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

2 7

[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

2 1

[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

1 0

[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

2 3

[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

1 27

[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

1 2

[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

3 15

[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

3 294

[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

3 9

[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

2 10

[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

2 14

[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

3 10

[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

1 0

[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

1 0

[alsa-devel] Alsa Application using the "default" plugin
by Hauke Krüger 07 Aug '16

07 Aug '16

Hi everyone, I have created an ALSA audio application which works fine for any kind of hardware device (all devices addressed by "hw:n,m"). Today, I planned to implement the functionality to also address plugins. My test case is the "default" plugin. When using this, I see significant differences: 1) Function "snd_pcm_hw_params_any" returns a value of 1. For hardware devices it returns 0. There is no documentation of the return type from that function. What does the 1 say? 2) I want to simultaneously capture and playback. Therefore I call "snd_pcm_link". This function fails with the error saying that this function is not implemented. Is that the intended behavior and how would I synchronize input and output? What I do to fix this is to start input and output ("snd_pcm_start") independently if "snd_pcm_link" fails. 3) The program stops working on the second call to "snd_pcm_writei": The function call simply never returns. I guess that the pcm output interface does not run at this moment and the output buffer has been completely filled before. All other functions calls work as expected but the described behavior prevents the application from working properly. And - to repeat - the same code works for all hardware soundcards "hw:n,m" which I tested. What is the recommended way to address the "default" plugin pcm device? Best regards Hauke

1 0

[alsa-devel] ASoC: sun4i-i2s: Restricting the MCLK
by Thomas Niederprüm 05 Aug '16

05 Aug '16

I am using the new sun4i-i2s driver in linux-next to run a sta326 codec via the pin header on a cubietruck (this involves some soldering as described in [1]). The codec can only be configured as slave and needs a master clock for its internal PLL. It is therefore great that, unlike many other SOCs, the a20 provides a MCLK that can be used to sync the codec. However, in my case there is a restriction imposed on the MCLK by the codec. As stated in the datasheet [2], the master clock should be at maximum 12.288 MHz/11.2896 MHz or otherwise the codec will not work (confirmed by experiment!). Since the mod clock of the a20's dai block is running at twice this frequency, I need to assure that the MLCK divider is at least 2. Actually it is rather straight forward to achieve this by changing the breaking condition in the divider calculation, as shown by the following one-line patch: diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index 687a8f8..8920844 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -206,7 +206,7 @@ static int sun4i_i2s_set_clk_rate(struct sun4i_i2s *i2s, clk_rate, rate); - if ((bclk_div >= 0) && (mclk_div >= 0)) + if ((bclk_div >= 0) && (mclk_div >= 1)) break; } Of course, this is just a hack to fix my use case and I am wondering what would be the best way to handle my corner case more generally? I see one option in introducing a DT property mlck_mindiv, or similar, that defaults to 1 and can be overwritten at the board level. I'm willing to prepare a proper patch to tackle this problem but right now I'm wondering if there a major objections to this approach or whether there is any better solution to the problem. Best regards, Thomas [1] https://hifiduino.wordpress.com/2014/03/07/cubieboard-for-audio/ [2] www.st.com/resource/en/datasheet/sta326.pdf

2 2

[alsa-devel] [PATCH v4 1/4] mfd: Add DT bindings documentation for Samsung Exynos LPASS
by Sylwester Nawrocki 05 Aug '16

05 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> --- There were few rather significant additions since v3 and I had to drop tags: Acked-by: Rob Herring <robh(a)kernel.org> Acked-by: Krzysztof Kozlowski <k.kozlowski(a)samsung.com> 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 | 21 +++++++++++++++++++++ 1 file changed, 21 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..7e97c0d --- /dev/null +++ b/Documentation/devicetree/bindings/mfd/samsung,exynos5433-lpass.txt @@ -0,0 +1,21 @@ + +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, WDT 0...1 +devices. + +Bindings of the sub-nodes are described in: + Documentation/devicetree/bindings/serial/samsung_uart.txt + Documentation/devicetree/bindings/sound/samsung-i2s.txt + Documentation/devicetree/bindings/dma/arm-pl330.txt -- 1.9.1

3 3

[alsa-devel] Stale data interleaved in capture driver
by Rob Nertney 05 Aug '16

05 Aug '16

My DMA is configured with a debug mode in which it sends an a simple increasing counter value in the format I wish to receive: an IRQ every 32Bytes (half-frame). it transfers 32KBytes of data, and then restarts from it's original offset. This is tested in bare-metal mode as working. I wrote my handler as described in the high-frequency interrupts described here: http://www.alsa-project.org/~tiwai/writing-an-alsa-driver/ch05s07.html My data is valid until 1/4 buffer full, then it interleaves stale (lower numbers) in, replacing the desired numbers. It's sporadic, and it appears to repeat, as if ALSA never returns to the 0-offset of dma_area. I've dumped a snip below. If the formatting is lost, I can post a google drive or an image somewhere. Thanks, Rob 00004000: 8182 8183 8184 8185 00004008: 8186 8187 8188 8189 00004010: 8190 8191 8192 1 00004018: 2 3 4 5 00004020: 6 7 8 8201 00004028: 8202 8203 8204 8205 00004030: 8206 8207 8208 8209 00004038: 8210 8211 8212 8213 00004040: 8214 8215 8216 8217 00004048: 8218 8219 8220 8221 00004050: 8222 8223 8224 8225 00004058: 8226 8227 8228 8229 00004060: 8230 8231 8232 8233 00004068: 8234 8235 8236 8237 00004070: 8238 8239 8240 49 00004078: 50 51 52 53 00004080: 54 55 56 8249 00004088: 8250 8251 8252 8253 00004090: 8254 8255 8256 65 00004098: 66 67 68 69 000040a0: 70 71 72 73 000040a8: 74 75 76 77 000040b0: 78 79 80 8273 000040b8: 8274 8275 8276 8277 000040c0: 8278 8279 8280 8281 000040c8: 8282 8283 8284 8285 000040d0: 8286 8287 8288 8289 000040d8: 8290 8291 8292 8293 000040e0: 8294 8295 8296 8297 000040e8: 8298 8299 8300 8301 000040f0: 8302 8303 8304 8305 000040f8: 8306 8307 8308 8309 00004100: 8310 8311 8312 121

2 6

[alsa-devel] [GIT PULL] Skylake firmware update
by Vinod Koul 05 Aug '16

05 Aug '16

Hi, Please PULL to recive updated Skylake audio firmware. The following changes since commit 6bc2c609988823744d1a45d33ee370c8310ec305: amdgpu: update stoney UVD firmware (2016-07-29 14:40:03 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/firmware.git skl for you to fetch changes up to c170c8d95794d6aedbaeea44674daaa96baf04f7: linux-firmware: intel: Update Skylake audio firmware (2016-08-04 16:09:21 +0530) ---------------------------------------------------------------- Vinod Koul (1): linux-firmware: intel: Update Skylake audio firmware WHENCE | 4 ++-- intel/dsp_fw_release.bin | 2 +- intel/dsp_fw_release_v958.bin | Bin 0 -> 221184 bytes 3 files changed, 3 insertions(+), 3 deletions(-) create mode 100644 intel/dsp_fw_release_v958.bin Thanks -- ~Vinod

2 1

[alsa-devel] [asoc:topic/simple 20/20] sound/soc/sh/rcar/rsrc-card.c:193:9: error: implicit declaration of function 'asoc_simple_card_parse_clk_cpu'
by kbuild test robot 05 Aug '16

05 Aug '16

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/simple head: c9a235da8a61cf9af7653fca341ded0881a64eca commit: c9a235da8a61cf9af7653fca341ded0881a64eca [20/20] ASoC: rsrc-card: use asoc_simple_card_parse_clk() config: arm-shmobile_defconfig (attached as .config) compiler: arm-linux-gnueabi-gcc (Debian 5.4.0-6) 5.4.0 20160609 reproduce: wget https://git.kernel.org/cgit/linux/kernel/git/wfg/lkp-tests.git/plain/sbin/m… -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout c9a235da8a61cf9af7653fca341ded0881a64eca # save the attached .config to linux build tree make.cross ARCH=arm All errors (new ones prefixed by >>): sound/soc/sh/rcar/rsrc-card.c: In function 'rsrc_card_parse_links': >> sound/soc/sh/rcar/rsrc-card.c:193:9: error: implicit declaration of function 'asoc_simple_card_parse_clk_cpu' [-Werror=implicit-function-declaration] ret = asoc_simple_card_parse_clk_cpu(np, dai_link, dai_props); ^ >> sound/soc/sh/rcar/rsrc-card.c:232:9: error: implicit declaration of function 'asoc_simple_card_parse_clk_codec' [-Werror=implicit-function-declaration] ret = asoc_simple_card_parse_clk_codec(np, dai_link, dai_props); ^ cc1: some warnings being treated as errors vim +/asoc_simple_card_parse_clk_cpu +193 sound/soc/sh/rcar/rsrc-card.c 187 dai_link->dpcm_merged_format = 1; 188 dai_link->cpu_of_node = args.np; 189 ret = snd_soc_of_get_dai_name(np, &dai_link->cpu_dai_name); 190 if (ret < 0) 191 return ret; 192 > 193 ret = asoc_simple_card_parse_clk_cpu(np, dai_link, dai_props); 194 if (ret < 0) 195 return ret; 196 197 ret = asoc_simple_card_set_dailink_name(dev, dai_link, 198 "fe.%s", 199 dai_link->cpu_dai_name); 200 if (ret < 0) 201 return ret; 202 203 /* 204 * In soc_bind_dai_link() will check cpu name after 205 * of_node matching if dai_link has cpu_dai_name. 206 * but, it will never match if name was created by 207 * fmt_single_name() remove cpu_dai_name if cpu_args 208 * was 0. See: 209 * fmt_single_name() 210 * fmt_multiple_name() 211 */ 212 if (!args.args_count) 213 dai_link->cpu_dai_name = NULL; 214 } else { 215 const struct rsrc_card_of_data *of_data; 216 217 of_data = of_device_get_match_data(dev); 218 219 /* FE is dummy */ 220 dai_link->cpu_of_node = NULL; 221 dai_link->cpu_dai_name = "snd-soc-dummy-dai"; 222 dai_link->cpu_name = "snd-soc-dummy"; 223 224 /* BE settings */ 225 dai_link->no_pcm = 1; 226 dai_link->be_hw_params_fixup = rsrc_card_be_hw_params_fixup; 227 dai_link->codec_of_node = args.np; 228 ret = snd_soc_of_get_dai_name(np, &dai_link->codec_dai_name); 229 if (ret < 0) 230 return ret; 231 > 232 ret = asoc_simple_card_parse_clk_codec(np, dai_link, dai_props); 233 if (ret < 0) 234 return ret; 235 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [asoc:topic/simple 19/20] sound/soc/generic/simple-card.c:350:8: error: implicit declaration of function 'asoc_simple_card_parse_clk_cpu'
by kbuild test robot 05 Aug '16

05 Aug '16

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/simple head: c9a235da8a61cf9af7653fca341ded0881a64eca commit: 28abd99b6e40741cd0e75be20817f23a3044d338 [19/20] ASoC: simple-card: use asoc_simple_card_parse_clk() config: i386-randconfig-s0-08050951 (attached as .config) compiler: gcc-6 (Debian 6.1.1-9) 6.1.1 20160705 reproduce: git checkout 28abd99b6e40741cd0e75be20817f23a3044d338 # save the attached .config to linux build tree make ARCH=i386 All errors (new ones prefixed by >>): sound/soc/generic/simple-card.c: In function 'asoc_simple_card_dai_link_of': >> sound/soc/generic/simple-card.c:350:8: error: implicit declaration of function 'asoc_simple_card_parse_clk_cpu' [-Werror=implicit-function-declaration] ret = asoc_simple_card_parse_clk_cpu(cpu, dai_link, cpu_dai); ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ >> sound/soc/generic/simple-card.c:354:8: error: implicit declaration of function 'asoc_simple_card_parse_clk_codec' [-Werror=implicit-function-declaration] ret = asoc_simple_card_parse_clk_codec(codec, dai_link, codec_dai); ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ cc1: some warnings being treated as errors -- /kbuild/src/rand/sound/soc/generic/simple-card.c: In function 'asoc_simple_card_dai_link_of': >> /kbuild/src/rand/sound/soc/generic/simple-card.c:350:8: error: implicit declaration of function 'asoc_simple_card_parse_clk_cpu' [-Werror=implicit-function-declaration] ret = asoc_simple_card_parse_clk_cpu(cpu, dai_link, cpu_dai); ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ >> /kbuild/src/rand/sound/soc/generic/simple-card.c:354:8: error: implicit declaration of function 'asoc_simple_card_parse_clk_codec' [-Werror=implicit-function-declaration] ret = asoc_simple_card_parse_clk_codec(codec, dai_link, codec_dai); ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ cc1: some warnings being treated as errors vim +/asoc_simple_card_parse_clk_cpu +350 sound/soc/generic/simple-card.c 344 ret = asoc_simple_card_sub_parse_of(plat, NULL, 345 &dai_link->platform_of_node, 346 NULL, NULL); 347 if (ret < 0) 348 goto dai_link_of_err; 349 > 350 ret = asoc_simple_card_parse_clk_cpu(cpu, dai_link, cpu_dai); 351 if (ret < 0) 352 goto dai_link_of_err; 353 > 354 ret = asoc_simple_card_parse_clk_codec(codec, dai_link, codec_dai); 355 if (ret < 0) 356 goto dai_link_of_err; 357 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

Re: [alsa-devel] [PATCH 10/11] soc: ti: knav_qmss_queue: use of_property_read_bool
by Julia Lawall 05 Aug '16

05 Aug '16

On Fri, 5 Aug 2016, Robin Murphy wrote: > Hi Julia, > > On 05/08/16 09:56, Julia Lawall wrote: > > Use of_property_read_bool to check for the existence of a property. > > This caught my eye since Rob told me off for doing the same recently[1]. > > > The semantic patch that makes this change is as follows: > > (http://coccinelle.lip6.fr/) > > > > // <smpl> > > @@ > > expression e1,e2; > > statement S2,S1; > > @@ > > - if (of_get_property(e1,e2,NULL)) > > + if (of_property_read_bool(e1,e2)) > > S1 else S2 > > // </smpl> > > > > Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> > > > > --- > > drivers/soc/ti/knav_qmss_queue.c | 2 +- > > 1 file changed, 1 insertion(+), 1 deletion(-) > > > > diff --git a/drivers/soc/ti/knav_qmss_queue.c b/drivers/soc/ti/knav_qmss_queue.c > > index b73e353..56b5d7c 100644 > > --- a/drivers/soc/ti/knav_qmss_queue.c > > +++ b/drivers/soc/ti/knav_qmss_queue.c > > @@ -1240,7 +1240,7 @@ static int knav_setup_queue_range(struct knav_device *kdev, > > if (of_get_property(node, "qalloc-by-id", NULL)) > > According to the binding, "qalloc-by-id" _is_ a boolean property, so > this one really does deserve to be of_property_read_bool()... > > > range->flags |= RANGE_RESERVED; > > > > - if (of_get_property(node, "accumulator", NULL)) { > > + if (of_property_read_bool(node, "accumulator")) { > > ...whereas "accumulator" must have a value, so this isn't technically > appropriate. In general, most of these "if the property exists, read the > property and do stuff" checks are probably a sign of code that could be > simplified by refactoring the "do stuff" step to just specifically > handle the "read the property" step returning -EINVAL when it's not present. Thanks for the very helpful feedback. I will rethink the patch set in light of this information. julia > Robin. > > [1]:https://www.mail-archive.com/iommu@lists.linux-foundation.org/msg13375.h… > > > ret = knav_init_acc_range(kdev, node, range); > > if (ret < 0) { > > devm_kfree(dev, range); > > > > > > _______________________________________________ > > linux-arm-kernel mailing list > > linux-arm-kernel(a)lists.infradead.org > > http://lists.infradead.org/mailman/listinfo/linux-arm-kernel > > > > -- > To unsubscribe from this list: send the line "unsubscribe kernel-janitors" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html >

