities ALSA - Ext hda: remove bus_parse_capabilities include/sound/hda_register.h | 36 ++++++ include/sound/hdaudio.h | 13 +++ include/sound/hdaudio_ext.h | 12 -- sound/hda/ext/hdac_ext_controller.c | 91 ++------------- sound/hda/ext/hdac_ext_stream.c | 46 ++++---- sound/hda/hdac_controller.c | 76 +++++++++++++ sound/pci/hda/hda_controller.c | 200 ++++++++++++++++++++++++++++++++- sound/pci/hda/hda_controller.h | 3 + sound/pci/hda/hda_intel.c | 16 +++ sound/soc/intel/skylake/skl-messages.c | 4 +- sound/soc/intel/skylake/skl-pcm.c | 8 +- sound/soc/intel/skylake/skl.c | 6 +- 12 files changed, 383 insertions(+), 128 deletions(-) -- 1.9.1

3 15

[alsa-devel] [PATCH 0/3] ASoC: remove snd_soc_pcm_set/get_drvdata()
by Kuninori Morimoto 09 Aug '16

09 Aug '16

Hi Mark snd_soc_pcm_set_drvdata() will set driver data to rtd->dev, but driver data of rtd->dev is already used as "rtd" on soc_post_component_init(). static int soc_post_component_init(xxx) { ... dev_set_drvdata(rtd->dev, rtd); ... } But, ASoC has snd_soc_pcm_set/get_drvdata() to set rtd->dev. It will overwrite existing data and will break system. This patch tries to solve this issue. Intel haswell and qcom is using these functions. ** Note ** I don't have these boards, thus, I did compile test only. Kuninori Morimoto (3): ASoC: sst-haswell-pcm: don't use snd_soc_pcm_set_drvdata() ASoC: lpass-platform: don't use snd_soc_pcm_set_drvdata() ASoC: remove snd_soc_pcm_set/get_drvdata() include/sound/soc.h | 11 ----------- sound/soc/intel/haswell/sst-haswell-pcm.c | 1 - sound/soc/qcom/lpass-platform.c | 14 +++++++------- sound/soc/qcom/lpass.h | 2 ++ 4 files changed, 9 insertions(+), 19 deletions(-)

2 7

[alsa-devel] [PATCH 0/6] ASoC: simple-card: add init_dai / canonicalize_dailink
by Kuninori Morimoto 09 Aug '16

09 Aug '16

Hi Mark These add init_dai / canonicalize_dailink to simple utiles. These are part of simple-card cleanup Kuninori Morimoto (6): ASoC: simple-card-utils: add asoc_simple_card_init_dai() ASoC: simple-card: use asoc_simple_card_init_dai() ASoC: rsrc-card: use asoc_simple_card_init_dai() ASoC: simple-card-utils: add asoc_simple_card_canonicalize_dailink() ASoC: simple-card: use asoc_simple_card_canonicalize_dailink() ASoC: rsrc-card: use asoc_simple_card_canonicalize_dailink() include/sound/simple_card_utils.h | 5 +++++ sound/soc/generic/simple-card-utils.c | 42 ++++++++++++++++++++++++++++++++++++++++++ sound/soc/generic/simple-card.c | 47 +++++------------------------------------------ sound/soc/sh/rcar/rsrc-card.c | 32 +++++--------------------------- 4 files changed, 57 insertions(+), 69 deletions(-) -- 1.9.1

2 10

[alsa-devel] [PATCH 0/2] alsabat: test scripts and reference amixer config files
by focus.luo＠linux.intel.com 09 Aug '16

09 Aug '16

From: Focus Luo <focus.luo(a)linux.intel.com> The patch set supplies some automated test scripts and reference amixer config files for linux audio driver testing by using alsa-bat as test tool on Intel platforms. The first patch includes automated test scripts for linux audio driver based on alsa-lib interface by using alsabat as test tool. It supports analog audio and display audio(HDMI/DP) interface. The second patch supplies some amixer reference config file on Intel platforms. Focus Luo (2): alsabat: automation test scripts alsabat: add amixer config files bat/tests/README | 32 ++ bat/tests/alsabat_main.sh | 178 ++++++++++ bat/tests/analog_audio_playback_and_capture.sh | 84 +++++ bat/tests/asound_state/asound.state.Broadwell | 439 +++++++++++++++++++++++++ bat/tests/asound_state/asound.state.Haswell | 342 +++++++++++++++++++ bat/tests/asound_state/asound.state.Skylake | 437 ++++++++++++++++++++++++ bat/tests/dp_audio_playback.sh | 83 +++++ bat/tests/dp_audio_subdevice_number.sh | 68 ++++ bat/tests/hdmi_audio_playback.sh | 81 +++++ bat/tests/hdmi_audio_subdevice_number.sh | 65 ++++ bat/tests/map_test_case | 8 + 11 files changed, 1817 insertions(+) create mode 100644 bat/tests/README create mode 100755 bat/tests/alsabat_main.sh create mode 100755 bat/tests/analog_audio_playback_and_capture.sh create mode 100644 bat/tests/asound_state/asound.state.Broadwell create mode 100644 bat/tests/asound_state/asound.state.Haswell create mode 100644 bat/tests/asound_state/asound.state.Skylake create mode 100755 bat/tests/dp_audio_playback.sh create mode 100755 bat/tests/dp_audio_subdevice_number.sh create mode 100755 bat/tests/hdmi_audio_playback.sh create mode 100755 bat/tests/hdmi_audio_subdevice_number.sh create mode 100644 bat/tests/map_test_case -- 1.9.1

5 8

[alsa-devel] [PATCH] ALSA: usb-audio: Add quirk for ELP HD USB Camera
by Vittorio Gambaletta (VittGam) 09 Aug '16

09 Aug '16

The ELP HD USB Camera (05a3:9420) needs this quirk for suppressing the unsupported sample rate inquiry. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=98481 Cc: <stable(a)vger.kernel.org> Cc: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Vittorio Gambaletta <linuxbugs(a)vittgam.net> --- --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1138,6 +1138,7 @@ 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(0x05A3, 0x9420): /* ELP HD USB Camera */ 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 1

[alsa-devel] [PATCH] ASoC: simple-card: fix module load "unknown symbol" issue
by Kevin Hilman 09 Aug '16

09 Aug '16

An ASoC driver using simple-card, when built as a module gets the following error at module load time: [ 23.571873] simple_card_utils: Unknown symbol snd_soc_of_parse_daifmt (err 0) [ 23.712726] simple_card_utils: Unknown symbol snd_soc_of_parse_card_name (err 0) fix this by including simple-card-utils.o symbols in the module build its symols are resolved correctly. Tested on Davinci DA850-EVM using snd_soc_edma, snc_soc_davinci_mcasp, with all of ASoC built as modules. Cc: Sekhar Nori <nsekhar(a)ti.com> Signed-off-by: Kevin Hilman <khilman(a)baylibre.com> --- sound/soc/generic/Makefile | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/generic/Makefile b/sound/soc/generic/Makefile index 45602ca8536e..ff0c55eee4f7 100644 --- a/sound/soc/generic/Makefile +++ b/sound/soc/generic/Makefile @@ -2,4 +2,8 @@ obj-$(CONFIG_SND_SIMPLE_CARD_UTILS) := simple-card-utils.o snd-soc-simple-card-objs := simple-card.o +ifneq ($(CONFIG_SND_SIMPLE_CARD_UTILS),) +snd-soc-simple-card-objs += simple-card-utils.o +endif + obj-$(CONFIG_SND_SIMPLE_CARD) += snd-soc-simple-card.o -- 2.9.0

4 9

[alsa-devel] [PATCH] ASoC: nau8810: Add driver for Nuvoton codec chip NAU88C10
by John Hsu 09 Aug '16