1 0

[alsa-devel] [PATCH 00/11] use of_property_read_bool
by Julia Lawall 05 Aug '16

05 Aug '16

Use of_property_read_bool to check for the existence of a property. The semantic patch that makes this change is as follows: (http://coccinelle.lip6.fr/) // <smpl> @@ expression e1,e2; statement S2,S1; @@ - if (of_get_property(e1,e2,NULL)) + if (of_property_read_bool(e1,e2)) S1 else S2 // </smpl> --- arch/powerpc/sysdev/mpic.c | 2 +- drivers/i2c/busses/i2c-mpc.c | 2 +- drivers/mmc/host/sdhci-of-esdhc.c | 2 +- drivers/net/ethernet/freescale/xgmac_mdio.c | 3 +-- drivers/phy/phy-qcom-ufs.c | 2 +- drivers/pinctrl/nomadik/pinctrl-nomadik.c | 2 +- drivers/soc/fsl/qe/qe_tdm.c | 2 +- drivers/soc/ti/knav_qmss_queue.c | 2 +- drivers/tty/serial/atmel_serial.c | 8 ++++---- drivers/usb/host/fsl-mph-dr-of.c | 6 +++--- sound/soc/codecs/ab8500-codec.c | 10 +++++----- sound/soc/sh/rcar/ssi.c | 2 +- sound/soc/soc-core.c | 2 +- 13 files changed, 22 insertions(+), 23 deletions(-)

2 2

[alsa-devel] [PATCH 0/2 v2] DesignWare HDMI I2S suport
by Kuninori Morimoto 05 Aug '16

05 Aug '16

Hi Thierry, Russell, Mark These are v2 of DesignWare HDMI I2S support patches. It will use ALSA SoC hdmi-codec driver, but we can't use it as-is at this point. 1) patch tidyup hdmi-codec driver to enable dw-hdmi I2S support. 2) patch is based on it. Difficult is that these 2 patches are under different maintainer. Kuninori Morimoto (2): 1) ASoC: hdmi-codec: enable multi probe for same device 2) drm: bridge: add DesignWare HDMI I2S audio support drivers/gpu/drm/bridge/Kconfig | 8 ++++++++ drivers/gpu/drm/bridge/Makefile | 1 + drivers/gpu/drm/bridge/dw-hdmi-audio.h | 7 +++++++ drivers/gpu/drm/bridge/dw-hdmi-i2s-audio.c | 130 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ drivers/gpu/drm/bridge/dw-hdmi.c | 22 ++++++++++++++++++-- drivers/gpu/drm/bridge/dw-hdmi.h | 21 +++++++++++++++++++ sound/soc/codecs/hdmi-codec.c | 67 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++--- 7 files changed, 251 insertions(+), 5 deletions(-) -- 1.9.1

2 3

[alsa-devel] [PATCH v7] ASoC: rockchip: Add machine driver for RK3399 GRU Boards
by Xing Zheng 05 Aug '16

05 Aug '16

Because we need to support the multiple codecs (MAX98357A/RT5514/DA7219) on the RK3399 GRU boards, this patch can help us to support these codecs. Signed-off-by: Xing Zheng <zhengxing(a)rock-chips.com> Acked-by: Rob Herring <robh(a)kernel.org> --- Changes in v7: - add default MCLK and PLL settings for the da7219_init() so that the accessory detection works as expected Changes in v6: - remove enable/disable dapm pins, and to switch codecs via user space - fix "Speaker" dapm route for max98357a - add SND_JACK_LINEOUT flag for da7219 jack detection Changes in v5: - fix the build warning for missing I2C/CLKDEV_LOOKUP dependency Changes in v4: - the compatible name needs to be based on a single reference design, and the name of the board is called "GRU" - use one i2sX card "rockchip,cpu = <&i2s0>" instead of array that is based on GRU - rename DOC to rockchip,rk3399-gru-sound.txt - rename compatible to rockchip,rk3399-gru-sound - rename source code to rk3399_gru_sound.c Changes in v3: - rename DOC to rockchip,rk3399-max98357a-rt5514-da7219.txt - rename compatible to rockchip,rk3399-max98357a-rt5514-da7219 - rename source code to rk3399_max98357a_rt5514_da7219.c Changes in v2: - use the FS 256 to set mclks of the max98357a and rt5514 danamically - add more sample rate for da7219 .../bindings/sound/rockchip,rk3399-gru-sound.txt | 15 + sound/soc/rockchip/Kconfig | 11 + sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rk3399_gru_sound.c | 337 ++++++++++++++++++++ 4 files changed, 365 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt create mode 100644 sound/soc/rockchip/rk3399_gru_sound.c diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt b/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt new file mode 100644 index 0000000..f19b6c8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt @@ -0,0 +1,15 @@ +ROCKCHIP with MAX98357A/RT5514/DA7219 codecs on GRU boards + +Required properties: +- compatible: "rockchip,rk3399-gru-sound" +- rockchip,cpu: The phandle of the Rockchip I2S controller that's + connected to the codecs +- rockchip,codec: The phandle of the MAX98357A/RT5514/DA7219 codecs + +Example: + +sound { + compatible = "rockchip,rk3399-gru-sound"; + rockchip,cpu = <&i2s0>; + rockchip,codec = <&max98357a &rt5514 &da7219>; +}; diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig index f1e0c70..6d39032 100644 --- a/sound/soc/rockchip/Kconfig +++ b/sound/soc/rockchip/Kconfig @@ -41,3 +41,14 @@ config SND_SOC_ROCKCHIP_RT5645 help Say Y or M here if you want to add support for SoC audio on Rockchip boards using the RT5645/RT5650 codec, such as Veyron. + +config SND_SOC_RK3399_GRU_SOUND + tristate "ASoC support multiple codecs for Rockchip RK3399 GRU boards" + depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB && CLKDEV_LOOKUP + select SND_SOC_ROCKCHIP_I2S + select SND_SOC_MAX98357A + select SND_SOC_RT5514 + select SND_SOC_DA7219 + help + Say Y or M here if you want to add support multiple codecs for SoC + audio on Rockchip RK3399 GRU boards. diff --git a/sound/soc/rockchip/Makefile b/sound/soc/rockchip/Makefile index c0bf560..84e5c7c 100644 --- a/sound/soc/rockchip/Makefile +++ b/sound/soc/rockchip/Makefile @@ -7,6 +7,8 @@ obj-$(CONFIG_SND_SOC_ROCKCHIP_SPDIF) += snd-soc-rockchip-spdif.o snd-soc-rockchip-max98090-objs := rockchip_max98090.o snd-soc-rockchip-rt5645-objs := rockchip_rt5645.o +snd-soc-rk3399-gru-sound-objs := rk3399_gru_sound.o obj-$(CONFIG_SND_SOC_ROCKCHIP_MAX98090) += snd-soc-rockchip-max98090.o obj-$(CONFIG_SND_SOC_ROCKCHIP_RT5645) += snd-soc-rockchip-rt5645.o +obj-$(CONFIG_SND_SOC_RK3399_GRU_SOUND) += snd-soc-rk3399-gru-sound.o diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c new file mode 100644 index 0000000..9933703 --- /dev/null +++ b/sound/soc/rockchip/rk3399_gru_sound.c @@ -0,0 +1,337 @@ +/* + * Rockchip machine ASoC driver for boards using MAX98357A/RT5514/DA7219 + * + * Copyright (c) 2016, ROCKCHIP CORPORATION. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <linux/delay.h> +#include <linux/spi/spi.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include "rockchip_i2s.h" +#include "../codecs/da7219.h" +#include "../codecs/da7219-aad.h" +#include "../codecs/rt5514.h" + +#define DRV_NAME "rk3399-gru-sound" + +#define SOUND_FS 256 + +static struct snd_soc_jack rockchip_sound_jack; + +static const struct snd_soc_dapm_widget rockchip_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_SPK("Speakers", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), +}; + +static const struct snd_soc_dapm_route rockchip_dapm_routes[] = { + /* Input Lines */ + {"MIC", NULL, "Headset Mic"}, + {"DMIC1L", NULL, "Int Mic"}, + {"DMIC1R", NULL, "Int Mic"}, + + /* Output Lines */ + {"Headphones", NULL, "HPL"}, + {"Headphones", NULL, "HPR"}, + {"Speakers", NULL, "Speaker"}, +}; + +static const struct snd_kcontrol_new rockchip_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphones"), + SOC_DAPM_PIN_SWITCH("Speakers"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Int Mic"), +}; + +static int rockchip_sound_max98357a_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + unsigned int mclk; + int ret; + + /* max98357a supports these sample rates */ + switch (params_rate(params)) { + case 8000: + case 16000: + case 48000: + case 96000: + mclk = params_rate(params) * SOUND_FS; + break; + default: + dev_err(rtd->card->dev, "%s() doesn't support this sample rate: %d\n", + __func__, params_rate(params)); + return -EINVAL; + } + + ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, 0, mclk, 0); + if (ret) { + dev_err(rtd->card->dev, "%s() error setting sysclk to %u: %d\n", + __func__, mclk, ret); + return ret; + } + + return 0; +} + +static int rockchip_sound_rt5514_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + unsigned int mclk; + int ret; + + mclk = params_rate(params) * SOUND_FS; + + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, + SND_SOC_CLOCK_OUT); + if (ret < 0) { + dev_err(rtd->card->dev, "Can't set cpu clock out %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, RT5514_SCLK_S_MCLK, + mclk, SND_SOC_CLOCK_IN); + if (ret) { + dev_err(rtd->card->dev, "%s() error setting sysclk to %u: %d\n", + __func__, params_rate(params) * 512, ret); + return ret; + } + + return 0; +} + +static int rockchip_sound_da7219_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int mclk, ret; + + /* in bypass mode, the mclk has to be one of the frequencies below */ + switch (params_rate(params)) { + case 8000: + case 16000: + case 24000: + case 32000: + case 48000: + case 64000: + case 96000: + mclk = 12288000; + break; + case 11025: + case 22050: + case 44100: + case 88200: + mclk = 11289600; + break; + default: + return -EINVAL; + } + + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, + SND_SOC_CLOCK_OUT); + if (ret < 0) { + dev_err(codec_dai->dev, "Can't set cpu clock out %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "Can't set codec clock in %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_MCLK, 0, 0); + if (ret < 0) { + dev_err(codec_dai->dev, "Can't set pll sysclk mclk %d\n", ret); + return ret; + } + + return 0; +} + +static int rockchip_sound_da7219_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec_dais[0]->codec; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + /* We need default MCLK and PLL settings for the accessory detection */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 12288000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "Init can't set codec clock in %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_MCLK, 0, 0); + if (ret < 0) { + dev_err(codec_dai->dev, "Init can't set pll sysclk mclk %d\n", ret); + return ret; + } + + /* Enable Headset and 4 Buttons Jack detection */ + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_LINEOUT | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, + &rockchip_sound_jack, NULL, 0); + + if (ret) { + dev_err(rtd->card->dev, "New Headset Jack failed! (%d)\n", ret); + return ret; + } + + da7219_aad_jack_det(codec, &rockchip_sound_jack); + + return 0; +} + +static struct snd_soc_ops rockchip_sound_max98357a_ops = { + .hw_params = rockchip_sound_max98357a_hw_params, +}; + +static struct snd_soc_ops rockchip_sound_rt5514_ops = { + .hw_params = rockchip_sound_rt5514_hw_params, +}; + +static struct snd_soc_ops rockchip_sound_da7219_ops = { + .hw_params = rockchip_sound_da7219_hw_params, +}; + +enum { + DAILINK_MAX98357A, + DAILINK_RT5514, + DAILINK_DA7219, +}; + +static struct snd_soc_dai_link rockchip_dailinks[] = { + [DAILINK_MAX98357A] = { + .name = "MAX98357A", + .stream_name = "MAX98357A PCM", + .codec_dai_name = "HiFi", + .ops = &rockchip_sound_max98357a_ops, + /* set max98357a as slave */ + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + }, + [DAILINK_RT5514] = { + .name = "RT5514", + .stream_name = "RT5514 PCM", + .codec_dai_name = "rt5514-aif1", + .ops = &rockchip_sound_rt5514_ops, + /* set rt5514 as slave */ + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + }, + [DAILINK_DA7219] = { + .name = "DA7219", + .stream_name = "DA7219 PCM", + .codec_dai_name = "da7219-hifi", + .init = rockchip_sound_da7219_init, + .ops = &rockchip_sound_da7219_ops, + /* set da7219 as slave */ + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + }, +}; + +static struct snd_soc_card rockchip_sound_card = { + .name = "rk3399-gru-sound", + .owner = THIS_MODULE, + .dai_link = rockchip_dailinks, + .num_links = ARRAY_SIZE(rockchip_dailinks), + .dapm_widgets = rockchip_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rockchip_dapm_widgets), + .dapm_routes = rockchip_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rockchip_dapm_routes), + .controls = rockchip_controls, + .num_controls = ARRAY_SIZE(rockchip_controls), +}; + +static int rockchip_sound_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &rockchip_sound_card; + struct device_node *cpu_node; + int i, ret; + + cpu_node = of_parse_phandle(pdev->dev.of_node, "rockchip,cpu", 0); + if (!cpu_node) { + dev_err(&pdev->dev, "Property 'rockchip,cpu' missing or invalid\n"); + return -EINVAL; + } + + for (i = 0; i < card->num_links; i++) { + rockchip_dailinks[i].platform_of_node = cpu_node; + rockchip_dailinks[i].cpu_of_node = cpu_node; + + rockchip_dailinks[i].codec_of_node = + of_parse_phandle(pdev->dev.of_node, "rockchip,codec", i); + if (!rockchip_dailinks[i].codec_of_node) { + dev_err(&pdev->dev, + "Property[%d] 'rockchip,codec' missing or invalid\n", i); + return -EINVAL; + } + } + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + + ret = devm_snd_soc_register_card(&pdev->dev, card); + if (ret) + dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", + __func__, ret); + + return ret; +} + +static const struct of_device_id rockchip_sound_of_match[] = { + { .compatible = "rockchip,rk3399-gru-sound", }, + {}, +}; + +static struct platform_driver rockchip_sound_driver = { + .probe = rockchip_sound_probe, + .driver = { + .name = DRV_NAME, + .of_match_table = rockchip_sound_of_match, +#ifdef CONFIG_PM + .pm = &snd_soc_pm_ops, +#endif + }, +}; + +module_platform_driver(rockchip_sound_driver); + +MODULE_AUTHOR("Xing Zheng <zhengxing(a)rock-chips.com>"); +MODULE_DESCRIPTION("Rockchip ASoC Machine Driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, rockchip_sound_of_match); -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: dwc: Drop DMA channel names assignment
by Sylwester Nawrocki 05 Aug '16