09 Aug '16

The driver is for codec NAU88C10 of Nuvoton Technology Corporation. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- .../devicetree/bindings/sound/nau8810.txt | 16 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/nau8810.c | 833 +++++++++++++++++++++ sound/soc/codecs/nau8810.h | 272 +++++++ 5 files changed, 1128 insertions(+) create mode 100755 Documentation/devicetree/bindings/sound/nau8810.txt create mode 100755 sound/soc/codecs/nau8810.c create mode 100755 sound/soc/codecs/nau8810.h diff --git a/Documentation/devicetree/bindings/sound/nau8810.txt b/Documentation/devicetree/bindings/sound/nau8810.txt new file mode 100755 index 0000000..05830e4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nau8810.txt @@ -0,0 +1,16 @@ +NAU8810 audio CODEC + +This device supports I2C only. + +Required properties: + + - compatible : "nuvoton,nau8810" + + - reg : the I2C address of the device. + +Example: + +codec: nau8810@1a { + compatible = "nuvoton,nau8810"; + reg = <0x1a>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index cb57f8f..bbd7324 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -91,6 +91,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX9877 if I2C select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C + select SND_SOC_NAU8810 if I2C select SND_SOC_NAU8825 if I2C select SND_SOC_HDMI_CODEC select SND_SOC_PCM1681 if I2C @@ -1046,6 +1047,10 @@ config SND_SOC_MC13783 config SND_SOC_ML26124 tristate +config SND_SOC_NAU8810 + tristate "Nuvoton Technology Corporation NAU88C10 CODEC" + depends on I2C + config SND_SOC_NAU8825 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 550a174..3c15bc5 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -84,6 +84,7 @@ 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-nau8810-objs := nau8810.o snd-soc-nau8825-objs := nau8825.o snd-soc-hdmi-codec-objs := hdmi-codec.o snd-soc-pcm1681-objs := pcm1681.o @@ -301,6 +302,7 @@ 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-$(CONFIG_SND_SOC_NAU8810) += snd-soc-nau8810.o obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o diff --git a/sound/soc/codecs/nau8810.c b/sound/soc/codecs/nau8810.c new file mode 100755 index 0000000..d03e9c76 --- /dev/null +++ b/sound/soc/codecs/nau8810.c @@ -0,0 +1,833 @@ +/* + * nau8810.c -- NAU8810 ALSA Soc Audio driver + * + * Copyright 2016 Nuvoton Technology Corp. + * + * Author: David Lin <ctlin0(a)nuvoton.com> + * + * Based on WM8974.c + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "nau8810.h" + + +static const int nau8810_mclk_scaler[] = { 10, 15, 20, 30, 40, 60, 80, 120 }; + +static const struct reg_default nau8810_reg_defaults[] = { + { NAU8810_REG_POWER1, 0x0000 }, + { NAU8810_REG_POWER2, 0x0000 }, + { NAU8810_REG_POWER3, 0x0000 }, + { NAU8810_REG_IFACE, 0x0050 }, + { NAU8810_REG_COMP, 0x0000 }, + { NAU8810_REG_CLOCK, 0x0140 }, + { NAU8810_REG_SMPLR, 0x0000 }, + { NAU8810_REG_DAC, 0x0000 }, + { NAU8810_REG_DACGAIN, 0x00FF }, + { NAU8810_REG_ADC, 0x0100 }, + { NAU8810_REG_ADCGAIN, 0x00FF }, + { NAU8810_REG_EQ1, 0x012C }, + { NAU8810_REG_EQ2, 0x002C }, + { NAU8810_REG_EQ3, 0x002C }, + { NAU8810_REG_EQ4, 0x002C }, + { NAU8810_REG_EQ5, 0x002C }, + { NAU8810_REG_DACLIM1, 0x0032 }, + { NAU8810_REG_DACLIM2, 0x0000 }, + { NAU8810_REG_NOTCH1, 0x0000 }, + { NAU8810_REG_NOTCH2, 0x0000 }, + { NAU8810_REG_NOTCH3, 0x0000 }, + { NAU8810_REG_NOTCH4, 0x0000 }, + { NAU8810_REG_ALC1, 0x0038 }, + { NAU8810_REG_ALC2, 0x000B }, + { NAU8810_REG_ALC3, 0x0032 }, + { NAU8810_REG_NOISEGATE, 0x0000 }, + { NAU8810_REG_PLLN, 0x0008 }, + { NAU8810_REG_PLLK1, 0x000C }, + { NAU8810_REG_PLLK2, 0x0093 }, + { NAU8810_REG_PLLK3, 0x00E9 }, + { NAU8810_REG_ATTEN, 0x0000 }, + { NAU8810_REG_INPUT_SIGNAL, 0x0003 }, + { NAU8810_REG_PGAGAIN, 0x0010 }, + { NAU8810_REG_ADCBOOST, 0x0100 }, + { NAU8810_REG_OUTPUT, 0x0002 }, + { NAU8810_REG_SPKMIX, 0x0001 }, + { NAU8810_REG_SPKGAIN, 0x0039 }, + { NAU8810_REG_MONOMIX, 0x0001 }, + { NAU8810_REG_POWER4, 0x0000 }, + { NAU8810_REG_TSLOTCTL1, 0x0000 }, + { NAU8810_REG_TSLOTCTL2, 0x0020 }, + { NAU8810_REG_DEVICE_REVID, 0x00EF }, + { NAU8810_REG_I2C_DEVICEID, 0x001A }, + { NAU8810_REG_ADDITIONID, 0x00CA }, + { NAU8810_REG_RESERVE, 0x0124 }, + { NAU8810_REG_OUTCTL, 0x0001 }, + { NAU8810_REG_ALC1ENHAN1, 0x0000 }, + { NAU8810_REG_ALC1ENHAN2, 0x0039 }, + { NAU8810_REG_MISCCTL, 0x0000 }, + { NAU8810_REG_OUTTIEOFF, 0x0000 }, + { NAU8810_REG_AGCP2POUT, 0x0000 }, + { NAU8810_REG_AGCPOUT, 0x0000 }, + { NAU8810_REG_AMTCTL, 0x0000 }, + { NAU8810_REG_OUTTIEOFFMAN, 0x0000 }, +}; + +static bool nau8810_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8810_REG_RESET ... NAU8810_REG_SMPLR: + case NAU8810_REG_DAC ... NAU8810_REG_DACGAIN: + case NAU8810_REG_ADC ... NAU8810_REG_ADCGAIN: + case NAU8810_REG_EQ1 ... NAU8810_REG_EQ5: + case NAU8810_REG_DACLIM1 ... NAU8810_REG_DACLIM2: + case NAU8810_REG_NOTCH1 ... NAU8810_REG_NOTCH4: + case NAU8810_REG_ALC1 ... NAU8810_REG_ATTEN: + case NAU8810_REG_INPUT_SIGNAL ... NAU8810_REG_PGAGAIN: + case NAU8810_REG_ADCBOOST: + case NAU8810_REG_OUTPUT ... NAU8810_REG_SPKMIX: + case NAU8810_REG_SPKGAIN: + case NAU8810_REG_MONOMIX: + case NAU8810_REG_POWER4 ... NAU8810_REG_TSLOTCTL2: + case NAU8810_REG_DEVICE_REVID ... NAU8810_REG_RESERVE: + case NAU8810_REG_ALC1ENHAN1 ... NAU8810_REG_ALC1ENHAN2: + case NAU8810_REG_MISCCTL: + case NAU8810_REG_OUTTIEOFF ... NAU8810_REG_OUTTIEOFFMAN: + return true; + default: + return false; + } +} + +static bool nau8810_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8810_REG_RESET ... NAU8810_REG_SMPLR: + case NAU8810_REG_DAC ... NAU8810_REG_DACGAIN: + case NAU8810_REG_ADC ... NAU8810_REG_ADCGAIN: + case NAU8810_REG_EQ1 ... NAU8810_REG_EQ5: + case NAU8810_REG_DACLIM1 ... NAU8810_REG_DACLIM2: + case NAU8810_REG_NOTCH1 ... NAU8810_REG_NOTCH4: + case NAU8810_REG_ALC1 ... NAU8810_REG_ATTEN: + case NAU8810_REG_INPUT_SIGNAL ... NAU8810_REG_PGAGAIN: + case NAU8810_REG_ADCBOOST: + case NAU8810_REG_OUTPUT ... NAU8810_REG_SPKMIX: + case NAU8810_REG_SPKGAIN: + case NAU8810_REG_MONOMIX: + case NAU8810_REG_POWER4 ... NAU8810_REG_TSLOTCTL2: + case NAU8810_REG_ALC1ENHAN1 ... NAU8810_REG_ALC1ENHAN2: + case NAU8810_REG_MISCCTL: + case NAU8810_REG_OUTTIEOFF ... NAU8810_REG_OUTTIEOFFMAN: + return true; + default: + return false; + } +} + +static bool nau8810_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8810_REG_RESET: + case NAU8810_REG_EQ1 ... NAU8810_REG_EQ5: + case NAU8810_REG_NOTCH1 ... NAU8810_REG_NOTCH4: + return true; + default: + return false; + } +} + +static int nau8810_reg_write(void *context, unsigned int reg, + unsigned int value) +{ + struct i2c_client *client = context; + uint8_t buf[2]; + __be16 *out = (void *)buf; + int ret; + + *out = cpu_to_be16((reg << 9) | value); + ret = i2c_master_send(client, buf, sizeof(buf)); + if (ret == sizeof(buf)) + return 0; + else if (ret < 0) + return ret; + else + return -EIO; +} + +static int nau8810_reg_read(void *context, unsigned int reg, + unsigned int *value) +{ + struct i2c_client *client = context; + struct i2c_msg xfer[2]; + uint8_t reg_buf; + uint16_t val_buf; + int ret; + + reg_buf = (uint8_t)(reg << 1); + xfer[0].addr = client->addr; + xfer[0].len = sizeof(reg_buf); + xfer[0].buf = &reg_buf; + xfer[0].flags = 0; + + xfer[1].addr = client->addr; + xfer[1].len = sizeof(val_buf); + xfer[1].buf = (uint8_t *)&val_buf; + xfer[1].flags = I2C_M_RD; + + ret = i2c_transfer(client->adapter, xfer, ARRAY_SIZE(xfer)); + if (ret < 0) + return ret; + else if (ret != ARRAY_SIZE(xfer)) + return -EIO; + + *value = be16_to_cpu(val_buf); + + return 0; +} + +static const char * const nau8810_companding[] = { + "Off", "NC", "u-law", "A-law" +}; +static const char * const nau8810_deemp[] = { + "None", "32kHz", "44.1kHz", "48kHz" +}; +static const char * const nau8810_eqmode[] = {"Capture", "Playback" }; +static const char * const nau8810_bw[] = {"Narrow", "Wide" }; +static const char * const nau8810_eq1[] = { + "80Hz", "105Hz", "135Hz", "175Hz" +}; +static const char * const nau8810_eq2[] = { + "230Hz", "300Hz", "385Hz", "500Hz" +}; +static const char * const nau8810_eq3[] = { + "650Hz", "850Hz", "1.1kHz", "1.4kHz" +}; +static const char * const nau8810_eq4[] = { + "1.8kHz", "2.4kHz", "3.2kHz", "4.1kHz" +}; +static const char * const nau8810_eq5[] = { + "5.3kHz", "6.9kHz", "9kHz", "11.7kHz" +}; +static const char * const nau8810_alc[] = {"Normal", "Limiter" }; + +static const struct soc_enum nau8810_enum[] = { + SOC_ENUM_SINGLE(NAU8810_REG_COMP, NAU8810_ADCCM_SFT, + ARRAY_SIZE(nau8810_companding), nau8810_companding), + SOC_ENUM_SINGLE(NAU8810_REG_COMP, NAU8810_DACCM_SFT, + ARRAY_SIZE(nau8810_companding), nau8810_companding), + SOC_ENUM_SINGLE(NAU8810_REG_DAC, NAU8810_DEEMP_SFT, + ARRAY_SIZE(nau8810_deemp), nau8810_deemp), + SOC_ENUM_SINGLE(NAU8810_REG_EQ1, NAU8810_EQM_SFT, + ARRAY_SIZE(nau8810_eqmode), nau8810_eqmode), + + SOC_ENUM_SINGLE(NAU8810_REG_EQ1, NAU8810_EQ1CF_SFT, + ARRAY_SIZE(nau8810_eq1), nau8810_eq1), + SOC_ENUM_SINGLE(NAU8810_REG_EQ2, NAU8810_EQ2BW_SFT, + ARRAY_SIZE(nau8810_bw), nau8810_bw), + SOC_ENUM_SINGLE(NAU8810_REG_EQ2, NAU8810_EQ2CF_SFT, + ARRAY_SIZE(nau8810_eq2), nau8810_eq2), + SOC_ENUM_SINGLE(NAU8810_REG_EQ3, NAU8810_EQ3BW_SFT, + ARRAY_SIZE(nau8810_bw), nau8810_bw), + + SOC_ENUM_SINGLE(NAU8810_REG_EQ3, NAU8810_EQ3CF_SFT, + ARRAY_SIZE(nau8810_eq3), nau8810_eq3), + SOC_ENUM_SINGLE(NAU8810_REG_EQ4, NAU8810_EQ4BW_SFT, + ARRAY_SIZE(nau8810_bw), nau8810_bw), + SOC_ENUM_SINGLE(NAU8810_REG_EQ4, NAU8810_EQ4CF_SFT, + ARRAY_SIZE(nau8810_eq4), nau8810_eq4), + SOC_ENUM_SINGLE(NAU8810_REG_EQ5, NAU8810_EQ5CF_SFT, + ARRAY_SIZE(nau8810_eq5), nau8810_eq5), + + SOC_ENUM_SINGLE(NAU8810_REG_ALC3, NAU8810_ALCM_SFT, + ARRAY_SIZE(nau8810_alc), nau8810_alc), +}; + +static const DECLARE_TLV_DB_SCALE(digital_tlv, -12750, 50, 1); +static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); +static const DECLARE_TLV_DB_SCALE(inpga_tlv, -1200, 75, 0); +static const DECLARE_TLV_DB_SCALE(spk_tlv, -5700, 100, 0); + +static const struct snd_kcontrol_new nau8810_snd_controls[] = { + + SOC_ENUM("ADC Companding", nau8810_enum[0]), + SOC_ENUM("DAC Companding", nau8810_enum[1]), + SOC_ENUM("DAC De-emphasis", nau8810_enum[2]), + SOC_ENUM("EQ Function", nau8810_enum[3]), + SOC_ENUM("EQ1 Cut Off", nau8810_enum[4]), + SOC_ENUM("EQ2 Bandwidth", nau8810_enum[5]), + SOC_ENUM("EQ2 Cut Off", nau8810_enum[6]), + SOC_ENUM("EQ3 Bandwidth", nau8810_enum[7]), + SOC_ENUM("EQ3 Cut Off", nau8810_enum[8]), + SOC_ENUM("EQ4 Bandwidth", nau8810_enum[9]), + SOC_ENUM("EQ4 Cut Off", nau8810_enum[10]), + SOC_ENUM("EQ5 Cut Off", nau8810_enum[11]), + SOC_ENUM("ALC Mode", nau8810_enum[12]), + + SOC_SINGLE("Digital Loopback Switch", NAU8810_REG_COMP, + NAU8810_ADDAP_SFT, 1, 0), + + SOC_SINGLE("DAC Inversion Switch", NAU8810_REG_DAC, + NAU8810_DACPL_SFT, 1, 0), + SOC_SINGLE_TLV("Playback Gain", NAU8810_REG_DACGAIN, + NAU8810_DACGAIN_SFT, 0xff, 0, digital_tlv), + + SOC_SINGLE("High Pass Filter Switch", NAU8810_REG_ADC, + NAU8810_HPFEN_SFT, 1, 0), + SOC_SINGLE("High Pass Cut Off", NAU8810_REG_ADC, + NAU8810_HPF_SFT, 0x7, 0), + + SOC_SINGLE("ADC Inversion Switch", NAU8810_REG_ADC, + NAU8810_ADCPL_SFT, 1, 0), + SOC_SINGLE_TLV("Capture Gain", NAU8810_REG_ADCGAIN, + NAU8810_ADCGAIN_SFT, 0xff, 0, digital_tlv), + + SOC_SINGLE_TLV("EQ1 Gain", NAU8810_REG_EQ1, + NAU8810_EQ1GC_SFT, 0x18, 1, eq_tlv), + SOC_SINGLE_TLV("EQ2 Gain", NAU8810_REG_EQ2, + NAU8810_EQ2GC_SFT, 0x18, 1, eq_tlv), + SOC_SINGLE_TLV("EQ3 Gain", NAU8810_REG_EQ3, + NAU8810_EQ3GC_SFT, 0x18, 1, eq_tlv), + SOC_SINGLE_TLV("EQ4 Gain", NAU8810_REG_EQ4, + NAU8810_EQ4GC_SFT, 0x18, 1, eq_tlv), + SOC_SINGLE_TLV("EQ5 Gain", NAU8810_REG_EQ5, + NAU8810_EQ5GC_SFT, 0x18, 1, eq_tlv), + + SOC_SINGLE("DAC Limiter Switch", NAU8810_REG_DACLIM1, + NAU8810_DACLIMEN_SFT, 1, 0), + SOC_SINGLE("DAC Limiter Decay", NAU8810_REG_DACLIM1, + NAU8810_DACLIMDCY_SFT, 0xf, 0), + SOC_SINGLE("DAC Limiter Attack", NAU8810_REG_DACLIM1, + NAU8810_DACLIMATK_SFT, 0xf, 0), + + SOC_SINGLE("DAC Limiter Threshold", NAU8810_REG_DACLIM2, + NAU8810_DACLIMTHL_SFT, 0x7, 0), + SOC_SINGLE("DAC Limiter Boost", NAU8810_REG_DACLIM2, + NAU8810_DACLIMBST_SFT, 0xf, 0), + + SOC_SINGLE("ALC Enable Switch", NAU8810_REG_ALC1, + NAU8810_ALCEN_SFT, 1, 0), + SOC_SINGLE("ALC Max Gain", NAU8810_REG_ALC1, + NAU8810_ALCMXGAIN_SFT, 0x7, 0), + SOC_SINGLE("ALC Min Gain", NAU8810_REG_ALC1, + NAU8810_ALCMINGAIN_SFT, 0x7, 0), + + SOC_SINGLE("ALC ZC Switch", NAU8810_REG_ALC2, + NAU8810_ALCZC_SFT, 1, 0), + SOC_SINGLE("ALC Hold", NAU8810_REG_ALC2, + NAU8810_ALCHT_SFT, 0xf, 0), + SOC_SINGLE("ALC Target", NAU8810_REG_ALC2, + NAU8810_ALCSL_SFT, 0xf, 0), + + SOC_SINGLE("ALC Decay", NAU8810_REG_ALC3, + NAU8810_ALCDCY_SFT, 0xf, 0), + SOC_SINGLE("ALC Attack", NAU8810_REG_ALC3, + NAU8810_ALCATK_SFT, 0xf, 0), + + SOC_SINGLE("ALC Noise Gate Switch", NAU8810_REG_NOISEGATE, + NAU8810_ALCNEN_SFT, 1, 0), + SOC_SINGLE("ALC Noise Gate Threshold", NAU8810_REG_NOISEGATE, + NAU8810_ALCNTH_SFT, 0x7, 0), + + SOC_SINGLE("PGA ZC Switch", NAU8810_REG_PGAGAIN, + NAU8810_PGAZC_SFT, 1, 0), + SOC_SINGLE_TLV("PGA Volume", NAU8810_REG_PGAGAIN, + NAU8810_PGAGAIN_SFT, 0x3f, 0, inpga_tlv), + + SOC_SINGLE("Speaker ZC Switch", NAU8810_REG_SPKGAIN, + NAU8810_SPKZC_SFT, 1, 0), + SOC_SINGLE("Speaker Mute Switch", NAU8810_REG_SPKGAIN, + NAU8810_SPKMT_SFT, 1, 0), + SOC_SINGLE_TLV("Speaker Volume", NAU8810_REG_SPKGAIN, + NAU8810_SPKGAIN_SFT, 0x3f, 0, spk_tlv), + + SOC_SINGLE("Capture Boost(+20dB)", NAU8810_REG_ADCBOOST, + NAU8810_PGABST_SFT, 1, 0), + SOC_SINGLE("Mono Mute Switch", NAU8810_REG_MONOMIX, + NAU8810_MOUTMXMT_SFT, 1, 0), + + SOC_SINGLE("DAC Oversampling Rate(128x) Switch", NAU8810_REG_DAC, + NAU8810_DACOS_SFT, 1, 0), + SOC_SINGLE("ADC Oversampling Rate(128x) Switch", NAU8810_REG_ADC, + NAU8810_ADCOS_SFT, 1, 0), +}; + +/* Speaker Output Mixer */ +static const struct snd_kcontrol_new nau8810_speaker_mixer_controls[] = { + SOC_DAPM_SINGLE("Line Bypass Switch", NAU8810_REG_SPKMIX, + NAU8810_BYPSPK_SFT, 1, 0), + SOC_DAPM_SINGLE("PCM Playback Switch", NAU8810_REG_SPKMIX, + NAU8810_DACSPK_SFT, 1, 0), +}; + +/* Mono Output Mixer */ +static const struct snd_kcontrol_new nau8810_mono_mixer_controls[] = { + SOC_DAPM_SINGLE("Line Bypass Switch", NAU8810_REG_MONOMIX, + NAU8810_BYPMOUT_SFT, 1, 0), + SOC_DAPM_SINGLE("PCM Playback Switch", NAU8810_REG_MONOMIX, + NAU8810_DACMOUT_SFT, 1, 0), +}; + +/* PGA Mute */ +static const struct snd_kcontrol_new nau8810_inpga_mute[] = { + SOC_DAPM_SINGLE("PGA Mute Switch", NAU8810_REG_PGAGAIN, + NAU8810_PGAMT_SFT, 1, 0), +}; + +/* Input PGA */ +static const struct snd_kcontrol_new nau8810_inpga[] = { + SOC_DAPM_SINGLE("MicN Switch", NAU8810_REG_INPUT_SIGNAL, + NAU8810_NMICPGA_SFT, 1, 0), + SOC_DAPM_SINGLE("MicP Switch", NAU8810_REG_INPUT_SIGNAL, + NAU8810_PMICPGA_SFT, 1, 0), +}; + +/* Mic Input boost vol */ +static const struct snd_kcontrol_new nau8810_mic_boost_controls = + SOC_DAPM_SINGLE("Mic Volume", NAU8810_REG_ADCBOOST, + NAU8810_PMICBSTGAIN_SFT, 0x7, 0); + +static const struct snd_soc_dapm_widget nau8810_dapm_widgets[] = { + SND_SOC_DAPM_MIXER("Speaker Mixer", NAU8810_REG_POWER3, + NAU8810_SPKMX_EN_SFT, 0, &nau8810_speaker_mixer_controls[0], + ARRAY_SIZE(nau8810_speaker_mixer_controls)), + SND_SOC_DAPM_MIXER("Mono Mixer", NAU8810_REG_POWER3, + NAU8810_MOUTMX_EN_SFT, 0, &nau8810_mono_mixer_controls[0], + ARRAY_SIZE(nau8810_mono_mixer_controls)), + SND_SOC_DAPM_DAC("DAC", "HiFi Playback", NAU8810_REG_POWER3, + NAU8810_DAC_EN_SFT, 0), + SND_SOC_DAPM_ADC("ADC", "HiFi Capture", NAU8810_REG_POWER2, + NAU8810_ADC_EN_SFT, 0), + SND_SOC_DAPM_PGA("SpkN Out", NAU8810_REG_POWER3, + NAU8810_NSPK_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("SpkP Out", NAU8810_REG_POWER3, + NAU8810_PSPK_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mono Out", NAU8810_REG_POWER3, + NAU8810_MOUT_EN_SFT, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("Input PGA", NAU8810_REG_POWER2, + NAU8810_PGA_EN_SFT, 0, nau8810_inpga, + ARRAY_SIZE(nau8810_inpga)), + SND_SOC_DAPM_MIXER("Input Boost Stage", NAU8810_REG_POWER2, + NAU8810_BST_EN_SFT, 0, nau8810_inpga_mute, + ARRAY_SIZE(nau8810_inpga_mute)), + + SND_SOC_DAPM_SUPPLY("Mic Bias", NAU8810_REG_POWER1, + NAU8810_MICBIAS_EN_SFT, 0, NULL, 0), + + SND_SOC_DAPM_INPUT("MICN"), + SND_SOC_DAPM_INPUT("MICP"), + SND_SOC_DAPM_OUTPUT("MONOOUT"), + SND_SOC_DAPM_OUTPUT("SPKOUTP"), + SND_SOC_DAPM_OUTPUT("SPKOUTN"), +}; + +static const struct snd_soc_dapm_route nau8810_dapm_routes[] = { + /* Mono output mixer */ + {"Mono Mixer", "PCM Playback Switch", "DAC"}, + {"Mono Mixer", "Line Bypass Switch", "Input Boost Stage"}, + + /* Speaker output mixer */ + {"Speaker Mixer", "PCM Playback Switch", "DAC"}, + {"Speaker Mixer", "Line Bypass Switch", "Input Boost Stage"}, + + /* Outputs */ + {"Mono Out", NULL, "Mono Mixer"}, + {"MONOOUT", NULL, "Mono Out"}, + {"SpkN Out", NULL, "Speaker Mixer"}, + {"SpkP Out", NULL, "Speaker Mixer"}, + {"SPKOUTN", NULL, "SpkN Out"}, + {"SPKOUTP", NULL, "SpkP Out"}, + + /* Input Boost Stage */ + {"ADC", NULL, "Input Boost Stage"}, + {"Input Boost Stage", NULL, "Input PGA"}, + {"Input Boost Stage", NULL, "MICP"}, + + /* Input PGA */ + {"Input PGA", "MicN Switch", "MICN"}, + {"Input PGA", "MicP Switch", "MICP"}, +}; + +static int nau8810_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct nau8810 *nau8810 = snd_soc_codec_get_drvdata(codec); + + nau8810->sysclk = freq; + dev_dbg(nau8810->dev, "master sysclk %dHz\n", nau8810->sysclk); + + return 0; +} + +static int nau8810_config_clkdiv(struct nau8810 *nau8810, int div, int rate) +{ + struct regmap *regmap = nau8810->regmap; + int i, sclk, imclk; + + switch (nau8810->div_id) { + case NAU8810_MCLK_DIV_PLL: + /* master clock from PLL and enable PLL */ + regmap_update_bits(regmap, NAU8810_REG_CLOCK, + NAU8810_MCLKSEL_MASK, (div << NAU8810_MCLKSEL_SFT)); + regmap_update_bits(regmap, NAU8810_REG_POWER1, + NAU8810_PLL_EN, NAU8810_PLL_EN); + regmap_update_bits(regmap, NAU8810_REG_CLOCK, + NAU8810_CLKM_MASK, NAU8810_CLKM_PLL); + break; + + case NAU8810_MCLK_DIV_MCLK: + /* Configure the master clock prescaler div to make system + * clock to approximate the internal master clock (IMCLK); + * and large or equal to IMCLK. + */ + div = 0; + imclk = rate * 256; + for (i = 1; i < ARRAY_SIZE(nau8810_mclk_scaler); i++) { + sclk = (nau8810->sysclk * 10) / + nau8810_mclk_scaler[i]; + if (sclk < imclk) + break; + div = i; + } + dev_dbg(nau8810->dev, + "master clock prescaler %x for fs %d\n", div, rate); + + /* master clock from MCLK and disable PLL */ + regmap_update_bits(regmap, NAU8810_REG_CLOCK, + NAU8810_MCLKSEL_MASK, (div << NAU8810_MCLKSEL_SFT)); + regmap_update_bits(regmap, NAU8810_REG_CLOCK, + NAU8810_CLKM_MASK, NAU8810_CLKM_MCLK); + regmap_update_bits(regmap, NAU8810_REG_POWER1, + NAU8810_PLL_EN, 0); + break; + + case NAU8810_BCLK_DIV: + regmap_update_bits(regmap, NAU8810_REG_CLOCK, + NAU8810_BCLKSEL_MASK, (div << NAU8810_BCLKSEL_SFT)); + break; + + default: + return -EINVAL; + } + + return 0; +} + +static int nau8810_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) +{ + struct snd_soc_codec *codec = dai->codec; + struct nau8810 *nau8810 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + nau8810->div_id = div_id; + if (div_id != NAU8810_MCLK_DIV_MCLK) + /* Defer the master clock prescaler configuration to DAI + * hardware parameter if master clock from MCLK because + * it needs runtime fs information to get the proper div. + */ + ret = nau8810_config_clkdiv(nau8810, div, 0); + + return ret; +} + +static int nau8810_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct nau8810 *nau8810 = snd_soc_codec_get_drvdata(codec); + u16 ctrl1_val = 0, ctrl2_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + ctrl2_val |= NAU8810_CLKIO_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl1_val |= NAU8810_AIFMT_I2S; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl1_val |= NAU8810_AIFMT_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + ctrl1_val |= NAU8810_AIFMT_PCM_A; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + ctrl1_val |= NAU8810_BCLKP_IB | NAU8810_FSP_IF; + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl1_val |= NAU8810_BCLKP_IB; + break; + case SND_SOC_DAIFMT_NB_IF: + ctrl1_val |= NAU8810_FSP_IF; + break; + default: + return -EINVAL; + } + + regmap_update_bits(nau8810->regmap, NAU8810_REG_IFACE, + NAU8810_AIFMT_MASK | NAU8810_FSP_IF | + NAU8810_BCLKP_IB, ctrl1_val); + regmap_update_bits(nau8810->regmap, NAU8810_REG_CLOCK, + NAU8810_CLKIO_MASK, ctrl2_val); + + return 0; +} + +static int nau8810_pcm_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 nau8810 *nau8810 = snd_soc_codec_get_drvdata(codec); + int val_len = 0, val_rate = 0; + + switch (params_width(params)) { + case 16: + break; + case 20: + val_len |= NAU8810_WLEN_20; + break; + case 24: + val_len |= NAU8810_WLEN_24; + break; + case 32: + val_len |= NAU8810_WLEN_32; + break; + } + + switch (params_rate(params)) { + case 8000: + val_rate |= NAU8810_SMPLR_8K; + break; + case 11025: + val_rate |= NAU8810_SMPLR_12K; + break; + case 16000: + val_rate |= NAU8810_SMPLR_16K; + break; + case 22050: + val_rate |= NAU8810_SMPLR_24K; + break; + case 32000: + val_rate |= NAU8810_SMPLR_32K; + break; + case 44100: + case 48000: + break; + } + + regmap_update_bits(nau8810->regmap, NAU8810_REG_IFACE, + NAU8810_WLEN_MASK, val_len); + regmap_update_bits(nau8810->regmap, NAU8810_REG_SMPLR, + NAU8810_SMPLR_MASK, val_rate); + + /* If the master clock is from MCLK, provide the runtime FS for driver + * to get the master clock prescaler configuration. + */ + if (nau8810->div_id == NAU8810_MCLK_DIV_MCLK) + nau8810_config_clkdiv(nau8810, 0, params_rate(params)); + + return 0; +} + +static int nau8810_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct nau8810 *nau8810 = snd_soc_codec_get_drvdata(codec); + struct regmap *map = nau8810->regmap; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + regmap_update_bits(map, NAU8810_REG_POWER1, + NAU8810_REFIMP_MASK, NAU8810_REFIMP_80K); + break; + + case SND_SOC_BIAS_STANDBY: + regmap_update_bits(map, NAU8810_REG_POWER1, + NAU8810_IOBUF_EN | NAU8810_ABIAS_EN, + NAU8810_IOBUF_EN | NAU8810_ABIAS_EN); + + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { + regcache_sync(map); + regmap_update_bits(map, NAU8810_REG_POWER1, + NAU8810_REFIMP_MASK, NAU8810_REFIMP_3K); + mdelay(100); + } + regmap_update_bits(map, NAU8810_REG_POWER1, + NAU8810_REFIMP_MASK, NAU8810_REFIMP_300K); + break; + + case SND_SOC_BIAS_OFF: + regmap_write(map, NAU8810_REG_POWER1, 0); + regmap_write(map, NAU8810_REG_POWER2, 0); + regmap_write(map, NAU8810_REG_POWER3, 0); + break; + } + + return 0; +} + + +#define NAU8810_RATES (SNDRV_PCM_RATE_8000_48000) + +#define NAU8810_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static const struct snd_soc_dai_ops nau8810_ops = { + .hw_params = nau8810_pcm_hw_params, + .set_fmt = nau8810_set_dai_fmt, + .set_sysclk = nau8810_set_sysclk, + .set_clkdiv = nau8810_set_clkdiv, +}; + +static struct snd_soc_dai_driver nau8810_dai = { + .name = "nau8810-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, /* Only 1 channel of data */ + .rates = NAU8810_RATES, + .formats = NAU8810_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, /* Only 1 channel of data */ + .rates = NAU8810_RATES, + .formats = NAU8810_FORMATS, + }, + .ops = &nau8810_ops, + .symmetric_rates = 1, +}; + +static const struct regmap_config nau8810_regmap_config = { + .reg_bits = 7, + .val_bits = 9, + + .max_register = NAU8810_REG_MAX, + .readable_reg = nau8810_readable_reg, + .writeable_reg = nau8810_writeable_reg, + .volatile_reg = nau8810_volatile_reg, + .reg_read = nau8810_reg_read, + .reg_write = nau8810_reg_write, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = nau8810_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(nau8810_reg_defaults), +}; + +static int nau8810_probe(struct snd_soc_codec *codec) +{ + struct nau8810 *nau8810 = snd_soc_codec_get_drvdata(codec); + + regmap_write(nau8810->regmap, NAU8810_REG_RESET, 0x00); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_nau8810 = { + .probe = nau8810_probe, + .set_bias_level = nau8810_set_bias_level, + .suspend_bias_off = true, + + .controls = nau8810_snd_controls, + .num_controls = ARRAY_SIZE(nau8810_snd_controls), + .dapm_widgets = nau8810_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(nau8810_dapm_widgets), + .dapm_routes = nau8810_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(nau8810_dapm_routes), +}; + +static int nau8810_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct nau8810 *nau8810 = dev_get_platdata(dev); + + if (!nau8810) { + nau8810 = devm_kzalloc(dev, sizeof(*nau8810), GFP_KERNEL); + if (!nau8810) + return -ENOMEM; + } + i2c_set_clientdata(i2c, nau8810); + + nau8810->regmap = devm_regmap_init(dev, NULL, + i2c, &nau8810_regmap_config); + if (IS_ERR(nau8810->regmap)) + return PTR_ERR(nau8810->regmap); + + nau8810->dev = dev; + + return snd_soc_register_codec(dev, + &soc_codec_dev_nau8810, &nau8810_dai, 1); +} + +static int nau8810_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + + return 0; +} + +static const struct i2c_device_id nau8810_i2c_id[] = { + { "nau8810", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, nau8810_i2c_id); + +#ifdef CONFIG_OF +static const struct of_device_id nau8810_of_match[] = { + { .compatible = "nuvoton,nau8810", }, + { } +}; +MODULE_DEVICE_TABLE(of, nau8810_of_match); +#endif + +static struct i2c_driver nau8810_i2c_driver = { + .driver = { + .name = "nau8810", + .of_match_table = of_match_ptr(nau8810_of_match), + }, + .probe = nau8810_i2c_probe, + .remove = nau8810_i2c_remove, + .id_table = nau8810_i2c_id, +}; + +module_i2c_driver(nau8810_i2c_driver); + +MODULE_DESCRIPTION("ASoC NAU8810 driver"); +MODULE_AUTHOR("David Lin <ctlin0(a)nuvoton.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/nau8810.h b/sound/soc/codecs/nau8810.h new file mode 100755 index 0000000..174d26e --- /dev/null +++ b/sound/soc/codecs/nau8810.h @@ -0,0 +1,272 @@ +/* + * NAU8810 ALSA SoC audio driver + * + * Copyright 2016 Nuvoton Technology Corp. + * Author: David Lin <ctlin0(a)nuvoton.