05 Aug '16

The dw_configure_dai_by_dt() function and further dev->{play,capture} _dma_data.dt data structures seem to be used in this driver only in case of a system using devicetree, thus chan_name assignments have no effect since they will be ignored in dmaengine_pcm_request_chan_of() call and will be substituted with values taken form dmaengine_pcm_dma_channel_names[] table ("tx", "rx"). Also there is no any "TX", "RX" dma-names entries in arch/arm/boot/dts, only lower case "tx", "rx" seem to be used. Lastly, this driver doesn't set SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME flag when registering a dmaengine PCM to indicate the chan_name should be used. My intention is to eventually remove the struct snd_dmaengine_dai_dma_data chan_name field as there is also a chan_names[] field in struct snd_dmaengine_pcm_config which can be used for same purpose. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/dwc/designware_i2s.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index dc97f43..2998954 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -577,7 +577,6 @@ static int dw_configure_dai_by_dt(struct dw_i2s_dev *dev, dev->capability |= DWC_I2S_PLAY; dev->play_dma_data.dt.addr = res->start + I2S_TXDMA; dev->play_dma_data.dt.addr_width = bus_widths[idx]; - dev->play_dma_data.dt.chan_name = "TX"; dev->play_dma_data.dt.fifo_size = fifo_depth * (fifo_width[idx2]) >> 8; dev->play_dma_data.dt.maxburst = 16; @@ -588,7 +587,6 @@ static int dw_configure_dai_by_dt(struct dw_i2s_dev *dev, dev->capability |= DWC_I2S_RECORD; dev->capture_dma_data.dt.addr = res->start + I2S_RXDMA; dev->capture_dma_data.dt.addr_width = bus_widths[idx]; - dev->capture_dma_data.dt.chan_name = "RX"; dev->capture_dma_data.dt.fifo_size = fifo_depth * (fifo_width[idx2] >> 8); dev->capture_dma_data.dt.maxburst = 16; -- 1.9.1

2 1

[alsa-devel] [PATCH 0/3] ASoC: topology: Add support for BE DAIs
by mengdong.lin＠linux.intel.com 05 Aug '16

05 Aug '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> There is topology ABI udpate in this series. Topology core can check size of each ABI objects to detect version mismatch between user space and kernel. This series adds support for BE (Back End) DAIs: - Define the type and ABI struct for Backend DAIs. - The topology core can configure an existing BE DAI or create new BE DAIs. The user space series "topology: Add support for BE DAIs" is also submitted. Is it possible to meet the timeline of v4.8? There will be 2 pairs of kernel & user space series for PCM & BE links later. Mengdong Lin (3): ASoC: topology: ABI - Add the types for BE DAI ASoC: topology: Add support for configuring existing BE DAIs ASoC: topology: Able to create BE DAIs include/sound/soc-topology.h | 1 + include/uapi/sound/asoc.h | 34 +++++++++- sound/soc/soc-core.c | 3 +- sound/soc/soc-topology.c | 156 ++++++++++++++++++++++++++++++++++++++++++- 4 files changed, 189 insertions(+), 5 deletions(-) -- 2.5.0

2 5

[alsa-devel] [PATCH RESEND 7/8] ASoC: samsung: s3c24xx_uda134x: debug/error trace cleanup
by Sylwester Nawrocki 05 Aug '16

05 Aug '16

Switch from pr_* to dev_* macros and drop some debug traces. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- Rebased onto topic/samsung branch. sound/soc/samsung/s3c24xx_uda134x.c | 30 +++++++++++++++--------------- 1 file changed, 15 insertions(+), 15 deletions(-) diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c index 13e52fb..1619e76 100644 --- a/sound/soc/samsung/s3c24xx_uda134x.c +++ b/sound/soc/samsung/s3c24xx_uda134x.c @@ -62,18 +62,18 @@ static int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream) #endif mutex_lock(&clk_lock); - pr_debug("%s %d\n", __func__, clk_users); + if (clk_users == 0) { xtal = clk_get(rtd->dev, "xtal"); if (IS_ERR(xtal)) { - printk(KERN_ERR "%s cannot get xtal\n", __func__); + dev_err(rtd->dev, "%s cannot get xtal\n", __func__); ret = PTR_ERR(xtal); } else { pclk = clk_get(&s3c24xx_uda134x_snd_device->dev, "pclk"); if (IS_ERR(pclk)) { - printk(KERN_ERR "%s cannot get pclk\n", - __func__); + dev_err(rtd->dev, "%s cannot get pclk\n", + __func__); clk_put(xtal); ret = PTR_ERR(pclk); } @@ -99,8 +99,8 @@ static int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream) SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); if (ret < 0) - printk(KERN_ERR "%s cannot set constraints\n", - __func__); + dev_err(rtd->dev, "%s cannot set constraints\n", + __func__); #endif } return ret; @@ -109,7 +109,6 @@ static int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream) static void s3c24xx_uda134x_shutdown(struct snd_pcm_substream *substream) { mutex_lock(&clk_lock); - pr_debug("%s %d\n", __func__, clk_users); clk_users -= 1; if (clk_users == 0) { clk_put(xtal); @@ -156,18 +155,19 @@ static int s3c24xx_uda134x_hw_params(struct snd_pcm_substream *substream, clk_source = S3C24XX_CLKSRC_PCLK; div = bi % 33; } - pr_debug("%s desired rate %lu, %d\n", __func__, rate, bi); + + dev_dbg(rtd->dev, "%s desired rate %lu, %d\n", __func__, rate, bi); clk = (fs_mode == S3C2410_IISMOD_384FS ? 384 : 256) * rate; - pr_debug("%s will use: %s %s %d sysclk %d err %ld\n", __func__, - fs_mode == S3C2410_IISMOD_384FS ? "384FS" : "256FS", - clk_source == S3C24XX_CLKSRC_MPLL ? "MPLLin" : "PCLK", - div, clk, err); + + dev_dbg(rtd->dev, "%s will use: %s %s %d sysclk %d err %ld\n", __func__, + fs_mode == S3C2410_IISMOD_384FS ? "384FS" : "256FS", + clk_source == S3C24XX_CLKSRC_MPLL ? "MPLLin" : "PCLK", + div, clk, err); if ((err * 100 / rate) > 5) { - printk(KERN_ERR "S3C24XX_UDA134X: effective frequency " - "too different from desired (%ld%%)\n", - err * 100 / rate); + dev_err(rtd->dev, "effective frequency too different " + "from desired (%ld%%)\n", err * 100 / rate); return -EINVAL; } -- 1.9.1

2 4

[alsa-devel] [PATCH 00/12] ASoC: Intel: Skylake: Add support for library load
by Vinod Koul 05 Aug '16

05 Aug '16

This patch series adds support for library loading. A module maybe part of base firmware or in a different library. A library may have one or more modules. The names of library to be loaded are passed as topology manifest data. The library list is loaded after base FW during dsp init. So we add support library load IPC and the usage in broxton ops for loading libs. To load a list of lib(s) coming from topology, we need to split DSP init, into object creation and fw_init. The fw_init is invoked from soc-probe after topology is initialized. Now fw is initialized later, so module UUID query was move to runtime as the firmware is not available at topology initialization. While at it, also fix the parsing UUIDs only once for SKL and BXT ops and one comment code style. Lastly this series adds support for specifying processing domain for modules. Kranthi G (1): ASoC: Intel: Skylake: add support for tplg manifest load Ramesh Babu (2): ASoC: Intel: Skylake: Add library loading IPCs ASoC: Intel: Skylake: Add library loading support Senthilnathan Veppur (2): ASoC: Intel: Skylake: add additional args to module parsing ASoC: Intel: Skylake: Add module processing domain support Vinod Koul (7): ASoC: Intel: Skylake: Check list empty while getting module info ASoC: Intel: Skylake: Move modules query to runtime ASoC: Intel: Skylake: modify skl_get_dsp_ops() ASoC: Intel: Skylake: split fw and dsp initialization ASoC: Intel: Skylake: Parse UUIDs once ASoC: Intel: Bxt: Parse UUIDs once ASoC: Intel: Skylake: Fix a comment style sound/soc/intel/skylake/bxt-sst.c | 114 ++++++++++++++++++++++++--- sound/soc/intel/skylake/skl-messages.c | 39 +++++---- sound/soc/intel/skylake/skl-pcm.c | 20 +++++ sound/soc/intel/skylake/skl-sst-dsp.h | 7 +- sound/soc/intel/skylake/skl-sst-ipc.c | 29 +++++++ sound/soc/intel/skylake/skl-sst-ipc.h | 12 ++- sound/soc/intel/skylake/skl-sst-utils.c | 14 +++- sound/soc/intel/skylake/skl-sst.c | 43 ++++++---- sound/soc/intel/skylake/skl-topology.c | 55 +++++++++++-- sound/soc/intel/skylake/skl-topology.h | 2 +- sound/soc/intel/skylake/skl-tplg-interface.h | 15 +++- sound/soc/intel/skylake/skl.h | 2 + 12 files changed, 297 insertions(+), 55 deletions(-) -- 1.9.1

2 24

[alsa-devel] [PATCH 0/3] DesignWare HDMI I2S suport
by Kuninori Morimoto 05 Aug '16

05 Aug '16

Hi Mark, Thierry, Russell These are DesignWare HDMI I2S support patches. It will use ALSA SoC hdmi-codec driver, but we can't use it as-is. So, 2), 3) patches modify hdmi-codec style. Kuninori Morimoto (3): 1) drm: bridge: add DesignWare HDMI I2S audio support 2) ASoC: hdmi-codec: callback function will be called with private data 3) ASoC: hdmi-codec: enable multi probe for same device drivers/gpu/drm/bridge/Kconfig | 8 ++ drivers/gpu/drm/bridge/Makefile | 1 + drivers/gpu/drm/bridge/dw-hdmi-audio.h | 7 ++ drivers/gpu/drm/bridge/dw-hdmi-i2s-audio.c | 123 +++++++++++++++++++++++++++++ drivers/gpu/drm/bridge/dw-hdmi.c | 22 +++++- drivers/gpu/drm/bridge/dw-hdmi.h | 21 +++++ include/sound/hdmi-codec.h | 13 +-- sound/soc/codecs/hdmi-codec.c | 81 ++++++++++++++++--- 8 files changed, 259 insertions(+), 17 deletions(-) create mode 100644 drivers/gpu/drm/bridge/dw-hdmi-i2s-audio.c -- 1.9.1

5 16

Re: [alsa-devel] [PATCH] drm/i915/dp: DP audio API changes for MST
by Ville Syrjälä 05 Aug '16