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __NAU8810_H__ +#define __NAU8810_H__ + +#define NAU8810_REG_RESET 0x00 +#define NAU8810_REG_POWER1 0x01 +#define NAU8810_REG_POWER2 0x02 +#define NAU8810_REG_POWER3 0x03 +#define NAU8810_REG_IFACE 0x04 +#define NAU8810_REG_COMP 0x05 +#define NAU8810_REG_CLOCK 0x06 +#define NAU8810_REG_SMPLR 0x07 +#define NAU8810_REG_DAC 0x0A +#define NAU8810_REG_DACGAIN 0x0B +#define NAU8810_REG_ADC 0x0E +#define NAU8810_REG_ADCGAIN 0x0F +#define NAU8810_REG_EQ1 0x12 +#define NAU8810_REG_EQ2 0x13 +#define NAU8810_REG_EQ3 0x14 +#define NAU8810_REG_EQ4 0x15 +#define NAU8810_REG_EQ5 0x16 +#define NAU8810_REG_DACLIM1 0x18 +#define NAU8810_REG_DACLIM2 0x19 +#define NAU8810_REG_NOTCH1 0x1B +#define NAU8810_REG_NOTCH2 0x1C +#define NAU8810_REG_NOTCH3 0x1D +#define NAU8810_REG_NOTCH4 0x1E +#define NAU8810_REG_ALC1 0x20 +#define NAU8810_REG_ALC2 0x21 +#define NAU8810_REG_ALC3 0x22 +#define NAU8810_REG_NOISEGATE 0x23 +#define NAU8810_REG_PLLN 0x24 +#define NAU8810_REG_PLLK1 0x25 +#define NAU8810_REG_PLLK2 0x26 +#define NAU8810_REG_PLLK3 0x27 +#define NAU8810_REG_ATTEN 0x28 +#define NAU8810_REG_INPUT_SIGNAL 0x2C +#define NAU8810_REG_PGAGAIN 0x2D +#define NAU8810_REG_ADCBOOST 0x2F +#define NAU8810_REG_OUTPUT 0x31 +#define NAU8810_REG_SPKMIX 0x32 +#define NAU8810_REG_SPKGAIN 0x36 +#define NAU8810_REG_MONOMIX 0x38 +#define NAU8810_REG_POWER4 0x3A +#define NAU8810_REG_TSLOTCTL1 0x3B +#define NAU8810_REG_TSLOTCTL2 0x3C +#define NAU8810_REG_DEVICE_REVID 0x3E +#define NAU8810_REG_I2C_DEVICEID 0x3F +#define NAU8810_REG_ADDITIONID 0x40 +#define NAU8810_REG_RESERVE 0x41 +#define NAU8810_REG_OUTCTL 0x45 +#define NAU8810_REG_ALC1ENHAN1 0x46 +#define NAU8810_REG_ALC1ENHAN2 0x47 +#define NAU8810_REG_MISCCTL 0x49 +#define NAU8810_REG_OUTTIEOFF 0x4B +#define NAU8810_REG_AGCP2POUT 0x4C +#define NAU8810_REG_AGCPOUT 0x4D +#define NAU8810_REG_AMTCTL 0x4E +#define NAU8810_REG_OUTTIEOFFMAN 0x4F +#define NAU8810_REG_MAX NAU8810_REG_OUTTIEOFFMAN + + +/* NAU8810_REG_POWER1 (0x1) */ +#define NAU8810_DCBUF_EN (0x1 << 8) +#define NAU8810_PLL_EN (0x1 << 5) +#define NAU8810_MICBIAS_EN_SFT 4 +#define NAU8810_ABIAS_EN (0x1 << 3) +#define NAU8810_IOBUF_EN (0x1 << 2) +#define NAU8810_REFIMP_MASK 0x3 +#define NAU8810_REFIMP_DIS 0x0 +#define NAU8810_REFIMP_80K 0x1 +#define NAU8810_REFIMP_300K 0x2 +#define NAU8810_REFIMP_3K 0x3 + +/* NAU8810_REG_POWER2 (0x2) */ +#define NAU8810_BST_EN_SFT 4 +#define NAU8810_PGA_EN_SFT 2 +#define NAU8810_ADC_EN_SFT 0 + +/* NAU8810_REG_POWER3 (0x3) */ +#define NAU8810_DAC_EN_SFT 0 +#define NAU8810_SPKMX_EN_SFT 2 +#define NAU8810_MOUTMX_EN_SFT 3 +#define NAU8810_PSPK_EN_SFT 5 +#define NAU8810_NSPK_EN_SFT 6 +#define NAU8810_MOUT_EN_SFT 7 + +/* NAU8810_REG_IFACE (0x4) */ +#define NAU8810_AIFMT_SFT 3 +#define NAU8810_AIFMT_MASK (0x3 << NAU8810_AIFMT_SFT) +#define NAU8810_AIFMT_RIGHT (0x0 << NAU8810_AIFMT_SFT) +#define NAU8810_AIFMT_LEFT (0x1 << NAU8810_AIFMT_SFT) +#define NAU8810_AIFMT_I2S (0x2 << NAU8810_AIFMT_SFT) +#define NAU8810_AIFMT_PCM_A (0x3 << NAU8810_AIFMT_SFT) +#define NAU8810_WLEN_SFT 5 +#define NAU8810_WLEN_MASK (0x3 << NAU8810_WLEN_SFT) +#define NAU8810_WLEN_16 (0x0 << NAU8810_WLEN_SFT) +#define NAU8810_WLEN_20 (0x1 << NAU8810_WLEN_SFT) +#define NAU8810_WLEN_24 (0x2 << NAU8810_WLEN_SFT) +#define NAU8810_WLEN_32 (0x3 << NAU8810_WLEN_SFT) +#define NAU8810_FSP_IF (0x1 << 7) +#define NAU8810_BCLKP_IB (0x1 << 8) + +/* NAU8810_REG_COMP (0x5) */ +#define NAU8810_ADDAP_SFT 0 +#define NAU8810_ADCCM_SFT 1 +#define NAU8810_DACCM_SFT 3 + +/* NAU8810_REG_CLOCK (0x6) */ +#define NAU8810_CLKIO_MASK 0x1 +#define NAU8810_CLKIO_SLAVE 0x0 +#define NAU8810_CLKIO_MASTER 0x1 +#define NAU8810_BCLKSEL_SFT 2 +#define NAU8810_BCLKSEL_MASK (0x7 << NAU8810_BCLKSEL_SFT) +#define NAU8810_BCLKDIV_1 (0x0 << NAU8810_BCLKSEL_SFT) +#define NAU8810_BCLKDIV_2 (0x1 << NAU8810_BCLKSEL_SFT) +#define NAU8810_BCLKDIV_4 (0x2 << NAU8810_BCLKSEL_SFT) +#define NAU8810_BCLKDIV_8 (0x3 << NAU8810_BCLKSEL_SFT) +#define NAU8810_BCLKDIV_16 (0x4 << NAU8810_BCLKSEL_SFT) +#define NAU8810_BCLKDIV_32 (0x5 << NAU8810_BCLKSEL_SFT) +#define NAU8810_MCLKSEL_SFT 5 +#define NAU8810_MCLKSEL_MASK (0x7 << NAU8810_MCLKSEL_SFT) +#define NAU8810_CLKM_SFT 8 +#define NAU8810_CLKM_MASK (0x1 << NAU8810_CLKM_SFT) +#define NAU8810_CLKM_MCLK (0x0 << NAU8810_CLKM_SFT) +#define NAU8810_CLKM_PLL (0x1 << NAU8810_CLKM_SFT) + +/* NAU8810_REG_SMPLR (0x7) */ +#define NAU8810_SMPLR_SFT 1 +#define NAU8810_SMPLR_MASK (0x7 << NAU8810_SMPLR_SFT) +#define NAU8810_SMPLR_48K (0x0 << NAU8810_SMPLR_SFT) +#define NAU8810_SMPLR_32K (0x1 << NAU8810_SMPLR_SFT) +#define NAU8810_SMPLR_24K (0x2 << NAU8810_SMPLR_SFT) +#define NAU8810_SMPLR_16K (0x3 << NAU8810_SMPLR_SFT) +#define NAU8810_SMPLR_12K (0x4 << NAU8810_SMPLR_SFT) +#define NAU8810_SMPLR_8K (0x5 << NAU8810_SMPLR_SFT) + +/* NAU8810_REG_DAC (0xA) */ +#define NAU8810_DACPL_SFT 0 +#define NAU8810_DACOS_SFT 3 +#define NAU8810_DEEMP_SFT 4 + +/* NAU8810_REG_DACGAIN (0xB) */ +#define NAU8810_DACGAIN_SFT 0 + +/* NAU8810_REG_ADC (0xE) */ +#define NAU8810_ADCPL_SFT 0 +#define NAU8810_ADCOS_SFT 3 +#define NAU8810_HPF_SFT 4 +#define NAU8810_HPFEN_SFT 8 + +/* NAU8810_REG_ADCGAIN (0xF) */ +#define NAU8810_ADCGAIN_SFT 0 + +/* NAU8810_REG_EQ1 (0x12) */ +#define NAU8810_EQ1GC_SFT 0 +#define NAU8810_EQ1CF_SFT 5 +#define NAU8810_EQM_SFT 8 + +/* NAU8810_REG_EQ2 (0x13) */ +#define NAU8810_EQ2GC_SFT 0 +#define NAU8810_EQ2CF_SFT 5 +#define NAU8810_EQ2BW_SFT 8 + +/* NAU8810_REG_EQ3 (0x14) */ +#define NAU8810_EQ3GC_SFT 0 +#define NAU8810_EQ3CF_SFT 5 +#define NAU8810_EQ3BW_SFT 8 + +/* NAU8810_REG_EQ4 (0x15) */ +#define NAU8810_EQ4GC_SFT 0 +#define NAU8810_EQ4CF_SFT 5 +#define NAU8810_EQ4BW_SFT 8 + +/* NAU8810_REG_EQ5 (0x16) */ +#define NAU8810_EQ5GC_SFT 0 +#define NAU8810_EQ5CF_SFT 5 + +/* NAU8810_REG_DACLIM1 (0x18) */ +#define NAU8810_DACLIMATK_SFT 0 +#define NAU8810_DACLIMDCY_SFT 4 +#define NAU8810_DACLIMEN_SFT 8 + +/* NAU8810_REG_DACLIM2 (0x19) */ +#define NAU8810_DACLIMBST_SFT 0 +#define NAU8810_DACLIMTHL_SFT 4 + +/* NAU8810_REG_ALC1 (0x20) */ +#define NAU8810_ALCMINGAIN_SFT 0 +#define NAU8810_ALCMXGAIN_SFT 3 +#define NAU8810_ALCEN_SFT 8 + +/* NAU8810_REG_ALC2 (0x21) */ +#define NAU8810_ALCSL_SFT 0 +#define NAU8810_ALCHT_SFT 4 +#define NAU8810_ALCZC_SFT 8 + +/* NAU8810_REG_ALC3 (0x22) */ +#define NAU8810_ALCATK_SFT 0 +#define NAU8810_ALCDCY_SFT 4 +#define NAU8810_ALCM_SFT 8 + +/* NAU8810_REG_NOISEGATE (0x23) */ +#define NAU8810_ALCNTH_SFT 0 +#define NAU8810_ALCNEN_SFT 3 + +/* NAU8810_REG_PLLN (0x24) */ +#define NAU8810_PLLN_MASK 0xF +#define NAU8810_PLLMCLK_DIV2 (0x1 << 4) + +/* NAU8810_REG_PLLK1 (0x25) */ +#define NAU8810_PLLK1_MASK 0x3F + +/* NAU8810_REG_PLLK2 (0x26) */ +#define NAU8810_PLLK2_MASK 0x1FF + +/* NAU8810_REG_PLLK3 (0x27) */ +#define NAU8810_PLLK3_MASK 0x1FF + +/* NAU8810_REG_INPUT_SIGNAL (0x2C) */ +#define NAU8810_PMICPGA_SFT 0 +#define NAU8810_NMICPGA_SFT 1 + +/* NAU8810_REG_PGAGAIN (0x2D) */ +#define NAU8810_PGAGAIN_SFT 0 +#define NAU8810_PGAMT_SFT 6 +#define NAU8810_PGAZC_SFT 7 + +/* NAU8810_REG_ADCBOOST (0x2F) */ +#define NAU8810_PMICBSTGAIN_SFT 4 +#define NAU8810_PGABST_SFT 8 + +/* NAU8810_REG_SPKMIX (0x32) */ +#define NAU8810_DACSPK_SFT 0 +#define NAU8810_BYPSPK_SFT 1 + +/* NAU8810_REG_SPKGAIN (0x36) */ +#define NAU8810_SPKGAIN_SFT 0 +#define NAU8810_SPKMT_SFT 6 +#define NAU8810_SPKZC_SFT 7 + +/* NAU8810_REG_MONOMIX (0x38) */ +#define NAU8810_DACMOUT_SFT 0 +#define NAU8810_BYPMOUT_SFT 1 +#define NAU8810_MOUTMXMT_SFT 6 + + +/* Clock divider Id's */ +enum { + NAU8810_MCLK_DIV_PLL, + NAU8810_MCLK_DIV_MCLK, + NAU8810_BCLK_DIV, +}; + +struct nau8810 { + struct device *dev; + struct regmap *regmap; + int sysclk; + int div_id; +}; + +#endif -- 2.6.4