05 Aug '16

On Thu, Aug 04, 2016 at 10:18:52AM -0700, Jim Bride wrote: > On Wed, Aug 03, 2016 at 10:08:12PM +0300, Ville Syrjälä wrote: > > On Tue, Aug 02, 2016 at 07:14:30PM -0700, Dhinakaran Pandiyan wrote: > > > DP MST provides the capability to send multiple video and audio streams via > > > one single port. This requires the API's between i915 and audio drivers to > > > distinguish between audio capable displays connected to a port. This patch > > > adds this support via an additional parameter 'int dev_id'. The existing > > > parameter 'port' does not change it's meaning. > > > > > > dev_id = > > > MST : pipe that the stream originates from > > > Non-MST : -1 > > > > > > Affected APIs: > > > struct i915_audio_component_ops > > > - int (*sync_audio_rate)(struct device *, int port, int rate); > > > + int (*sync_audio_rate)(struct device *, int port, int dev_id, > > > > Does the term 'dev_id' have some special meaning on the audio side? On > > the i915 side things would be less confusing if we just called it > > 'pipe'. > > Yeah, it does. All of the documentation on the audio side is written > in terms of device ID, so they asked for that nomenclature. And is the device ID always the same as the pipe? Until now we've made due with passing the port instead of the pipe, so either the audio side didn't use the device ID, or its meaning changes based on how we drive things, or they dug it out from somewhere else based on the port? > > Jim > > > > > + int rate); > > > > > > - int (*get_eld)(struct device *, int port, bool *enabled, > > > - unsigned char *buf, int max_bytes); > > > + int (*get_eld)(struct device *, int port, int dev_id, > > > + bool *enabled, unsigned char *buf, int max_bytes); > > > > > > struct i915_audio_component_audio_ops > > > - void (*pin_eld_notify)(void *audio_ptr, int port); > > > + void (*pin_eld_notify)(void *audio_ptr, int port, int dev_id); > > > > > > This patch makes dummy changes in the audio drivers for build to succeed. > > > > > > Signed-off-by: Dhinakaran Pandiyan <dhinakaran.pandiyan(a)intel.com> > > > --- > > > drivers/gpu/drm/i915/i915_drv.h | 2 +- > > > drivers/gpu/drm/i915/intel_audio.c | 82 +++++++++++++++++++++++++++++--------- > > > include/drm/i915_component.h | 6 +-- > > > include/sound/hda_i915.h | 11 ++--- > > > sound/hda/hdac_i915.c | 9 +++-- > > > sound/pci/hda/patch_hdmi.c | 7 ++-- > > > 6 files changed, 82 insertions(+), 35 deletions(-) > > > > > > diff --git a/drivers/gpu/drm/i915/i915_drv.h b/drivers/gpu/drm/i915/i915_drv.h > > > index 65ada5d..8e4c8c8 100644 > > > --- a/drivers/gpu/drm/i915/i915_drv.h > > > +++ b/drivers/gpu/drm/i915/i915_drv.h > > > @@ -2054,7 +2054,7 @@ struct drm_i915_private { > > > /* perform PHY state sanity checks? */ > > > bool chv_phy_assert[2]; > > > > > > - struct intel_encoder *dig_port_map[I915_MAX_PORTS]; > > > + struct intel_encoder *av_enc_map[I915_MAX_PIPES]; > > > > > > /* > > > * NOTE: This is the dri1/ums dungeon, don't add stuff here. Your patch > > > diff --git a/drivers/gpu/drm/i915/intel_audio.c b/drivers/gpu/drm/i915/intel_audio.c > > > index 8c493de..cbe44c8 100644 > > > --- a/drivers/gpu/drm/i915/intel_audio.c > > > +++ b/drivers/gpu/drm/i915/intel_audio.c > > > @@ -500,6 +500,9 @@ void intel_audio_codec_enable(struct intel_encoder *intel_encoder) > > > struct i915_audio_component *acomp = dev_priv->audio_component; > > > struct intel_digital_port *intel_dig_port = enc_to_dig_port(encoder); > > > enum port port = intel_dig_port->port; > > > + enum pipe pipe = crtc->pipe; > > > + int dev_id = -1; > > > + > > > > > > connector = drm_select_eld(encoder); > > > if (!connector) > > > @@ -522,14 +525,19 @@ void intel_audio_codec_enable(struct intel_encoder *intel_encoder) > > > dev_priv->display.audio_codec_enable(connector, intel_encoder, > > > adjusted_mode); > > > > > > + if (intel_encoder->type == INTEL_OUTPUT_DP_MST) > > > + dev_id = pipe; > > > + > > > mutex_lock(&dev_priv->av_mutex); > > > intel_encoder->audio_connector = connector; > > > + > > > /* referred in audio callbacks */ > > > - dev_priv->dig_port_map[port] = intel_encoder; > > > + dev_priv->av_enc_map[pipe] = intel_encoder; > > > mutex_unlock(&dev_priv->av_mutex); > > > > > > if (acomp && acomp->audio_ops && acomp->audio_ops->pin_eld_notify) > > > - acomp->audio_ops->pin_eld_notify(acomp->audio_ops->audio_ptr, (int) port); > > > + acomp->audio_ops->pin_eld_notify(acomp->audio_ops->audio_ptr, > > > + (int) port, dev_id); > > > } > > > > > > /** > > > @@ -542,22 +550,29 @@ void intel_audio_codec_enable(struct intel_encoder *intel_encoder) > > > void intel_audio_codec_disable(struct intel_encoder *intel_encoder) > > > { > > > struct drm_encoder *encoder = &intel_encoder->base; > > > + struct intel_crtc *crtc = to_intel_crtc(encoder->crtc); > > > struct drm_device *dev = encoder->dev; > > > struct drm_i915_private *dev_priv = to_i915(dev); > > > struct i915_audio_component *acomp = dev_priv->audio_component; > > > struct intel_digital_port *intel_dig_port = enc_to_dig_port(encoder); > > > enum port port = intel_dig_port->port; > > > + enum pipe pipe = crtc->pipe; > > > + int dev_id = -1; > > > + > > > + if (intel_encoder->type == INTEL_OUTPUT_DP_MST) > > > + dev_id = pipe; > > > > > > if (dev_priv->display.audio_codec_disable) > > > dev_priv->display.audio_codec_disable(intel_encoder); > > > > > > mutex_lock(&dev_priv->av_mutex); > > > intel_encoder->audio_connector = NULL; > > > - dev_priv->dig_port_map[port] = NULL; > > > + dev_priv->av_enc_map[pipe] = NULL; > > > mutex_unlock(&dev_priv->av_mutex); > > > > > > if (acomp && acomp->audio_ops && acomp->audio_ops->pin_eld_notify) > > > - acomp->audio_ops->pin_eld_notify(acomp->audio_ops->audio_ptr, (int) port); > > > + acomp->audio_ops->pin_eld_notify(acomp->audio_ops->audio_ptr, > > > + (int) port, dev_id); > > > } > > > > > > /** > > > @@ -628,8 +643,32 @@ static int i915_audio_component_get_cdclk_freq(struct device *dev) > > > return dev_priv->cdclk_freq; > > > } > > > > > > -static int i915_audio_component_sync_audio_rate(struct device *dev, > > > - int port, int rate) > > > +static struct intel_encoder *get_saved_encoder(struct intel_encoder *av_enc_map[], > > > + int port, int dev_id) > > > +{ > > > + > > > + enum pipe pipe; > > > + struct drm_encoder *encoder; > > > + > > > + if (dev_id >= 0 && dev_id < I915_MAX_PIPES) > > > + return av_enc_map[dev_id]; > > > + > > > + for (pipe = PIPE_A; pipe < I915_MAX_PIPES; pipe++) { > > > + > > > + if (!av_enc_map[pipe]) > > > + continue; > > > + > > > + encoder = &av_enc_map[pipe]->base; > > > + if (port == enc_to_dig_port(encoder)->port) > > > + return av_enc_map[pipe]; > > > + } > > > + > > > + return NULL; > > > + > > > +} > > > + > > > +static int i915_audio_component_sync_audio_rate(struct device *dev, int port, > > > + int dev_id, int rate) > > > { > > > struct drm_i915_private *dev_priv = dev_to_i915(dev); > > > struct intel_encoder *intel_encoder; > > > @@ -649,15 +688,16 @@ static int i915_audio_component_sync_audio_rate(struct device *dev, > > > return 0; > > > > > > mutex_lock(&dev_priv->av_mutex); > > > + > > > /* 1. get the pipe */ > > > - intel_encoder = dev_priv->dig_port_map[port]; > > > - /* intel_encoder might be NULL for DP MST */ > > > + intel_encoder = get_saved_encoder(dev_priv->av_enc_map, port, dev_id); > > > if (!intel_encoder || !intel_encoder->base.crtc || > > > intel_encoder->type != INTEL_OUTPUT_HDMI) { > > > DRM_DEBUG_KMS("no valid port %c\n", port_name(port)); > > > err = -ENODEV; > > > goto unlock; > > > } > > > + > > > crtc = to_intel_crtc(intel_encoder->base.crtc); > > > pipe = crtc->pipe; > > > if (pipe == INVALID_PIPE) { > > > @@ -702,7 +742,7 @@ static int i915_audio_component_sync_audio_rate(struct device *dev, > > > } > > > > > > static int i915_audio_component_get_eld(struct device *dev, int port, > > > - bool *enabled, > > > + int dev_id, bool *enabled, > > > unsigned char *buf, int max_bytes) > > > { > > > struct drm_i915_private *dev_priv = dev_to_i915(dev); > > > @@ -710,17 +750,21 @@ static int i915_audio_component_get_eld(struct device *dev, int port, > > > const u8 *eld; > > > int ret = -EINVAL; > > > > > > + > > > mutex_lock(&dev_priv->av_mutex); > > > - intel_encoder = dev_priv->dig_port_map[port]; > > > - /* intel_encoder might be NULL for DP MST */ > > > - if (intel_encoder) { > > > - ret = 0; > > > - *enabled = intel_encoder->audio_connector != NULL; > > > - if (*enabled) { > > > - eld = intel_encoder->audio_connector->eld; > > > - ret = drm_eld_size(eld); > > > - memcpy(buf, eld, min(max_bytes, ret)); > > > - } > > > + > > > + intel_encoder = get_saved_encoder(dev_priv->av_enc_map, port, dev_id); > > > + if (!intel_encoder) { > > > + mutex_unlock(&dev_priv->av_mutex); > > > + return ret; > > > + } > > > + > > > + ret = 0; > > > + *enabled = intel_encoder->audio_connector != NULL; > > > + if (*enabled) { > > > + eld = intel_encoder->audio_connector->eld; > > > + ret = drm_eld_size(eld); > > > + memcpy(buf, eld, min(max_bytes, ret)); > > > } > > > > > > mutex_unlock(&dev_priv->av_mutex); > > > diff --git a/include/drm/i915_component.h b/include/drm/i915_component.h > > > index b46fa0e..1419c98 100644 > > > --- a/include/drm/i915_component.h > > > +++ b/include/drm/i915_component.h > > > @@ -64,7 +64,7 @@ struct i915_audio_component_ops { > > > * Called from audio driver. After audio driver sets the > > > * sample rate, it will call this function to set n/cts > > > */ > > > - int (*sync_audio_rate)(struct device *, int port, int rate); > > > + int (*sync_audio_rate)(struct device *, int port, int dev_id, int rate); > > > /** > > > * @get_eld: fill the audio state and ELD bytes for the given port > > > * > > > @@ -77,7 +77,7 @@ struct i915_audio_component_ops { > > > * Note that the returned size may be over @max_bytes. Then it > > > * implies that only a part of ELD has been copied to the buffer. > > > */ > > > - int (*get_eld)(struct device *, int port, bool *enabled, > > > + int (*get_eld)(struct device *, int port, int dev_id, bool *enabled, > > > unsigned char *buf, int max_bytes); > > > }; > > > > > > @@ -97,7 +97,7 @@ struct i915_audio_component_audio_ops { > > > * status accordingly (even when the HDA controller is in power save > > > * mode). > > > */ > > > - void (*pin_eld_notify)(void *audio_ptr, int port); > > > + void (*pin_eld_notify)(void *audio_ptr, int port, int dev_id); > > > }; > > > > > > /** > > > diff --git a/include/sound/hda_i915.h b/include/sound/hda_i915.h > > > index 796cabf..5ab972e 100644 > > > --- a/include/sound/hda_i915.h > > > +++ b/include/sound/hda_i915.h > > > @@ -10,8 +10,9 @@ > > > int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable); > > > int snd_hdac_display_power(struct hdac_bus *bus, bool enable); > > > void snd_hdac_i915_set_bclk(struct hdac_bus *bus); > > > -int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, int rate); > > > -int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, > > > +int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, > > > + int dev_id, int rate); > > > +int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, int dev_id, > > > bool *audio_enabled, char *buffer, int max_bytes); > > > int snd_hdac_i915_init(struct hdac_bus *bus); > > > int snd_hdac_i915_exit(struct hdac_bus *bus); > > > @@ -29,13 +30,13 @@ static inline void snd_hdac_i915_set_bclk(struct hdac_bus *bus) > > > { > > > } > > > static inline int snd_hdac_sync_audio_rate(struct hdac_device *codec, > > > - hda_nid_t nid, int rate) > > > + hda_nid_t nid, int dev_id, int rate) > > > { > > > return 0; > > > } > > > static inline int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, > > > - bool *audio_enabled, char *buffer, > > > - int max_bytes) > > > + int dev_id, bool *audio_enabled, > > > + char *buffer, int max_bytes) > > > { > > > return -ENODEV; > > > } > > > diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c > > > index c9af022..6ea63ac 100644 > > > --- a/sound/hda/hdac_i915.c > > > +++ b/sound/hda/hdac_i915.c > > > @@ -201,7 +201,8 @@ static int pin2port(struct hdac_device *codec, hda_nid_t pin_nid) > > > * This function sets N/CTS value based on the given sample rate. > > > * Returns zero for success, or a negative error code. > > > */ > > > -int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, int rate) > > > +int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, > > > + int dev_id, int rate) > > > { > > > struct hdac_bus *bus = codec->bus; > > > struct i915_audio_component *acomp = bus->audio_component; > > > @@ -212,7 +213,7 @@ int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, int rate) > > > port = pin2port(codec, nid); > > > if (port < 0) > > > return -EINVAL; > > > - return acomp->ops->sync_audio_rate(acomp->dev, port, rate); > > > + return acomp->ops->sync_audio_rate(acomp->dev, port, dev_id, rate); > > > } > > > EXPORT_SYMBOL_GPL(snd_hdac_sync_audio_rate); > > > > > > @@ -236,7 +237,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_sync_audio_rate); > > > * thus it may be over @max_bytes. If it's over @max_bytes, it implies > > > * that only a part of ELD bytes have been fetched. > > > */ > > > -int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, > > > +int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, int dev_id, > > > bool *audio_enabled, char *buffer, int max_bytes) > > > { > > > struct hdac_bus *bus = codec->bus; > > > @@ -249,7 +250,7 @@ int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, > > > port = pin2port(codec, nid); > > > if (port < 0) > > > return -EINVAL; > > > - return acomp->ops->get_eld(acomp->dev, port, audio_enabled, > > > + return acomp->ops->get_eld(acomp->dev, port, dev_id, audio_enabled, > > > buffer, max_bytes); > > > } > > > EXPORT_SYMBOL_GPL(snd_hdac_acomp_get_eld); > > > diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c > > > index d0d5ad8..077d48a 100644 > > > --- a/sound/pci/hda/patch_hdmi.c > > > +++ b/sound/pci/hda/patch_hdmi.c > > > @@ -1485,7 +1485,7 @@ static void sync_eld_via_acomp(struct hda_codec *codec, > > > > > > mutex_lock(&per_pin->lock); > > > eld->monitor_present = false; > > > - size = snd_hdac_acomp_get_eld(&codec->core, per_pin->pin_nid, > > > + size = snd_hdac_acomp_get_eld(&codec->core, per_pin->pin_nid, -1, > > > &eld->monitor_present, eld->eld_buffer, > > > ELD_MAX_SIZE); > > > if (size > 0) { > > > @@ -1739,7 +1739,8 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, > > > /* Call sync_audio_rate to set the N/CTS/M manually if necessary */ > > > /* Todo: add DP1.2 MST audio support later */ > > > if (codec_has_acomp(codec)) > > > - snd_hdac_sync_audio_rate(&codec->core, pin_nid, runtime->rate); > > > + snd_hdac_sync_audio_rate(&codec->core, pin_nid, -1, > > > + runtime->rate); > > > > > > non_pcm = check_non_pcm_per_cvt(codec, cvt_nid); > > > mutex_lock(&per_pin->lock); > > > @@ -2285,7 +2286,7 @@ static void haswell_set_power_state(struct hda_codec *codec, hda_nid_t fg, > > > snd_hda_codec_set_power_to_all(codec, fg, power_state); > > > } > > > > > > -static void intel_pin_eld_notify(void *audio_ptr, int port) > > > +static void intel_pin_eld_notify(void *audio_ptr, int port, int dev_id) > > > { > > > struct hda_codec *codec = audio_ptr; > > > int pin_nid; > > > -- > > > 2.5.0 > > > > -- > > Ville Syrjälä > > Intel OTC -- Ville Syrjälä Intel OTC

4 5

[alsa-devel] Question about struct snd_soc_dai() :: cpu_dai->codec
by Kuninori Morimoto 05 Aug '16

05 Aug '16

Hi ALSA SoC My current headache is ALSA SoC's each modules (= Card/Codec/CPU/Platform) doesn't care about "unbind/rmmod". For example, if someone unbinded/rmmoded "Codec", Card or other modules doesn't know about it. Thus, user can continue to use this sound card, and kernel will be Oops. So, I would like to solve this issue, and for this purpose, now I'm investigating about ALSA SoC structures. It is now super complex, and I noticed that it is having duplicated struct members. If possible, I would like to cleanup this issue. During this investigating, I noticed 1 strange operation ${LINUX}/sound/soc/soc-core.c :: snd_soc_runtime_set_dai_fmt int snd_soc_runtime_set_dai_fmt(xxx) { ... /* Flip the polarity for the "CPU" end of a CODEC<->CODEC link */ if (cpu_dai->codec) { ... } ... } struct snd_soc_dai { ... /* parent platform/codec */ struct snd_soc_codec *codec; struct snd_soc_component *component; ... } According to snd_soc_dai, *codec is its "parent". I wonder does "cpu_dai" really can have "codec" parent ?? I think this is not correct, and we can remove this operation ?

6 38

[alsa-devel] DP audio API changes for identifying displays connected to a port
by Dhinakaran Pandiyan 05 Aug '16

05 Aug '16

An additional parameter has been added to the API's between i915 and audio drivers to identify display devices that can be connected to the same port. Currently only the port identity is being shared, but this is not sufficient as multiple displays can be driven from the same port (E.g. DP MST). This patch makes the API changes with dummy changes in the audio driver (Thanks Libin). Implementation to fully enable DP MST audio will follow later. Dhinakaran Pandiyan(1): drm/i915/dp: DP audio API changes for MST

5 15

[alsa-devel] [PATCH 1/2] ASoC: nau8825: fix bug in playback when suspend
by John Hsu 05 Aug '16

05 Aug '16

In chromium, the following steps will make codec function fail. \1. plug in headphones, Play music \2. run "powerd_dbus_suspend" \3. resume from S3 After resume, the jack detection will restart and make configuration for the headset. Meanwhile, the playback prepares and starts to work. The two sequences will conflict and make wrong register configuration. Originally, the driver adds protection for the case when it finds the playback is active. But the "powerd_dbus_suspend" command will close the pcm stream before suspend. Therefore, the driver can't detect the playback after resume, and the protection not works. For the issue, the driver raises protection every time after resume. The protection will release after jack detection and configuration completes, and then the playback just will goes on. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 51 +++++++++++----------------------------------- 1 file changed, 12 insertions(+), 39 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 5c9707a..20d2f15 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -212,31 +212,6 @@ static const unsigned short logtable[256] = { 0xfa2f, 0xfaea, 0xfba5, 0xfc60, 0xfd1a, 0xfdd4, 0xfe8e, 0xff47 }; -static struct snd_soc_dai *nau8825_get_codec_dai(struct nau8825 *nau8825) -{ - struct snd_soc_codec *codec = snd_soc_dapm_to_codec(nau8825->dapm); - struct snd_soc_component *component = &codec->component; - struct snd_soc_dai *codec_dai, *_dai; - - list_for_each_entry_safe(codec_dai, _dai, &component->dai_list, list) { - if (!strncmp(codec_dai->name, NUVOTON_CODEC_DAI, - strlen(NUVOTON_CODEC_DAI))) - return codec_dai; - } - return NULL; -} - -static bool nau8825_dai_is_active(struct nau8825 *nau8825) -{ - struct snd_soc_dai *codec_dai = nau8825_get_codec_dai(nau8825); - - if (codec_dai) { - if (codec_dai->playback_active || codec_dai->capture_active) - return true; - } - return false; -} - /** * nau8825_sema_acquire - acquire the semaphore of nau88l25 * @nau8825: component to register the codec private data with @@ -1205,6 +1180,8 @@ static int nau8825_hw_params(struct snd_pcm_substream *substream, struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); unsigned int val_len = 0; + nau8825_sema_acquire(nau8825, 2 * HZ); + switch (params_width(params)) { case 16: val_len |= NAU8825_I2S_DL_16; @@ -1225,6 +1202,9 @@ static int nau8825_hw_params(struct snd_pcm_substream *substream, regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL1, NAU8825_I2S_DL_MASK, val_len); + /* Release the semaphone. */ + nau8825_sema_release(nau8825); + return 0; } @@ -1234,6 +1214,8 @@ static int nau8825_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); unsigned int ctrl1_val = 0, ctrl2_val = 0; + nau8825_sema_acquire(nau8825, 2 * HZ); + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: ctrl2_val |= NAU8825_I2S_MS_MASTER; @@ -1282,6 +1264,9 @@ static int nau8825_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, NAU8825_I2S_MS_MASK, ctrl2_val); + /* Release the semaphone. */ + nau8825_sema_release(nau8825); + return 0; } @@ -2241,20 +2226,8 @@ static int __maybe_unused nau8825_resume(struct snd_soc_codec *codec) regcache_cache_only(nau8825->regmap, false); regcache_sync(nau8825->regmap); - if (nau8825_is_jack_inserted(nau8825->regmap)) { - /* If the jack is inserted, we need to check whether the play- - * back is active before suspend. If active, the driver has to - * raise the protection for cross talk function to avoid the - * playback recovers before cross talk process finish. Other- - * wise, the playback will be interfered by cross talk func- - * tion. It is better to apply hardware related parameters - * before starting playback or record. - */ - if (nau8825_dai_is_active(nau8825)) { - nau8825->xtalk_protect = true; - nau8825_sema_acquire(nau8825, 0); - } - } + nau8825->xtalk_protect = true; + nau8825_sema_acquire(nau8825, 0); enable_irq(nau8825->irq); return 0; -- 2.6.4

2 3

[alsa-devel] [PATCH] ASoC: samsung: Fix clock handling in S3C24XX_UDA134X card
by Sylwester Nawrocki 04 Aug '16

04 Aug '16

There is no "pclk" alias in the s3c2440 clk driver for "soc-audio" device so related clk_get() fails, which prevents any operation of the S3C24XX_UDA134X sound card. Instead we get the clock on behalf of the I2S device, i.e. we use the I2S block gate clock which has PCLK is its parent clock. Without this patch there is an error like: s3c24xx_uda134x_startup cannot get pclk ASoC: UDA134X startup failed: -2 Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/samsung/s3c24xx_uda134x.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c index 50849e1..92e88bc 100644 --- a/sound/soc/samsung/s3c24xx_uda134x.c +++ b/sound/soc/samsung/s3c24xx_uda134x.c @@ -58,10 +58,12 @@ static struct platform_device *s3c24xx_uda134x_snd_device; static int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream) { - int ret = 0; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; #ifdef ENFORCE_RATES struct snd_pcm_runtime *runtime = substream->runtime; #endif + int ret = 0; mutex_lock(&clk_lock); pr_debug("%s %d\n", __func__, clk_users); @@ -71,8 +73,7 @@ static int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream) printk(KERN_ERR "%s cannot get xtal\n", __func__); ret = PTR_ERR(xtal); } else { - pclk = clk_get(&s3c24xx_uda134x_snd_device->dev, - "pclk"); + pclk = clk_get(cpu_dai->dev, "iis"); if (IS_ERR(pclk)) { printk(KERN_ERR "%s cannot get pclk\n", __func__); -- 1.9.1

1 0

[alsa-devel] [PATCH 0/7] ASoC: samsung: DMA parameters handling cleanup
by Sylwester Nawrocki 04 Aug '16

04 Aug '16

The motivation for this series is to remove the s3c_dma_params data structure and use struct snd_dmaengine_dai_dma_data instead, which removes some kind of indirection and makes the code easier to follow. Sylwester Nawrocki (7): ASoC: samsung: Drop usage of struct s3c_dma_params from i2s.c ASoC: samsung: Drop usage of struct s3c_dma_params from s3c24xx-i2s.c ASoC: samsung: Drop usage of struct s3c_dma_params from s3c2412-i2s.c ASoC: samsung: Drop usage of struct s3c_dma_params from ac97.c ASoC: samsung: Drop usage of struct s3c_dma_params from spdif.c ASoC: samsung: Drop usage of of struct s3c_dma_params from pcm.c ASoC: samsung: Remove unused now unused struct s3c_dma_params sound/soc/samsung/ac97.c | 28 +++++++++++----------- sound/soc/samsung/dma.h | 20 +++------------- sound/soc/samsung/dmaengine.c | 31 ------------------------ sound/soc/samsung/i2s.c | 48 +++++++++++++++++++------------------- sound/soc/samsung/idma.c | 1 - sound/soc/samsung/pcm.c | 26 ++++++++++----------- sound/soc/samsung/s3c-i2s-v2.c | 3 +-- sound/soc/samsung/s3c-i2s-v2.h | 4 ++-- sound/soc/samsung/s3c2412-i2s.c | 24 +++++++++---------- sound/soc/samsung/s3c24xx-i2s.c | 24 +++++++++---------- sound/soc/samsung/smdk_wm8580pcm.c | 1 - sound/soc/samsung/smdk_wm8994pcm.c | 1 - sound/soc/samsung/spdif.c | 12 +++++----- 13 files changed, 86 insertions(+), 137 deletions(-) -- 1.9.1

1 7

[alsa-devel] [PATCH] ALSA: hda - Fix headset mic detection problem for two dell machines
by Hui Wang 04 Aug '16

04 Aug '16

One of the machines has ALC255 on it, another one has ALC298 on it. On the machine with the codec ALC298, it also has the speaker volume problem, so we add the fixup chained to ALC298_FIXUP_SPK_VOLUME rather than adding a group of pin definition in the pin quirk table, since the speak volume problem does not happen on other machines yet. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ce5f1ba..574b1b4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5513,6 +5513,8 @@ static const struct hda_fixup alc269_fixups[] = { [ALC298_FIXUP_SPK_VOLUME] = { .type = HDA_FIXUP_FUNC, .v.func = alc298_fixup_speaker_volume, + .chained = true, + .chain_id = ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, }, }; @@ -5836,6 +5838,10 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x1b, 0x01014020}, {0x21, 0x0221103f}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x14, 0x90170130}, + {0x1b, 0x02011020}, + {0x21, 0x0221103f}), + SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x14, 0x90170150}, {0x1b, 0x02011020}, {0x21, 0x0221105f}), -- 1.9.1

2 1

[alsa-devel] atmel_ssc_dai, duplex audio support?
by Peter Meerwald-Stadler 03 Aug '16

03 Aug '16

Hello, it appears that atmel_ssc_dai does not support duplex audio, i.e. when first starting a playback stream and then starting a record stream, the playback stops problem seems to be in atmel_ssc_startup(): /* Reset the SSC to keep it at a clean status */ ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); which is performed unconditionally when a stream starts code was introduced with ASoC: atmel_ssc_dai: refactor the startup and shutdown, commit cbaadf0f90d6 tested with the 4.1 branch of linux-at91 git thanks, regards, p. -- Peter Meerwald-Stadler +43-664-2444418 (mobile)

1 0

[alsa-devel] [PATCH] ALSA: hda: Fix krealloc() with __GFP_ZERO usage
by Takashi Iwai 03 Aug '16

03 Aug '16

krealloc() doesn't work always properly with __GFP_ZERO flag as expected. For clearing the reallocated area, we need to clear explicitly instead. Reported-by: Joe Perches <joe(a)perches.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/array.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/hda/array.c b/sound/hda/array.c index 516795baa7db..5dfa610e4471 100644 --- a/sound/hda/array.c +++ b/sound/hda/array.c @@ -21,13 +21,15 @@ void *snd_array_new(struct snd_array *array) return NULL; if (array->used >= array->alloced) { int num = array->alloced + array->alloc_align; + int oldsize = array->alloced * array->elem_size; int size = (num + 1) * array->elem_size; void *nlist; if (snd_BUG_ON(num >= 4096)) return NULL; - nlist = krealloc(array->list, size, GFP_KERNEL | __GFP_ZERO); + nlist = krealloc(array->list, size, GFP_KERNEL); if (!nlist) return NULL; + memset(nlist + oldsize, 0, size - oldsize); array->list = nlist; array->alloced = num; } -- 2.9.2

1 0

[alsa-devel] [PATCH] ALSA: hda: add AMD Bonaire AZ PCI ID with proper driver caps
by Maruthi Srinivas Bayyavarapu 03 Aug '16

03 Aug '16

This commit fixes garbled audio on Bonaire HDMI Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu(a)amd.com> Reviewed-by: Alex Deucher <alexander.deucher(a)amd.com> --- sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 6f8ea13..89dacf9 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2265,6 +2265,8 @@ static const struct pci_device_id azx_ids[] = { { PCI_DEVICE(0x1022, 0x780d), .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB }, /* ATI HDMI */ + { PCI_DEVICE(0x1002, 0x0002), + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, { PCI_DEVICE(0x1002, 0x1308), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, { PCI_DEVICE(0x1002, 0x157a), -- 1.9.1

2 1

[alsa-devel] [PATCH] sound: Fix empty menuconfig SND_ARM
by Eugeniu Rosca 03 Aug '16

03 Aug '16

commit 3c8f7710c1c4 ("ASoC: fix broken pxa SoC support") has removed the SND_ARM dependency from SND_PXA2XX_LIB and SND_PXA2XX_LIB_AC97, by moving these config entries outside of the "if SND_ARM ... endif" construct. However, by placing these 2 symbols right between the SND_ARM menuconfig definition and the first SND_ARM menu entry, the side effect is that the SND_ARM menu becomes empty and all the config entries caught between "if SND_ARM ... endif" no more belong to menuconfig SND_ARM, but to its parent (menuconfig SND). Fix this. Signed-off-by: Eugeniu Rosca <rosca.eugeniu(a)gmail.com> --- sound/arm/Kconfig | 15 +++++++-------- 1 file changed, 7 insertions(+), 8 deletions(-) diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig index e040621..65171f6 100644 --- a/sound/arm/Kconfig +++ b/sound/arm/Kconfig @@ -9,14 +9,6 @@ menuconfig SND_ARM Drivers that are implemented on ASoC can be found in "ALSA for SoC audio support" section. -config SND_PXA2XX_LIB - tristate - select SND_AC97_CODEC if SND_PXA2XX_LIB_AC97 - select SND_DMAENGINE_PCM - -config SND_PXA2XX_LIB_AC97 - bool - if SND_ARM config SND_ARMAACI @@ -42,3 +34,10 @@ config SND_PXA2XX_AC97 endif # SND_ARM +config SND_PXA2XX_LIB + tristate + select SND_AC97_CODEC if SND_PXA2XX_LIB_AC97 + select SND_DMAENGINE_PCM + +config SND_PXA2XX_LIB_AC97 + bool -- 2.9.2

3 5

[alsa-devel] [PATCH v2] ASoC: simple-card-utils: add missing MODULE_xxx()
by Kuninori Morimoto 03 Aug '16