2 8

[alsa-devel] [PATCH 00/27 v2] ALSA: seq: obsolete change of address limit
by Takashi Sakamoto 08 Aug '16

08 Aug '16

Hi, For purpose of this patchset, please refer to my previous post: [alsa-devel] obsolete change of address limit http://mailman.alsa-project.org/pipermail/alsa-devel/2016-August/110937.html Changes in v2: - Use '_IOC_SIZE' macro to calculate the size of argument of ioctl command. - Keep redundant longer name. - Improve comments. - Merge some patches relevant to the same features. I give up using 'dir' field in each ioctl command to decide copying, because some ioctl commands for ALSA sequencer core includes bugs against actual implementations. Takashi Sakamoto (27): ALSA: seq: add const qualifier to table of functions for ioctl ALSA: seq: fulfill callback entry for ioctl ALSA: seq: copy ioctl data from user space to kernel stack ALSA: seq: add documentation for snd_seq_kernel_client_ctl ALSA: seq: obsolete address mode in compatibility layer ALSA: seq: obsolete change of address limit in in-kernel path for ioctl ALSA: seq: obsolete address limit helper ALSA: seq: optimize pversion function to new design ALSA: seq: optimize client_id function to new design ALSA: seq: optimize system_info function to new design ALSA: seq: optimize running mode function to new design ALSA: seq: optimize get/set_client_info function to new design ALSA: seq: optimize create/delete_port function to new design ALSA: seq: optimize get/set_port_info function to new design ALSA: seq: optimize subscribe_port function to new design ALSA: seq: optimize create/delete_queue function to new design ALSA: seq: optimize get/set_queue_info function to new design ALSA: seq: optimize get/set_queue_info function to new design ALSA: seq: optimize get_named_queue function to new design ALSA: seq: optimize get_queue_status function to new design ALSA: seq: optimize get/set_queue_tempo function to new design ALSA: seq: optimize get/set_queue_timer function to new design ALSA: seq: optimize get/set_queue_client function to new design ALSA: seq: optimize get/set_client_pool function to new design ALSA: seq: optimize remove_events function to new design ALSA: seq: optimize get_subscription function to new design ALSA: seq: optimize query functions for subscribes, next client and port to new design sound/core/seq/seq_clientmgr.c | 750 +++++++++++++++++++---------------------- sound/core/seq/seq_compat.c | 26 +- 2 files changed, 353 insertions(+), 423 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH 0/2] ASoC: da7218: Small clocking related updates for driver
by Adam Thomson 08 Aug '16

08 Aug '16

This patch set contains two updates relating to clocking. The first removes the 32KHz PLL feature as this is no longer supported, and the second provides MCLK effciency improvements to avoid unnecessary MCLK enabling. Changes are based on v4.8-rc1 Linux kernel. Adam Thomson (2): ASoC: da7218: Remove 32KHz PLL mode from driver ASoC: da7218: Improve driver efficiency with regards to MCLK usage sound/soc/codecs/da7218.c | 34 +++++++++++++++------------------- sound/soc/codecs/da7218.h | 2 -- 2 files changed, 15 insertions(+), 21 deletions(-) -- 1.9.3

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[alsa-devel] [PATCH 0/3] drm/i2c: tda998x ASoC hdmi-codec support + BeagleBoneBlack audio support
by Jyri Sarha 08 Aug '16

08 Aug '16

These patches are just a rebase of the same patches from my "[PATCH v4 0/7] Implement generic ASoC HDMI codec and use it in tda998x" -patch series. I have only made updates required by efc9194bcff8 ("ASoC: hdmi-codec: callback function will be called with private data") -patch. The first patch changes tda998x pdata, so if there are any out of tree users of tda998x-driver the out of tree code needs to be updated. Jyri Sarha (3): drm/i2c: tda998x: Improve tda998x_configure_audio() audio related pdata drm/i2c: tda998x: Register ASoC hdmi-codec and add audio DT binding ARM: dts: am335x-boneblack: Add HDMI audio support .../devicetree/bindings/display/bridge/tda998x.txt | 18 ++ arch/arm/boot/dts/am335x-boneblack.dts | 71 +++++- drivers/gpu/drm/i2c/Kconfig | 1 + drivers/gpu/drm/i2c/tda998x_drv.c | 276 ++++++++++++++++++--- include/drm/i2c/tda998x.h | 24 +- include/dt-bindings/display/tda998x.h | 7 + 6 files changed, 342 insertions(+), 55 deletions(-) create mode 100644 include/dt-bindings/display/tda998x.h -- 1.9.1

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[alsa-devel] [PATCH 000/159] ASoC: codec cleanup - codec duplicate functions
by Kuninori Morimoto 08 Aug '16

08 Aug '16

Hi Mark These are codec duplicate functions cleanup patches - v1 We would like to switch to "component" style base ASoC in the future instead of current "CPU/Codec/Platform/Card" style. Current "component" already has callback functions, and "codec" is based on "component", but "codec" is still requesting/using its original callback. This is bacause of historical reason, but this is the time to cleanup. Lars's opinion is that this kind of cleanup can be final stage, but I think these are needed for "component" base style switching These are very big patch-set, but it doesn't add new features. [001/159] - [158/159] move callback functions from codec to component, and last [159/159] will remove codec side callback. We need this kind of cleanup for .probe/.remove too, and on Platform too. This cares codec duplicate callback function only as 1st step. I will post codec .probe/.remove cleanup if these are accepted, and will post platform side cleanup too. I build-tested these patches with make allyesconfig on x86, and tested on Lager board. Best regards --- Kuninori Morimoto

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[alsa-devel] How to post cleanup patches ?
by Kuninori Morimoto 08 Aug '16