03 Aug '16

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> simple-card-utils might be used as module, but MODULE_xxx() information was missed. This patch adds it. Otherwise, we will have below error, and can't use it. Specil thanks to Kevin. > insmod simple-card-utils.ko simple_card_utils: module license 'unspecified' taints kernel. Disabling lock debugging due to kernel taint simple_card_utils: Unknown symbol snd_soc_of_parse_daifmt (err 0) simple_card_utils: Unknown symbol snd_soc_of_parse_card_name (err 0) insmod: can't insert 'simple-card-utils.ko': \ unknown symbol in module, or unknown parameter Reported-by: Kevin Hilman <khilman(a)baylibre.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - remove MODULE_ALIAS() sound/soc/generic/Makefile | 6 +++--- sound/soc/generic/simple-card-utils.c | 6 ++++++ 2 files changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/soc/generic/Makefile b/sound/soc/generic/Makefile index 45602ca..2d53c8d 100644 --- a/sound/soc/generic/Makefile +++ b/sound/soc/generic/Makefile @@ -1,5 +1,5 @@ -obj-$(CONFIG_SND_SIMPLE_CARD_UTILS) := simple-card-utils.o - +snd-soc-simple-card-utils-objs := simple-card-utils.o snd-soc-simple-card-objs := simple-card.o -obj-$(CONFIG_SND_SIMPLE_CARD) += snd-soc-simple-card.o +obj-$(CONFIG_SND_SIMPLE_CARD_UTILS) += snd-soc-simple-card-utils.o +obj-$(CONFIG_SND_SIMPLE_CARD) += snd-soc-simple-card.o diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index d89a9a1..9599de6 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -7,6 +7,7 @@ * 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/of.h> #include <sound/simple_card_utils.h> @@ -95,3 +96,8 @@ int asoc_simple_card_parse_card_name(struct snd_soc_card *card, return 0; } EXPORT_SYMBOL_GPL(asoc_simple_card_parse_card_name); + +/* Module information */ +MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>"); +MODULE_DESCRIPTION("ALSA SoC Simple Card Utils"); +MODULE_LICENSE("GPL v2"); -- 1.9.1

1 0

[alsa-devel] [PATCH 1/2] ASoC: cs42l73: Constify cs42l73_mclk_coeffs and cs42l73_mclkx_coeffs tables
by Axel Lin 02 Aug '16

02 Aug '16

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs42l73.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 42a8fd4..8524e22 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -794,7 +794,7 @@ struct cs42l73_mclk_div { u8 mmcc; }; -static struct cs42l73_mclk_div cs42l73_mclk_coeffs[] = { +static const struct cs42l73_mclk_div cs42l73_mclk_coeffs[] = { /* MCLK, Sample Rate, xMMCC[5:0] */ {5644800, 11025, 0x30}, {5644800, 22050, 0x20}, @@ -844,7 +844,7 @@ struct cs42l73_mclkx_div { u8 mclkdiv; }; -static struct cs42l73_mclkx_div cs42l73_mclkx_coeffs[] = { +static const struct cs42l73_mclkx_div cs42l73_mclkx_coeffs[] = { {5644800, 1, 0}, /* 5644800 */ {6000000, 1, 0}, /* 6000000 */ {6144000, 1, 0}, /* 6144000 */ -- 2.7.4

3 3

[alsa-devel] Improving status timestamp accuracy
by Alan Young 02 Aug '16

02 Aug '16

I am looking at ways to get more accurate status timestamp information for various SoC drivers. The data that is obtained by snd_pcm_status(). One route would be to implement the more accurate timestamp mechanisms that currently are only available for HDA and Skylake (which I think is the SoC version of HDA). Looking at the code however, I think that may be unnecessary, at least for my purposes. It may not actually be practical in many cases. A call to snd_pcm_status() result in snd_pcm_update_hw_ptr0() being called. This gets the current output position (pos) via substream->ops->pointer(substream) and then makes all the other calculations based on the result. In theory, the result of substream->ops->pointer() could be sample accurate but in practice it is very unlikely to be better than period accurate. In fact, if I read it right, it will just about always be accurate to the point of the last period interrupt. Even when a DMA driver claims support of DMA_RESIDUE_GRANULARITY_BURST, it is often the case that the actual granularity is a period. The consequence of all that is that, for most drivers, the accuracy of a status report is period time. The result values, tstamp & audio_tstamp, are calculated using the current time and the pos estimate from above. snd_pcm_update_hw_ptr0() is also called when there is a DMA interrupt. At that time the calculate results will be accurate, or at worst consistently inaccurate (there could be a constant offset). It would be useful if a snd_pcm_status() call would simply return the results from the point of the last interrupt, and not try to estimate a current value based on the inaccurate substream->ops->pointer() result. It could either: (a) return the result from the time of the last interrupt, in which case tstamp would be in the past and driver_tstamp would be now; or (b) update audio_tstamp based on the elapsed time since it was recorded. (b) effectively abandons the idea that a current position report will be accurate outside of an interrupt callback but, even if it is, doing so is unlikely to result in any loss of accuracy in practice (assuming a drift of better than 40ppm and period time < 100ms). Any comments on either of these approaches? I guess (b) is more compatible with the current model. Alan.

6 24

[alsa-devel] [PATCH 1170/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/firewire/fireworks/fireworks.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/firewire/fireworks/fireworks.c b/sound/firewire/fireworks/fireworks.c index 71a0613..3390b4e 100644 --- a/sound/firewire/fireworks/fireworks.c +++ b/sound/firewire/fireworks/fireworks.c @@ -27,16 +27,16 @@ static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; unsigned int snd_efw_resp_buf_size = 1024; bool snd_efw_resp_buf_debug = false; -module_param_array(index, int, NULL, 0444); +module_param_array(index, int, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(index, "card index"); -module_param_array(id, charp, NULL, 0444); +module_param_array(id, charp, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(id, "ID string"); -module_param_array(enable, bool, NULL, 0444); +module_param_array(enable, bool, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(enable, "enable Fireworks sound card"); -module_param_named(resp_buf_size, snd_efw_resp_buf_size, uint, 0444); +module_param_named(resp_buf_size, snd_efw_resp_buf_size, uint, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(resp_buf_size, "response buffer size (max 4096, default 1024)"); -module_param_named(resp_buf_debug, snd_efw_resp_buf_debug, bool, 0444); +module_param_named(resp_buf_debug, snd_efw_resp_buf_debug, bool, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(resp_buf_debug, "store all responses to buffer"); static DEFINE_MUTEX(devices_mutex); -- 2.9.2

2 1

[alsa-devel] [PATCH 1169/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/firewire/bebob/bebob.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c index f7e2cbd..ec67fc6 100644 --- a/sound/firewire/bebob/bebob.c +++ b/sound/firewire/bebob/bebob.c @@ -22,11 +22,11 @@ static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; -module_param_array(index, int, NULL, 0444); +module_param_array(index, int, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(index, "card index"); -module_param_array(id, charp, NULL, 0444); +module_param_array(id, charp, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(id, "ID string"); -module_param_array(enable, bool, NULL, 0444); +module_param_array(enable, bool, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(enable, "enable BeBoB sound card"); static DEFINE_MUTEX(devices_mutex); -- 2.9.2

2 1

[alsa-devel] [PATCH 1271/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/soc/omap/mcbsp.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 4a16e77..3938336 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -857,7 +857,7 @@ unlock: return size; } -static DEVICE_ATTR(dma_op_mode, 0644, dma_op_mode_show, dma_op_mode_store); +static DEVICE_ATTR(dma_op_mode, S_IRUSR | S_IWUSR | S_IRGRP | S_IROTH, dma_op_mode_show, dma_op_mode_store); static const struct attribute *additional_attrs[] = { &dev_attr_max_tx_thres.attr, @@ -926,7 +926,7 @@ out: return size; } -static DEVICE_ATTR(st_taps, 0644, st_taps_show, st_taps_store); +static DEVICE_ATTR(st_taps, S_IRUSR | S_IWUSR | S_IRGRP | S_IROTH, st_taps_show, st_taps_store); static const struct attribute *sidetone_attrs[] = { &dev_attr_st_taps.attr, -- 2.9.2

2 1

[alsa-devel] [PATCH 1285/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/usb/usx2y/usbusx2yaudio.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index dd40ca9..a98ddcf 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -61,7 +61,7 @@ #ifdef USX2Y_NRPACKS_VARIABLE static int nrpacks = USX2Y_NRPACKS; /* number of packets per urb */ #define nr_of_packs() nrpacks - module_param(nrpacks, int, 0444); + module_param(nrpacks, int, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(nrpacks, "Number of packets per URB."); #else #define nr_of_packs() USX2Y_NRPACKS -- 2.9.2

1 0

[alsa-devel] [PATCH 1284/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/usb/usx2y/usbusx2y.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c index 91e0e2a..86f0d0b 100644 --- a/sound/usb/usx2y/usbusx2y.c +++ b/sound/usb/usx2y/usbusx2y.c @@ -156,11 +156,11 @@ static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-max */ static char* id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* Id for this card */ static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ -module_param_array(index, int, NULL, 0444); +module_param_array(index, int, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(index, "Index value for "NAME_ALLCAPS"."); -module_param_array(id, charp, NULL, 0444); +module_param_array(id, charp, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(id, "ID string for "NAME_ALLCAPS"."); -module_param_array(enable, bool, NULL, 0444); +module_param_array(enable, bool, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(enable, "Enable "NAME_ALLCAPS"."); -- 2.9.2

1 0

[alsa-devel] [PATCH 1283/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/usb/usx2y/us122l.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c index cf5dc33..5ba4254 100644 --- a/sound/usb/usx2y/us122l.c +++ b/sound/usb/usx2y/us122l.c @@ -39,11 +39,11 @@ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* Id for this card */ /* Enable this card */ static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; -module_param_array(index, int, NULL, 0444); +module_param_array(index, int, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(index, "Index value for "NAME_ALLCAPS"."); -module_param_array(id, charp, NULL, 0444); +module_param_array(id, charp, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(id, "ID string for "NAME_ALLCAPS"."); -module_param_array(enable, bool, NULL, 0444); +module_param_array(enable, bool, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(enable, "Enable "NAME_ALLCAPS"."); static int snd_us122l_card_used[SNDRV_CARDS]; -- 2.9.2

1 0

[alsa-devel] [PATCH 1282/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/usb/misc/ua101.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/usb/misc/ua101.c b/sound/usb/misc/ua101.c index c19a5dd..c48d874 100644 --- a/sound/usb/misc/ua101.c +++ b/sound/usb/misc/ua101.c @@ -55,13 +55,13 @@ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; static unsigned int queue_length = 21; -module_param_array(index, int, NULL, 0444); +module_param_array(index, int, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(index, "card index"); -module_param_array(id, charp, NULL, 0444); +module_param_array(id, charp, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(id, "ID string"); -module_param_array(enable, bool, NULL, 0444); +module_param_array(enable, bool, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(enable, "enable card"); -module_param(queue_length, uint, 0644); +module_param(queue_length, uint, S_IRUSR | S_IWUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(queue_length, "USB queue length in microframes, " __stringify(MIN_QUEUE_LENGTH)"-"__stringify(MAX_QUEUE_LENGTH)); -- 2.9.2

1 0

[alsa-devel] [PATCH 1281/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/usb/hiface/chip.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/usb/hiface/chip.c b/sound/usb/hiface/chip.c index 2670d646..d2f4d9a 100644 --- a/sound/usb/hiface/chip.c +++ b/sound/usb/hiface/chip.c @@ -50,11 +50,11 @@ static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card #define DRIVER_NAME "snd-usb-hiface" #define CARD_NAME "hiFace" -module_param_array(index, int, NULL, 0444); +module_param_array(index, int, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard."); -module_param_array(id, charp, NULL, 0444); +module_param_array(id, charp, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard."); -module_param_array(enable, bool, NULL, 0444); +module_param_array(enable, bool, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard."); static DEFINE_MUTEX(register_mutex); -- 2.9.2

1 0

[alsa-devel] [PATCH 1280/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/usb/card.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 69860da..4808173 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -84,24 +84,24 @@ static bool ignore_ctl_error; static bool autoclock = true; static char *quirk_alias[SNDRV_CARDS]; -module_param_array(index, int, NULL, 0444); +module_param_array(index, int, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(index, "Index value for the USB audio adapter."); -module_param_array(id, charp, NULL, 0444); +module_param_array(id, charp, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(id, "ID string for the USB audio adapter."); -module_param_array(enable, bool, NULL, 0444); +module_param_array(enable, bool, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(enable, "Enable USB audio adapter."); -module_param_array(vid, int, NULL, 0444); +module_param_array(vid, int, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(vid, "Vendor ID for the USB audio device."); -module_param_array(pid, int, NULL, 0444); +module_param_array(pid, int, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(pid, "Product ID for the USB audio device."); -module_param_array(device_setup, int, NULL, 0444); +module_param_array(device_setup, int, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(device_setup, "Specific device setup (if needed)."); -module_param(ignore_ctl_error, bool, 0444); +module_param(ignore_ctl_error, bool, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(ignore_ctl_error, "Ignore errors from USB controller for mixer interfaces."); -module_param(autoclock, bool, 0444); +module_param(autoclock, bool, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(autoclock, "Enable auto-clock selection for UAC2 devices (default: yes)."); -module_param_array(quirk_alias, charp, NULL, 0444); +module_param_array(quirk_alias, charp, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(quirk_alias, "Quirk aliases, e.g. 0123abcd:5678beef."); /* -- 2.9.2

1 0

[alsa-devel] [PATCH 1279/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/usb/caiaq/device.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index b871ba4..42c255d 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -58,11 +58,11 @@ static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-max */ static char* id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* Id for this card */ static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ -module_param_array(index, int, NULL, 0444); +module_param_array(index, int, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(index, "Index value for the caiaq sound device"); -module_param_array(id, charp, NULL, 0444); +module_param_array(id, charp, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(id, "ID string for the caiaq soundcard."); -module_param_array(enable, bool, NULL, 0444); +module_param_array(enable, bool, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(enable, "Enable the caiaq soundcard."); enum { -- 2.9.2

1 0

[alsa-devel] [PATCH 1278/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/usb/6fire/chip.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/usb/6fire/chip.c b/sound/usb/6fire/chip.c index dcddfc3..afce091 100644 --- a/sound/usb/6fire/chip.c +++ b/sound/usb/6fire/chip.c @@ -38,11 +38,11 @@ static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable card */ static struct sfire_chip *chips[SNDRV_CARDS] = SNDRV_DEFAULT_PTR; static struct usb_device *devices[SNDRV_CARDS] = SNDRV_DEFAULT_PTR; -module_param_array(index, int, NULL, 0444); +module_param_array(index, int, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(index, "Index value for the 6fire sound device"); -module_param_array(id, charp, NULL, 0444); +module_param_array(id, charp, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(id, "ID string for the 6fire sound device."); -module_param_array(enable, bool, NULL, 0444); +module_param_array(enable, bool, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(enable, "Enable the 6fire sound device."); static DEFINE_MUTEX(register_mutex); -- 2.9.2

1 0

[alsa-devel] [PATCH 1277/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/sparc/dbri.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index 0190cb6..4cc609e 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -82,11 +82,11 @@ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ /* Enable this card */ static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; -module_param_array(index, int, NULL, 0444); +module_param_array(index, int, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(index, "Index value for Sun DBRI soundcard."); -module_param_array(id, charp, NULL, 0444); +module_param_array(id, charp, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(id, "ID string for Sun DBRI soundcard."); -module_param_array(enable, bool, NULL, 0444); +module_param_array(enable, bool, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(enable, "Enable Sun DBRI soundcard."); #undef DBRI_DEBUG @@ -99,7 +99,7 @@ MODULE_PARM_DESC(enable, "Enable Sun DBRI soundcard."); #define D_DESC (1<<5) static int dbri_debug; -module_param(dbri_debug, int, 0644); +module_param(dbri_debug, int, S_IRUSR | S_IWUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(dbri_debug, "Debug value for Sun DBRI soundcard."); #ifdef DBRI_DEBUG -- 2.9.2