08 Aug '16

Hi Mark Now codec driver and component driver has duplicated callback function, and snd_soc_register_codec() just copied it to component side. int snd_soc_register_codec(xxx) { ... if (codec_drv->controls) { codec->component.controls = codec_drv->controls; codec->component.num_controls = codec_drv->num_controls; } if (codec_drv->dapm_widgets) { codec->component.dapm_widgets = codec_drv->dapm_widgets; codec->component.num_dapm_widgets = codec_drv->num_dapm_widgets; } if (codec_drv->dapm_routes) { codec->component.dapm_routes = codec_drv->dapm_routes; codec->component.num_dapm_routes = codec_drv->num_dapm_routes; } ... } I think we can cleanup this duplicated function, similar things happen on .probe, .remove, and platform side too. I would like to post cleanup patches for these. As 1st step, remove codec duplicate callback. But my concern is that you hate big-patch-set, and it is almost 70 patches. Can I post these patches to ML ? or should I use git pull request ? or I shouldn't post ? I will post main patches as sample patch Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH 0/7][REBASE] ASoC: simple-card: add parse_clk / parse_dai
by Kuninori Morimoto 08 Aug '16

08 Aug '16

Hi Mark These are rebase version patches for parse_clk / parse_dai on simple utiles. I think today's topic/simple is missing [1/7] patch, but have [2/7] [3/7]. Because of this, it has bulid error, and you can't apply [4/7]. These patches add new functions on utils.c, and I noticed that it will have conflict with fix/simple branch. Because of these reasons, I think you want to redo topic/simple branch (?). These patches are based on fix/simple branch. Kuninori Morimoto (7): ASoC: simple-card-utils: add asoc_simple_card_parse_clk() ASoC: simple-card: use asoc_simple_card_parse_clk() ASoC: rsrc-card: use asoc_simple_card_parse_clk() ASoC: simple-card-utils: add asoc_simple_card_parse_dai() ASoC: simple-card: use asoc_simple_card_parse_dai() ASoC: rsrc-card: use asoc_simple_card_parse_dai() ASoC: simple-card: remove asoc_simple_card_sub_parse_of() include/sound/simple_card_utils.h | 25 ++++++++ sound/soc/generic/simple-card-utils.c | 69 ++++++++++++++++++++ sound/soc/generic/simple-card.c | 116 ++++++++++------------------------ sound/soc/sh/rcar/rsrc-card.c | 93 ++++++--------------------- 4 files changed, 146 insertions(+), 157 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH 0/7] ASoC: simple-card: add parse_clk / parse_dai
by Kuninori Morimoto 08 Aug '16

08 Aug '16

Hi Mark These add parse_clk / parse_dai to simple utiles. I know that we want to use "devm_" version of of_clk_get(), and now I'm posting its patch to kernel ML, but it still under discussion. Thus, this patch is using current style (with small fixup). I will replace it to devm version if it was accepted. Kuninori Morimoto (7): ASoC: simple-card-utils: add asoc_simple_card_parse_clk() ASoC: simple-card: use asoc_simple_card_parse_clk() ASoC: rsrc-card: use asoc_simple_card_parse_clk() ASoC: simple-card-utils: add asoc_simple_card_parse_dai() ASoC: simple-card: use asoc_simple_card_parse_dai() ASoC: rsrc-card: use asoc_simple_card_parse_dai() ASoC: simple-card: remove asoc_simple_card_sub_parse_of() include/sound/simple_card_utils.h | 25 ++++++++ sound/soc/generic/simple-card-utils.c | 69 ++++++++++++++++++++ sound/soc/generic/simple-card.c | 116 ++++++++++------------------------ sound/soc/sh/rcar/rsrc-card.c | 93 ++++++--------------------- 4 files changed, 146 insertions(+), 157 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH 0/4] ASoC: da7213: Device clocking updates and fixes
by Adam Thomson 08 Aug '16

08 Aug '16

This patch set contains a series of patches relating to device clocking. The changes are as follows: 1) Set correct default BCLKs per WCLK to achieve all DAI formats. Currently only 16-bit formats would operate correctly. 2) Only use MCLK as required, when an audio stream is active. Can be disabled all other times. 3) Refactoring of sysclk() and pll() functions to make them cleaner, removing need for unnecessary private flags, and resolve incorrect 32Khz mode configuration. 4) Improve 32KHz PLL locking with some small register updates when configuring PLL and when DAI is enabled. Patches are based on v4.7 Linux kernel Adam Thomson (4): ASoC: da7213: Default to 64 BCLKs per WCLK to support all formats ASoC: da7213: Improve driver efficiency with regards to MCLK usage ASoC: da7213: Refactor sysclk(), pll() functions to improve handling ASoC: da7213: Improve 32KHz mode PLL locking sound/soc/codecs/da7213.c | 134 +++++++++++++++++++++++++++++----------------- sound/soc/codecs/da7213.h | 12 +++-- 2 files changed, 94 insertions(+), 52 deletions(-) -- 1.9.3

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[alsa-devel] Audible Pop noise with CS4271 codec
by Vishal Thanki 08 Aug '16

08 Aug '16

Hi all, On a custom built am335x based board with CS4271 codec, I am observing the audible pop/click sound during playback start and playback stop (using aplay). The pop sound is observed with kernel 4.0 as well as on kernel 4.4. I have not tried the recent kernels but since there is no change in the CS4271 codec driver, I am assuming the problem can be reproduced in the latest kernel as well. I have experimented a bit with conditional "soft reset" (enable_soft_reset flag) which is added in driver and which seems to have little influence in pop sound. My experiment is like following: 1) Start the board with "soft reset" enabled in CS4271 driver 2) Play sine wave tone for 5 seconds using aplay --> The pop sound is present during starting and stopping of the playback 3) Play the silence file (no audio data - filled with zero) using aplay --> The pop sound is present only during starting of the playback 4) Play the silence file again using aplay --> The pop sound is present only during starting of the playback Now disable the "soft reset" in driver and 1) Play sine wave tone for 5 seconds using aplay --> The pop sound is present during starting and stopping of the playback 2) Play the silence file using aplay --> The pop sound is present only during starting of the playback 3) Play the silence file again using aplay --> The pop sound is gone 4) Play the silence file successively multiple times --> The pop sound not observed 5) Now play the sine wave tone again --> The pop sound is present during startup and stop. 6) Now play the silent file again --> The pop sound is present during startup, and no pop sound during stop. The driver handles the muting and unmuting of DAC in mute_stream DAI callback which looks okay to me. So I have added delays in mute_stream callback after "unmute" of DACs and noticed that unmuting of DAC is not causing the pop noise, but it is heard as soon as the audio data starts flowing (after .trigger() ?). I could also not figure out what problem the "soft reset" is addressing and how it is influencing the pop noise in successive silence file playback. Any help in further debugging would be greatly appreciated. Thanks, Vishal

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[alsa-devel] [PATCH] alsa-python: alsaseq: added binding for delete_port
by Maurizio Berti 07 Aug '16

07 Aug '16

Hello, I was going to send this patch via git, but I've troubles setting it up against gmail, then I hope I'm not doing something wrong by posting this directly here. I found out that pyalsa doesn't provide delete_port, then I added it; please, note, I'm not a programmer, I just do this for fun and I actually have no experience in C programming: I just understand (most of) it by reading the code, so I just used some knowledge and logic to add this binding. So far, it looks fine to me and works without any problem. Signed-off-by: MaurizioB <maurizio.berti(a)gmail.com> diff --git a/pyalsa/alsaseq.c b/pyalsa/alsaseq.c index ce332ed..2ad4d55 100644 --- a/pyalsa/alsaseq.c +++ b/pyalsa/alsaseq.c @@ -2343,6 +2343,44 @@ Sequencer_create_simple_port(SequencerObject *self, return PyInt_FromLong(port); } + +/** alsaseq.Sequencer delete_port() method: __doc__ */ +PyDoc_STRVAR(Sequencer_delete_port__doc__, + "delete_port(port_id) -> int" + "\n" + "Delete an existing port.\n" + "\n" + "Parameters:\n" + " port_id -- id of the port\n" + "Raises:\n" + " SequencerError: if ALSA can't delete the port" +); + + +/** alsaseq.Sequencer delete_port() method */ +static PyObject * +Sequencer_delete_port(SequencerObject *self, + PyObject *args, + PyObject *kwds) { + int port_id = 0; + int ret; + char *kwlist[] = { "port_id", NULL}; + + if (!PyArg_ParseTupleAndKeywords(args, kwds, "|i", kwlist, + &port_id)) { + return NULL; + } + + ret = snd_seq_delete_port(self->handle, port_id); + if (ret < 0) { + RAISESND(ret, "Failed to delete port"); + return NULL; + } + + Py_RETURN_NONE; +} + + /** alsaseq.Sequencer connection_list() method: __doc__ */ PyDoc_STRVAR(Sequencer_connection_list__doc__, "connection_list() -> list\n" @@ -3248,6 +3286,10 @@ static PyMethodDef Sequencer_methods[] = { (PyCFunction) Sequencer_create_simple_port, METH_VARARGS | METH_KEYWORDS, Sequencer_create_simple_port__doc__ }, + {"delete_port", + (PyCFunction) Sequencer_delete_port, + METH_VARARGS | METH_KEYWORDS, + Sequencer_delete_port__doc__ }, {"connection_list", (PyCFunction) Sequencer_connection_list, METH_VARARGS, -- 2.4.10 -- È difficile avere una convinzione precisa quando si parla delle ragioni del cuore. - "Sostiene Pereira", Antonio Tabucchi http://www.jidesk.net

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