1 0

[alsa-devel] [PATCH 1276/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/sparc/cs4231.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/sparc/cs4231.c b/sound/sparc/cs4231.c index 30bdc97..3ce2d9b 100644 --- a/sound/sparc/cs4231.c +++ b/sound/sparc/cs4231.c @@ -42,11 +42,11 @@ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ /* Enable this card */ static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; -module_param_array(index, int, NULL, 0444); +module_param_array(index, int, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(index, "Index value for Sun CS4231 soundcard."); -module_param_array(id, charp, NULL, 0444); +module_param_array(id, charp, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(id, "ID string for Sun CS4231 soundcard."); -module_param_array(enable, bool, NULL, 0444); +module_param_array(enable, bool, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(enable, "Enable Sun CS4231 soundcard."); MODULE_AUTHOR("Jaroslav Kysela, Derrick J. Brashear and David S. Miller"); MODULE_DESCRIPTION("Sun CS4231"); -- 2.9.2

1 0

[alsa-devel] [PATCH 1275/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/sparc/amd7930.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/sparc/amd7930.c b/sound/sparc/amd7930.c index 35c1f6a..08edc86 100644 --- a/sound/sparc/amd7930.c +++ b/sound/sparc/amd7930.c @@ -52,11 +52,11 @@ static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ -module_param_array(index, int, NULL, 0444); +module_param_array(index, int, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(index, "Index value for Sun AMD7930 soundcard."); -module_param_array(id, charp, NULL, 0444); +module_param_array(id, charp, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(id, "ID string for Sun AMD7930 soundcard."); -module_param_array(enable, bool, NULL, 0444); +module_param_array(enable, bool, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(enable, "Enable Sun AMD7930 soundcard."); MODULE_AUTHOR("Thomas K. Dyas and David S. Miller"); MODULE_DESCRIPTION("Sun AMD7930"); -- 2.9.2

1 0

[alsa-devel] [PATCH 1274/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/sound_core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/sound_core.c b/sound/sound_core.c index 99b73c6..2c3c09c 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -154,7 +154,7 @@ static int preclaim_oss = 1; static int preclaim_oss = 0; #endif -module_param(preclaim_oss, int, 0444); +module_param(preclaim_oss, int, S_IRUSR | S_IRGRP | S_IROTH); static int soundcore_open(struct inode *, struct file *); -- 2.9.2

1 0

[alsa-devel] [PATCH 1273/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c446485..3d0ebc7 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2263,7 +2263,7 @@ static ssize_t dapm_widget_show(struct device *dev, return count; } -static DEVICE_ATTR(dapm_widget, 0444, dapm_widget_show, NULL); +static DEVICE_ATTR(dapm_widget, S_IRUSR | S_IRGRP | S_IROTH, dapm_widget_show, NULL); struct attribute *soc_dapm_dev_attrs[] = { &dev_attr_dapm_widget.attr, -- 2.9.2

1 0

[alsa-devel] [PATCH 1272/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/soc/soc-core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 16369ca..9da79cd 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -157,7 +157,7 @@ static ssize_t codec_reg_show(struct device *dev, return soc_codec_reg_show(rtd->codec, buf, PAGE_SIZE, 0); } -static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL); +static DEVICE_ATTR(codec_reg, S_IRUSR | S_IRGRP | S_IROTH, codec_reg_show, NULL); static ssize_t pmdown_time_show(struct device *dev, struct device_attribute *attr, char *buf) @@ -181,7 +181,7 @@ static ssize_t pmdown_time_set(struct device *dev, return count; } -static DEVICE_ATTR(pmdown_time, 0644, pmdown_time_show, pmdown_time_set); +static DEVICE_ATTR(pmdown_time, S_IRUSR | S_IWUSR | S_IRGRP | S_IROTH, pmdown_time_show, pmdown_time_set); static struct attribute *soc_dev_attrs[] = { &dev_attr_codec_reg.attr, -- 2.9.2

1 0

[alsa-devel] [PATCH 1270/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/soc/codecs/wm8962.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index f3109da..58ac63c 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3231,7 +3231,7 @@ static ssize_t wm8962_beep_set(struct device *dev, return count; } -static DEVICE_ATTR(beep, 0200, NULL, wm8962_beep_set); +static DEVICE_ATTR(beep, S_IWUSR, NULL, wm8962_beep_set); static void wm8962_init_beep(struct snd_soc_codec *codec) { -- 2.9.2

1 0

[alsa-devel] [PATCH 1269/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/soc/codecs/tlv320aic26.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index 2c904d7..27f9f45 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -286,7 +286,7 @@ static ssize_t aic26_keyclick_set(struct device *dev, return count; } -static DEVICE_ATTR(keyclick, 0644, aic26_keyclick_show, aic26_keyclick_set); +static DEVICE_ATTR(keyclick, S_IRUSR | S_IWUSR | S_IRGRP | S_IROTH, aic26_keyclick_show, aic26_keyclick_set); /* --------------------------------------------------------------------- * SoC CODEC portion of driver: probe and release routines -- 2.9.2

1 0

[alsa-devel] [PATCH 1268/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/soc/codecs/cs42l56.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index eec1ff8..a6a8edf 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -1050,7 +1050,7 @@ static ssize_t cs42l56_beep_set(struct device *dev, return count; } -static DEVICE_ATTR(beep, 0200, NULL, cs42l56_beep_set); +static DEVICE_ATTR(beep, S_IWUSR, NULL, cs42l56_beep_set); static void cs42l56_init_beep(struct snd_soc_codec *codec) { -- 2.9.2

1 0

[alsa-devel] [PATCH 1267/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/soc/codecs/cs42l52.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 47b97fc..4323e17 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -976,7 +976,7 @@ static ssize_t cs42l52_beep_set(struct device *dev, return count; } -static DEVICE_ATTR(beep, 0200, NULL, cs42l52_beep_set); +static DEVICE_ATTR(beep, S_IWUSR, NULL, cs42l52_beep_set); static void cs42l52_init_beep(struct snd_soc_codec *codec) { -- 2.9.2

1 0

[alsa-devel] [PATCH 1266/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/sh/sh_dac_audio.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/sh/sh_dac_audio.c b/sound/sh/sh_dac_audio.c index abf9c0c..c5f9c9a 100644 --- a/sound/sh/sh_dac_audio.c +++ b/sound/sh/sh_dac_audio.c @@ -45,9 +45,9 @@ MODULE_SUPPORTED_DEVICE("{{SuperH DAC audio support}}"); /* Module Parameters */ static int index = SNDRV_DEFAULT_IDX1; static char *id = SNDRV_DEFAULT_STR1; -module_param(index, int, 0444); +module_param(index, int, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(index, "Index value for SuperH DAC audio."); -module_param(id, charp, 0444); +module_param(id, charp, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(id, "ID string for SuperH DAC audio."); /* main struct */ -- 2.9.2

1 0

[alsa-devel] [PATCH 1265/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/sh/aica.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/sh/aica.c b/sound/sh/aica.c index ad3d9ae..e6bfee8 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -56,11 +56,11 @@ MODULE_FIRMWARE("aica_firmware.bin"); static int index = -1; static char *id; static bool enable = 1; -module_param(index, int, 0444); +module_param(index, int, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard."); -module_param(id, charp, 0444); +module_param(id, charp, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard."); -module_param(enable, bool, 0644); +module_param(enable, bool, S_IRUSR | S_IWUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard."); /* Use workqueue */ -- 2.9.2

1 0

[alsa-devel] [PATCH 1264/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/ppc/snd_ps3.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c index 3682425..594d694 100644 --- a/sound/ppc/snd_ps3.c +++ b/sound/ppc/snd_ps3.c @@ -51,15 +51,15 @@ static struct snd_ps3_card_info the_card; static int snd_ps3_start_delay = CONFIG_SND_PS3_DEFAULT_START_DELAY; -module_param_named(start_delay, snd_ps3_start_delay, uint, 0644); +module_param_named(start_delay, snd_ps3_start_delay, uint, S_IRUSR | S_IWUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(start_delay, "time to insert silent data in ms"); static int index = SNDRV_DEFAULT_IDX1; static char *id = SNDRV_DEFAULT_STR1; -module_param(index, int, 0444); +module_param(index, int, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(index, "Index value for PS3 soundchip."); -module_param(id, charp, 0444); +module_param(id, charp, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(id, "ID string for PS3 soundchip."); -- 2.9.2

1 0

[alsa-devel] [PATCH 1263/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/ppc/powermac.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/ppc/powermac.c b/sound/ppc/powermac.c index 33c6be9..d6498cd 100644 --- a/sound/ppc/powermac.c +++ b/sound/ppc/powermac.c @@ -38,11 +38,11 @@ static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */ static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ static bool enable_beep = 1; -module_param(index, int, 0444); +module_param(index, int, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(index, "Index value for " CHIP_NAME " soundchip."); -module_param(id, charp, 0444); +module_param(id, charp, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(id, "ID string for " CHIP_NAME " soundchip."); -module_param(enable_beep, bool, 0444); +module_param(enable_beep, bool, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(enable_beep, "Enable beep using PCM."); static struct platform_device *device; -- 2.9.2

1 0

[alsa-devel] [PATCH 1262/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/pcmcia/vx/vxpocket.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pcmcia/vx/vxpocket.c b/sound/pcmcia/vx/vxpocket.c index b16f42d..4a8d286 100644 --- a/sound/pcmcia/vx/vxpocket.c +++ b/sound/pcmcia/vx/vxpocket.c @@ -42,13 +42,13 @@ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable switches */ static int ibl[SNDRV_CARDS]; -module_param_array(index, int, NULL, 0444); +module_param_array(index, int, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(index, "Index value for VXPocket soundcard."); -module_param_array(id, charp, NULL, 0444); +module_param_array(id, charp, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(id, "ID string for VXPocket soundcard."); -module_param_array(enable, bool, NULL, 0444); +module_param_array(enable, bool, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(enable, "Enable VXPocket soundcard."); -module_param_array(ibl, int, NULL, 0444); +module_param_array(ibl, int, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(ibl, "Capture IBL size for VXPocket soundcard."); -- 2.9.2

1 0

[alsa-devel] [PATCH 1261/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/pcmcia/pdaudiocf/pdaudiocf.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.c b/sound/pcmcia/pdaudiocf/pdaudiocf.c index 07f4b33..a487786 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf.c @@ -41,11 +41,11 @@ static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable switches */ -module_param_array(index, int, NULL, 0444); +module_param_array(index, int, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard."); -module_param_array(id, charp, NULL, 0444); +module_param_array(id, charp, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard."); -module_param_array(enable, bool, NULL, 0444); +module_param_array(enable, bool, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard."); /* -- 2.9.2

1 0

[alsa-devel] [PATCH 1260/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/pci/ymfpci/ymfpci.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c index 812e27a..7794dc4 100644 --- a/sound/pci/ymfpci/ymfpci.c +++ b/sound/pci/ymfpci/ymfpci.c @@ -49,21 +49,21 @@ static long joystick_port[SNDRV_CARDS]; #endif static bool rear_switch[SNDRV_CARDS]; -module_param_array(index, int, NULL, 0444); +module_param_array(index, int, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(index, "Index value for the Yamaha DS-1 PCI soundcard."); -module_param_array(id, charp, NULL, 0444); +module_param_array(id, charp, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(id, "ID string for the Yamaha DS-1 PCI soundcard."); -module_param_array(enable, bool, NULL, 0444); +module_param_array(enable, bool, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(enable, "Enable Yamaha DS-1 soundcard."); -module_param_array(mpu_port, long, NULL, 0444); +module_param_array(mpu_port, long, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(mpu_port, "MPU-401 Port."); -module_param_array(fm_port, long, NULL, 0444); +module_param_array(fm_port, long, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(fm_port, "FM OPL-3 Port."); #ifdef SUPPORT_JOYSTICK -module_param_array(joystick_port, long, NULL, 0444); +module_param_array(joystick_port, long, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(joystick_port, "Joystick port address"); #endif -module_param_array(rear_switch, bool, NULL, 0444); +module_param_array(rear_switch, bool, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(rear_switch, "Enable shared rear/line-in switch"); static const struct pci_device_id snd_ymfpci_ids[] = { -- 2.9.2

1 0

[alsa-devel] [PATCH 1259/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/pci/vx222/vx222.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/pci/vx222/vx222.c b/sound/pci/vx222/vx222.c index ecbaf47..3731554 100644 --- a/sound/pci/vx222/vx222.c +++ b/sound/pci/vx222/vx222.c @@ -41,15 +41,15 @@ static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card static bool mic[SNDRV_CARDS]; /* microphone */ static int ibl[SNDRV_CARDS]; /* microphone */ -module_param_array(index, int, NULL, 0444); +module_param_array(index, int, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(index, "Index value for Digigram " CARD_NAME " soundcard."); -module_param_array(id, charp, NULL, 0444); +module_param_array(id, charp, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(id, "ID string for Digigram " CARD_NAME " soundcard."); -module_param_array(enable, bool, NULL, 0444); +module_param_array(enable, bool, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(enable, "Enable Digigram " CARD_NAME " soundcard."); -module_param_array(mic, bool, NULL, 0444); +module_param_array(mic, bool, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(mic, "Enable Microphone."); -module_param_array(ibl, int, NULL, 0444); +module_param_array(ibl, int, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(ibl, "Capture IBL size."); /* -- 2.9.2

1 0

[alsa-devel] [PATCH 1258/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/pci/via82xx_modem.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c index 99b9137..98cd658 100644 --- a/sound/pci/via82xx_modem.c +++ b/sound/pci/via82xx_modem.c @@ -58,16 +58,16 @@ static int index = -2; /* Exclude the first card */ static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ static int ac97_clock = 48000; -module_param(index, int, 0444); +module_param(index, int, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(index, "Index value for VIA 82xx bridge."); -module_param(id, charp, 0444); +module_param(id, charp, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(id, "ID string for VIA 82xx bridge."); -module_param(ac97_clock, int, 0444); +module_param(ac97_clock, int, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(ac97_clock, "AC'97 codec clock (default 48000Hz)."); /* just for backward compatibility */ static bool enable; -module_param(enable, bool, 0444); +module_param(enable, bool, S_IRUSR | S_IRGRP | S_IROTH); /* -- 2.9.2

1 0

[alsa-devel] [PATCH 1257/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/pci/via82xx.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 3dd038b..0d23ae7 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -88,30 +88,30 @@ static int dxs_support; static int dxs_init_volume = 31; static int nodelay; -module_param(index, int, 0444); +module_param(index, int, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(index, "Index value for VIA 82xx bridge."); -module_param(id, charp, 0444); +module_param(id, charp, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(id, "ID string for VIA 82xx bridge."); -module_param(mpu_port, long, 0444); +module_param(mpu_port, long, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(mpu_port, "MPU-401 port. (VT82C686x only)"); #ifdef SUPPORT_JOYSTICK -module_param(joystick, bool, 0444); +module_param(joystick, bool, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(joystick, "Enable joystick. (VT82C686x only)"); #endif -module_param(ac97_clock, int, 0444); +module_param(ac97_clock, int, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(ac97_clock, "AC'97 codec clock (default 48000Hz)."); -module_param(ac97_quirk, charp, 0444); +module_param(ac97_quirk, charp, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(ac97_quirk, "AC'97 workaround for strange hardware."); -module_param(dxs_support, int, 0444); +module_param(dxs_support, int, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(dxs_support, "Support for DXS channels (0 = auto, 1 = enable, 2 = disable, 3 = 48k only, 4 = no VRA, 5 = enable any sample rate)"); -module_param(dxs_init_volume, int, 0644); +module_param(dxs_init_volume, int, S_IRUSR | S_IWUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(dxs_init_volume, "initial DXS volume (0-31)"); -module_param(nodelay, int, 0444); +module_param(nodelay, int, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(nodelay, "Disable 500ms init delay"); /* just for backward compatibility */ static bool enable; -module_param(enable, bool, 0444); +module_param(enable, bool, S_IRUSR | S_IRGRP | S_IROTH); /* revision numbers for via686 */ -- 2.9.2

1 0

[alsa-devel] [PATCH 1256/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/pci/trident/trident.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/pci/trident/trident.c b/sound/pci/trident/trident.c index cedf13b..0d56223 100644 --- a/sound/pci/trident/trident.c +++ b/sound/pci/trident/trident.c @@ -51,15 +51,15 @@ static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card static int pcm_channels[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 32}; static int wavetable_size[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 8192}; -module_param_array(index, int, NULL, 0444); +module_param_array(index, int, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(index, "Index value for Trident 4DWave PCI soundcard."); -module_param_array(id, charp, NULL, 0444); +module_param_array(id, charp, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(id, "ID string for Trident 4DWave PCI soundcard."); -module_param_array(enable, bool, NULL, 0444); +module_param_array(enable, bool, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(enable, "Enable Trident 4DWave PCI soundcard."); -module_param_array(pcm_channels, int, NULL, 0444); +module_param_array(pcm_channels, int, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(pcm_channels, "Number of hardware channels assigned for PCM."); -module_param_array(wavetable_size, int, NULL, 0444); +module_param_array(wavetable_size, int, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(wavetable_size, "Maximum memory size in kB for wavetable synth."); static const struct pci_device_id snd_trident_ids[] = { -- 2.9.2

1 0

[alsa-devel] [PATCH 1255/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/pci/sonicvibes.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c index 1b6fad7..2072879 100644 --- a/sound/pci/sonicvibes.c +++ b/sound/pci/sonicvibes.c @@ -56,17 +56,17 @@ static bool reverb[SNDRV_CARDS]; static bool mge[SNDRV_CARDS]; static unsigned int dmaio = 0x7a00; /* DDMA i/o address */ -module_param_array(index, int, NULL, 0444); +module_param_array(index, int, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(index, "Index value for S3 SonicVibes soundcard."); -module_param_array(id, charp, NULL, 0444); +module_param_array(id, charp, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(id, "ID string for S3 SonicVibes soundcard."); -module_param_array(enable, bool, NULL, 0444); +module_param_array(enable, bool, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(enable, "Enable S3 SonicVibes soundcard."); -module_param_array(reverb, bool, NULL, 0444); +module_param_array(reverb, bool, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(reverb, "Enable reverb (SRAM is present) for S3 SonicVibes soundcard."); -module_param_array(mge, bool, NULL, 0444); +module_param_array(mge, bool, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(mge, "MIC Gain Enable for S3 SonicVibes soundcard."); -module_param(dmaio, uint, 0444); +module_param(dmaio, uint, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(dmaio, "DDMA i/o base address for S3 SonicVibes soundcard."); /* -- 2.9.2

1 0

[alsa-devel] [PATCH 1254/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/pci/sis7019.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index f3860b8..1b68640 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -43,13 +43,13 @@ static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ static bool enable = 1; static int codecs = 1; -module_param(index, int, 0444); +module_param(index, int, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(index, "Index value for SiS7019 Audio Accelerator."); -module_param(id, charp, 0444); +module_param(id, charp, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(id, "ID string for SiS7019 Audio Accelerator."); -module_param(enable, bool, 0444); +module_param(enable, bool, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(enable, "Enable SiS7019 Audio Accelerator."); -module_param(codecs, int, 0444); +module_param(codecs, int, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(codecs, "Set bit to indicate that codec number is expected to be present (default 1)"); static const struct pci_device_id snd_sis7019_ids[] = { -- 2.9.2

1 0

[alsa-devel] [PATCH 1253/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/pci/rme9652/rme9652.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index fdbc0aa..6b977c7 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -41,13 +41,13 @@ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ static bool precise_ptr[SNDRV_CARDS]; /* Enable precise pointer */ -module_param_array(index, int, NULL, 0444); +module_param_array(index, int, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(index, "Index value for RME Digi9652 (Hammerfall) soundcard."); -module_param_array(id, charp, NULL, 0444); +module_param_array(id, charp, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(id, "ID string for RME Digi9652 (Hammerfall) soundcard."); -module_param_array(enable, bool, NULL, 0444); +module_param_array(enable, bool, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(enable, "Enable/disable specific RME96{52,36} soundcards."); -module_param_array(precise_ptr, bool, NULL, 0444); +module_param_array(precise_ptr, bool, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(precise_ptr, "Enable precise pointer (doesn't work reliably)."); MODULE_AUTHOR("Paul Davis <pbd(a)op.net>, Winfried Ritsch"); MODULE_DESCRIPTION("RME Digi9652/Digi9636"); -- 2.9.2

1 0

[alsa-devel] [PATCH 1252/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/pci/rme9652/hdspm.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index a4a999a..7358fd5 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -154,13 +154,13 @@ static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;/* Enable this card */ -module_param_array(index, int, NULL, 0444); +module_param_array(index, int, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(index, "Index value for RME HDSPM interface."); -module_param_array(id, charp, NULL, 0444); +module_param_array(id, charp, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(id, "ID string for RME HDSPM interface."); -module_param_array(enable, bool, NULL, 0444); +module_param_array(enable, bool, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(enable, "Enable/disable specific HDSPM soundcards."); -- 2.9.2

1 0

[alsa-devel] [PATCH 1251/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/pci/rme9652/hdsp.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 7c8941b..337451c 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -48,11 +48,11 @@ static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ -module_param_array(index, int, NULL, 0444); +module_param_array(index, int, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(index, "Index value for RME Hammerfall DSP interface."); -module_param_array(id, charp, NULL, 0444); +module_param_array(id, charp, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(id, "ID string for RME Hammerfall DSP interface."); -module_param_array(enable, bool, NULL, 0444); +module_param_array(enable, bool, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(enable, "Enable/disable specific Hammerfall DSP soundcards."); MODULE_AUTHOR("Paul Davis <paul(a)linuxaudiosystems.com>, Marcus Andersson, Thomas Charbonnel <thomas(a)undata.org>"); MODULE_DESCRIPTION("RME Hammerfall DSP"); -- 2.9.2

1 0

[alsa-devel] [PATCH 1250/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/pci/rme96.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index 41c31db..23c3909 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -55,11 +55,11 @@ static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ -module_param_array(index, int, NULL, 0444); +module_param_array(index, int, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(index, "Index value for RME Digi96 soundcard."); -module_param_array(id, charp, NULL, 0444); +module_param_array(id, charp, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(id, "ID string for RME Digi96 soundcard."); -module_param_array(enable, bool, NULL, 0444); +module_param_array(enable, bool, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(enable, "Enable RME Digi96 soundcard."); /* -- 2.9.2

1 0

[alsa-devel] [PATCH 1249/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/pci/rme32.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c index cd94ac5..c932c09 100644 --- a/sound/pci/rme32.c +++ b/sound/pci/rme32.c @@ -91,13 +91,13 @@ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ static bool fullduplex[SNDRV_CARDS]; // = {[0 ... (SNDRV_CARDS - 1)] = 1}; -module_param_array(index, int, NULL, 0444); +module_param_array(index, int, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(index, "Index value for RME Digi32 soundcard."); -module_param_array(id, charp, NULL, 0444); +module_param_array(id, charp, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(id, "ID string for RME Digi32 soundcard."); -module_param_array(enable, bool, NULL, 0444); +module_param_array(enable, bool, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(enable, "Enable RME Digi32 soundcard."); -module_param_array(fullduplex, bool, NULL, 0444); +module_param_array(fullduplex, bool, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(fullduplex, "Support full-duplex mode."); MODULE_AUTHOR("Martin Langer <martin-langer(a)gmx.de>, Pilo Chambert <pilo.c(a)wanadoo.fr>"); MODULE_DESCRIPTION("RME Digi32, Digi32/8, Digi32 PRO"); -- 2.9.2

1 0

[alsa-devel] [PATCH 1248/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/pci/riptide/riptide.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index 94639d6..e73a45b 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -130,19 +130,19 @@ static int joystick_port[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS - 1)] = 0x200 }; static int mpu_port[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS - 1)] = 0x330 }; static int opl3_port[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS - 1)] = 0x388 }; -module_param_array(index, int, NULL, 0444); +module_param_array(index, int, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(index, "Index value for Riptide soundcard."); -module_param_array(id, charp, NULL, 0444); +module_param_array(id, charp, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(id, "ID string for Riptide soundcard."); -module_param_array(enable, bool, NULL, 0444); +module_param_array(enable, bool, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(enable, "Enable Riptide soundcard."); #ifdef SUPPORT_JOYSTICK -module_param_array(joystick_port, int, NULL, 0444); +module_param_array(joystick_port, int, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(joystick_port, "Joystick port # for Riptide soundcard."); #endif -module_param_array(mpu_port, int, NULL, 0444); +module_param_array(mpu_port, int, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(mpu_port, "MPU401 port # for Riptide driver."); -module_param_array(opl3_port, int, NULL, 0444); +module_param_array(opl3_port, int, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(opl3_port, "OPL3 port # for Riptide driver."); /* -- 2.9.2

1 0

[alsa-devel] [PATCH 1247/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/pci/pcxhr/pcxhr.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index 9293235..0f5e99c 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -55,13 +55,13 @@ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;/* Enable this card */ static bool mono[SNDRV_CARDS]; /* capture mono only */ -module_param_array(index, int, NULL, 0444); +module_param_array(index, int, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(index, "Index value for Digigram " DRIVER_NAME " soundcard"); -module_param_array(id, charp, NULL, 0444); +module_param_array(id, charp, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(id, "ID string for Digigram " DRIVER_NAME " soundcard"); -module_param_array(enable, bool, NULL, 0444); +module_param_array(enable, bool, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(enable, "Enable Digigram " DRIVER_NAME " soundcard"); -module_param_array(mono, bool, NULL, 0444); +module_param_array(mono, bool, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(mono, "Mono capture mode (default is stereo)"); enum { -- 2.9.2

1 0

[alsa-devel] [PATCH 1246/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/pci/oxygen/virtuoso.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 83de6fb..7f81d42 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -34,11 +34,11 @@ static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; -module_param_array(index, int, NULL, 0444); +module_param_array(index, int, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(index, "card index"); -module_param_array(id, charp, NULL, 0444); +module_param_array(id, charp, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(id, "ID string"); -module_param_array(enable, bool, NULL, 0444); +module_param_array(enable, bool, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(enable, "enable card"); static const struct pci_device_id xonar_ids[] = { -- 2.9.2

1 0

[alsa-devel] [PATCH 1245/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/pci/oxygen/se6x.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/oxygen/se6x.c b/sound/pci/oxygen/se6x.c index f70d514..e27bcef 100644 --- a/sound/pci/oxygen/se6x.c +++ b/sound/pci/oxygen/se6x.c @@ -45,11 +45,11 @@ static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; -module_param_array(index, int, NULL, 0444); +module_param_array(index, int, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(index, "card index"); -module_param_array(id, charp, NULL, 0444); +module_param_array(id, charp, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(id, "ID string"); -module_param_array(enable, bool, NULL, 0444); +module_param_array(enable, bool, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(enable, "enable card"); static const struct pci_device_id se6x_ids[] = { -- 2.9.2

1 0

[alsa-devel] [PATCH 1244/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/pci/oxygen/oxygen.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index 74afb6b..a33e9c3 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -76,11 +76,11 @@ static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; -module_param_array(index, int, NULL, 0444); +module_param_array(index, int, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(index, "card index"); -module_param_array(id, charp, NULL, 0444); +module_param_array(id, charp, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(id, "ID string"); -module_param_array(enable, bool, NULL, 0444); +module_param_array(enable, bool, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(enable, "enable card"); enum { -- 2.9.2

1 0

[alsa-devel] [PATCH 1243/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/pci/nm256/nm256.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c index 4735e27..6235b1e 100644 --- a/sound/pci/nm256/nm256.c +++ b/sound/pci/nm256/nm256.c @@ -64,30 +64,30 @@ static bool vaio_hack; /* disabled */ static bool reset_workaround; static bool reset_workaround_2; -module_param(index, int, 0444); +module_param(index, int, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard."); -module_param(id, charp, 0444); +module_param(id, charp, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard."); -module_param(playback_bufsize, int, 0444); +module_param(playback_bufsize, int, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(playback_bufsize, "DAC frame size in kB for " CARD_NAME " soundcard."); -module_param(capture_bufsize, int, 0444); +module_param(capture_bufsize, int, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(capture_bufsize, "ADC frame size in kB for " CARD_NAME " soundcard."); -module_param(force_ac97, bool, 0444); +module_param(force_ac97, bool, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(force_ac97, "Force to use AC97 codec for " CARD_NAME " soundcard."); -module_param(buffer_top, int, 0444); +module_param(buffer_top, int, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(buffer_top, "Set the top address of audio buffer for " CARD_NAME " soundcard."); -module_param(use_cache, bool, 0444); +module_param(use_cache, bool, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(use_cache, "Enable the cache for coefficient table access."); -module_param(vaio_hack, bool, 0444); +module_param(vaio_hack, bool, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(vaio_hack, "Enable workaround for Sony VAIO notebooks."); -module_param(reset_workaround, bool, 0444); +module_param(reset_workaround, bool, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(reset_workaround, "Enable AC97 RESET workaround for some laptops."); -module_param(reset_workaround_2, bool, 0444); +module_param(reset_workaround_2, bool, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(reset_workaround_2, "Enable extended AC97 RESET workaround for some other laptops."); /* just for backward compatibility */ static bool enable; -module_param(enable, bool, 0444); +module_param(enable, bool, S_IRUSR | S_IRGRP | S_IROTH); -- 2.9.2

1 0